diff --git a/Android.mk b/Android.mk
deleted file mode 100644
index aa90dd3..0000000
--- a/Android.mk
+++ /dev/null
@@ -1,30 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-MY_WEBRTC_ROOT_PATH := $(call my-dir)
-
-
-include $(MY_WEBRTC_ROOT_PATH)/src/system_wrappers/source/Android.mk
-
-# audio processing
-include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/resampler/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/signal_processing_library/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/vad/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aec/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aecm/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/agc/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/ns/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/utility/Android.mk
-
-
-# build .so
-include $(MY_WEBRTC_ROOT_PATH)/android-webrtc.mk
-
-# build test apps
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/main/test/process_test/Android.mk
diff --git a/MODULE_LICENSE_BSD b/MODULE_LICENSE_BSD
deleted file mode 100644
index e69de29..0000000
--- a/MODULE_LICENSE_BSD
+++ /dev/null
diff --git a/NOTICE b/NOTICE
deleted file mode 100644
index 1c39e5b..0000000
--- a/NOTICE
+++ /dev/null
@@ -1,111 +0,0 @@
-===============================================================================
-
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-===============================================================================
-
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- *
- */
-
-===============================================================================
-
-/*
- * http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
- * Copyright Takuya OOURA, 1996-2001
- *
- * You may use, copy, modify and distribute this code for any purpose (include
- * commercial use) and without fee. Please refer to this package when you modify
- * this code.
- *
- * Changes by the WebRTC authors:
- *    - Trivial type modifications.
- *    - Minimal code subset to do rdft of length 128.
- *    - Optimizations because of known length.
- *
- *  All changes are covered by the WebRTC license and IP grant:
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-===============================================================================
-
-/*
- * http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
- * Copyright Takuya OOURA, 1996-2001
- *
- * You may use, copy, modify and distribute this code for any purpose (include
- * commercial use) and without fee. Please refer to this package when you modify
- * this code.
- *
- * Changes:
- * Trivial type modifications by the WebRTC authors.
- */
-
-===============================================================================
-
-/*Boost Software License - Version 1.0 - August 17th, 2003
-
-Permission is hereby granted, free of charge, to any person or organization
-obtaining a copy of the software and accompanying documentation covered by
-this license (the "Software") to use, reproduce, display, distribute,
-execute, and transmit the Software, and to prepare derivative works of the
-Software, and to permit third-parties to whom the Software is furnished to
-do so, all subject to the following:
-
-The copyright notices in the Software and this entire statement, including
-the above license grant, this restriction and the following disclaimer,
-must be included in all copies of the Software, in whole or in part, and
-all derivative works of the Software, unless such copies or derivative
-works are solely in the form of machine-executable object code generated by
-a source language processor.
-
-THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
-IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
-FITNESS FOR A PARTICULAR PURPOSE, TITLE AND NON-INFRINGEMENT. IN NO EVENT
-SHALL THE COPYRIGHT HOLDERS OR ANYONE DISTRIBUTING THE SOFTWARE BE LIABLE
-FOR ANY DAMAGES OR OTHER LIABILITY, WHETHER IN CONTRACT, TORT OR OTHERWISE,
-ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
-DEALINGS IN THE SOFTWARE.*/
-
-===============================================================================
-
-//          Copyright Steven J. Ross 2001 - 2009.
-// Distributed under the Boost Software License, Version 1.0.
-//    (See accompanying file LICENSE_1_0.txt or copy at
-//          http://www.boost.org/LICENSE_1_0.txt)
-
-//  See http://www.boost.org/ for updates, documentation, and revision history.
-
-/*
-Some improvements suggested by:
-Phil Endecott and Frank Gennari
-Cygwin fix provided by:
-Scott McMurray
-*/
-
-
diff --git a/android-webrtc.mk b/android-webrtc.mk
deleted file mode 100644
index 01d6a9e..0000000
--- a/android-webrtc.mk
+++ /dev/null
@@ -1,96 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-MY_APM_WHOLE_STATIC_LIBRARIES := \
-    libwebrtc_spl \
-    libwebrtc_resampler \
-    libwebrtc_apm \
-    libwebrtc_apm_utility \
-    libwebrtc_vad \
-    libwebrtc_ns \
-    libwebrtc_agc \
-    libwebrtc_aec \
-    libwebrtc_aecm
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc_audio_preprocessing
-LOCAL_MODULE_TAGS := optional
-LOCAL_LDFLAGS :=
-
-LOCAL_WHOLE_STATIC_LIBRARIES := \
-    $(MY_APM_WHOLE_STATIC_LIBRARIES) \
-    libwebrtc_system_wrappers \
-
-LOCAL_SHARED_LIBRARIES := \
-    libcutils \
-    libdl \
-    libstlport
-
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_SHARED_LIBRARY)
-
-###
-
-#LOCAL_PATH := $(call my-dir)
-#
-#include $(CLEAR_VARS)
-#
-#LOCAL_ARM_MODE := arm
-#LOCAL_MODULE := libwebrtc
-#LOCAL_MODULE_TAGS := optional
-#LOCAL_LDFLAGS :=
-#
-#LOCAL_WHOLE_STATIC_LIBRARIES := \
-#    libwebrtc_system_wrappers \
-#    libwebrtc_audio_device \
-#    libwebrtc_pcm16b \
-#    libwebrtc_cng \
-#    libwebrtc_audio_coding \
-#    libwebrtc_rtp_rtcp \
-#    libwebrtc_media_file \
-#    libwebrtc_udp_transport \
-#    libwebrtc_utility \
-#    libwebrtc_neteq \
-#    libwebrtc_audio_conference_mixer \
-#    libwebrtc_isac \
-#    libwebrtc_ilbc \
-#    libwebrtc_isacfix \
-#    libwebrtc_g722 \
-#    libwebrtc_g711 \
-#    libwebrtc_voe_core \
-#    libwebrtc_video_render \
-#    libwebrtc_video_capture \
-#    libwebrtc_i420 \
-#    libwebrtc_video_coding \
-#    libwebrtc_video_processing \
-#    libwebrtc_vp8 \
-#    libwebrtc_vie_core \
-#    libwebrtc_vplib \
-#    libwebrtc_jpeg \
-#    libwebrtc_vpx
-#
-#LOCAL_STATIC_LIBRARIES :=
-#LOCAL_SHARED_LIBRARIES := \
-#    libcutils \
-#    libdl \
-#    libstlport \
-#    libjpeg \
-#    libGLESv2 \
-#    libOpenSLES \
-#    libwebrtc_audio_preprocessing
-#
-#LOCAL_ADDITIONAL_DEPENDENCIES :=
-#
-#include external/stlport/libstlport.mk
-#include $(BUILD_SHARED_LIBRARY)
diff --git a/src/common_audio/OWNERS b/src/common_audio/OWNERS
deleted file mode 100644
index 0eb967b..0000000
--- a/src/common_audio/OWNERS
+++ /dev/null
@@ -1 +0,0 @@
-bjornv@google.com
diff --git a/src/common_audio/resampler/OWNERS b/src/common_audio/resampler/OWNERS
deleted file mode 100644
index cf595df..0000000
--- a/src/common_audio/resampler/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-bjornv@google.com
-tlegrand@google.com
-jks@google.com
diff --git a/src/common_audio/resampler/main/interface/resampler.h b/src/common_audio/resampler/main/interface/resampler.h
deleted file mode 100644
index a03ff18..0000000
--- a/src/common_audio/resampler/main/interface/resampler.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * A wrapper for resampling a numerous amount of sampling combinations.
- */
-
-#ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
-#define WEBRTC_RESAMPLER_RESAMPLER_H_
-
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-enum ResamplerType
-{
-    // 4 MSB = Number of channels
-    // 4 LSB = Synchronous or asynchronous
-
-    kResamplerSynchronous = 0x10,
-    kResamplerAsynchronous = 0x11,
-    kResamplerSynchronousStereo = 0x20,
-    kResamplerAsynchronousStereo = 0x21,
-    kResamplerInvalid = 0xff
-};
-
-enum ResamplerMode
-{
-    kResamplerMode1To1,
-    kResamplerMode1To2,
-    kResamplerMode1To3,
-    kResamplerMode1To4,
-    kResamplerMode1To6,
-    kResamplerMode2To3,
-    kResamplerMode2To11,
-    kResamplerMode4To11,
-    kResamplerMode8To11,
-    kResamplerMode11To16,
-    kResamplerMode11To32,
-    kResamplerMode2To1,
-    kResamplerMode3To1,
-    kResamplerMode4To1,
-    kResamplerMode6To1,
-    kResamplerMode3To2,
-    kResamplerMode11To2,
-    kResamplerMode11To4,
-    kResamplerMode11To8
-};
-
-class Resampler
-{
-
-public:
-    Resampler();
-    Resampler(int inFreq, int outFreq, ResamplerType type);
-    ~Resampler();
-
-    // Reset all states
-    int Reset(int inFreq, int outFreq, ResamplerType type);
-
-    // Reset all states if any parameter has changed
-    int ResetIfNeeded(int inFreq, int outFreq, ResamplerType type);
-
-    // Synchronous resampling, all output samples are written to samplesOut
-    int Push(const WebRtc_Word16* samplesIn, int lengthIn, WebRtc_Word16* samplesOut,
-             int maxLen, int &outLen);
-
-    // Asynchronous resampling, input
-    int Insert(WebRtc_Word16* samplesIn, int lengthIn);
-
-    // Asynchronous resampling output, remaining samples are buffered
-    int Pull(WebRtc_Word16* samplesOut, int desiredLen, int &outLen);
-
-private:
-    // Generic pointers since we don't know what states we'll need
-    void* state1_;
-    void* state2_;
-    void* state3_;
-
-    // Storage if needed
-    WebRtc_Word16* in_buffer_;
-    WebRtc_Word16* out_buffer_;
-    int in_buffer_size_;
-    int out_buffer_size_;
-    int in_buffer_size_max_;
-    int out_buffer_size_max_;
-
-    // State
-    int my_in_frequency_khz_;
-    int my_out_frequency_khz_;
-    ResamplerMode my_mode_;
-    ResamplerType my_type_;
-
-    // Extra instance for stereo
-    Resampler* slave_left_;
-    Resampler* slave_right_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_RESAMPLER_RESAMPLER_H_
diff --git a/src/common_audio/resampler/main/source/Android.mk b/src/common_audio/resampler/main/source/Android.mk
deleted file mode 100644
index b78e3af..0000000
--- a/src/common_audio/resampler/main/source/Android.mk
+++ /dev/null
@@ -1,58 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_resampler
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := resampler.cc
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS := 
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
-    '-DNO_HEAPCHECKER' \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' 
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../.. \
-    $(LOCAL_PATH)/../interface \
-    $(LOCAL_PATH)/../../../signal_processing_library/main/interface 
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS := 
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
-    libdl \
-    libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
-LOCAL_LDLIBS += -ldl -lpthread
-endif
-
-ifneq ($(TARGET_SIMULATOR),true)
-LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/common_audio/resampler/main/source/resampler.cc b/src/common_audio/resampler/main/source/resampler.cc
deleted file mode 100644
index f866739..0000000
--- a/src/common_audio/resampler/main/source/resampler.cc
+++ /dev/null
@@ -1,981 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * A wrapper for resampling a numerous amount of sampling combinations.
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "signal_processing_library.h"
-#include "resampler.h"
-
-
-namespace webrtc
-{
-
-Resampler::Resampler()
-{
-    state1_ = NULL;
-    state2_ = NULL;
-    state3_ = NULL;
-    in_buffer_ = NULL;
-    out_buffer_ = NULL;
-    in_buffer_size_ = 0;
-    out_buffer_size_ = 0;
-    in_buffer_size_max_ = 0;
-    out_buffer_size_max_ = 0;
-    // we need a reset before we will work
-    my_in_frequency_khz_ = 0;
-    my_out_frequency_khz_ = 0;
-    my_mode_ = kResamplerMode1To1;
-    my_type_ = kResamplerInvalid;
-    slave_left_ = NULL;
-    slave_right_ = NULL;
-}
-
-Resampler::Resampler(int inFreq, int outFreq, ResamplerType type)
-{
-    state1_ = NULL;
-    state2_ = NULL;
-    state3_ = NULL;
-    in_buffer_ = NULL;
-    out_buffer_ = NULL;
-    in_buffer_size_ = 0;
-    out_buffer_size_ = 0;
-    in_buffer_size_max_ = 0;
-    out_buffer_size_max_ = 0;
-    // we need a reset before we will work
-    my_in_frequency_khz_ = 0;
-    my_out_frequency_khz_ = 0;
-    my_mode_ = kResamplerMode1To1;
-    my_type_ = kResamplerInvalid;
-    slave_left_ = NULL;
-    slave_right_ = NULL;
-
-    int res = Reset(inFreq, outFreq, type);
-
-}
-
-Resampler::~Resampler()
-{
-    if (state1_)
-    {
-        free(state1_);
-    }
-    if (state2_)
-    {
-        free(state2_);
-    }
-    if (state3_)
-    {
-        free(state3_);
-    }
-    if (in_buffer_)
-    {
-        free(in_buffer_);
-    }
-    if (out_buffer_)
-    {
-        free(out_buffer_);
-    }
-    if (slave_left_)
-    {
-        delete slave_left_;
-    }
-    if (slave_right_)
-    {
-        delete slave_right_;
-    }
-}
-
-int Resampler::ResetIfNeeded(int inFreq, int outFreq, ResamplerType type)
-{
-    int tmpInFreq_kHz = inFreq / 1000;
-    int tmpOutFreq_kHz = outFreq / 1000;
-
-    if ((tmpInFreq_kHz != my_in_frequency_khz_) || (tmpOutFreq_kHz != my_out_frequency_khz_)
-            || (type != my_type_))
-    {
-        return Reset(inFreq, outFreq, type);
-    } else
-    {
-        return 0;
-    }
-}
-
-int Resampler::Reset(int inFreq, int outFreq, ResamplerType type)
-{
-
-    if (state1_)
-    {
-        free(state1_);
-        state1_ = NULL;
-    }
-    if (state2_)
-    {
-        free(state2_);
-        state2_ = NULL;
-    }
-    if (state3_)
-    {
-        free(state3_);
-        state3_ = NULL;
-    }
-    if (in_buffer_)
-    {
-        free(in_buffer_);
-        in_buffer_ = NULL;
-    }
-    if (out_buffer_)
-    {
-        free(out_buffer_);
-        out_buffer_ = NULL;
-    }
-    if (slave_left_)
-    {
-        delete slave_left_;
-        slave_left_ = NULL;
-    }
-    if (slave_right_)
-    {
-        delete slave_right_;
-        slave_right_ = NULL;
-    }
-
-    in_buffer_size_ = 0;
-    out_buffer_size_ = 0;
-    in_buffer_size_max_ = 0;
-    out_buffer_size_max_ = 0;
-
-    // This might be overridden if parameters are not accepted.
-    my_type_ = type;
-
-    // Start with a math exercise, Euclid's algorithm to find the gcd:
-
-    int a = inFreq;
-    int b = outFreq;
-    int c = a % b;
-    while (c != 0)
-    {
-        a = b;
-        b = c;
-        c = a % b;
-    }
-    // b is now the gcd;
-
-    // We need to track what domain we're in.
-    my_in_frequency_khz_ = inFreq / 1000;
-    my_out_frequency_khz_ = outFreq / 1000;
-
-    // Scale with GCD
-    inFreq = inFreq / b;
-    outFreq = outFreq / b;
-
-    // Do we need stereo?
-    if ((my_type_ & 0xf0) == 0x20)
-    {
-        // Change type to mono
-        type = (ResamplerType)((int)type & 0x0f + 0x10);
-        slave_left_ = new Resampler(inFreq, outFreq, type);
-        slave_right_ = new Resampler(inFreq, outFreq, type);
-    }
-
-    if (inFreq == outFreq)
-    {
-        my_mode_ = kResamplerMode1To1;
-    } else if (inFreq == 1)
-    {
-        switch (outFreq)
-        {
-            case 2:
-                my_mode_ = kResamplerMode1To2;
-                break;
-            case 3:
-                my_mode_ = kResamplerMode1To3;
-                break;
-            case 4:
-                my_mode_ = kResamplerMode1To4;
-                break;
-            case 6:
-                my_mode_ = kResamplerMode1To6;
-                break;
-            default:
-                my_type_ = kResamplerInvalid;
-                break;
-        }
-    } else if (outFreq == 1)
-    {
-        switch (inFreq)
-        {
-            case 2:
-                my_mode_ = kResamplerMode2To1;
-                break;
-            case 3:
-                my_mode_ = kResamplerMode3To1;
-                break;
-            case 4:
-                my_mode_ = kResamplerMode4To1;
-                break;
-            case 6:
-                my_mode_ = kResamplerMode6To1;
-                break;
-            default:
-                my_type_ = kResamplerInvalid;
-                break;
-        }
-    } else if ((inFreq == 2) && (outFreq == 3))
-    {
-        my_mode_ = kResamplerMode2To3;
-    } else if ((inFreq == 2) && (outFreq == 11))
-    {
-        my_mode_ = kResamplerMode2To11;
-    } else if ((inFreq == 4) && (outFreq == 11))
-    {
-        my_mode_ = kResamplerMode4To11;
-    } else if ((inFreq == 8) && (outFreq == 11))
-    {
-        my_mode_ = kResamplerMode8To11;
-    } else if ((inFreq == 3) && (outFreq == 2))
-    {
-        my_mode_ = kResamplerMode3To2;
-    } else if ((inFreq == 11) && (outFreq == 2))
-    {
-        my_mode_ = kResamplerMode11To2;
-    } else if ((inFreq == 11) && (outFreq == 4))
-    {
-        my_mode_ = kResamplerMode11To4;
-    } else if ((inFreq == 11) && (outFreq == 16))
-    {
-        my_mode_ = kResamplerMode11To16;
-    } else if ((inFreq == 11) && (outFreq == 32))
-    {
-        my_mode_ = kResamplerMode11To32;
-    } else if ((inFreq == 11) && (outFreq == 8))
-    {
-        my_mode_ = kResamplerMode11To8;
-    } else
-    {
-        my_type_ = kResamplerInvalid;
-        return -1;
-    }
-
-    // Now create the states we need
-    switch (my_mode_)
-    {
-        case kResamplerMode1To1:
-            // No state needed;
-            break;
-        case kResamplerMode1To2:
-            state1_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-            break;
-        case kResamplerMode1To3:
-            state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
-            WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state1_);
-            break;
-        case kResamplerMode1To4:
-            // 1:2
-            state1_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-            // 2:4
-            state2_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
-            break;
-        case kResamplerMode1To6:
-            // 1:2
-            state1_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-            // 2:6
-            state2_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
-            WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state2_);
-            break;
-        case kResamplerMode2To3:
-            // 2:6
-            state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
-            WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state1_);
-            // 6:3
-            state2_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
-            break;
-        case kResamplerMode2To11:
-            state1_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-
-            state2_ = malloc(sizeof(WebRtcSpl_State8khzTo22khz));
-            WebRtcSpl_ResetResample8khzTo22khz((WebRtcSpl_State8khzTo22khz *)state2_);
-            break;
-        case kResamplerMode4To11:
-            state1_ = malloc(sizeof(WebRtcSpl_State8khzTo22khz));
-            WebRtcSpl_ResetResample8khzTo22khz((WebRtcSpl_State8khzTo22khz *)state1_);
-            break;
-        case kResamplerMode8To11:
-            state1_ = malloc(sizeof(WebRtcSpl_State16khzTo22khz));
-            WebRtcSpl_ResetResample16khzTo22khz((WebRtcSpl_State16khzTo22khz *)state1_);
-            break;
-        case kResamplerMode11To16:
-            state1_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-
-            state2_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
-            WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state2_);
-            break;
-        case kResamplerMode11To32:
-            // 11 -> 22
-            state1_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-
-            // 22 -> 16
-            state2_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
-            WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state2_);
-
-            // 16 -> 32
-            state3_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state3_, 0, 8 * sizeof(WebRtc_Word32));
-
-            break;
-        case kResamplerMode2To1:
-            state1_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-            break;
-        case kResamplerMode3To1:
-            state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
-            WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state1_);
-            break;
-        case kResamplerMode4To1:
-            // 4:2
-            state1_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-            // 2:1
-            state2_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
-            break;
-        case kResamplerMode6To1:
-            // 6:2
-            state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
-            WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state1_);
-            // 2:1
-            state2_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
-            break;
-        case kResamplerMode3To2:
-            // 3:6
-            state1_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-            // 6:2
-            state2_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
-            WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state2_);
-            break;
-        case kResamplerMode11To2:
-            state1_ = malloc(sizeof(WebRtcSpl_State22khzTo8khz));
-            WebRtcSpl_ResetResample22khzTo8khz((WebRtcSpl_State22khzTo8khz *)state1_);
-
-            state2_ = malloc(8 * sizeof(WebRtc_Word32));
-            memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
-
-            break;
-        case kResamplerMode11To4:
-            state1_ = malloc(sizeof(WebRtcSpl_State22khzTo8khz));
-            WebRtcSpl_ResetResample22khzTo8khz((WebRtcSpl_State22khzTo8khz *)state1_);
-            break;
-        case kResamplerMode11To8:
-            state1_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
-            WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state1_);
-            break;
-
-    }
-
-    return 0;
-}
-
-// Synchronous resampling, all output samples are written to samplesOut
-int Resampler::Push(const WebRtc_Word16 * samplesIn, int lengthIn, WebRtc_Word16* samplesOut,
-                    int maxLen, int &outLen)
-{
-    // Check that the resampler is not in asynchronous mode
-    if (my_type_ & 0x0f)
-    {
-        return -1;
-    }
-
-    // Do we have a stereo signal?
-    if ((my_type_ & 0xf0) == 0x20)
-    {
-
-        // Split up the signal and call the slave object for each channel
-
-        WebRtc_Word16* left = (WebRtc_Word16*)malloc(lengthIn * sizeof(WebRtc_Word16) / 2);
-        WebRtc_Word16* right = (WebRtc_Word16*)malloc(lengthIn * sizeof(WebRtc_Word16) / 2);
-        WebRtc_Word16* out_left = (WebRtc_Word16*)malloc(maxLen / 2 * sizeof(WebRtc_Word16));
-        WebRtc_Word16* out_right =
-                (WebRtc_Word16*)malloc(maxLen / 2 * sizeof(WebRtc_Word16));
-        int res = 0;
-        for (int i = 0; i < lengthIn; i += 2)
-        {
-            left[i >> 1] = samplesIn[i];
-            right[i >> 1] = samplesIn[i + 1];
-        }
-
-        // It's OK to overwrite the local parameter, since it's just a copy
-        lengthIn = lengthIn / 2;
-
-        int actualOutLen_left = 0;
-        int actualOutLen_right = 0;
-        // Do resampling for right channel
-        res |= slave_left_->Push(left, lengthIn, out_left, maxLen / 2, actualOutLen_left);
-        res |= slave_right_->Push(right, lengthIn, out_right, maxLen / 2, actualOutLen_right);
-        if (res || (actualOutLen_left != actualOutLen_right))
-        {
-            free(left);
-            free(right);
-            free(out_left);
-            free(out_right);
-            return -1;
-        }
-
-        // Reassemble the signal
-        for (int i = 0; i < actualOutLen_left; i++)
-        {
-            samplesOut[i * 2] = out_left[i];
-            samplesOut[i * 2 + 1] = out_right[i];
-        }
-        outLen = 2 * actualOutLen_left;
-
-        free(left);
-        free(right);
-        free(out_left);
-        free(out_right);
-
-        return 0;
-    }
-
-    // Container for temp samples
-    WebRtc_Word16* tmp;
-    // tmp data for resampling routines
-    WebRtc_Word32* tmp_mem;
-
-    switch (my_mode_)
-    {
-        case kResamplerMode1To1:
-            memcpy(samplesOut, samplesIn, lengthIn * sizeof(WebRtc_Word16));
-            outLen = lengthIn;
-            break;
-        case kResamplerMode1To2:
-            if (maxLen < (lengthIn * 2))
-            {
-                return -1;
-            }
-            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
-            outLen = lengthIn * 2;
-            return 0;
-        case kResamplerMode1To3:
-
-            // We can only handle blocks of 160 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 160) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < (lengthIn * 3))
-            {
-                return -1;
-            }
-            tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
-
-            for (int i = 0; i < lengthIn; i += 160)
-            {
-                WebRtcSpl_Resample16khzTo48khz(samplesIn + i, samplesOut + i * 3,
-                                               (WebRtcSpl_State16khzTo48khz *)state1_,
-                                               tmp_mem);
-            }
-            outLen = lengthIn * 3;
-            free(tmp_mem);
-            return 0;
-        case kResamplerMode1To4:
-            if (maxLen < (lengthIn * 4))
-            {
-                return -1;
-            }
-
-            tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
-            // 1:2
-            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
-            // 2:4
-            WebRtcSpl_UpsampleBy2(tmp, lengthIn * 2, samplesOut, (WebRtc_Word32*)state2_);
-            outLen = lengthIn * 4;
-            free(tmp);
-            return 0;
-        case kResamplerMode1To6:
-            // We can only handle blocks of 80 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 80) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < (lengthIn * 6))
-            {
-                return -1;
-            }
-
-            //1:2
-
-            tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
-            tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
-
-            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
-            outLen = lengthIn * 2;
-
-            for (int i = 0; i < outLen; i += 160)
-            {
-                WebRtcSpl_Resample16khzTo48khz(tmp + i, samplesOut + i * 3,
-                                               (WebRtcSpl_State16khzTo48khz *)state2_,
-                                               tmp_mem);
-            }
-            outLen = outLen * 3;
-            free(tmp_mem);
-            free(tmp);
-
-            return 0;
-        case kResamplerMode2To3:
-            if (maxLen < (lengthIn * 3 / 2))
-            {
-                return -1;
-            }
-            // 2:6
-            // We can only handle blocks of 160 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 160) != 0)
-            {
-                return -1;
-            }
-            tmp = static_cast<WebRtc_Word16*> (malloc(sizeof(WebRtc_Word16) * lengthIn * 3));
-            tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
-            for (int i = 0; i < lengthIn; i += 160)
-            {
-                WebRtcSpl_Resample16khzTo48khz(samplesIn + i, tmp + i * 3,
-                                               (WebRtcSpl_State16khzTo48khz *)state1_,
-                                               tmp_mem);
-            }
-            lengthIn = lengthIn * 3;
-            // 6:3
-            WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut, (WebRtc_Word32*)state2_);
-            outLen = lengthIn / 2;
-            free(tmp);
-            free(tmp_mem);
-            return 0;
-        case kResamplerMode2To11:
-
-            // We can only handle blocks of 80 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 80) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < ((lengthIn * 11) / 2))
-            {
-                return -1;
-            }
-            tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
-            // 1:2
-            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
-            lengthIn *= 2;
-
-            tmp_mem = (WebRtc_Word32*)malloc(98 * sizeof(WebRtc_Word32));
-
-            for (int i = 0; i < lengthIn; i += 80)
-            {
-                WebRtcSpl_Resample8khzTo22khz(tmp + i, samplesOut + (i * 11) / 4,
-                                              (WebRtcSpl_State8khzTo22khz *)state2_,
-                                              tmp_mem);
-            }
-            outLen = (lengthIn * 11) / 4;
-            free(tmp_mem);
-            free(tmp);
-            return 0;
-        case kResamplerMode4To11:
-
-            // We can only handle blocks of 80 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 80) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < ((lengthIn * 11) / 4))
-            {
-                return -1;
-            }
-            tmp_mem = (WebRtc_Word32*)malloc(98 * sizeof(WebRtc_Word32));
-
-            for (int i = 0; i < lengthIn; i += 80)
-            {
-                WebRtcSpl_Resample8khzTo22khz(samplesIn + i, samplesOut + (i * 11) / 4,
-                                              (WebRtcSpl_State8khzTo22khz *)state1_,
-                                              tmp_mem);
-            }
-            outLen = (lengthIn * 11) / 4;
-            free(tmp_mem);
-            return 0;
-        case kResamplerMode8To11:
-            // We can only handle blocks of 160 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 160) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < ((lengthIn * 11) / 8))
-            {
-                return -1;
-            }
-            tmp_mem = (WebRtc_Word32*)malloc(88 * sizeof(WebRtc_Word32));
-
-            for (int i = 0; i < lengthIn; i += 160)
-            {
-                WebRtcSpl_Resample16khzTo22khz(samplesIn + i, samplesOut + (i * 11) / 8,
-                                               (WebRtcSpl_State16khzTo22khz *)state1_,
-                                               tmp_mem);
-            }
-            outLen = (lengthIn * 11) / 8;
-            free(tmp_mem);
-            return 0;
-
-        case kResamplerMode11To16:
-            // We can only handle blocks of 110 samples
-            if ((lengthIn % 110) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < ((lengthIn * 16) / 11))
-            {
-                return -1;
-            }
-
-            tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
-            tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn * 2));
-
-            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
-
-            for (int i = 0; i < (lengthIn * 2); i += 220)
-            {
-                WebRtcSpl_Resample22khzTo16khz(tmp + i, samplesOut + (i / 220) * 160,
-                                               (WebRtcSpl_State22khzTo16khz *)state2_,
-                                               tmp_mem);
-            }
-
-            outLen = (lengthIn * 16) / 11;
-
-            free(tmp_mem);
-            free(tmp);
-            return 0;
-
-        case kResamplerMode11To32:
-
-            // We can only handle blocks of 110 samples
-            if ((lengthIn % 110) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < ((lengthIn * 32) / 11))
-            {
-                return -1;
-            }
-
-            tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
-            tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn * 2));
-
-            // 11 -> 22 kHz in samplesOut
-            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
-
-            // 22 -> 16 in tmp
-            for (int i = 0; i < (lengthIn * 2); i += 220)
-            {
-                WebRtcSpl_Resample22khzTo16khz(samplesOut + i, tmp + (i / 220) * 160,
-                                               (WebRtcSpl_State22khzTo16khz *)state2_,
-                                               tmp_mem);
-            }
-
-            // 16 -> 32 in samplesOut
-            WebRtcSpl_UpsampleBy2(tmp, (lengthIn * 16) / 11, samplesOut,
-                                  (WebRtc_Word32*)state3_);
-
-            outLen = (lengthIn * 32) / 11;
-
-            free(tmp_mem);
-            free(tmp);
-            return 0;
-
-        case kResamplerMode2To1:
-            if (maxLen < (lengthIn / 2))
-            {
-                return -1;
-            }
-            WebRtcSpl_DownsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
-            outLen = lengthIn / 2;
-            return 0;
-        case kResamplerMode3To1:
-            // We can only handle blocks of 480 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 480) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < (lengthIn / 3))
-            {
-                return -1;
-            }
-            tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
-
-            for (int i = 0; i < lengthIn; i += 480)
-            {
-                WebRtcSpl_Resample48khzTo16khz(samplesIn + i, samplesOut + i / 3,
-                                               (WebRtcSpl_State48khzTo16khz *)state1_,
-                                               tmp_mem);
-            }
-            outLen = lengthIn / 3;
-            free(tmp_mem);
-            return 0;
-        case kResamplerMode4To1:
-            if (maxLen < (lengthIn / 4))
-            {
-                return -1;
-            }
-            tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * lengthIn / 2);
-            // 4:2
-            WebRtcSpl_DownsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
-            // 2:1
-            WebRtcSpl_DownsampleBy2(tmp, lengthIn / 2, samplesOut, (WebRtc_Word32*)state2_);
-            outLen = lengthIn / 4;
-            free(tmp);
-            return 0;
-
-        case kResamplerMode6To1:
-            // We can only handle blocks of 480 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 480) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < (lengthIn / 6))
-            {
-                return -1;
-            }
-
-            tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
-            tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn) / 3);
-
-            for (int i = 0; i < lengthIn; i += 480)
-            {
-                WebRtcSpl_Resample48khzTo16khz(samplesIn + i, tmp + i / 3,
-                                               (WebRtcSpl_State48khzTo16khz *)state1_,
-                                               tmp_mem);
-            }
-            outLen = lengthIn / 3;
-            free(tmp_mem);
-            WebRtcSpl_DownsampleBy2(tmp, outLen, samplesOut, (WebRtc_Word32*)state2_);
-            free(tmp);
-            outLen = outLen / 2;
-            return 0;
-        case kResamplerMode3To2:
-            if (maxLen < (lengthIn * 2 / 3))
-            {
-                return -1;
-            }
-            // 3:6
-            tmp = static_cast<WebRtc_Word16*> (malloc(sizeof(WebRtc_Word16) * lengthIn * 2));
-            WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
-            lengthIn *= 2;
-            // 6:2
-            // We can only handle blocks of 480 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 480) != 0)
-            {
-                free(tmp);
-                return -1;
-            }
-            tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
-            for (int i = 0; i < lengthIn; i += 480)
-            {
-                WebRtcSpl_Resample48khzTo16khz(tmp + i, samplesOut + i / 3,
-                                               (WebRtcSpl_State48khzTo16khz *)state2_,
-                                               tmp_mem);
-            }
-            outLen = lengthIn / 3;
-            free(tmp);
-            free(tmp_mem);
-            return 0;
-        case kResamplerMode11To2:
-            // We can only handle blocks of 220 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 220) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < ((lengthIn * 2) / 11))
-            {
-                return -1;
-            }
-            tmp_mem = (WebRtc_Word32*)malloc(126 * sizeof(WebRtc_Word32));
-            tmp = (WebRtc_Word16*)malloc((lengthIn * 4) / 11 * sizeof(WebRtc_Word16));
-
-            for (int i = 0; i < lengthIn; i += 220)
-            {
-                WebRtcSpl_Resample22khzTo8khz(samplesIn + i, tmp + (i * 4) / 11,
-                                              (WebRtcSpl_State22khzTo8khz *)state1_,
-                                              tmp_mem);
-            }
-            lengthIn = (lengthIn * 4) / 11;
-
-            WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut, (WebRtc_Word32*)state2_);
-            outLen = lengthIn / 2;
-
-            free(tmp_mem);
-            free(tmp);
-            return 0;
-        case kResamplerMode11To4:
-            // We can only handle blocks of 220 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 220) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < ((lengthIn * 4) / 11))
-            {
-                return -1;
-            }
-            tmp_mem = (WebRtc_Word32*)malloc(126 * sizeof(WebRtc_Word32));
-
-            for (int i = 0; i < lengthIn; i += 220)
-            {
-                WebRtcSpl_Resample22khzTo8khz(samplesIn + i, samplesOut + (i * 4) / 11,
-                                              (WebRtcSpl_State22khzTo8khz *)state1_,
-                                              tmp_mem);
-            }
-            outLen = (lengthIn * 4) / 11;
-            free(tmp_mem);
-            return 0;
-        case kResamplerMode11To8:
-            // We can only handle blocks of 160 samples
-            // Can be fixed, but I don't think it's needed
-            if ((lengthIn % 220) != 0)
-            {
-                return -1;
-            }
-            if (maxLen < ((lengthIn * 8) / 11))
-            {
-                return -1;
-            }
-            tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
-
-            for (int i = 0; i < lengthIn; i += 220)
-            {
-                WebRtcSpl_Resample22khzTo16khz(samplesIn + i, samplesOut + (i * 8) / 11,
-                                               (WebRtcSpl_State22khzTo16khz *)state1_,
-                                               tmp_mem);
-            }
-            outLen = (lengthIn * 8) / 11;
-            free(tmp_mem);
-            return 0;
-            break;
-
-    }
-    return 0;
-}
-
-// Asynchronous resampling, input
-int Resampler::Insert(WebRtc_Word16 * samplesIn, int lengthIn)
-{
-    if (my_type_ != kResamplerAsynchronous)
-    {
-        return -1;
-    }
-    int sizeNeeded, tenMsblock;
-
-    // Determine need for size of outBuffer
-    sizeNeeded = out_buffer_size_ + ((lengthIn + in_buffer_size_) * my_out_frequency_khz_)
-            / my_in_frequency_khz_;
-    if (sizeNeeded > out_buffer_size_max_)
-    {
-        // Round the value upwards to complete 10 ms blocks
-        tenMsblock = my_out_frequency_khz_ * 10;
-        sizeNeeded = (sizeNeeded / tenMsblock + 1) * tenMsblock;
-        out_buffer_ = (WebRtc_Word16*)realloc(out_buffer_, sizeNeeded * sizeof(WebRtc_Word16));
-        out_buffer_size_max_ = sizeNeeded;
-    }
-
-    // If we need to use inBuffer, make sure all input data fits there.
-
-    tenMsblock = my_in_frequency_khz_ * 10;
-    if (in_buffer_size_ || (lengthIn % tenMsblock))
-    {
-        // Check if input buffer size is enough
-        if ((in_buffer_size_ + lengthIn) > in_buffer_size_max_)
-        {
-            // Round the value upwards to complete 10 ms blocks
-            sizeNeeded = ((in_buffer_size_ + lengthIn) / tenMsblock + 1) * tenMsblock;
-            in_buffer_ = (WebRtc_Word16*)realloc(in_buffer_,
-                                                 sizeNeeded * sizeof(WebRtc_Word16));
-            in_buffer_size_max_ = sizeNeeded;
-        }
-        // Copy in data to input buffer
-        memcpy(in_buffer_ + in_buffer_size_, samplesIn, lengthIn * sizeof(WebRtc_Word16));
-
-        // Resample all available 10 ms blocks
-        int lenOut;
-        int dataLenToResample = (in_buffer_size_ / tenMsblock) * tenMsblock;
-        Push(in_buffer_, dataLenToResample, out_buffer_ + out_buffer_size_,
-             out_buffer_size_max_ - out_buffer_size_, lenOut);
-        out_buffer_size_ += lenOut;
-
-        // Save the rest
-        memmove(in_buffer_, in_buffer_ + dataLenToResample,
-                (in_buffer_size_ - dataLenToResample) * sizeof(WebRtc_Word16));
-        in_buffer_size_ -= dataLenToResample;
-    } else
-    {
-        // Just resample
-        int lenOut;
-        Push(in_buffer_, lengthIn, out_buffer_ + out_buffer_size_,
-             out_buffer_size_max_ - out_buffer_size_, lenOut);
-        out_buffer_size_ += lenOut;
-    }
-
-    return 0;
-}
-
-// Asynchronous resampling output, remaining samples are buffered
-int Resampler::Pull(WebRtc_Word16* samplesOut, int desiredLen, int &outLen)
-{
-    if (my_type_ != kResamplerAsynchronous)
-    {
-        return -1;
-    }
-
-    // Check that we have enough data
-    if (desiredLen <= out_buffer_size_)
-    {
-        // Give out the date
-        memcpy(samplesOut, out_buffer_, desiredLen * sizeof(WebRtc_Word32));
-
-        // Shuffle down remaining
-        memmove(out_buffer_, out_buffer_ + desiredLen,
-                (out_buffer_size_ - desiredLen) * sizeof(WebRtc_Word16));
-
-        // Update remaining size
-        out_buffer_size_ -= desiredLen;
-
-        return 0;
-    } else
-    {
-        return -1;
-    }
-}
-
-} // namespace webrtc
diff --git a/src/common_audio/resampler/main/source/resampler.gyp b/src/common_audio/resampler/main/source/resampler.gyp
deleted file mode 100644
index 8baf870..0000000
--- a/src/common_audio/resampler/main/source/resampler.gyp
+++ /dev/null
@@ -1,40 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../../../common_settings.gypi', # Common settings
-  ],
-  'targets': [
-    {
-      'target_name': 'resampler',
-      'type': '<(library)',
-      'dependencies': [
-        '../../../signal_processing_library/main/source/spl.gyp:spl',
-      ],
-      'include_dirs': [
-        '../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-        ],
-      },
-      'sources': [
-        '../interface/resampler.h',
-        'resampler.cc',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/common_audio/signal_processing_library/OWNERS b/src/common_audio/signal_processing_library/OWNERS
deleted file mode 100644
index cf595df..0000000
--- a/src/common_audio/signal_processing_library/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-bjornv@google.com
-tlegrand@google.com
-jks@google.com
diff --git a/src/common_audio/signal_processing_library/main/interface/signal_processing_library.h b/src/common_audio/signal_processing_library/main/interface/signal_processing_library.h
deleted file mode 100644
index 414e045..0000000
--- a/src/common_audio/signal_processing_library/main/interface/signal_processing_library.h
+++ /dev/null
@@ -1,1771 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes all of the fix point signal processing library (SPL) function
- * descriptions and declarations.
- * For specific function calls, see bottom of file.
- */
-
-#ifndef WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
-#define WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
-
-#include <string.h>
-#include "typedefs.h"
-
-#ifdef ARM_WINM
-#include <Armintr.h> // intrinsic file for windows mobile
-#endif
-
-#ifdef WEBRTC_ANDROID
-#define WEBRTC_SPL_INLINE_CALLS
-#define SPL_NO_DOUBLE_IMPLEMENTATIONS
-#endif
-
-// Macros specific for the fixed point implementation
-#define WEBRTC_SPL_WORD16_MAX       32767
-#define WEBRTC_SPL_WORD16_MIN       -32768
-#define WEBRTC_SPL_WORD32_MAX       (WebRtc_Word32)0x7fffffff
-#define WEBRTC_SPL_WORD32_MIN       (WebRtc_Word32)0x80000000
-#define WEBRTC_SPL_MAX_LPC_ORDER    14
-#define WEBRTC_SPL_MAX_SEED_USED    0x80000000L
-#define WEBRTC_SPL_MIN(A, B)        (A < B ? A : B) // Get min value
-#define WEBRTC_SPL_MAX(A, B)        (A > B ? A : B) // Get max value
-#define WEBRTC_SPL_ABS_W16(a)\
-    (((WebRtc_Word16)a >= 0) ? ((WebRtc_Word16)a) : -((WebRtc_Word16)a))
-#define WEBRTC_SPL_ABS_W32(a)\
-    (((WebRtc_Word32)a >= 0) ? ((WebRtc_Word32)a) : -((WebRtc_Word32)a))
-
-#if (defined WEBRTC_TARGET_PC)||(defined __TARGET_XSCALE)
-#define WEBRTC_SPL_GET_BYTE(a, nr)  (((WebRtc_Word8 *)a)[nr])
-#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index)  \
-  (((WebRtc_Word8 *)d_ptr)[index] = (val))
-#elif defined WEBRTC_BIG_ENDIAN
-#define WEBRTC_SPL_GET_BYTE(a, nr)\
-    ((((WebRtc_Word16 *)a)[nr >> 1]) >> (((nr + 1) & 0x1) * 8) & 0x00ff)
-#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index)                          \
-  ((WebRtc_Word16 *)d_ptr)[index >> 1] = \
-      ((((WebRtc_Word16 *)d_ptr)[index >> 1])                           \
-       & (0x00ff << (8 * ((index) & 0x1)))) | (val << (8 * ((index + 1) & 0x1)))
-#else
-#define WEBRTC_SPL_GET_BYTE(a,nr)                                       \
-  ((((WebRtc_Word16 *)(a))[(nr) >> 1]) >> (((nr) & 0x1) * 8) & 0x00ff)
-#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index)                          \
-  ((WebRtc_Word16 *)(d_ptr))[(index) >> 1] = \
-      ((((WebRtc_Word16 *)(d_ptr))[(index) >> 1])                       \
-       & (0x00ff << (8 * (((index) + 1) & 0x1)))) |                     \
-      ((val) << (8 * ((index) & 0x1)))
-#endif
-
-#ifndef WEBRTC_ANDROID
-#define WEBRTC_SPL_MUL(a, b)                                    \
-  ((WebRtc_Word32) ((WebRtc_Word32)(a) * (WebRtc_Word32)(b)))
-#endif
-
-#define WEBRTC_SPL_UMUL(a, b)                                           \
-  ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)))
-#define WEBRTC_SPL_UMUL_RSFT16(a, b)\
-    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)) >> 16)
-#define WEBRTC_SPL_UMUL_16_16(a, b)\
-    ((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b))
-#define WEBRTC_SPL_UMUL_16_16_RSFT16(a, b)\
-    (((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b)) >> 16)
-#define WEBRTC_SPL_UMUL_32_16(a, b)\
-    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)))
-#define WEBRTC_SPL_UMUL_32_16_RSFT16(a, b)\
-    ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)) >> 16)
-#define WEBRTC_SPL_MUL_16_U16(a, b)\
-    ((WebRtc_Word32)(WebRtc_Word16)(a) * (WebRtc_UWord16)(b))
-#define WEBRTC_SPL_DIV(a, b)    \
-  ((WebRtc_Word32) ((WebRtc_Word32)(a) / (WebRtc_Word32)(b)))
-#define WEBRTC_SPL_UDIV(a, b)   \
-  ((WebRtc_UWord32) ((WebRtc_UWord32)(a) / (WebRtc_UWord32)(b)))
-
-#define WEBRTC_SPL_MUL_16_32_RSFT11(a, b)\
-  ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 5)                            \
-     + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x0200) >> 10))
-#define WEBRTC_SPL_MUL_16_32_RSFT14(a, b)\
-  ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 2)                            \
-   + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x1000) >> 13))
-#define WEBRTC_SPL_MUL_16_32_RSFT15(a, b)                               \
-  ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 1)                            \
-   + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x2000) >> 14))
-
-#ifndef WEBRTC_ANDROID
-#define WEBRTC_SPL_MUL_16_32_RSFT16(a, b)                               \
-  (WEBRTC_SPL_MUL_16_16(a, b >> 16)                                     \
-   + ((WEBRTC_SPL_MUL_16_16(a, (b & 0xffff) >> 1) + 0x4000) >> 15))
-#define WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, b32)                    \
-  ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(a32a, b32)               \
-                   + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32) >> 16)))
-#define WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, b32)                         \
-  ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16((                        \
-      (WebRtc_Word16)(a32 >> 16)), b32) +                               \
-                   (WEBRTC_SPL_MUL_16_32_RSFT16((                       \
-                       (WebRtc_Word16)((a32 & 0x0000FFFF) >> 1)), b32) >> 15)))
-#endif
-
-#ifdef ARM_WINM
-#define WEBRTC_SPL_MUL_16_16(a, b)                      \
-  _SmulLo_SW_SL((WebRtc_Word16)(a), (WebRtc_Word16)(b))
-#elif !defined (WEBRTC_ANDROID)
-#define WEBRTC_SPL_MUL_16_16(a, b)                                      \
-    ((WebRtc_Word32) (((WebRtc_Word16)(a)) * ((WebRtc_Word16)(b))))
-#endif
-
-#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c)      \
-  (WEBRTC_SPL_MUL_16_16(a, b) >> (c))
-
-#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c)                   \
-  ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) \
-                                  (((WebRtc_Word32)1) << ((c) - 1)))) >> (c))
-#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b)\
-    ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) (1 << 14))) >> 15)
-
-// C + the 32 most significant bits of A * B
-#define WEBRTC_SPL_SCALEDIFF32(A, B, C)                                 \
-  (C + (B >> 16) * A + (((WebRtc_UWord32)(0x0000FFFF & B) * A) >> 16))
-
-#define WEBRTC_SPL_ADD_SAT_W32(a, b)    WebRtcSpl_AddSatW32(a, b)
-#define WEBRTC_SPL_SAT(a, b, c)         (b > a ? a : b < c ? c : b)
-#define WEBRTC_SPL_MUL_32_16(a, b)      ((a) * (b))
-
-#define WEBRTC_SPL_SUB_SAT_W32(a, b)    WebRtcSpl_SubSatW32(a, b)
-#define WEBRTC_SPL_ADD_SAT_W16(a, b)    WebRtcSpl_AddSatW16(a, b)
-#define WEBRTC_SPL_SUB_SAT_W16(a, b)    WebRtcSpl_SubSatW16(a, b)
-
-// We cannot do casting here due to signed/unsigned problem
-#define WEBRTC_SPL_IS_NEG(a)            ((a) & 0x80000000)
-// Shifting with negative numbers allowed
-// Positive means left shift
-#define WEBRTC_SPL_SHIFT_W16(x, c)              \
-  (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
-#define WEBRTC_SPL_SHIFT_W32(x, c)              \
-  (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
-
-// Shifting with negative numbers not allowed
-// We cannot do casting here due to signed/unsigned problem
-#define WEBRTC_SPL_RSHIFT_W16(x, c)     ((x) >> (c))
-#define WEBRTC_SPL_LSHIFT_W16(x, c)     ((x) << (c))
-#define WEBRTC_SPL_RSHIFT_W32(x, c)     ((x) >> (c))
-#define WEBRTC_SPL_LSHIFT_W32(x, c)     ((x) << (c))
-
-#define WEBRTC_SPL_RSHIFT_U16(x, c)     ((WebRtc_UWord16)(x) >> (c))
-#define WEBRTC_SPL_LSHIFT_U16(x, c)     ((WebRtc_UWord16)(x) << (c))
-#define WEBRTC_SPL_RSHIFT_U32(x, c)     ((WebRtc_UWord32)(x) >> (c))
-#define WEBRTC_SPL_LSHIFT_U32(x, c)     ((WebRtc_UWord32)(x) << (c))
-
-#define WEBRTC_SPL_VNEW(t, n)           (t *) malloc (sizeof (t) * (n))
-#define WEBRTC_SPL_FREE                 free
-
-#define WEBRTC_SPL_RAND(a)\
-  ((WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT((a), 18816, 7) & 0x00007fff))
-
-#ifdef __cplusplus
-extern "C"
-{
-#endif
-
-#define WEBRTC_SPL_MEMCPY_W8(v1, v2, length)    \
-  memcpy(v1, v2, (length) * sizeof(char))
-#define WEBRTC_SPL_MEMCPY_W16(v1, v2, length)           \
-  memcpy(v1, v2, (length) * sizeof(WebRtc_Word16))
-
-#define WEBRTC_SPL_MEMMOVE_W16(v1, v2, length)          \
-  memmove(v1, v2, (length) * sizeof(WebRtc_Word16))
-
-// Trigonometric tables used for quick lookup
-// default declarations
-extern WebRtc_Word16 WebRtcSpl_kCosTable[];
-extern WebRtc_Word16 WebRtcSpl_kSinTable[];
-extern WebRtc_Word16 WebRtcSpl_kSinTable1024[];
-// Hanning table
-extern WebRtc_Word16 WebRtcSpl_kHanningTable[];
-// Random table
-extern WebRtc_Word16 WebRtcSpl_kRandNTable[];
-
-#ifndef WEBRTC_SPL_INLINE_CALLS
-WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 var1, WebRtc_Word16 var2);
-WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1, WebRtc_Word16 var2);
-WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 var1, WebRtc_Word32 var2);
-WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 var1, WebRtc_Word32 var2);
-WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 value);
-int WebRtcSpl_NormW32(WebRtc_Word32 value);
-int WebRtcSpl_NormW16(WebRtc_Word16 value);
-int WebRtcSpl_NormU32(WebRtc_UWord32 value);
-#else
-#include "spl_inl.h"
-#endif
-
-// Get SPL Version
-WebRtc_Word16 WebRtcSpl_get_version(char* version,
-                                    WebRtc_Word16 length_in_bytes);
-
-int WebRtcSpl_GetScalingSquare(WebRtc_Word16* in_vector,
-                               int in_vector_length,
-                               int times);
-
-// Copy and set operations. Implementation in copy_set_operations.c.
-// Descriptions at bottom of file.
-void WebRtcSpl_MemSetW16(WebRtc_Word16* vector,
-                         WebRtc_Word16 set_value,
-                         int vector_length);
-void WebRtcSpl_MemSetW32(WebRtc_Word32* vector,
-                         WebRtc_Word32 set_value,
-                         int vector_length);
-void WebRtcSpl_MemCpyReversedOrder(WebRtc_Word16* out_vector,
-                                   WebRtc_Word16* in_vector,
-                                   int vector_length);
-WebRtc_Word16 WebRtcSpl_CopyFromEndW16(G_CONST WebRtc_Word16* in_vector,
-                                       WebRtc_Word16 in_vector_length,
-                                       WebRtc_Word16 samples,
-                                       WebRtc_Word16* out_vector);
-WebRtc_Word16 WebRtcSpl_ZerosArrayW16(WebRtc_Word16* vector,
-                                      WebRtc_Word16 vector_length);
-WebRtc_Word16 WebRtcSpl_ZerosArrayW32(WebRtc_Word32* vector,
-                                      WebRtc_Word16 vector_length);
-WebRtc_Word16 WebRtcSpl_OnesArrayW16(WebRtc_Word16* vector,
-                                     WebRtc_Word16 vector_length);
-WebRtc_Word16 WebRtcSpl_OnesArrayW32(WebRtc_Word32* vector,
-                                     WebRtc_Word16 vector_length);
-// End: Copy and set operations.
-
-// Minimum and maximum operations. Implementation in min_max_operations.c.
-// Descriptions at bottom of file.
-WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(G_CONST WebRtc_Word16* vector,
-                                       WebRtc_Word16 length);
-WebRtc_Word32 WebRtcSpl_MaxAbsValueW32(G_CONST WebRtc_Word32* vector,
-                                       WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MinValueW16(G_CONST WebRtc_Word16* vector,
-                                    WebRtc_Word16 length);
-WebRtc_Word32 WebRtcSpl_MinValueW32(G_CONST WebRtc_Word32* vector,
-                                    WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MaxValueW16(G_CONST WebRtc_Word16* vector,
-                                    WebRtc_Word16 length);
-
-WebRtc_Word16 WebRtcSpl_MaxAbsIndexW16(G_CONST WebRtc_Word16* vector,
-                                       WebRtc_Word16 length);
-WebRtc_Word32 WebRtcSpl_MaxValueW32(G_CONST WebRtc_Word32* vector,
-                                    WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MinIndexW16(G_CONST WebRtc_Word16* vector,
-                                    WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MinIndexW32(G_CONST WebRtc_Word32* vector,
-                                    WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MaxIndexW16(G_CONST WebRtc_Word16* vector,
-                                    WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MaxIndexW32(G_CONST WebRtc_Word32* vector,
-                                    WebRtc_Word16 length);
-// End: Minimum and maximum operations.
-
-// Vector scaling operations. Implementation in vector_scaling_operations.c.
-// Description at bottom of file.
-void WebRtcSpl_VectorBitShiftW16(WebRtc_Word16* out_vector,
-                                 WebRtc_Word16 vector_length,
-                                 G_CONST WebRtc_Word16* in_vector,
-                                 WebRtc_Word16 right_shifts);
-void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32* out_vector,
-                                 WebRtc_Word16 vector_length,
-                                 G_CONST WebRtc_Word32* in_vector,
-                                 WebRtc_Word16 right_shifts);
-void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16* out_vector,
-                                      WebRtc_Word16 vector_length,
-                                      G_CONST WebRtc_Word32* in_vector,
-                                      WebRtc_Word16 right_shifts);
-
-void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16* in_vector,
-                           WebRtc_Word16* out_vector,
-                           WebRtc_Word16 gain,
-                           WebRtc_Word16 vector_length,
-                           WebRtc_Word16 right_shifts);
-void WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16* in_vector,
-                                  WebRtc_Word16* out_vector,
-                                  WebRtc_Word16 gain,
-                                  WebRtc_Word16 vector_length,
-                                  WebRtc_Word16 right_shifts);
-void WebRtcSpl_ScaleAndAddVectors(G_CONST WebRtc_Word16* in_vector1,
-                                  WebRtc_Word16 gain1, int right_shifts1,
-                                  G_CONST WebRtc_Word16* in_vector2,
-                                  WebRtc_Word16 gain2, int right_shifts2,
-                                  WebRtc_Word16* out_vector,
-                                  int vector_length);
-// End: Vector scaling operations.
-
-// iLBC specific functions. Implementations in ilbc_specific_functions.c.
-// Description at bottom of file.
-void WebRtcSpl_ScaleAndAddVectorsWithRound(WebRtc_Word16* in_vector1,
-                                           WebRtc_Word16 scale1,
-                                           WebRtc_Word16* in_vector2,
-                                           WebRtc_Word16 scale2,
-                                           WebRtc_Word16 right_shifts,
-                                           WebRtc_Word16* out_vector,
-                                           WebRtc_Word16 vector_length);
-void WebRtcSpl_ReverseOrderMultArrayElements(WebRtc_Word16* out_vector,
-                                             G_CONST WebRtc_Word16* in_vector,
-                                             G_CONST WebRtc_Word16* window,
-                                             WebRtc_Word16 vector_length,
-                                             WebRtc_Word16 right_shifts);
-void WebRtcSpl_ElementwiseVectorMult(WebRtc_Word16* out_vector,
-                                     G_CONST WebRtc_Word16* in_vector,
-                                     G_CONST WebRtc_Word16* window,
-                                     WebRtc_Word16 vector_length,
-                                     WebRtc_Word16 right_shifts);
-void WebRtcSpl_AddVectorsAndShift(WebRtc_Word16* out_vector,
-                                  G_CONST WebRtc_Word16* in_vector1,
-                                  G_CONST WebRtc_Word16* in_vector2,
-                                  WebRtc_Word16 vector_length,
-                                  WebRtc_Word16 right_shifts);
-void WebRtcSpl_AddAffineVectorToVector(WebRtc_Word16* out_vector,
-                                       WebRtc_Word16* in_vector,
-                                       WebRtc_Word16 gain,
-                                       WebRtc_Word32 add_constant,
-                                       WebRtc_Word16 right_shifts,
-                                       int vector_length);
-void WebRtcSpl_AffineTransformVector(WebRtc_Word16* out_vector,
-                                     WebRtc_Word16* in_vector,
-                                     WebRtc_Word16 gain,
-                                     WebRtc_Word32 add_constant,
-                                     WebRtc_Word16 right_shifts,
-                                     int vector_length);
-// End: iLBC specific functions.
-
-// Signal processing operations. Descriptions at bottom of this file.
-int WebRtcSpl_AutoCorrelation(G_CONST WebRtc_Word16* vector,
-                              int vector_length, int order,
-                              WebRtc_Word32* result_vector,
-                              int* scale);
-WebRtc_Word16 WebRtcSpl_LevinsonDurbin(WebRtc_Word32* auto_corr,
-                                       WebRtc_Word16* lpc_coef,
-                                       WebRtc_Word16* refl_coef,
-                                       WebRtc_Word16 order);
-void WebRtcSpl_ReflCoefToLpc(G_CONST WebRtc_Word16* refl_coef,
-                             int use_order,
-                             WebRtc_Word16* lpc_coef);
-void WebRtcSpl_LpcToReflCoef(WebRtc_Word16* lpc_coef,
-                             int use_order,
-                             WebRtc_Word16* refl_coef);
-void WebRtcSpl_AutoCorrToReflCoef(G_CONST WebRtc_Word32* auto_corr,
-                                  int use_order,
-                                  WebRtc_Word16* refl_coef);
-void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_corr,
-                                WebRtc_Word16* vector1,
-                                WebRtc_Word16* vector2,
-                                WebRtc_Word16 dim_vector,
-                                WebRtc_Word16 dim_cross_corr,
-                                WebRtc_Word16 right_shifts,
-                                WebRtc_Word16 step_vector2);
-void WebRtcSpl_GetHanningWindow(WebRtc_Word16* window, WebRtc_Word16 size);
-void WebRtcSpl_SqrtOfOneMinusXSquared(WebRtc_Word16* in_vector,
-                                      int vector_length,
-                                      WebRtc_Word16* out_vector);
-// End: Signal processing operations.
-
-// Randomization functions. Implementations collected in randomization_functions.c and
-// descriptions at bottom of this file.
-WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32* seed);
-WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32* seed);
-WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32* seed);
-WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
-                                   WebRtc_Word16 vector_length,
-                                   WebRtc_UWord32* seed);
-// End: Randomization functions.
-
-// Math functions
-WebRtc_Word32 WebRtcSpl_Sqrt(WebRtc_Word32 value);
-WebRtc_Word32 WebRtcSpl_SqrtFloor(WebRtc_Word32 value);
-
-// Divisions. Implementations collected in division_operations.c and
-// descriptions at bottom of this file.
-WebRtc_UWord32 WebRtcSpl_DivU32U16(WebRtc_UWord32 num, WebRtc_UWord16 den);
-WebRtc_Word32 WebRtcSpl_DivW32W16(WebRtc_Word32 num, WebRtc_Word16 den);
-WebRtc_Word16 WebRtcSpl_DivW32W16ResW16(WebRtc_Word32 num, WebRtc_Word16 den);
-WebRtc_Word32 WebRtcSpl_DivResultInQ31(WebRtc_Word32 num, WebRtc_Word32 den);
-WebRtc_Word32 WebRtcSpl_DivW32HiLow(WebRtc_Word32 num, WebRtc_Word16 den_hi,
-                                    WebRtc_Word16 den_low);
-// End: Divisions.
-
-WebRtc_Word32 WebRtcSpl_Energy(WebRtc_Word16* vector,
-                               int vector_length,
-                               int* scale_factor);
-
-WebRtc_Word32 WebRtcSpl_DotProductWithScale(WebRtc_Word16* vector1,
-                                            WebRtc_Word16* vector2,
-                                            int vector_length,
-                                            int scaling);
-
-// Filter operations.
-int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* ar_coef, int ar_coef_length,
-                       G_CONST WebRtc_Word16* in_vector, int in_vector_length,
-                       WebRtc_Word16* filter_state, int filter_state_length,
-                       WebRtc_Word16* filter_state_low,
-                       int filter_state_low_length, WebRtc_Word16* out_vector,
-                       WebRtc_Word16* out_vector_low, int out_vector_low_length);
-
-void WebRtcSpl_FilterMAFastQ12(WebRtc_Word16* in_vector,
-                               WebRtc_Word16* out_vector,
-                               WebRtc_Word16* ma_coef,
-                               WebRtc_Word16 ma_coef_length,
-                               WebRtc_Word16 vector_length);
-void WebRtcSpl_FilterARFastQ12(WebRtc_Word16* in_vector,
-                               WebRtc_Word16* out_vector,
-                               WebRtc_Word16* ar_coef,
-                               WebRtc_Word16 ar_coef_length,
-                               WebRtc_Word16 vector_length);
-int WebRtcSpl_DownsampleFast(WebRtc_Word16* in_vector,
-                             WebRtc_Word16 in_vector_length,
-                             WebRtc_Word16* out_vector,
-                             WebRtc_Word16 out_vector_length,
-                             WebRtc_Word16* ma_coef,
-                             WebRtc_Word16 ma_coef_length,
-                             WebRtc_Word16 factor,
-                             WebRtc_Word16 delay);
-// End: Filter operations.
-
-// FFT operations
-int WebRtcSpl_ComplexFFT(WebRtc_Word16 vector[], int stages, int mode);
-int WebRtcSpl_ComplexIFFT(WebRtc_Word16 vector[], int stages, int mode);
-#if (defined ARM9E_GCC) || (defined ARM_WINM) || (defined ANDROID_AECOPT)
-int WebRtcSpl_ComplexFFT2(WebRtc_Word16 in_vector[],
-                          WebRtc_Word16 out_vector[],
-                          int stages, int mode);
-int WebRtcSpl_ComplexIFFT2(WebRtc_Word16 in_vector[],
-                           WebRtc_Word16 out_vector[],
-                           int stages, int mode);
-#endif
-void WebRtcSpl_ComplexBitReverse(WebRtc_Word16 vector[], int stages);
-// End: FFT operations
-
-/************************************************************
- *
- * RESAMPLING FUNCTIONS AND THEIR STRUCTS ARE DEFINED BELOW
- *
- ************************************************************/
-
-/*******************************************************************
- * resample.c
- *
- * Includes the following resampling combinations
- * 22 kHz -> 16 kHz
- * 16 kHz -> 22 kHz
- * 22 kHz ->  8 kHz
- *  8 kHz -> 22 kHz
- *
- ******************************************************************/
-
-// state structure for 22 -> 16 resampler
-typedef struct
-{
-    WebRtc_Word32 S_22_44[8];
-    WebRtc_Word32 S_44_32[8];
-    WebRtc_Word32 S_32_16[8];
-} WebRtcSpl_State22khzTo16khz;
-
-void WebRtcSpl_Resample22khzTo16khz(const WebRtc_Word16* in,
-                                    WebRtc_Word16* out,
-                                    WebRtcSpl_State22khzTo16khz* state,
-                                    WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state);
-
-// state structure for 16 -> 22 resampler
-typedef struct
-{
-    WebRtc_Word32 S_16_32[8];
-    WebRtc_Word32 S_32_22[8];
-} WebRtcSpl_State16khzTo22khz;
-
-void WebRtcSpl_Resample16khzTo22khz(const WebRtc_Word16* in,
-                                    WebRtc_Word16* out,
-                                    WebRtcSpl_State16khzTo22khz* state,
-                                    WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state);
-
-// state structure for 22 -> 8 resampler
-typedef struct
-{
-    WebRtc_Word32 S_22_22[16];
-    WebRtc_Word32 S_22_16[8];
-    WebRtc_Word32 S_16_8[8];
-} WebRtcSpl_State22khzTo8khz;
-
-void WebRtcSpl_Resample22khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                   WebRtcSpl_State22khzTo8khz* state,
-                                   WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state);
-
-// state structure for 8 -> 22 resampler
-typedef struct
-{
-    WebRtc_Word32 S_8_16[8];
-    WebRtc_Word32 S_16_11[8];
-    WebRtc_Word32 S_11_22[8];
-} WebRtcSpl_State8khzTo22khz;
-
-void WebRtcSpl_Resample8khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                   WebRtcSpl_State8khzTo22khz* state,
-                                   WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state);
-
-/*******************************************************************
- * resample_fractional.c
- * Functions for internal use in the other resample functions
- *
- * Includes the following resampling combinations
- * 48 kHz -> 32 kHz
- * 32 kHz -> 24 kHz
- * 44 kHz -> 32 kHz
- *
- ******************************************************************/
-
-void WebRtcSpl_Resample48khzTo32khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
-                                    const WebRtc_Word32 K);
-
-void WebRtcSpl_Resample32khzTo24khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
-                                    const WebRtc_Word32 K);
-
-void WebRtcSpl_Resample44khzTo32khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
-                                    const WebRtc_Word32 K);
-
-/*******************************************************************
- * resample_48khz.c
- *
- * Includes the following resampling combinations
- * 48 kHz -> 16 kHz
- * 16 kHz -> 48 kHz
- * 48 kHz ->  8 kHz
- *  8 kHz -> 48 kHz
- *
- ******************************************************************/
-
-typedef struct
-{
-    WebRtc_Word32 S_48_48[16];
-    WebRtc_Word32 S_48_32[8];
-    WebRtc_Word32 S_32_16[8];
-} WebRtcSpl_State48khzTo16khz;
-
-void WebRtcSpl_Resample48khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                    WebRtcSpl_State48khzTo16khz* state,
-                                    WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state);
-
-typedef struct
-{
-    WebRtc_Word32 S_16_32[8];
-    WebRtc_Word32 S_32_24[8];
-    WebRtc_Word32 S_24_48[8];
-} WebRtcSpl_State16khzTo48khz;
-
-void WebRtcSpl_Resample16khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                    WebRtcSpl_State16khzTo48khz* state,
-                                    WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state);
-
-typedef struct
-{
-    WebRtc_Word32 S_48_24[8];
-    WebRtc_Word32 S_24_24[16];
-    WebRtc_Word32 S_24_16[8];
-    WebRtc_Word32 S_16_8[8];
-} WebRtcSpl_State48khzTo8khz;
-
-void WebRtcSpl_Resample48khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                   WebRtcSpl_State48khzTo8khz* state,
-                                   WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state);
-
-typedef struct
-{
-    WebRtc_Word32 S_8_16[8];
-    WebRtc_Word32 S_16_12[8];
-    WebRtc_Word32 S_12_24[8];
-    WebRtc_Word32 S_24_48[8];
-} WebRtcSpl_State8khzTo48khz;
-
-void WebRtcSpl_Resample8khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                   WebRtcSpl_State8khzTo48khz* state,
-                                   WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state);
-
-/*******************************************************************
- * resample_by_2.c
- *
- * Includes down and up sampling by a factor of two.
- *
- ******************************************************************/
-
-void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
-                             WebRtc_Word16* out, WebRtc_Word32* filtState);
-
-void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len, WebRtc_Word16* out,
-                           WebRtc_Word32* filtState);
-
-/************************************************************
- * END OF RESAMPLING FUNCTIONS
- ************************************************************/
-void WebRtcSpl_AnalysisQMF(const WebRtc_Word16* in_data,
-                           WebRtc_Word16* low_band,
-                           WebRtc_Word16* high_band,
-                           WebRtc_Word32* filter_state1,
-                           WebRtc_Word32* filter_state2);
-void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band,
-                            const WebRtc_Word16* high_band,
-                            WebRtc_Word16* out_data,
-                            WebRtc_Word32* filter_state1,
-                            WebRtc_Word32* filter_state2);
-
-#ifdef __cplusplus
-}
-#endif // __cplusplus
-#endif // WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
-
-//
-// WebRtcSpl_AddSatW16(...)
-// WebRtcSpl_AddSatW32(...)
-//
-// Returns the result of a saturated 16-bit, respectively 32-bit, addition of
-// the numbers specified by the |var1| and |var2| parameters.
-//
-// Input:
-//      - var1      : Input variable 1
-//      - var2      : Input variable 2
-//
-// Return value     : Added and saturated value
-//
-
-//
-// WebRtcSpl_SubSatW16(...)
-// WebRtcSpl_SubSatW32(...)
-//
-// Returns the result of a saturated 16-bit, respectively 32-bit, subtraction
-// of the numbers specified by the |var1| and |var2| parameters.
-//
-// Input:
-//      - var1      : Input variable 1
-//      - var2      : Input variable 2
-//
-// Returned value   : Subtracted and saturated value
-//
-
-//
-// WebRtcSpl_GetSizeInBits(...)
-//
-// Returns the # of bits that are needed at the most to represent the number
-// specified by the |value| parameter.
-//
-// Input:
-//      - value     : Input value
-//
-// Return value     : Number of bits needed to represent |value|
-//
-
-//
-// WebRtcSpl_NormW32(...)
-//
-// Norm returns the # of left shifts required to 32-bit normalize the 32-bit
-// signed number specified by the |value| parameter.
-//
-// Input:
-//      - value     : Input value
-//
-// Return value     : Number of bit shifts needed to 32-bit normalize |value|
-//
-
-//
-// WebRtcSpl_NormW16(...)
-//
-// Norm returns the # of left shifts required to 16-bit normalize the 16-bit
-// signed number specified by the |value| parameter.
-//
-// Input:
-//      - value     : Input value
-//
-// Return value     : Number of bit shifts needed to 32-bit normalize |value|
-//
-
-//
-// WebRtcSpl_NormU32(...)
-//
-// Norm returns the # of left shifts required to 32-bit normalize the unsigned
-// 32-bit number specified by the |value| parameter.
-//
-// Input:
-//      - value     : Input value
-//
-// Return value     : Number of bit shifts needed to 32-bit normalize |value|
-//
-
-//
-// WebRtcSpl_GetScalingSquare(...)
-//
-// Returns the # of bits required to scale the samples specified in the
-// |in_vector| parameter so that, if the squares of the samples are added the
-// # of times specified by the |times| parameter, the 32-bit addition will not
-// overflow (result in WebRtc_Word32).
-//
-// Input:
-//      - in_vector         : Input vector to check scaling on
-//      - in_vector_length  : Samples in |in_vector|
-//      - times             : Number of additions to be performed
-//
-// Return value             : Number of right bit shifts needed to avoid
-//                            overflow in the addition calculation
-//
-
-//
-// WebRtcSpl_MemSetW16(...)
-//
-// Sets all the values in the WebRtc_Word16 vector |vector| of length
-// |vector_length| to the specified value |set_value|
-//
-// Input:
-//      - vector        : Pointer to the WebRtc_Word16 vector
-//      - set_value     : Value specified
-//      - vector_length : Length of vector
-//
-
-//
-// WebRtcSpl_MemSetW32(...)
-//
-// Sets all the values in the WebRtc_Word32 vector |vector| of length
-// |vector_length| to the specified value |set_value|
-//
-// Input:
-//      - vector        : Pointer to the WebRtc_Word16 vector
-//      - set_value     : Value specified
-//      - vector_length : Length of vector
-//
-
-//
-// WebRtcSpl_MemCpyReversedOrder(...)
-//
-// Copies all the values from the source WebRtc_Word16 vector |in_vector| to a
-// destination WebRtc_Word16 vector |out_vector|. It is done in reversed order,
-// meaning that the first sample of |in_vector| is copied to the last sample of
-// the |out_vector|. The procedure continues until the last sample of
-// |in_vector| has been copied to the first sample of |out_vector|. This
-// creates a reversed vector. Used in e.g. prediction in iLBC.
-//
-// Input:
-//      - in_vector     : Pointer to the first sample in a WebRtc_Word16 vector
-//                        of length |length|
-//      - vector_length : Number of elements to copy
-//
-// Output:
-//      - out_vector    : Pointer to the last sample in a WebRtc_Word16 vector
-//                        of length |length|
-//
-
-//
-// WebRtcSpl_CopyFromEndW16(...)
-//
-// Copies the rightmost |samples| of |in_vector| (of length |in_vector_length|)
-// to the vector |out_vector|.
-//
-// Input:
-//      - in_vector         : Input vector
-//      - in_vector_length  : Number of samples in |in_vector|
-//      - samples           : Number of samples to extract (from right side)
-//                            from |in_vector|
-//
-// Output:
-//      - out_vector        : Vector with the requested samples
-//
-// Return value             : Number of copied samples in |out_vector|
-//
-
-//
-// WebRtcSpl_ZerosArrayW16(...)
-// WebRtcSpl_ZerosArrayW32(...)
-//
-// Inserts the value "zero" in all positions of a w16 and a w32 vector
-// respectively.
-//
-// Input:
-//      - vector_length : Number of samples in vector
-//
-// Output:
-//      - vector        : Vector containing all zeros
-//
-// Return value         : Number of samples in vector
-//
-
-//
-// WebRtcSpl_OnesArrayW16(...)
-// WebRtcSpl_OnesArrayW32(...)
-//
-// Inserts the value "one" in all positions of a w16 and a w32 vector
-// respectively.
-//
-// Input:
-//      - vector_length : Number of samples in vector
-//
-// Output:
-//      - vector        : Vector containing all ones
-//
-// Return value         : Number of samples in vector
-//
-
-//
-// WebRtcSpl_MinValueW16(...)
-// WebRtcSpl_MinValueW32(...)
-//
-// Returns the minimum value of a vector
-//
-// Input:
-//      - vector        : Input vector
-//      - vector_length : Number of samples in vector
-//
-// Return value         : Minimum sample value in vector
-//
-
-//
-// WebRtcSpl_MaxValueW16(...)
-// WebRtcSpl_MaxValueW32(...)
-//
-// Returns the maximum value of a vector
-//
-// Input:
-//      - vector        : Input vector
-//      - vector_length : Number of samples in vector
-//
-// Return value         : Maximum sample value in vector
-//
-
-//
-// WebRtcSpl_MaxAbsValueW16(...)
-// WebRtcSpl_MaxAbsValueW32(...)
-//
-// Returns the largest absolute value of a vector
-//
-// Input:
-//      - vector        : Input vector
-//      - vector_length : Number of samples in vector
-//
-// Return value         : Maximum absolute value in vector
-//
-
-//
-// WebRtcSpl_MaxAbsIndexW16(...)
-//
-// Returns the vector index to the largest absolute value of a vector
-//
-// Input:
-//      - vector        : Input vector
-//      - vector_length : Number of samples in vector
-//
-// Return value         : Index to maximum absolute value in vector
-//
-
-//
-// WebRtcSpl_MinIndexW16(...)
-// WebRtcSpl_MinIndexW32(...)
-//
-// Returns the vector index to the minimum sample value of a vector
-//
-// Input:
-//      - vector        : Input vector
-//      - vector_length : Number of samples in vector
-//
-// Return value         : Index to minimum sample value in vector
-//
-
-//
-// WebRtcSpl_MaxIndexW16(...)
-// WebRtcSpl_MaxIndexW32(...)
-//
-// Returns the vector index to the maximum sample value of a vector
-//
-// Input:
-//      - vector        : Input vector
-//      - vector_length : Number of samples in vector
-//
-// Return value         : Index to maximum sample value in vector
-//
-
-//
-// WebRtcSpl_VectorBitShiftW16(...)
-// WebRtcSpl_VectorBitShiftW32(...)
-//
-// Bit shifts all the values in a vector up or downwards. Different calls for
-// WebRtc_Word16 and WebRtc_Word32 vectors respectively.
-//
-// Input:
-//      - vector_length : Length of vector
-//      - in_vector     : Pointer to the vector that should be bit shifted
-//      - right_shifts  : Number of right bit shifts (negative value gives left
-//                        shifts)
-//
-// Output:
-//      - out_vector    : Pointer to the result vector (can be the same as
-//                        |in_vector|)
-//
-
-//
-// WebRtcSpl_VectorBitShiftW32ToW16(...)
-//
-// Bit shifts all the values in a WebRtc_Word32 vector up or downwards and
-// stores the result as a WebRtc_Word16 vector
-//
-// Input:
-//      - vector_length : Length of vector
-//      - in_vector     : Pointer to the vector that should be bit shifted
-//      - right_shifts  : Number of right bit shifts (negative value gives left
-//                        shifts)
-//
-// Output:
-//      - out_vector    : Pointer to the result vector (can be the same as
-//                        |in_vector|)
-//
-
-//
-// WebRtcSpl_ScaleVector(...)
-//
-// Performs the vector operation:
-//  out_vector[k] = (gain*in_vector[k])>>right_shifts
-//
-// Input:
-//      - in_vector     : Input vector
-//      - gain          : Scaling gain
-//      - vector_length : Elements in the |in_vector|
-//      - right_shifts  : Number of right bit shifts applied
-//
-// Output:
-//      - out_vector    : Output vector (can be the same as |in_vector|)
-//
-
-//
-// WebRtcSpl_ScaleVectorWithSat(...)
-//
-// Performs the vector operation:
-//  out_vector[k] = SATURATE( (gain*in_vector[k])>>right_shifts )
-//
-// Input:
-//      - in_vector     : Input vector
-//      - gain          : Scaling gain
-//      - vector_length : Elements in the |in_vector|
-//      - right_shifts  : Number of right bit shifts applied
-//
-// Output:
-//      - out_vector    : Output vector (can be the same as |in_vector|)
-//
-
-//
-// WebRtcSpl_ScaleAndAddVectors(...)
-//
-// Performs the vector operation:
-//  out_vector[k] = (gain1*in_vector1[k])>>right_shifts1
-//                  + (gain2*in_vector2[k])>>right_shifts2
-//
-// Input:
-//      - in_vector1    : Input vector 1
-//      - gain1         : Gain to be used for vector 1
-//      - right_shifts1 : Right bit shift to be used for vector 1
-//      - in_vector2    : Input vector 2
-//      - gain2         : Gain to be used for vector 2
-//      - right_shifts2 : Right bit shift to be used for vector 2
-//      - vector_length : Elements in the input vectors
-//
-// Output:
-//      - out_vector    : Output vector
-//
-
-//
-// WebRtcSpl_ScaleAndAddVectorsWithRound(...)
-//
-// Performs the vector operation:
-//
-//  out_vector[k] = ((scale1*in_vector1[k]) + (scale2*in_vector2[k])
-//                      + round_value) >> right_shifts
-//
-//      where:
-//
-//  round_value = (1<<right_shifts)>>1
-//
-// Input:
-//      - in_vector1    : Input vector 1
-//      - scale1        : Gain to be used for vector 1
-//      - in_vector2    : Input vector 2
-//      - scale2        : Gain to be used for vector 2
-//      - right_shifts  : Number of right bit shifts to be applied
-//      - vector_length : Number of elements in the input vectors
-//
-// Output:
-//      - out_vector    : Output vector
-//
-
-//
-// WebRtcSpl_ReverseOrderMultArrayElements(...)
-//
-// Performs the vector operation:
-//  out_vector[n] = (in_vector[n]*window[-n])>>right_shifts
-//
-// Input:
-//      - in_vector     : Input vector
-//      - window        : Window vector (should be reversed). The pointer
-//                        should be set to the last value in the vector
-//      - right_shifts  : Number of right bit shift to be applied after the
-//                        multiplication
-//      - vector_length : Number of elements in |in_vector|
-//
-// Output:
-//      - out_vector    : Output vector (can be same as |in_vector|)
-//
-
-//
-// WebRtcSpl_ElementwiseVectorMult(...)
-//
-// Performs the vector operation:
-//  out_vector[n] = (in_vector[n]*window[n])>>right_shifts
-//
-// Input:
-//      - in_vector     : Input vector
-//      - window        : Window vector.
-//      - right_shifts  : Number of right bit shift to be applied after the
-//                        multiplication
-//      - vector_length : Number of elements in |in_vector|
-//
-// Output:
-//      - out_vector    : Output vector (can be same as |in_vector|)
-//
-
-//
-// WebRtcSpl_AddVectorsAndShift(...)
-//
-// Performs the vector operation:
-//  out_vector[k] = (in_vector1[k] + in_vector2[k])>>right_shifts
-//
-// Input:
-//      - in_vector1    : Input vector 1
-//      - in_vector2    : Input vector 2
-//      - right_shifts  : Number of right bit shift to be applied after the
-//                        multiplication
-//      - vector_length : Number of elements in |in_vector1| and |in_vector2|
-//
-// Output:
-//      - out_vector    : Output vector (can be same as |in_vector1|)
-//
-
-//
-// WebRtcSpl_AddAffineVectorToVector(...)
-//
-// Adds an affine transformed vector to another vector |out_vector|, i.e,
-// performs
-//  out_vector[k] += (in_vector[k]*gain+add_constant)>>right_shifts
-//
-// Input:
-//      - in_vector     : Input vector
-//      - gain          : Gain value, used to multiply the in vector with
-//      - add_constant  : Constant value to add (usually 1<<(right_shifts-1),
-//                        but others can be used as well
-//      - right_shifts  : Number of right bit shifts (0-16)
-//      - vector_length : Number of samples in |in_vector| and |out_vector|
-//
-// Output:
-//      - out_vector    : Vector with the output
-//
-
-//
-// WebRtcSpl_AffineTransformVector(...)
-//
-// Affine transforms a vector, i.e, performs
-//  out_vector[k] = (in_vector[k]*gain+add_constant)>>right_shifts
-//
-// Input:
-//      - in_vector     : Input vector
-//      - gain          : Gain value, used to multiply the in vector with
-//      - add_constant  : Constant value to add (usually 1<<(right_shifts-1),
-//                        but others can be used as well
-//      - right_shifts  : Number of right bit shifts (0-16)
-//      - vector_length : Number of samples in |in_vector| and |out_vector|
-//
-// Output:
-//      - out_vector    : Vector with the output
-//
-
-//
-// WebRtcSpl_AutoCorrelation(...)
-//
-// A 32-bit fix-point implementation of auto-correlation computation
-//
-// Input:
-//      - vector        : Vector to calculate autocorrelation upon
-//      - vector_length : Length (in samples) of |vector|
-//      - order         : The order up to which the autocorrelation should be
-//                        calculated
-//
-// Output:
-//      - result_vector : auto-correlation values (values should be seen
-//                        relative to each other since the absolute values
-//                        might have been down shifted to avoid overflow)
-//
-//      - scale         : The number of left shifts required to obtain the
-//                        auto-correlation in Q0
-//
-// Return value         : Number of samples in |result_vector|, i.e., (order+1)
-//
-
-//
-// WebRtcSpl_LevinsonDurbin(...)
-//
-// A 32-bit fix-point implementation of the Levinson-Durbin algorithm that
-// does NOT use the 64 bit class
-//
-// Input:
-//      - auto_corr : Vector with autocorrelation values of length >=
-//                    |use_order|+1
-//      - use_order : The LPC filter order (support up to order 20)
-//
-// Output:
-//      - lpc_coef  : lpc_coef[0..use_order] LPC coefficients in Q12
-//      - refl_coef : refl_coef[0...use_order-1]| Reflection coefficients in
-//                    Q15
-//
-// Return value     : 1 for stable 0 for unstable
-//
-
-//
-// WebRtcSpl_ReflCoefToLpc(...)
-//
-// Converts reflection coefficients |refl_coef| to LPC coefficients |lpc_coef|.
-// This version is a 16 bit operation.
-//
-// NOTE: The 16 bit refl_coef -> lpc_coef conversion might result in a
-// "slightly unstable" filter (i.e., a pole just outside the unit circle) in
-// "rare" cases even if the reflection coefficients are stable.
-//
-// Input:
-//      - refl_coef : Reflection coefficients in Q15 that should be converted
-//                    to LPC coefficients
-//      - use_order : Number of coefficients in |refl_coef|
-//
-// Output:
-//      - lpc_coef  : LPC coefficients in Q12
-//
-
-//
-// WebRtcSpl_LpcToReflCoef(...)
-//
-// Converts LPC coefficients |lpc_coef| to reflection coefficients |refl_coef|.
-// This version is a 16 bit operation.
-// The conversion is implemented by the step-down algorithm.
-//
-// Input:
-//      - lpc_coef  : LPC coefficients in Q12, that should be converted to
-//                    reflection coefficients
-//      - use_order : Number of coefficients in |lpc_coef|
-//
-// Output:
-//      - refl_coef : Reflection coefficients in Q15.
-//
-
-//
-// WebRtcSpl_AutoCorrToReflCoef(...)
-//
-// Calculates reflection coefficients (16 bit) from auto-correlation values
-//
-// Input:
-//      - auto_corr : Auto-correlation values
-//      - use_order : Number of coefficients wanted be calculated
-//
-// Output:
-//      - refl_coef : Reflection coefficients in Q15.
-//
-
-//
-// WebRtcSpl_CrossCorrelation(...)
-//
-// Calculates the cross-correlation between two sequences |vector1| and
-// |vector2|. |vector1| is fixed and |vector2| slides as the pointer is
-// increased with the amount |step_vector2|
-//
-// Input:
-//      - vector1           : First sequence (fixed throughout the correlation)
-//      - vector2           : Second sequence (slides |step_vector2| for each
-//                            new correlation)
-//      - dim_vector        : Number of samples to use in the cross-correlation
-//      - dim_cross_corr    : Number of cross-correlations to calculate (the
-//                            start position for |vector2| is updated for each
-//                            new one)
-//      - right_shifts      : Number of right bit shifts to use. This will
-//                            become the output Q-domain.
-//      - step_vector2      : How many (positive or negative) steps the
-//                            |vector2| pointer should be updated for each new
-//                            cross-correlation value.
-//
-// Output:
-//      - cross_corr        : The cross-correlation in Q(-right_shifts)
-//
-
-//
-// WebRtcSpl_GetHanningWindow(...)
-//
-// Creates (the first half of) a Hanning window. Size must be at least 1 and
-// at most 512.
-//
-// Input:
-//      - size      : Length of the requested Hanning window (1 to 512)
-//
-// Output:
-//      - window    : Hanning vector in Q14.
-//
-
-//
-// WebRtcSpl_SqrtOfOneMinusXSquared(...)
-//
-// Calculates y[k] = sqrt(1 - x[k]^2) for each element of the input vector
-// |in_vector|. Input and output values are in Q15.
-//
-// Inputs:
-//      - in_vector     : Values to calculate sqrt(1 - x^2) of
-//      - vector_length : Length of vector |in_vector|
-//
-// Output:
-//      - out_vector    : Output values in Q15
-//
-
-//
-// WebRtcSpl_IncreaseSeed(...)
-//
-// Increases the seed (and returns the new value)
-//
-// Input:
-//      - seed      : Seed for random calculation
-//
-// Output:
-//      - seed      : Updated seed value
-//
-// Return value     : The new seed value
-//
-
-//
-// WebRtcSpl_RandU(...)
-//
-// Produces a uniformly distributed value in the WebRtc_Word16 range
-//
-// Input:
-//      - seed      : Seed for random calculation
-//
-// Output:
-//      - seed      : Updated seed value
-//
-// Return value     : Uniformly distributed value in the range
-//                    [Word16_MIN...Word16_MAX]
-//
-
-//
-// WebRtcSpl_RandN(...)
-//
-// Produces a normal distributed value in the WebRtc_Word16 range
-//
-// Input:
-//      - seed      : Seed for random calculation
-//
-// Output:
-//      - seed      : Updated seed value
-//
-// Return value     : N(0,1) value in the Q13 domain
-//
-
-//
-// WebRtcSpl_RandUArray(...)
-//
-// Produces a uniformly distributed vector with elements in the WebRtc_Word16
-// range
-//
-// Input:
-//      - vector_length : Samples wanted in the vector
-//      - seed          : Seed for random calculation
-//
-// Output:
-//      - vector        : Vector with the uniform values
-//      - seed          : Updated seed value
-//
-// Return value         : Number of samples in vector, i.e., |vector_length|
-//
-
-//
-// WebRtcSpl_Sqrt(...)
-//
-// Returns the square root of the input value |value|. The precision of this
-// function is integer precision, i.e., sqrt(8) gives 2 as answer.
-// If |value| is a negative number then 0 is returned.
-//
-// Algorithm:
-//
-// A sixth order Taylor Series expansion is used here to compute the square
-// root of a number y^0.5 = (1+x)^0.5
-// where
-// x = y-1
-//   = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
-// 0.5 <= x < 1
-//
-// Input:
-//      - value     : Value to calculate sqrt of
-//
-// Return value     : Result of the sqrt calculation
-//
-
-//
-// WebRtcSpl_SqrtFloor(...)
-//
-// Returns the square root of the input value |value|. The precision of this
-// function is rounding down integer precision, i.e., sqrt(8) gives 2 as answer.
-// If |value| is a negative number then 0 is returned.
-//
-// Algorithm:
-//
-// An iterative 4 cylce/bit routine
-//
-// Input:
-//      - value     : Value to calculate sqrt of
-//
-// Return value     : Result of the sqrt calculation
-//
-
-//
-// WebRtcSpl_DivU32U16(...)
-//
-// Divides a WebRtc_UWord32 |num| by a WebRtc_UWord16 |den|.
-//
-// If |den|==0, (WebRtc_UWord32)0xFFFFFFFF is returned.
-//
-// Input:
-//      - num       : Numerator
-//      - den       : Denominator
-//
-// Return value     : Result of the division (as a WebRtc_UWord32), i.e., the
-//                    integer part of num/den.
-//
-
-//
-// WebRtcSpl_DivW32W16(...)
-//
-// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|.
-//
-// If |den|==0, (WebRtc_Word32)0x7FFFFFFF is returned.
-//
-// Input:
-//      - num       : Numerator
-//      - den       : Denominator
-//
-// Return value     : Result of the division (as a WebRtc_Word32), i.e., the
-//                    integer part of num/den.
-//
-
-//
-// WebRtcSpl_DivW32W16ResW16(...)
-//
-// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|, assuming that the
-// result is less than 32768, otherwise an unpredictable result will occur.
-//
-// If |den|==0, (WebRtc_Word16)0x7FFF is returned.
-//
-// Input:
-//      - num       : Numerator
-//      - den       : Denominator
-//
-// Return value     : Result of the division (as a WebRtc_Word16), i.e., the
-//                    integer part of num/den.
-//
-
-//
-// WebRtcSpl_DivResultInQ31(...)
-//
-// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|, assuming that the
-// absolute value of the denominator is larger than the numerator, otherwise
-// an unpredictable result will occur.
-//
-// Input:
-//      - num       : Numerator
-//      - den       : Denominator
-//
-// Return value     : Result of the division in Q31.
-//
-
-//
-// WebRtcSpl_DivW32HiLow(...)
-//
-// Divides a WebRtc_Word32 |num| by a denominator in hi, low format. The
-// absolute value of the denominator has to be larger (or equal to) the
-// numerator.
-//
-// Input:
-//      - num       : Numerator
-//      - den_hi    : High part of denominator
-//      - den_low   : Low part of denominator
-//
-// Return value     : Divided value in Q31
-//
-
-//
-// WebRtcSpl_Energy(...)
-//
-// Calculates the energy of a vector
-//
-// Input:
-//      - vector        : Vector which the energy should be calculated on
-//      - vector_length : Number of samples in vector
-//
-// Output:
-//      - scale_factor  : Number of left bit shifts needed to get the physical
-//                        energy value, i.e, to get the Q0 value
-//
-// Return value         : Energy value in Q(-|scale_factor|)
-//
-
-//
-// WebRtcSpl_FilterAR(...)
-//
-// Performs a 32-bit AR filtering on a vector in Q12
-//
-// Input:
-//  - ar_coef                   : AR-coefficient vector (values in Q12),
-//                                ar_coef[0] must be 4096.
-//  - ar_coef_length            : Number of coefficients in |ar_coef|.
-//  - in_vector                 : Vector to be filtered.
-//  - in_vector_length          : Number of samples in |in_vector|.
-//  - filter_state              : Current state (higher part) of the filter.
-//  - filter_state_length       : Length (in samples) of |filter_state|.
-//  - filter_state_low          : Current state (lower part) of the filter.
-//  - filter_state_low_length   : Length (in samples) of |filter_state_low|.
-//  - out_vector_low_length     : Maximum length (in samples) of
-//                                |out_vector_low|.
-//
-// Output:
-//  - filter_state              : Updated state (upper part) vector.
-//  - filter_state_low          : Updated state (lower part) vector.
-//  - out_vector                : Vector containing the upper part of the
-//                                filtered values.
-//  - out_vector_low            : Vector containing the lower part of the
-//                                filtered values.
-//
-// Return value                 : Number of samples in the |out_vector|.
-//
-
-//
-// WebRtcSpl_FilterMAFastQ12(...)
-//
-// Performs a MA filtering on a vector in Q12
-//
-// Input:
-//      - in_vector         : Input samples (state in positions
-//                            in_vector[-order] .. in_vector[-1])
-//      - ma_coef           : Filter coefficients (in Q12)
-//      - ma_coef_length    : Number of B coefficients (order+1)
-//      - vector_length     : Number of samples to be filtered
-//
-// Output:
-//      - out_vector        : Filtered samples
-//
-
-//
-// WebRtcSpl_FilterARFastQ12(...)
-//
-// Performs a AR filtering on a vector in Q12
-//
-// Input:
-//      - in_vector         : Input samples
-//      - out_vector        : State information in positions
-//                            out_vector[-order] .. out_vector[-1]
-//      - ar_coef           : Filter coefficients (in Q12)
-//      - ar_coef_length    : Number of B coefficients (order+1)
-//      - vector_length     : Number of samples to be filtered
-//
-// Output:
-//      - out_vector        : Filtered samples
-//
-
-//
-// WebRtcSpl_DownsampleFast(...)
-//
-// Performs a MA down sampling filter on a vector
-//
-// Input:
-//      - in_vector         : Input samples (state in positions
-//                            in_vector[-order] .. in_vector[-1])
-//      - in_vector_length  : Number of samples in |in_vector| to be filtered.
-//                            This must be at least
-//                            |delay| + |factor|*(|out_vector_length|-1) + 1)
-//      - out_vector_length : Number of down sampled samples desired
-//      - ma_coef           : Filter coefficients (in Q12)
-//      - ma_coef_length    : Number of B coefficients (order+1)
-//      - factor            : Decimation factor
-//      - delay             : Delay of filter (compensated for in out_vector)
-//
-// Output:
-//      - out_vector        : Filtered samples
-//
-// Return value             : 0 if OK, -1 if |in_vector| is too short
-//
-
-//
-// WebRtcSpl_DotProductWithScale(...)
-//
-// Calculates the dot product between two (WebRtc_Word16) vectors
-//
-// Input:
-//      - vector1       : Vector 1
-//      - vector2       : Vector 2
-//      - vector_length : Number of samples used in the dot product
-//      - scaling       : The number of right bit shifts to apply on each term
-//                        during calculation to avoid overflow, i.e., the
-//                        output will be in Q(-|scaling|)
-//
-// Return value         : The dot product in Q(-scaling)
-//
-
-//
-// WebRtcSpl_ComplexIFFT(...)
-//
-// Complex Inverse FFT
-//
-// Computes an inverse complex 2^|stages|-point FFT on the input vector, which
-// is in bit-reversed order. The original content of the vector is destroyed in
-// the process, since the input is overwritten by the output, normal-ordered,
-// FFT vector. With X as the input complex vector, y as the output complex
-// vector and with M = 2^|stages|, the following is computed:
-//
-//        M-1
-// y(k) = sum[X(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
-//        i=0
-//
-// The implementations are optimized for speed, not for code size. It uses the
-// decimation-in-time algorithm with radix-2 butterfly technique.
-//
-// Input:
-//      - vector    : In pointer to complex vector containing 2^|stages|
-//                    real elements interleaved with 2^|stages| imaginary
-//                    elements.
-//                    [ReImReImReIm....]
-//                    The elements are in Q(-scale) domain, see more on Return
-//                    Value below.
-//
-//      - stages    : Number of FFT stages. Must be at least 3 and at most 10,
-//                    since the table WebRtcSpl_kSinTable1024[] is 1024
-//                    elements long.
-//
-//      - mode      : This parameter gives the user to choose how the FFT
-//                    should work.
-//                    mode==0: Low-complexity and Low-accuracy mode
-//                    mode==1: High-complexity and High-accuracy mode
-//
-// Output:
-//      - vector    : Out pointer to the FFT vector (the same as input).
-//
-// Return Value     : The scale value that tells the number of left bit shifts
-//                    that the elements in the |vector| should be shifted with
-//                    in order to get Q0 values, i.e. the physically correct
-//                    values. The scale parameter is always 0 or positive,
-//                    except if N>1024 (|stages|>10), which returns a scale
-//                    value of -1, indicating error.
-//
-
-#if (defined ARM9E_GCC) || (defined ARM_WINM) || (defined ANDROID_AECOPT)
-//
-// WebRtcSpl_ComplexIFFT2(...)
-//
-// Complex or Real inverse FFT, for ARM processor only
-//
-// Computes a 2^|stages|-point FFT on the input vector, which can be or not be
-// in bit-reversed order. If it is bit-reversed, the original content of the
-// vector could be overwritten by the output by setting the first two arguments
-// the same. With X as the input complex vector, y as the output complex vector
-// and with M = 2^|stages|, the following is computed:
-//
-//        M-1
-// y(k) = sum[X(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
-//        i=0
-//
-// The implementations are optimized for speed, not for code size. It uses the
-// decimation-in-time algorithm with radix-2 butterfly technique.
-//
-// Arguments:
-//      - in_vector     : In pointer to complex vector containing 2^|stages|
-//                        real elements interleaved with 2^|stages| imaginary
-//                        elements. [ReImReImReIm....]
-//                        The elements are in Q(-scale) domain.
-//      - out_vector    : Output pointer to vector containing 2^|stages| real
-//                        elements interleaved with 2^|stages| imaginary
-//                        elements. [ReImReImReIm....]
-//                        The output is in the Q0 domain.
-//      - stages        : Number of FFT stages. Must be at least 3 and at most
-//                        10.
-//      - mode          : Dummy input.
-//
-// Return value         : The scale parameter is always 0, except if N>1024,
-//                        which returns a scale value of -1, indicating error.
-//
-#endif
-
-//
-// WebRtcSpl_ComplexFFT(...)
-//
-// Complex FFT
-//
-// Computes a complex 2^|stages|-point FFT on the input vector, which is in
-// bit-reversed order. The original content of the vector is destroyed in
-// the process, since the input is overwritten by the output, normal-ordered,
-// FFT vector. With x as the input complex vector, Y as the output complex
-// vector and with M = 2^|stages|, the following is computed:
-//
-//              M-1
-// Y(k) = 1/M * sum[x(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
-//              i=0
-//
-// The implementations are optimized for speed, not for code size. It uses the
-// decimation-in-time algorithm with radix-2 butterfly technique.
-//
-// This routine prevents overflow by scaling by 2 before each FFT stage. This is
-// a fixed scaling, for proper normalization - there will be log2(n) passes, so
-// this results in an overall factor of 1/n, distributed to maximize arithmetic
-// accuracy.
-//
-// Input:
-//      - vector    : In pointer to complex vector containing 2^|stages| real
-//                    elements interleaved with 2^|stages| imaginary elements.
-//                    [ReImReImReIm....]
-//                    The output is in the Q0 domain.
-//
-//      - stages    : Number of FFT stages. Must be at least 3 and at most 10,
-//                    since the table WebRtcSpl_kSinTable1024[] is 1024
-//                    elements long.
-//
-//      - mode      : This parameter gives the user to choose how the FFT
-//                    should work.
-//                    mode==0: Low-complexity and Low-accuracy mode
-//                    mode==1: High-complexity and High-accuracy mode
-//
-// Output:
-//      - vector    : The output FFT vector is in the Q0 domain.
-//
-// Return value     : The scale parameter is always 0, except if N>1024,
-//                    which returns a scale value of -1, indicating error.
-//
-
-#if (defined ARM9E_GCC) || (defined ARM_WINM) || (defined ANDROID_AECOPT)
-//
-// WebRtcSpl_ComplexFFT2(...)
-//
-// Complex or Real FFT, for ARM processor only
-//
-// Computes a 2^|stages|-point FFT on the input vector, which can be or not be
-// in bit-reversed order. If it is bit-reversed, the original content of the
-// vector could be overwritten by the output by setting the first two arguments
-// the same. With x as the input complex vector, Y as the output complex vector
-// and with M = 2^|stages|, the following is computed:
-//
-//              M-1
-// Y(k) = 1/M * sum[x(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
-//              i=0
-//
-// The implementations are optimized for speed, not for code size. It uses the
-// decimation-in-time algorithm with radix-2 butterfly technique.
-//
-// Arguments:
-//      - in_vector     : In pointer to complex vector containing 2^|stages|
-//                        real elements interleaved with 2^|stages| imaginary
-//                        elements. [ReImReImReIm....]
-//      - out_vector    : Output pointer to vector containing 2^|stages| real
-//                        elements interleaved with 2^|stages| imaginary
-//                        elements. [ReImReImReIm....]
-//                        The output is in the Q0 domain.
-//      - stages        : Number of FFT stages. Must be at least 3 and at most
-//                        10.
-//      - mode          : Dummy input
-//
-// Return value         : The scale parameter is always 0, except if N>1024,
-//                        which returns a scale value of -1, indicating error.
-//
-#endif
-
-//
-// WebRtcSpl_ComplexBitReverse(...)
-//
-// Complex Bit Reverse
-//
-// This function bit-reverses the position of elements in the complex input
-// vector into the output vector.
-//
-// If you bit-reverse a linear-order array, you obtain a bit-reversed order
-// array. If you bit-reverse a bit-reversed order array, you obtain a
-// linear-order array.
-//
-// Input:
-//      - vector    : In pointer to complex vector containing 2^|stages| real
-//                    elements interleaved with 2^|stages| imaginary elements.
-//                    [ReImReImReIm....]
-//      - stages    : Number of FFT stages. Must be at least 3 and at most 10,
-//                    since the table WebRtcSpl_kSinTable1024[] is 1024
-//                    elements long.
-//
-// Output:
-//      - vector    : Out pointer to complex vector in bit-reversed order.
-//                    The input vector is over written.
-//
-
-//
-// WebRtcSpl_AnalysisQMF(...)
-//
-// Splits a 0-2*F Hz signal into two sub bands: 0-F Hz and F-2*F Hz. The
-// current version has F = 8000, therefore, a super-wideband audio signal is
-// split to lower-band 0-8 kHz and upper-band 8-16 kHz.
-//
-// Input:
-//      - in_data       : Wide band speech signal, 320 samples (10 ms)
-//
-// Input & Output:
-//      - filter_state1 : Filter state for first All-pass filter
-//      - filter_state2 : Filter state for second All-pass filter
-//
-// Output:
-//      - low_band      : Lower-band signal 0-8 kHz band, 160 samples (10 ms)
-//      - high_band     : Upper-band signal 8-16 kHz band (flipped in frequency
-//                        domain), 160 samples (10 ms)
-//
-
-//
-// WebRtcSpl_SynthesisQMF(...)
-//
-// Combines the two sub bands (0-F and F-2*F Hz) into a signal of 0-2*F
-// Hz, (current version has F = 8000 Hz). So the filter combines lower-band
-// (0-8 kHz) and upper-band (8-16 kHz) channels to obtain super-wideband 0-16
-// kHz audio.
-//
-// Input:
-//      - low_band      : The signal with the 0-8 kHz band, 160 samples (10 ms)
-//      - high_band     : The signal with the 8-16 kHz band, 160 samples (10 ms)
-//
-// Input & Output:
-//      - filter_state1 : Filter state for first All-pass filter
-//      - filter_state2 : Filter state for second All-pass filter
-//
-// Output:
-//      - out_data      : Super-wideband speech signal, 0-16 kHz
-//
-
-// WebRtc_Word16 WebRtcSpl_get_version(...)
-//
-// This function gives the version string of the Signal Processing Library.
-//
-// Input:
-//      - length_in_bytes   : The size of Allocated space (in Bytes) where
-//                            the version number is written to (in string format).
-//
-// Output:
-//      - version           : Pointer to a buffer where the version number is written to.
-//
diff --git a/src/common_audio/signal_processing_library/main/interface/spl_inl.h b/src/common_audio/signal_processing_library/main/interface/spl_inl.h
deleted file mode 100644
index 8716ca9..0000000
--- a/src/common_audio/signal_processing_library/main/interface/spl_inl.h
+++ /dev/null
@@ -1,293 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-// This header file includes the inline functions in
-// the fix point signal processing library.
-
-#ifndef WEBRTC_SPL_SPL_INL_H_
-#define WEBRTC_SPL_SPL_INL_H_
-
-#ifdef WEBRTC_SPL_INLINE_CALLS
-
-#ifdef WEBRTC_ANDROID
-
-WEBRTC_INLINE WebRtc_Word32 WEBRTC_SPL_MUL(WebRtc_Word32 a, WebRtc_Word32 b)
-{
-    WebRtc_Word32 tmp;
-    __asm__("mul %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
-    return tmp;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WEBRTC_SPL_MUL_16_32_RSFT16(WebRtc_Word16 a,
-                                                        WebRtc_Word32 b)
-{
-    WebRtc_Word32 tmp;
-    __asm__("smulwb %0, %1, %2":"=r"(tmp):"r"(b), "r"(a));
-    return tmp;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32(WebRtc_Word16 a,
-                                                      WebRtc_Word16 b,
-                                                      WebRtc_Word32 c)
-{
-    WebRtc_Word32 tmp;
-    __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(tmp) : "r"(b), "r"(a));
-    __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(tmp), "r"(c));
-    return tmp;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32BI(
-        WebRtc_Word32 a,
-        WebRtc_Word32 b)
-{
-    WebRtc_Word32 tmp;
-    __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
-    return tmp;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WEBRTC_SPL_MUL_16_16(WebRtc_Word16 a,
-                                                 WebRtc_Word16 b)
-{
-    WebRtc_Word32 tmp;
-    __asm__("smulbb %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
-    return tmp;
-}
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
-                                                WebRtc_Word16 b)
-{
-    WebRtc_Word32 s_sum;
-
-    __asm__("qadd16 %0, %1, %2":"=r"(s_sum):"r"(a), "r"(b));
-
-    return (WebRtc_Word16) s_sum;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
-                                                WebRtc_Word32 l_var2)
-{
-    WebRtc_Word32 l_sum;
-
-    __asm__("qadd %0, %1, %2":"=r"(l_sum):"r"(l_var1), "r"(l_var2));
-
-    return l_sum;
-}
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
-                                                WebRtc_Word16 var2)
-{
-    WebRtc_Word32 s_sub;
-
-    __asm__("qsub16 %0, %1, %2":"=r"(s_sub):"r"(var1), "r"(var2));
-
-    return (WebRtc_Word16)s_sub;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
-                                                WebRtc_Word32 l_var2)
-{
-    WebRtc_Word32 l_sub;
-
-    __asm__("qsub %0, %1, %2":"=r"(l_sub):"r"(l_var1), "r"(l_var2));
-
-    return l_sub;
-}
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n)
-{
-    WebRtc_Word32 tmp;
-
-    __asm__("clz %0, %1":"=r"(tmp):"r"(n));
-
-    return (WebRtc_Word16)(32 - tmp);
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormW32(WebRtc_Word32 a)
-{
-    WebRtc_Word32 tmp;
-
-    if (a <= 0) a ^= 0xFFFFFFFF;
-
-    __asm__("clz %0, %1":"=r"(tmp):"r"(a));
-
-    return tmp - 1;
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormW16(WebRtc_Word16 a)
-{
-    int zeros;
-
-    if (a <= 0) a ^= 0xFFFF;
-
-    if (!(0xFF80 & a)) zeros = 8; else zeros = 0;
-    if (!(0xF800 & (a << zeros))) zeros += 4;
-    if (!(0xE000 & (a << zeros))) zeros += 2;
-    if (!(0xC000 & (a << zeros))) zeros += 1;
-
-    return zeros;
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormU32(WebRtc_UWord32 a)
-{
-    int tmp;
-
-    if (a == 0) return 0;
-
-    __asm__("clz %0, %1":"=r"(tmp):"r"(a));
-
-    return tmp;
-}
-
-#else
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
-                                                WebRtc_Word16 b)
-{
-    WebRtc_Word32 s_sum = (WebRtc_Word32) a + (WebRtc_Word32) b;
-
-    if (s_sum > WEBRTC_SPL_WORD16_MAX)
-    s_sum = WEBRTC_SPL_WORD16_MAX;
-    else if (s_sum < WEBRTC_SPL_WORD16_MIN)
-    s_sum = WEBRTC_SPL_WORD16_MIN;
-
-    return (WebRtc_Word16)s_sum;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
-                                                WebRtc_Word32 l_var2)
-{
-    WebRtc_Word32 l_sum;
-
-    // perform long addition
-    l_sum = l_var1 + l_var2;
-
-    // check for under or overflow
-    if (WEBRTC_SPL_IS_NEG (l_var1))
-    {
-        if (WEBRTC_SPL_IS_NEG (l_var2) && !WEBRTC_SPL_IS_NEG (l_sum))
-        {
-            l_sum = (WebRtc_Word32)0x80000000;
-        }
-    }
-    else
-    {
-        if (!WEBRTC_SPL_IS_NEG (l_var2) && WEBRTC_SPL_IS_NEG (l_sum))
-        {
-            l_sum = (WebRtc_Word32)0x7FFFFFFF;
-        }
-    }
-
-    return l_sum;
-}
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_SubSatW16( WebRtc_Word16 var1,
-                                                 WebRtc_Word16 var2)
-{
-    WebRtc_Word32 l_diff;
-    WebRtc_Word16 s_diff;
-
-    // perform subtraction
-    l_diff = (WebRtc_Word32)var1 - (WebRtc_Word32)var2;
-
-    // default setting
-    s_diff = (WebRtc_Word16) l_diff;
-
-    // check for overflow
-    if (l_diff > (WebRtc_Word32)32767)
-    s_diff = (WebRtc_Word16)32767;
-
-    // check for underflow
-    if (l_diff < (WebRtc_Word32)-32768)
-    s_diff = (WebRtc_Word16)-32768;
-
-    return s_diff;
-}
-
-WEBRTC_INLINE WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
-                                                WebRtc_Word32 l_var2)
-{
-    WebRtc_Word32 l_diff;
-
-    // perform subtraction
-    l_diff = l_var1 - l_var2;
-
-    // check for underflow
-    if ((l_var1 < 0) && (l_var2 > 0) && (l_diff > 0))
-    l_diff = (WebRtc_Word32)0x80000000;
-    // check for overflow
-    if ((l_var1 > 0) && (l_var2 < 0) && (l_diff < 0))
-    l_diff = (WebRtc_Word32)0x7FFFFFFF;
-
-    return l_diff;
-}
-
-WEBRTC_INLINE WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n)
-{
-
-    int bits;
-
-    if ((0xFFFF0000 & n)) bits = 16; else bits = 0;
-    if ((0x0000FF00 & (n >> bits))) bits += 8;
-    if ((0x000000F0 & (n >> bits))) bits += 4;
-    if ((0x0000000C & (n >> bits))) bits += 2;
-    if ((0x00000002 & (n >> bits))) bits += 1;
-    if ((0x00000001 & (n >> bits))) bits += 1;
-
-    return bits;
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormW32(WebRtc_Word32 a)
-{
-    int zeros;
-
-    if (a <= 0) a ^= 0xFFFFFFFF;
-
-    if (!(0xFFFF8000 & a)) zeros = 16; else zeros = 0;
-    if (!(0xFF800000 & (a << zeros))) zeros += 8;
-    if (!(0xF8000000 & (a << zeros))) zeros += 4;
-    if (!(0xE0000000 & (a << zeros))) zeros += 2;
-    if (!(0xC0000000 & (a << zeros))) zeros += 1;
-
-    return zeros;
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormW16(WebRtc_Word16 a)
-{
-    int zeros;
-
-    if (a <= 0) a ^= 0xFFFF;
-
-    if (!(0xFF80 & a)) zeros = 8; else zeros = 0;
-    if (!(0xF800 & (a << zeros))) zeros += 4;
-    if (!(0xE000 & (a << zeros))) zeros += 2;
-    if (!(0xC000 & (a << zeros))) zeros += 1;
-
-    return zeros;
-}
-
-WEBRTC_INLINE int WebRtcSpl_NormU32(WebRtc_UWord32 a)
-{
-    int zeros;
-
-    if (a == 0) return 0;
-
-    if (!(0xFFFF0000 & a)) zeros = 16; else zeros = 0;
-    if (!(0xFF000000 & (a << zeros))) zeros += 8;
-    if (!(0xF0000000 & (a << zeros))) zeros += 4;
-    if (!(0xC0000000 & (a << zeros))) zeros += 2;
-    if (!(0x80000000 & (a << zeros))) zeros += 1;
-
-    return zeros;
-}
-
-#endif // WEBRTC_ANDROID
-#endif // WEBRTC_SPL_INLINE_CALLS
-#endif // WEBRTC_SPL_SPL_INL_H_
diff --git a/src/common_audio/signal_processing_library/main/source/Android.mk b/src/common_audio/signal_processing_library/main/source/Android.mk
deleted file mode 100644
index 8b08676..0000000
--- a/src/common_audio/signal_processing_library/main/source/Android.mk
+++ /dev/null
@@ -1,99 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_spl
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := add_sat_w16.c \
-    add_sat_w32.c \
-    auto_corr_to_refl_coef.c \
-    auto_correlation.c \
-    complex_fft.c \
-    complex_ifft.c \
-    complex_bit_reverse.c \
-    copy_set_operations.c \
-    cos_table.c \
-    cross_correlation.c \
-    division_operations.c \
-    dot_product_with_scale.c \
-    downsample_fast.c \
-    energy.c \
-    filter_ar.c \
-    filter_ar_fast_q12.c \
-    filter_ma_fast_q12.c \
-    get_hanning_window.c \
-    get_scaling_square.c \
-    get_size_in_bits.c \
-    hanning_table.c \
-    ilbc_specific_functions.c \
-    levinson_durbin.c \
-    lpc_to_refl_coef.c \
-    min_max_operations.c \
-    norm_u32.c \
-    norm_w16.c \
-    norm_w32.c \
-    randn_table.c \
-    randomization_functions.c \
-    refl_coef_to_lpc.c \
-    resample.c \
-    resample_48khz.c \
-    resample_by_2.c \
-    resample_by_2_internal.c \
-    resample_fractional.c \
-    sin_table.c \
-    sin_table_1024.c \
-    spl_sqrt.c \
-    spl_version.c \
-    splitting_filter.c \
-    sqrt_of_one_minus_x_squared.c \
-    sub_sat_w16.c \
-    sub_sat_w32.c \
-    vector_scaling_operations.c
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=  
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
-    '-DNO_HEAPCHECKER' \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' 
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../.. \
-    $(LOCAL_PATH)/../interface 
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS := 
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libstlport
-
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
-LOCAL_LDLIBS += -ldl -lpthread
-endif
-
-ifneq ($(TARGET_SIMULATOR),true)
-LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/common_audio/signal_processing_library/main/source/add_sat_w16.c b/src/common_audio/signal_processing_library/main/source/add_sat_w16.c
deleted file mode 100644
index d103999..0000000
--- a/src/common_audio/signal_processing_library/main/source/add_sat_w16.c
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_AddSatW16().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 var1, WebRtc_Word16 var2)
-{
-    WebRtc_Word32 s_sum = (WebRtc_Word32)var1 + (WebRtc_Word32)var2;
-
-    if (s_sum > WEBRTC_SPL_WORD16_MAX)
-        s_sum = WEBRTC_SPL_WORD16_MAX;
-    else if (s_sum < WEBRTC_SPL_WORD16_MIN)
-        s_sum = WEBRTC_SPL_WORD16_MIN;
-
-    return (WebRtc_Word16)s_sum;
-}
-
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/add_sat_w32.c b/src/common_audio/signal_processing_library/main/source/add_sat_w32.c
deleted file mode 100644
index 6d83e75..0000000
--- a/src/common_audio/signal_processing_library/main/source/add_sat_w32.c
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_AddSatW32().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 var1, WebRtc_Word32 var2)
-{
-    WebRtc_Word32 l_sum;
-
-    // perform long addition
-    l_sum = var1 + var2;
-
-    // check for under or overflow
-    if (WEBRTC_SPL_IS_NEG(var1))
-    {
-        if (WEBRTC_SPL_IS_NEG(var2) && !WEBRTC_SPL_IS_NEG(l_sum))
-        {
-            l_sum = (WebRtc_Word32)0x80000000;
-        }
-    } else
-    {
-        if (!WEBRTC_SPL_IS_NEG(var2) && WEBRTC_SPL_IS_NEG(l_sum))
-        {
-            l_sum = (WebRtc_Word32)0x7FFFFFFF;
-        }
-    }
-
-    return l_sum;
-}
-
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/auto_corr_to_refl_coef.c b/src/common_audio/signal_processing_library/main/source/auto_corr_to_refl_coef.c
deleted file mode 100644
index b7e8858..0000000
--- a/src/common_audio/signal_processing_library/main/source/auto_corr_to_refl_coef.c
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_AutoCorrToReflCoef().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_AutoCorrToReflCoef(G_CONST WebRtc_Word32 *R, int use_order, WebRtc_Word16 *K)
-{
-    int i, n;
-    WebRtc_Word16 tmp;
-    G_CONST WebRtc_Word32 *rptr;
-    WebRtc_Word32 L_num, L_den;
-    WebRtc_Word16 *acfptr, *pptr, *wptr, *p1ptr, *w1ptr, ACF[WEBRTC_SPL_MAX_LPC_ORDER],
-            P[WEBRTC_SPL_MAX_LPC_ORDER], W[WEBRTC_SPL_MAX_LPC_ORDER];
-
-    // Initialize loop and pointers.
-    acfptr = ACF;
-    rptr = R;
-    pptr = P;
-    p1ptr = &P[1];
-    w1ptr = &W[1];
-    wptr = w1ptr;
-
-    // First loop; n=0. Determine shifting.
-    tmp = WebRtcSpl_NormW32(*R);
-    *acfptr = (WebRtc_Word16)((*rptr++ << tmp) >> 16);
-    *pptr++ = *acfptr++;
-
-    // Initialize ACF, P and W.
-    for (i = 1; i <= use_order; i++)
-    {
-        *acfptr = (WebRtc_Word16)((*rptr++ << tmp) >> 16);
-        *wptr++ = *acfptr;
-        *pptr++ = *acfptr++;
-    }
-
-    // Compute reflection coefficients.
-    for (n = 1; n <= use_order; n++, K++)
-    {
-        tmp = WEBRTC_SPL_ABS_W16(*p1ptr);
-        if (*P < tmp)
-        {
-            for (i = n; i <= use_order; i++)
-                *K++ = 0;
-
-            return;
-        }
-
-        // Division: WebRtcSpl_div(tmp, *P)
-        *K = 0;
-        if (tmp != 0)
-        {
-            L_num = tmp;
-            L_den = *P;
-            i = 15;
-            while (i--)
-            {
-                (*K) <<= 1;
-                L_num <<= 1;
-                if (L_num >= L_den)
-                {
-                    L_num -= L_den;
-                    (*K)++;
-                }
-            }
-            if (*p1ptr > 0)
-                *K = -*K;
-        }
-
-        // Last iteration; don't do Schur recursion.
-        if (n == use_order)
-            return;
-
-        // Schur recursion.
-        pptr = P;
-        wptr = w1ptr;
-        tmp = (WebRtc_Word16)(((WebRtc_Word32)*p1ptr * (WebRtc_Word32)*K + 16384) >> 15);
-        *pptr = WEBRTC_SPL_ADD_SAT_W16( *pptr, tmp );
-        pptr++;
-        for (i = 1; i <= use_order - n; i++)
-        {
-            tmp = (WebRtc_Word16)(((WebRtc_Word32)*wptr * (WebRtc_Word32)*K + 16384) >> 15);
-            *pptr = WEBRTC_SPL_ADD_SAT_W16( *(pptr+1), tmp );
-            pptr++;
-            tmp = (WebRtc_Word16)(((WebRtc_Word32)*pptr * (WebRtc_Word32)*K + 16384) >> 15);
-            *wptr = WEBRTC_SPL_ADD_SAT_W16( *wptr, tmp );
-            wptr++;
-        }
-    }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/auto_correlation.c b/src/common_audio/signal_processing_library/main/source/auto_correlation.c
deleted file mode 100644
index a00fde4..0000000
--- a/src/common_audio/signal_processing_library/main/source/auto_correlation.c
+++ /dev/null
@@ -1,141 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_AutoCorrelation().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-int WebRtcSpl_AutoCorrelation(G_CONST WebRtc_Word16* in_vector,
-                              int in_vector_length,
-                              int order,
-                              WebRtc_Word32* result,
-                              int* scale)
-{
-    WebRtc_Word32 sum;
-    int i, j;
-    WebRtc_Word16 smax; // Sample max
-    G_CONST WebRtc_Word16* xptr1;
-    G_CONST WebRtc_Word16* xptr2;
-    WebRtc_Word32* resultptr;
-    int scaling = 0;
-
-#ifdef _ARM_OPT_
-#pragma message("NOTE: _ARM_OPT_ optimizations are used")
-    WebRtc_Word16 loops4;
-#endif
-
-    if (order < 0)
-        order = in_vector_length;
-
-    // Find the max. sample
-    smax = WebRtcSpl_MaxAbsValueW16(in_vector, in_vector_length);
-
-    // In order to avoid overflow when computing the sum we should scale the samples so that
-    // (in_vector_length * smax * smax) will not overflow.
-
-    if (smax == 0)
-    {
-        scaling = 0;
-    } else
-    {
-        int nbits = WebRtcSpl_GetSizeInBits(in_vector_length); // # of bits in the sum loop
-        int t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax)); // # of bits to normalize smax
-
-        if (t > nbits)
-        {
-            scaling = 0;
-        } else
-        {
-            scaling = nbits - t;
-        }
-
-    }
-
-    resultptr = result;
-
-    // Perform the actual correlation calculation
-    for (i = 0; i < order + 1; i++)
-    {
-        int loops = (in_vector_length - i);
-        sum = 0;
-        xptr1 = in_vector;
-        xptr2 = &in_vector[i];
-#ifndef _ARM_OPT_
-        for (j = loops; j > 0; j--)
-        {
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1++, *xptr2++, scaling);
-        }
-#else
-        loops4 = (loops >> 2) << 2;
-
-        if (scaling == 0)
-        {
-            for (j = 0; j < loops4; j = j + 4)
-            {
-                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
-                xptr1++;
-                xptr2++;
-            }
-
-            for (j = loops4; j < loops; j++)
-            {
-                sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
-                xptr1++;
-                xptr2++;
-            }
-        }
-        else
-        {
-            for (j = 0; j < loops4; j = j + 4)
-            {
-                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
-                xptr1++;
-                xptr2++;
-                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
-                xptr1++;
-                xptr2++;
-            }
-
-            for (j = loops4; j < loops; j++)
-            {
-                sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
-                xptr1++;
-                xptr2++;
-            }
-        }
-
-#endif
-        *resultptr++ = sum;
-    }
-
-    *scale = scaling;
-
-    return order + 1;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/complex_bit_reverse.c b/src/common_audio/signal_processing_library/main/source/complex_bit_reverse.c
deleted file mode 100644
index 85c76f8..0000000
--- a/src/common_audio/signal_processing_library/main/source/complex_bit_reverse.c
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_ComplexBitReverse().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_ComplexBitReverse(WebRtc_Word16 frfi[], int stages)
-{
-    int mr, nn, n, l, m;
-    WebRtc_Word16 tr, ti;
-
-    n = 1 << stages;
-
-    mr = 0;
-    nn = n - 1;
-
-    // decimation in time - re-order data
-    for (m = 1; m <= nn; ++m)
-    {
-        l = n;
-        do
-        {
-            l >>= 1;
-        } while (mr + l > nn);
-        mr = (mr & (l - 1)) + l;
-
-        if (mr <= m)
-            continue;
-
-        tr = frfi[2 * m];
-        frfi[2 * m] = frfi[2 * mr];
-        frfi[2 * mr] = tr;
-
-        ti = frfi[2 * m + 1];
-        frfi[2 * m + 1] = frfi[2 * mr + 1];
-        frfi[2 * mr + 1] = ti;
-    }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/complex_fft.c b/src/common_audio/signal_processing_library/main/source/complex_fft.c
deleted file mode 100644
index b6f0c4e..0000000
--- a/src/common_audio/signal_processing_library/main/source/complex_fft.c
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_ComplexFFT().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define CFFTSFT 14
-#define CFFTRND 1
-#define CFFTRND2 16384
-
-#if (defined ARM9E_GCC) || (defined ARM_WINM) || (defined ANDROID_AECOPT)
-extern "C" int FFT_4OFQ14(void *src, void *dest, int NC, int shift);
-
-// For detailed description of the fft functions, check the readme files in fft_ARM9E folder.
-int WebRtcSpl_ComplexFFT2(WebRtc_Word16 frfi[], WebRtc_Word16 frfiOut[], int stages, int mode)
-{
-    return FFT_4OFQ14(frfi, frfiOut, 1 << stages, 0);
-}
-#endif
-
-int WebRtcSpl_ComplexFFT(WebRtc_Word16 frfi[], int stages, int mode)
-{
-    int i, j, l, k, istep, n, m;
-    WebRtc_Word16 wr, wi;
-    WebRtc_Word32 tr32, ti32, qr32, qi32;
-
-    /* The 1024-value is a constant given from the size of WebRtcSpl_kSinTable1024[],
-     * and should not be changed depending on the input parameter 'stages'
-     */
-    n = 1 << stages;
-    if (n > 1024)
-        return -1;
-
-    l = 1;
-    k = 10 - 1; /* Constant for given WebRtcSpl_kSinTable1024[]. Do not change
-         depending on the input parameter 'stages' */
-
-    if (mode == 0)
-    {
-        // mode==0: Low-complexity and Low-accuracy mode
-        while (l < n)
-        {
-            istep = l << 1;
-
-            for (m = 0; m < l; ++m)
-            {
-                j = m << k;
-
-                /* The 256-value is a constant given as 1/4 of the size of
-                 * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
-                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
-                 */
-                wr = WebRtcSpl_kSinTable1024[j + 256];
-                wi = -WebRtcSpl_kSinTable1024[j];
-
-                for (i = m; i < n; i += istep)
-                {
-                    j = i + l;
-
-                    tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
-                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1])), 15);
-
-                    ti32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
-                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j])), 15);
-
-                    qr32 = (WebRtc_Word32)frfi[2 * i];
-                    qi32 = (WebRtc_Word32)frfi[2 * i + 1];
-                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, 1);
-                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, 1);
-                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, 1);
-                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, 1);
-                }
-            }
-
-            --k;
-            l = istep;
-
-        }
-
-    } else
-    {
-        // mode==1: High-complexity and High-accuracy mode
-        while (l < n)
-        {
-            istep = l << 1;
-
-            for (m = 0; m < l; ++m)
-            {
-                j = m << k;
-
-                /* The 256-value is a constant given as 1/4 of the size of
-                 * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
-                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
-                 */
-                wr = WebRtcSpl_kSinTable1024[j + 256];
-                wi = -WebRtcSpl_kSinTable1024[j];
-
-                for (i = m; i < n; i += istep)
-                {
-                    j = i + l;
-
-                    tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
-                            - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CFFTRND),
-                            15 - CFFTSFT);
-
-                    ti32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
-                            + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CFFTRND), 15 - CFFTSFT);
-
-                    qr32 = ((WebRtc_Word32)frfi[2 * i]) << CFFTSFT;
-                    qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CFFTSFT;
-                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qr32 - tr32 + CFFTRND2), 1 + CFFTSFT);
-                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qi32 - ti32 + CFFTRND2), 1 + CFFTSFT);
-                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qr32 + tr32 + CFFTRND2), 1 + CFFTSFT);
-                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qi32 + ti32 + CFFTRND2), 1 + CFFTSFT);
-                }
-            }
-
-            --k;
-            l = istep;
-        }
-    }
-    return 0;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/complex_ifft.c b/src/common_audio/signal_processing_library/main/source/complex_ifft.c
deleted file mode 100644
index 184b8de..0000000
--- a/src/common_audio/signal_processing_library/main/source/complex_ifft.c
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_ComplexIFFT().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define CIFFTSFT 14
-#define CIFFTRND 1
-
-#if (defined ARM9E_GCC) || (defined ARM_WINM) || (defined ANDROID_AECOPT)
-extern "C" int FFT_4OIQ14(void *src, void *dest, int NC, int shift);
-
-// For detailed description of the fft functions, check the readme files in fft_ARM9E folder.
-int WebRtcSpl_ComplexIFFT2(WebRtc_Word16 frfi[], WebRtc_Word16 frfiOut[], int stages, int mode)
-{
-    FFT_4OIQ14(frfi, frfiOut, 1 << stages, 0);
-    return 0;
-}
-#endif
-
-int WebRtcSpl_ComplexIFFT(WebRtc_Word16 frfi[], int stages, int mode)
-{
-    int i, j, l, k, istep, n, m, scale, shift;
-    WebRtc_Word16 wr, wi;
-    WebRtc_Word32 tr32, ti32, qr32, qi32;
-    WebRtc_Word32 tmp32, round2;
-
-    /* The 1024-value is a constant given from the size of WebRtcSpl_kSinTable1024[],
-     * and should not be changed depending on the input parameter 'stages'
-     */
-    n = 1 << stages;
-    if (n > 1024)
-        return -1;
-
-    scale = 0;
-
-    l = 1;
-    k = 10 - 1; /* Constant for given WebRtcSpl_kSinTable1024[]. Do not change
-         depending on the input parameter 'stages' */
-
-    while (l < n)
-    {
-        // variable scaling, depending upon data
-        shift = 0;
-        round2 = 8192;
-
-        tmp32 = (WebRtc_Word32)WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
-        if (tmp32 > 13573)
-        {
-            shift++;
-            scale++;
-            round2 <<= 1;
-        }
-        if (tmp32 > 27146)
-        {
-            shift++;
-            scale++;
-            round2 <<= 1;
-        }
-
-        istep = l << 1;
-
-        if (mode == 0)
-        {
-            // mode==0: Low-complexity and Low-accuracy mode
-            for (m = 0; m < l; ++m)
-            {
-                j = m << k;
-
-                /* The 256-value is a constant given as 1/4 of the size of
-                 * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
-                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
-                 */
-                wr = WebRtcSpl_kSinTable1024[j + 256];
-                wi = WebRtcSpl_kSinTable1024[j];
-
-                for (i = m; i < n; i += istep)
-                {
-                    j = i + l;
-
-                    tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j], 0)
-                            - WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j + 1], 0)), 15);
-
-                    ti32 = WEBRTC_SPL_RSHIFT_W32(
-                            (WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j + 1], 0)
-                                    + WEBRTC_SPL_MUL_16_16_RSFT(wi,frfi[2*j],0)), 15);
-
-                    qr32 = (WebRtc_Word32)frfi[2 * i];
-                    qi32 = (WebRtc_Word32)frfi[2 * i + 1];
-                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, shift);
-                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, shift);
-                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, shift);
-                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, shift);
-                }
-            }
-        } else
-        {
-            // mode==1: High-complexity and High-accuracy mode
-
-            for (m = 0; m < l; ++m)
-            {
-                j = m << k;
-
-                /* The 256-value is a constant given as 1/4 of the size of
-                 * WebRtcSpl_kSinTable1024[], and should not be changed depending on the input
-                 * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
-                 */
-                wr = WebRtcSpl_kSinTable1024[j + 256];
-                wi = WebRtcSpl_kSinTable1024[j];
-
-                for (i = m; i < n; i += istep)
-                {
-                    j = i + l;
-
-                    tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j], 0)
-                            - WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j + 1], 0) + CIFFTRND),
-                            15 - CIFFTSFT);
-
-                    ti32 = WEBRTC_SPL_RSHIFT_W32(
-                                    (WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j + 1], 0)
-                                            + WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j], 0)
-                                            + CIFFTRND), 15 - CIFFTSFT);
-
-                    qr32 = ((WebRtc_Word32)frfi[2 * i]) << CIFFTSFT;
-                    qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CIFFTSFT;
-                    frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 - tr32+round2),
-                                                                       shift+CIFFTSFT);
-                    frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qi32 - ti32 + round2), shift + CIFFTSFT);
-                    frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 + tr32 + round2),
-                                                                       shift + CIFFTSFT);
-                    frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
-                            (qi32 + ti32 + round2), shift + CIFFTSFT);
-                }
-            }
-
-        }
-        --k;
-        l = istep;
-    }
-    return scale;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/copy_set_operations.c b/src/common_audio/signal_processing_library/main/source/copy_set_operations.c
deleted file mode 100644
index 8247337..0000000
--- a/src/common_audio/signal_processing_library/main/source/copy_set_operations.c
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the implementation of functions
- * WebRtcSpl_MemSetW16()
- * WebRtcSpl_MemSetW32()
- * WebRtcSpl_MemCpyReversedOrder()
- * WebRtcSpl_CopyFromEndW16()
- * WebRtcSpl_ZerosArrayW16()
- * WebRtcSpl_ZerosArrayW32()
- * WebRtcSpl_OnesArrayW16()
- * WebRtcSpl_OnesArrayW32()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include <string.h>
-#include "signal_processing_library.h"
-
-
-void WebRtcSpl_MemSetW16(WebRtc_Word16 *ptr, WebRtc_Word16 set_value, int length)
-{
-    int j;
-    WebRtc_Word16 *arrptr = ptr;
-
-    for (j = length; j > 0; j--)
-    {
-        *arrptr++ = set_value;
-    }
-}
-
-void WebRtcSpl_MemSetW32(WebRtc_Word32 *ptr, WebRtc_Word32 set_value, int length)
-{
-    int j;
-    WebRtc_Word32 *arrptr = ptr;
-
-    for (j = length; j > 0; j--)
-    {
-        *arrptr++ = set_value;
-    }
-}
-
-void WebRtcSpl_MemCpyReversedOrder(WebRtc_Word16* dest, WebRtc_Word16* source, int length)
-{
-    int j;
-    WebRtc_Word16* destPtr = dest;
-    WebRtc_Word16* sourcePtr = source;
-
-    for (j = 0; j < length; j++)
-    {
-        *destPtr-- = *sourcePtr++;
-    }
-}
-
-WebRtc_Word16 WebRtcSpl_CopyFromEndW16(G_CONST WebRtc_Word16 *vector_in,
-                                       WebRtc_Word16 length,
-                                       WebRtc_Word16 samples,
-                                       WebRtc_Word16 *vector_out)
-{
-    // Copy the last <samples> of the input vector to vector_out
-    WEBRTC_SPL_MEMCPY_W16(vector_out, &vector_in[length - samples], samples);
-
-    return samples;
-}
-
-WebRtc_Word16 WebRtcSpl_ZerosArrayW16(WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
-    WebRtcSpl_MemSetW16(vector, 0, length);
-    return length;
-}
-
-WebRtc_Word16 WebRtcSpl_ZerosArrayW32(WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
-    WebRtcSpl_MemSetW32(vector, 0, length);
-    return length;
-}
-
-WebRtc_Word16 WebRtcSpl_OnesArrayW16(WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 i;
-    WebRtc_Word16 *tmpvec = vector;
-    for (i = 0; i < length; i++)
-    {
-        *tmpvec++ = 1;
-    }
-    return length;
-}
-
-WebRtc_Word16 WebRtcSpl_OnesArrayW32(WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 i;
-    WebRtc_Word32 *tmpvec = vector;
-    for (i = 0; i < length; i++)
-    {
-        *tmpvec++ = 1;
-    }
-    return length;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/cos_table.c b/src/common_audio/signal_processing_library/main/source/cos_table.c
deleted file mode 100644
index 7dba4b0..0000000
--- a/src/common_audio/signal_processing_library/main/source/cos_table.c
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the 360 degree cos table.
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_kCosTable[] = {
-        8192,  8190,  8187,  8180,  8172,  8160,  8147,  8130,  8112,
-        8091,  8067,  8041,  8012,  7982,  7948,  7912,  7874,  7834,
-        7791,  7745,  7697,  7647,  7595,  7540,  7483,  7424,  7362,
-        7299,  7233,  7164,  7094,  7021,  6947,  6870,  6791,  6710,
-        6627,  6542,  6455,  6366,  6275,  6182,  6087,  5991,  5892,
-        5792,  5690,  5586,  5481,  5374,  5265,  5155,  5043,  4930,
-        4815,  4698,  4580,  4461,  4341,  4219,  4096,  3971,  3845,
-        3719,  3591,  3462,  3331,  3200,  3068,  2935,  2801,  2667,
-        2531,  2395,  2258,  2120,  1981,  1842,  1703,  1563,  1422,
-        1281,  1140,   998,   856,   713,   571,   428,   285,   142,
-           0,  -142,  -285,  -428,  -571,  -713,  -856,  -998, -1140,
-       -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395,
-       -2531, -2667, -2801, -2935, -3068, -3200, -3331, -3462, -3591,
-       -3719, -3845, -3971, -4095, -4219, -4341, -4461, -4580, -4698,
-       -4815, -4930, -5043, -5155, -5265, -5374, -5481, -5586, -5690,
-       -5792, -5892, -5991, -6087, -6182, -6275, -6366, -6455, -6542,
-       -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164, -7233,
-       -7299, -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745,
-       -7791, -7834, -7874, -7912, -7948, -7982, -8012, -8041, -8067,
-       -8091, -8112, -8130, -8147, -8160, -8172, -8180, -8187, -8190,
-       -8191, -8190, -8187, -8180, -8172, -8160, -8147, -8130, -8112,
-       -8091, -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
-       -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362,
-       -7299, -7233, -7164, -7094, -7021, -6947, -6870, -6791, -6710,
-       -6627, -6542, -6455, -6366, -6275, -6182, -6087, -5991, -5892,
-       -5792, -5690, -5586, -5481, -5374, -5265, -5155, -5043, -4930,
-       -4815, -4698, -4580, -4461, -4341, -4219, -4096, -3971, -3845,
-       -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801, -2667,
-       -2531, -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422,
-       -1281, -1140,  -998,  -856,  -713,  -571,  -428,  -285,  -142,
-           0,   142,   285,   428,   571,   713,   856,   998,  1140,
-        1281,  1422,  1563,  1703,  1842,  1981,  2120,  2258,  2395,
-        2531,  2667,  2801,  2935,  3068,  3200,  3331,  3462,  3591,
-        3719,  3845,  3971,  4095,  4219,  4341,  4461,  4580,  4698,
-        4815,  4930,  5043,  5155,  5265,  5374,  5481,  5586,  5690,
-        5792,  5892,  5991,  6087,  6182,  6275,  6366,  6455,  6542,
-        6627,  6710,  6791,  6870,  6947,  7021,  7094,  7164,  7233,
-        7299,  7362,  7424,  7483,  7540,  7595,  7647,  7697,  7745,
-        7791,  7834,  7874,  7912,  7948,  7982,  8012,  8041,  8067,
-        8091,  8112,  8130,  8147,  8160,  8172,  8180,  8187,  8190
-};
diff --git a/src/common_audio/signal_processing_library/main/source/cross_correlation.c b/src/common_audio/signal_processing_library/main/source/cross_correlation.c
deleted file mode 100644
index 1133d09..0000000
--- a/src/common_audio/signal_processing_library/main/source/cross_correlation.c
+++ /dev/null
@@ -1,267 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_CrossCorrelation().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_correlation, WebRtc_Word16* seq1,
-                                WebRtc_Word16* seq2, WebRtc_Word16 dim_seq,
-                                WebRtc_Word16 dim_cross_correlation,
-                                WebRtc_Word16 right_shifts,
-                                WebRtc_Word16 step_seq2)
-{
-    int i, j;
-    WebRtc_Word16* seq1Ptr;
-    WebRtc_Word16* seq2Ptr;
-    WebRtc_Word32* CrossCorrPtr;
-
-#ifdef _XSCALE_OPT_
-
-#ifdef _WIN32
-#pragma message("NOTE: _XSCALE_OPT_ optimizations are used (overrides _ARM_OPT_ and requires /QRxscale compiler flag)")
-#endif
-
-    __int64 macc40;
-
-    int iseq1[250];
-    int iseq2[250];
-    int iseq3[250];
-    int * iseq1Ptr;
-    int * iseq2Ptr;
-    int * iseq3Ptr;
-    int len, i_len;
-
-    seq1Ptr = seq1;
-    iseq1Ptr = iseq1;
-    for(i = 0; i < ((dim_seq + 1) >> 1); i++)
-    {
-        *iseq1Ptr = (unsigned short)*seq1Ptr++;
-        *iseq1Ptr++ |= (WebRtc_Word32)*seq1Ptr++ << 16;
-
-    }
-
-    if(dim_seq%2)
-    {
-        *(iseq1Ptr-1) &= 0x0000ffff;
-    }
-    *iseq1Ptr = 0;
-    iseq1Ptr++;
-    *iseq1Ptr = 0;
-    iseq1Ptr++;
-    *iseq1Ptr = 0;
-
-    if(step_seq2 < 0)
-    {
-        seq2Ptr = seq2 - dim_cross_correlation + 1;
-        CrossCorrPtr = &cross_correlation[dim_cross_correlation - 1];
-    }
-    else
-    {
-        seq2Ptr = seq2;
-        CrossCorrPtr = cross_correlation;
-    }
-
-    len = dim_seq + dim_cross_correlation - 1;
-    i_len = (len + 1) >> 1;
-    iseq2Ptr = iseq2;
-
-    iseq3Ptr = iseq3;
-    for(i = 0; i < i_len; i++)
-    {
-        *iseq2Ptr = (unsigned short)*seq2Ptr++;
-        *iseq3Ptr = (unsigned short)*seq2Ptr;
-        *iseq2Ptr++ |= (WebRtc_Word32)*seq2Ptr++ << 16;
-        *iseq3Ptr++ |= (WebRtc_Word32)*seq2Ptr << 16;
-    }
-
-    if(len % 2)
-    {
-        iseq2[i_len - 1] &= 0x0000ffff;
-        iseq3[i_len - 1] = 0;
-    }
-    else
-    iseq3[i_len - 1] &= 0x0000ffff;
-
-    iseq2[i_len] = 0;
-    iseq3[i_len] = 0;
-    iseq2[i_len + 1] = 0;
-    iseq3[i_len + 1] = 0;
-    iseq2[i_len + 2] = 0;
-    iseq3[i_len + 2] = 0;
-
-    // Set pointer to start value
-    iseq2Ptr = iseq2;
-    iseq3Ptr = iseq3;
-
-    i_len = (dim_seq + 7) >> 3;
-    for (i = 0; i < dim_cross_correlation; i++)
-    {
-
-        iseq1Ptr = iseq1;
-
-        macc40 = 0;
-
-        _WriteCoProcessor(macc40, 0);
-
-        if((i & 1))
-        {
-            iseq3Ptr = iseq3 + (i >> 1);
-            for (j = i_len; j > 0; j--)
-            {
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
-            }
-        }
-        else
-        {
-            iseq2Ptr = iseq2 + (i >> 1);
-            for (j = i_len; j > 0; j--)
-            {
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
-                _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
-            }
-
-        }
-
-        macc40 = _ReadCoProcessor(0);
-        *CrossCorrPtr = (WebRtc_Word32)(macc40 >> right_shifts);
-        CrossCorrPtr += step_seq2;
-    }
-#else // #ifdef _XSCALE_OPT_
-#ifdef _ARM_OPT_
-    WebRtc_Word16 dim_seq8 = (dim_seq >> 3) << 3;
-#endif
-
-    CrossCorrPtr = cross_correlation;
-
-    for (i = 0; i < dim_cross_correlation; i++)
-    {
-        // Set the pointer to the static vector, set the pointer to the sliding vector
-        // and initialize cross_correlation
-        seq1Ptr = seq1;
-        seq2Ptr = seq2 + (step_seq2 * i);
-        (*CrossCorrPtr) = 0;
-
-#ifndef _ARM_OPT_ 
-#ifdef _WIN32
-#pragma message("NOTE: default implementation is used")
-#endif
-        // Perform the cross correlation
-        for (j = 0; j < dim_seq; j++)
-        {
-            (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr), right_shifts);
-            seq1Ptr++;
-            seq2Ptr++;
-        }
-#else
-#ifdef _WIN32
-#pragma message("NOTE: _ARM_OPT_ optimizations are used")
-#endif
-        if (right_shifts == 0)
-        {
-            // Perform the optimized cross correlation
-            for (j = 0; j < dim_seq8; j = j + 8)
-            {
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-            }
-
-            for (j = dim_seq8; j < dim_seq; j++)
-            {
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
-                seq1Ptr++;
-                seq2Ptr++;
-            }
-        }
-        else // right_shifts != 0
-
-        {
-            // Perform the optimized cross correlation
-            for (j = 0; j < dim_seq8; j = j + 8)
-            {
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-            }
-
-            for (j = dim_seq8; j < dim_seq; j++)
-            {
-                (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
-                                                             right_shifts);
-                seq1Ptr++;
-                seq2Ptr++;
-            }
-        }
-#endif
-        CrossCorrPtr++;
-    }
-#endif
-}
diff --git a/src/common_audio/signal_processing_library/main/source/division_operations.c b/src/common_audio/signal_processing_library/main/source/division_operations.c
deleted file mode 100644
index b143373..0000000
--- a/src/common_audio/signal_processing_library/main/source/division_operations.c
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the divisions
- * WebRtcSpl_DivU32U16()
- * WebRtcSpl_DivW32W16()
- * WebRtcSpl_DivW32W16ResW16()
- * WebRtcSpl_DivResultInQ31()
- * WebRtcSpl_DivW32HiLow()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_UWord32 WebRtcSpl_DivU32U16(WebRtc_UWord32 num, WebRtc_UWord16 den)
-{
-    // Guard against division with 0
-    if (den != 0)
-    {
-        return (WebRtc_UWord32)(num / den);
-    } else
-    {
-        return (WebRtc_UWord32)0xFFFFFFFF;
-    }
-}
-
-WebRtc_Word32 WebRtcSpl_DivW32W16(WebRtc_Word32 num, WebRtc_Word16 den)
-{
-    // Guard against division with 0
-    if (den != 0)
-    {
-        return (WebRtc_Word32)(num / den);
-    } else
-    {
-        return (WebRtc_Word32)0x7FFFFFFF;
-    }
-}
-
-WebRtc_Word16 WebRtcSpl_DivW32W16ResW16(WebRtc_Word32 num, WebRtc_Word16 den)
-{
-    // Guard against division with 0
-    if (den != 0)
-    {
-        return (WebRtc_Word16)(num / den);
-    } else
-    {
-        return (WebRtc_Word16)0x7FFF;
-    }
-}
-
-WebRtc_Word32 WebRtcSpl_DivResultInQ31(WebRtc_Word32 num, WebRtc_Word32 den)
-{
-    WebRtc_Word32 L_num = num;
-    WebRtc_Word32 L_den = den;
-    WebRtc_Word32 div = 0;
-    int k = 31;
-    int change_sign = 0;
-
-    if (num == 0)
-        return 0;
-
-    if (num < 0)
-    {
-        change_sign++;
-        L_num = -num;
-    }
-    if (den < 0)
-    {
-        change_sign++;
-        L_den = -den;
-    }
-    while (k--)
-    {
-        div <<= 1;
-        L_num <<= 1;
-        if (L_num >= L_den)
-        {
-            L_num -= L_den;
-            div++;
-        }
-    }
-    if (change_sign == 1)
-    {
-        div = -div;
-    }
-    return div;
-}
-
-WebRtc_Word32 WebRtcSpl_DivW32HiLow(WebRtc_Word32 num, WebRtc_Word16 den_hi,
-                                    WebRtc_Word16 den_low)
-{
-    WebRtc_Word16 approx, tmp_hi, tmp_low, num_hi, num_low;
-    WebRtc_Word32 tmpW32;
-
-    approx = (WebRtc_Word16)WebRtcSpl_DivW32W16((WebRtc_Word32)0x1FFFFFFF, den_hi);
-    // result in Q14 (Note: 3FFFFFFF = 0.5 in Q30)
-
-    // tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30)
-    tmpW32 = (WEBRTC_SPL_MUL_16_16(den_hi, approx) << 1)
-            + ((WEBRTC_SPL_MUL_16_16(den_low, approx) >> 15) << 1);
-    // tmpW32 = den * approx
-
-    tmpW32 = (WebRtc_Word32)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx))
-
-    // Store tmpW32 in hi and low format
-    tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
-    tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
-    // tmpW32 = 1/den in Q29
-    tmpW32 = ((WEBRTC_SPL_MUL_16_16(tmp_hi, approx) + (WEBRTC_SPL_MUL_16_16(tmp_low, approx)
-            >> 15)) << 1);
-
-    // 1/den in hi and low format
-    tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
-    tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
-    // Store num in hi and low format
-    num_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(num, 16);
-    num_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((num
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)num_hi, 16)), 1);
-
-    // num * (1/den) by 32 bit multiplication (result in Q28)
-
-    tmpW32 = (WEBRTC_SPL_MUL_16_16(num_hi, tmp_hi) + (WEBRTC_SPL_MUL_16_16(num_hi, tmp_low)
-            >> 15) + (WEBRTC_SPL_MUL_16_16(num_low, tmp_hi) >> 15));
-
-    // Put result in Q31 (convert from Q28)
-    tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
-
-    return tmpW32;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/dot_product_with_scale.c b/src/common_audio/signal_processing_library/main/source/dot_product_with_scale.c
deleted file mode 100644
index 6e085fd..0000000
--- a/src/common_audio/signal_processing_library/main/source/dot_product_with_scale.c
+++ /dev/null
@@ -1,91 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_DotProductWithScale().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word32 WebRtcSpl_DotProductWithScale(WebRtc_Word16 *vector1, WebRtc_Word16 *vector2,
-                                            int length, int scaling)
-{
-    WebRtc_Word32 sum;
-    int i;
-#ifdef _ARM_OPT_
-#pragma message("NOTE: _ARM_OPT_ optimizations are used")
-    WebRtc_Word16 len4 = (length >> 2) << 2;
-#endif
-
-    sum = 0;
-
-#ifndef _ARM_OPT_
-    for (i = 0; i < length; i++)
-    {
-        sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1++, *vector2++, scaling);
-    }
-#else
-    if (scaling == 0)
-    {
-        for (i = 0; i < len4; i = i + 4)
-        {
-            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
-            vector1++;
-            vector2++;
-        }
-
-        for (i = len4; i < length; i++)
-        {
-            sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
-            vector1++;
-            vector2++;
-        }
-    }
-    else
-    {
-        for (i = 0; i < len4; i = i + 4)
-        {
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
-            vector1++;
-            vector2++;
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
-            vector1++;
-            vector2++;
-        }
-
-        for (i = len4; i < length; i++)
-        {
-            sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
-            vector1++;
-            vector2++;
-        }
-    }
-#endif
-
-    return sum;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/downsample_fast.c b/src/common_audio/signal_processing_library/main/source/downsample_fast.c
deleted file mode 100644
index 9338275..0000000
--- a/src/common_audio/signal_processing_library/main/source/downsample_fast.c
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_DownsampleFast().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-int WebRtcSpl_DownsampleFast(WebRtc_Word16 *in_ptr, WebRtc_Word16 in_length,
-                             WebRtc_Word16 *out_ptr, WebRtc_Word16 out_length,
-                             WebRtc_Word16 *B, WebRtc_Word16 B_length, WebRtc_Word16 factor,
-                             WebRtc_Word16 delay)
-{
-    WebRtc_Word32 o;
-    int i, j;
-
-    WebRtc_Word16 *downsampled_ptr = out_ptr;
-    WebRtc_Word16 *b_ptr;
-    WebRtc_Word16 *x_ptr;
-    WebRtc_Word16 endpos = delay
-            + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(factor, (out_length - 1)) + 1;
-
-    if (in_length < endpos)
-    {
-        return -1;
-    }
-
-    for (i = delay; i < endpos; i += factor)
-    {
-        b_ptr = &B[0];
-        x_ptr = &in_ptr[i];
-
-        o = (WebRtc_Word32)2048; // Round val
-
-        for (j = 0; j < B_length; j++)
-        {
-            o += WEBRTC_SPL_MUL_16_16(*b_ptr++, *x_ptr--);
-        }
-
-        o = WEBRTC_SPL_RSHIFT_W32(o, 12);
-
-        // If output is higher than 32768, saturate it. Same with negative side
-
-        *downsampled_ptr++ = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, o, -32768);
-    }
-
-    return 0;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/energy.c b/src/common_audio/signal_processing_library/main/source/energy.c
deleted file mode 100644
index e8fdf94..0000000
--- a/src/common_audio/signal_processing_library/main/source/energy.c
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_Energy().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word32 WebRtcSpl_Energy(WebRtc_Word16* vector, int vector_length, int* scale_factor)
-{
-    WebRtc_Word32 en = 0;
-    int i;
-    int scaling = WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length);
-    int looptimes = vector_length;
-    WebRtc_Word16 *vectorptr = vector;
-
-    for (i = 0; i < looptimes; i++)
-    {
-        en += WEBRTC_SPL_MUL_16_16_RSFT(*vectorptr, *vectorptr, scaling);
-        vectorptr++;
-    }
-    *scale_factor = scaling;
-
-    return en;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/filter_ar.c b/src/common_audio/signal_processing_library/main/source/filter_ar.c
deleted file mode 100644
index 30a56c1..0000000
--- a/src/common_audio/signal_processing_library/main/source/filter_ar.c
+++ /dev/null
@@ -1,92 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_FilterAR().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* a,
-                       int a_length,
-                       G_CONST WebRtc_Word16* x,
-                       int x_length,
-                       WebRtc_Word16* state,
-                       int state_length,
-                       WebRtc_Word16* state_low,
-                       int state_low_length,
-                       WebRtc_Word16* filtered,
-                       WebRtc_Word16* filtered_low,
-                       int filtered_low_length)
-{
-    WebRtc_Word32 o;
-    WebRtc_Word32 oLOW;
-    int i, j, stop;
-    G_CONST WebRtc_Word16* x_ptr = &x[0];
-    WebRtc_Word16* filteredFINAL_ptr = filtered;
-    WebRtc_Word16* filteredFINAL_LOW_ptr = filtered_low;
-
-    state_low_length = state_low_length;
-    filtered_low_length = filtered_low_length;
-
-    for (i = 0; i < x_length; i++)
-    {
-        // Calculate filtered[i] and filtered_low[i]
-        G_CONST WebRtc_Word16* a_ptr = &a[1];
-        WebRtc_Word16* filtered_ptr = &filtered[i - 1];
-        WebRtc_Word16* filtered_low_ptr = &filtered_low[i - 1];
-        WebRtc_Word16* state_ptr = &state[state_length - 1];
-        WebRtc_Word16* state_low_ptr = &state_low[state_length - 1];
-
-        o = (WebRtc_Word32)(*x_ptr++) << 12;
-        oLOW = (WebRtc_Word32)0;
-
-        stop = (i < a_length) ? i + 1 : a_length;
-        for (j = 1; j < stop; j++)
-        {
-            o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *filtered_ptr--);
-            oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *filtered_low_ptr--);
-        }
-        for (j = i + 1; j < a_length; j++)
-        {
-            o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *state_ptr--);
-            oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *state_low_ptr--);
-        }
-
-        o += (oLOW >> 12);
-        *filteredFINAL_ptr = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
-        *filteredFINAL_LOW_ptr++ = (WebRtc_Word16)(o - ((WebRtc_Word32)(*filteredFINAL_ptr++)
-                << 12));
-    }
-
-    // Save the filter state
-    if (x_length >= state_length)
-    {
-        WebRtcSpl_CopyFromEndW16(filtered, x_length, a_length - 1, state);
-        WebRtcSpl_CopyFromEndW16(filtered_low, x_length, a_length - 1, state_low);
-    } else
-    {
-        for (i = 0; i < state_length - x_length; i++)
-        {
-            state[i] = state[i + x_length];
-            state_low[i] = state_low[i + x_length];
-        }
-        for (i = 0; i < x_length; i++)
-        {
-            state[state_length - x_length + i] = filtered[i];
-            state[state_length - x_length + i] = filtered_low[i];
-        }
-    }
-
-    return x_length;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/filter_ar_fast_q12.c b/src/common_audio/signal_processing_library/main/source/filter_ar_fast_q12.c
deleted file mode 100644
index 6184da3..0000000
--- a/src/common_audio/signal_processing_library/main/source/filter_ar_fast_q12.c
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_FilterARFastQ12().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_FilterARFastQ12(WebRtc_Word16 *in, WebRtc_Word16 *out, WebRtc_Word16 *A,
-                               WebRtc_Word16 A_length, WebRtc_Word16 length)
-{
-    WebRtc_Word32 o;
-    int i, j;
-
-    WebRtc_Word16 *x_ptr = &in[0];
-    WebRtc_Word16 *filtered_ptr = &out[0];
-
-    for (i = 0; i < length; i++)
-    {
-        // Calculate filtered[i]
-        G_CONST WebRtc_Word16 *a_ptr = &A[0];
-        WebRtc_Word16 *state_ptr = &out[i - 1];
-
-        o = WEBRTC_SPL_MUL_16_16(*x_ptr++, *a_ptr++);
-
-        for (j = 1; j < A_length; j++)
-        {
-            o -= WEBRTC_SPL_MUL_16_16(*a_ptr++,*state_ptr--);
-        }
-
-        // Saturate the output
-        o = WEBRTC_SPL_SAT((WebRtc_Word32)134215679, o, (WebRtc_Word32)-134217728);
-
-        *filtered_ptr++ = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
-    }
-
-    return;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/filter_ma_fast_q12.c b/src/common_audio/signal_processing_library/main/source/filter_ma_fast_q12.c
deleted file mode 100644
index 19ad9b1..0000000
--- a/src/common_audio/signal_processing_library/main/source/filter_ma_fast_q12.c
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_FilterMAFastQ12().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_FilterMAFastQ12(WebRtc_Word16* in_ptr,
-                               WebRtc_Word16* out_ptr,
-                               WebRtc_Word16* B,
-                               WebRtc_Word16 B_length,
-                               WebRtc_Word16 length)
-{
-    WebRtc_Word32 o;
-    int i, j;
-    for (i = 0; i < length; i++)
-    {
-        G_CONST WebRtc_Word16* b_ptr = &B[0];
-        G_CONST WebRtc_Word16* x_ptr = &in_ptr[i];
-
-        o = (WebRtc_Word32)0;
-
-        for (j = 0; j < B_length; j++)
-        {
-            o += WEBRTC_SPL_MUL_16_16(*b_ptr++, *x_ptr--);
-        }
-
-        // If output is higher than 32768, saturate it. Same with negative side
-        // 2^27 = 134217728, which corresponds to 32768 in Q12
-
-        // Saturate the output
-        o = WEBRTC_SPL_SAT((WebRtc_Word32)134215679, o, (WebRtc_Word32)-134217728);
-
-        *out_ptr++ = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
-    }
-    return;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/get_hanning_window.c b/src/common_audio/signal_processing_library/main/source/get_hanning_window.c
deleted file mode 100644
index 2845c83..0000000
--- a/src/common_audio/signal_processing_library/main/source/get_hanning_window.c
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_GetHanningWindow().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_GetHanningWindow(WebRtc_Word16 *v, WebRtc_Word16 size)
-{
-    int jj;
-    WebRtc_Word16 *vptr1;
-
-    WebRtc_Word32 index;
-    WebRtc_Word32 factor = ((WebRtc_Word32)0x40000000);
-
-    factor = WebRtcSpl_DivW32W16(factor, size);
-    if (size < 513)
-        index = (WebRtc_Word32)-0x200000;
-    else
-        index = (WebRtc_Word32)-0x100000;
-    vptr1 = v;
-
-    for (jj = 0; jj < size; jj++)
-    {
-        index += factor;
-        (*vptr1++) = WebRtcSpl_kHanningTable[index >> 22];
-    }
-
-}
diff --git a/src/common_audio/signal_processing_library/main/source/get_scaling_square.c b/src/common_audio/signal_processing_library/main/source/get_scaling_square.c
deleted file mode 100644
index dccbf33..0000000
--- a/src/common_audio/signal_processing_library/main/source/get_scaling_square.c
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_GetScalingSquare().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-int WebRtcSpl_GetScalingSquare(WebRtc_Word16 *in_vector, int in_vector_length, int times)
-{
-    int nbits = WebRtcSpl_GetSizeInBits(times);
-    int i;
-    WebRtc_Word16 smax = -1;
-    WebRtc_Word16 sabs;
-    WebRtc_Word16 *sptr = in_vector;
-    int t;
-    int looptimes = in_vector_length;
-
-    for (i = looptimes; i > 0; i--)
-    {
-        sabs = (*sptr > 0 ? *sptr++ : -*sptr++);
-        smax = (sabs > smax ? sabs : smax);
-    }
-    t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
-
-    if (smax == 0)
-    {
-        return 0; // Since norm(0) returns 0
-    } else
-    {
-        return (t > nbits) ? 0 : nbits - t;
-    }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/get_size_in_bits.c b/src/common_audio/signal_processing_library/main/source/get_size_in_bits.c
deleted file mode 100644
index 53853f0..0000000
--- a/src/common_audio/signal_processing_library/main/source/get_size_in_bits.c
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_GetSizeInBits().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 value)
-{
-
-    int bits = 0;
-
-    // Fast binary search to find the number of bits used
-    if ((0xFFFF0000 & value))
-        bits = 16;
-    if ((0x0000FF00 & (value >> bits)))
-        bits += 8;
-    if ((0x000000F0 & (value >> bits)))
-        bits += 4;
-    if ((0x0000000C & (value >> bits)))
-        bits += 2;
-    if ((0x00000002 & (value >> bits)))
-        bits += 1;
-    if ((0x00000001 & (value >> bits)))
-        bits += 1;
-
-    return bits;
-}
-
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/hanning_table.c b/src/common_audio/signal_processing_library/main/source/hanning_table.c
deleted file mode 100644
index 112d0e5..0000000
--- a/src/common_audio/signal_processing_library/main/source/hanning_table.c
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the Hanning table with 256 entries.
- *
- */
-
-#include "signal_processing_library.h"
-
-// Hanning table with 256 entries
-WebRtc_Word16 WebRtcSpl_kHanningTable[] = {
-    1,      2,      6,     10,     15,     22,     30,     39,
-   50,     62,     75,     89,    104,    121,    138,    157,
-  178,    199,    222,    246,    271,    297,    324,    353,
-  383,    413,    446,    479,    513,    549,    586,    624,
-  663,    703,    744,    787,    830,    875,    920,    967,
- 1015,   1064,   1114,   1165,   1218,   1271,   1325,   1381,
- 1437,   1494,   1553,   1612,   1673,   1734,   1796,   1859,
- 1924,   1989,   2055,   2122,   2190,   2259,   2329,   2399,
- 2471,   2543,   2617,   2691,   2765,   2841,   2918,   2995,
- 3073,   3152,   3232,   3312,   3393,   3475,   3558,   3641,
- 3725,   3809,   3895,   3980,   4067,   4154,   4242,   4330,
- 4419,   4509,   4599,   4689,   4781,   4872,   4964,   5057,
- 5150,   5244,   5338,   5432,   5527,   5622,   5718,   5814,
- 5910,   6007,   6104,   6202,   6299,   6397,   6495,   6594,
- 6693,   6791,   6891,   6990,   7090,   7189,   7289,   7389,
- 7489,   7589,   7690,   7790,   7890,   7991,   8091,   8192,
- 8293,   8393,   8494,   8594,   8694,   8795,   8895,   8995,
- 9095,   9195,   9294,   9394,   9493,   9593,   9691,   9790,
- 9889,   9987,  10085,  10182,  10280,  10377,  10474,  10570,
-10666,  10762,  10857,  10952,  11046,  11140,  11234,  11327,
-11420,  11512,  11603,  11695,  11785,  11875,  11965,  12054,
-12142,  12230,  12317,  12404,  12489,  12575,  12659,  12743,
-12826,  12909,  12991,  13072,  13152,  13232,  13311,  13389,
-13466,  13543,  13619,  13693,  13767,  13841,  13913,  13985,
-14055,  14125,  14194,  14262,  14329,  14395,  14460,  14525,
-14588,  14650,  14711,  14772,  14831,  14890,  14947,  15003,
-15059,  15113,  15166,  15219,  15270,  15320,  15369,  15417,
-15464,  15509,  15554,  15597,  15640,  15681,  15721,  15760,
-15798,  15835,  15871,  15905,  15938,  15971,  16001,  16031,
-16060,  16087,  16113,  16138,  16162,  16185,  16206,  16227,
-16246,  16263,  16280,  16295,  16309,  16322,  16334,  16345,
-16354,  16362,  16369,  16374,  16378,  16382,  16383,  16384
-};
diff --git a/src/common_audio/signal_processing_library/main/source/ilbc_specific_functions.c b/src/common_audio/signal_processing_library/main/source/ilbc_specific_functions.c
deleted file mode 100644
index 5a9e577..0000000
--- a/src/common_audio/signal_processing_library/main/source/ilbc_specific_functions.c
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the iLBC specific functions
- * WebRtcSpl_ScaleAndAddVectorsWithRound()
- * WebRtcSpl_ReverseOrderMultArrayElements()
- * WebRtcSpl_ElementwiseVectorMult()
- * WebRtcSpl_AddVectorsAndShift()
- * WebRtcSpl_AddAffineVectorToVector()
- * WebRtcSpl_AffineTransformVector()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_ScaleAndAddVectorsWithRound(WebRtc_Word16 *vector1, WebRtc_Word16 scale1,
-                                           WebRtc_Word16 *vector2, WebRtc_Word16 scale2,
-                                           WebRtc_Word16 right_shifts, WebRtc_Word16 *out,
-                                           WebRtc_Word16 vector_length)
-{
-    int i;
-    WebRtc_Word16 roundVal;
-    roundVal = 1 << right_shifts;
-    roundVal = roundVal >> 1;
-    for (i = 0; i < vector_length; i++)
-    {
-        out[i] = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(vector1[i], scale1)
-                + WEBRTC_SPL_MUL_16_16(vector2[i], scale2) + roundVal) >> right_shifts);
-    }
-}
-
-void WebRtcSpl_ReverseOrderMultArrayElements(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in,
-                                             G_CONST WebRtc_Word16 *win,
-                                             WebRtc_Word16 vector_length,
-                                             WebRtc_Word16 right_shifts)
-{
-    int i;
-    WebRtc_Word16 *outptr = out;
-    G_CONST WebRtc_Word16 *inptr = in;
-    G_CONST WebRtc_Word16 *winptr = win;
-    for (i = 0; i < vector_length; i++)
-    {
-        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
-                                                               *winptr--, right_shifts);
-    }
-}
-
-void WebRtcSpl_ElementwiseVectorMult(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in,
-                                     G_CONST WebRtc_Word16 *win, WebRtc_Word16 vector_length,
-                                     WebRtc_Word16 right_shifts)
-{
-    int i;
-    WebRtc_Word16 *outptr = out;
-    G_CONST WebRtc_Word16 *inptr = in;
-    G_CONST WebRtc_Word16 *winptr = win;
-    for (i = 0; i < vector_length; i++)
-    {
-        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
-                                                               *winptr++, right_shifts);
-    }
-}
-
-void WebRtcSpl_AddVectorsAndShift(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in1,
-                                  G_CONST WebRtc_Word16 *in2, WebRtc_Word16 vector_length,
-                                  WebRtc_Word16 right_shifts)
-{
-    int i;
-    WebRtc_Word16 *outptr = out;
-    G_CONST WebRtc_Word16 *in1ptr = in1;
-    G_CONST WebRtc_Word16 *in2ptr = in2;
-    for (i = vector_length; i > 0; i--)
-    {
-        (*outptr++) = (WebRtc_Word16)(((*in1ptr++) + (*in2ptr++)) >> right_shifts);
-    }
-}
-
-void WebRtcSpl_AddAffineVectorToVector(WebRtc_Word16 *out, WebRtc_Word16 *in,
-                                       WebRtc_Word16 gain, WebRtc_Word32 add_constant,
-                                       WebRtc_Word16 right_shifts, int vector_length)
-{
-    WebRtc_Word16 *inPtr;
-    WebRtc_Word16 *outPtr;
-    int i;
-
-    inPtr = in;
-    outPtr = out;
-    for (i = 0; i < vector_length; i++)
-    {
-        (*outPtr++) += (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
-                + (WebRtc_Word32)add_constant) >> right_shifts);
-    }
-}
-
-void WebRtcSpl_AffineTransformVector(WebRtc_Word16 *out, WebRtc_Word16 *in,
-                                     WebRtc_Word16 gain, WebRtc_Word32 add_constant,
-                                     WebRtc_Word16 right_shifts, int vector_length)
-{
-    WebRtc_Word16 *inPtr;
-    WebRtc_Word16 *outPtr;
-    int i;
-
-    inPtr = in;
-    outPtr = out;
-    for (i = 0; i < vector_length; i++)
-    {
-        (*outPtr++) = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
-                + (WebRtc_Word32)add_constant) >> right_shifts);
-    }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/levinson_durbin.c b/src/common_audio/signal_processing_library/main/source/levinson_durbin.c
deleted file mode 100644
index 4e11cdb..0000000
--- a/src/common_audio/signal_processing_library/main/source/levinson_durbin.c
+++ /dev/null
@@ -1,259 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_LevinsonDurbin().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define SPL_LEVINSON_MAXORDER 20
-
-WebRtc_Word16 WebRtcSpl_LevinsonDurbin(WebRtc_Word32 *R, WebRtc_Word16 *A, WebRtc_Word16 *K,
-                                       WebRtc_Word16 order)
-{
-    WebRtc_Word16 i, j;
-    // Auto-correlation coefficients in high precision
-    WebRtc_Word16 R_hi[SPL_LEVINSON_MAXORDER + 1], R_low[SPL_LEVINSON_MAXORDER + 1];
-    // LPC coefficients in high precision
-    WebRtc_Word16 A_hi[SPL_LEVINSON_MAXORDER + 1], A_low[SPL_LEVINSON_MAXORDER + 1];
-    // LPC coefficients for next iteration
-    WebRtc_Word16 A_upd_hi[SPL_LEVINSON_MAXORDER + 1], A_upd_low[SPL_LEVINSON_MAXORDER + 1];
-    // Reflection coefficient in high precision
-    WebRtc_Word16 K_hi, K_low;
-    // Prediction gain Alpha in high precision and with scale factor
-    WebRtc_Word16 Alpha_hi, Alpha_low, Alpha_exp;
-    WebRtc_Word16 tmp_hi, tmp_low;
-    WebRtc_Word32 temp1W32, temp2W32, temp3W32;
-    WebRtc_Word16 norm;
-
-    // Normalize the autocorrelation R[0]...R[order+1]
-
-    norm = WebRtcSpl_NormW32(R[0]);
-
-    for (i = order; i >= 0; i--)
-    {
-        temp1W32 = WEBRTC_SPL_LSHIFT_W32(R[i], norm);
-        // Put R in hi and low format
-        R_hi[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-        R_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)), 1);
-    }
-
-    // K = A[1] = -R[1] / R[0]
-
-    temp2W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[1],16)
-            + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[1],1); // R[1] in Q31
-    temp3W32 = WEBRTC_SPL_ABS_W32(temp2W32); // abs R[1]
-    temp1W32 = WebRtcSpl_DivW32HiLow(temp3W32, R_hi[0], R_low[0]); // abs(R[1])/R[0] in Q31
-    // Put back the sign on R[1]
-    if (temp2W32 > 0)
-    {
-        temp1W32 = -temp1W32;
-    }
-
-    // Put K in hi and low format
-    K_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-    K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
-
-    // Store first reflection coefficient
-    K[0] = K_hi;
-
-    temp1W32 = WEBRTC_SPL_RSHIFT_W32(temp1W32, 4); // A[1] in Q27
-
-    // Put A[1] in hi and low format
-    A_hi[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-    A_low[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[1], 16)), 1);
-
-    // Alpha = R[0] * (1-K^2)
-
-    temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14) + WEBRTC_SPL_MUL_16_16(K_hi, K_hi))
-            << 1); // temp1W32 = k^2 in Q31
-
-    temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
-    temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; // temp1W32 = (1 - K[0]*K[0]) in Q31
-
-    // Store temp1W32 = 1 - K[0]*K[0] on hi and low format
-    tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-    tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
-    // Calculate Alpha in Q31
-    temp1W32 = ((WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_hi)
-            + (WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_low) >> 15)
-            + (WEBRTC_SPL_MUL_16_16(R_low[0], tmp_hi) >> 15)) << 1);
-
-    // Normalize Alpha and put it in hi and low format
-
-    Alpha_exp = WebRtcSpl_NormW32(temp1W32);
-    temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, Alpha_exp);
-    Alpha_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-    Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-            - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
-
-    // Perform the iterative calculations in the Levinson-Durbin algorithm
-
-    for (i = 2; i <= order; i++)
-    {
-        /*                    ----
-         temp1W32 =  R[i] + > R[j]*A[i-j]
-         /
-         ----
-         j=1..i-1
-         */
-
-        temp1W32 = 0;
-
-        for (j = 1; j < i; j++)
-        {
-            // temp1W32 is in Q31
-            temp1W32 += ((WEBRTC_SPL_MUL_16_16(R_hi[j], A_hi[i-j]) << 1)
-                    + (((WEBRTC_SPL_MUL_16_16(R_hi[j], A_low[i-j]) >> 15)
-                            + (WEBRTC_SPL_MUL_16_16(R_low[j], A_hi[i-j]) >> 15)) << 1));
-        }
-
-        temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4);
-        temp1W32 += (WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)
-                + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[i], 1));
-
-        // K = -temp1W32 / Alpha
-        temp2W32 = WEBRTC_SPL_ABS_W32(temp1W32); // abs(temp1W32)
-        temp3W32 = WebRtcSpl_DivW32HiLow(temp2W32, Alpha_hi, Alpha_low); // abs(temp1W32)/Alpha
-
-        // Put the sign of temp1W32 back again
-        if (temp1W32 > 0)
-        {
-            temp3W32 = -temp3W32;
-        }
-
-        // Use the Alpha shifts from earlier to de-normalize
-        norm = WebRtcSpl_NormW32(temp3W32);
-        if ((Alpha_exp <= norm) || (temp3W32 == 0))
-        {
-            temp3W32 = WEBRTC_SPL_LSHIFT_W32(temp3W32, Alpha_exp);
-        } else
-        {
-            if (temp3W32 > 0)
-            {
-                temp3W32 = (WebRtc_Word32)0x7fffffffL;
-            } else
-            {
-                temp3W32 = (WebRtc_Word32)0x80000000L;
-            }
-        }
-
-        // Put K on hi and low format
-        K_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
-        K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32
-                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
-
-        // Store Reflection coefficient in Q15
-        K[i - 1] = K_hi;
-
-        // Test for unstable filter.
-        // If unstable return 0 and let the user decide what to do in that case
-
-        if ((WebRtc_Word32)WEBRTC_SPL_ABS_W16(K_hi) > (WebRtc_Word32)32750)
-        {
-            return 0; // Unstable filter
-        }
-
-        /*
-         Compute updated LPC coefficient: Anew[i]
-         Anew[j]= A[j] + K*A[i-j]   for j=1..i-1
-         Anew[i]= K
-         */
-
-        for (j = 1; j < i; j++)
-        {
-            // temp1W32 = A[j] in Q27
-            temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[j],16)
-                    + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[j],1);
-
-            // temp1W32 += K*A[i-j] in Q27
-            temp1W32 += ((WEBRTC_SPL_MUL_16_16(K_hi, A_hi[i-j])
-                    + (WEBRTC_SPL_MUL_16_16(K_hi, A_low[i-j]) >> 15)
-                    + (WEBRTC_SPL_MUL_16_16(K_low, A_hi[i-j]) >> 15)) << 1);
-
-            // Put Anew in hi and low format
-            A_upd_hi[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-            A_upd_low[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-                    - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[j], 16)), 1);
-        }
-
-        // temp3W32 = K in Q27 (Convert from Q31 to Q27)
-        temp3W32 = WEBRTC_SPL_RSHIFT_W32(temp3W32, 4);
-
-        // Store Anew in hi and low format
-        A_upd_hi[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
-        A_upd_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32
-                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[i], 16)), 1);
-
-        // Alpha = Alpha * (1-K^2)
-
-        temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14)
-                + WEBRTC_SPL_MUL_16_16(K_hi, K_hi)) << 1); // K*K in Q31
-
-        temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
-        temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; // 1 - K*K  in Q31
-
-        // Convert 1- K^2 in hi and low format
-        tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-        tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
-        // Calculate Alpha = Alpha * (1-K^2) in Q31
-        temp1W32 = ((WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_hi)
-                + (WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_low) >> 15)
-                + (WEBRTC_SPL_MUL_16_16(Alpha_low, tmp_hi) >> 15)) << 1);
-
-        // Normalize Alpha and store it on hi and low format
-
-        norm = WebRtcSpl_NormW32(temp1W32);
-        temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, norm);
-
-        Alpha_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
-        Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
-                - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
-
-        // Update the total normalization of Alpha
-        Alpha_exp = Alpha_exp + norm;
-
-        // Update A[]
-
-        for (j = 1; j <= i; j++)
-        {
-            A_hi[j] = A_upd_hi[j];
-            A_low[j] = A_upd_low[j];
-        }
-    }
-
-    /*
-     Set A[0] to 1.0 and store the A[i] i=1...order in Q12
-     (Convert from Q27 and use rounding)
-     */
-
-    A[0] = 4096;
-
-    for (i = 1; i <= order; i++)
-    {
-        // temp1W32 in Q27
-        temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[i], 16)
-                + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[i], 1);
-        // Round and store upper word
-        A[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32<<1)+(WebRtc_Word32)32768, 16);
-    }
-    return 1; // Stable filters
-}
diff --git a/src/common_audio/signal_processing_library/main/source/lpc_to_refl_coef.c b/src/common_audio/signal_processing_library/main/source/lpc_to_refl_coef.c
deleted file mode 100644
index 2cb83c2..0000000
--- a/src/common_audio/signal_processing_library/main/source/lpc_to_refl_coef.c
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_LpcToReflCoef().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER 50
-
-void WebRtcSpl_LpcToReflCoef(WebRtc_Word16* a16, int use_order, WebRtc_Word16* k16)
-{
-    int m, k;
-    WebRtc_Word32 tmp32[SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER];
-    WebRtc_Word32 tmp_inv_denom32;
-    WebRtc_Word16 tmp_inv_denom16;
-
-    k16[use_order - 1] = WEBRTC_SPL_LSHIFT_W16(a16[use_order], 3); //Q12<<3 => Q15
-    for (m = use_order - 1; m > 0; m--)
-    {
-        // (1 - k^2) in Q30
-        tmp_inv_denom32 = ((WebRtc_Word32)1073741823) - WEBRTC_SPL_MUL_16_16(k16[m], k16[m]);
-        // (1 - k^2) in Q15
-        tmp_inv_denom16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp_inv_denom32, 15);
-
-        for (k = 1; k <= m; k++)
-        {
-            // tmp[k] = (a[k] - RC[m] * a[m-k+1]) / (1.0 - RC[m]*RC[m]);
-
-            // [Q12<<16 - (Q15*Q12)<<1] = [Q28 - Q28] = Q28
-            tmp32[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)a16[k], 16)
-                    - WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1);
-
-            tmp32[k] = WebRtcSpl_DivW32W16(tmp32[k], tmp_inv_denom16); //Q28/Q15 = Q13
-        }
-
-        for (k = 1; k < m; k++)
-        {
-            a16[k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32[k], 1); //Q13>>1 => Q12
-        }
-
-        tmp32[m] = WEBRTC_SPL_SAT(8191, tmp32[m], -8191);
-        k16[m - 1] = (WebRtc_Word16)WEBRTC_SPL_LSHIFT_W32(tmp32[m], 2); //Q13<<2 => Q15
-    }
-    return;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/min_max_operations.c b/src/common_audio/signal_processing_library/main/source/min_max_operations.c
deleted file mode 100644
index cf5e9a7..0000000
--- a/src/common_audio/signal_processing_library/main/source/min_max_operations.c
+++ /dev/null
@@ -1,305 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the implementation of functions
- * WebRtcSpl_MaxAbsValueW16()
- * WebRtcSpl_MaxAbsIndexW16()
- * WebRtcSpl_MaxAbsValueW32()
- * WebRtcSpl_MaxValueW16()
- * WebRtcSpl_MaxIndexW16()
- * WebRtcSpl_MaxValueW32()
- * WebRtcSpl_MaxIndexW32()
- * WebRtcSpl_MinValueW16()
- * WebRtcSpl_MinIndexW16()
- * WebRtcSpl_MinValueW32()
- * WebRtcSpl_MinIndexW32()
- *
- * The description header can be found in signal_processing_library.h.
- *
- */
-
-#include "signal_processing_library.h"
-
-// Maximum absolute value of word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word32 tempMax = 0;
-    WebRtc_Word32 absVal;
-    WebRtc_Word16 totMax;
-    int i;
-    G_CONST WebRtc_Word16 *tmpvector = vector;
-
-#ifdef _ARM_OPT_
-#pragma message("NOTE: _ARM_OPT_ optimizations are used")
-
-    WebRtc_Word16 len4 = (length >> 2) << 2;
-
-    for (i = 0; i < len4; i = i + 4)
-    {
-        absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
-        if (absVal > tempMax)
-        {
-            tempMax = absVal;
-        }
-        tmpvector++;
-        absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
-        if (absVal > tempMax)
-        {
-            tempMax = absVal;
-        }
-        tmpvector++;
-        absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
-        if (absVal > tempMax)
-        {
-            tempMax = absVal;
-        }
-        tmpvector++;
-        absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
-        if (absVal > tempMax)
-        {
-            tempMax = absVal;
-        }
-        tmpvector++;
-    }
-
-    for (i = len4; i < len; i++)
-    {
-        absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
-        if (absVal > tempMax)
-        {
-            tempMax = absVal;
-        }
-        tmpvector++;
-    }
-#else
-    for (i = 0; i < length; i++)
-    {
-        absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
-        if (absVal > tempMax)
-        {
-            tempMax = absVal;
-        }
-        tmpvector++;
-    }
-    totMax = (WebRtc_Word16)WEBRTC_SPL_MIN(tempMax, WEBRTC_SPL_WORD16_MAX);
-    return totMax;
-#endif
-}
-
-// Index of maximum absolute value in a  word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxAbsIndexW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 tempMax;
-    WebRtc_Word16 absTemp;
-    WebRtc_Word16 tempMaxIndex = 0;
-    WebRtc_Word16 i = 0;
-    G_CONST WebRtc_Word16 *tmpvector = vector;
-
-    tempMax = WEBRTC_SPL_ABS_W16(*tmpvector);
-    tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        absTemp = WEBRTC_SPL_ABS_W16(*tmpvector);
-        tmpvector++;
-        if (absTemp > tempMax)
-        {
-            tempMax = absTemp;
-            tempMaxIndex = i;
-        }
-    }
-    return tempMaxIndex;
-}
-
-// Maximum absolute value of word32 vector.
-WebRtc_Word32 WebRtcSpl_MaxAbsValueW32(G_CONST WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
-    WebRtc_UWord32 tempMax = 0;
-    WebRtc_UWord32 absVal;
-    WebRtc_Word32 retval;
-    int i;
-    G_CONST WebRtc_Word32 *tmpvector = vector;
-
-    for (i = 0; i < length; i++)
-    {
-        absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
-        if (absVal > tempMax)
-        {
-            tempMax = absVal;
-        }
-        tmpvector++;
-    }
-    retval = (WebRtc_Word32)(WEBRTC_SPL_MIN(tempMax, WEBRTC_SPL_WORD32_MAX));
-    return retval;
-}
-
-// Maximum value of word16 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word16 WebRtcSpl_MaxValueW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 tempMax;
-    WebRtc_Word16 i;
-    G_CONST WebRtc_Word16 *tmpvector = vector;
-
-    tempMax = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ > tempMax)
-            tempMax = vector[i];
-    }
-    return tempMax;
-}
-#else
-#pragma message(">> WebRtcSpl_MaxValueW16 is excluded from this build")
-#endif
-
-// Index of maximum value in a word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxIndexW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 tempMax;
-    WebRtc_Word16 tempMaxIndex = 0;
-    WebRtc_Word16 i = 0;
-    G_CONST WebRtc_Word16 *tmpvector = vector;
-
-    tempMax = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ > tempMax)
-        {
-            tempMax = vector[i];
-            tempMaxIndex = i;
-        }
-    }
-    return tempMaxIndex;
-}
-
-// Maximum value of word32 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word32 WebRtcSpl_MaxValueW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word32 tempMax;
-    WebRtc_Word16 i;
-    G_CONST WebRtc_Word32 *tmpvector = vector;
-
-    tempMax = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ > tempMax)
-            tempMax = vector[i];
-    }
-    return tempMax;
-}
-#else
-#pragma message(">> WebRtcSpl_MaxValueW32 is excluded from this build")
-#endif
-
-// Index of maximum value in a word32 vector.
-WebRtc_Word16 WebRtcSpl_MaxIndexW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word32 tempMax;
-    WebRtc_Word16 tempMaxIndex = 0;
-    WebRtc_Word16 i = 0;
-    G_CONST WebRtc_Word32 *tmpvector = vector;
-
-    tempMax = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ > tempMax)
-        {
-            tempMax = vector[i];
-            tempMaxIndex = i;
-        }
-    }
-    return tempMaxIndex;
-}
-
-// Minimum value of word16 vector.
-WebRtc_Word16 WebRtcSpl_MinValueW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 tempMin;
-    WebRtc_Word16 i;
-    G_CONST WebRtc_Word16 *tmpvector = vector;
-
-    // Find the minimum value
-    tempMin = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ < tempMin)
-            tempMin = (vector[i]);
-    }
-    return tempMin;
-}
-
-// Index of minimum value in a word16 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word16 WebRtcSpl_MinIndexW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word16 tempMin;
-    WebRtc_Word16 tempMinIndex = 0;
-    WebRtc_Word16 i = 0;
-    G_CONST WebRtc_Word16* tmpvector = vector;
-
-    // Find index of smallest value
-    tempMin = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ < tempMin)
-        {
-            tempMin = vector[i];
-            tempMinIndex = i;
-        }
-    }
-    return tempMinIndex;
-}
-#else
-#pragma message(">> WebRtcSpl_MinIndexW16 is excluded from this build")
-#endif
-
-// Minimum value of word32 vector.
-WebRtc_Word32 WebRtcSpl_MinValueW32(G_CONST WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
-    WebRtc_Word32 tempMin;
-    WebRtc_Word16 i;
-    G_CONST WebRtc_Word32 *tmpvector = vector;
-
-    // Find the minimum value
-    tempMin = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ < tempMin)
-            tempMin = (vector[i]);
-    }
-    return tempMin;
-}
-
-// Index of minimum value in a word32 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word16 WebRtcSpl_MinIndexW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
-{
-    WebRtc_Word32 tempMin;
-    WebRtc_Word16 tempMinIndex = 0;
-    WebRtc_Word16 i = 0;
-    G_CONST WebRtc_Word32 *tmpvector = vector;
-
-    // Find index of smallest value
-    tempMin = *tmpvector++;
-    for (i = 1; i < length; i++)
-    {
-        if (*tmpvector++ < tempMin)
-        {
-            tempMin = vector[i];
-            tempMinIndex = i;
-        }
-    }
-    return tempMinIndex;
-}
-#else
-#pragma message(">> WebRtcSpl_MinIndexW32 is excluded from this build")
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/norm_u32.c b/src/common_audio/signal_processing_library/main/source/norm_u32.c
deleted file mode 100644
index c903a64..0000000
--- a/src/common_audio/signal_processing_library/main/source/norm_u32.c
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_NormU32().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-int WebRtcSpl_NormU32(WebRtc_UWord32 value)
-{
-    int zeros = 0;
-
-    if (value == 0)
-        return 0;
-
-    if (!(0xFFFF0000 & value))
-        zeros = 16;
-    if (!(0xFF000000 & (value << zeros)))
-        zeros += 8;
-    if (!(0xF0000000 & (value << zeros)))
-        zeros += 4;
-    if (!(0xC0000000 & (value << zeros)))
-        zeros += 2;
-    if (!(0x80000000 & (value << zeros)))
-        zeros += 1;
-
-    return zeros;
-}
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/norm_w16.c b/src/common_audio/signal_processing_library/main/source/norm_w16.c
deleted file mode 100644
index be6711d..0000000
--- a/src/common_audio/signal_processing_library/main/source/norm_w16.c
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_NormW16().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-int WebRtcSpl_NormW16(WebRtc_Word16 value)
-{
-    int zeros = 0;
-
-    if (value <= 0)
-        value ^= 0xFFFF;
-
-    if ( !(0xFF80 & value))
-        zeros = 8;
-    if ( !(0xF800 & (value << zeros)))
-        zeros += 4;
-    if ( !(0xE000 & (value << zeros)))
-        zeros += 2;
-    if ( !(0xC000 & (value << zeros)))
-        zeros += 1;
-
-    return zeros;
-}
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/norm_w32.c b/src/common_audio/signal_processing_library/main/source/norm_w32.c
deleted file mode 100644
index d456335..0000000
--- a/src/common_audio/signal_processing_library/main/source/norm_w32.c
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_NormW32().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-int WebRtcSpl_NormW32(WebRtc_Word32 value)
-{
-    int zeros = 0;
-
-    if (value <= 0)
-        value ^= 0xFFFFFFFF;
-
-    // Fast binary search to determine the number of left shifts required to 32-bit normalize
-    // the value
-    if (!(0xFFFF8000 & value))
-        zeros = 16;
-    if (!(0xFF800000 & (value << zeros)))
-        zeros += 8;
-    if (!(0xF8000000 & (value << zeros)))
-        zeros += 4;
-    if (!(0xE0000000 & (value << zeros)))
-        zeros += 2;
-    if (!(0xC0000000 & (value << zeros)))
-        zeros += 1;
-
-    return zeros;
-}
-
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/randn_table.c b/src/common_audio/signal_processing_library/main/source/randn_table.c
deleted file mode 100644
index 734fa79..0000000
--- a/src/common_audio/signal_processing_library/main/source/randn_table.c
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * Table with 512 samples from a normal distribution with mean 1 and std 1
- * The values are shifted up 13 steps (multiplied by 8192)
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_kRandNTable[] =
-{
-    9178,    -7260,       40,    10189,     4894,    -3531,   -13779,    14764,
-   -4008,    -8884,    -8990,     1008,     7368,     5184,     3251,    -5817,
-   -9786,     5963,     1770,     8066,    -7135,    10772,    -2298,     1361,
-    6484,     2241,    -8633,      792,      199,    -3344,     6553,   -10079,
-  -15040,       95,    11608,   -12469,    14161,    -4176,     2476,     6403,
-   13685,   -16005,     6646,     2239,    10916,    -3004,     -602,    -3141,
-    2142,    14144,    -5829,     5305,     8209,     4713,     2697,    -5112,
-   16092,    -1210,    -2891,    -6631,    -5360,   -11878,    -6781,    -2739,
-   -6392,      536,    10923,    10872,     5059,    -4748,    -7770,     5477,
-      38,    -1025,    -2892,     1638,     6304,    14375,   -11028,     1553,
-   -1565,    10762,     -393,     4040,     5257,    12310,     6554,    -4799,
-    4899,    -6354,     1603,    -1048,    -2220,     8247,     -186,    -8944,
-  -12004,     2332,     4801,    -4933,     6371,      131,     8614,    -5927,
-   -8287,   -22760,     4033,   -15162,     3385,     3246,     3153,    -5250,
-    3766,      784,     6494,      -62,     3531,    -1582,    15572,      662,
-   -3952,     -330,    -3196,      669,     7236,    -2678,    -6569,    23319,
-   -8645,     -741,    14830,   -15976,     4903,      315,   -11342,    10311,
-    1858,    -7777,     2145,     5436,     5677,     -113,   -10033,      826,
-   -1353,    17210,     7768,      986,    -1471,     8291,    -4982,     8207,
-  -14911,    -6255,    -2449,   -11881,    -7059,   -11703,    -4338,     8025,
-    7538,    -2823,   -12490,     9470,    -1613,    -2529,   -10092,    -7807,
-    9480,     6970,   -12844,     5123,     3532,     4816,     4803,    -8455,
-   -5045,    14032,    -4378,    -1643,     5756,   -11041,    -2732,   -16618,
-   -6430,   -18375,    -3320,     6098,     5131,    -4269,    -8840,     2482,
-   -7048,     1547,   -21890,    -6505,    -7414,     -424,   -11722,     7955,
-    1653,   -17299,     1823,      473,    -9232,     3337,     1111,      873,
-    4018,    -8982,     9889,     3531,   -11763,    -3799,     7373,    -4539,
-    3231,     7054,    -8537,     7616,     6244,    16635,      447,    -2915,
-   13967,      705,    -2669,    -1520,    -1771,   -16188,     5956,     5117,
-    6371,    -9936,    -1448,     2480,     5128,     7550,    -8130,     5236,
-    8213,    -6443,     7707,    -1950,   -13811,     7218,     7031,    -3883,
-      67,     5731,    -2874,    13480,    -3743,     9298,    -3280,     3552,
-   -4425,      -18,    -3785,    -9988,    -5357,     5477,   -11794,     2117,
-    1416,    -9935,     3376,      802,    -5079,    -8243,    12652,       66,
-    3653,    -2368,     6781,   -21895,    -7227,     2487,     7839,     -385,
-    6646,    -7016,    -4658,     5531,    -1705,      834,      129,     3694,
-   -1343,     2238,   -22640,    -6417,   -11139,    11301,    -2945,    -3494,
-   -5626,      185,    -3615,    -2041,    -7972,    -3106,      -60,   -23497,
-   -1566,    17064,     3519,     2518,      304,    -6805,   -10269,     2105,
-    1936,     -426,     -736,    -8122,    -1467,     4238,    -6939,   -13309,
-     360,     7402,    -7970,    12576,     3287,    12194,    -6289,   -16006,
-    9171,     4042,    -9193,     9123,    -2512,     6388,    -4734,    -8739,
-    1028,    -5406,    -1696,     5889,     -666,    -4736,     4971,     3565,
-    9362,    -6292,     3876,    -3652,   -19666,     7523,    -4061,      391,
-  -11773,     7502,    -3763,     4929,    -9478,    13278,     2805,     4496,
-    7814,    16419,    12455,   -14773,     2127,    -2746,     3763,     4847,
-    3698,     6978,     4751,    -6957,    -3581,      -45,     6252,     1513,
-   -4797,    -7925,    11270,    16188,    -2359,    -5269,     9376,   -10777,
-    7262,    20031,    -6515,    -2208,    -5353,     8085,    -1341,    -1303,
-    7333,     5576,     3625,     5763,    -7931,     9833,    -3371,   -10305,
-    6534,   -13539,    -9971,      997,     8464,    -4064,    -1495,     1857,
-   13624,     5458,     9490,   -11086,    -4524,    12022,     -550,     -198,
-     408,    -8455,    -7068,    10289,     9712,    -3366,     9028,    -7621,
-   -5243,     2362,     6909,     4672,    -4933,    -1799,     4709,    -4563,
-     -62,     -566,     1624,    -7010,    14730,   -17791,    -3697,    -2344,
-   -1741,     7099,    -9509,    -6855,    -1989,     3495,    -2289,     2031,
-   12784,      891,    14189,    -3963,    -5683,      421,   -12575,     1724,
-  -12682,    -5970,    -8169,     3143,    -1824,    -5488,    -5130,     8536,
-   12799,      794,     5738,     3459,   -11689,     -258,    -3738,    -3775,
-   -8742,     2333,     8312,    -9383,    10331,    13119,     8398,    10644,
-  -19433,    -6446,   -16277,   -11793,    16284,     9345,    15222,    15834,
-    2009,    -7349,      130,   -14547,      338,    -5998,     3337,    21492,
-    2406,     7703,     -951,    11196,     -564,     3406,     2217,     4806,
-    2374,    -5797,    11839,     8940,   -11874,    18213,     2855,    10492
-};
diff --git a/src/common_audio/signal_processing_library/main/source/randomization_functions.c b/src/common_audio/signal_processing_library/main/source/randomization_functions.c
deleted file mode 100644
index 6bc87c7..0000000
--- a/src/common_audio/signal_processing_library/main/source/randomization_functions.c
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the randomization functions
- * WebRtcSpl_IncreaseSeed()
- * WebRtcSpl_RandU()
- * WebRtcSpl_RandN()
- * WebRtcSpl_RandUArray()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32 *seed)
-{
-    seed[0] = (seed[0] * ((WebRtc_Word32)69069) + 1) & (WEBRTC_SPL_MAX_SEED_USED - 1);
-    return seed[0];
-}
-
-WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32 *seed)
-{
-    return (WebRtc_Word16)(WebRtcSpl_IncreaseSeed(seed) >> 16);
-}
-
-WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32 *seed)
-{
-    return WebRtcSpl_kRandNTable[WebRtcSpl_IncreaseSeed(seed) >> 23];
-}
-
-// Creates an array of uniformly distributed variables
-WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
-                                   WebRtc_Word16 vector_length,
-                                   WebRtc_UWord32* seed)
-{
-    int i;
-    for (i = 0; i < vector_length; i++)
-    {
-        vector[i] = WebRtcSpl_RandU(seed);
-    }
-    return vector_length;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/refl_coef_to_lpc.c b/src/common_audio/signal_processing_library/main/source/refl_coef_to_lpc.c
deleted file mode 100644
index d07804d..0000000
--- a/src/common_audio/signal_processing_library/main/source/refl_coef_to_lpc.c
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_ReflCoefToLpc().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_ReflCoefToLpc(G_CONST WebRtc_Word16 *k, int use_order, WebRtc_Word16 *a)
-{
-    WebRtc_Word16 any[WEBRTC_SPL_MAX_LPC_ORDER + 1];
-    WebRtc_Word16 *aptr, *aptr2, *anyptr;
-    G_CONST WebRtc_Word16 *kptr;
-    int m, i;
-
-    kptr = k;
-    *a = 4096; // i.e., (Word16_MAX >> 3)+1.
-    *any = *a;
-    a[1] = WEBRTC_SPL_RSHIFT_W16((*k), 3);
-
-    for (m = 1; m < use_order; m++)
-    {
-        kptr++;
-        aptr = a;
-        aptr++;
-        aptr2 = &a[m];
-        anyptr = any;
-        anyptr++;
-
-        any[m + 1] = WEBRTC_SPL_RSHIFT_W16((*kptr), 3);
-        for (i = 0; i < m; i++)
-        {
-            *anyptr = (*aptr)
-                    + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((*aptr2), (*kptr), 15);
-            anyptr++;
-            aptr++;
-            aptr2--;
-        }
-
-        aptr = a;
-        anyptr = any;
-        for (i = 0; i < (m + 2); i++)
-        {
-            *aptr = *anyptr;
-            aptr++;
-            anyptr++;
-        }
-    }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/resample.c b/src/common_audio/signal_processing_library/main/source/resample.c
deleted file mode 100644
index 19d1778..0000000
--- a/src/common_audio/signal_processing_library/main/source/resample.c
+++ /dev/null
@@ -1,505 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the resampling functions for 22 kHz.
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-#include "resample_by_2_internal.h"
-
-// Declaration of internally used functions
-static void WebRtcSpl_32khzTo22khzIntToShort(const WebRtc_Word32 *In, WebRtc_Word16 *Out,
-                                             const WebRtc_Word32 K);
-
-void WebRtcSpl_32khzTo22khzIntToInt(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
-                                    const WebRtc_Word32 K);
-
-// interpolation coefficients
-static const WebRtc_Word16 kCoefficients32To22[5][9] = {
-        {127, -712,  2359, -6333, 23456, 16775, -3695,  945, -154},
-        {-39,  230,  -830,  2785, 32366, -2324,   760, -218,   38},
-        {117, -663,  2222, -6133, 26634, 13070, -3174,  831, -137},
-        {-77,  457, -1677,  5958, 31175, -4136,  1405, -408,   71},
-        { 98, -560,  1900, -5406, 29240,  9423, -2480,  663, -110}
-};
-
-//////////////////////
-// 22 kHz -> 16 kHz //
-//////////////////////
-
-// number of subblocks; options: 1, 2, 4, 5, 10
-#define SUB_BLOCKS_22_16    5
-
-// 22 -> 16 resampler
-void WebRtcSpl_Resample22khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                    WebRtcSpl_State22khzTo16khz* state, WebRtc_Word32* tmpmem)
-{
-    int k;
-
-    // process two blocks of 10/SUB_BLOCKS_22_16 ms (to reduce temp buffer size)
-    for (k = 0; k < SUB_BLOCKS_22_16; k++)
-    {
-        ///// 22 --> 44 /////
-        // WebRtc_Word16  in[220/SUB_BLOCKS_22_16]
-        // WebRtc_Word32 out[440/SUB_BLOCKS_22_16]
-        /////
-        WebRtcSpl_UpBy2ShortToInt(in, 220 / SUB_BLOCKS_22_16, tmpmem + 16, state->S_22_44);
-
-        ///// 44 --> 32 /////
-        // WebRtc_Word32  in[440/SUB_BLOCKS_22_16]
-        // WebRtc_Word32 out[320/SUB_BLOCKS_22_16]
-        /////
-        // copy state to and from input array
-        tmpmem[8] = state->S_44_32[0];
-        tmpmem[9] = state->S_44_32[1];
-        tmpmem[10] = state->S_44_32[2];
-        tmpmem[11] = state->S_44_32[3];
-        tmpmem[12] = state->S_44_32[4];
-        tmpmem[13] = state->S_44_32[5];
-        tmpmem[14] = state->S_44_32[6];
-        tmpmem[15] = state->S_44_32[7];
-        state->S_44_32[0] = tmpmem[440 / SUB_BLOCKS_22_16 + 8];
-        state->S_44_32[1] = tmpmem[440 / SUB_BLOCKS_22_16 + 9];
-        state->S_44_32[2] = tmpmem[440 / SUB_BLOCKS_22_16 + 10];
-        state->S_44_32[3] = tmpmem[440 / SUB_BLOCKS_22_16 + 11];
-        state->S_44_32[4] = tmpmem[440 / SUB_BLOCKS_22_16 + 12];
-        state->S_44_32[5] = tmpmem[440 / SUB_BLOCKS_22_16 + 13];
-        state->S_44_32[6] = tmpmem[440 / SUB_BLOCKS_22_16 + 14];
-        state->S_44_32[7] = tmpmem[440 / SUB_BLOCKS_22_16 + 15];
-
-        WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 40 / SUB_BLOCKS_22_16);
-
-        ///// 32 --> 16 /////
-        // WebRtc_Word32  in[320/SUB_BLOCKS_22_16]
-        // WebRtc_Word32 out[160/SUB_BLOCKS_22_16]
-        /////
-        WebRtcSpl_DownBy2IntToShort(tmpmem, 320 / SUB_BLOCKS_22_16, out, state->S_32_16);
-
-        // move input/output pointers 10/SUB_BLOCKS_22_16 ms seconds ahead
-        in += 220 / SUB_BLOCKS_22_16;
-        out += 160 / SUB_BLOCKS_22_16;
-    }
-}
-
-// initialize state of 22 -> 16 resampler
-void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state)
-{
-    int k;
-    for (k = 0; k < 8; k++)
-    {
-        state->S_22_44[k] = 0;
-        state->S_44_32[k] = 0;
-        state->S_32_16[k] = 0;
-    }
-}
-
-//////////////////////
-// 16 kHz -> 22 kHz //
-//////////////////////
-
-// number of subblocks; options: 1, 2, 4, 5, 10
-#define SUB_BLOCKS_16_22    4
-
-// 16 -> 22 resampler
-void WebRtcSpl_Resample16khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                    WebRtcSpl_State16khzTo22khz* state, WebRtc_Word32* tmpmem)
-{
-    int k;
-
-    // process two blocks of 10/SUB_BLOCKS_16_22 ms (to reduce temp buffer size)
-    for (k = 0; k < SUB_BLOCKS_16_22; k++)
-    {
-        ///// 16 --> 32 /////
-        // WebRtc_Word16  in[160/SUB_BLOCKS_16_22]
-        // WebRtc_Word32 out[320/SUB_BLOCKS_16_22]
-        /////
-        WebRtcSpl_UpBy2ShortToInt(in, 160 / SUB_BLOCKS_16_22, tmpmem + 8, state->S_16_32);
-
-        ///// 32 --> 22 /////
-        // WebRtc_Word32  in[320/SUB_BLOCKS_16_22]
-        // WebRtc_Word32 out[220/SUB_BLOCKS_16_22]
-        /////
-        // copy state to and from input array
-        tmpmem[0] = state->S_32_22[0];
-        tmpmem[1] = state->S_32_22[1];
-        tmpmem[2] = state->S_32_22[2];
-        tmpmem[3] = state->S_32_22[3];
-        tmpmem[4] = state->S_32_22[4];
-        tmpmem[5] = state->S_32_22[5];
-        tmpmem[6] = state->S_32_22[6];
-        tmpmem[7] = state->S_32_22[7];
-        state->S_32_22[0] = tmpmem[320 / SUB_BLOCKS_16_22];
-        state->S_32_22[1] = tmpmem[320 / SUB_BLOCKS_16_22 + 1];
-        state->S_32_22[2] = tmpmem[320 / SUB_BLOCKS_16_22 + 2];
-        state->S_32_22[3] = tmpmem[320 / SUB_BLOCKS_16_22 + 3];
-        state->S_32_22[4] = tmpmem[320 / SUB_BLOCKS_16_22 + 4];
-        state->S_32_22[5] = tmpmem[320 / SUB_BLOCKS_16_22 + 5];
-        state->S_32_22[6] = tmpmem[320 / SUB_BLOCKS_16_22 + 6];
-        state->S_32_22[7] = tmpmem[320 / SUB_BLOCKS_16_22 + 7];
-
-        WebRtcSpl_32khzTo22khzIntToShort(tmpmem, out, 20 / SUB_BLOCKS_16_22);
-
-        // move input/output pointers 10/SUB_BLOCKS_16_22 ms seconds ahead
-        in += 160 / SUB_BLOCKS_16_22;
-        out += 220 / SUB_BLOCKS_16_22;
-    }
-}
-
-// initialize state of 16 -> 22 resampler
-void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state)
-{
-    int k;
-    for (k = 0; k < 8; k++)
-    {
-        state->S_16_32[k] = 0;
-        state->S_32_22[k] = 0;
-    }
-}
-
-//////////////////////
-// 22 kHz ->  8 kHz //
-//////////////////////
-
-// number of subblocks; options: 1, 2, 5, 10
-#define SUB_BLOCKS_22_8     2
-
-// 22 -> 8 resampler
-void WebRtcSpl_Resample22khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                   WebRtcSpl_State22khzTo8khz* state, WebRtc_Word32* tmpmem)
-{
-    int k;
-
-    // process two blocks of 10/SUB_BLOCKS_22_8 ms (to reduce temp buffer size)
-    for (k = 0; k < SUB_BLOCKS_22_8; k++)
-    {
-        ///// 22 --> 22 lowpass /////
-        // WebRtc_Word16  in[220/SUB_BLOCKS_22_8]
-        // WebRtc_Word32 out[220/SUB_BLOCKS_22_8]
-        /////
-        WebRtcSpl_LPBy2ShortToInt(in, 220 / SUB_BLOCKS_22_8, tmpmem + 16, state->S_22_22);
-
-        ///// 22 --> 16 /////
-        // WebRtc_Word32  in[220/SUB_BLOCKS_22_8]
-        // WebRtc_Word32 out[160/SUB_BLOCKS_22_8]
-        /////
-        // copy state to and from input array
-        tmpmem[8] = state->S_22_16[0];
-        tmpmem[9] = state->S_22_16[1];
-        tmpmem[10] = state->S_22_16[2];
-        tmpmem[11] = state->S_22_16[3];
-        tmpmem[12] = state->S_22_16[4];
-        tmpmem[13] = state->S_22_16[5];
-        tmpmem[14] = state->S_22_16[6];
-        tmpmem[15] = state->S_22_16[7];
-        state->S_22_16[0] = tmpmem[220 / SUB_BLOCKS_22_8 + 8];
-        state->S_22_16[1] = tmpmem[220 / SUB_BLOCKS_22_8 + 9];
-        state->S_22_16[2] = tmpmem[220 / SUB_BLOCKS_22_8 + 10];
-        state->S_22_16[3] = tmpmem[220 / SUB_BLOCKS_22_8 + 11];
-        state->S_22_16[4] = tmpmem[220 / SUB_BLOCKS_22_8 + 12];
-        state->S_22_16[5] = tmpmem[220 / SUB_BLOCKS_22_8 + 13];
-        state->S_22_16[6] = tmpmem[220 / SUB_BLOCKS_22_8 + 14];
-        state->S_22_16[7] = tmpmem[220 / SUB_BLOCKS_22_8 + 15];
-
-        WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 20 / SUB_BLOCKS_22_8);
-
-        ///// 16 --> 8 /////
-        // WebRtc_Word32 in[160/SUB_BLOCKS_22_8]
-        // WebRtc_Word32 out[80/SUB_BLOCKS_22_8]
-        /////
-        WebRtcSpl_DownBy2IntToShort(tmpmem, 160 / SUB_BLOCKS_22_8, out, state->S_16_8);
-
-        // move input/output pointers 10/SUB_BLOCKS_22_8 ms seconds ahead
-        in += 220 / SUB_BLOCKS_22_8;
-        out += 80 / SUB_BLOCKS_22_8;
-    }
-}
-
-// initialize state of 22 -> 8 resampler
-void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state)
-{
-    int k;
-    for (k = 0; k < 8; k++)
-    {
-        state->S_22_22[k] = 0;
-        state->S_22_22[k + 8] = 0;
-        state->S_22_16[k] = 0;
-        state->S_16_8[k] = 0;
-    }
-}
-
-//////////////////////
-//  8 kHz -> 22 kHz //
-//////////////////////
-
-// number of subblocks; options: 1, 2, 5, 10
-#define SUB_BLOCKS_8_22     2
-
-// 8 -> 22 resampler
-void WebRtcSpl_Resample8khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                   WebRtcSpl_State8khzTo22khz* state, WebRtc_Word32* tmpmem)
-{
-    int k;
-
-    // process two blocks of 10/SUB_BLOCKS_8_22 ms (to reduce temp buffer size)
-    for (k = 0; k < SUB_BLOCKS_8_22; k++)
-    {
-        ///// 8 --> 16 /////
-        // WebRtc_Word16  in[80/SUB_BLOCKS_8_22]
-        // WebRtc_Word32 out[160/SUB_BLOCKS_8_22]
-        /////
-        WebRtcSpl_UpBy2ShortToInt(in, 80 / SUB_BLOCKS_8_22, tmpmem + 18, state->S_8_16);
-
-        ///// 16 --> 11 /////
-        // WebRtc_Word32  in[160/SUB_BLOCKS_8_22]
-        // WebRtc_Word32 out[110/SUB_BLOCKS_8_22]
-        /////
-        // copy state to and from input array
-        tmpmem[10] = state->S_16_11[0];
-        tmpmem[11] = state->S_16_11[1];
-        tmpmem[12] = state->S_16_11[2];
-        tmpmem[13] = state->S_16_11[3];
-        tmpmem[14] = state->S_16_11[4];
-        tmpmem[15] = state->S_16_11[5];
-        tmpmem[16] = state->S_16_11[6];
-        tmpmem[17] = state->S_16_11[7];
-        state->S_16_11[0] = tmpmem[160 / SUB_BLOCKS_8_22 + 10];
-        state->S_16_11[1] = tmpmem[160 / SUB_BLOCKS_8_22 + 11];
-        state->S_16_11[2] = tmpmem[160 / SUB_BLOCKS_8_22 + 12];
-        state->S_16_11[3] = tmpmem[160 / SUB_BLOCKS_8_22 + 13];
-        state->S_16_11[4] = tmpmem[160 / SUB_BLOCKS_8_22 + 14];
-        state->S_16_11[5] = tmpmem[160 / SUB_BLOCKS_8_22 + 15];
-        state->S_16_11[6] = tmpmem[160 / SUB_BLOCKS_8_22 + 16];
-        state->S_16_11[7] = tmpmem[160 / SUB_BLOCKS_8_22 + 17];
-
-        WebRtcSpl_32khzTo22khzIntToInt(tmpmem + 10, tmpmem, 10 / SUB_BLOCKS_8_22);
-
-        ///// 11 --> 22 /////
-        // WebRtc_Word32  in[110/SUB_BLOCKS_8_22]
-        // WebRtc_Word16 out[220/SUB_BLOCKS_8_22]
-        /////
-        WebRtcSpl_UpBy2IntToShort(tmpmem, 110 / SUB_BLOCKS_8_22, out, state->S_11_22);
-
-        // move input/output pointers 10/SUB_BLOCKS_8_22 ms seconds ahead
-        in += 80 / SUB_BLOCKS_8_22;
-        out += 220 / SUB_BLOCKS_8_22;
-    }
-}
-
-// initialize state of 8 -> 22 resampler
-void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state)
-{
-    int k;
-    for (k = 0; k < 8; k++)
-    {
-        state->S_8_16[k] = 0;
-        state->S_16_11[k] = 0;
-        state->S_11_22[k] = 0;
-    }
-}
-
-// compute two inner-products and store them to output array
-static void WebRtcSpl_DotProdIntToInt(const WebRtc_Word32* in1, const WebRtc_Word32* in2,
-                                      const WebRtc_Word16* coef_ptr, WebRtc_Word32* out1,
-                                      WebRtc_Word32* out2)
-{
-    WebRtc_Word32 tmp1 = 16384;
-    WebRtc_Word32 tmp2 = 16384;
-    WebRtc_Word16 coef;
-
-    coef = coef_ptr[0];
-    tmp1 += coef * in1[0];
-    tmp2 += coef * in2[-0];
-
-    coef = coef_ptr[1];
-    tmp1 += coef * in1[1];
-    tmp2 += coef * in2[-1];
-
-    coef = coef_ptr[2];
-    tmp1 += coef * in1[2];
-    tmp2 += coef * in2[-2];
-
-    coef = coef_ptr[3];
-    tmp1 += coef * in1[3];
-    tmp2 += coef * in2[-3];
-
-    coef = coef_ptr[4];
-    tmp1 += coef * in1[4];
-    tmp2 += coef * in2[-4];
-
-    coef = coef_ptr[5];
-    tmp1 += coef * in1[5];
-    tmp2 += coef * in2[-5];
-
-    coef = coef_ptr[6];
-    tmp1 += coef * in1[6];
-    tmp2 += coef * in2[-6];
-
-    coef = coef_ptr[7];
-    tmp1 += coef * in1[7];
-    tmp2 += coef * in2[-7];
-
-    coef = coef_ptr[8];
-    *out1 = tmp1 + coef * in1[8];
-    *out2 = tmp2 + coef * in2[-8];
-}
-
-// compute two inner-products and store them to output array
-static void WebRtcSpl_DotProdIntToShort(const WebRtc_Word32* in1, const WebRtc_Word32* in2,
-                                        const WebRtc_Word16* coef_ptr, WebRtc_Word16* out1,
-                                        WebRtc_Word16* out2)
-{
-    WebRtc_Word32 tmp1 = 16384;
-    WebRtc_Word32 tmp2 = 16384;
-    WebRtc_Word16 coef;
-
-    coef = coef_ptr[0];
-    tmp1 += coef * in1[0];
-    tmp2 += coef * in2[-0];
-
-    coef = coef_ptr[1];
-    tmp1 += coef * in1[1];
-    tmp2 += coef * in2[-1];
-
-    coef = coef_ptr[2];
-    tmp1 += coef * in1[2];
-    tmp2 += coef * in2[-2];
-
-    coef = coef_ptr[3];
-    tmp1 += coef * in1[3];
-    tmp2 += coef * in2[-3];
-
-    coef = coef_ptr[4];
-    tmp1 += coef * in1[4];
-    tmp2 += coef * in2[-4];
-
-    coef = coef_ptr[5];
-    tmp1 += coef * in1[5];
-    tmp2 += coef * in2[-5];
-
-    coef = coef_ptr[6];
-    tmp1 += coef * in1[6];
-    tmp2 += coef * in2[-6];
-
-    coef = coef_ptr[7];
-    tmp1 += coef * in1[7];
-    tmp2 += coef * in2[-7];
-
-    coef = coef_ptr[8];
-    tmp1 += coef * in1[8];
-    tmp2 += coef * in2[-8];
-
-    // scale down, round and saturate
-    tmp1 >>= 15;
-    if (tmp1 > (WebRtc_Word32)0x00007FFF)
-        tmp1 = 0x00007FFF;
-    if (tmp1 < (WebRtc_Word32)0xFFFF8000)
-        tmp1 = 0xFFFF8000;
-    tmp2 >>= 15;
-    if (tmp2 > (WebRtc_Word32)0x00007FFF)
-        tmp2 = 0x00007FFF;
-    if (tmp2 < (WebRtc_Word32)0xFFFF8000)
-        tmp2 = 0xFFFF8000;
-    *out1 = (WebRtc_Word16)tmp1;
-    *out2 = (WebRtc_Word16)tmp2;
-}
-
-//   Resampling ratio: 11/16
-// input:  WebRtc_Word32 (normalized, not saturated) :: size 16 * K
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 11 * K
-//      K: Number of blocks
-
-void WebRtcSpl_32khzTo22khzIntToInt(const WebRtc_Word32* In,
-                                    WebRtc_Word32* Out,
-                                    const WebRtc_Word32 K)
-{
-    /////////////////////////////////////////////////////////////
-    // Filter operation:
-    //
-    // Perform resampling (16 input samples -> 11 output samples);
-    // process in sub blocks of size 16 samples.
-    WebRtc_Word32 m;
-
-    for (m = 0; m < K; m++)
-    {
-        // first output sample
-        Out[0] = ((WebRtc_Word32)In[3] << 15) + (1 << 14);
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_DotProdIntToInt(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_DotProdIntToInt(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_DotProdIntToInt(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_DotProdIntToInt(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_DotProdIntToInt(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
-
-        // update pointers
-        In += 16;
-        Out += 11;
-    }
-}
-
-//   Resampling ratio: 11/16
-// input:  WebRtc_Word32 (normalized, not saturated) :: size 16 * K
-// output: WebRtc_Word16 (saturated) :: size 11 * K
-//      K: Number of blocks
-
-void WebRtcSpl_32khzTo22khzIntToShort(const WebRtc_Word32 *In,
-                                      WebRtc_Word16 *Out,
-                                      const WebRtc_Word32 K)
-{
-    /////////////////////////////////////////////////////////////
-    // Filter operation:
-    //
-    // Perform resampling (16 input samples -> 11 output samples);
-    // process in sub blocks of size 16 samples.
-    WebRtc_Word32 tmp;
-    WebRtc_Word32 m;
-
-    for (m = 0; m < K; m++)
-    {
-        // first output sample
-        tmp = In[3];
-        if (tmp > (WebRtc_Word32)0x00007FFF)
-            tmp = 0x00007FFF;
-        if (tmp < (WebRtc_Word32)0xFFFF8000)
-            tmp = 0xFFFF8000;
-        Out[0] = (WebRtc_Word16)tmp;
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_DotProdIntToShort(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_DotProdIntToShort(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_DotProdIntToShort(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_DotProdIntToShort(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_DotProdIntToShort(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
-
-        // update pointers
-        In += 16;
-        Out += 11;
-    }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/resample_48khz.c b/src/common_audio/signal_processing_library/main/source/resample_48khz.c
deleted file mode 100644
index 31cbe6b..0000000
--- a/src/common_audio/signal_processing_library/main/source/resample_48khz.c
+++ /dev/null
@@ -1,186 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains resampling functions between 48 kHz and nb/wb.
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include <string.h>
-#include "signal_processing_library.h"
-#include "resample_by_2_internal.h"
-
-////////////////////////////
-///// 48 kHz -> 16 kHz /////
-////////////////////////////
-
-// 48 -> 16 resampler
-void WebRtcSpl_Resample48khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                    WebRtcSpl_State48khzTo16khz* state, WebRtc_Word32* tmpmem)
-{
-    ///// 48 --> 48(LP) /////
-    // WebRtc_Word16  in[480]
-    // WebRtc_Word32 out[480]
-    /////
-    WebRtcSpl_LPBy2ShortToInt(in, 480, tmpmem + 16, state->S_48_48);
-
-    ///// 48 --> 32 /////
-    // WebRtc_Word32  in[480]
-    // WebRtc_Word32 out[320]
-    /////
-    // copy state to and from input array
-    memcpy(tmpmem + 8, state->S_48_32, 8 * sizeof(WebRtc_Word32));
-    memcpy(state->S_48_32, tmpmem + 488, 8 * sizeof(WebRtc_Word32));
-    WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 160);
-
-    ///// 32 --> 16 /////
-    // WebRtc_Word32  in[320]
-    // WebRtc_Word16 out[160]
-    /////
-    WebRtcSpl_DownBy2IntToShort(tmpmem, 320, out, state->S_32_16);
-}
-
-// initialize state of 48 -> 16 resampler
-void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state)
-{
-    memset(state->S_48_48, 0, 16 * sizeof(WebRtc_Word32));
-    memset(state->S_48_32, 0, 8 * sizeof(WebRtc_Word32));
-    memset(state->S_32_16, 0, 8 * sizeof(WebRtc_Word32));
-}
-
-////////////////////////////
-///// 16 kHz -> 48 kHz /////
-////////////////////////////
-
-// 16 -> 48 resampler
-void WebRtcSpl_Resample16khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                    WebRtcSpl_State16khzTo48khz* state, WebRtc_Word32* tmpmem)
-{
-    ///// 16 --> 32 /////
-    // WebRtc_Word16  in[160]
-    // WebRtc_Word32 out[320]
-    /////
-    WebRtcSpl_UpBy2ShortToInt(in, 160, tmpmem + 16, state->S_16_32);
-
-    ///// 32 --> 24 /////
-    // WebRtc_Word32  in[320]
-    // WebRtc_Word32 out[240]
-    // copy state to and from input array
-    /////
-    memcpy(tmpmem + 8, state->S_32_24, 8 * sizeof(WebRtc_Word32));
-    memcpy(state->S_32_24, tmpmem + 328, 8 * sizeof(WebRtc_Word32));
-    WebRtcSpl_Resample32khzTo24khz(tmpmem + 8, tmpmem, 80);
-
-    ///// 24 --> 48 /////
-    // WebRtc_Word32  in[240]
-    // WebRtc_Word16 out[480]
-    /////
-    WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
-}
-
-// initialize state of 16 -> 48 resampler
-void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state)
-{
-    memset(state->S_16_32, 0, 8 * sizeof(WebRtc_Word32));
-    memset(state->S_32_24, 0, 8 * sizeof(WebRtc_Word32));
-    memset(state->S_24_48, 0, 8 * sizeof(WebRtc_Word32));
-}
-
-////////////////////////////
-///// 48 kHz ->  8 kHz /////
-////////////////////////////
-
-// 48 -> 8 resampler
-void WebRtcSpl_Resample48khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                   WebRtcSpl_State48khzTo8khz* state, WebRtc_Word32* tmpmem)
-{
-    ///// 48 --> 24 /////
-    // WebRtc_Word16  in[480]
-    // WebRtc_Word32 out[240]
-    /////
-    WebRtcSpl_DownBy2ShortToInt(in, 480, tmpmem + 256, state->S_48_24);
-
-    ///// 24 --> 24(LP) /////
-    // WebRtc_Word32  in[240]
-    // WebRtc_Word32 out[240]
-    /////
-    WebRtcSpl_LPBy2IntToInt(tmpmem + 256, 240, tmpmem + 16, state->S_24_24);
-
-    ///// 24 --> 16 /////
-    // WebRtc_Word32  in[240]
-    // WebRtc_Word32 out[160]
-    /////
-    // copy state to and from input array
-    memcpy(tmpmem + 8, state->S_24_16, 8 * sizeof(WebRtc_Word32));
-    memcpy(state->S_24_16, tmpmem + 248, 8 * sizeof(WebRtc_Word32));
-    WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 80);
-
-    ///// 16 --> 8 /////
-    // WebRtc_Word32  in[160]
-    // WebRtc_Word16 out[80]
-    /////
-    WebRtcSpl_DownBy2IntToShort(tmpmem, 160, out, state->S_16_8);
-}
-
-// initialize state of 48 -> 8 resampler
-void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state)
-{
-    memset(state->S_48_24, 0, 8 * sizeof(WebRtc_Word32));
-    memset(state->S_24_24, 0, 16 * sizeof(WebRtc_Word32));
-    memset(state->S_24_16, 0, 8 * sizeof(WebRtc_Word32));
-    memset(state->S_16_8, 0, 8 * sizeof(WebRtc_Word32));
-}
-
-////////////////////////////
-/////  8 kHz -> 48 kHz /////
-////////////////////////////
-
-// 8 -> 48 resampler
-void WebRtcSpl_Resample8khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
-                                   WebRtcSpl_State8khzTo48khz* state, WebRtc_Word32* tmpmem)
-{
-    ///// 8 --> 16 /////
-    // WebRtc_Word16  in[80]
-    // WebRtc_Word32 out[160]
-    /////
-    WebRtcSpl_UpBy2ShortToInt(in, 80, tmpmem + 264, state->S_8_16);
-
-    ///// 16 --> 12 /////
-    // WebRtc_Word32  in[160]
-    // WebRtc_Word32 out[120]
-    /////
-    // copy state to and from input array
-    memcpy(tmpmem + 256, state->S_16_12, 8 * sizeof(WebRtc_Word32));
-    memcpy(state->S_16_12, tmpmem + 416, 8 * sizeof(WebRtc_Word32));
-    WebRtcSpl_Resample32khzTo24khz(tmpmem + 256, tmpmem + 240, 40);
-
-    ///// 12 --> 24 /////
-    // WebRtc_Word32  in[120]
-    // WebRtc_Word16 out[240]
-    /////
-    WebRtcSpl_UpBy2IntToInt(tmpmem + 240, 120, tmpmem, state->S_12_24);
-
-    ///// 24 --> 48 /////
-    // WebRtc_Word32  in[240]
-    // WebRtc_Word16 out[480]
-    /////
-    WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
-}
-
-// initialize state of 8 -> 48 resampler
-void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state)
-{
-    memset(state->S_8_16, 0, 8 * sizeof(WebRtc_Word32));
-    memset(state->S_16_12, 0, 8 * sizeof(WebRtc_Word32));
-    memset(state->S_12_24, 0, 8 * sizeof(WebRtc_Word32));
-    memset(state->S_24_48, 0, 8 * sizeof(WebRtc_Word32));
-}
diff --git a/src/common_audio/signal_processing_library/main/source/resample_by_2.c b/src/common_audio/signal_processing_library/main/source/resample_by_2.c
deleted file mode 100644
index 7ed4cfd..0000000
--- a/src/common_audio/signal_processing_library/main/source/resample_by_2.c
+++ /dev/null
@@ -1,135 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the resampling by two functions.
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-// allpass filter coefficients.
-static const WebRtc_UWord16 kResampleAllpass1[3] = {3284, 24441, 49528};
-static const WebRtc_UWord16 kResampleAllpass2[3] = {12199, 37471, 60255};
-
-// decimator
-void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
-                             WebRtc_Word16* out, WebRtc_Word32* filtState)
-{
-    const WebRtc_Word16 *inptr;
-    WebRtc_Word16 *outptr;
-    WebRtc_Word32 *state;
-    WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
-    WebRtc_Word16 i;
-
-    // local versions of pointers to input and output arrays
-    inptr = in; // input array
-    outptr = out; // output array (of length len/2)
-    state = filtState; // filter state array; length = 8
-
-    for (i = (len >> 1); i > 0; i--)
-    {
-        // lower allpass filter
-        in32 = (WebRtc_Word32)(*inptr++) << 10;
-        diff = in32 - state[1];
-        tmp1 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[0], diff, state[0] );
-        state[0] = in32;
-        diff = tmp1 - state[2];
-        tmp2 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[1], diff, state[1] );
-        state[1] = tmp1;
-        diff = tmp2 - state[3];
-        state[3] = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[2], diff, state[2] );
-        state[2] = tmp2;
-
-        // upper allpass filter
-        in32 = (WebRtc_Word32)(*inptr++) << 10;
-        diff = in32 - state[5];
-        tmp1 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[0], diff, state[4] );
-        state[4] = in32;
-        diff = tmp1 - state[6];
-        tmp2 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[1], diff, state[5] );
-        state[5] = tmp1;
-        diff = tmp2 - state[7];
-        state[7] = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[2], diff, state[6] );
-        state[6] = tmp2;
-
-        // add two allpass outputs, divide by two and round
-        out32 = (state[3] + state[7] + 1024) >> 11;
-
-        // limit amplitude to prevent wrap-around, and write to output array
-        if (out32 > 32767)
-            *outptr++ = 32767;
-        else if (out32 < -32768)
-            *outptr++ = -32768;
-        else
-            *outptr++ = (WebRtc_Word16)out32;
-    }
-}
-
-void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len, WebRtc_Word16* out,
-                           WebRtc_Word32* filtState)
-{
-    const WebRtc_Word16 *inptr;
-    WebRtc_Word16 *outptr;
-    WebRtc_Word32 *state;
-    WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
-    WebRtc_Word16 i;
-
-    // local versions of pointers to input and output arrays
-    inptr = in; // input array
-    outptr = out; // output array (of length len*2)
-    state = filtState; // filter state array; length = 8
-
-    for (i = len; i > 0; i--)
-    {
-        // lower allpass filter
-        in32 = (WebRtc_Word32)(*inptr++) << 10;
-        diff = in32 - state[1];
-        tmp1 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[0], diff, state[0] );
-        state[0] = in32;
-        diff = tmp1 - state[2];
-        tmp2 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[1], diff, state[1] );
-        state[1] = tmp1;
-        diff = tmp2 - state[3];
-        state[3] = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass1[2], diff, state[2] );
-        state[2] = tmp2;
-
-        // round; limit amplitude to prevent wrap-around; write to output array
-        out32 = (state[3] + 512) >> 10;
-        if (out32 > 32767)
-            *outptr++ = 32767;
-        else if (out32 < -32768)
-            *outptr++ = -32768;
-        else
-            *outptr++ = (WebRtc_Word16)out32;
-
-        // upper allpass filter
-        diff = in32 - state[5];
-        tmp1 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[0], diff, state[4] );
-        state[4] = in32;
-        diff = tmp1 - state[6];
-        tmp2 = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[1], diff, state[5] );
-        state[5] = tmp1;
-        diff = tmp2 - state[7];
-        state[7] = WEBRTC_SPL_SCALEDIFF32( kResampleAllpass2[2], diff, state[6] );
-        state[6] = tmp2;
-
-        // round; limit amplitude to prevent wrap-around; write to output array
-        out32 = (state[7] + 512) >> 10;
-        if (out32 > 32767)
-            *outptr++ = 32767;
-        else if (out32 < -32768)
-            *outptr++ = -32768;
-        else
-            *outptr++ = (WebRtc_Word16)out32;
-    }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/resample_by_2_internal.c b/src/common_audio/signal_processing_library/main/source/resample_by_2_internal.c
deleted file mode 100644
index cbd2395..0000000
--- a/src/common_audio/signal_processing_library/main/source/resample_by_2_internal.c
+++ /dev/null
@@ -1,679 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file contains some internal resampling functions.
- *
- */
-
-#include "resample_by_2_internal.h"
-
-// allpass filter coefficients.
-static const WebRtc_Word16 kResampleAllpass[2][3] = {
-        {821, 6110, 12382},
-        {3050, 9368, 15063}
-};
-
-//
-//   decimator
-// input:  WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) OVERWRITTEN!
-// output: WebRtc_Word16 (saturated) (of length len/2)
-// state:  filter state array; length = 8
-
-void WebRtcSpl_DownBy2IntToShort(WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
-                                 WebRtc_Word32 *state)
-{
-    WebRtc_Word32 tmp0, tmp1, diff;
-    WebRtc_Word32 i;
-
-    len >>= 1;
-
-    // lower allpass filter (operates on even input samples)
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = in[i << 1];
-        diff = tmp0 - state[1];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[0] + diff * kResampleAllpass[1][0];
-        state[0] = tmp0;
-        diff = tmp1 - state[2];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[1] + diff * kResampleAllpass[1][1];
-        state[1] = tmp1;
-        diff = tmp0 - state[3];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[3] = state[2] + diff * kResampleAllpass[1][2];
-        state[2] = tmp0;
-
-        // divide by two and store temporarily
-        in[i << 1] = (state[3] >> 1);
-    }
-
-    in++;
-
-    // upper allpass filter (operates on odd input samples)
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = in[i << 1];
-        diff = tmp0 - state[5];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[4] + diff * kResampleAllpass[0][0];
-        state[4] = tmp0;
-        diff = tmp1 - state[6];
-        // scale down and round
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[5] + diff * kResampleAllpass[0][1];
-        state[5] = tmp1;
-        diff = tmp0 - state[7];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[7] = state[6] + diff * kResampleAllpass[0][2];
-        state[6] = tmp0;
-
-        // divide by two and store temporarily
-        in[i << 1] = (state[7] >> 1);
-    }
-
-    in--;
-
-    // combine allpass outputs
-    for (i = 0; i < len; i += 2)
-    {
-        // divide by two, add both allpass outputs and round
-        tmp0 = (in[i << 1] + in[(i << 1) + 1]) >> 15;
-        tmp1 = (in[(i << 1) + 2] + in[(i << 1) + 3]) >> 15;
-        if (tmp0 > (WebRtc_Word32)0x00007FFF)
-            tmp0 = 0x00007FFF;
-        if (tmp0 < (WebRtc_Word32)0xFFFF8000)
-            tmp0 = 0xFFFF8000;
-        out[i] = (WebRtc_Word16)tmp0;
-        if (tmp1 > (WebRtc_Word32)0x00007FFF)
-            tmp1 = 0x00007FFF;
-        if (tmp1 < (WebRtc_Word32)0xFFFF8000)
-            tmp1 = 0xFFFF8000;
-        out[i + 1] = (WebRtc_Word16)tmp1;
-    }
-}
-
-//
-//   decimator
-// input:  WebRtc_Word16
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) (of length len/2)
-// state:  filter state array; length = 8
-
-void WebRtcSpl_DownBy2ShortToInt(const WebRtc_Word16 *in,
-                                  WebRtc_Word32 len,
-                                  WebRtc_Word32 *out,
-                                  WebRtc_Word32 *state)
-{
-    WebRtc_Word32 tmp0, tmp1, diff;
-    WebRtc_Word32 i;
-
-    len >>= 1;
-
-    // lower allpass filter (operates on even input samples)
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
-        diff = tmp0 - state[1];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[0] + diff * kResampleAllpass[1][0];
-        state[0] = tmp0;
-        diff = tmp1 - state[2];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[1] + diff * kResampleAllpass[1][1];
-        state[1] = tmp1;
-        diff = tmp0 - state[3];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[3] = state[2] + diff * kResampleAllpass[1][2];
-        state[2] = tmp0;
-
-        // divide by two and store temporarily
-        out[i] = (state[3] >> 1);
-    }
-
-    in++;
-
-    // upper allpass filter (operates on odd input samples)
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
-        diff = tmp0 - state[5];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[4] + diff * kResampleAllpass[0][0];
-        state[4] = tmp0;
-        diff = tmp1 - state[6];
-        // scale down and round
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[5] + diff * kResampleAllpass[0][1];
-        state[5] = tmp1;
-        diff = tmp0 - state[7];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[7] = state[6] + diff * kResampleAllpass[0][2];
-        state[6] = tmp0;
-
-        // divide by two and store temporarily
-        out[i] += (state[7] >> 1);
-    }
-
-    in--;
-}
-
-//
-//   interpolator
-// input:  WebRtc_Word16
-// output: WebRtc_Word32 (normalized, not saturated) (of length len*2)
-// state:  filter state array; length = 8
-void WebRtcSpl_UpBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
-                               WebRtc_Word32 *state)
-{
-    WebRtc_Word32 tmp0, tmp1, diff;
-    WebRtc_Word32 i;
-
-    // upper allpass filter (generates odd output samples)
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = ((WebRtc_Word32)in[i] << 15) + (1 << 14);
-        diff = tmp0 - state[5];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[4] + diff * kResampleAllpass[0][0];
-        state[4] = tmp0;
-        diff = tmp1 - state[6];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[5] + diff * kResampleAllpass[0][1];
-        state[5] = tmp1;
-        diff = tmp0 - state[7];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[7] = state[6] + diff * kResampleAllpass[0][2];
-        state[6] = tmp0;
-
-        // scale down, round and store
-        out[i << 1] = state[7] >> 15;
-    }
-
-    out++;
-
-    // lower allpass filter (generates even output samples)
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = ((WebRtc_Word32)in[i] << 15) + (1 << 14);
-        diff = tmp0 - state[1];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[0] + diff * kResampleAllpass[1][0];
-        state[0] = tmp0;
-        diff = tmp1 - state[2];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[1] + diff * kResampleAllpass[1][1];
-        state[1] = tmp1;
-        diff = tmp0 - state[3];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[3] = state[2] + diff * kResampleAllpass[1][2];
-        state[2] = tmp0;
-
-        // scale down, round and store
-        out[i << 1] = state[3] >> 15;
-    }
-}
-
-//
-//   interpolator
-// input:  WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) (of length len*2)
-// state:  filter state array; length = 8
-void WebRtcSpl_UpBy2IntToInt(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
-                             WebRtc_Word32 *state)
-{
-    WebRtc_Word32 tmp0, tmp1, diff;
-    WebRtc_Word32 i;
-
-    // upper allpass filter (generates odd output samples)
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = in[i];
-        diff = tmp0 - state[5];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[4] + diff * kResampleAllpass[0][0];
-        state[4] = tmp0;
-        diff = tmp1 - state[6];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[5] + diff * kResampleAllpass[0][1];
-        state[5] = tmp1;
-        diff = tmp0 - state[7];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[7] = state[6] + diff * kResampleAllpass[0][2];
-        state[6] = tmp0;
-
-        // scale down, round and store
-        out[i << 1] = state[7];
-    }
-
-    out++;
-
-    // lower allpass filter (generates even output samples)
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = in[i];
-        diff = tmp0 - state[1];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[0] + diff * kResampleAllpass[1][0];
-        state[0] = tmp0;
-        diff = tmp1 - state[2];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[1] + diff * kResampleAllpass[1][1];
-        state[1] = tmp1;
-        diff = tmp0 - state[3];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[3] = state[2] + diff * kResampleAllpass[1][2];
-        state[2] = tmp0;
-
-        // scale down, round and store
-        out[i << 1] = state[3];
-    }
-}
-
-//
-//   interpolator
-// input:  WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
-// output: WebRtc_Word16 (saturated) (of length len*2)
-// state:  filter state array; length = 8
-void WebRtcSpl_UpBy2IntToShort(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
-                               WebRtc_Word32 *state)
-{
-    WebRtc_Word32 tmp0, tmp1, diff;
-    WebRtc_Word32 i;
-
-    // upper allpass filter (generates odd output samples)
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = in[i];
-        diff = tmp0 - state[5];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[4] + diff * kResampleAllpass[0][0];
-        state[4] = tmp0;
-        diff = tmp1 - state[6];
-        // scale down and round
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[5] + diff * kResampleAllpass[0][1];
-        state[5] = tmp1;
-        diff = tmp0 - state[7];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[7] = state[6] + diff * kResampleAllpass[0][2];
-        state[6] = tmp0;
-
-        // scale down, saturate and store
-        tmp1 = state[7] >> 15;
-        if (tmp1 > (WebRtc_Word32)0x00007FFF)
-            tmp1 = 0x00007FFF;
-        if (tmp1 < (WebRtc_Word32)0xFFFF8000)
-            tmp1 = 0xFFFF8000;
-        out[i << 1] = (WebRtc_Word16)tmp1;
-    }
-
-    out++;
-
-    // lower allpass filter (generates even output samples)
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = in[i];
-        diff = tmp0 - state[1];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[0] + diff * kResampleAllpass[1][0];
-        state[0] = tmp0;
-        diff = tmp1 - state[2];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[1] + diff * kResampleAllpass[1][1];
-        state[1] = tmp1;
-        diff = tmp0 - state[3];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[3] = state[2] + diff * kResampleAllpass[1][2];
-        state[2] = tmp0;
-
-        // scale down, saturate and store
-        tmp1 = state[3] >> 15;
-        if (tmp1 > (WebRtc_Word32)0x00007FFF)
-            tmp1 = 0x00007FFF;
-        if (tmp1 < (WebRtc_Word32)0xFFFF8000)
-            tmp1 = 0xFFFF8000;
-        out[i << 1] = (WebRtc_Word16)tmp1;
-    }
-}
-
-//   lowpass filter
-// input:  WebRtc_Word16
-// output: WebRtc_Word32 (normalized, not saturated)
-// state:  filter state array; length = 8
-void WebRtcSpl_LPBy2ShortToInt(const WebRtc_Word16* in, WebRtc_Word32 len, WebRtc_Word32* out,
-                               WebRtc_Word32* state)
-{
-    WebRtc_Word32 tmp0, tmp1, diff;
-    WebRtc_Word32 i;
-
-    len >>= 1;
-
-    // lower allpass filter: odd input -> even output samples
-    in++;
-    // initial state of polyphase delay element
-    tmp0 = state[12];
-    for (i = 0; i < len; i++)
-    {
-        diff = tmp0 - state[1];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[0] + diff * kResampleAllpass[1][0];
-        state[0] = tmp0;
-        diff = tmp1 - state[2];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[1] + diff * kResampleAllpass[1][1];
-        state[1] = tmp1;
-        diff = tmp0 - state[3];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[3] = state[2] + diff * kResampleAllpass[1][2];
-        state[2] = tmp0;
-
-        // scale down, round and store
-        out[i << 1] = state[3] >> 1;
-        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
-    }
-    in--;
-
-    // upper allpass filter: even input -> even output samples
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
-        diff = tmp0 - state[5];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[4] + diff * kResampleAllpass[0][0];
-        state[4] = tmp0;
-        diff = tmp1 - state[6];
-        // scale down and round
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[5] + diff * kResampleAllpass[0][1];
-        state[5] = tmp1;
-        diff = tmp0 - state[7];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[7] = state[6] + diff * kResampleAllpass[0][2];
-        state[6] = tmp0;
-
-        // average the two allpass outputs, scale down and store
-        out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
-    }
-
-    // switch to odd output samples
-    out++;
-
-    // lower allpass filter: even input -> odd output samples
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
-        diff = tmp0 - state[9];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[8] + diff * kResampleAllpass[1][0];
-        state[8] = tmp0;
-        diff = tmp1 - state[10];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[9] + diff * kResampleAllpass[1][1];
-        state[9] = tmp1;
-        diff = tmp0 - state[11];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[11] = state[10] + diff * kResampleAllpass[1][2];
-        state[10] = tmp0;
-
-        // scale down, round and store
-        out[i << 1] = state[11] >> 1;
-    }
-
-    // upper allpass filter: odd input -> odd output samples
-    in++;
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
-        diff = tmp0 - state[13];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[12] + diff * kResampleAllpass[0][0];
-        state[12] = tmp0;
-        diff = tmp1 - state[14];
-        // scale down and round
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[13] + diff * kResampleAllpass[0][1];
-        state[13] = tmp1;
-        diff = tmp0 - state[15];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[15] = state[14] + diff * kResampleAllpass[0][2];
-        state[14] = tmp0;
-
-        // average the two allpass outputs, scale down and store
-        out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
-    }
-}
-
-//   lowpass filter
-// input:  WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
-// output: WebRtc_Word32 (normalized, not saturated)
-// state:  filter state array; length = 8
-void WebRtcSpl_LPBy2IntToInt(const WebRtc_Word32* in, WebRtc_Word32 len, WebRtc_Word32* out,
-                             WebRtc_Word32* state)
-{
-    WebRtc_Word32 tmp0, tmp1, diff;
-    WebRtc_Word32 i;
-
-    len >>= 1;
-
-    // lower allpass filter: odd input -> even output samples
-    in++;
-    // initial state of polyphase delay element
-    tmp0 = state[12];
-    for (i = 0; i < len; i++)
-    {
-        diff = tmp0 - state[1];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[0] + diff * kResampleAllpass[1][0];
-        state[0] = tmp0;
-        diff = tmp1 - state[2];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[1] + diff * kResampleAllpass[1][1];
-        state[1] = tmp1;
-        diff = tmp0 - state[3];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[3] = state[2] + diff * kResampleAllpass[1][2];
-        state[2] = tmp0;
-
-        // scale down, round and store
-        out[i << 1] = state[3] >> 1;
-        tmp0 = in[i << 1];
-    }
-    in--;
-
-    // upper allpass filter: even input -> even output samples
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = in[i << 1];
-        diff = tmp0 - state[5];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[4] + diff * kResampleAllpass[0][0];
-        state[4] = tmp0;
-        diff = tmp1 - state[6];
-        // scale down and round
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[5] + diff * kResampleAllpass[0][1];
-        state[5] = tmp1;
-        diff = tmp0 - state[7];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[7] = state[6] + diff * kResampleAllpass[0][2];
-        state[6] = tmp0;
-
-        // average the two allpass outputs, scale down and store
-        out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
-    }
-
-    // switch to odd output samples
-    out++;
-
-    // lower allpass filter: even input -> odd output samples
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = in[i << 1];
-        diff = tmp0 - state[9];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[8] + diff * kResampleAllpass[1][0];
-        state[8] = tmp0;
-        diff = tmp1 - state[10];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[9] + diff * kResampleAllpass[1][1];
-        state[9] = tmp1;
-        diff = tmp0 - state[11];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[11] = state[10] + diff * kResampleAllpass[1][2];
-        state[10] = tmp0;
-
-        // scale down, round and store
-        out[i << 1] = state[11] >> 1;
-    }
-
-    // upper allpass filter: odd input -> odd output samples
-    in++;
-    for (i = 0; i < len; i++)
-    {
-        tmp0 = in[i << 1];
-        diff = tmp0 - state[13];
-        // scale down and round
-        diff = (diff + (1 << 13)) >> 14;
-        tmp1 = state[12] + diff * kResampleAllpass[0][0];
-        state[12] = tmp0;
-        diff = tmp1 - state[14];
-        // scale down and round
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        tmp0 = state[13] + diff * kResampleAllpass[0][1];
-        state[13] = tmp1;
-        diff = tmp0 - state[15];
-        // scale down and truncate
-        diff = diff >> 14;
-        if (diff < 0)
-            diff += 1;
-        state[15] = state[14] + diff * kResampleAllpass[0][2];
-        state[14] = tmp0;
-
-        // average the two allpass outputs, scale down and store
-        out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
-    }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/resample_by_2_internal.h b/src/common_audio/signal_processing_library/main/source/resample_by_2_internal.h
deleted file mode 100644
index b6ac9f0..0000000
--- a/src/common_audio/signal_processing_library/main/source/resample_by_2_internal.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file contains some internal resampling functions.
- *
- */
-
-#ifndef WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
-#define WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
-
-#include "typedefs.h"
-
-/*******************************************************************
- * resample_by_2_fast.c
- * Functions for internal use in the other resample functions
- ******************************************************************/
-void WebRtcSpl_DownBy2IntToShort(WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
-                                 WebRtc_Word32 *state);
-
-void WebRtcSpl_DownBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len,
-                                 WebRtc_Word32 *out, WebRtc_Word32 *state);
-
-void WebRtcSpl_UpBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len,
-                               WebRtc_Word32 *out, WebRtc_Word32 *state);
-
-void WebRtcSpl_UpBy2IntToInt(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
-                             WebRtc_Word32 *state);
-
-void WebRtcSpl_UpBy2IntToShort(const WebRtc_Word32 *in, WebRtc_Word32 len,
-                               WebRtc_Word16 *out, WebRtc_Word32 *state);
-
-void WebRtcSpl_LPBy2ShortToInt(const WebRtc_Word16* in, WebRtc_Word32 len,
-                               WebRtc_Word32* out, WebRtc_Word32* state);
-
-void WebRtcSpl_LPBy2IntToInt(const WebRtc_Word32* in, WebRtc_Word32 len, WebRtc_Word32* out,
-                             WebRtc_Word32* state);
-
-#endif // WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
diff --git a/src/common_audio/signal_processing_library/main/source/resample_fractional.c b/src/common_audio/signal_processing_library/main/source/resample_fractional.c
deleted file mode 100644
index 51003d4..0000000
--- a/src/common_audio/signal_processing_library/main/source/resample_fractional.c
+++ /dev/null
@@ -1,242 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the resampling functions between 48, 44, 32 and 24 kHz.
- * The description headers can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-// interpolation coefficients
-static const WebRtc_Word16 kCoefficients48To32[2][8] = {
-        {778, -2050, 1087, 23285, 12903, -3783, 441, 222},
-        {222, 441, -3783, 12903, 23285, 1087, -2050, 778}
-};
-
-static const WebRtc_Word16 kCoefficients32To24[3][8] = {
-        {767, -2362, 2434, 24406, 10620, -3838, 721, 90},
-        {386, -381, -2646, 19062, 19062, -2646, -381, 386},
-        {90, 721, -3838, 10620, 24406, 2434, -2362, 767}
-};
-
-static const WebRtc_Word16 kCoefficients44To32[4][9] = {
-        {117, -669, 2245, -6183, 26267, 13529, -3245, 845, -138},
-        {-101, 612, -2283, 8532, 29790, -5138, 1789, -524, 91},
-        {50, -292, 1016, -3064, 32010, 3933, -1147, 315, -53},
-        {-156, 974, -3863, 18603, 21691, -6246, 2353, -712, 126}
-};
-
-//   Resampling ratio: 2/3
-// input:  WebRtc_Word32 (normalized, not saturated) :: size 3 * K
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 2 * K
-//      K: number of blocks
-
-void WebRtcSpl_Resample48khzTo32khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
-                                    const WebRtc_Word32 K)
-{
-    /////////////////////////////////////////////////////////////
-    // Filter operation:
-    //
-    // Perform resampling (3 input samples -> 2 output samples);
-    // process in sub blocks of size 3 samples.
-    WebRtc_Word32 tmp;
-    WebRtc_Word32 m;
-
-    for (m = 0; m < K; m++)
-    {
-        tmp = 1 << 14;
-        tmp += kCoefficients48To32[0][0] * In[0];
-        tmp += kCoefficients48To32[0][1] * In[1];
-        tmp += kCoefficients48To32[0][2] * In[2];
-        tmp += kCoefficients48To32[0][3] * In[3];
-        tmp += kCoefficients48To32[0][4] * In[4];
-        tmp += kCoefficients48To32[0][5] * In[5];
-        tmp += kCoefficients48To32[0][6] * In[6];
-        tmp += kCoefficients48To32[0][7] * In[7];
-        Out[0] = tmp;
-
-        tmp = 1 << 14;
-        tmp += kCoefficients48To32[1][0] * In[1];
-        tmp += kCoefficients48To32[1][1] * In[2];
-        tmp += kCoefficients48To32[1][2] * In[3];
-        tmp += kCoefficients48To32[1][3] * In[4];
-        tmp += kCoefficients48To32[1][4] * In[5];
-        tmp += kCoefficients48To32[1][5] * In[6];
-        tmp += kCoefficients48To32[1][6] * In[7];
-        tmp += kCoefficients48To32[1][7] * In[8];
-        Out[1] = tmp;
-
-        // update pointers
-        In += 3;
-        Out += 2;
-    }
-}
-
-//   Resampling ratio: 3/4
-// input:  WebRtc_Word32 (normalized, not saturated) :: size 4 * K
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 3 * K
-//      K: number of blocks
-
-void WebRtcSpl_Resample32khzTo24khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
-                                    const WebRtc_Word32 K)
-{
-    /////////////////////////////////////////////////////////////
-    // Filter operation:
-    //
-    // Perform resampling (4 input samples -> 3 output samples);
-    // process in sub blocks of size 4 samples.
-    WebRtc_Word32 m;
-    WebRtc_Word32 tmp;
-
-    for (m = 0; m < K; m++)
-    {
-        tmp = 1 << 14;
-        tmp += kCoefficients32To24[0][0] * In[0];
-        tmp += kCoefficients32To24[0][1] * In[1];
-        tmp += kCoefficients32To24[0][2] * In[2];
-        tmp += kCoefficients32To24[0][3] * In[3];
-        tmp += kCoefficients32To24[0][4] * In[4];
-        tmp += kCoefficients32To24[0][5] * In[5];
-        tmp += kCoefficients32To24[0][6] * In[6];
-        tmp += kCoefficients32To24[0][7] * In[7];
-        Out[0] = tmp;
-
-        tmp = 1 << 14;
-        tmp += kCoefficients32To24[1][0] * In[1];
-        tmp += kCoefficients32To24[1][1] * In[2];
-        tmp += kCoefficients32To24[1][2] * In[3];
-        tmp += kCoefficients32To24[1][3] * In[4];
-        tmp += kCoefficients32To24[1][4] * In[5];
-        tmp += kCoefficients32To24[1][5] * In[6];
-        tmp += kCoefficients32To24[1][6] * In[7];
-        tmp += kCoefficients32To24[1][7] * In[8];
-        Out[1] = tmp;
-
-        tmp = 1 << 14;
-        tmp += kCoefficients32To24[2][0] * In[2];
-        tmp += kCoefficients32To24[2][1] * In[3];
-        tmp += kCoefficients32To24[2][2] * In[4];
-        tmp += kCoefficients32To24[2][3] * In[5];
-        tmp += kCoefficients32To24[2][4] * In[6];
-        tmp += kCoefficients32To24[2][5] * In[7];
-        tmp += kCoefficients32To24[2][6] * In[8];
-        tmp += kCoefficients32To24[2][7] * In[9];
-        Out[2] = tmp;
-
-        // update pointers
-        In += 4;
-        Out += 3;
-    }
-}
-
-//
-// fractional resampling filters
-//   Fout = 11/16 * Fin
-//   Fout =  8/11 * Fin
-//
-
-// compute two inner-products and store them to output array
-static void WebRtcSpl_ResampDotProduct(const WebRtc_Word32 *in1, const WebRtc_Word32 *in2,
-                               const WebRtc_Word16 *coef_ptr, WebRtc_Word32 *out1,
-                               WebRtc_Word32 *out2)
-{
-    WebRtc_Word32 tmp1 = 16384;
-    WebRtc_Word32 tmp2 = 16384;
-    WebRtc_Word16 coef;
-
-    coef = coef_ptr[0];
-    tmp1 += coef * in1[0];
-    tmp2 += coef * in2[-0];
-
-    coef = coef_ptr[1];
-    tmp1 += coef * in1[1];
-    tmp2 += coef * in2[-1];
-
-    coef = coef_ptr[2];
-    tmp1 += coef * in1[2];
-    tmp2 += coef * in2[-2];
-
-    coef = coef_ptr[3];
-    tmp1 += coef * in1[3];
-    tmp2 += coef * in2[-3];
-
-    coef = coef_ptr[4];
-    tmp1 += coef * in1[4];
-    tmp2 += coef * in2[-4];
-
-    coef = coef_ptr[5];
-    tmp1 += coef * in1[5];
-    tmp2 += coef * in2[-5];
-
-    coef = coef_ptr[6];
-    tmp1 += coef * in1[6];
-    tmp2 += coef * in2[-6];
-
-    coef = coef_ptr[7];
-    tmp1 += coef * in1[7];
-    tmp2 += coef * in2[-7];
-
-    coef = coef_ptr[8];
-    *out1 = tmp1 + coef * in1[8];
-    *out2 = tmp2 + coef * in2[-8];
-}
-
-//   Resampling ratio: 8/11
-// input:  WebRtc_Word32 (normalized, not saturated) :: size 11 * K
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size  8 * K
-//      K: number of blocks
-
-void WebRtcSpl_Resample44khzTo32khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
-                                    const WebRtc_Word32 K)
-{
-    /////////////////////////////////////////////////////////////
-    // Filter operation:
-    //
-    // Perform resampling (11 input samples -> 8 output samples);
-    // process in sub blocks of size 11 samples.
-    WebRtc_Word32 tmp;
-    WebRtc_Word32 m;
-
-    for (m = 0; m < K; m++)
-    {
-        tmp = 1 << 14;
-
-        // first output sample
-        Out[0] = ((WebRtc_Word32)In[3] << 15) + tmp;
-
-        // sum and accumulate filter coefficients and input samples
-        tmp += kCoefficients44To32[3][0] * In[5];
-        tmp += kCoefficients44To32[3][1] * In[6];
-        tmp += kCoefficients44To32[3][2] * In[7];
-        tmp += kCoefficients44To32[3][3] * In[8];
-        tmp += kCoefficients44To32[3][4] * In[9];
-        tmp += kCoefficients44To32[3][5] * In[10];
-        tmp += kCoefficients44To32[3][6] * In[11];
-        tmp += kCoefficients44To32[3][7] * In[12];
-        tmp += kCoefficients44To32[3][8] * In[13];
-        Out[4] = tmp;
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_ResampDotProduct(&In[0], &In[17], kCoefficients44To32[0], &Out[1], &Out[7]);
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_ResampDotProduct(&In[2], &In[15], kCoefficients44To32[1], &Out[2], &Out[6]);
-
-        // sum and accumulate filter coefficients and input samples
-        WebRtcSpl_ResampDotProduct(&In[3], &In[14], kCoefficients44To32[2], &Out[3], &Out[5]);
-
-        // update pointers
-        In += 11;
-        Out += 8;
-    }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/sin_table.c b/src/common_audio/signal_processing_library/main/source/sin_table.c
deleted file mode 100644
index ea44666..0000000
--- a/src/common_audio/signal_processing_library/main/source/sin_table.c
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the 360 degree sine table.
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_kSinTable[] = {
-        0,    142,    285,    428,    571,    713,    856,    998,   1140,
-     1281,   1422,   1563,   1703,   1842,   1981,   2120,   2258,   2395,
-     2531,   2667,   2801,   2935,   3068,   3200,   3331,   3462,   3591,
-     3719,   3845,   3971,   4095,   4219,   4341,   4461,   4580,   4698,
-     4815,   4930,   5043,   5155,   5265,   5374,   5481,   5586,   5690,
-     5792,   5892,   5991,   6087,   6182,   6275,   6366,   6455,   6542,
-     6627,   6710,   6791,   6870,   6947,   7021,   7094,   7164,   7233,
-     7299,   7362,   7424,   7483,   7540,   7595,   7647,   7697,   7745,
-     7791,   7834,   7874,   7912,   7948,   7982,   8012,   8041,   8067,
-     8091,   8112,   8130,   8147,   8160,   8172,   8180,   8187,   8190,
-     8191,   8190,   8187,   8180,   8172,   8160,   8147,   8130,   8112,
-     8091,   8067,   8041,   8012,   7982,   7948,   7912,   7874,   7834,
-     7791,   7745,   7697,   7647,   7595,   7540,   7483,   7424,   7362,
-     7299,   7233,   7164,   7094,   7021,   6947,   6870,   6791,   6710,
-     6627,   6542,   6455,   6366,   6275,   6182,   6087,   5991,   5892,
-     5792,   5690,   5586,   5481,   5374,   5265,   5155,   5043,   4930,
-     4815,   4698,   4580,   4461,   4341,   4219,   4096,   3971,   3845,
-     3719,   3591,   3462,   3331,   3200,   3068,   2935,   2801,   2667,
-     2531,   2395,   2258,   2120,   1981,   1842,   1703,   1563,   1422,
-     1281,   1140,    998,    856,    713,    571,    428,    285,    142,
-        0,   -142,   -285,   -428,   -571,   -713,   -856,   -998,  -1140,
-    -1281,  -1422,  -1563,  -1703,  -1842,  -1981,  -2120,  -2258,  -2395,
-    -2531,  -2667,  -2801,  -2935,  -3068,  -3200,  -3331,  -3462,  -3591,
-    -3719,  -3845,  -3971,  -4095,  -4219,  -4341,  -4461,  -4580,  -4698,
-    -4815,  -4930,  -5043,  -5155,  -5265,  -5374,  -5481,  -5586,  -5690,
-    -5792,  -5892,  -5991,  -6087,  -6182,  -6275,  -6366,  -6455,  -6542,
-    -6627,  -6710,  -6791,  -6870,  -6947,  -7021,  -7094,  -7164,  -7233,
-    -7299,  -7362,  -7424,  -7483,  -7540,  -7595,  -7647,  -7697,  -7745,
-    -7791,  -7834,  -7874,  -7912,  -7948,  -7982,  -8012,  -8041,  -8067,
-    -8091,  -8112,  -8130,  -8147,  -8160,  -8172,  -8180,  -8187,  -8190,
-    -8191,  -8190,  -8187,  -8180,  -8172,  -8160,  -8147,  -8130,  -8112,
-    -8091,  -8067,  -8041,  -8012,  -7982,  -7948,  -7912,  -7874,  -7834,
-    -7791,  -7745,  -7697,  -7647,  -7595,  -7540,  -7483,  -7424,  -7362,
-    -7299,  -7233,  -7164,  -7094,  -7021,  -6947,  -6870,  -6791,  -6710,
-    -6627,  -6542,  -6455,  -6366,  -6275,  -6182,  -6087,  -5991,  -5892,
-    -5792,  -5690,  -5586,  -5481,  -5374,  -5265,  -5155,  -5043,  -4930,
-    -4815,  -4698,  -4580,  -4461,  -4341,  -4219,  -4096,  -3971,  -3845,
-    -3719,  -3591,  -3462,  -3331,  -3200,  -3068,  -2935,  -2801,  -2667,
-    -2531,  -2395,  -2258,  -2120,  -1981,  -1842,  -1703,  -1563,  -1422,
-    -1281,  -1140,   -998,   -856,   -713,   -571,   -428,   -285,   -142
-};
diff --git a/src/common_audio/signal_processing_library/main/source/sin_table_1024.c b/src/common_audio/signal_processing_library/main/source/sin_table_1024.c
deleted file mode 100644
index a2007f9..0000000
--- a/src/common_audio/signal_processing_library/main/source/sin_table_1024.c
+++ /dev/null
@@ -1,150 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the 1024 point sine table.
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_kSinTable1024[] =
-{
-      0,    201,    402,    603,    804,   1005,   1206,   1406,
-   1607,   1808,   2009,   2209,   2410,   2610,   2811,   3011,
-   3211,   3411,   3611,   3811,   4011,   4210,   4409,   4608,
-   4807,   5006,   5205,   5403,   5601,   5799,   5997,   6195,
-   6392,   6589,   6786,   6982,   7179,   7375,   7571,   7766,
-   7961,   8156,   8351,   8545,   8739,   8932,   9126,   9319,
-   9511,   9703,   9895,  10087,  10278,  10469,  10659,  10849,
-  11038,  11227,  11416,  11604,  11792,  11980,  12166,  12353,
-  12539,  12724,  12909,  13094,  13278,  13462,  13645,  13827,
-  14009,  14191,  14372,  14552,  14732,  14911,  15090,  15268,
-  15446,  15623,  15799,  15975,  16150,  16325,  16499,  16672,
-  16845,  17017,  17189,  17360,  17530,  17699,  17868,  18036,
-  18204,  18371,  18537,  18702,  18867,  19031,  19194,  19357,
-  19519,  19680,  19840,  20000,  20159,  20317,  20474,  20631,
-  20787,  20942,  21096,  21249,  21402,  21554,  21705,  21855,
-  22004,  22153,  22301,  22448,  22594,  22739,  22883,  23027,
-  23169,  23311,  23452,  23592,  23731,  23869,  24006,  24143,
-  24278,  24413,  24546,  24679,  24811,  24942,  25072,  25201,
-  25329,  25456,  25582,  25707,  25831,  25954,  26077,  26198,
-  26318,  26437,  26556,  26673,  26789,  26905,  27019,  27132,
-  27244,  27355,  27466,  27575,  27683,  27790,  27896,  28001,
-  28105,  28208,  28309,  28410,  28510,  28608,  28706,  28802,
-  28897,  28992,  29085,  29177,  29268,  29358,  29446,  29534,
-  29621,  29706,  29790,  29873,  29955,  30036,  30116,  30195,
-  30272,  30349,  30424,  30498,  30571,  30643,  30713,  30783,
-  30851,  30918,  30984,  31049,
-  31113,  31175,  31236,  31297,
-  31356,  31413,  31470,  31525,  31580,  31633,  31684,  31735,
-  31785,  31833,  31880,  31926,  31970,  32014,  32056,  32097,
-  32137,  32176,  32213,  32249,  32284,  32318,  32350,  32382,
-  32412,  32441,  32468,  32495,  32520,  32544,  32567,  32588,
-  32609,  32628,  32646,  32662,  32678,  32692,  32705,  32717,
-  32727,  32736,  32744,  32751,  32757,  32761,  32764,  32766,
-  32767,  32766,  32764,  32761,  32757,  32751,  32744,  32736,
-  32727,  32717,  32705,  32692,  32678,  32662,  32646,  32628,
-  32609,  32588,  32567,  32544,  32520,  32495,  32468,  32441,
-  32412,  32382,  32350,  32318,  32284,  32249,  32213,  32176,
-  32137,  32097,  32056,  32014,  31970,  31926,  31880,  31833,
-  31785,  31735,  31684,  31633,  31580,  31525,  31470,  31413,
-  31356,  31297,  31236,  31175,  31113,  31049,  30984,  30918,
-  30851,  30783,  30713,  30643,  30571,  30498,  30424,  30349,
-  30272,  30195,  30116,  30036,  29955,  29873,  29790,  29706,
-  29621,  29534,  29446,  29358,  29268,  29177,  29085,  28992,
-  28897,  28802,  28706,  28608,  28510,  28410,  28309,  28208,
-  28105,  28001,  27896,  27790,  27683,  27575,  27466,  27355,
-  27244,  27132,  27019,  26905,  26789,  26673,  26556,  26437,
-  26318,  26198,  26077,  25954,  25831,  25707,  25582,  25456,
-  25329,  25201,  25072,  24942,  24811,  24679,  24546,  24413,
-  24278,  24143,  24006,  23869,  23731,  23592,  23452,  23311,
-  23169,  23027,  22883,  22739,  22594,  22448,  22301,  22153,
-  22004,  21855,  21705,  21554,  21402,  21249,  21096,  20942,
-  20787,  20631,  20474,  20317,  20159,  20000,  19840,  19680,
-  19519,  19357,  19194,  19031,  18867,  18702,  18537,  18371,
-  18204,  18036,  17868,  17699,  17530,  17360,  17189,  17017,
-  16845,  16672,  16499,  16325,  16150,  15975,  15799,  15623,
-  15446,  15268,  15090,  14911,  14732,  14552,  14372,  14191,
-  14009,  13827,  13645,  13462,  13278,  13094,  12909,  12724,
-  12539,  12353,  12166,  11980,  11792,  11604,  11416,  11227,
-  11038,  10849,  10659,  10469,  10278,  10087,   9895,   9703,
-   9511,   9319,   9126,   8932,   8739,   8545,   8351,   8156,
-   7961,   7766,   7571,   7375,   7179,   6982,   6786,   6589,
-   6392,   6195,   5997,   5799,   5601,   5403,   5205,   5006,
-   4807,   4608,   4409,   4210,   4011,   3811,   3611,   3411,
-   3211,   3011,   2811,   2610,   2410,   2209,   2009,   1808,
-   1607,   1406,   1206,   1005,    804,    603,    402,    201,
-      0,   -201,   -402,   -603,   -804,  -1005,  -1206,  -1406,
-  -1607,  -1808,  -2009,  -2209,  -2410,  -2610,  -2811,  -3011,
-  -3211,  -3411,  -3611,  -3811,  -4011,  -4210,  -4409,  -4608,
-  -4807,  -5006,  -5205,  -5403,  -5601,  -5799,  -5997,  -6195,
-  -6392,  -6589,  -6786,  -6982,  -7179,  -7375,  -7571,  -7766,
-  -7961,  -8156,  -8351,  -8545,  -8739,  -8932,  -9126,  -9319,
-  -9511,  -9703,  -9895, -10087, -10278, -10469, -10659, -10849,
- -11038, -11227, -11416, -11604, -11792, -11980, -12166, -12353,
- -12539, -12724, -12909, -13094, -13278, -13462, -13645, -13827,
- -14009, -14191, -14372, -14552, -14732, -14911, -15090, -15268,
- -15446, -15623, -15799, -15975, -16150, -16325, -16499, -16672,
- -16845, -17017, -17189, -17360, -17530, -17699, -17868, -18036,
- -18204, -18371, -18537, -18702, -18867, -19031, -19194, -19357,
- -19519, -19680, -19840, -20000, -20159, -20317, -20474, -20631,
- -20787, -20942, -21096, -21249, -21402, -21554, -21705, -21855,
- -22004, -22153, -22301, -22448, -22594, -22739, -22883, -23027,
- -23169, -23311, -23452, -23592, -23731, -23869, -24006, -24143,
- -24278, -24413, -24546, -24679, -24811, -24942, -25072, -25201,
- -25329, -25456, -25582, -25707, -25831, -25954, -26077, -26198,
- -26318, -26437, -26556, -26673, -26789, -26905, -27019, -27132,
- -27244, -27355, -27466, -27575, -27683, -27790, -27896, -28001,
- -28105, -28208, -28309, -28410, -28510, -28608, -28706, -28802,
- -28897, -28992, -29085, -29177, -29268, -29358, -29446, -29534,
- -29621, -29706, -29790, -29873, -29955, -30036, -30116, -30195,
- -30272, -30349, -30424, -30498, -30571, -30643, -30713, -30783,
- -30851, -30918, -30984, -31049, -31113, -31175, -31236, -31297,
- -31356, -31413, -31470, -31525, -31580, -31633, -31684, -31735,
- -31785, -31833, -31880, -31926, -31970, -32014, -32056, -32097,
- -32137, -32176, -32213, -32249, -32284, -32318, -32350, -32382,
- -32412, -32441, -32468, -32495, -32520, -32544, -32567, -32588,
- -32609, -32628, -32646, -32662, -32678, -32692, -32705, -32717,
- -32727, -32736, -32744, -32751, -32757, -32761, -32764, -32766,
- -32767, -32766, -32764, -32761, -32757, -32751, -32744, -32736,
- -32727, -32717, -32705, -32692, -32678, -32662, -32646, -32628,
- -32609, -32588, -32567, -32544, -32520, -32495, -32468, -32441,
- -32412, -32382, -32350, -32318, -32284, -32249, -32213, -32176,
- -32137, -32097, -32056, -32014, -31970, -31926, -31880, -31833,
- -31785, -31735, -31684, -31633, -31580, -31525, -31470, -31413,
- -31356, -31297, -31236, -31175, -31113, -31049, -30984, -30918,
- -30851, -30783, -30713, -30643, -30571, -30498, -30424, -30349,
- -30272, -30195, -30116, -30036, -29955, -29873, -29790, -29706,
- -29621, -29534, -29446, -29358, -29268, -29177, -29085, -28992,
- -28897, -28802, -28706, -28608, -28510, -28410, -28309, -28208,
- -28105, -28001, -27896, -27790, -27683, -27575, -27466, -27355,
- -27244, -27132, -27019, -26905, -26789, -26673, -26556, -26437,
- -26318, -26198, -26077, -25954, -25831, -25707, -25582, -25456,
- -25329, -25201, -25072, -24942, -24811, -24679, -24546, -24413,
- -24278, -24143, -24006, -23869, -23731, -23592, -23452, -23311,
- -23169, -23027, -22883, -22739, -22594, -22448, -22301, -22153,
- -22004, -21855, -21705, -21554, -21402, -21249, -21096, -20942,
- -20787, -20631, -20474, -20317, -20159, -20000, -19840, -19680,
- -19519, -19357, -19194, -19031, -18867, -18702, -18537, -18371,
- -18204, -18036, -17868, -17699, -17530, -17360, -17189, -17017,
- -16845, -16672, -16499, -16325, -16150, -15975, -15799, -15623,
- -15446, -15268, -15090, -14911, -14732, -14552, -14372, -14191,
- -14009, -13827, -13645, -13462, -13278, -13094, -12909, -12724,
- -12539, -12353, -12166, -11980, -11792, -11604, -11416, -11227,
- -11038, -10849, -10659, -10469, -10278, -10087,  -9895,  -9703,
-  -9511,  -9319,  -9126,  -8932,  -8739,  -8545,  -8351,  -8156,
-  -7961,  -7766,  -7571,  -7375,  -7179,  -6982,  -6786,  -6589,
-  -6392,  -6195,  -5997,  -5799,  -5601,  -5403,  -5205,  -5006,
-  -4807,  -4608,  -4409,  -4210,  -4011,  -3811,  -3611,  -3411,
-  -3211,  -3011,  -2811,  -2610,  -2410,  -2209,  -2009,  -1808,
-  -1607,  -1406,  -1206,  -1005,   -804,   -603,   -402,   -201,
-};
diff --git a/src/common_audio/signal_processing_library/main/source/spl.gyp b/src/common_audio/signal_processing_library/main/source/spl.gyp
deleted file mode 100644
index 9c052fc..0000000
--- a/src/common_audio/signal_processing_library/main/source/spl.gyp
+++ /dev/null
@@ -1,84 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../../../common_settings.gypi', # Common settings
-  ],
-  'targets': [
-    {
-      'target_name': 'spl',
-      'type': '<(library)',
-      'include_dirs': [
-        '../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-        ],
-      },
-      'sources': [
-        '../interface/signal_processing_library.h',
-        '../interface/spl_inl.h',
-        'add_sat_w16.c',
-        'add_sat_w32.c',
-        'auto_corr_to_refl_coef.c',
-        'auto_correlation.c',
-        'complex_fft.c',
-        'complex_ifft.c',
-        'complex_bit_reverse.c',
-        'copy_set_operations.c',
-        'cos_table.c',
-        'cross_correlation.c',
-        'division_operations.c',
-        'dot_product_with_scale.c',
-        'downsample_fast.c',
-        'energy.c',
-        'filter_ar.c',
-        'filter_ar_fast_q12.c',
-        'filter_ma_fast_q12.c',
-        'get_hanning_window.c',
-        'get_scaling_square.c',
-        'get_size_in_bits.c',
-        'hanning_table.c',
-        'ilbc_specific_functions.c',
-        'levinson_durbin.c',
-        'lpc_to_refl_coef.c',
-        'min_max_operations.c',
-        'norm_u32.c',
-        'norm_w16.c',
-        'norm_w32.c',
-        'randn_table.c',
-        'randomization_functions.c',
-        'refl_coef_to_lpc.c',
-        'resample.c',
-        'resample_48khz.c',
-        'resample_by_2.c',
-        'resample_by_2_internal.c',
-        'resample_by_2_internal.h',
-        'resample_fractional.c',
-        'sin_table.c',
-        'sin_table_1024.c',
-        'spl_sqrt.c',
-        'spl_sqrt_floor.c',
-        'spl_version.c',
-        'splitting_filter.c',
-        'sqrt_of_one_minus_x_squared.c',
-        'sub_sat_w16.c',
-        'sub_sat_w32.c',
-        'vector_scaling_operations.c',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/common_audio/signal_processing_library/main/source/spl_sqrt.c b/src/common_audio/signal_processing_library/main/source/spl_sqrt.c
deleted file mode 100644
index cfe2cd3..0000000
--- a/src/common_audio/signal_processing_library/main/source/spl_sqrt.c
+++ /dev/null
@@ -1,184 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_Sqrt().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word32 WebRtcSpl_SqrtLocal(WebRtc_Word32 in);
-
-WebRtc_Word32 WebRtcSpl_SqrtLocal(WebRtc_Word32 in)
-{
-
-    WebRtc_Word16 x_half, t16;
-    WebRtc_Word32 A, B, x2;
-
-    /* The following block performs:
-     y=in/2
-     x=y-2^30
-     x_half=x/2^31
-     t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
-         + 0.875*((x_half)^5)
-     */
-
-    B = in;
-
-    B = WEBRTC_SPL_RSHIFT_W32(B, 1); // B = in/2
-    B = B - ((WebRtc_Word32)0x40000000); // B = in/2 - 1/2
-    x_half = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(B, 16);// x_half = x/2 = (in-1)/2
-    B = B + ((WebRtc_Word32)0x40000000); // B = 1 + x/2
-    B = B + ((WebRtc_Word32)0x40000000); // Add 0.5 twice (since 1.0 does not exist in Q31)
-
-    x2 = ((WebRtc_Word32)x_half) * ((WebRtc_Word32)x_half) * 2; // A = (x/2)^2
-    A = -x2; // A = -(x/2)^2
-    B = B + (A >> 1); // B = 1 + x/2 - 0.5*(x/2)^2
-
-    A = WEBRTC_SPL_RSHIFT_W32(A, 16);
-    A = A * A * 2; // A = (x/2)^4
-    t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
-    B = B + WEBRTC_SPL_MUL_16_16(-20480, t16) * 2; // B = B - 0.625*A
-    // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4
-
-    t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
-    A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = (x/2)^5
-    t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
-    B = B + WEBRTC_SPL_MUL_16_16(28672, t16) * 2; // B = B + 0.875*A
-    // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4 + 0.875*(x/2)^5
-
-    t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(x2, 16);
-    A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = x/2^3
-
-    B = B + (A >> 1); // B = B + 0.5*A
-    // After this, B = 1 + x/2 - 0.5*(x/2)^2 + 0.5*(x/2)^3 - 0.625*(x/2)^4 + 0.875*(x/2)^5
-
-    B = B + ((WebRtc_Word32)32768); // Round off bit
-
-    return B;
-}
-
-WebRtc_Word32 WebRtcSpl_Sqrt(WebRtc_Word32 value)
-{
-    /*
-     Algorithm:
-
-     Six term Taylor Series is used here to compute the square root of a number
-     y^0.5 = (1+x)^0.5 where x = y-1
-     = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
-     0.5 <= x < 1
-
-     Example of how the algorithm works, with ut=sqrt(in), and
-     with in=73632 and ut=271 (even shift value case):
-
-     in=73632
-     y= in/131072
-     x=y-1
-     t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
-     ut=t*(1/sqrt(2))*512
-
-     or:
-
-     in=73632
-     in2=73632*2^14
-     y= in2/2^31
-     x=y-1
-     t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
-     ut=t*(1/sqrt(2))
-     ut2=ut*2^9
-
-     which gives:
-
-     in  = 73632
-     in2 = 1206386688
-     y   = 0.56176757812500
-     x   = -0.43823242187500
-     t   = 0.74973506527313
-     ut  = 0.53014274874797
-     ut2 = 2.714330873589594e+002
-
-     or:
-
-     in=73632
-     in2=73632*2^14
-     y=in2/2
-     x=y-2^30
-     x_half=x/2^31
-     t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
-         + 0.875*((x_half)^5)
-     ut=t*(1/sqrt(2))
-     ut2=ut*2^9
-
-     which gives:
-
-     in  = 73632
-     in2 = 1206386688
-     y   = 603193344
-     x   = -470548480
-     x_half =  -0.21911621093750
-     t   = 0.74973506527313
-     ut  = 0.53014274874797
-     ut2 = 2.714330873589594e+002
-
-     */
-
-    WebRtc_Word16 x_norm, nshift, t16, sh;
-    WebRtc_Word32 A;
-
-    WebRtc_Word16 k_sqrt_2 = 23170; // 1/sqrt2 (==5a82)
-
-    A = value;
-
-    if (A == 0)
-        return (WebRtc_Word32)0; // sqrt(0) = 0
-
-    sh = WebRtcSpl_NormW32(A); // # shifts to normalize A
-    A = WEBRTC_SPL_LSHIFT_W32(A, sh); // Normalize A
-    if (A < (WEBRTC_SPL_WORD32_MAX - 32767))
-    {
-        A = A + ((WebRtc_Word32)32768); // Round off bit
-    } else
-    {
-        A = WEBRTC_SPL_WORD32_MAX;
-    }
-
-    x_norm = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16); // x_norm = AH
-
-    nshift = WEBRTC_SPL_RSHIFT_W16(sh, 1); // nshift = sh>>1
-    nshift = -nshift; // Negate the power for later de-normalization
-
-    A = (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x_norm, 16);
-    A = WEBRTC_SPL_ABS_W32(A); // A = abs(x_norm<<16)
-    A = WebRtcSpl_SqrtLocal(A); // A = sqrt(A)
-
-    if ((-2 * nshift) == sh)
-    { // Even shift value case
-
-        t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16); // t16 = AH
-
-        A = WEBRTC_SPL_MUL_16_16(k_sqrt_2, t16) * 2; // A = 1/sqrt(2)*t16
-        A = A + ((WebRtc_Word32)32768); // Round off
-        A = A & ((WebRtc_Word32)0x7fff0000); // Round off
-
-        A = WEBRTC_SPL_RSHIFT_W32(A, 15); // A = A>>16
-
-    } else
-    {
-        A = WEBRTC_SPL_RSHIFT_W32(A, 16); // A = A>>16
-    }
-
-    A = A & ((WebRtc_Word32)0x0000ffff);
-    A = (WebRtc_Word32)WEBRTC_SPL_SHIFT_W32(A, nshift); // De-normalize the result
-
-    return A;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/spl_sqrt_floor.c b/src/common_audio/signal_processing_library/main/source/spl_sqrt_floor.c
deleted file mode 100644
index aa36459..0000000
--- a/src/common_audio/signal_processing_library/main/source/spl_sqrt_floor.c
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the function WebRtcSpl_SqrtFloor().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define WEBRTC_SPL_SQRT_ITER(N)                 \
-  try1 = root + (1 << (N));                     \
-  if (value >= try1 << (N))                     \
-  {                                             \
-    value -= try1 << (N);                       \
-    root |= 2 << (N);                           \
-  }
-
-// (out) Square root of input parameter
-WebRtc_Word32 WebRtcSpl_SqrtFloor(WebRtc_Word32 value)
-{
-    // new routine for performance, 4 cycles/bit in ARM
-    // output precision is 16 bits
-
-    WebRtc_Word32 root = 0, try1;
-
-    WEBRTC_SPL_SQRT_ITER (15);
-    WEBRTC_SPL_SQRT_ITER (14);
-    WEBRTC_SPL_SQRT_ITER (13);
-    WEBRTC_SPL_SQRT_ITER (12);
-    WEBRTC_SPL_SQRT_ITER (11);
-    WEBRTC_SPL_SQRT_ITER (10);
-    WEBRTC_SPL_SQRT_ITER ( 9);
-    WEBRTC_SPL_SQRT_ITER ( 8);
-    WEBRTC_SPL_SQRT_ITER ( 7);
-    WEBRTC_SPL_SQRT_ITER ( 6);
-    WEBRTC_SPL_SQRT_ITER ( 5);
-    WEBRTC_SPL_SQRT_ITER ( 4);
-    WEBRTC_SPL_SQRT_ITER ( 3);
-    WEBRTC_SPL_SQRT_ITER ( 2);
-    WEBRTC_SPL_SQRT_ITER ( 1);
-    WEBRTC_SPL_SQRT_ITER ( 0);
-
-    return root >> 1;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/spl_version.c b/src/common_audio/signal_processing_library/main/source/spl_version.c
deleted file mode 100644
index 936925e..0000000
--- a/src/common_audio/signal_processing_library/main/source/spl_version.c
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_get_version().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include <string.h>
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_get_version(char* version, WebRtc_Word16 length_in_bytes)
-{
-    strncpy(version, "1.2.0", length_in_bytes);
-    return 0;
-}
diff --git a/src/common_audio/signal_processing_library/main/source/splitting_filter.c b/src/common_audio/signal_processing_library/main/source/splitting_filter.c
deleted file mode 100644
index 98442f4..0000000
--- a/src/common_audio/signal_processing_library/main/source/splitting_filter.c
+++ /dev/null
@@ -1,200 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the splitting filter functions.
- *
- */
-
-#include "signal_processing_library.h"
-
-// Number of samples in a low/high-band frame.
-enum
-{
-    kBandFrameLength = 160
-};
-
-// QMF filter coefficients in Q16.
-static const WebRtc_UWord16 WebRtcSpl_kAllPassFilter1[3] = {6418, 36982, 57261};
-static const WebRtc_UWord16 WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcSpl_AllPassQMF(...)
-//
-// Allpass filter used by the analysis and synthesis parts of the QMF filter.
-//
-// Input:
-//    - in_data             : Input data sequence (Q10)
-//    - data_length         : Length of data sequence (>2)
-//    - filter_coefficients : Filter coefficients (length 3, Q16)
-//
-// Input & Output:
-//    - filter_state        : Filter state (length 6, Q10).
-//
-// Output:
-//    - out_data            : Output data sequence (Q10), length equal to
-//                            |data_length|
-//
-
-void WebRtcSpl_AllPassQMF(WebRtc_Word32* in_data, const WebRtc_Word16 data_length,
-                          WebRtc_Word32* out_data, const WebRtc_UWord16* filter_coefficients,
-                          WebRtc_Word32* filter_state)
-{
-    // The procedure is to filter the input with three first order all pass filters
-    // (cascade operations).
-    //
-    //         a_3 + q^-1    a_2 + q^-1    a_1 + q^-1
-    // y[n] =  -----------   -----------   -----------   x[n]
-    //         1 + a_3q^-1   1 + a_2q^-1   1 + a_1q^-1
-    //
-    // The input vector |filter_coefficients| includes these three filter coefficients.
-    // The filter state contains the in_data state, in_data[-1], followed by
-    // the out_data state, out_data[-1]. This is repeated for each cascade.
-    // The first cascade filter will filter the |in_data| and store the output in
-    // |out_data|. The second will the take the |out_data| as input and make an
-    // intermediate storage in |in_data|, to save memory. The third, and final, cascade
-    // filter operation takes the |in_data| (which is the output from the previous cascade
-    // filter) and store the output in |out_data|.
-    // Note that the input vector values are changed during the process.
-    WebRtc_Word16 k;
-    WebRtc_Word32 diff;
-    // First all-pass cascade; filter from in_data to out_data.
-
-    // Let y_i[n] indicate the output of cascade filter i (with filter coefficient a_i) at
-    // vector position n. Then the final output will be y[n] = y_3[n]
-
-    // First loop, use the states stored in memory.
-    // "diff" should be safe from wrap around since max values are 2^25
-    diff = WEBRTC_SPL_SUB_SAT_W32(in_data[0], filter_state[1]); // = (x[0] - y_1[-1])
-    // y_1[0] =  x[-1] + a_1 * (x[0] - y_1[-1])
-    out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, filter_state[0]);
-
-    // For the remaining loops, use previous values.
-    for (k = 1; k < data_length; k++)
-    {
-        diff = WEBRTC_SPL_SUB_SAT_W32(in_data[k], out_data[k - 1]); // = (x[n] - y_1[n-1])
-        // y_1[n] =  x[n-1] + a_1 * (x[n] - y_1[n-1])
-        out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, in_data[k - 1]);
-    }
-
-    // Update states.
-    filter_state[0] = in_data[data_length - 1]; // x[N-1], becomes x[-1] next time
-    filter_state[1] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
-
-    // Second all-pass cascade; filter from out_data to in_data.
-    diff = WEBRTC_SPL_SUB_SAT_W32(out_data[0], filter_state[3]); // = (y_1[0] - y_2[-1])
-    // y_2[0] =  y_1[-1] + a_2 * (y_1[0] - y_2[-1])
-    in_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, filter_state[2]);
-    for (k = 1; k < data_length; k++)
-    {
-        diff = WEBRTC_SPL_SUB_SAT_W32(out_data[k], in_data[k - 1]); // =(y_1[n] - y_2[n-1])
-        // y_2[0] =  y_1[-1] + a_2 * (y_1[0] - y_2[-1])
-        in_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, out_data[k-1]);
-    }
-
-    filter_state[2] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
-    filter_state[3] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
-
-    // Third all-pass cascade; filter from in_data to out_data.
-    diff = WEBRTC_SPL_SUB_SAT_W32(in_data[0], filter_state[5]); // = (y_2[0] - y[-1])
-    // y[0] =  y_2[-1] + a_3 * (y_2[0] - y[-1])
-    out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, filter_state[4]);
-    for (k = 1; k < data_length; k++)
-    {
-        diff = WEBRTC_SPL_SUB_SAT_W32(in_data[k], out_data[k - 1]); // = (y_2[n] - y[n-1])
-        // y[n] =  y_2[n-1] + a_3 * (y_2[n] - y[n-1])
-        out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, in_data[k-1]);
-    }
-    filter_state[4] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
-    filter_state[5] = out_data[data_length - 1]; // y[N-1], becomes y[-1] next time
-}
-
-void WebRtcSpl_AnalysisQMF(const WebRtc_Word16* in_data, WebRtc_Word16* low_band,
-                           WebRtc_Word16* high_band, WebRtc_Word32* filter_state1,
-                           WebRtc_Word32* filter_state2)
-{
-    WebRtc_Word16 i;
-    WebRtc_Word16 k;
-    WebRtc_Word32 tmp;
-    WebRtc_Word32 half_in1[kBandFrameLength];
-    WebRtc_Word32 half_in2[kBandFrameLength];
-    WebRtc_Word32 filter1[kBandFrameLength];
-    WebRtc_Word32 filter2[kBandFrameLength];
-
-    // Split even and odd samples. Also shift them to Q10.
-    for (i = 0, k = 0; i < kBandFrameLength; i++, k += 2)
-    {
-        half_in2[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k], 10);
-        half_in1[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k + 1], 10);
-    }
-
-    // All pass filter even and odd samples, independently.
-    WebRtcSpl_AllPassQMF(half_in1, kBandFrameLength, filter1, WebRtcSpl_kAllPassFilter1,
-                         filter_state1);
-    WebRtcSpl_AllPassQMF(half_in2, kBandFrameLength, filter2, WebRtcSpl_kAllPassFilter2,
-                         filter_state2);
-
-    // Take the sum and difference of filtered version of odd and even
-    // branches to get upper & lower band.
-    for (i = 0; i < kBandFrameLength; i++)
-    {
-        tmp = filter1[i] + filter2[i] + 1024;
-        tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
-        low_band[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
-                tmp, WEBRTC_SPL_WORD16_MIN);
-
-        tmp = filter1[i] - filter2[i] + 1024;
-        tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
-        high_band[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
-                tmp, WEBRTC_SPL_WORD16_MIN);
-    }
-}
-
-void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band, const WebRtc_Word16* high_band,
-                            WebRtc_Word16* out_data, WebRtc_Word32* filter_state1,
-                            WebRtc_Word32* filter_state2)
-{
-    WebRtc_Word32 tmp;
-    WebRtc_Word32 half_in1[kBandFrameLength];
-    WebRtc_Word32 half_in2[kBandFrameLength];
-    WebRtc_Word32 filter1[kBandFrameLength];
-    WebRtc_Word32 filter2[kBandFrameLength];
-    WebRtc_Word16 i;
-    WebRtc_Word16 k;
-
-    // Obtain the sum and difference channels out of upper and lower-band channels.
-    // Also shift to Q10 domain.
-    for (i = 0; i < kBandFrameLength; i++)
-    {
-        tmp = (WebRtc_Word32)low_band[i] + (WebRtc_Word32)high_band[i];
-        half_in1[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
-        tmp = (WebRtc_Word32)low_band[i] - (WebRtc_Word32)high_band[i];
-        half_in2[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
-    }
-
-    // all-pass filter the sum and difference channels
-    WebRtcSpl_AllPassQMF(half_in1, kBandFrameLength, filter1, WebRtcSpl_kAllPassFilter2,
-                         filter_state1);
-    WebRtcSpl_AllPassQMF(half_in2, kBandFrameLength, filter2, WebRtcSpl_kAllPassFilter1,
-                         filter_state2);
-
-    // The filtered signals are even and odd samples of the output. Combine
-    // them. The signals are Q10 should shift them back to Q0 and take care of
-    // saturation.
-    for (i = 0, k = 0; i < kBandFrameLength; i++)
-    {
-        tmp = WEBRTC_SPL_RSHIFT_W32(filter2[i] + 512, 10);
-        out_data[k++] = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, tmp, -32768);
-
-        tmp = WEBRTC_SPL_RSHIFT_W32(filter1[i] + 512, 10);
-        out_data[k++] = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, tmp, -32768);
-    }
-
-}
diff --git a/src/common_audio/signal_processing_library/main/source/sqrt_of_one_minus_x_squared.c b/src/common_audio/signal_processing_library/main/source/sqrt_of_one_minus_x_squared.c
deleted file mode 100644
index 9fb2c73..0000000
--- a/src/common_audio/signal_processing_library/main/source/sqrt_of_one_minus_x_squared.c
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_SqrtOfOneMinusXSquared().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_SqrtOfOneMinusXSquared(WebRtc_Word16 *xQ15, int vector_length,
-                                      WebRtc_Word16 *yQ15)
-{
-    WebRtc_Word32 sq;
-    int m;
-    WebRtc_Word16 tmp;
-
-    for (m = 0; m < vector_length; m++)
-    {
-        tmp = xQ15[m];
-        sq = WEBRTC_SPL_MUL_16_16(tmp, tmp); // x^2 in Q30
-        sq = 1073741823 - sq; // 1-x^2, where 1 ~= 0.99999999906 is 1073741823 in Q30
-        sq = WebRtcSpl_Sqrt(sq); // sqrt(1-x^2) in Q15
-        yQ15[m] = (WebRtc_Word16)sq;
-    }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/sub_sat_w16.c b/src/common_audio/signal_processing_library/main/source/sub_sat_w16.c
deleted file mode 100644
index a48c3d5..0000000
--- a/src/common_audio/signal_processing_library/main/source/sub_sat_w16.c
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_SubSatW16().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-#ifndef XSCALE_OPT
-
-WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1, WebRtc_Word16 var2)
-{
-    WebRtc_Word32 l_diff;
-    WebRtc_Word16 s_diff;
-
-    // perform subtraction
-    l_diff = (WebRtc_Word32)var1 - (WebRtc_Word32)var2;
-
-    // default setting
-    s_diff = (WebRtc_Word16)l_diff;
-
-    // check for overflow
-    if (l_diff > (WebRtc_Word32)32767)
-        s_diff = (WebRtc_Word16)32767;
-
-    // check for underflow
-    if (l_diff < (WebRtc_Word32)-32768)
-        s_diff = (WebRtc_Word16)-32768;
-
-    return s_diff;
-}
-
-#else
-#pragma message(">> WebRtcSpl_SubSatW16.c is excluded from this build")
-#endif // XSCALE_OPT
-#endif // SPL_NO_DOUBLE_IMPLEMENTATIONS
diff --git a/src/common_audio/signal_processing_library/main/source/sub_sat_w32.c b/src/common_audio/signal_processing_library/main/source/sub_sat_w32.c
deleted file mode 100644
index add3675..0000000
--- a/src/common_audio/signal_processing_library/main/source/sub_sat_w32.c
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_SubSatW32().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifndef SPL_NO_DOUBLE_IMPLEMENTATIONS
-
-WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 var1, WebRtc_Word32 var2)
-{
-    WebRtc_Word32 l_diff;
-
-    // perform subtraction
-    l_diff = var1 - var2;
-
-    // check for underflow
-    if ((var1 < 0) && (var2 > 0) && (l_diff > 0))
-        l_diff = (WebRtc_Word32)0x80000000;
-    // check for overflow
-    if ((var1 > 0) && (var2 < 0) && (l_diff < 0))
-        l_diff = (WebRtc_Word32)0x7FFFFFFF;
-
-    return l_diff;
-}
-
-#endif
diff --git a/src/common_audio/signal_processing_library/main/source/vector_scaling_operations.c b/src/common_audio/signal_processing_library/main/source/vector_scaling_operations.c
deleted file mode 100644
index 47362ee..0000000
--- a/src/common_audio/signal_processing_library/main/source/vector_scaling_operations.c
+++ /dev/null
@@ -1,151 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the functions
- * WebRtcSpl_VectorBitShiftW16()
- * WebRtcSpl_VectorBitShiftW32()
- * WebRtcSpl_VectorBitShiftW32ToW16()
- * WebRtcSpl_ScaleVector()
- * WebRtcSpl_ScaleVectorWithSat()
- * WebRtcSpl_ScaleAndAddVectors()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_VectorBitShiftW16(WebRtc_Word16 *res,
-                             WebRtc_Word16 length,
-                             G_CONST WebRtc_Word16 *in,
-                             WebRtc_Word16 right_shifts)
-{
-    int i;
-
-    if (right_shifts > 0)
-    {
-        for (i = length; i > 0; i--)
-        {
-            (*res++) = ((*in++) >> right_shifts);
-        }
-    } else
-    {
-        for (i = length; i > 0; i--)
-        {
-            (*res++) = ((*in++) << (-right_shifts));
-        }
-    }
-}
-
-void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32 *out_vector,
-                             WebRtc_Word16 vector_length,
-                             G_CONST WebRtc_Word32 *in_vector,
-                             WebRtc_Word16 right_shifts)
-{
-    int i;
-
-    if (right_shifts > 0)
-    {
-        for (i = vector_length; i > 0; i--)
-        {
-            (*out_vector++) = ((*in_vector++) >> right_shifts);
-        }
-    } else
-    {
-        for (i = vector_length; i > 0; i--)
-        {
-            (*out_vector++) = ((*in_vector++) << (-right_shifts));
-        }
-    }
-}
-
-void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16 *res,
-                                  WebRtc_Word16 length,
-                                  G_CONST WebRtc_Word32 *in,
-                                  WebRtc_Word16 right_shifts)
-{
-    int i;
-
-    if (right_shifts >= 0)
-    {
-        for (i = length; i > 0; i--)
-        {
-            (*res++) = (WebRtc_Word16)((*in++) >> right_shifts);
-        }
-    } else
-    {
-        WebRtc_Word16 left_shifts = -right_shifts;
-        for (i = length; i > 0; i--)
-        {
-            (*res++) = (WebRtc_Word16)((*in++) << left_shifts);
-        }
-    }
-}
-
-void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector,
-                           WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
-                           WebRtc_Word16 right_shifts)
-{
-    // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
-    int i;
-    G_CONST WebRtc_Word16 *inptr;
-    WebRtc_Word16 *outptr;
-
-    inptr = in_vector;
-    outptr = out_vector;
-
-    for (i = 0; i < in_vector_length; i++)
-    {
-        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
-    }
-}
-
-void WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector,
-                                 WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
-                                 WebRtc_Word16 right_shifts)
-{
-    // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
-    int i;
-    WebRtc_Word32 tmpW32;
-    G_CONST WebRtc_Word16 *inptr;
-    WebRtc_Word16 *outptr;
-
-    inptr = in_vector;
-    outptr = out_vector;
-
-    for (i = 0; i < in_vector_length; i++)
-    {
-        tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
-        ( *outptr++) = (WebRtc_Word16)WEBRTC_SPL_SAT(32767, tmpW32, -32768);
-    }
-}
-
-void WebRtcSpl_ScaleAndAddVectors(G_CONST WebRtc_Word16 *in1, WebRtc_Word16 gain1, int shift1,
-                                  G_CONST WebRtc_Word16 *in2, WebRtc_Word16 gain2, int shift2,
-                                  WebRtc_Word16 *out, int vector_length)
-{
-    // Performs vector operation: out = (gain1*in1)>>shift1 + (gain2*in2)>>shift2
-    int i;
-    G_CONST WebRtc_Word16 *in1ptr;
-    G_CONST WebRtc_Word16 *in2ptr;
-    WebRtc_Word16 *outptr;
-
-    in1ptr = in1;
-    in2ptr = in2;
-    outptr = out;
-
-    for (i = 0; i < vector_length; i++)
-    {
-        (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gain1, *in1ptr++, shift1)
-                + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gain2, *in2ptr++, shift2);
-    }
-}
diff --git a/src/common_audio/signal_processing_library/main/source/webrtc_fft_4ofq14_gcc_android.s b/src/common_audio/signal_processing_library/main/source/webrtc_fft_4ofq14_gcc_android.s
deleted file mode 100644
index c1a893b..0000000
--- a/src/common_audio/signal_processing_library/main/source/webrtc_fft_4ofq14_gcc_android.s
+++ /dev/null
@@ -1,227 +0,0 @@
-  .globl FFT_4OFQ14
-
-FFT_4OFQ14:
-  stmdb       sp!, {r4 - r11, lr}
-  ldr         lr, =s_Q14S_8
-  ldr         lr, [lr]
-  cmp         r2, lr
-  movgt       r0, #1
-  ldmgtia     sp!, {r4 - r11, pc}
-  stmdb       sp!, {r1, r2}
-  mov         r3, #0
-  mov         r2, r2
-
-LBL1:
-  add         r12, r0, r3, lsl #2
-  add         r12, r12, r2, lsr #1
-  ldrsh       r5, [r12, #2]
-  ldrsh       r4, [r12], +r2
-  ldrsh       r9, [r12, #2]
-  ldrsh       r8, [r12], +r2
-  ldrsh       r7, [r12, #2]
-  ldrsh       r6, [r12], +r2
-  ldrsh       r11, [r12, #2]
-  ldrsh       r10, [r12], +r2
-  add         r4, r4, r6
-  add         r5, r5, r7
-  sub         r6, r4, r6, lsl #1
-  sub         r7, r5, r7, lsl #1
-  sub         r12, r8, r10
-  sub         lr, r9, r11
-  add         r10, r8, r10
-  add         r11, r9, r11
-  sub         r9, r4, r10
-  sub         r8, r5, r11
-  add         r4, r4, r10
-  add         r5, r5, r11
-  sub         r10, r6, lr
-  add         r11, r7, r12
-  add         r6, r6, lr
-  sub         r7, r7, r12
-  ldr         lr, =t_Q14R_rad8
-  ldrsh       lr, [lr]
-  stmdb       sp!, {r2}
-  add         r12, r6, r7
-  mul         r6, r12, lr
-  rsb         r12, r12, r7, lsl #1
-  mul         r7, r12, lr
-  sub         r12, r11, r10
-  mul         r10, r12, lr
-  sub         r12, r12, r11, lsl #1
-  mul         r11, r12, lr
-  ldmia       sp!, {r2}
-  stmdb       sp!, {r4 - r11}
-  add         r4, r0, r3, lsl #2
-  ldrsh       r7, [r4, #2]
-  ldrsh       r6, [r4], +r2
-  ldrsh       r11, [r4, #2]
-  ldrsh       r10, [r4], +r2
-  ldrsh       r9, [r4, #2]
-  ldrsh       r8, [r4], +r2
-  ldrsh       lr, [r4, #2]
-  ldrsh       r12, [r4], +r2
-  mov         r7, r7, asr #3
-  mov         r6, r6, asr #3
-  add         r6, r6, r8, asr #3
-  add         r7, r7, r9, asr #3
-  sub         r8, r6, r8, asr #2
-  sub         r9, r7, r9, asr #2
-  sub         r4, r10, r12
-  sub         r5, r11, lr
-  add         r10, r10, r12
-  add         r11, r11, lr
-  add         r6, r6, r10, asr #3
-  add         r7, r7, r11, asr #3
-  sub         r10, r6, r10, asr #2
-  sub         r11, r7, r11, asr #2
-  sub         r12, r8, r5, asr #3
-  add         lr, r9, r4, asr #3
-  add         r8, r8, r5, asr #3
-  sub         r9, r9, r4, asr #3
-  ldmia       sp!, {r4, r5}
-  add         r6, r6, r4, asr #3
-  add         r7, r7, r5, asr #3
-  sub         r4, r6, r4, asr #2
-  sub         r5, r7, r5, asr #2
-  strh        r7, [r1, #2]
-  strh        r6, [r1], #4
-  ldmia       sp!, {r6, r7}
-  add         r8, r8, r6, asr #17
-  add         r9, r9, r7, asr #17
-  sub         r6, r8, r6, asr #16
-  sub         r7, r9, r7, asr #16
-  strh        r9, [r1, #2]
-  strh        r8, [r1], #4
-  ldmia       sp!, {r8, r9}
-  add         r10, r10, r8, asr #3
-  sub         r11, r11, r9, asr #3
-  sub         r8, r10, r8, asr #2
-  add         r9, r11, r9, asr #2
-  strh        r11, [r1, #2]
-  strh        r10, [r1], #4
-  ldmia       sp!, {r10, r11}
-  add         r12, r12, r10, asr #17
-  add         lr, lr, r11, asr #17
-  sub         r10, r12, r10, asr #16
-  sub         r11, lr, r11, asr #16
-  strh        lr, [r1, #2]
-  strh        r12, [r1], #4
-  strh        r5, [r1, #2]
-  strh        r4, [r1], #4
-  strh        r7, [r1, #2]
-  strh        r6, [r1], #4
-  strh        r9, [r1, #2]
-  strh        r8, [r1], #4
-  strh        r11, [r1, #2]
-  strh        r10, [r1], #4
-  eor         r3, r3, r2, lsr #4
-  tst         r3, r2, lsr #4
-  bne         LBL1
-
-  eor         r3, r3, r2, lsr #5
-  tst         r3, r2, lsr #5
-  bne         LBL1
-
-  mov         r12, r2, lsr #6
-
-LBL2:
-  eor         r3, r3, r12
-  tst         r3, r12
-  bne         LBL1
-
-  movs        r12, r12, lsr #1
-  bne         LBL2
-
-  ldmia       sp!, {r1, r2}
-  mov         r3, r2, lsr #3
-  mov         r2, #0x20
-  ldr         r0, =t_Q14S_8
-  cmp         r3, #1
-  beq         LBL3
-
-LBL6:
-  mov         r3, r3, lsr #2
-  stmdb       sp!, {r1, r3}
-  add         r12, r2, r2, lsl #1
-  add         r1, r1, r12
-  sub         r3, r3, #1, 16
-
-LBL5:
-  add         r3, r3, r2, lsl #14
-
-LBL4:
-  ldrsh       r6, [r0], #2
-  ldrsh       r7, [r0], #2
-  ldrsh       r8, [r0], #2
-  ldrsh       r9, [r0], #2
-  ldrsh       r10, [r0], #2
-  ldrsh       r11, [r0], #2
-  ldrsh       r5, [r1, #2]
-  ldrsh       r4, [r1], -r2
-  sub         lr, r5, r4
-  mul         r12, lr, r11
-  add         lr, r10, r11, lsl #1
-  mla         r11, r5, r10, r12
-  mla         r10, r4, lr, r12
-  ldrsh       r5, [r1, #2]
-  ldrsh       r4, [r1], -r2
-  sub         lr, r5, r4
-  mul         r12, lr, r9
-  add         lr, r8, r9, lsl #1
-  mla         r9, r5, r8, r12
-  mla         r8, r4, lr, r12
-  ldrsh       r5, [r1, #2]
-  ldrsh       r4, [r1], -r2
-  sub         lr, r5, r4
-  mul         r12, lr, r7
-  add         lr, r6, r7, lsl #1
-  mla         r7, r5, r6, r12
-  mla         r6, r4, lr, r12
-  ldrsh       r5, [r1, #2]
-  ldrsh       r4, [r1]
-  mov         r5, r5, asr #2
-  mov         r4, r4, asr #2
-  add         r12, r4, r6, asr #16
-  add         lr, r5, r7, asr #16
-  sub         r4, r4, r6, asr #16
-  sub         r5, r5, r7, asr #16
-  add         r6, r8, r10
-  add         r7, r9, r11
-  sub         r8, r8, r10
-  sub         r9, r9, r11
-  add         r10, r12, r6, asr #16
-  add         r11, lr, r7, asr #16
-  strh        r11, [r1, #2]
-  strh        r10, [r1], +r2
-  add         r10, r4, r9, asr #16
-  sub         r11, r5, r8, asr #16
-  strh        r11, [r1, #2]
-  strh        r10, [r1], +r2
-  sub         r10, r12, r6, asr #16
-  sub         r11, lr, r7, asr #16
-  strh        r11, [r1, #2]
-  strh        r10, [r1], +r2
-  sub         r10, r4, r9, asr #16
-  add         r11, r5, r8, asr #16
-  strh        r11, [r1, #2]
-  strh        r10, [r1], #4
-  subs        r3, r3, #1, 16
-  bge         LBL4
-  add         r12, r2, r2, lsl #1
-  add         r1, r1, r12
-  sub         r0, r0, r12
-  sub         r3, r3, #1
-  movs        lr, r3, lsl #16
-  bne         LBL5
-  add         r0, r0, r12
-  ldmia       sp!, {r1, r3}
-  mov         r2, r2, lsl #2
-  cmp         r3, #2
-  bgt         LBL6
-
-LBL3:
-  mov         r0, #0
-  ldmia       sp!, {r4 - r11, pc}
-  andeq       r3, r1, r0, lsr #32
-  andeq       r10, r1, r12, ror #31
-  andeq       r3, r1, r8, lsr #32
diff --git a/src/common_audio/signal_processing_library/main/source/webrtc_fft_4oiq14_gcc_android.s b/src/common_audio/signal_processing_library/main/source/webrtc_fft_4oiq14_gcc_android.s
deleted file mode 100644
index cc93291..0000000
--- a/src/common_audio/signal_processing_library/main/source/webrtc_fft_4oiq14_gcc_android.s
+++ /dev/null
@@ -1,221 +0,0 @@
-  .globl FFT_4OIQ14
-
-FFT_4OIQ14:
-  stmdb       sp!, {r4 - r11, lr}
-  ldr         lr, =s_Q14S_8
-  ldr         lr, [lr]
-  cmp         r2, lr
-  movgt       r0, #1
-  ldmgtia     sp!, {r4 - r11, pc}
-  stmdb       sp!, {r1, r2}
-  mov         r3, #0
-  mov         r2, r2
-
-LBL1:
-  add         r12, r0, r3, lsl #2
-  add         r12, r12, r2, lsr #1
-  ldrsh       r5, [r12, #2]
-  ldrsh       r4, [r12], +r2
-  ldrsh       r9, [r12, #2]
-  ldrsh       r8, [r12], +r2
-  ldrsh       r7, [r12, #2]
-  ldrsh       r6, [r12], +r2
-  ldrsh       r11, [r12, #2]
-  ldrsh       r10, [r12], +r2
-  add         r4, r4, r6
-  add         r5, r5, r7
-  sub         r6, r4, r6, lsl #1
-  sub         r7, r5, r7, lsl #1
-  sub         r12, r8, r10
-  sub         lr, r9, r11
-  add         r10, r8, r10
-  add         r11, r9, r11
-  sub         r9, r4, r10
-  sub         r8, r5, r11
-  add         r4, r4, r10
-  add         r5, r5, r11
-  add         r10, r6, lr
-  sub         r11, r7, r12
-  sub         r6, r6, lr
-  add         r7, r7, r12
-  ldr         lr, =t_Q14R_rad8
-  ldrsh       lr, [lr]
-  stmdb       sp!, {r2}
-  sub         r12, r6, r7
-  mul         r6, r12, lr
-  add         r12, r12, r7, lsl #1
-  mul         r7, r12, lr
-  sub         r12, r10, r11
-  mul         r11, r12, lr
-  sub         r12, r12, r10, lsl #1
-  mul         r10, r12, lr
-  ldmia       sp!, {r2}
-  stmdb       sp!, {r4 - r11}
-  add         r4, r0, r3, lsl #2
-  ldrsh       r7, [r4, #2]
-  ldrsh       r6, [r4], +r2
-  ldrsh       r11, [r4, #2]
-  ldrsh       r10, [r4], +r2
-  ldrsh       r9, [r4, #2]
-  ldrsh       r8, [r4], +r2
-  ldrsh       lr, [r4, #2]
-  ldrsh       r12, [r4], +r2
-  add         r6, r6, r8
-  add         r7, r7, r9
-  sub         r8, r6, r8, lsl #1
-  sub         r9, r7, r9, lsl #1
-  sub         r4, r10, r12
-  sub         r5, r11, lr
-  add         r10, r10, r12
-  add         r11, r11, lr
-  add         r6, r6, r10
-  add         r7, r7, r11
-  sub         r10, r6, r10, lsl #1
-  sub         r11, r7, r11, lsl #1
-  add         r12, r8, r5
-  sub         lr, r9, r4
-  sub         r8, r8, r5
-  add         r9, r9, r4
-  ldmia       sp!, {r4, r5}
-  add         r6, r6, r4
-  add         r7, r7, r5
-  sub         r4, r6, r4, lsl #1
-  sub         r5, r7, r5, lsl #1
-  strh        r7, [r1, #2]
-  strh        r6, [r1], #4
-  ldmia       sp!, {r6, r7}
-  add         r8, r8, r6, asr #14
-  add         r9, r9, r7, asr #14
-  sub         r6, r8, r6, asr #13
-  sub         r7, r9, r7, asr #13
-  strh        r9, [r1, #2]
-  strh        r8, [r1], #4
-  ldmia       sp!, {r8, r9}
-  sub         r10, r10, r8
-  add         r11, r11, r9
-  add         r8, r10, r8, lsl #1
-  sub         r9, r11, r9, lsl #1
-  strh        r11, [r1, #2]
-  strh        r10, [r1], #4
-  ldmia       sp!, {r10, r11}
-  add         r12, r12, r10, asr #14
-  add         lr, lr, r11, asr #14
-  sub         r10, r12, r10, asr #13
-  sub         r11, lr, r11, asr #13
-  strh        lr, [r1, #2]
-  strh        r12, [r1], #4
-  strh        r5, [r1, #2]
-  strh        r4, [r1], #4
-  strh        r7, [r1, #2]
-  strh        r6, [r1], #4
-  strh        r9, [r1, #2]
-  strh        r8, [r1], #4
-  strh        r11, [r1, #2]
-  strh        r10, [r1], #4
-  eor         r3, r3, r2, lsr #4
-  tst         r3, r2, lsr #4
-  bne         LBL1
-  eor         r3, r3, r2, lsr #5
-  tst         r3, r2, lsr #5
-  bne         LBL1
-  mov         r12, r2, lsr #6
-
-
-LBL2:
-  eor         r3, r3, r12
-  tst         r3, r12
-  bne         LBL1
-  movs        r12, r12, lsr #1
-  bne         LBL2
-  ldmia       sp!, {r1, r2}
-  mov         r3, r2, lsr #3
-  mov         r2, #0x20
-  ldr         r0, =t_Q14S_8
-  cmp         r3, #1
-  beq         LBL3
-
-LBL6:
-  mov         r3, r3, lsr #2
-  stmdb       sp!, {r1, r3}
-  add         r12, r2, r2, lsl #1
-  add         r1, r1, r12
-  sub         r3, r3, #1, 16
-
-LBL5:
-  add         r3, r3, r2, lsl #14
-
-LBL4:
-  ldrsh       r6, [r0], #2
-  ldrsh       r7, [r0], #2
-  ldrsh       r8, [r0], #2
-  ldrsh       r9, [r0], #2
-  ldrsh       r10, [r0], #2
-  ldrsh       r11, [r0], #2
-  ldrsh       r5, [r1, #2]
-  ldrsh       r4, [r1], -r2
-  sub         lr, r4, r5
-  mul         r12, lr, r11
-  add         r11, r10, r11, lsl #1
-  mla         r10, r4, r10, r12
-  mla         r11, r5, r11, r12
-  ldrsh       r5, [r1, #2]
-  ldrsh       r4, [r1], -r2
-  sub         lr, r4, r5
-  mul         r12, lr, r9
-  add         r9, r8, r9, lsl #1
-  mla         r8, r4, r8, r12
-  mla         r9, r5, r9, r12
-  ldrsh       r5, [r1, #2]
-  ldrsh       r4, [r1], -r2
-  sub         lr, r4, r5
-  mul         r12, lr, r7
-  add         r7, r6, r7, lsl #1
-  mla         r6, r4, r6, r12
-  mla         r7, r5, r7, r12
-  ldrsh       r5, [r1, #2]
-  ldrsh       r4, [r1]
-  add         r12, r4, r6, asr #14
-  add         lr, r5, r7, asr #14
-  sub         r4, r4, r6, asr #14
-  sub         r5, r5, r7, asr #14
-  add         r6, r8, r10
-  add         r7, r9, r11
-  sub         r8, r8, r10
-  sub         r9, r9, r11
-  add         r10, r12, r6, asr #14
-  add         r11, lr, r7, asr #14
-  strh        r11, [r1, #2]
-  strh        r10, [r1], +r2
-  sub         r10, r4, r9, asr #14
-  add         r11, r5, r8, asr #14
-  strh        r11, [r1, #2]
-  strh        r10, [r1], +r2
-  sub         r10, r12, r6, asr #14
-  sub         r11, lr, r7, asr #14
-  strh        r11, [r1, #2]
-  strh        r10, [r1], +r2
-  add         r10, r4, r9, asr #14
-  sub         r11, r5, r8, asr #14
-  strh        r11, [r1, #2]
-  strh        r10, [r1], #4
-  subs        r3, r3, #1, 16
-  bge         LBL4
-  add         r12, r2, r2, lsl #1
-  add         r1, r1, r12
-  sub         r0, r0, r12
-  sub         r3, r3, #1
-  movs        lr, r3, lsl #16
-  bne         LBL5
-  add         r0, r0, r12
-  ldmia       sp!, {r1, r3}
-  mov         r2, r2, lsl #2
-  cmp         r3, #2
-  bgt         LBL6
-
-LBL3:
-  mov         r0, #0
-  ldmia       sp!, {r4 - r11, pc}
-  andeq       r3, r1, r0, lsr #32
-  andeq       r10, r1, r12, ror #31
-  andeq       r3, r1, r8, lsr #32
-
diff --git a/src/common_audio/signal_processing_library/main/source/webrtc_fft_t_1024_8.c b/src/common_audio/signal_processing_library/main/source/webrtc_fft_t_1024_8.c
deleted file mode 100644
index b587380..0000000
--- a/src/common_audio/signal_processing_library/main/source/webrtc_fft_t_1024_8.c
+++ /dev/null
@@ -1,704 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the Q14 radix-8 tables used in ARM9e optimizations.
- *
- */
-
-extern const int s_Q14S_8;
-const int s_Q14S_8 = 1024;
-extern const unsigned short t_Q14S_8[2032];
-const unsigned short t_Q14S_8[2032] = {
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
-  0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
-  0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
-  0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
-  0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
-  0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
-  0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x396b,0x0646 ,0x3cc8,0x0324 ,0x35eb,0x0964 ,
-  0x3249,0x0c7c ,0x396b,0x0646 ,0x2aaa,0x1294 ,
-  0x2aaa,0x1294 ,0x35eb,0x0964 ,0x1e7e,0x1b5d ,
-  0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
-  0x1a46,0x1e2b ,0x2e88,0x0f8d ,0x0471,0x2afb ,
-  0x11a8,0x238e ,0x2aaa,0x1294 ,0xf721,0x3179 ,
-  0x08df,0x289a ,0x26b3,0x1590 ,0xea02,0x36e5 ,
-  0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
-  0xf721,0x3179 ,0x1e7e,0x1b5d ,0xd178,0x3e15 ,
-  0xee58,0x3537 ,0x1a46,0x1e2b ,0xc695,0x3fb1 ,
-  0xe5ba,0x3871 ,0x15fe,0x20e7 ,0xbcf0,0x3fec ,
-  0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
-  0xd556,0x3d3f ,0x0d48,0x2620 ,0xae2e,0x3c42 ,
-  0xcdb7,0x3ec5 ,0x08df,0x289a ,0xa963,0x3871 ,
-  0xc695,0x3fb1 ,0x0471,0x2afb ,0xa678,0x3368 ,
-  0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
-  0xba09,0x3fb1 ,0xfb8f,0x2f6c ,0xa678,0x2620 ,
-  0xb4be,0x3ec5 ,0xf721,0x3179 ,0xa963,0x1e2b ,
-  0xb02d,0x3d3f ,0xf2b8,0x3368 ,0xae2e,0x1590 ,
-  0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
-  0xa963,0x3871 ,0xea02,0x36e5 ,0xbcf0,0x0324 ,
-  0xa73b,0x3537 ,0xe5ba,0x3871 ,0xc695,0xf9ba ,
-  0xa5ed,0x3179 ,0xe182,0x39db ,0xd178,0xf073 ,
-  0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
-  0xa5ed,0x289a ,0xd94d,0x3c42 ,0xea02,0xdf19 ,
-  0xa73b,0x238e ,0xd556,0x3d3f ,0xf721,0xd766 ,
-  0xa963,0x1e2b ,0xd178,0x3e15 ,0x0471,0xd094 ,
-  0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
-  0xb02d,0x1294 ,0xca15,0x3f4f ,0x1e7e,0xc625 ,
-  0xb4be,0x0c7c ,0xc695,0x3fb1 ,0x2aaa,0xc2c1 ,
-  0xba09,0x0646 ,0xc338,0x3fec ,0x35eb,0xc0b1 ,
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x3e69,0x0192 ,0x3f36,0x00c9 ,0x3d9a,0x025b ,
-  0x3cc8,0x0324 ,0x3e69,0x0192 ,0x3b1e,0x04b5 ,
-  0x3b1e,0x04b5 ,0x3d9a,0x025b ,0x388e,0x070e ,
-  0x396b,0x0646 ,0x3cc8,0x0324 ,0x35eb,0x0964 ,
-  0x37af,0x07d6 ,0x3bf4,0x03ed ,0x3334,0x0bb7 ,
-  0x35eb,0x0964 ,0x3b1e,0x04b5 ,0x306c,0x0e06 ,
-  0x341e,0x0af1 ,0x3a46,0x057e ,0x2d93,0x1050 ,
-  0x3249,0x0c7c ,0x396b,0x0646 ,0x2aaa,0x1294 ,
-  0x306c,0x0e06 ,0x388e,0x070e ,0x27b3,0x14d2 ,
-  0x2e88,0x0f8d ,0x37af,0x07d6 ,0x24ae,0x1709 ,
-  0x2c9d,0x1112 ,0x36ce,0x089d ,0x219c,0x1937 ,
-  0x2aaa,0x1294 ,0x35eb,0x0964 ,0x1e7e,0x1b5d ,
-  0x28b2,0x1413 ,0x3505,0x0a2b ,0x1b56,0x1d79 ,
-  0x26b3,0x1590 ,0x341e,0x0af1 ,0x1824,0x1f8c ,
-  0x24ae,0x1709 ,0x3334,0x0bb7 ,0x14ea,0x2193 ,
-  0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
-  0x2093,0x19ef ,0x315b,0x0d41 ,0x0e61,0x257e ,
-  0x1e7e,0x1b5d ,0x306c,0x0e06 ,0x0b14,0x2760 ,
-  0x1c64,0x1cc6 ,0x2f7b,0x0eca ,0x07c4,0x2935 ,
-  0x1a46,0x1e2b ,0x2e88,0x0f8d ,0x0471,0x2afb ,
-  0x1824,0x1f8c ,0x2d93,0x1050 ,0x011c,0x2cb2 ,
-  0x15fe,0x20e7 ,0x2c9d,0x1112 ,0xfdc7,0x2e5a ,
-  0x13d5,0x223d ,0x2ba4,0x11d3 ,0xfa73,0x2ff2 ,
-  0x11a8,0x238e ,0x2aaa,0x1294 ,0xf721,0x3179 ,
-  0x0f79,0x24da ,0x29af,0x1354 ,0xf3d2,0x32ef ,
-  0x0d48,0x2620 ,0x28b2,0x1413 ,0xf087,0x3453 ,
-  0x0b14,0x2760 ,0x27b3,0x14d2 ,0xed41,0x35a5 ,
-  0x08df,0x289a ,0x26b3,0x1590 ,0xea02,0x36e5 ,
-  0x06a9,0x29ce ,0x25b1,0x164c ,0xe6cb,0x3812 ,
-  0x0471,0x2afb ,0x24ae,0x1709 ,0xe39c,0x392b ,
-  0x0239,0x2c21 ,0x23a9,0x17c4 ,0xe077,0x3a30 ,
-  0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
-  0xfdc7,0x2e5a ,0x219c,0x1937 ,0xda4f,0x3bfd ,
-  0xfb8f,0x2f6c ,0x2093,0x19ef ,0xd74e,0x3cc5 ,
-  0xf957,0x3076 ,0x1f89,0x1aa7 ,0xd45c,0x3d78 ,
-  0xf721,0x3179 ,0x1e7e,0x1b5d ,0xd178,0x3e15 ,
-  0xf4ec,0x3274 ,0x1d72,0x1c12 ,0xcea5,0x3e9d ,
-  0xf2b8,0x3368 ,0x1c64,0x1cc6 ,0xcbe2,0x3f0f ,
-  0xf087,0x3453 ,0x1b56,0x1d79 ,0xc932,0x3f6b ,
-  0xee58,0x3537 ,0x1a46,0x1e2b ,0xc695,0x3fb1 ,
-  0xec2b,0x3612 ,0x1935,0x1edc ,0xc40c,0x3fe1 ,
-  0xea02,0x36e5 ,0x1824,0x1f8c ,0xc197,0x3ffb ,
-  0xe7dc,0x37b0 ,0x1711,0x203a ,0xbf38,0x3fff ,
-  0xe5ba,0x3871 ,0x15fe,0x20e7 ,0xbcf0,0x3fec ,
-  0xe39c,0x392b ,0x14ea,0x2193 ,0xbabf,0x3fc4 ,
-  0xe182,0x39db ,0x13d5,0x223d ,0xb8a6,0x3f85 ,
-  0xdf6d,0x3a82 ,0x12bf,0x22e7 ,0xb6a5,0x3f30 ,
-  0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
-  0xdb52,0x3bb6 ,0x1091,0x2435 ,0xb2f2,0x3e45 ,
-  0xd94d,0x3c42 ,0x0f79,0x24da ,0xb140,0x3daf ,
-  0xd74e,0x3cc5 ,0x0e61,0x257e ,0xafa9,0x3d03 ,
-  0xd556,0x3d3f ,0x0d48,0x2620 ,0xae2e,0x3c42 ,
-  0xd363,0x3daf ,0x0c2e,0x26c1 ,0xacd0,0x3b6d ,
-  0xd178,0x3e15 ,0x0b14,0x2760 ,0xab8e,0x3a82 ,
-  0xcf94,0x3e72 ,0x09fa,0x27fe ,0xaa6a,0x3984 ,
-  0xcdb7,0x3ec5 ,0x08df,0x289a ,0xa963,0x3871 ,
-  0xcbe2,0x3f0f ,0x07c4,0x2935 ,0xa87b,0x374b ,
-  0xca15,0x3f4f ,0x06a9,0x29ce ,0xa7b1,0x3612 ,
-  0xc851,0x3f85 ,0x058d,0x2a65 ,0xa705,0x34c6 ,
-  0xc695,0x3fb1 ,0x0471,0x2afb ,0xa678,0x3368 ,
-  0xc4e2,0x3fd4 ,0x0355,0x2b8f ,0xa60b,0x31f8 ,
-  0xc338,0x3fec ,0x0239,0x2c21 ,0xa5bc,0x3076 ,
-  0xc197,0x3ffb ,0x011c,0x2cb2 ,0xa58d,0x2ee4 ,
-  0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
-  0xbe73,0x3ffb ,0xfee4,0x2dcf ,0xa58d,0x2b8f ,
-  0xbcf0,0x3fec ,0xfdc7,0x2e5a ,0xa5bc,0x29ce ,
-  0xbb77,0x3fd4 ,0xfcab,0x2ee4 ,0xa60b,0x27fe ,
-  0xba09,0x3fb1 ,0xfb8f,0x2f6c ,0xa678,0x2620 ,
-  0xb8a6,0x3f85 ,0xfa73,0x2ff2 ,0xa705,0x2435 ,
-  0xb74d,0x3f4f ,0xf957,0x3076 ,0xa7b1,0x223d ,
-  0xb600,0x3f0f ,0xf83c,0x30f9 ,0xa87b,0x203a ,
-  0xb4be,0x3ec5 ,0xf721,0x3179 ,0xa963,0x1e2b ,
-  0xb388,0x3e72 ,0xf606,0x31f8 ,0xaa6a,0x1c12 ,
-  0xb25e,0x3e15 ,0xf4ec,0x3274 ,0xab8e,0x19ef ,
-  0xb140,0x3daf ,0xf3d2,0x32ef ,0xacd0,0x17c4 ,
-  0xb02d,0x3d3f ,0xf2b8,0x3368 ,0xae2e,0x1590 ,
-  0xaf28,0x3cc5 ,0xf19f,0x33df ,0xafa9,0x1354 ,
-  0xae2e,0x3c42 ,0xf087,0x3453 ,0xb140,0x1112 ,
-  0xad41,0x3bb6 ,0xef6f,0x34c6 ,0xb2f2,0x0eca ,
-  0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
-  0xab8e,0x3a82 ,0xed41,0x35a5 ,0xb6a5,0x0a2b ,
-  0xaac8,0x39db ,0xec2b,0x3612 ,0xb8a6,0x07d6 ,
-  0xaa0f,0x392b ,0xeb16,0x367d ,0xbabf,0x057e ,
-  0xa963,0x3871 ,0xea02,0x36e5 ,0xbcf0,0x0324 ,
-  0xa8c5,0x37b0 ,0xe8ef,0x374b ,0xbf38,0x00c9 ,
-  0xa834,0x36e5 ,0xe7dc,0x37b0 ,0xc197,0xfe6e ,
-  0xa7b1,0x3612 ,0xe6cb,0x3812 ,0xc40c,0xfc13 ,
-  0xa73b,0x3537 ,0xe5ba,0x3871 ,0xc695,0xf9ba ,
-  0xa6d3,0x3453 ,0xe4aa,0x38cf ,0xc932,0xf763 ,
-  0xa678,0x3368 ,0xe39c,0x392b ,0xcbe2,0xf50f ,
-  0xa62c,0x3274 ,0xe28e,0x3984 ,0xcea5,0xf2bf ,
-  0xa5ed,0x3179 ,0xe182,0x39db ,0xd178,0xf073 ,
-  0xa5bc,0x3076 ,0xe077,0x3a30 ,0xd45c,0xee2d ,
-  0xa599,0x2f6c ,0xdf6d,0x3a82 ,0xd74e,0xebed ,
-  0xa585,0x2e5a ,0xde64,0x3ad3 ,0xda4f,0xe9b4 ,
-  0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
-  0xa585,0x2c21 ,0xdc57,0x3b6d ,0xe077,0xe559 ,
-  0xa599,0x2afb ,0xdb52,0x3bb6 ,0xe39c,0xe33a ,
-  0xa5bc,0x29ce ,0xda4f,0x3bfd ,0xe6cb,0xe124 ,
-  0xa5ed,0x289a ,0xd94d,0x3c42 ,0xea02,0xdf19 ,
-  0xa62c,0x2760 ,0xd84d,0x3c85 ,0xed41,0xdd19 ,
-  0xa678,0x2620 ,0xd74e,0x3cc5 ,0xf087,0xdb26 ,
-  0xa6d3,0x24da ,0xd651,0x3d03 ,0xf3d2,0xd93f ,
-  0xa73b,0x238e ,0xd556,0x3d3f ,0xf721,0xd766 ,
-  0xa7b1,0x223d ,0xd45c,0x3d78 ,0xfa73,0xd59b ,
-  0xa834,0x20e7 ,0xd363,0x3daf ,0xfdc7,0xd3df ,
-  0xa8c5,0x1f8c ,0xd26d,0x3de3 ,0x011c,0xd231 ,
-  0xa963,0x1e2b ,0xd178,0x3e15 ,0x0471,0xd094 ,
-  0xaa0f,0x1cc6 ,0xd085,0x3e45 ,0x07c4,0xcf07 ,
-  0xaac8,0x1b5d ,0xcf94,0x3e72 ,0x0b14,0xcd8c ,
-  0xab8e,0x19ef ,0xcea5,0x3e9d ,0x0e61,0xcc21 ,
-  0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
-  0xad41,0x1709 ,0xcccc,0x3eeb ,0x14ea,0xc983 ,
-  0xae2e,0x1590 ,0xcbe2,0x3f0f ,0x1824,0xc850 ,
-  0xaf28,0x1413 ,0xcafb,0x3f30 ,0x1b56,0xc731 ,
-  0xb02d,0x1294 ,0xca15,0x3f4f ,0x1e7e,0xc625 ,
-  0xb140,0x1112 ,0xc932,0x3f6b ,0x219c,0xc52d ,
-  0xb25e,0x0f8d ,0xc851,0x3f85 ,0x24ae,0xc44a ,
-  0xb388,0x0e06 ,0xc772,0x3f9c ,0x27b3,0xc37b ,
-  0xb4be,0x0c7c ,0xc695,0x3fb1 ,0x2aaa,0xc2c1 ,
-  0xb600,0x0af1 ,0xc5ba,0x3fc4 ,0x2d93,0xc21d ,
-  0xb74d,0x0964 ,0xc4e2,0x3fd4 ,0x306c,0xc18e ,
-  0xb8a6,0x07d6 ,0xc40c,0x3fe1 ,0x3334,0xc115 ,
-  0xba09,0x0646 ,0xc338,0x3fec ,0x35eb,0xc0b1 ,
-  0xbb77,0x04b5 ,0xc266,0x3ff5 ,0x388e,0xc064 ,
-  0xbcf0,0x0324 ,0xc197,0x3ffb ,0x3b1e,0xc02c ,
-  0xbe73,0x0192 ,0xc0ca,0x3fff ,0x3d9a,0xc00b ,
-  0x4000,0x0000 ,0x3f9b,0x0065 ,0x3f36,0x00c9 ,
-  0x3ed0,0x012e ,0x3e69,0x0192 ,0x3e02,0x01f7 ,
-  0x3d9a,0x025b ,0x3d31,0x02c0 ,0x3cc8,0x0324 ,
-  0x3c5f,0x0388 ,0x3bf4,0x03ed ,0x3b8a,0x0451 ,
-  0x3b1e,0x04b5 ,0x3ab2,0x051a ,0x3a46,0x057e ,
-  0x39d9,0x05e2 ,0x396b,0x0646 ,0x38fd,0x06aa ,
-  0x388e,0x070e ,0x381f,0x0772 ,0x37af,0x07d6 ,
-  0x373f,0x0839 ,0x36ce,0x089d ,0x365d,0x0901 ,
-  0x35eb,0x0964 ,0x3578,0x09c7 ,0x3505,0x0a2b ,
-  0x3492,0x0a8e ,0x341e,0x0af1 ,0x33a9,0x0b54 ,
-  0x3334,0x0bb7 ,0x32bf,0x0c1a ,0x3249,0x0c7c ,
-  0x31d2,0x0cdf ,0x315b,0x0d41 ,0x30e4,0x0da4 ,
-  0x306c,0x0e06 ,0x2ff4,0x0e68 ,0x2f7b,0x0eca ,
-  0x2f02,0x0f2b ,0x2e88,0x0f8d ,0x2e0e,0x0fee ,
-  0x2d93,0x1050 ,0x2d18,0x10b1 ,0x2c9d,0x1112 ,
-  0x2c21,0x1173 ,0x2ba4,0x11d3 ,0x2b28,0x1234 ,
-  0x2aaa,0x1294 ,0x2a2d,0x12f4 ,0x29af,0x1354 ,
-  0x2931,0x13b4 ,0x28b2,0x1413 ,0x2833,0x1473 ,
-  0x27b3,0x14d2 ,0x2733,0x1531 ,0x26b3,0x1590 ,
-  0x2632,0x15ee ,0x25b1,0x164c ,0x252f,0x16ab ,
-  0x24ae,0x1709 ,0x242b,0x1766 ,0x23a9,0x17c4 ,
-  0x2326,0x1821 ,0x22a3,0x187e ,0x221f,0x18db ,
-  0x219c,0x1937 ,0x2117,0x1993 ,0x2093,0x19ef ,
-  0x200e,0x1a4b ,0x1f89,0x1aa7 ,0x1f04,0x1b02 ,
-  0x1e7e,0x1b5d ,0x1df8,0x1bb8 ,0x1d72,0x1c12 ,
-  0x1ceb,0x1c6c ,0x1c64,0x1cc6 ,0x1bdd,0x1d20 ,
-  0x1b56,0x1d79 ,0x1ace,0x1dd3 ,0x1a46,0x1e2b ,
-  0x19be,0x1e84 ,0x1935,0x1edc ,0x18ad,0x1f34 ,
-  0x1824,0x1f8c ,0x179b,0x1fe3 ,0x1711,0x203a ,
-  0x1688,0x2091 ,0x15fe,0x20e7 ,0x1574,0x213d ,
-  0x14ea,0x2193 ,0x145f,0x21e8 ,0x13d5,0x223d ,
-  0x134a,0x2292 ,0x12bf,0x22e7 ,0x1234,0x233b ,
-  0x11a8,0x238e ,0x111d,0x23e2 ,0x1091,0x2435 ,
-  0x1005,0x2488 ,0x0f79,0x24da ,0x0eed,0x252c ,
-  0x0e61,0x257e ,0x0dd4,0x25cf ,0x0d48,0x2620 ,
-  0x0cbb,0x2671 ,0x0c2e,0x26c1 ,0x0ba1,0x2711 ,
-  0x0b14,0x2760 ,0x0a87,0x27af ,0x09fa,0x27fe ,
-  0x096d,0x284c ,0x08df,0x289a ,0x0852,0x28e7 ,
-  0x07c4,0x2935 ,0x0736,0x2981 ,0x06a9,0x29ce ,
-  0x061b,0x2a1a ,0x058d,0x2a65 ,0x04ff,0x2ab0 ,
-  0x0471,0x2afb ,0x03e3,0x2b45 ,0x0355,0x2b8f ,
-  0x02c7,0x2bd8 ,0x0239,0x2c21 ,0x01aa,0x2c6a ,
-  0x011c,0x2cb2 ,0x008e,0x2cfa ,0x0000,0x2d41 ,
-  0xff72,0x2d88 ,0xfee4,0x2dcf ,0xfe56,0x2e15 ,
-  0xfdc7,0x2e5a ,0xfd39,0x2e9f ,0xfcab,0x2ee4 ,
-  0xfc1d,0x2f28 ,0xfb8f,0x2f6c ,0xfb01,0x2faf ,
-  0xfa73,0x2ff2 ,0xf9e5,0x3034 ,0xf957,0x3076 ,
-  0xf8ca,0x30b8 ,0xf83c,0x30f9 ,0xf7ae,0x3139 ,
-  0xf721,0x3179 ,0xf693,0x31b9 ,0xf606,0x31f8 ,
-  0xf579,0x3236 ,0xf4ec,0x3274 ,0xf45f,0x32b2 ,
-  0xf3d2,0x32ef ,0xf345,0x332c ,0xf2b8,0x3368 ,
-  0xf22c,0x33a3 ,0xf19f,0x33df ,0xf113,0x3419 ,
-  0xf087,0x3453 ,0xeffb,0x348d ,0xef6f,0x34c6 ,
-  0xeee3,0x34ff ,0xee58,0x3537 ,0xedcc,0x356e ,
-  0xed41,0x35a5 ,0xecb6,0x35dc ,0xec2b,0x3612 ,
-  0xeba1,0x3648 ,0xeb16,0x367d ,0xea8c,0x36b1 ,
-  0xea02,0x36e5 ,0xe978,0x3718 ,0xe8ef,0x374b ,
-  0xe865,0x377e ,0xe7dc,0x37b0 ,0xe753,0x37e1 ,
-  0xe6cb,0x3812 ,0xe642,0x3842 ,0xe5ba,0x3871 ,
-  0xe532,0x38a1 ,0xe4aa,0x38cf ,0xe423,0x38fd ,
-  0xe39c,0x392b ,0xe315,0x3958 ,0xe28e,0x3984 ,
-  0xe208,0x39b0 ,0xe182,0x39db ,0xe0fc,0x3a06 ,
-  0xe077,0x3a30 ,0xdff2,0x3a59 ,0xdf6d,0x3a82 ,
-  0xdee9,0x3aab ,0xde64,0x3ad3 ,0xdde1,0x3afa ,
-  0xdd5d,0x3b21 ,0xdcda,0x3b47 ,0xdc57,0x3b6d ,
-  0xdbd5,0x3b92 ,0xdb52,0x3bb6 ,0xdad1,0x3bda ,
-  0xda4f,0x3bfd ,0xd9ce,0x3c20 ,0xd94d,0x3c42 ,
-  0xd8cd,0x3c64 ,0xd84d,0x3c85 ,0xd7cd,0x3ca5 ,
-  0xd74e,0x3cc5 ,0xd6cf,0x3ce4 ,0xd651,0x3d03 ,
-  0xd5d3,0x3d21 ,0xd556,0x3d3f ,0xd4d8,0x3d5b ,
-  0xd45c,0x3d78 ,0xd3df,0x3d93 ,0xd363,0x3daf ,
-  0xd2e8,0x3dc9 ,0xd26d,0x3de3 ,0xd1f2,0x3dfc ,
-  0xd178,0x3e15 ,0xd0fe,0x3e2d ,0xd085,0x3e45 ,
-  0xd00c,0x3e5c ,0xcf94,0x3e72 ,0xcf1c,0x3e88 ,
-  0xcea5,0x3e9d ,0xce2e,0x3eb1 ,0xcdb7,0x3ec5 ,
-  0xcd41,0x3ed8 ,0xcccc,0x3eeb ,0xcc57,0x3efd ,
-  0xcbe2,0x3f0f ,0xcb6e,0x3f20 ,0xcafb,0x3f30 ,
-  0xca88,0x3f40 ,0xca15,0x3f4f ,0xc9a3,0x3f5d ,
-  0xc932,0x3f6b ,0xc8c1,0x3f78 ,0xc851,0x3f85 ,
-  0xc7e1,0x3f91 ,0xc772,0x3f9c ,0xc703,0x3fa7 ,
-  0xc695,0x3fb1 ,0xc627,0x3fbb ,0xc5ba,0x3fc4 ,
-  0xc54e,0x3fcc ,0xc4e2,0x3fd4 ,0xc476,0x3fdb ,
-  0xc40c,0x3fe1 ,0xc3a1,0x3fe7 ,0xc338,0x3fec ,
-  0xc2cf,0x3ff1 ,0xc266,0x3ff5 ,0xc1fe,0x3ff8 ,
-  0xc197,0x3ffb ,0xc130,0x3ffd ,0xc0ca,0x3fff ,
-  0xc065,0x4000 ,0xc000,0x4000 ,0xbf9c,0x4000 ,
-  0xbf38,0x3fff ,0xbed5,0x3ffd ,0xbe73,0x3ffb ,
-  0xbe11,0x3ff8 ,0xbdb0,0x3ff5 ,0xbd50,0x3ff1 ,
-  0xbcf0,0x3fec ,0xbc91,0x3fe7 ,0xbc32,0x3fe1 ,
-  0xbbd4,0x3fdb ,0xbb77,0x3fd4 ,0xbb1b,0x3fcc ,
-  0xbabf,0x3fc4 ,0xba64,0x3fbb ,0xba09,0x3fb1 ,
-  0xb9af,0x3fa7 ,0xb956,0x3f9c ,0xb8fd,0x3f91 ,
-  0xb8a6,0x3f85 ,0xb84f,0x3f78 ,0xb7f8,0x3f6b ,
-  0xb7a2,0x3f5d ,0xb74d,0x3f4f ,0xb6f9,0x3f40 ,
-  0xb6a5,0x3f30 ,0xb652,0x3f20 ,0xb600,0x3f0f ,
-  0xb5af,0x3efd ,0xb55e,0x3eeb ,0xb50e,0x3ed8 ,
-  0xb4be,0x3ec5 ,0xb470,0x3eb1 ,0xb422,0x3e9d ,
-  0xb3d5,0x3e88 ,0xb388,0x3e72 ,0xb33d,0x3e5c ,
-  0xb2f2,0x3e45 ,0xb2a7,0x3e2d ,0xb25e,0x3e15 ,
-  0xb215,0x3dfc ,0xb1cd,0x3de3 ,0xb186,0x3dc9 ,
-  0xb140,0x3daf ,0xb0fa,0x3d93 ,0xb0b5,0x3d78 ,
-  0xb071,0x3d5b ,0xb02d,0x3d3f ,0xafeb,0x3d21 ,
-  0xafa9,0x3d03 ,0xaf68,0x3ce4 ,0xaf28,0x3cc5 ,
-  0xaee8,0x3ca5 ,0xaea9,0x3c85 ,0xae6b,0x3c64 ,
-  0xae2e,0x3c42 ,0xadf2,0x3c20 ,0xadb6,0x3bfd ,
-  0xad7b,0x3bda ,0xad41,0x3bb6 ,0xad08,0x3b92 ,
-  0xacd0,0x3b6d ,0xac98,0x3b47 ,0xac61,0x3b21 ,
-  0xac2b,0x3afa ,0xabf6,0x3ad3 ,0xabc2,0x3aab ,
-  0xab8e,0x3a82 ,0xab5b,0x3a59 ,0xab29,0x3a30 ,
-  0xaaf8,0x3a06 ,0xaac8,0x39db ,0xaa98,0x39b0 ,
-  0xaa6a,0x3984 ,0xaa3c,0x3958 ,0xaa0f,0x392b ,
-  0xa9e3,0x38fd ,0xa9b7,0x38cf ,0xa98d,0x38a1 ,
-  0xa963,0x3871 ,0xa93a,0x3842 ,0xa912,0x3812 ,
-  0xa8eb,0x37e1 ,0xa8c5,0x37b0 ,0xa89f,0x377e ,
-  0xa87b,0x374b ,0xa857,0x3718 ,0xa834,0x36e5 ,
-  0xa812,0x36b1 ,0xa7f1,0x367d ,0xa7d0,0x3648 ,
-  0xa7b1,0x3612 ,0xa792,0x35dc ,0xa774,0x35a5 ,
-  0xa757,0x356e ,0xa73b,0x3537 ,0xa71f,0x34ff ,
-  0xa705,0x34c6 ,0xa6eb,0x348d ,0xa6d3,0x3453 ,
-  0xa6bb,0x3419 ,0xa6a4,0x33df ,0xa68e,0x33a3 ,
-  0xa678,0x3368 ,0xa664,0x332c ,0xa650,0x32ef ,
-  0xa63e,0x32b2 ,0xa62c,0x3274 ,0xa61b,0x3236 ,
-  0xa60b,0x31f8 ,0xa5fb,0x31b9 ,0xa5ed,0x3179 ,
-  0xa5e0,0x3139 ,0xa5d3,0x30f9 ,0xa5c7,0x30b8 ,
-  0xa5bc,0x3076 ,0xa5b2,0x3034 ,0xa5a9,0x2ff2 ,
-  0xa5a1,0x2faf ,0xa599,0x2f6c ,0xa593,0x2f28 ,
-  0xa58d,0x2ee4 ,0xa588,0x2e9f ,0xa585,0x2e5a ,
-  0xa581,0x2e15 ,0xa57f,0x2dcf ,0xa57e,0x2d88 ,
-  0xa57e,0x2d41 ,0xa57e,0x2cfa ,0xa57f,0x2cb2 ,
-  0xa581,0x2c6a ,0xa585,0x2c21 ,0xa588,0x2bd8 ,
-  0xa58d,0x2b8f ,0xa593,0x2b45 ,0xa599,0x2afb ,
-  0xa5a1,0x2ab0 ,0xa5a9,0x2a65 ,0xa5b2,0x2a1a ,
-  0xa5bc,0x29ce ,0xa5c7,0x2981 ,0xa5d3,0x2935 ,
-  0xa5e0,0x28e7 ,0xa5ed,0x289a ,0xa5fb,0x284c ,
-  0xa60b,0x27fe ,0xa61b,0x27af ,0xa62c,0x2760 ,
-  0xa63e,0x2711 ,0xa650,0x26c1 ,0xa664,0x2671 ,
-  0xa678,0x2620 ,0xa68e,0x25cf ,0xa6a4,0x257e ,
-  0xa6bb,0x252c ,0xa6d3,0x24da ,0xa6eb,0x2488 ,
-  0xa705,0x2435 ,0xa71f,0x23e2 ,0xa73b,0x238e ,
-  0xa757,0x233b ,0xa774,0x22e7 ,0xa792,0x2292 ,
-  0xa7b1,0x223d ,0xa7d0,0x21e8 ,0xa7f1,0x2193 ,
-  0xa812,0x213d ,0xa834,0x20e7 ,0xa857,0x2091 ,
-  0xa87b,0x203a ,0xa89f,0x1fe3 ,0xa8c5,0x1f8c ,
-  0xa8eb,0x1f34 ,0xa912,0x1edc ,0xa93a,0x1e84 ,
-  0xa963,0x1e2b ,0xa98d,0x1dd3 ,0xa9b7,0x1d79 ,
-  0xa9e3,0x1d20 ,0xaa0f,0x1cc6 ,0xaa3c,0x1c6c ,
-  0xaa6a,0x1c12 ,0xaa98,0x1bb8 ,0xaac8,0x1b5d ,
-  0xaaf8,0x1b02 ,0xab29,0x1aa7 ,0xab5b,0x1a4b ,
-  0xab8e,0x19ef ,0xabc2,0x1993 ,0xabf6,0x1937 ,
-  0xac2b,0x18db ,0xac61,0x187e ,0xac98,0x1821 ,
-  0xacd0,0x17c4 ,0xad08,0x1766 ,0xad41,0x1709 ,
-  0xad7b,0x16ab ,0xadb6,0x164c ,0xadf2,0x15ee ,
-  0xae2e,0x1590 ,0xae6b,0x1531 ,0xaea9,0x14d2 ,
-  0xaee8,0x1473 ,0xaf28,0x1413 ,0xaf68,0x13b4 ,
-  0xafa9,0x1354 ,0xafeb,0x12f4 ,0xb02d,0x1294 ,
-  0xb071,0x1234 ,0xb0b5,0x11d3 ,0xb0fa,0x1173 ,
-  0xb140,0x1112 ,0xb186,0x10b1 ,0xb1cd,0x1050 ,
-  0xb215,0x0fee ,0xb25e,0x0f8d ,0xb2a7,0x0f2b ,
-  0xb2f2,0x0eca ,0xb33d,0x0e68 ,0xb388,0x0e06 ,
-  0xb3d5,0x0da4 ,0xb422,0x0d41 ,0xb470,0x0cdf ,
-  0xb4be,0x0c7c ,0xb50e,0x0c1a ,0xb55e,0x0bb7 ,
-  0xb5af,0x0b54 ,0xb600,0x0af1 ,0xb652,0x0a8e ,
-  0xb6a5,0x0a2b ,0xb6f9,0x09c7 ,0xb74d,0x0964 ,
-  0xb7a2,0x0901 ,0xb7f8,0x089d ,0xb84f,0x0839 ,
-  0xb8a6,0x07d6 ,0xb8fd,0x0772 ,0xb956,0x070e ,
-  0xb9af,0x06aa ,0xba09,0x0646 ,0xba64,0x05e2 ,
-  0xbabf,0x057e ,0xbb1b,0x051a ,0xbb77,0x04b5 ,
-  0xbbd4,0x0451 ,0xbc32,0x03ed ,0xbc91,0x0388 ,
-  0xbcf0,0x0324 ,0xbd50,0x02c0 ,0xbdb0,0x025b ,
-  0xbe11,0x01f7 ,0xbe73,0x0192 ,0xbed5,0x012e ,
-  0xbf38,0x00c9 ,0xbf9c,0x0065 };
-
-
-extern const int s_Q14R_8;
-const int s_Q14R_8 = 1024;
-extern const unsigned short t_Q14R_8[2032];
-const unsigned short t_Q14R_8[2032] = {
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
-  0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
-  0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
-  0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
-  0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
-  0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
-  0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x3fb1,0x0646 ,0x3fec,0x0324 ,0x3f4f,0x0964 ,
-  0x3ec5,0x0c7c ,0x3fb1,0x0646 ,0x3d3f,0x1294 ,
-  0x3d3f,0x1294 ,0x3f4f,0x0964 ,0x39db,0x1b5d ,
-  0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
-  0x3871,0x1e2b ,0x3e15,0x0f8d ,0x2f6c,0x2afb ,
-  0x3537,0x238e ,0x3d3f,0x1294 ,0x289a,0x3179 ,
-  0x3179,0x289a ,0x3c42,0x1590 ,0x20e7,0x36e5 ,
-  0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
-  0x289a,0x3179 ,0x39db,0x1b5d ,0x0f8d,0x3e15 ,
-  0x238e,0x3537 ,0x3871,0x1e2b ,0x0646,0x3fb1 ,
-  0x1e2b,0x3871 ,0x36e5,0x20e7 ,0xfcdc,0x3fec ,
-  0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
-  0x1294,0x3d3f ,0x3368,0x2620 ,0xea70,0x3c42 ,
-  0x0c7c,0x3ec5 ,0x3179,0x289a ,0xe1d5,0x3871 ,
-  0x0646,0x3fb1 ,0x2f6c,0x2afb ,0xd9e0,0x3368 ,
-  0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
-  0xf9ba,0x3fb1 ,0x2afb,0x2f6c ,0xcc98,0x2620 ,
-  0xf384,0x3ec5 ,0x289a,0x3179 ,0xc78f,0x1e2b ,
-  0xed6c,0x3d3f ,0x2620,0x3368 ,0xc3be,0x1590 ,
-  0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
-  0xe1d5,0x3871 ,0x20e7,0x36e5 ,0xc014,0x0324 ,
-  0xdc72,0x3537 ,0x1e2b,0x3871 ,0xc04f,0xf9ba ,
-  0xd766,0x3179 ,0x1b5d,0x39db ,0xc1eb,0xf073 ,
-  0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
-  0xce87,0x289a ,0x1590,0x3c42 ,0xc91b,0xdf19 ,
-  0xcac9,0x238e ,0x1294,0x3d3f ,0xce87,0xd766 ,
-  0xc78f,0x1e2b ,0x0f8d,0x3e15 ,0xd505,0xd094 ,
-  0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
-  0xc2c1,0x1294 ,0x0964,0x3f4f ,0xe4a3,0xc625 ,
-  0xc13b,0x0c7c ,0x0646,0x3fb1 ,0xed6c,0xc2c1 ,
-  0xc04f,0x0646 ,0x0324,0x3fec ,0xf69c,0xc0b1 ,
-  0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
-  0x3ffb,0x0192 ,0x3fff,0x00c9 ,0x3ff5,0x025b ,
-  0x3fec,0x0324 ,0x3ffb,0x0192 ,0x3fd4,0x04b5 ,
-  0x3fd4,0x04b5 ,0x3ff5,0x025b ,0x3f9c,0x070e ,
-  0x3fb1,0x0646 ,0x3fec,0x0324 ,0x3f4f,0x0964 ,
-  0x3f85,0x07d6 ,0x3fe1,0x03ed ,0x3eeb,0x0bb7 ,
-  0x3f4f,0x0964 ,0x3fd4,0x04b5 ,0x3e72,0x0e06 ,
-  0x3f0f,0x0af1 ,0x3fc4,0x057e ,0x3de3,0x1050 ,
-  0x3ec5,0x0c7c ,0x3fb1,0x0646 ,0x3d3f,0x1294 ,
-  0x3e72,0x0e06 ,0x3f9c,0x070e ,0x3c85,0x14d2 ,
-  0x3e15,0x0f8d ,0x3f85,0x07d6 ,0x3bb6,0x1709 ,
-  0x3daf,0x1112 ,0x3f6b,0x089d ,0x3ad3,0x1937 ,
-  0x3d3f,0x1294 ,0x3f4f,0x0964 ,0x39db,0x1b5d ,
-  0x3cc5,0x1413 ,0x3f30,0x0a2b ,0x38cf,0x1d79 ,
-  0x3c42,0x1590 ,0x3f0f,0x0af1 ,0x37b0,0x1f8c ,
-  0x3bb6,0x1709 ,0x3eeb,0x0bb7 ,0x367d,0x2193 ,
-  0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
-  0x3a82,0x19ef ,0x3e9d,0x0d41 ,0x33df,0x257e ,
-  0x39db,0x1b5d ,0x3e72,0x0e06 ,0x3274,0x2760 ,
-  0x392b,0x1cc6 ,0x3e45,0x0eca ,0x30f9,0x2935 ,
-  0x3871,0x1e2b ,0x3e15,0x0f8d ,0x2f6c,0x2afb ,
-  0x37b0,0x1f8c ,0x3de3,0x1050 ,0x2dcf,0x2cb2 ,
-  0x36e5,0x20e7 ,0x3daf,0x1112 ,0x2c21,0x2e5a ,
-  0x3612,0x223d ,0x3d78,0x11d3 ,0x2a65,0x2ff2 ,
-  0x3537,0x238e ,0x3d3f,0x1294 ,0x289a,0x3179 ,
-  0x3453,0x24da ,0x3d03,0x1354 ,0x26c1,0x32ef ,
-  0x3368,0x2620 ,0x3cc5,0x1413 ,0x24da,0x3453 ,
-  0x3274,0x2760 ,0x3c85,0x14d2 ,0x22e7,0x35a5 ,
-  0x3179,0x289a ,0x3c42,0x1590 ,0x20e7,0x36e5 ,
-  0x3076,0x29ce ,0x3bfd,0x164c ,0x1edc,0x3812 ,
-  0x2f6c,0x2afb ,0x3bb6,0x1709 ,0x1cc6,0x392b ,
-  0x2e5a,0x2c21 ,0x3b6d,0x17c4 ,0x1aa7,0x3a30 ,
-  0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
-  0x2c21,0x2e5a ,0x3ad3,0x1937 ,0x164c,0x3bfd ,
-  0x2afb,0x2f6c ,0x3a82,0x19ef ,0x1413,0x3cc5 ,
-  0x29ce,0x3076 ,0x3a30,0x1aa7 ,0x11d3,0x3d78 ,
-  0x289a,0x3179 ,0x39db,0x1b5d ,0x0f8d,0x3e15 ,
-  0x2760,0x3274 ,0x3984,0x1c12 ,0x0d41,0x3e9d ,
-  0x2620,0x3368 ,0x392b,0x1cc6 ,0x0af1,0x3f0f ,
-  0x24da,0x3453 ,0x38cf,0x1d79 ,0x089d,0x3f6b ,
-  0x238e,0x3537 ,0x3871,0x1e2b ,0x0646,0x3fb1 ,
-  0x223d,0x3612 ,0x3812,0x1edc ,0x03ed,0x3fe1 ,
-  0x20e7,0x36e5 ,0x37b0,0x1f8c ,0x0192,0x3ffb ,
-  0x1f8c,0x37b0 ,0x374b,0x203a ,0xff37,0x3fff ,
-  0x1e2b,0x3871 ,0x36e5,0x20e7 ,0xfcdc,0x3fec ,
-  0x1cc6,0x392b ,0x367d,0x2193 ,0xfa82,0x3fc4 ,
-  0x1b5d,0x39db ,0x3612,0x223d ,0xf82a,0x3f85 ,
-  0x19ef,0x3a82 ,0x35a5,0x22e7 ,0xf5d5,0x3f30 ,
-  0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
-  0x1709,0x3bb6 ,0x34c6,0x2435 ,0xf136,0x3e45 ,
-  0x1590,0x3c42 ,0x3453,0x24da ,0xeeee,0x3daf ,
-  0x1413,0x3cc5 ,0x33df,0x257e ,0xecac,0x3d03 ,
-  0x1294,0x3d3f ,0x3368,0x2620 ,0xea70,0x3c42 ,
-  0x1112,0x3daf ,0x32ef,0x26c1 ,0xe83c,0x3b6d ,
-  0x0f8d,0x3e15 ,0x3274,0x2760 ,0xe611,0x3a82 ,
-  0x0e06,0x3e72 ,0x31f8,0x27fe ,0xe3ee,0x3984 ,
-  0x0c7c,0x3ec5 ,0x3179,0x289a ,0xe1d5,0x3871 ,
-  0x0af1,0x3f0f ,0x30f9,0x2935 ,0xdfc6,0x374b ,
-  0x0964,0x3f4f ,0x3076,0x29ce ,0xddc3,0x3612 ,
-  0x07d6,0x3f85 ,0x2ff2,0x2a65 ,0xdbcb,0x34c6 ,
-  0x0646,0x3fb1 ,0x2f6c,0x2afb ,0xd9e0,0x3368 ,
-  0x04b5,0x3fd4 ,0x2ee4,0x2b8f ,0xd802,0x31f8 ,
-  0x0324,0x3fec ,0x2e5a,0x2c21 ,0xd632,0x3076 ,
-  0x0192,0x3ffb ,0x2dcf,0x2cb2 ,0xd471,0x2ee4 ,
-  0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
-  0xfe6e,0x3ffb ,0x2cb2,0x2dcf ,0xd11c,0x2b8f ,
-  0xfcdc,0x3fec ,0x2c21,0x2e5a ,0xcf8a,0x29ce ,
-  0xfb4b,0x3fd4 ,0x2b8f,0x2ee4 ,0xce08,0x27fe ,
-  0xf9ba,0x3fb1 ,0x2afb,0x2f6c ,0xcc98,0x2620 ,
-  0xf82a,0x3f85 ,0x2a65,0x2ff2 ,0xcb3a,0x2435 ,
-  0xf69c,0x3f4f ,0x29ce,0x3076 ,0xc9ee,0x223d ,
-  0xf50f,0x3f0f ,0x2935,0x30f9 ,0xc8b5,0x203a ,
-  0xf384,0x3ec5 ,0x289a,0x3179 ,0xc78f,0x1e2b ,
-  0xf1fa,0x3e72 ,0x27fe,0x31f8 ,0xc67c,0x1c12 ,
-  0xf073,0x3e15 ,0x2760,0x3274 ,0xc57e,0x19ef ,
-  0xeeee,0x3daf ,0x26c1,0x32ef ,0xc493,0x17c4 ,
-  0xed6c,0x3d3f ,0x2620,0x3368 ,0xc3be,0x1590 ,
-  0xebed,0x3cc5 ,0x257e,0x33df ,0xc2fd,0x1354 ,
-  0xea70,0x3c42 ,0x24da,0x3453 ,0xc251,0x1112 ,
-  0xe8f7,0x3bb6 ,0x2435,0x34c6 ,0xc1bb,0x0eca ,
-  0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
-  0xe611,0x3a82 ,0x22e7,0x35a5 ,0xc0d0,0x0a2b ,
-  0xe4a3,0x39db ,0x223d,0x3612 ,0xc07b,0x07d6 ,
-  0xe33a,0x392b ,0x2193,0x367d ,0xc03c,0x057e ,
-  0xe1d5,0x3871 ,0x20e7,0x36e5 ,0xc014,0x0324 ,
-  0xe074,0x37b0 ,0x203a,0x374b ,0xc001,0x00c9 ,
-  0xdf19,0x36e5 ,0x1f8c,0x37b0 ,0xc005,0xfe6e ,
-  0xddc3,0x3612 ,0x1edc,0x3812 ,0xc01f,0xfc13 ,
-  0xdc72,0x3537 ,0x1e2b,0x3871 ,0xc04f,0xf9ba ,
-  0xdb26,0x3453 ,0x1d79,0x38cf ,0xc095,0xf763 ,
-  0xd9e0,0x3368 ,0x1cc6,0x392b ,0xc0f1,0xf50f ,
-  0xd8a0,0x3274 ,0x1c12,0x3984 ,0xc163,0xf2bf ,
-  0xd766,0x3179 ,0x1b5d,0x39db ,0xc1eb,0xf073 ,
-  0xd632,0x3076 ,0x1aa7,0x3a30 ,0xc288,0xee2d ,
-  0xd505,0x2f6c ,0x19ef,0x3a82 ,0xc33b,0xebed ,
-  0xd3df,0x2e5a ,0x1937,0x3ad3 ,0xc403,0xe9b4 ,
-  0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
-  0xd1a6,0x2c21 ,0x17c4,0x3b6d ,0xc5d0,0xe559 ,
-  0xd094,0x2afb ,0x1709,0x3bb6 ,0xc6d5,0xe33a ,
-  0xcf8a,0x29ce ,0x164c,0x3bfd ,0xc7ee,0xe124 ,
-  0xce87,0x289a ,0x1590,0x3c42 ,0xc91b,0xdf19 ,
-  0xcd8c,0x2760 ,0x14d2,0x3c85 ,0xca5b,0xdd19 ,
-  0xcc98,0x2620 ,0x1413,0x3cc5 ,0xcbad,0xdb26 ,
-  0xcbad,0x24da ,0x1354,0x3d03 ,0xcd11,0xd93f ,
-  0xcac9,0x238e ,0x1294,0x3d3f ,0xce87,0xd766 ,
-  0xc9ee,0x223d ,0x11d3,0x3d78 ,0xd00e,0xd59b ,
-  0xc91b,0x20e7 ,0x1112,0x3daf ,0xd1a6,0xd3df ,
-  0xc850,0x1f8c ,0x1050,0x3de3 ,0xd34e,0xd231 ,
-  0xc78f,0x1e2b ,0x0f8d,0x3e15 ,0xd505,0xd094 ,
-  0xc6d5,0x1cc6 ,0x0eca,0x3e45 ,0xd6cb,0xcf07 ,
-  0xc625,0x1b5d ,0x0e06,0x3e72 ,0xd8a0,0xcd8c ,
-  0xc57e,0x19ef ,0x0d41,0x3e9d ,0xda82,0xcc21 ,
-  0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
-  0xc44a,0x1709 ,0x0bb7,0x3eeb ,0xde6d,0xc983 ,
-  0xc3be,0x1590 ,0x0af1,0x3f0f ,0xe074,0xc850 ,
-  0xc33b,0x1413 ,0x0a2b,0x3f30 ,0xe287,0xc731 ,
-  0xc2c1,0x1294 ,0x0964,0x3f4f ,0xe4a3,0xc625 ,
-  0xc251,0x1112 ,0x089d,0x3f6b ,0xe6c9,0xc52d ,
-  0xc1eb,0x0f8d ,0x07d6,0x3f85 ,0xe8f7,0xc44a ,
-  0xc18e,0x0e06 ,0x070e,0x3f9c ,0xeb2e,0xc37b ,
-  0xc13b,0x0c7c ,0x0646,0x3fb1 ,0xed6c,0xc2c1 ,
-  0xc0f1,0x0af1 ,0x057e,0x3fc4 ,0xefb0,0xc21d ,
-  0xc0b1,0x0964 ,0x04b5,0x3fd4 ,0xf1fa,0xc18e ,
-  0xc07b,0x07d6 ,0x03ed,0x3fe1 ,0xf449,0xc115 ,
-  0xc04f,0x0646 ,0x0324,0x3fec ,0xf69c,0xc0b1 ,
-  0xc02c,0x04b5 ,0x025b,0x3ff5 ,0xf8f2,0xc064 ,
-  0xc014,0x0324 ,0x0192,0x3ffb ,0xfb4b,0xc02c ,
-  0xc005,0x0192 ,0x00c9,0x3fff ,0xfda5,0xc00b ,
-  0x4000,0x0000 ,0x4000,0x0065 ,0x3fff,0x00c9 ,
-  0x3ffd,0x012e ,0x3ffb,0x0192 ,0x3ff8,0x01f7 ,
-  0x3ff5,0x025b ,0x3ff1,0x02c0 ,0x3fec,0x0324 ,
-  0x3fe7,0x0388 ,0x3fe1,0x03ed ,0x3fdb,0x0451 ,
-  0x3fd4,0x04b5 ,0x3fcc,0x051a ,0x3fc4,0x057e ,
-  0x3fbb,0x05e2 ,0x3fb1,0x0646 ,0x3fa7,0x06aa ,
-  0x3f9c,0x070e ,0x3f91,0x0772 ,0x3f85,0x07d6 ,
-  0x3f78,0x0839 ,0x3f6b,0x089d ,0x3f5d,0x0901 ,
-  0x3f4f,0x0964 ,0x3f40,0x09c7 ,0x3f30,0x0a2b ,
-  0x3f20,0x0a8e ,0x3f0f,0x0af1 ,0x3efd,0x0b54 ,
-  0x3eeb,0x0bb7 ,0x3ed8,0x0c1a ,0x3ec5,0x0c7c ,
-  0x3eb1,0x0cdf ,0x3e9d,0x0d41 ,0x3e88,0x0da4 ,
-  0x3e72,0x0e06 ,0x3e5c,0x0e68 ,0x3e45,0x0eca ,
-  0x3e2d,0x0f2b ,0x3e15,0x0f8d ,0x3dfc,0x0fee ,
-  0x3de3,0x1050 ,0x3dc9,0x10b1 ,0x3daf,0x1112 ,
-  0x3d93,0x1173 ,0x3d78,0x11d3 ,0x3d5b,0x1234 ,
-  0x3d3f,0x1294 ,0x3d21,0x12f4 ,0x3d03,0x1354 ,
-  0x3ce4,0x13b4 ,0x3cc5,0x1413 ,0x3ca5,0x1473 ,
-  0x3c85,0x14d2 ,0x3c64,0x1531 ,0x3c42,0x1590 ,
-  0x3c20,0x15ee ,0x3bfd,0x164c ,0x3bda,0x16ab ,
-  0x3bb6,0x1709 ,0x3b92,0x1766 ,0x3b6d,0x17c4 ,
-  0x3b47,0x1821 ,0x3b21,0x187e ,0x3afa,0x18db ,
-  0x3ad3,0x1937 ,0x3aab,0x1993 ,0x3a82,0x19ef ,
-  0x3a59,0x1a4b ,0x3a30,0x1aa7 ,0x3a06,0x1b02 ,
-  0x39db,0x1b5d ,0x39b0,0x1bb8 ,0x3984,0x1c12 ,
-  0x3958,0x1c6c ,0x392b,0x1cc6 ,0x38fd,0x1d20 ,
-  0x38cf,0x1d79 ,0x38a1,0x1dd3 ,0x3871,0x1e2b ,
-  0x3842,0x1e84 ,0x3812,0x1edc ,0x37e1,0x1f34 ,
-  0x37b0,0x1f8c ,0x377e,0x1fe3 ,0x374b,0x203a ,
-  0x3718,0x2091 ,0x36e5,0x20e7 ,0x36b1,0x213d ,
-  0x367d,0x2193 ,0x3648,0x21e8 ,0x3612,0x223d ,
-  0x35dc,0x2292 ,0x35a5,0x22e7 ,0x356e,0x233b ,
-  0x3537,0x238e ,0x34ff,0x23e2 ,0x34c6,0x2435 ,
-  0x348d,0x2488 ,0x3453,0x24da ,0x3419,0x252c ,
-  0x33df,0x257e ,0x33a3,0x25cf ,0x3368,0x2620 ,
-  0x332c,0x2671 ,0x32ef,0x26c1 ,0x32b2,0x2711 ,
-  0x3274,0x2760 ,0x3236,0x27af ,0x31f8,0x27fe ,
-  0x31b9,0x284c ,0x3179,0x289a ,0x3139,0x28e7 ,
-  0x30f9,0x2935 ,0x30b8,0x2981 ,0x3076,0x29ce ,
-  0x3034,0x2a1a ,0x2ff2,0x2a65 ,0x2faf,0x2ab0 ,
-  0x2f6c,0x2afb ,0x2f28,0x2b45 ,0x2ee4,0x2b8f ,
-  0x2e9f,0x2bd8 ,0x2e5a,0x2c21 ,0x2e15,0x2c6a ,
-  0x2dcf,0x2cb2 ,0x2d88,0x2cfa ,0x2d41,0x2d41 ,
-  0x2cfa,0x2d88 ,0x2cb2,0x2dcf ,0x2c6a,0x2e15 ,
-  0x2c21,0x2e5a ,0x2bd8,0x2e9f ,0x2b8f,0x2ee4 ,
-  0x2b45,0x2f28 ,0x2afb,0x2f6c ,0x2ab0,0x2faf ,
-  0x2a65,0x2ff2 ,0x2a1a,0x3034 ,0x29ce,0x3076 ,
-  0x2981,0x30b8 ,0x2935,0x30f9 ,0x28e7,0x3139 ,
-  0x289a,0x3179 ,0x284c,0x31b9 ,0x27fe,0x31f8 ,
-  0x27af,0x3236 ,0x2760,0x3274 ,0x2711,0x32b2 ,
-  0x26c1,0x32ef ,0x2671,0x332c ,0x2620,0x3368 ,
-  0x25cf,0x33a3 ,0x257e,0x33df ,0x252c,0x3419 ,
-  0x24da,0x3453 ,0x2488,0x348d ,0x2435,0x34c6 ,
-  0x23e2,0x34ff ,0x238e,0x3537 ,0x233b,0x356e ,
-  0x22e7,0x35a5 ,0x2292,0x35dc ,0x223d,0x3612 ,
-  0x21e8,0x3648 ,0x2193,0x367d ,0x213d,0x36b1 ,
-  0x20e7,0x36e5 ,0x2091,0x3718 ,0x203a,0x374b ,
-  0x1fe3,0x377e ,0x1f8c,0x37b0 ,0x1f34,0x37e1 ,
-  0x1edc,0x3812 ,0x1e84,0x3842 ,0x1e2b,0x3871 ,
-  0x1dd3,0x38a1 ,0x1d79,0x38cf ,0x1d20,0x38fd ,
-  0x1cc6,0x392b ,0x1c6c,0x3958 ,0x1c12,0x3984 ,
-  0x1bb8,0x39b0 ,0x1b5d,0x39db ,0x1b02,0x3a06 ,
-  0x1aa7,0x3a30 ,0x1a4b,0x3a59 ,0x19ef,0x3a82 ,
-  0x1993,0x3aab ,0x1937,0x3ad3 ,0x18db,0x3afa ,
-  0x187e,0x3b21 ,0x1821,0x3b47 ,0x17c4,0x3b6d ,
-  0x1766,0x3b92 ,0x1709,0x3bb6 ,0x16ab,0x3bda ,
-  0x164c,0x3bfd ,0x15ee,0x3c20 ,0x1590,0x3c42 ,
-  0x1531,0x3c64 ,0x14d2,0x3c85 ,0x1473,0x3ca5 ,
-  0x1413,0x3cc5 ,0x13b4,0x3ce4 ,0x1354,0x3d03 ,
-  0x12f4,0x3d21 ,0x1294,0x3d3f ,0x1234,0x3d5b ,
-  0x11d3,0x3d78 ,0x1173,0x3d93 ,0x1112,0x3daf ,
-  0x10b1,0x3dc9 ,0x1050,0x3de3 ,0x0fee,0x3dfc ,
-  0x0f8d,0x3e15 ,0x0f2b,0x3e2d ,0x0eca,0x3e45 ,
-  0x0e68,0x3e5c ,0x0e06,0x3e72 ,0x0da4,0x3e88 ,
-  0x0d41,0x3e9d ,0x0cdf,0x3eb1 ,0x0c7c,0x3ec5 ,
-  0x0c1a,0x3ed8 ,0x0bb7,0x3eeb ,0x0b54,0x3efd ,
-  0x0af1,0x3f0f ,0x0a8e,0x3f20 ,0x0a2b,0x3f30 ,
-  0x09c7,0x3f40 ,0x0964,0x3f4f ,0x0901,0x3f5d ,
-  0x089d,0x3f6b ,0x0839,0x3f78 ,0x07d6,0x3f85 ,
-  0x0772,0x3f91 ,0x070e,0x3f9c ,0x06aa,0x3fa7 ,
-  0x0646,0x3fb1 ,0x05e2,0x3fbb ,0x057e,0x3fc4 ,
-  0x051a,0x3fcc ,0x04b5,0x3fd4 ,0x0451,0x3fdb ,
-  0x03ed,0x3fe1 ,0x0388,0x3fe7 ,0x0324,0x3fec ,
-  0x02c0,0x3ff1 ,0x025b,0x3ff5 ,0x01f7,0x3ff8 ,
-  0x0192,0x3ffb ,0x012e,0x3ffd ,0x00c9,0x3fff ,
-  0x0065,0x4000 ,0x0000,0x4000 ,0xff9b,0x4000 ,
-  0xff37,0x3fff ,0xfed2,0x3ffd ,0xfe6e,0x3ffb ,
-  0xfe09,0x3ff8 ,0xfda5,0x3ff5 ,0xfd40,0x3ff1 ,
-  0xfcdc,0x3fec ,0xfc78,0x3fe7 ,0xfc13,0x3fe1 ,
-  0xfbaf,0x3fdb ,0xfb4b,0x3fd4 ,0xfae6,0x3fcc ,
-  0xfa82,0x3fc4 ,0xfa1e,0x3fbb ,0xf9ba,0x3fb1 ,
-  0xf956,0x3fa7 ,0xf8f2,0x3f9c ,0xf88e,0x3f91 ,
-  0xf82a,0x3f85 ,0xf7c7,0x3f78 ,0xf763,0x3f6b ,
-  0xf6ff,0x3f5d ,0xf69c,0x3f4f ,0xf639,0x3f40 ,
-  0xf5d5,0x3f30 ,0xf572,0x3f20 ,0xf50f,0x3f0f ,
-  0xf4ac,0x3efd ,0xf449,0x3eeb ,0xf3e6,0x3ed8 ,
-  0xf384,0x3ec5 ,0xf321,0x3eb1 ,0xf2bf,0x3e9d ,
-  0xf25c,0x3e88 ,0xf1fa,0x3e72 ,0xf198,0x3e5c ,
-  0xf136,0x3e45 ,0xf0d5,0x3e2d ,0xf073,0x3e15 ,
-  0xf012,0x3dfc ,0xefb0,0x3de3 ,0xef4f,0x3dc9 ,
-  0xeeee,0x3daf ,0xee8d,0x3d93 ,0xee2d,0x3d78 ,
-  0xedcc,0x3d5b ,0xed6c,0x3d3f ,0xed0c,0x3d21 ,
-  0xecac,0x3d03 ,0xec4c,0x3ce4 ,0xebed,0x3cc5 ,
-  0xeb8d,0x3ca5 ,0xeb2e,0x3c85 ,0xeacf,0x3c64 ,
-  0xea70,0x3c42 ,0xea12,0x3c20 ,0xe9b4,0x3bfd ,
-  0xe955,0x3bda ,0xe8f7,0x3bb6 ,0xe89a,0x3b92 ,
-  0xe83c,0x3b6d ,0xe7df,0x3b47 ,0xe782,0x3b21 ,
-  0xe725,0x3afa ,0xe6c9,0x3ad3 ,0xe66d,0x3aab ,
-  0xe611,0x3a82 ,0xe5b5,0x3a59 ,0xe559,0x3a30 ,
-  0xe4fe,0x3a06 ,0xe4a3,0x39db ,0xe448,0x39b0 ,
-  0xe3ee,0x3984 ,0xe394,0x3958 ,0xe33a,0x392b ,
-  0xe2e0,0x38fd ,0xe287,0x38cf ,0xe22d,0x38a1 ,
-  0xe1d5,0x3871 ,0xe17c,0x3842 ,0xe124,0x3812 ,
-  0xe0cc,0x37e1 ,0xe074,0x37b0 ,0xe01d,0x377e ,
-  0xdfc6,0x374b ,0xdf6f,0x3718 ,0xdf19,0x36e5 ,
-  0xdec3,0x36b1 ,0xde6d,0x367d ,0xde18,0x3648 ,
-  0xddc3,0x3612 ,0xdd6e,0x35dc ,0xdd19,0x35a5 ,
-  0xdcc5,0x356e ,0xdc72,0x3537 ,0xdc1e,0x34ff ,
-  0xdbcb,0x34c6 ,0xdb78,0x348d ,0xdb26,0x3453 ,
-  0xdad4,0x3419 ,0xda82,0x33df ,0xda31,0x33a3 ,
-  0xd9e0,0x3368 ,0xd98f,0x332c ,0xd93f,0x32ef ,
-  0xd8ef,0x32b2 ,0xd8a0,0x3274 ,0xd851,0x3236 ,
-  0xd802,0x31f8 ,0xd7b4,0x31b9 ,0xd766,0x3179 ,
-  0xd719,0x3139 ,0xd6cb,0x30f9 ,0xd67f,0x30b8 ,
-  0xd632,0x3076 ,0xd5e6,0x3034 ,0xd59b,0x2ff2 ,
-  0xd550,0x2faf ,0xd505,0x2f6c ,0xd4bb,0x2f28 ,
-  0xd471,0x2ee4 ,0xd428,0x2e9f ,0xd3df,0x2e5a ,
-  0xd396,0x2e15 ,0xd34e,0x2dcf ,0xd306,0x2d88 ,
-  0xd2bf,0x2d41 ,0xd278,0x2cfa ,0xd231,0x2cb2 ,
-  0xd1eb,0x2c6a ,0xd1a6,0x2c21 ,0xd161,0x2bd8 ,
-  0xd11c,0x2b8f ,0xd0d8,0x2b45 ,0xd094,0x2afb ,
-  0xd051,0x2ab0 ,0xd00e,0x2a65 ,0xcfcc,0x2a1a ,
-  0xcf8a,0x29ce ,0xcf48,0x2981 ,0xcf07,0x2935 ,
-  0xcec7,0x28e7 ,0xce87,0x289a ,0xce47,0x284c ,
-  0xce08,0x27fe ,0xcdca,0x27af ,0xcd8c,0x2760 ,
-  0xcd4e,0x2711 ,0xcd11,0x26c1 ,0xccd4,0x2671 ,
-  0xcc98,0x2620 ,0xcc5d,0x25cf ,0xcc21,0x257e ,
-  0xcbe7,0x252c ,0xcbad,0x24da ,0xcb73,0x2488 ,
-  0xcb3a,0x2435 ,0xcb01,0x23e2 ,0xcac9,0x238e ,
-  0xca92,0x233b ,0xca5b,0x22e7 ,0xca24,0x2292 ,
-  0xc9ee,0x223d ,0xc9b8,0x21e8 ,0xc983,0x2193 ,
-  0xc94f,0x213d ,0xc91b,0x20e7 ,0xc8e8,0x2091 ,
-  0xc8b5,0x203a ,0xc882,0x1fe3 ,0xc850,0x1f8c ,
-  0xc81f,0x1f34 ,0xc7ee,0x1edc ,0xc7be,0x1e84 ,
-  0xc78f,0x1e2b ,0xc75f,0x1dd3 ,0xc731,0x1d79 ,
-  0xc703,0x1d20 ,0xc6d5,0x1cc6 ,0xc6a8,0x1c6c ,
-  0xc67c,0x1c12 ,0xc650,0x1bb8 ,0xc625,0x1b5d ,
-  0xc5fa,0x1b02 ,0xc5d0,0x1aa7 ,0xc5a7,0x1a4b ,
-  0xc57e,0x19ef ,0xc555,0x1993 ,0xc52d,0x1937 ,
-  0xc506,0x18db ,0xc4df,0x187e ,0xc4b9,0x1821 ,
-  0xc493,0x17c4 ,0xc46e,0x1766 ,0xc44a,0x1709 ,
-  0xc426,0x16ab ,0xc403,0x164c ,0xc3e0,0x15ee ,
-  0xc3be,0x1590 ,0xc39c,0x1531 ,0xc37b,0x14d2 ,
-  0xc35b,0x1473 ,0xc33b,0x1413 ,0xc31c,0x13b4 ,
-  0xc2fd,0x1354 ,0xc2df,0x12f4 ,0xc2c1,0x1294 ,
-  0xc2a5,0x1234 ,0xc288,0x11d3 ,0xc26d,0x1173 ,
-  0xc251,0x1112 ,0xc237,0x10b1 ,0xc21d,0x1050 ,
-  0xc204,0x0fee ,0xc1eb,0x0f8d ,0xc1d3,0x0f2b ,
-  0xc1bb,0x0eca ,0xc1a4,0x0e68 ,0xc18e,0x0e06 ,
-  0xc178,0x0da4 ,0xc163,0x0d41 ,0xc14f,0x0cdf ,
-  0xc13b,0x0c7c ,0xc128,0x0c1a ,0xc115,0x0bb7 ,
-  0xc103,0x0b54 ,0xc0f1,0x0af1 ,0xc0e0,0x0a8e ,
-  0xc0d0,0x0a2b ,0xc0c0,0x09c7 ,0xc0b1,0x0964 ,
-  0xc0a3,0x0901 ,0xc095,0x089d ,0xc088,0x0839 ,
-  0xc07b,0x07d6 ,0xc06f,0x0772 ,0xc064,0x070e ,
-  0xc059,0x06aa ,0xc04f,0x0646 ,0xc045,0x05e2 ,
-  0xc03c,0x057e ,0xc034,0x051a ,0xc02c,0x04b5 ,
-  0xc025,0x0451 ,0xc01f,0x03ed ,0xc019,0x0388 ,
-  0xc014,0x0324 ,0xc00f,0x02c0 ,0xc00b,0x025b ,
-  0xc008,0x01f7 ,0xc005,0x0192 ,0xc003,0x012e ,
-  0xc001,0x00c9 ,0xc000,0x0065 };
diff --git a/src/common_audio/signal_processing_library/main/source/webrtc_fft_t_rad.c b/src/common_audio/signal_processing_library/main/source/webrtc_fft_t_rad.c
deleted file mode 100644
index 13fbd9f..0000000
--- a/src/common_audio/signal_processing_library/main/source/webrtc_fft_t_rad.c
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the Q14 radix-2 tables used in ARM9E optimization routines.
- *
- */
-
-extern const unsigned short t_Q14S_rad8[2];
-const unsigned short t_Q14S_rad8[2] = {  0x0000,0x2d41 };
-
-//extern const int t_Q30S_rad8[2];
-//const int t_Q30S_rad8[2] = {  0x00000000,0x2d413ccd };
-
-extern const unsigned short t_Q14R_rad8[2];
-const unsigned short t_Q14R_rad8[2] = {  0x2d41,0x2d41 };
-
-//extern const int t_Q30R_rad8[2];
-//const int t_Q30R_rad8[2] = {  0x2d413ccd,0x2d413ccd };
diff --git a/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.cc b/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.cc
deleted file mode 100644
index 5adc339..0000000
--- a/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.cc
+++ /dev/null
@@ -1,479 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the SPL unit_test.
- *
- */
-
-#include "unit_test.h"
-#include "signal_processing_library.h"
-
-class SplEnvironment : public ::testing::Environment {
- public:
-  virtual void SetUp() {
-  }
-  virtual void TearDown() {
-  }
-};
-
-SplTest::SplTest()
-{
-}
-
-void SplTest::SetUp() {
-}
-
-void SplTest::TearDown() {
-}
-
-TEST_F(SplTest, MacroTest) {
-    // Macros with inputs.
-    int A = 10;
-    int B = 21;
-    int a = -3;
-    int b = WEBRTC_SPL_WORD32_MAX;
-    int nr = 2;
-    int d_ptr1 = 0;
-    int d_ptr2 = 0;
-
-    EXPECT_EQ(10, WEBRTC_SPL_MIN(A, B));
-    EXPECT_EQ(21, WEBRTC_SPL_MAX(A, B));
-
-    EXPECT_EQ(3, WEBRTC_SPL_ABS_W16(a));
-    EXPECT_EQ(3, WEBRTC_SPL_ABS_W32(a));
-    EXPECT_EQ(0, WEBRTC_SPL_GET_BYTE(&B, nr));
-    WEBRTC_SPL_SET_BYTE(&d_ptr2, 1, nr);
-    EXPECT_EQ(65536, d_ptr2);
-
-    EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
-    EXPECT_EQ(-2147483645, WEBRTC_SPL_MUL(a, b));
-    EXPECT_EQ(-2147483645, WEBRTC_SPL_UMUL(a, b));
-    b = WEBRTC_SPL_WORD16_MAX >> 1;
-    EXPECT_EQ(65535, WEBRTC_SPL_UMUL_RSFT16(a, b));
-    EXPECT_EQ(1073627139, WEBRTC_SPL_UMUL_16_16(a, b));
-    EXPECT_EQ(16382, WEBRTC_SPL_UMUL_16_16_RSFT16(a, b));
-    EXPECT_EQ(-49149, WEBRTC_SPL_UMUL_32_16(a, b));
-    EXPECT_EQ(65535, WEBRTC_SPL_UMUL_32_16_RSFT16(a, b));
-    EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));
-
-    a = b;
-    b = -3;
-    EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b));
-    EXPECT_EQ(0, WEBRTC_SPL_UDIV(a, b));
-
-    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
-    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
-    EXPECT_EQ(-3, WEBRTC_SPL_MUL_16_32_RSFT14(a, b));
-    EXPECT_EQ(-24, WEBRTC_SPL_MUL_16_32_RSFT11(a, b));
-
-    int a32 = WEBRTC_SPL_WORD32_MAX;
-    int a32a = (WEBRTC_SPL_WORD32_MAX >> 16);
-    int a32b = (WEBRTC_SPL_WORD32_MAX & 0x0000ffff);
-    EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, A));
-    EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, A));
-
-    EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_16(a, b));
-    EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2));
-
-    EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2));
-    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b));
-
-    EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W32(a, b));
-    EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
-    EXPECT_EQ(21, WEBRTC_SPL_SAT(a, B, A));
-    EXPECT_EQ(-49149, WEBRTC_SPL_MUL_32_16(a, b));
-
-    EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W32(a, b));
-    EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W16(a, b));
-    EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W16(a, b));
-
-    EXPECT_TRUE(WEBRTC_SPL_IS_NEG(b));
-
-    // Shifting with negative numbers allowed
-    // Positive means left shift
-    EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W16(a, 1));
-    EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W32(a, 1));
-
-    // Shifting with negative numbers not allowed
-    // We cannot do casting here due to signed/unsigned problem
-    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W16(a, 1));
-    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W16(a, 1));
-    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W32(a, 1));
-    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
-
-    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_U16(a, 1));
-    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_U16(a, 1));
-    EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_U32(a, 1));
-    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_U32(a, 1));
-
-    EXPECT_EQ(1470, WEBRTC_SPL_RAND(A));
-}
-
-TEST_F(SplTest, InlineTest) {
-
-    WebRtc_Word16 a = 121;
-    WebRtc_Word16 b = -17;
-    WebRtc_Word32 A = 111121;
-    WebRtc_Word32 B = -1711;
-    char* bVersion = (char*) malloc(8);
-
-    EXPECT_EQ(104, WebRtcSpl_AddSatW16(a, b));
-    EXPECT_EQ(138, WebRtcSpl_SubSatW16(a, b));
-
-    EXPECT_EQ(109410, WebRtcSpl_AddSatW32(A, B));
-    EXPECT_EQ(112832, WebRtcSpl_SubSatW32(A, B));
-
-    EXPECT_EQ(17, WebRtcSpl_GetSizeInBits(A));
-    EXPECT_EQ(14, WebRtcSpl_NormW32(A));
-    EXPECT_EQ(4, WebRtcSpl_NormW16(B));
-    EXPECT_EQ(15, WebRtcSpl_NormU32(A));
-
-    EXPECT_EQ(0, WebRtcSpl_get_version(bVersion, 8));
-}
-
-TEST_F(SplTest, MathOperationsTest) {
-
-    int A = 117;
-    WebRtc_Word32 num = 117;
-    WebRtc_Word32 den = -5;
-    WebRtc_UWord16 denU = 5;
-    EXPECT_EQ(10, WebRtcSpl_Sqrt(A));
-    EXPECT_EQ(10, WebRtcSpl_SqrtFloor(A));
-
-
-    EXPECT_EQ(-91772805, WebRtcSpl_DivResultInQ31(den, num));
-    EXPECT_EQ(-23, WebRtcSpl_DivW32W16ResW16(num, (WebRtc_Word16)den));
-    EXPECT_EQ(-23, WebRtcSpl_DivW32W16(num, (WebRtc_Word16)den));
-    EXPECT_EQ(23, WebRtcSpl_DivU32U16(num, denU));
-    EXPECT_EQ(0, WebRtcSpl_DivW32HiLow(128, 0, 256));
-}
-
-TEST_F(SplTest, BasicArrayOperationsTest) {
-
-
-    int B[] = {4, 12, 133, 1100};
-    int Bs[] = {2, 6, 66, 550};
-    WebRtc_UWord8* b8 = (WebRtc_UWord8*) malloc(4);
-    WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word32* b32 = (WebRtc_Word32*) malloc(4);
-
-    WebRtc_UWord8* bTmp8 = (WebRtc_UWord8*) malloc(4);
-    WebRtc_Word16* bTmp16 = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word32* bTmp32 = (WebRtc_Word32*) malloc(4);
-
-    WebRtcSpl_MemSetW16(b16, 3, 4);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(3, b16[kk]);
-    }
-    EXPECT_EQ(4, WebRtcSpl_ZerosArrayW16(b16, 4));
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(0, b16[kk]);
-    }
-    EXPECT_EQ(4, WebRtcSpl_OnesArrayW16(b16, 4));
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(1, b16[kk]);
-    }
-    WebRtcSpl_MemSetW32(b32, 3, 4);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(3, b32[kk]);
-    }
-    EXPECT_EQ(4, WebRtcSpl_ZerosArrayW32(b32, 4));
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(0, b32[kk]);
-    }
-    EXPECT_EQ(4, WebRtcSpl_OnesArrayW32(b32, 4));
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(1, b32[kk]);
-    }
-    for (int kk = 0; kk < 4; ++kk) {
-        bTmp8[kk] = (WebRtc_Word8)kk;
-        bTmp16[kk] = (WebRtc_Word16)kk;
-        bTmp32[kk] = (WebRtc_Word32)kk;
-    }
-    WEBRTC_SPL_MEMCPY_W8(b8, bTmp8, 4);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(b8[kk], bTmp8[kk]);
-    }
-    WEBRTC_SPL_MEMCPY_W16(b16, bTmp16, 4);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(b16[kk], bTmp16[kk]);
-    }
-//    WEBRTC_SPL_MEMCPY_W32(b32, bTmp32, 4);
-//    for (int kk = 0; kk < 4; ++kk) {
-//        EXPECT_EQ(b32[kk], bTmp32[kk]);
-//    }
-    EXPECT_EQ(2, WebRtcSpl_CopyFromEndW16(b16, 4, 2, bTmp16));
-    for (int kk = 0; kk < 2; ++kk) {
-        EXPECT_EQ(kk+2, bTmp16[kk]);
-    }
-
-    for (int kk = 0; kk < 4; ++kk) {
-        b32[kk] = B[kk];
-        b16[kk] = (WebRtc_Word16)B[kk];
-    }
-    WebRtcSpl_VectorBitShiftW32ToW16(bTmp16, 4, b32, 1);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
-    }
-    WebRtcSpl_VectorBitShiftW16(bTmp16, 4, b16, 1);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
-    }
-    WebRtcSpl_VectorBitShiftW32(bTmp32, 4, b32, 1);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ((B[kk]>>1), bTmp32[kk]);
-    }
-
-    WebRtcSpl_MemCpyReversedOrder(&bTmp16[3], b16, 4);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(b16[3-kk], bTmp16[kk]);
-    }
-
-}
-
-TEST_F(SplTest, MinMaxOperationsTest) {
-
-
-    int B[] = {4, 12, 133, -1100};
-    WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word32* b32 = (WebRtc_Word32*) malloc(4);
-
-    for (int kk = 0; kk < 4; ++kk) {
-        b16[kk] = B[kk];
-        b32[kk] = B[kk];
-    }
-
-    EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW16(b16, 4));
-    EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW32(b32, 4));
-    EXPECT_EQ(133, WebRtcSpl_MaxValueW16(b16, 4));
-    EXPECT_EQ(133, WebRtcSpl_MaxValueW32(b32, 4));
-    EXPECT_EQ(3, WebRtcSpl_MaxAbsIndexW16(b16, 4));
-    EXPECT_EQ(2, WebRtcSpl_MaxIndexW16(b16, 4));
-    EXPECT_EQ(2, WebRtcSpl_MaxIndexW32(b32, 4));
-
-    EXPECT_EQ(-1100, WebRtcSpl_MinValueW16(b16, 4));
-    EXPECT_EQ(-1100, WebRtcSpl_MinValueW32(b32, 4));
-    EXPECT_EQ(3, WebRtcSpl_MinIndexW16(b16, 4));
-    EXPECT_EQ(3, WebRtcSpl_MinIndexW32(b32, 4));
-
-    EXPECT_EQ(0, WebRtcSpl_GetScalingSquare(b16, 4, 1));
-
-}
-
-TEST_F(SplTest, VectorOperationsTest) {
-
-
-    int B[] = {4, 12, 133, 1100};
-    WebRtc_Word16* a16 = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word32* b32 = (WebRtc_Word32*) malloc(4);
-    WebRtc_Word16* bTmp16 = (WebRtc_Word16*) malloc(4);
-
-    for (int kk = 0; kk < 4; ++kk) {
-        a16[kk] = B[kk];
-        b16[kk] = B[kk];
-    }
-
-    WebRtcSpl_AffineTransformVector(bTmp16, b16, 3, 7, 2, 4);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ((B[kk]*3+7)>>2, bTmp16[kk]);
-    }
-    WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16, 4);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ((B[kk]*3+B[kk]*2+2)>>2, bTmp16[kk]);
-    }
-
-    WebRtcSpl_AddAffineVectorToVector(bTmp16, b16, 3, 7, 2, 4);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(((B[kk]*3+B[kk]*2+2)>>2)+((b16[kk]*3+7)>>2), bTmp16[kk]);
-    }
-
-    WebRtcSpl_CrossCorrelation(b32, b16, bTmp16, 4, 2, 2, 0);
-    for (int kk = 0; kk < 2; ++kk) {
-        EXPECT_EQ(614236, b32[kk]);
-    }
-//    EXPECT_EQ(, WebRtcSpl_DotProduct(b16, bTmp16, 4));
-    EXPECT_EQ(306962, WebRtcSpl_DotProductWithScale(b16, b16, 4, 2));
-
-    WebRtcSpl_ScaleVector(b16, bTmp16, 13, 4, 2);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
-    }
-    WebRtcSpl_ScaleVectorWithSat(b16, bTmp16, 13, 4, 2);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
-    }
-    WebRtcSpl_ScaleAndAddVectors(a16, 13, 2, b16, 7, 2, bTmp16, 4);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(((a16[kk]*13)>>2)+((b16[kk]*7)>>2), bTmp16[kk]);
-    }
-
-    WebRtcSpl_AddVectorsAndShift(bTmp16, a16, b16, 4, 2);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
-    }
-    WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3], 4, 2);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ((a16[kk]*b16[3-kk])>>2, bTmp16[kk]);
-    }
-    WebRtcSpl_ElementwiseVectorMult(bTmp16, a16, b16, 4, 6);
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ((a16[kk]*b16[kk])>>6, bTmp16[kk]);
-    }
-
-    WebRtcSpl_SqrtOfOneMinusXSquared(b16, 4, bTmp16);
-    for (int kk = 0; kk < 3; ++kk) {
-        EXPECT_EQ(32767, bTmp16[kk]);
-    }
-    EXPECT_EQ(32749, bTmp16[3]);
-}
-
-TEST_F(SplTest, EstimatorsTest) {
-
-
-    int B[] = {4, 12, 133, 1100};
-    WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word32* b32 = (WebRtc_Word32*) malloc(4);
-    WebRtc_Word16* bTmp16 = (WebRtc_Word16*) malloc(4);
-
-    for (int kk = 0; kk < 4; ++kk) {
-        b16[kk] = B[kk];
-        b32[kk] = B[kk];
-    }
-
-    EXPECT_EQ(0, WebRtcSpl_LevinsonDurbin(b32, b16, bTmp16, 2));
-
-}
-
-TEST_F(SplTest, FilterTest) {
-
-
-    WebRtc_Word16 A[] = {1, 2, 33, 100};
-    WebRtc_Word16 A5[] = {1, 2, 33, 100, -5};
-    WebRtc_Word16 B[] = {4, 12, 133, 110};
-    WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word16* bTmp16 = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word16* bTmp16Low = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word16* bState = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word16* bStateLow = (WebRtc_Word16*) malloc(4);
-
-    WebRtcSpl_ZerosArrayW16(bState, 4);
-    WebRtcSpl_ZerosArrayW16(bStateLow, 4);
-
-    for (int kk = 0; kk < 4; ++kk) {
-        b16[kk] = A[kk];
-    }
-
-    // MA filters
-    WebRtcSpl_FilterMAFastQ12(b16, bTmp16, B, 4, 4);
-    for (int kk = 0; kk < 4; ++kk) {
-        //EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-    }
-    // AR filters
-    WebRtcSpl_FilterARFastQ12(b16, bTmp16, A, 4, 4);
-    for (int kk = 0; kk < 4; ++kk) {
-//        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-    }
-    EXPECT_EQ(4, WebRtcSpl_FilterAR(A5, 5, b16, 4, bState, 4, bStateLow, 4, bTmp16, bTmp16Low, 4));
-
-}
-
-TEST_F(SplTest, RandTest) {
-
-
-    WebRtc_Word16 BU[] = {3653, 12446, 8525, 30691};
-    WebRtc_Word16 BN[] = {3459, -11689, -258, -3738};
-    WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
-    WebRtc_UWord32* bSeed = (WebRtc_UWord32*) malloc(1);
-
-    bSeed[0] = 100000;
-
-    EXPECT_EQ(464449057, WebRtcSpl_IncreaseSeed(bSeed));
-    EXPECT_EQ(31565, WebRtcSpl_RandU(bSeed));
-    EXPECT_EQ(-9786, WebRtcSpl_RandN(bSeed));
-    EXPECT_EQ(4, WebRtcSpl_RandUArray(b16, 4, bSeed));
-    for (int kk = 0; kk < 4; ++kk) {
-        EXPECT_EQ(BU[kk], b16[kk]);
-    }
-}
-
-TEST_F(SplTest, SignalProcessingTest) {
-
-
-    int A[] = {1, 2, 33, 100};
-    WebRtc_Word16* b16 = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word32* b32 = (WebRtc_Word32*) malloc(4);
-
-    WebRtc_Word16* bTmp16 = (WebRtc_Word16*) malloc(4);
-    WebRtc_Word32* bTmp32 = (WebRtc_Word32*) malloc(4);
-
-    int bScale = 0;
-
-    for (int kk = 0; kk < 4; ++kk) {
-        b16[kk] = A[kk];
-        b32[kk] = A[kk];
-    }
-
-    EXPECT_EQ(2, WebRtcSpl_AutoCorrelation(b16, 4, 1, bTmp32, &bScale));
-    WebRtcSpl_ReflCoefToLpc(b16, 4, bTmp16);
-//    for (int kk = 0; kk < 4; ++kk) {
-//        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-//    }
-    WebRtcSpl_LpcToReflCoef(bTmp16, 4, b16);
-//    for (int kk = 0; kk < 4; ++kk) {
-//        EXPECT_EQ(a16[kk], b16[kk]);
-//    }
-    WebRtcSpl_AutoCorrToReflCoef(b32, 4, bTmp16);
-//    for (int kk = 0; kk < 4; ++kk) {
-//        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-//    }
-    WebRtcSpl_GetHanningWindow(bTmp16, 4);
-//    for (int kk = 0; kk < 4; ++kk) {
-//        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-//    }
-
-    for (int kk = 0; kk < 4; ++kk) {
-        b16[kk] = A[kk];
-    }
-    EXPECT_EQ(11094 , WebRtcSpl_Energy(b16, 4, &bScale));
-    EXPECT_EQ(0, bScale);
-}
-
-TEST_F(SplTest, FFTTest) {
-
-
-    WebRtc_Word16 B[] = {1, 2, 33, 100,
-            2, 3, 34, 101,
-            3, 4, 35, 102,
-            4, 5, 36, 103};
-
-    EXPECT_EQ(0, WebRtcSpl_ComplexFFT(B, 3, 1));
-//    for (int kk = 0; kk < 16; ++kk) {
-//        EXPECT_EQ(A[kk], B[kk]);
-//    }
-    EXPECT_EQ(0, WebRtcSpl_ComplexIFFT(B, 3, 1));
-//    for (int kk = 0; kk < 16; ++kk) {
-//        EXPECT_EQ(A[kk], B[kk]);
-//    }
-    WebRtcSpl_ComplexBitReverse(B, 3);
-    for (int kk = 0; kk < 16; ++kk) {
-        //EXPECT_EQ(A[kk], B[kk]);
-    }
-}
-
-int main(int argc, char** argv) {
-  ::testing::InitGoogleTest(&argc, argv);
-  SplEnvironment* env = new SplEnvironment;
-  ::testing::AddGlobalTestEnvironment(env);
-
-  return RUN_ALL_TESTS();
-}
diff --git a/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.h b/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.h
deleted file mode 100644
index d7babe7..0000000
--- a/src/common_audio/signal_processing_library/main/test/unit_test/unit_test.h
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This header file contains the function WebRtcSpl_CopyFromBeginU8().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#ifndef WEBRTC_SPL_UNIT_TEST_H_
-#define WEBRTC_SPL_UNIT_TEST_H_
-
-#include <gtest/gtest.h>
-
-class SplTest: public ::testing::Test
-{
-protected:
-    SplTest();
-    virtual void SetUp();
-    virtual void TearDown();
-};
-
-#endif  // WEBRTC_SPL_UNIT_TEST_H_
diff --git a/src/common_audio/vad/OWNERS b/src/common_audio/vad/OWNERS
deleted file mode 100644
index 9132851..0000000
--- a/src/common_audio/vad/OWNERS
+++ /dev/null
@@ -1,2 +0,0 @@
-bjornv@google.com
-jks@google.com
diff --git a/src/common_audio/vad/main/interface/webrtc_vad.h b/src/common_audio/vad/main/interface/webrtc_vad.h
deleted file mode 100644
index 6e3eb74..0000000
--- a/src/common_audio/vad/main/interface/webrtc_vad.h
+++ /dev/null
@@ -1,159 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the VAD API calls. Specific function calls are given below.
- */
-
-#ifndef WEBRTC_VAD_WEBRTC_VAD_H_
-#define WEBRTC_VAD_WEBRTC_VAD_H_
-
-#include "typedefs.h"
-
-typedef struct WebRtcVadInst VadInst;
-
-#ifdef __cplusplus
-extern "C"
-{
-#endif
-
-/****************************************************************************
- * WebRtcVad_get_version(...)
- *
- * This function returns the version number of the code.
- *
- * Output:
- *      - version       : Pointer to a buffer where the version info will
- *                        be stored.
- * Input:
- *      - size_bytes    : Size of the buffer.
- *
- */
-WebRtc_Word16 WebRtcVad_get_version(char *version, size_t size_bytes);
-
-/****************************************************************************
- * WebRtcVad_AssignSize(...) 
- *
- * This functions get the size needed for storing the instance for encoder
- * and decoder, respectively
- *
- * Input/Output:
- *      - size_in_bytes : Pointer to integer where the size is returned
- *
- * Return value         : 0
- */
-WebRtc_Word16 WebRtcVad_AssignSize(int *size_in_bytes);
-
-/****************************************************************************
- * WebRtcVad_Assign(...) 
- *
- * This functions Assigns memory for the instances.
- *
- * Input:
- *        - vad_inst_addr :  Address to where to assign memory
- * Output:
- *        - vad_inst      :  Pointer to the instance that should be created
- *
- * Return value           :  0 - Ok
- *                          -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Assign(VadInst **vad_inst, void *vad_inst_addr);
-
-/****************************************************************************
- * WebRtcVad_Create(...)
- *
- * This function creates an instance to the VAD structure
- *
- * Input:
- *      - vad_inst      : Pointer to VAD instance that should be created
- *
- * Output:
- *      - vad_inst      : Pointer to created VAD instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Create(VadInst **vad_inst);
-
-/****************************************************************************
- * WebRtcVad_Free(...)
- *
- * This function frees the dynamic memory of a specified VAD instance
- *
- * Input:
- *      - vad_inst      : Pointer to VAD instance that should be freed
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Free(VadInst *vad_inst);
-
-/****************************************************************************
- * WebRtcVad_Init(...)
- *
- * This function initializes a VAD instance
- *
- * Input:
- *      - vad_inst      : Instance that should be initialized
- *
- * Output:
- *      - vad_inst      : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Init(VadInst *vad_inst);
-
-/****************************************************************************
- * WebRtcVad_set_mode(...)
- *
- * This function initializes a VAD instance
- *
- * Input:
- *      - vad_inst      : VAD instance
- *      - mode          : Aggressiveness setting (0, 1, 2, or 3) 
- *
- * Output:
- *      - vad_inst      : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-WebRtc_Word16 WebRtcVad_set_mode(VadInst *vad_inst, WebRtc_Word16 mode);
-
-/****************************************************************************
- * WebRtcVad_Process(...)
- * 
- * This functions does a VAD for the inserted speech frame
- *
- * Input
- *        - vad_inst     : VAD Instance. Needs to be initiated before call.
- *        - fs           : sampling frequency (Hz): 8000, 16000, or 32000
- *        - speech_frame : Pointer to speech frame buffer
- *        - frame_length : Length of speech frame buffer in number of samples
- *
- * Output:
- *        - vad_inst     : Updated VAD instance
- *
- * Return value          :  1 - Active Voice
- *                          0 - Non-active Voice
- *                         -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Process(VadInst *vad_inst,
-                                WebRtc_Word16 fs,
-                                WebRtc_Word16 *speech_frame,
-                                WebRtc_Word16 frame_length);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_VAD_WEBRTC_VAD_H_
diff --git a/src/common_audio/vad/main/source/Android.mk b/src/common_audio/vad/main/source/Android.mk
deleted file mode 100644
index fba6ca1..0000000
--- a/src/common_audio/vad/main/source/Android.mk
+++ /dev/null
@@ -1,61 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_vad
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := webrtc_vad.c \
-    vad_const.c \
-    vad_core.c \
-    vad_filterbank.c \
-    vad_gmm.c \
-    vad_sp.c
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=  
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
-    '-DNO_HEAPCHECKER' \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' 
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../.. \
-    $(LOCAL_PATH)/../interface \
-    $(LOCAL_PATH)/../../../signal_processing_library/main/interface 
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS := 
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libdl \
-    libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
-LOCAL_LDLIBS += -ldl -lpthread
-endif
-
-ifneq ($(TARGET_SIMULATOR),true)
-LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/common_audio/vad/main/source/vad.gyp b/src/common_audio/vad/main/source/vad.gyp
deleted file mode 100644
index 754b684..0000000
--- a/src/common_audio/vad/main/source/vad.gyp
+++ /dev/null
@@ -1,51 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../../../common_settings.gypi', # Common settings
-  ],
-  'targets': [
-    {
-      'target_name': 'vad',
-      'type': '<(library)',
-      'dependencies': [
-        '../../../signal_processing_library/main/source/spl.gyp:spl',
-      ],
-      'include_dirs': [
-        '../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-        ],
-      },
-      'sources': [
-        '../interface/webrtc_vad.h',
-        'webrtc_vad.c',
-        'vad_const.c',
-        'vad_const.h',
-        'vad_defines.h',
-        'vad_core.c',
-        'vad_core.h',
-        'vad_filterbank.c',
-        'vad_filterbank.h',
-        'vad_gmm.c',
-        'vad_gmm.h',
-        'vad_sp.c',
-        'vad_sp.h',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/common_audio/vad/main/source/vad_const.c b/src/common_audio/vad/main/source/vad_const.c
deleted file mode 100644
index 47b6a4b..0000000
--- a/src/common_audio/vad/main/source/vad_const.c
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file includes the constant values used internally in VAD.
- */
-
-#include "vad_const.h"
-
-// Spectrum Weighting
-const WebRtc_Word16 kSpectrumWeight[6] = {6, 8, 10, 12, 14, 16};
-
-const WebRtc_Word16 kCompVar = 22005;
-
-// Constant 160*log10(2) in Q9
-const WebRtc_Word16 kLogConst = 24660;
-
-// Constant log2(exp(1)) in Q12
-const WebRtc_Word16 kLog10Const = 5909;
-
-// Q15
-const WebRtc_Word16 kNoiseUpdateConst = 655;
-const WebRtc_Word16 kSpeechUpdateConst = 6554;
-
-// Q8
-const WebRtc_Word16 kBackEta = 154;
-
-// Coefficients used by WebRtcVad_HpOutput, Q14
-const WebRtc_Word16 kHpZeroCoefs[3] = {6631, -13262, 6631};
-const WebRtc_Word16 kHpPoleCoefs[3] = {16384, -7756, 5620};
-
-// Allpass filter coefficients, upper and lower, in Q15
-// Upper: 0.64, Lower: 0.17
-const WebRtc_Word16 kAllPassCoefsQ15[2] = {20972, 5571};
-const WebRtc_Word16 kAllPassCoefsQ13[2] = {5243, 1392}; // Q13
-
-// Minimum difference between the two models, Q5
-const WebRtc_Word16 kMinimumDifference[6] = {544, 544, 576, 576, 576, 576};
-
-// Upper limit of mean value for speech model, Q7
-const WebRtc_Word16 kMaximumSpeech[6] = {11392, 11392, 11520, 11520, 11520, 11520};
-
-// Minimum value for mean value
-const WebRtc_Word16 kMinimumMean[2] = {640, 768};
-
-// Upper limit of mean value for noise model, Q7
-const WebRtc_Word16 kMaximumNoise[6] = {9216, 9088, 8960, 8832, 8704, 8576};
-
-// Adjustment for division with two in WebRtcVad_SplitFilter
-const WebRtc_Word16 kOffsetVector[6] = {368, 368, 272, 176, 176, 176};
-
-// Start values for the Gaussian models, Q7
-// Weights for the two Gaussians for the six channels (noise)
-const WebRtc_Word16 kNoiseDataWeights[12] = {34, 62, 72, 66, 53, 25, 94, 66, 56, 62, 75, 103};
-
-// Weights for the two Gaussians for the six channels (speech)
-const WebRtc_Word16 kSpeechDataWeights[12] = {48, 82, 45, 87, 50, 47, 80, 46, 83, 41, 78, 81};
-
-// Means for the two Gaussians for the six channels (noise)
-const WebRtc_Word16 kNoiseDataMeans[12] = {6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863,
-        7820, 7266, 5020, 4362};
-
-// Means for the two Gaussians for the six channels (speech)
-const WebRtc_Word16 kSpeechDataMeans[12] = {8306, 10085, 10078, 11823, 11843, 6309, 9473,
-        9571, 10879, 7581, 8180, 7483};
-
-// Stds for the two Gaussians for the six channels (noise)
-const WebRtc_Word16 kNoiseDataStds[12] = {378, 1064, 493, 582, 688, 593, 474, 697, 475, 688,
-        421, 455};
-
-// Stds for the two Gaussians for the six channels (speech)
-const WebRtc_Word16 kSpeechDataStds[12] = {555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540,
-        1079, 850};
diff --git a/src/common_audio/vad/main/source/vad_const.h b/src/common_audio/vad/main/source/vad_const.h
deleted file mode 100644
index 8980437..0000000
--- a/src/common_audio/vad/main/source/vad_const.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the declarations of the internally used constants.
- */
-
-#ifndef WEBRTC_VAD_CONST_H_
-#define WEBRTC_VAD_CONST_H_
-
-#include "typedefs.h"
-
-// TODO(ajm): give these internal-linkage by moving to the appropriate file
-// where possible, and otherwise tag with WebRtcVad_.
-
-// Spectrum Weighting
-extern const WebRtc_Word16 kSpectrumWeight[];
-extern const WebRtc_Word16 kCompVar;
-// Logarithm constant
-extern const WebRtc_Word16 kLogConst;
-extern const WebRtc_Word16 kLog10Const;
-// Q15
-extern const WebRtc_Word16 kNoiseUpdateConst;
-extern const WebRtc_Word16 kSpeechUpdateConst;
-// Q8
-extern const WebRtc_Word16 kBackEta;
-// Coefficients used by WebRtcVad_HpOutput, Q14
-extern const WebRtc_Word16 kHpZeroCoefs[];
-extern const WebRtc_Word16 kHpPoleCoefs[];
-// Allpass filter coefficients, upper and lower, in Q15 resp. Q13
-extern const WebRtc_Word16 kAllPassCoefsQ15[];
-extern const WebRtc_Word16 kAllPassCoefsQ13[];
-// Minimum difference between the two models, Q5
-extern const WebRtc_Word16 kMinimumDifference[];
-// Maximum value when updating the speech model, Q7
-extern const WebRtc_Word16 kMaximumSpeech[];
-// Minimum value for mean value
-extern const WebRtc_Word16 kMinimumMean[];
-// Upper limit of mean value for noise model, Q7
-extern const WebRtc_Word16 kMaximumNoise[];
-// Adjustment for division with two in WebRtcVad_SplitFilter
-extern const WebRtc_Word16 kOffsetVector[];
-// Start values for the Gaussian models, Q7
-extern const WebRtc_Word16 kNoiseDataWeights[];
-extern const WebRtc_Word16 kSpeechDataWeights[];
-extern const WebRtc_Word16 kNoiseDataMeans[];
-extern const WebRtc_Word16 kSpeechDataMeans[];
-extern const WebRtc_Word16 kNoiseDataStds[];
-extern const WebRtc_Word16 kSpeechDataStds[];
-
-#endif // WEBRTC_VAD_CONST_H_
diff --git a/src/common_audio/vad/main/source/vad_core.c b/src/common_audio/vad/main/source/vad_core.c
deleted file mode 100644
index e882999..0000000
--- a/src/common_audio/vad/main/source/vad_core.c
+++ /dev/null
@@ -1,685 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the core functionality in VAD.
- * For function description, see vad_core.h.
- */
-
-#include "vad_core.h"
-#include "vad_const.h"
-#include "vad_defines.h"
-#include "vad_filterbank.h"
-#include "vad_gmm.h"
-#include "vad_sp.h"
-#include "signal_processing_library.h"
-
-static const int kInitCheck = 42;
-
-// Initialize VAD
-int WebRtcVad_InitCore(VadInstT *inst, short mode)
-{
-    int i;
-
-    // Initialization of struct
-    inst->vad = 1;
-    inst->frame_counter = 0;
-    inst->over_hang = 0;
-    inst->num_of_speech = 0;
-
-    // Initialization of downsampling filter state
-    inst->downsampling_filter_states[0] = 0;
-    inst->downsampling_filter_states[1] = 0;
-    inst->downsampling_filter_states[2] = 0;
-    inst->downsampling_filter_states[3] = 0;
-
-    // Read initial PDF parameters
-    for (i = 0; i < NUM_TABLE_VALUES; i++)
-    {
-        inst->noise_means[i] = kNoiseDataMeans[i];
-        inst->speech_means[i] = kSpeechDataMeans[i];
-        inst->noise_stds[i] = kNoiseDataStds[i];
-        inst->speech_stds[i] = kSpeechDataStds[i];
-    }
-
-    // Index and Minimum value vectors are initialized
-    for (i = 0; i < 16 * NUM_CHANNELS; i++)
-    {
-        inst->low_value_vector[i] = 10000;
-        inst->index_vector[i] = 0;
-    }
-
-    for (i = 0; i < 5; i++)
-    {
-        inst->upper_state[i] = 0;
-        inst->lower_state[i] = 0;
-    }
-
-    for (i = 0; i < 4; i++)
-    {
-        inst->hp_filter_state[i] = 0;
-    }
-
-    // Init mean value memory, for FindMin function
-    inst->mean_value[0] = 1600;
-    inst->mean_value[1] = 1600;
-    inst->mean_value[2] = 1600;
-    inst->mean_value[3] = 1600;
-    inst->mean_value[4] = 1600;
-    inst->mean_value[5] = 1600;
-
-    if (mode == 0)
-    {
-        // Quality mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_Q; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_Q; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_Q; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_Q; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_Q; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_Q; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_Q;
-        inst->individual[1] = INDIVIDUAL_20MS_Q;
-        inst->individual[2] = INDIVIDUAL_30MS_Q;
-
-        inst->total[0] = TOTAL_10MS_Q;
-        inst->total[1] = TOTAL_20MS_Q;
-        inst->total[2] = TOTAL_30MS_Q;
-    } else if (mode == 1)
-    {
-        // Low bitrate mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_LBR; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_LBR; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_LBR; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_LBR;
-        inst->individual[1] = INDIVIDUAL_20MS_LBR;
-        inst->individual[2] = INDIVIDUAL_30MS_LBR;
-
-        inst->total[0] = TOTAL_10MS_LBR;
-        inst->total[1] = TOTAL_20MS_LBR;
-        inst->total[2] = TOTAL_30MS_LBR;
-    } else if (mode == 2)
-    {
-        // Aggressive mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_AGG; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_AGG; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_AGG; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_AGG;
-        inst->individual[1] = INDIVIDUAL_20MS_AGG;
-        inst->individual[2] = INDIVIDUAL_30MS_AGG;
-
-        inst->total[0] = TOTAL_10MS_AGG;
-        inst->total[1] = TOTAL_20MS_AGG;
-        inst->total[2] = TOTAL_30MS_AGG;
-    } else
-    {
-        // Very aggressive mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_VAG; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_VAG; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_VAG; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_VAG;
-        inst->individual[1] = INDIVIDUAL_20MS_VAG;
-        inst->individual[2] = INDIVIDUAL_30MS_VAG;
-
-        inst->total[0] = TOTAL_10MS_VAG;
-        inst->total[1] = TOTAL_20MS_VAG;
-        inst->total[2] = TOTAL_30MS_VAG;
-    }
-
-    inst->init_flag = kInitCheck;
-
-    return 0;
-}
-
-// Set aggressiveness mode
-int WebRtcVad_set_mode_core(VadInstT *inst, short mode)
-{
-
-    if (mode == 0)
-    {
-        // Quality mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_Q; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_Q; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_Q; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_Q; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_Q; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_Q; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_Q;
-        inst->individual[1] = INDIVIDUAL_20MS_Q;
-        inst->individual[2] = INDIVIDUAL_30MS_Q;
-
-        inst->total[0] = TOTAL_10MS_Q;
-        inst->total[1] = TOTAL_20MS_Q;
-        inst->total[2] = TOTAL_30MS_Q;
-    } else if (mode == 1)
-    {
-        // Low bitrate mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_LBR; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_LBR; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_LBR; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_LBR; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_LBR;
-        inst->individual[1] = INDIVIDUAL_20MS_LBR;
-        inst->individual[2] = INDIVIDUAL_30MS_LBR;
-
-        inst->total[0] = TOTAL_10MS_LBR;
-        inst->total[1] = TOTAL_20MS_LBR;
-        inst->total[2] = TOTAL_30MS_LBR;
-    } else if (mode == 2)
-    {
-        // Aggressive mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_AGG; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_AGG; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_AGG; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_AGG; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_AGG;
-        inst->individual[1] = INDIVIDUAL_20MS_AGG;
-        inst->individual[2] = INDIVIDUAL_30MS_AGG;
-
-        inst->total[0] = TOTAL_10MS_AGG;
-        inst->total[1] = TOTAL_20MS_AGG;
-        inst->total[2] = TOTAL_30MS_AGG;
-    } else if (mode == 3)
-    {
-        // Very aggressive mode
-        inst->over_hang_max_1[0] = OHMAX1_10MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_1[1] = OHMAX1_20MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_1[2] = OHMAX1_30MS_VAG; // Overhang short speech burst
-        inst->over_hang_max_2[0] = OHMAX2_10MS_VAG; // Overhang long speech burst
-        inst->over_hang_max_2[1] = OHMAX2_20MS_VAG; // Overhang long speech burst
-        inst->over_hang_max_2[2] = OHMAX2_30MS_VAG; // Overhang long speech burst
-
-        inst->individual[0] = INDIVIDUAL_10MS_VAG;
-        inst->individual[1] = INDIVIDUAL_20MS_VAG;
-        inst->individual[2] = INDIVIDUAL_30MS_VAG;
-
-        inst->total[0] = TOTAL_10MS_VAG;
-        inst->total[1] = TOTAL_20MS_VAG;
-        inst->total[2] = TOTAL_30MS_VAG;
-    } else
-    {
-        return -1;
-    }
-
-    return 0;
-}
-
-// Calculate VAD decision by first extracting feature values and then calculate
-// probability for both speech and background noise.
-
-WebRtc_Word16 WebRtcVad_CalcVad32khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
-                                     int frame_length)
-{
-    WebRtc_Word16 len, vad;
-    WebRtc_Word16 speechWB[480]; // Downsampled speech frame: 960 samples (30ms in SWB)
-    WebRtc_Word16 speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
-
-
-    // Downsample signal 32->16->8 before doing VAD
-    WebRtcVad_Downsampling(speech_frame, speechWB, &(inst->downsampling_filter_states[2]),
-                           frame_length);
-    len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
-
-    WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states, len);
-    len = WEBRTC_SPL_RSHIFT_W16(len, 1);
-
-    // Do VAD on an 8 kHz signal
-    vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
-
-    return vad;
-}
-
-WebRtc_Word16 WebRtcVad_CalcVad16khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
-                                     int frame_length)
-{
-    WebRtc_Word16 len, vad;
-    WebRtc_Word16 speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
-
-    // Wideband: Downsample signal before doing VAD
-    WebRtcVad_Downsampling(speech_frame, speechNB, inst->downsampling_filter_states,
-                           frame_length);
-
-    len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
-    vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
-
-    return vad;
-}
-
-WebRtc_Word16 WebRtcVad_CalcVad8khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
-                                    int frame_length)
-{
-    WebRtc_Word16 feature_vector[NUM_CHANNELS], total_power;
-
-    // Get power in the bands
-    total_power = WebRtcVad_get_features(inst, speech_frame, frame_length, feature_vector);
-
-    // Make a VAD
-    inst->vad = WebRtcVad_GmmProbability(inst, feature_vector, total_power, frame_length);
-
-    return inst->vad;
-}
-
-// Calculate probability for both speech and background noise, and perform a
-// hypothesis-test.
-WebRtc_Word16 WebRtcVad_GmmProbability(VadInstT *inst, WebRtc_Word16 *feature_vector,
-                                       WebRtc_Word16 total_power, int frame_length)
-{
-    int n, k;
-    WebRtc_Word16 backval;
-    WebRtc_Word16 h0, h1;
-    WebRtc_Word16 ratvec, xval;
-    WebRtc_Word16 vadflag;
-    WebRtc_Word16 shifts0, shifts1;
-    WebRtc_Word16 tmp16, tmp16_1, tmp16_2;
-    WebRtc_Word16 diff, nr, pos;
-    WebRtc_Word16 nmk, nmk2, nmk3, smk, smk2, nsk, ssk;
-    WebRtc_Word16 delt, ndelt;
-    WebRtc_Word16 maxspe, maxmu;
-    WebRtc_Word16 deltaN[NUM_TABLE_VALUES], deltaS[NUM_TABLE_VALUES];
-    WebRtc_Word16 ngprvec[NUM_TABLE_VALUES], sgprvec[NUM_TABLE_VALUES];
-    WebRtc_Word32 h0test, h1test;
-    WebRtc_Word32 tmp32_1, tmp32_2;
-    WebRtc_Word32 dotVal;
-    WebRtc_Word32 nmid, smid;
-    WebRtc_Word32 probn[NUM_MODELS], probs[NUM_MODELS];
-    WebRtc_Word16 *nmean1ptr, *nmean2ptr, *smean1ptr, *smean2ptr, *nstd1ptr, *nstd2ptr,
-            *sstd1ptr, *sstd2ptr;
-    WebRtc_Word16 overhead1, overhead2, individualTest, totalTest;
-
-    // Set the thresholds to different values based on frame length
-    if (frame_length == 80)
-    {
-        // 80 input samples
-        overhead1 = inst->over_hang_max_1[0];
-        overhead2 = inst->over_hang_max_2[0];
-        individualTest = inst->individual[0];
-        totalTest = inst->total[0];
-    } else if (frame_length == 160)
-    {
-        // 160 input samples
-        overhead1 = inst->over_hang_max_1[1];
-        overhead2 = inst->over_hang_max_2[1];
-        individualTest = inst->individual[1];
-        totalTest = inst->total[1];
-    } else
-    {
-        // 240 input samples
-        overhead1 = inst->over_hang_max_1[2];
-        overhead2 = inst->over_hang_max_2[2];
-        individualTest = inst->individual[2];
-        totalTest = inst->total[2];
-    }
-
-    if (total_power > MIN_ENERGY)
-    { // If signal present at all
-
-        // Set pointers to the gaussian parameters
-        nmean1ptr = &inst->noise_means[0];
-        nmean2ptr = &inst->noise_means[NUM_CHANNELS];
-        smean1ptr = &inst->speech_means[0];
-        smean2ptr = &inst->speech_means[NUM_CHANNELS];
-        nstd1ptr = &inst->noise_stds[0];
-        nstd2ptr = &inst->noise_stds[NUM_CHANNELS];
-        sstd1ptr = &inst->speech_stds[0];
-        sstd2ptr = &inst->speech_stds[NUM_CHANNELS];
-
-        vadflag = 0;
-        dotVal = 0;
-        for (n = 0; n < NUM_CHANNELS; n++)
-        { // For all channels
-
-            pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
-            xval = feature_vector[n];
-
-            // Probability for Noise, Q7 * Q20 = Q27
-            tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean1ptr++, *nstd1ptr++,
-                                                    &deltaN[pos]);
-            probn[0] = (WebRtc_Word32)(kNoiseDataWeights[n] * tmp32_1);
-            tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean2ptr++, *nstd2ptr++,
-                                                    &deltaN[pos + 1]);
-            probn[1] = (WebRtc_Word32)(kNoiseDataWeights[n + NUM_CHANNELS] * tmp32_1);
-            h0test = probn[0] + probn[1]; // Q27
-            h0 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(h0test, 12); // Q15
-
-            // Probability for Speech
-            tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean1ptr++, *sstd1ptr++,
-                                                    &deltaS[pos]);
-            probs[0] = (WebRtc_Word32)(kSpeechDataWeights[n] * tmp32_1);
-            tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean2ptr++, *sstd2ptr++,
-                                                    &deltaS[pos + 1]);
-            probs[1] = (WebRtc_Word32)(kSpeechDataWeights[n + NUM_CHANNELS] * tmp32_1);
-            h1test = probs[0] + probs[1]; // Q27
-            h1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(h1test, 12); // Q15
-
-            // Get likelihood ratio. Approximate log2(H1/H0) with shifts0 - shifts1
-            shifts0 = WebRtcSpl_NormW32(h0test);
-            shifts1 = WebRtcSpl_NormW32(h1test);
-
-            if ((h0test > 0) && (h1test > 0))
-            {
-                ratvec = shifts0 - shifts1;
-            } else if (h1test > 0)
-            {
-                ratvec = 31 - shifts1;
-            } else if (h0test > 0)
-            {
-                ratvec = shifts0 - 31;
-            } else
-            {
-                ratvec = 0;
-            }
-
-            // VAD decision with spectrum weighting
-            dotVal += WEBRTC_SPL_MUL_16_16(ratvec, kSpectrumWeight[n]);
-
-            // Individual channel test
-            if ((ratvec << 2) > individualTest)
-            {
-                vadflag = 1;
-            }
-
-            // Probabilities used when updating model
-            if (h0 > 0)
-            {
-                tmp32_1 = probn[0] & 0xFFFFF000; // Q27
-                tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2); // Q29
-                ngprvec[pos] = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, h0);
-                ngprvec[pos + 1] = 16384 - ngprvec[pos];
-            } else
-            {
-                ngprvec[pos] = 16384;
-                ngprvec[pos + 1] = 0;
-            }
-
-            // Probabilities used when updating model
-            if (h1 > 0)
-            {
-                tmp32_1 = probs[0] & 0xFFFFF000;
-                tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2);
-                sgprvec[pos] = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, h1);
-                sgprvec[pos + 1] = 16384 - sgprvec[pos];
-            } else
-            {
-                sgprvec[pos] = 0;
-                sgprvec[pos + 1] = 0;
-            }
-        }
-
-        // Overall test
-        if (dotVal >= totalTest)
-        {
-            vadflag |= 1;
-        }
-
-        // Set pointers to the means and standard deviations.
-        nmean1ptr = &inst->noise_means[0];
-        smean1ptr = &inst->speech_means[0];
-        nstd1ptr = &inst->noise_stds[0];
-        sstd1ptr = &inst->speech_stds[0];
-
-        maxspe = 12800;
-
-        // Update the model's parameters
-        for (n = 0; n < NUM_CHANNELS; n++)
-        {
-
-            pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
-
-            // Get min value in past which is used for long term correction
-            backval = WebRtcVad_FindMinimum(inst, feature_vector[n], n); // Q4
-
-            // Compute the "global" mean, that is the sum of the two means weighted
-            nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr); // Q7 * Q7
-            nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+NUM_CHANNELS],
-                    *(nmean1ptr+NUM_CHANNELS));
-            tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 6); // Q8
-
-            for (k = 0; k < NUM_MODELS; k++)
-            {
-
-                nr = pos + k;
-
-                nmean2ptr = nmean1ptr + k * NUM_CHANNELS;
-                smean2ptr = smean1ptr + k * NUM_CHANNELS;
-                nstd2ptr = nstd1ptr + k * NUM_CHANNELS;
-                sstd2ptr = sstd1ptr + k * NUM_CHANNELS;
-                nmk = *nmean2ptr;
-                smk = *smean2ptr;
-                nsk = *nstd2ptr;
-                ssk = *sstd2ptr;
-
-                // Update noise mean vector if the frame consists of noise only
-                nmk2 = nmk;
-                if (!vadflag)
-                {
-                    // deltaN = (x-mu)/sigma^2
-                    // ngprvec[k] = probn[k]/(probn[0] + probn[1])
-
-                    delt = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ngprvec[nr],
-                            deltaN[nr], 11); // Q14*Q11
-                    nmk2 = nmk + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delt,
-                            kNoiseUpdateConst,
-                            22); // Q7+(Q14*Q15>>22)
-                }
-
-                // Long term correction of the noise mean
-                ndelt = WEBRTC_SPL_LSHIFT_W16(backval, 4);
-                ndelt -= tmp16_1; // Q8 - Q8
-                nmk3 = nmk2 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ndelt,
-                        kBackEta,
-                        9); // Q7+(Q8*Q8)>>9
-
-                // Control that the noise mean does not drift to much
-                tmp16 = WEBRTC_SPL_LSHIFT_W16(k+5, 7);
-                if (nmk3 < tmp16)
-                    nmk3 = tmp16;
-                tmp16 = WEBRTC_SPL_LSHIFT_W16(72+k-n, 7);
-                if (nmk3 > tmp16)
-                    nmk3 = tmp16;
-                *nmean2ptr = nmk3;
-
-                if (vadflag)
-                {
-                    // Update speech mean vector:
-                    // deltaS = (x-mu)/sigma^2
-                    // sgprvec[k] = probn[k]/(probn[0] + probn[1])
-
-                    delt = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sgprvec[nr],
-                            deltaS[nr],
-                            11); // (Q14*Q11)>>11=Q14
-                    tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delt,
-                            kSpeechUpdateConst,
-                            21) + 1;
-                    smk2 = smk + (tmp16 >> 1); // Q7 + (Q14 * Q15 >> 22)
-
-                    // Control that the speech mean does not drift to much
-                    maxmu = maxspe + 640;
-                    if (smk2 < kMinimumMean[k])
-                        smk2 = kMinimumMean[k];
-                    if (smk2 > maxmu)
-                        smk2 = maxmu;
-
-                    *smean2ptr = smk2;
-
-                    // (Q7>>3) = Q4
-                    tmp16 = WEBRTC_SPL_RSHIFT_W16((smk + 4), 3);
-
-                    tmp16 = feature_vector[n] - tmp16; // Q4
-                    tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaS[nr], tmp16, 3);
-                    tmp32_2 = tmp32_1 - (WebRtc_Word32)4096; // Q12
-                    tmp16 = WEBRTC_SPL_RSHIFT_W16((sgprvec[nr]), 2);
-                    tmp32_1 = (WebRtc_Word32)(tmp16 * tmp32_2);// (Q15>>3)*(Q14>>2)=Q12*Q12=Q24
-
-                    tmp32_2 = WEBRTC_SPL_RSHIFT_W32(tmp32_1, 4); // Q20
-
-                    // 0.1 * Q20 / Q7 = Q13
-                    if (tmp32_2 > 0)
-                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, ssk * 10);
-                    else
-                    {
-                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(-tmp32_2, ssk * 10);
-                        tmp16 = -tmp16;
-                    }
-                    // divide by 4 giving an update factor of 0.025
-                    tmp16 += 128; // Rounding
-                    ssk += WEBRTC_SPL_RSHIFT_W16(tmp16, 8);
-                    // Division with 8 plus Q7
-                    if (ssk < MIN_STD)
-                        ssk = MIN_STD;
-                    *sstd2ptr = ssk;
-                } else
-                {
-                    // Update GMM variance vectors
-                    // deltaN * (feature_vector[n] - nmk) - 1, Q11 * Q4
-                    tmp16 = feature_vector[n] - WEBRTC_SPL_RSHIFT_W16(nmk, 3);
-
-                    // (Q15>>3) * (Q14>>2) = Q12 * Q12 = Q24
-                    tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaN[nr], tmp16, 3) - 4096;
-                    tmp16 = WEBRTC_SPL_RSHIFT_W16((ngprvec[nr]+2), 2);
-                    tmp32_2 = (WebRtc_Word32)(tmp16 * tmp32_1);
-                    tmp32_1 = WEBRTC_SPL_RSHIFT_W32(tmp32_2, 14);
-                    // Q20  * approx 0.001 (2^-10=0.0009766)
-
-                    // Q20 / Q7 = Q13
-                    tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_1, nsk);
-                    if (tmp32_1 > 0)
-                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_1, nsk);
-                    else
-                    {
-                        tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(-tmp32_1, nsk);
-                        tmp16 = -tmp16;
-                    }
-                    tmp16 += 32; // Rounding
-                    nsk += WEBRTC_SPL_RSHIFT_W16(tmp16, 6);
-
-                    if (nsk < MIN_STD)
-                        nsk = MIN_STD;
-
-                    *nstd2ptr = nsk;
-                }
-            }
-
-            // Separate models if they are too close - nmid in Q14
-            nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr);
-            nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+NUM_CHANNELS], *nmean2ptr);
-
-            // smid in Q14
-            smid = WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n], *smean1ptr);
-            smid += WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n+NUM_CHANNELS], *smean2ptr);
-
-            // diff = "global" speech mean - "global" noise mean
-            diff = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(smid, 9);
-            tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 9);
-            diff -= tmp16;
-
-            if (diff < kMinimumDifference[n])
-            {
-
-                tmp16 = kMinimumDifference[n] - diff; // Q5
-
-                // tmp16_1 = ~0.8 * (kMinimumDifference - diff) in Q7
-                // tmp16_2 = ~0.2 * (kMinimumDifference - diff) in Q7
-                tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(13, tmp16, 2);
-                tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(3, tmp16, 2);
-
-                // First Gauss, speech model
-                tmp16 = tmp16_1 + *smean1ptr;
-                *smean1ptr = tmp16;
-                smid = WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n]);
-
-                // Second Gauss, speech model
-                tmp16 = tmp16_1 + *smean2ptr;
-                *smean2ptr = tmp16;
-                smid += WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n+NUM_CHANNELS]);
-
-                // First Gauss, noise model
-                tmp16 = *nmean1ptr - tmp16_2;
-                *nmean1ptr = tmp16;
-
-                nmid = WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n]);
-
-                // Second Gauss, noise model
-                tmp16 = *nmean2ptr - tmp16_2;
-                *nmean2ptr = tmp16;
-                nmid += WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n+NUM_CHANNELS]);
-            }
-
-            // Control that the speech & noise means do not drift to much
-            maxspe = kMaximumSpeech[n];
-            tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(smid, 7);
-            if (tmp16_2 > maxspe)
-            { // Upper limit of speech model
-                tmp16_2 -= maxspe;
-
-                *smean1ptr -= tmp16_2;
-                *smean2ptr -= tmp16_2;
-            }
-
-            tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 7);
-            if (tmp16_2 > kMaximumNoise[n])
-            {
-                tmp16_2 -= kMaximumNoise[n];
-
-                *nmean1ptr -= tmp16_2;
-                *nmean2ptr -= tmp16_2;
-            }
-
-            *nmean1ptr++;
-            *smean1ptr++;
-            *nstd1ptr++;
-            *sstd1ptr++;
-        }
-        inst->frame_counter++;
-    } else
-    {
-        vadflag = 0;
-    }
-
-    // Hangover smoothing
-    if (!vadflag)
-    {
-        if (inst->over_hang > 0)
-        {
-            vadflag = 2 + inst->over_hang;
-            inst->over_hang = inst->over_hang - 1;
-        }
-        inst->num_of_speech = 0;
-    } else
-    {
-        inst->num_of_speech = inst->num_of_speech + 1;
-        if (inst->num_of_speech > NSP_MAX)
-        {
-            inst->num_of_speech = NSP_MAX;
-            inst->over_hang = overhead2;
-        } else
-            inst->over_hang = overhead1;
-    }
-    return vadflag;
-}
diff --git a/src/common_audio/vad/main/source/vad_core.h b/src/common_audio/vad/main/source/vad_core.h
deleted file mode 100644
index 544caf5..0000000
--- a/src/common_audio/vad/main/source/vad_core.h
+++ /dev/null
@@ -1,132 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the descriptions of the core VAD calls.
- */
-
-#ifndef WEBRTC_VAD_CORE_H_
-#define WEBRTC_VAD_CORE_H_
-
-#include "typedefs.h"
-#include "vad_defines.h"
-
-typedef struct VadInstT_
-{
-
-    WebRtc_Word16 vad;
-    WebRtc_Word32 downsampling_filter_states[4];
-    WebRtc_Word16 noise_means[NUM_TABLE_VALUES];
-    WebRtc_Word16 speech_means[NUM_TABLE_VALUES];
-    WebRtc_Word16 noise_stds[NUM_TABLE_VALUES];
-    WebRtc_Word16 speech_stds[NUM_TABLE_VALUES];
-    WebRtc_Word32 frame_counter;
-    WebRtc_Word16 over_hang; // Over Hang
-    WebRtc_Word16 num_of_speech;
-    WebRtc_Word16 index_vector[16 * NUM_CHANNELS];
-    WebRtc_Word16 low_value_vector[16 * NUM_CHANNELS];
-    WebRtc_Word16 mean_value[NUM_CHANNELS];
-    WebRtc_Word16 upper_state[5];
-    WebRtc_Word16 lower_state[5];
-    WebRtc_Word16 hp_filter_state[4];
-    WebRtc_Word16 over_hang_max_1[3];
-    WebRtc_Word16 over_hang_max_2[3];
-    WebRtc_Word16 individual[3];
-    WebRtc_Word16 total[3];
-
-    short init_flag;
-
-} VadInstT;
-
-/****************************************************************************
- * WebRtcVad_InitCore(...)
- *
- * This function initializes a VAD instance
- *
- * Input:
- *      - inst      : Instance that should be initialized
- *      - mode      : Aggressiveness degree
- *                    0 (High quality) - 3 (Highly aggressive)
- *
- * Output:
- *      - inst      : Initialized instance
- *
- * Return value     :  0 - Ok
- *                    -1 - Error
- */
-int WebRtcVad_InitCore(VadInstT* inst, short mode);
-
-/****************************************************************************
- * WebRtcVad_set_mode_core(...)
- *
- * This function changes the VAD settings
- *
- * Input:
- *      - inst      : VAD instance
- *      - mode      : Aggressiveness degree
- *                    0 (High quality) - 3 (Highly aggressive)
- *
- * Output:
- *      - inst      : Changed  instance
- *
- * Return value     :  0 - Ok
- *                    -1 - Error
- */
-
-int WebRtcVad_set_mode_core(VadInstT* inst, short mode);
-
-/****************************************************************************
- * WebRtcVad_CalcVad32khz(...) 
- * WebRtcVad_CalcVad16khz(...) 
- * WebRtcVad_CalcVad8khz(...) 
- *
- * Calculate probability for active speech and make VAD decision.
- *
- * Input:
- *      - inst          : Instance that should be initialized
- *      - speech_frame  : Input speech frame
- *      - frame_length  : Number of input samples
- *
- * Output:
- *      - inst          : Updated filter states etc.
- *
- * Return value         : VAD decision
- *                        0 - No active speech
- *                        1-6 - Active speech
- */
-WebRtc_Word16 WebRtcVad_CalcVad32khz(VadInstT* inst, WebRtc_Word16* speech_frame,
-                                     int frame_length);
-WebRtc_Word16 WebRtcVad_CalcVad16khz(VadInstT* inst, WebRtc_Word16* speech_frame,
-                                     int frame_length);
-WebRtc_Word16 WebRtcVad_CalcVad8khz(VadInstT* inst, WebRtc_Word16* speech_frame,
-                                    int frame_length);
-
-/****************************************************************************
- * WebRtcVad_GmmProbability(...)
- *
- * This function calculates the probabilities for background noise and
- * speech using Gaussian Mixture Models. A hypothesis-test is performed to decide
- * which type of signal is most probable.
- *
- * Input:
- *      - inst              : Pointer to VAD instance
- *      - feature_vector    : Feature vector = log10(energy in frequency band)
- *      - total_power       : Total power in frame.
- *      - frame_length      : Number of input samples
- *
- * Output:
- *      VAD decision        : 0 - noise, 1 - speech
- *    
- */
-WebRtc_Word16 WebRtcVad_GmmProbability(VadInstT* inst, WebRtc_Word16* feature_vector,
-                                       WebRtc_Word16 total_power, int frame_length);
-
-#endif // WEBRTC_VAD_CORE_H_
diff --git a/src/common_audio/vad/main/source/vad_defines.h b/src/common_audio/vad/main/source/vad_defines.h
deleted file mode 100644
index b33af2e..0000000
--- a/src/common_audio/vad/main/source/vad_defines.h
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the macros used in VAD.
- */
-
-#ifndef WEBRTC_VAD_DEFINES_H_
-#define WEBRTC_VAD_DEFINES_H_
-
-#define NUM_CHANNELS        6   // Eight frequency bands
-#define NUM_MODELS          2   // Number of Gaussian models
-#define NUM_TABLE_VALUES    NUM_CHANNELS * NUM_MODELS
-
-#define MIN_ENERGY          10
-#define ALPHA1              6553    // 0.2 in Q15
-#define ALPHA2              32439   // 0.99 in Q15
-#define NSP_MAX             6       // Maximum number of VAD=1 frames in a row counted
-#define MIN_STD             384     // Minimum standard deviation
-// Mode 0, Quality thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_Q   24
-#define INDIVIDUAL_20MS_Q   21      // (log10(2)*66)<<2 ~=16
-#define INDIVIDUAL_30MS_Q   24
-
-#define TOTAL_10MS_Q        57
-#define TOTAL_20MS_Q        48
-#define TOTAL_30MS_Q        57
-
-#define OHMAX1_10MS_Q       8  // Max Overhang 1
-#define OHMAX2_10MS_Q       14 // Max Overhang 2
-#define OHMAX1_20MS_Q       4  // Max Overhang 1
-#define OHMAX2_20MS_Q       7  // Max Overhang 2
-#define OHMAX1_30MS_Q       3
-#define OHMAX2_30MS_Q       5
-
-// Mode 1, Low bitrate thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_LBR 37
-#define INDIVIDUAL_20MS_LBR 32
-#define INDIVIDUAL_30MS_LBR 37
-
-#define TOTAL_10MS_LBR      100
-#define TOTAL_20MS_LBR      80
-#define TOTAL_30MS_LBR      100
-
-#define OHMAX1_10MS_LBR     8  // Max Overhang 1
-#define OHMAX2_10MS_LBR     14 // Max Overhang 2
-#define OHMAX1_20MS_LBR     4
-#define OHMAX2_20MS_LBR     7
-
-#define OHMAX1_30MS_LBR     3
-#define OHMAX2_30MS_LBR     5
-
-// Mode 2, Very aggressive thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_AGG 82
-#define INDIVIDUAL_20MS_AGG 78
-#define INDIVIDUAL_30MS_AGG 82
-
-#define TOTAL_10MS_AGG      285 //580
-#define TOTAL_20MS_AGG      260
-#define TOTAL_30MS_AGG      285
-
-#define OHMAX1_10MS_AGG     6  // Max Overhang 1
-#define OHMAX2_10MS_AGG     9  // Max Overhang 2
-#define OHMAX1_20MS_AGG     3
-#define OHMAX2_20MS_AGG     5
-
-#define OHMAX1_30MS_AGG     2
-#define OHMAX2_30MS_AGG     3
-
-// Mode 3, Super aggressive thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_VAG 94
-#define INDIVIDUAL_20MS_VAG 94
-#define INDIVIDUAL_30MS_VAG 94
-
-#define TOTAL_10MS_VAG      1100 //1700
-#define TOTAL_20MS_VAG      1050
-#define TOTAL_30MS_VAG      1100
-
-#define OHMAX1_10MS_VAG     6  // Max Overhang 1
-#define OHMAX2_10MS_VAG     9  // Max Overhang 2
-#define OHMAX1_20MS_VAG     3
-#define OHMAX2_20MS_VAG     5
-
-#define OHMAX1_30MS_VAG     2
-#define OHMAX2_30MS_VAG     3
-
-#endif // WEBRTC_VAD_DEFINES_H_
diff --git a/src/common_audio/vad/main/source/vad_filterbank.c b/src/common_audio/vad/main/source/vad_filterbank.c
deleted file mode 100644
index 11392c9..0000000
--- a/src/common_audio/vad/main/source/vad_filterbank.c
+++ /dev/null
@@ -1,267 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the internal filterbank associated functions.
- * For function description, see vad_filterbank.h.
- */
-
-#include "vad_filterbank.h"
-#include "vad_defines.h"
-#include "vad_const.h"
-#include "signal_processing_library.h"
-
-void WebRtcVad_HpOutput(WebRtc_Word16 *in_vector,
-                        WebRtc_Word16 in_vector_length,
-                        WebRtc_Word16 *out_vector,
-                        WebRtc_Word16 *filter_state)
-{
-    WebRtc_Word16 i, *pi, *outPtr;
-    WebRtc_Word32 tmpW32;
-
-    pi = &in_vector[0];
-    outPtr = &out_vector[0];
-
-    // The sum of the absolute values of the impulse response:
-    // The zero/pole-filter has a max amplification of a single sample of: 1.4546
-    // Impulse response: 0.4047 -0.6179 -0.0266  0.1993  0.1035  -0.0194
-    // The all-zero section has a max amplification of a single sample of: 1.6189
-    // Impulse response: 0.4047 -0.8094  0.4047  0       0        0
-    // The all-pole section has a max amplification of a single sample of: 1.9931
-    // Impulse response: 1.0000  0.4734 -0.1189 -0.2187 -0.0627   0.04532
-
-    for (i = 0; i < in_vector_length; i++)
-    {
-        // all-zero section (filter coefficients in Q14)
-        tmpW32 = (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[0], (*pi));
-        tmpW32 += (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[1], filter_state[0]);
-        tmpW32 += (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[2], filter_state[1]); // Q14
-        filter_state[1] = filter_state[0];
-        filter_state[0] = *pi++;
-
-        // all-pole section
-        tmpW32 -= (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[1], filter_state[2]); // Q14
-        tmpW32 -= (WebRtc_Word32)WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[2], filter_state[3]);
-        filter_state[3] = filter_state[2];
-        filter_state[2] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32 (tmpW32, 14);
-        *outPtr++ = filter_state[2];
-    }
-}
-
-void WebRtcVad_Allpass(WebRtc_Word16 *in_vector,
-                       WebRtc_Word16 *out_vector,
-                       WebRtc_Word16 filter_coefficients,
-                       int vector_length,
-                       WebRtc_Word16 *filter_state)
-{
-    // The filter can only cause overflow (in the w16 output variable)
-    // if more than 4 consecutive input numbers are of maximum value and
-    // has the the same sign as the impulse responses first taps.
-    // First 6 taps of the impulse response: 0.6399 0.5905 -0.3779
-    // 0.2418 -0.1547 0.0990
-
-    int n;
-    WebRtc_Word16 tmp16;
-    WebRtc_Word32 tmp32, in32, state32;
-
-    state32 = WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)(*filter_state)), 16); // Q31
-
-    for (n = 0; n < vector_length; n++)
-    {
-
-        tmp32 = state32 + WEBRTC_SPL_MUL_16_16(filter_coefficients, (*in_vector));
-        tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
-        *out_vector++ = tmp16;
-        in32 = WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)(*in_vector)), 14);
-        state32 = in32 - WEBRTC_SPL_MUL_16_16(filter_coefficients, tmp16);
-        state32 = WEBRTC_SPL_LSHIFT_W32(state32, 1);
-        in_vector += 2;
-    }
-
-    *filter_state = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(state32, 16);
-}
-
-void WebRtcVad_SplitFilter(WebRtc_Word16 *in_vector,
-                           WebRtc_Word16 *out_vector_hp,
-                           WebRtc_Word16 *out_vector_lp,
-                           WebRtc_Word16 *upper_state,
-                           WebRtc_Word16 *lower_state,
-                           int in_vector_length)
-{
-    WebRtc_Word16 tmpOut;
-    int k, halflen;
-
-    // Downsampling by 2 and get two branches
-    halflen = WEBRTC_SPL_RSHIFT_W16(in_vector_length, 1);
-
-    // All-pass filtering upper branch
-    WebRtcVad_Allpass(&in_vector[0], out_vector_hp, kAllPassCoefsQ15[0], halflen, upper_state);
-
-    // All-pass filtering lower branch
-    WebRtcVad_Allpass(&in_vector[1], out_vector_lp, kAllPassCoefsQ15[1], halflen, lower_state);
-
-    // Make LP and HP signals
-    for (k = 0; k < halflen; k++)
-    {
-        tmpOut = *out_vector_hp;
-        *out_vector_hp++ -= *out_vector_lp;
-        *out_vector_lp++ += tmpOut;
-    }
-}
-
-WebRtc_Word16 WebRtcVad_get_features(VadInstT *inst,
-                                     WebRtc_Word16 *in_vector,
-                                     int frame_size,
-                                     WebRtc_Word16 *out_vector)
-{
-    int curlen, filtno;
-    WebRtc_Word16 vecHP1[120], vecLP1[120];
-    WebRtc_Word16 vecHP2[60], vecLP2[60];
-    WebRtc_Word16 *ptin;
-    WebRtc_Word16 *hptout, *lptout;
-    WebRtc_Word16 power = 0;
-
-    // Split at 2000 Hz and downsample
-    filtno = 0;
-    ptin = in_vector;
-    hptout = vecHP1;
-    lptout = vecLP1;
-    curlen = frame_size;
-    WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
-                  &inst->lower_state[filtno], curlen);
-
-    // Split at 3000 Hz and downsample
-    filtno = 1;
-    ptin = vecHP1;
-    hptout = vecHP2;
-    lptout = vecLP2;
-    curlen = WEBRTC_SPL_RSHIFT_W16(frame_size, 1);
-
-    WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
-                  &inst->lower_state[filtno], curlen);
-
-    // Energy in 3000 Hz - 4000 Hz
-    curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
-    WebRtcVad_LogOfEnergy(vecHP2, &out_vector[5], &power, kOffsetVector[5], curlen);
-
-    // Energy in 2000 Hz - 3000 Hz
-    WebRtcVad_LogOfEnergy(vecLP2, &out_vector[4], &power, kOffsetVector[4], curlen);
-
-    // Split at 1000 Hz and downsample
-    filtno = 2;
-    ptin = vecLP1;
-    hptout = vecHP2;
-    lptout = vecLP2;
-    curlen = WEBRTC_SPL_RSHIFT_W16(frame_size, 1);
-    WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
-                  &inst->lower_state[filtno], curlen);
-
-    // Energy in 1000 Hz - 2000 Hz
-    curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
-    WebRtcVad_LogOfEnergy(vecHP2, &out_vector[3], &power, kOffsetVector[3], curlen);
-
-    // Split at 500 Hz
-    filtno = 3;
-    ptin = vecLP2;
-    hptout = vecHP1;
-    lptout = vecLP1;
-
-    WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
-                  &inst->lower_state[filtno], curlen);
-
-    // Energy in 500 Hz - 1000 Hz
-    curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
-    WebRtcVad_LogOfEnergy(vecHP1, &out_vector[2], &power, kOffsetVector[2], curlen);
-    // Split at 250 Hz
-    filtno = 4;
-    ptin = vecLP1;
-    hptout = vecHP2;
-    lptout = vecLP2;
-
-    WebRtcVad_SplitFilter(ptin, hptout, lptout, &inst->upper_state[filtno],
-                  &inst->lower_state[filtno], curlen);
-
-    // Energy in 250 Hz - 500 Hz
-    curlen = WEBRTC_SPL_RSHIFT_W16(curlen, 1);
-    WebRtcVad_LogOfEnergy(vecHP2, &out_vector[1], &power, kOffsetVector[1], curlen);
-
-    // Remove DC and LFs
-    WebRtcVad_HpOutput(vecLP2, curlen, vecHP1, inst->hp_filter_state);
-
-    // Power in 80 Hz - 250 Hz
-    WebRtcVad_LogOfEnergy(vecHP1, &out_vector[0], &power, kOffsetVector[0], curlen);
-
-    return power;
-}
-
-void WebRtcVad_LogOfEnergy(WebRtc_Word16 *vector,
-                           WebRtc_Word16 *enerlogval,
-                           WebRtc_Word16 *power,
-                           WebRtc_Word16 offset,
-                           int vector_length)
-{
-    WebRtc_Word16 enerSum = 0;
-    WebRtc_Word16 zeros, frac, log2;
-    WebRtc_Word32 energy;
-
-    int shfts = 0, shfts2;
-
-    energy = WebRtcSpl_Energy(vector, vector_length, &shfts);
-
-    if (energy > 0)
-    {
-
-        shfts2 = 16 - WebRtcSpl_NormW32(energy);
-        shfts += shfts2;
-        // "shfts" is the total number of right shifts that has been done to enerSum.
-        enerSum = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(energy, -shfts2);
-
-        // Find:
-        // 160*log10(enerSum*2^shfts) = 160*log10(2)*log2(enerSum*2^shfts) =
-        // 160*log10(2)*(log2(enerSum) + log2(2^shfts)) =
-        // 160*log10(2)*(log2(enerSum) + shfts)
-
-        zeros = WebRtcSpl_NormU32(enerSum);
-        frac = (WebRtc_Word16)(((WebRtc_UWord32)((WebRtc_Word32)(enerSum) << zeros)
-                & 0x7FFFFFFF) >> 21);
-        log2 = (WebRtc_Word16)(((31 - zeros) << 10) + frac);
-
-        *enerlogval = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(kLogConst, log2, 19)
-                + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(shfts, kLogConst, 9);
-
-        if (*enerlogval < 0)
-        {
-            *enerlogval = 0;
-        }
-    } else
-    {
-        *enerlogval = 0;
-        shfts = -15;
-        enerSum = 0;
-    }
-
-    *enerlogval += offset;
-
-    // Total power in frame
-    if (*power <= MIN_ENERGY)
-    {
-        if (shfts > 0)
-        {
-            *power += MIN_ENERGY + 1;
-        } else if (WEBRTC_SPL_SHIFT_W16(enerSum, shfts) > MIN_ENERGY)
-        {
-            *power += MIN_ENERGY + 1;
-        } else
-        {
-            *power += WEBRTC_SPL_SHIFT_W16(enerSum, shfts);
-        }
-    }
-}
diff --git a/src/common_audio/vad/main/source/vad_filterbank.h b/src/common_audio/vad/main/source/vad_filterbank.h
deleted file mode 100644
index a5507ea..0000000
--- a/src/common_audio/vad/main/source/vad_filterbank.h
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the description of the internal VAD call
- * WebRtcVad_GaussianProbability.
- */
-
-#ifndef WEBRTC_VAD_FILTERBANK_H_
-#define WEBRTC_VAD_FILTERBANK_H_
-
-#include "vad_core.h"
-
-/****************************************************************************
- * WebRtcVad_HpOutput(...)
- *
- * This function removes DC from the lowest frequency band
- *
- * Input:
- *      - in_vector         : Samples in the frequency interval 0 - 250 Hz
- *      - in_vector_length  : Length of input and output vector
- *      - filter_state      : Current state of the filter
- *
- * Output:
- *      - out_vector        : Samples in the frequency interval 80 - 250 Hz
- *      - filter_state      : Updated state of the filter
- *
- */
-void WebRtcVad_HpOutput(WebRtc_Word16* in_vector,
-                        WebRtc_Word16  in_vector_length,
-                        WebRtc_Word16* out_vector,
-                        WebRtc_Word16* filter_state);
-
-/****************************************************************************
- * WebRtcVad_Allpass(...)
- *
- * This function is used when before splitting a speech file into 
- * different frequency bands
- *
- * Note! Do NOT let the arrays in_vector and out_vector correspond to the same address.
- *
- * Input:
- *      - in_vector             : (Q0)
- *      - filter_coefficients   : (Q15)
- *      - vector_length         : Length of input and output vector
- *      - filter_state          : Current state of the filter (Q(-1))
- *
- * Output:
- *      - out_vector            : Output speech signal (Q(-1))
- *      - filter_state          : Updated state of the filter (Q(-1))
- *
- */
-void WebRtcVad_Allpass(WebRtc_Word16* in_vector,
-                       WebRtc_Word16* outw16,
-                       WebRtc_Word16 filter_coefficients,
-                       int vector_length,
-                       WebRtc_Word16* filter_state);
-
-/****************************************************************************
- * WebRtcVad_SplitFilter(...)
- *
- * This function is used when before splitting a speech file into 
- * different frequency bands
- *
- * Input:
- *      - in_vector         : Input signal to be split into two frequency bands.
- *      - upper_state       : Current state of the upper filter
- *      - lower_state       : Current state of the lower filter
- *      - in_vector_length  : Length of input vector
- *
- * Output:
- *      - out_vector_hp     : Upper half of the spectrum
- *      - out_vector_lp     : Lower half of the spectrum
- *      - upper_state       : Updated state of the upper filter
- *      - lower_state       : Updated state of the lower filter
- *
- */
-void WebRtcVad_SplitFilter(WebRtc_Word16* in_vector,
-                           WebRtc_Word16* out_vector_hp,
-                           WebRtc_Word16* out_vector_lp,
-                           WebRtc_Word16* upper_state,
-                           WebRtc_Word16* lower_state,
-                           int in_vector_length);
-
-/****************************************************************************
- * WebRtcVad_get_features(...)
- *
- * This function is used to get the logarithm of the power of each of the 
- * 6 frequency bands used by the VAD:
- *        80 Hz - 250 Hz
- *        250 Hz - 500 Hz
- *        500 Hz - 1000 Hz
- *        1000 Hz - 2000 Hz
- *        2000 Hz - 3000 Hz
- *        3000 Hz - 4000 Hz 
- *
- * Input:
- *      - inst        : Pointer to VAD instance
- *      - in_vector   : Input speech signal
- *      - frame_size  : Frame size, in number of samples
- *
- * Output:
- *      - out_vector  : 10*log10(power in each freq. band), Q4
- *    
- * Return: total power in the signal (NOTE! This value is not exact since it
- *         is only used in a comparison.
- */
-WebRtc_Word16 WebRtcVad_get_features(VadInstT* inst,
-                                     WebRtc_Word16* in_vector,
-                                     int frame_size,
-                                     WebRtc_Word16* out_vector);
-
-/****************************************************************************
- * WebRtcVad_LogOfEnergy(...)
- *
- * This function is used to get the logarithm of the power of one frequency band.
- *
- * Input:
- *      - vector            : Input speech samples for one frequency band
- *      - offset            : Offset value for the current frequency band
- *      - vector_length     : Length of input vector
- *
- * Output:
- *      - enerlogval        : 10*log10(energy);
- *      - power             : Update total power in speech frame. NOTE! This value
- *                            is not exact since it is only used in a comparison.
- *     
- */
-void WebRtcVad_LogOfEnergy(WebRtc_Word16* vector,
-                           WebRtc_Word16* enerlogval,
-                           WebRtc_Word16* power,
-                           WebRtc_Word16 offset,
-                           int vector_length);
-
-#endif // WEBRTC_VAD_FILTERBANK_H_
diff --git a/src/common_audio/vad/main/source/vad_gmm.c b/src/common_audio/vad/main/source/vad_gmm.c
deleted file mode 100644
index 23d12fb..0000000
--- a/src/common_audio/vad/main/source/vad_gmm.c
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the internal VAD call
- * WebRtcVad_GaussianProbability. For function description, see vad_gmm.h.
- */
-
-#include "vad_gmm.h"
-#include "signal_processing_library.h"
-#include "vad_const.h"
-
-WebRtc_Word32 WebRtcVad_GaussianProbability(WebRtc_Word16 in_sample,
-                                            WebRtc_Word16 mean,
-                                            WebRtc_Word16 std,
-                                            WebRtc_Word16 *delta)
-{
-    WebRtc_Word16 tmp16, tmpDiv, tmpDiv2, expVal, tmp16_1, tmp16_2;
-    WebRtc_Word32 tmp32, y32;
-
-    // Calculate tmpDiv=1/std, in Q10
-    tmp32 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W16(std,1) + (WebRtc_Word32)131072; // 1 in Q17
-    tmpDiv = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32, std); // Q17/Q7 = Q10
-
-    // Calculate tmpDiv2=1/std^2, in Q14
-    tmp16 = WEBRTC_SPL_RSHIFT_W16(tmpDiv, 2); // From Q10 to Q8
-    tmpDiv2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 2); // (Q8 * Q8)>>2 = Q14
-
-    tmp16 = WEBRTC_SPL_LSHIFT_W16(in_sample, 3); // Q7
-    tmp16 = tmp16 - mean; // Q7 - Q7 = Q7
-
-    // To be used later, when updating noise/speech model
-    // delta = (x-m)/std^2, in Q11
-    *delta = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmpDiv2, tmp16, 10); //(Q14*Q7)>>10 = Q11
-
-    // Calculate tmp32=(x-m)^2/(2*std^2), in Q10
-    tmp32 = (WebRtc_Word32)WEBRTC_SPL_MUL_16_16_RSFT(*delta, tmp16, 9); // One shift for /2
-
-    // Calculate expVal ~= exp(-(x-m)^2/(2*std^2)) ~= exp2(-log2(exp(1))*tmp32)
-    if (tmp32 < kCompVar)
-    {
-        // Calculate tmp16 = log2(exp(1))*tmp32 , in Q10
-        tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)tmp32,
-                                                         kLog10Const, 12);
-        tmp16 = -tmp16;
-        tmp16_2 = (WebRtc_Word16)(0x0400 | (tmp16 & 0x03FF));
-        tmp16_1 = (WebRtc_Word16)(tmp16 ^ 0xFFFF);
-        tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(tmp16_1, 10);
-        tmp16 += 1;
-        // Calculate expVal=log2(-tmp32), in Q10
-        expVal = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)tmp16_2, tmp16);
-
-    } else
-    {
-        expVal = 0;
-    }
-
-    // Calculate y32=(1/std)*exp(-(x-m)^2/(2*std^2)), in Q20
-    y32 = WEBRTC_SPL_MUL_16_16(tmpDiv, expVal); // Q10 * Q10 = Q20
-
-    return y32; // Q20
-}
diff --git a/src/common_audio/vad/main/source/vad_gmm.h b/src/common_audio/vad/main/source/vad_gmm.h
deleted file mode 100644
index e0747fb..0000000
--- a/src/common_audio/vad/main/source/vad_gmm.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the description of the internal VAD call
- * WebRtcVad_GaussianProbability.
- */
-
-#ifndef WEBRTC_VAD_GMM_H_
-#define WEBRTC_VAD_GMM_H_
-
-#include "typedefs.h"
-
-/****************************************************************************
- * WebRtcVad_GaussianProbability(...)
- *
- * This function calculates the probability for the value 'in_sample', given that in_sample
- * comes from a normal distribution with mean 'mean' and standard deviation 'std'.
- *
- * Input:
- *      - in_sample     : Input sample in Q4
- *      - mean          : mean value in the statistical model, Q7
- *      - std           : standard deviation, Q7
- *
- * Output:
- *
- *      - delta         : Value used when updating the model, Q11
- *
- * Return:
- *      - out           : out = 1/std * exp(-(x-m)^2/(2*std^2));
- *                        Probability for x.
- *
- */
-WebRtc_Word32 WebRtcVad_GaussianProbability(WebRtc_Word16 in_sample,
-                                            WebRtc_Word16 mean,
-                                            WebRtc_Word16 std,
-                                            WebRtc_Word16 *delta);
-
-#endif // WEBRTC_VAD_GMM_H_
diff --git a/src/common_audio/vad/main/source/vad_sp.c b/src/common_audio/vad/main/source/vad_sp.c
deleted file mode 100644
index f347ab5..0000000
--- a/src/common_audio/vad/main/source/vad_sp.c
+++ /dev/null
@@ -1,231 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the VAD internal calls for Downsampling and
- * FindMinimum.
- * For function call descriptions; See vad_sp.h.
- */
-
-#include "vad_sp.h"
-#include "vad_defines.h"
-#include "vad_const.h"
-#include "signal_processing_library.h"
-
-// Downsampling filter based on the splitting filter and the allpass functions
-// in vad_filterbank.c
-void WebRtcVad_Downsampling(WebRtc_Word16* signal_in,
-                            WebRtc_Word16* signal_out,
-                            WebRtc_Word32* filter_state,
-                            int inlen)
-{
-    WebRtc_Word16 tmp16_1, tmp16_2;
-    WebRtc_Word32 tmp32_1, tmp32_2;
-    int n, halflen;
-
-    // Downsampling by 2 and get two branches
-    halflen = WEBRTC_SPL_RSHIFT_W16(inlen, 1);
-
-    tmp32_1 = filter_state[0];
-    tmp32_2 = filter_state[1];
-
-    // Filter coefficients in Q13, filter state in Q0
-    for (n = 0; n < halflen; n++)
-    {
-        // All-pass filtering upper branch
-        tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32_1, 1)
-                + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[0]),
-                                                           *signal_in, 14);
-        *signal_out = tmp16_1;
-        tmp32_1 = (WebRtc_Word32)(*signal_in++)
-                - (WebRtc_Word32)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[0]), tmp16_1, 12);
-
-        // All-pass filtering lower branch
-        tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32_2, 1)
-                + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[1]),
-                                                           *signal_in, 14);
-        *signal_out++ += tmp16_2;
-        tmp32_2 = (WebRtc_Word32)(*signal_in++)
-                - (WebRtc_Word32)WEBRTC_SPL_MUL_16_16_RSFT((kAllPassCoefsQ13[1]), tmp16_2, 12);
-    }
-    filter_state[0] = tmp32_1;
-    filter_state[1] = tmp32_2;
-}
-
-WebRtc_Word16 WebRtcVad_FindMinimum(VadInstT* inst,
-                                    WebRtc_Word16 x,
-                                    int n)
-{
-    int i, j, k, II = -1, offset;
-    WebRtc_Word16 meanV, alpha;
-    WebRtc_Word32 tmp32, tmp32_1;
-    WebRtc_Word16 *valptr, *idxptr, *p1, *p2, *p3;
-
-    // Offset to beginning of the 16 minimum values in memory
-    offset = WEBRTC_SPL_LSHIFT_W16(n, 4);
-
-    // Pointer to memory for the 16 minimum values and the age of each value
-    idxptr = &inst->index_vector[offset];
-    valptr = &inst->low_value_vector[offset];
-
-    // Each value in low_value_vector is getting 1 loop older.
-    // Update age of each value in indexVal, and remove old values.
-    for (i = 0; i < 16; i++)
-    {
-        p3 = idxptr + i;
-        if (*p3 != 100)
-        {
-            *p3 += 1;
-        } else
-        {
-            p1 = valptr + i + 1;
-            p2 = p3 + 1;
-            for (j = i; j < 16; j++)
-            {
-                *(valptr + j) = *p1++;
-                *(idxptr + j) = *p2++;
-            }
-            *(idxptr + 15) = 101;
-            *(valptr + 15) = 10000;
-        }
-    }
-
-    // Check if x smaller than any of the values in low_value_vector.
-    // If so, find position.
-    if (x < *(valptr + 7))
-    {
-        if (x < *(valptr + 3))
-        {
-            if (x < *(valptr + 1))
-            {
-                if (x < *valptr)
-                {
-                    II = 0;
-                } else
-                {
-                    II = 1;
-                }
-            } else if (x < *(valptr + 2))
-            {
-                II = 2;
-            } else
-            {
-                II = 3;
-            }
-        } else if (x < *(valptr + 5))
-        {
-            if (x < *(valptr + 4))
-            {
-                II = 4;
-            } else
-            {
-                II = 5;
-            }
-        } else if (x < *(valptr + 6))
-        {
-            II = 6;
-        } else
-        {
-            II = 7;
-        }
-    } else if (x < *(valptr + 15))
-    {
-        if (x < *(valptr + 11))
-        {
-            if (x < *(valptr + 9))
-            {
-                if (x < *(valptr + 8))
-                {
-                    II = 8;
-                } else
-                {
-                    II = 9;
-                }
-            } else if (x < *(valptr + 10))
-            {
-                II = 10;
-            } else
-            {
-                II = 11;
-            }
-        } else if (x < *(valptr + 13))
-        {
-            if (x < *(valptr + 12))
-            {
-                II = 12;
-            } else
-            {
-                II = 13;
-            }
-        } else if (x < *(valptr + 14))
-        {
-            II = 14;
-        } else
-        {
-            II = 15;
-        }
-    }
-
-    // Put new min value on right position and shift bigger values up
-    if (II > -1)
-    {
-        for (i = 15; i > II; i--)
-        {
-            k = i - 1;
-            *(valptr + i) = *(valptr + k);
-            *(idxptr + i) = *(idxptr + k);
-        }
-        *(valptr + II) = x;
-        *(idxptr + II) = 1;
-    }
-
-    meanV = 0;
-    if ((inst->frame_counter) > 4)
-    {
-        j = 5;
-    } else
-    {
-        j = inst->frame_counter;
-    }
-
-    if (j > 2)
-    {
-        meanV = *(valptr + 2);
-    } else if (j > 0)
-    {
-        meanV = *valptr;
-    } else
-    {
-        meanV = 1600;
-    }
-
-    if (inst->frame_counter > 0)
-    {
-        if (meanV < inst->mean_value[n])
-        {
-            alpha = (WebRtc_Word16)ALPHA1; // 0.2 in Q15
-        } else
-        {
-            alpha = (WebRtc_Word16)ALPHA2; // 0.99 in Q15
-        }
-    } else
-    {
-        alpha = 0;
-    }
-
-    tmp32 = WEBRTC_SPL_MUL_16_16((alpha+1), inst->mean_value[n]);
-    tmp32_1 = WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_WORD16_MAX - alpha, meanV);
-    tmp32 += tmp32_1;
-    tmp32 += 16384;
-    inst->mean_value[n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 15);
-
-    return inst->mean_value[n];
-}
diff --git a/src/common_audio/vad/main/source/vad_sp.h b/src/common_audio/vad/main/source/vad_sp.h
deleted file mode 100644
index ae15c11..0000000
--- a/src/common_audio/vad/main/source/vad_sp.h
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the VAD internal calls for Downsampling and FindMinimum.
- * Specific function calls are given below.
- */
-
-#ifndef WEBRTC_VAD_SP_H_
-#define WEBRTC_VAD_SP_H_
-
-#include "vad_core.h"
-
-/****************************************************************************
- * WebRtcVad_Downsampling(...)
- *
- * Downsamples the signal a factor 2, eg. 32->16 or 16->8
- *
- * Input:
- *      - signal_in     : Input signal
- *      - in_length     : Length of input signal in samples
- *
- * Input & Output:
- *      - filter_state  : Filter state for first all-pass filters
- *
- * Output:
- *      - signal_out    : Downsampled signal (of length len/2)
- */
-void WebRtcVad_Downsampling(WebRtc_Word16* signal_in,
-                            WebRtc_Word16* signal_out,
-                            WebRtc_Word32* filter_state,
-                            int in_length);
-
-/****************************************************************************
- * WebRtcVad_FindMinimum(...)
- *
- * Find the five lowest values of x in 100 frames long window. Return a mean
- * value of these five values.
- *
- * Input:
- *      - feature_value : Feature value
- *      - channel       : Channel number
- *
- * Input & Output:
- *      - inst          : State information
- *
- * Output:
- *      return value    : Weighted minimum value for a moving window.
- */
-WebRtc_Word16 WebRtcVad_FindMinimum(VadInstT* inst, WebRtc_Word16 feature_value, int channel);
-
-#endif // WEBRTC_VAD_SP_H_
diff --git a/src/common_audio/vad/main/source/webrtc_vad.c b/src/common_audio/vad/main/source/webrtc_vad.c
deleted file mode 100644
index dcfbda1..0000000
--- a/src/common_audio/vad/main/source/webrtc_vad.c
+++ /dev/null
@@ -1,197 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the VAD API calls. For a specific function call description,
- * see webrtc_vad.h
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "webrtc_vad.h"
-#include "vad_core.h"
-
-static const int kInitCheck = 42;
-
-WebRtc_Word16 WebRtcVad_get_version(char *version, size_t size_bytes)
-{
-    const char my_version[] = "VAD 1.2.0";
-
-    if (version == NULL)
-    {
-        return -1;
-    }
-
-    if (size_bytes < sizeof(my_version))
-    {
-        return -1;
-    }
-
-    memcpy(version, my_version, sizeof(my_version));
-    return 0;
-}
-
-WebRtc_Word16 WebRtcVad_AssignSize(int *size_in_bytes)
-{
-    *size_in_bytes = sizeof(VadInstT) * 2 / sizeof(WebRtc_Word16);
-    return 0;
-}
-
-WebRtc_Word16 WebRtcVad_Assign(VadInst **vad_inst, void *vad_inst_addr)
-{
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    if (vad_inst_addr != NULL)
-    {
-        *vad_inst = (VadInst*)vad_inst_addr;
-        return 0;
-    } else
-    {
-        return -1;
-    }
-}
-
-WebRtc_Word16 WebRtcVad_Create(VadInst **vad_inst)
-{
-
-    VadInstT *vad_ptr = NULL;
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    *vad_inst = NULL;
-
-    vad_ptr = (VadInstT *)malloc(sizeof(VadInstT));
-    *vad_inst = (VadInst *)vad_ptr;
-
-    if (vad_ptr == NULL)
-    {
-        return -1;
-    }
-
-    vad_ptr->init_flag = 0;
-
-    return 0;
-}
-
-WebRtc_Word16 WebRtcVad_Free(VadInst *vad_inst)
-{
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    free(vad_inst);
-    return 0;
-}
-
-WebRtc_Word16 WebRtcVad_Init(VadInst *vad_inst)
-{
-    short mode = 0; // Default high quality
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    return WebRtcVad_InitCore((VadInstT*)vad_inst, mode);
-}
-
-WebRtc_Word16 WebRtcVad_set_mode(VadInst *vad_inst, WebRtc_Word16 mode)
-{
-    VadInstT* vad_ptr;
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    vad_ptr = (VadInstT*)vad_inst;
-    if (vad_ptr->init_flag != kInitCheck)
-    {
-        return -1;
-    }
-
-    return WebRtcVad_set_mode_core((VadInstT*)vad_inst, mode);
-}
-
-WebRtc_Word16 WebRtcVad_Process(VadInst *vad_inst,
-                                WebRtc_Word16 fs,
-                                WebRtc_Word16 *speech_frame,
-                                WebRtc_Word16 frame_length)
-{
-    WebRtc_Word16 vad;
-    VadInstT* vad_ptr;
-
-    if (vad_inst == NULL)
-    {
-        return -1;
-    }
-
-    vad_ptr = (VadInstT*)vad_inst;
-    if (vad_ptr->init_flag != kInitCheck)
-    {
-        return -1;
-    }
-
-    if (speech_frame == NULL)
-    {
-        return -1;
-    }
-
-    if (fs == 32000)
-    {
-        if ((frame_length != 320) && (frame_length != 640) && (frame_length != 960))
-        {
-            return -1;
-        }
-        vad = WebRtcVad_CalcVad32khz((VadInstT*)vad_inst, speech_frame, frame_length);
-
-    } else if (fs == 16000)
-    {
-        if ((frame_length != 160) && (frame_length != 320) && (frame_length != 480))
-        {
-            return -1;
-        }
-        vad = WebRtcVad_CalcVad16khz((VadInstT*)vad_inst, speech_frame, frame_length);
-
-    } else if (fs == 8000)
-    {
-        if ((frame_length != 80) && (frame_length != 160) && (frame_length != 240))
-        {
-            return -1;
-        }
-        vad = WebRtcVad_CalcVad8khz((VadInstT*)vad_inst, speech_frame, frame_length);
-
-    } else
-    {
-        return -1; // Not a supported sampling frequency
-    }
-
-    if (vad > 0)
-    {
-        return 1;
-    } else if (vad == 0)
-    {
-        return 0;
-    } else
-    {
-        return -1;
-    }
-}
diff --git a/src/common_audio/vad/main/test/unit_test/unit_test.cc b/src/common_audio/vad/main/test/unit_test/unit_test.cc
deleted file mode 100644
index 8ac793e..0000000
--- a/src/common_audio/vad/main/test/unit_test/unit_test.cc
+++ /dev/null
@@ -1,123 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the implementation of the VAD unit tests.
- */
-
-#include <cstring>
-#include "unit_test.h"
-#include "webrtc_vad.h"
-
-
-class VadEnvironment : public ::testing::Environment {
- public:
-  virtual void SetUp() {
-  }
-
-  virtual void TearDown() {
-  }
-};
-
-VadTest::VadTest()
-{
-}
-
-void VadTest::SetUp() {
-}
-
-void VadTest::TearDown() {
-}
-
-TEST_F(VadTest, ApiTest) {
-    VadInst *vad_inst;
-    int i, j, k;
-    short zeros[960];
-    short speech[960];
-    char version[32];
-
-    // Valid test cases
-    int fs[3] = {8000, 16000, 32000};
-    int nMode[4] = {0, 1, 2, 3};
-    int framelen[3][3] = {{80, 160, 240},
-    {160, 320, 480}, {320, 640, 960}} ;
-    int vad_counter = 0;
-
-    memset(zeros, 0, sizeof(short) * 960);
-    memset(speech, 1, sizeof(short) * 960);
-    speech[13] = 1374;
-    speech[73] = -3747;
-
-
-
-    // WebRtcVad_get_version()
-    WebRtcVad_get_version(version);
-    //printf("API Test for %s\n", version);
-
-    // Null instance tests
-    EXPECT_EQ(-1, WebRtcVad_Create(NULL));
-    EXPECT_EQ(-1, WebRtcVad_Init(NULL));
-    EXPECT_EQ(-1, WebRtcVad_Assign(NULL, NULL));
-    EXPECT_EQ(-1, WebRtcVad_Free(NULL));
-    EXPECT_EQ(-1, WebRtcVad_set_mode(NULL, nMode[0]));
-    EXPECT_EQ(-1, WebRtcVad_Process(NULL, fs[0], speech,  framelen[0][0]));
-
-
-    EXPECT_EQ(WebRtcVad_Create(&vad_inst), 0);
-
-    // Not initialized tests
-    EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, fs[0], speech,  framelen[0][0]));
-    EXPECT_EQ(-1, WebRtcVad_set_mode(vad_inst, nMode[0]));
-
-    // WebRtcVad_Init() tests
-    EXPECT_EQ(WebRtcVad_Init(vad_inst), 0);
-
-    // WebRtcVad_set_mode() tests
-    EXPECT_EQ(-1, WebRtcVad_set_mode(vad_inst, -1));
-    EXPECT_EQ(-1, WebRtcVad_set_mode(vad_inst, 4));
-
-    for (i = 0; i < sizeof(nMode)/sizeof(nMode[0]); i++) {
-        EXPECT_EQ(WebRtcVad_set_mode(vad_inst, nMode[i]), 0);
-    }
-
-    // WebRtcVad_Process() tests
-    EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, fs[0], NULL,  framelen[0][0]));
-    EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, 12000, speech,  framelen[0][0]));
-    EXPECT_EQ(-1, WebRtcVad_Process(vad_inst, fs[0], speech,  framelen[1][1]));
-    EXPECT_EQ(WebRtcVad_Process(vad_inst, fs[0], zeros,  framelen[0][0]), 0);
-    for (i = 0; i < sizeof(fs)/sizeof(fs[0]); i++) {
-        for (j = 0; j < sizeof(framelen[0])/sizeof(framelen[0][0]); j++) {
-            for (k = 0; k < sizeof(nMode)/sizeof(nMode[0]); k++) {
-                EXPECT_EQ(WebRtcVad_set_mode(vad_inst, nMode[k]), 0);
-//                printf("%d\n", WebRtcVad_Process(vad_inst, fs[i], speech,  framelen[i][j]));
-                if (vad_counter < 9)
-                {
-                    EXPECT_EQ(WebRtcVad_Process(vad_inst, fs[i], speech,  framelen[i][j]), 1);
-                } else
-                {
-                    EXPECT_EQ(WebRtcVad_Process(vad_inst, fs[i], speech,  framelen[i][j]), 0);
-                }
-                vad_counter++;
-            }
-        }
-    }
-
-    EXPECT_EQ(0, WebRtcVad_Free(vad_inst));
-
-}
-
-int main(int argc, char** argv) {
-  ::testing::InitGoogleTest(&argc, argv);
-  VadEnvironment* env = new VadEnvironment;
-  ::testing::AddGlobalTestEnvironment(env);
-
-  return RUN_ALL_TESTS();
-}
diff --git a/src/common_audio/vad/main/test/unit_test/unit_test.h b/src/common_audio/vad/main/test/unit_test/unit_test.h
deleted file mode 100644
index 62dac11..0000000
--- a/src/common_audio/vad/main/test/unit_test/unit_test.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the declaration of the VAD unit test.
- */
-
-#ifndef WEBRTC_VAD_UNIT_TEST_H_
-#define WEBRTC_VAD_UNIT_TEST_H_
-
-#include <gtest/gtest.h>
-
-class VadTest : public ::testing::Test {
- protected:
-  VadTest();
-  virtual void SetUp();
-  virtual void TearDown();
-};
-
-#endif  // WEBRTC_VAD_UNIT_TEST_H_
diff --git a/src/common_types.h b/src/common_types.h
deleted file mode 100644
index 8b0b8a5..0000000
--- a/src/common_types.h
+++ /dev/null
@@ -1,595 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_COMMON_TYPES_H
-#define WEBRTC_COMMON_TYPES_H
-
-#include "typedefs.h"
-
-#ifdef WEBRTC_EXPORT
-    #define WEBRTC_DLLEXPORT _declspec(dllexport)
-#elif WEBRTC_DLL
-    #define WEBRTC_DLLEXPORT _declspec(dllimport)
-#else
-    #define WEBRTC_DLLEXPORT
-#endif
-
-#ifndef NULL
-    #define NULL 0
-#endif
-
-namespace webrtc {
-
-class InStream
-{
-public:
-    virtual int Read(void *buf,int len) = 0;
-    virtual int Rewind() {return -1;}
-    virtual ~InStream() {}
-protected:
-    InStream() {}
-};
-
-class OutStream
-{
-public:
-    virtual bool Write(const void *buf,int len) = 0;
-    virtual int Rewind() {return -1;}
-    virtual ~OutStream() {}
-protected:
-    OutStream() {}
-};
-
-enum TraceModule
-{
-    // not a module, triggered from the engine code
-    kTraceVoice              = 0x0001,
-    // not a module, triggered from the engine code
-    kTraceVideo              = 0x0002,
-    // not a module, triggered from the utility code
-    kTraceUtility            = 0x0003,
-    kTraceRtpRtcp            = 0x0004,
-    kTraceTransport          = 0x0005,
-    kTraceSrtp               = 0x0006,
-    kTraceAudioCoding        = 0x0007,
-    kTraceAudioMixerServer   = 0x0008,
-    kTraceAudioMixerClient   = 0x0009,
-    kTraceFile               = 0x000a,
-    kTraceAudioProcessing    = 0x000b,
-    kTraceVideoCoding        = 0x0010,
-    kTraceVideoMixer         = 0x0011,
-    kTraceAudioDevice        = 0x0012,
-    kTraceVideoRenderer      = 0x0014,
-    kTraceVideoCapture       = 0x0015,
-    kTraceVideoPreocessing   = 0x0016
-};
-
-enum TraceLevel
-{
-    kTraceNone               = 0x0000,    // no trace
-    kTraceStateInfo          = 0x0001,
-    kTraceWarning            = 0x0002,
-    kTraceError              = 0x0004,
-    kTraceCritical           = 0x0008,
-    kTraceApiCall            = 0x0010,
-    kTraceDefault            = 0x00ff,
-
-    kTraceModuleCall         = 0x0020,
-    kTraceMemory             = 0x0100,   // memory info
-    kTraceTimer              = 0x0200,   // timing info
-    kTraceStream             = 0x0400,   // "continuous" stream of data
-
-    // used for debug purposes
-    kTraceDebug              = 0x0800,  // debug
-    kTraceInfo               = 0x1000,  // debug info
-
-    kTraceAll                = 0xffff
-};
-
-// External Trace API
-class TraceCallback
-{
-public:
-    virtual void Print(const TraceLevel level,
-                       const char *traceString,
-                       const int length) = 0;
-protected:
-    virtual ~TraceCallback() {}
-    TraceCallback() {}
-};
-
-
-enum FileFormats
-{
-    kFileFormatWavFile        = 1,
-    kFileFormatCompressedFile = 2,
-    kFileFormatAviFile        = 3,
-    kFileFormatPreencodedFile = 4,
-    kFileFormatPcm16kHzFile   = 7,
-    kFileFormatPcm8kHzFile    = 8,
-    kFileFormatPcm32kHzFile   = 9
-};
-
-
-enum ProcessingTypes
-{
-    kPlaybackPerChannel = 0,
-    kPlaybackAllChannelsMixed,
-    kRecordingPerChannel,
-    kRecordingAllChannelsMixed
-};
-
-// Encryption enums
-enum CipherTypes
-{
-    kCipherNull               = 0,
-    kCipherAes128CounterMode  = 1
-};
-
-enum AuthenticationTypes
-{
-    kAuthNull       = 0,
-    kAuthHmacSha1   = 3
-};
-
-enum SecurityLevels
-{
-    kNoProtection                    = 0,
-    kEncryption                      = 1,
-    kAuthentication                  = 2,
-    kEncryptionAndAuthentication     = 3
-};
-
-class Encryption
-{
-public:
-    virtual void encrypt(
-        int channel_no,
-        unsigned char* in_data,
-        unsigned char* out_data,
-        int bytes_in,
-        int* bytes_out) = 0;
-
-    virtual void decrypt(
-        int channel_no,
-        unsigned char* in_data,
-        unsigned char* out_data,
-        int bytes_in,
-        int* bytes_out) = 0;
-
-    virtual void encrypt_rtcp(
-        int channel_no,
-        unsigned char* in_data,
-        unsigned char* out_data,
-        int bytes_in,
-        int* bytes_out) = 0;
-
-    virtual void decrypt_rtcp(
-        int channel_no,
-        unsigned char* in_data,
-        unsigned char* out_data,
-        int bytes_in,
-        int* bytes_out) = 0;
-
-protected:
-    virtual ~Encryption() {}
-    Encryption() {}
-};
-
-// External transport callback interface
-class Transport
-{
-public:
-    virtual int SendPacket(int channel, const void *data, int len) = 0;
-    virtual int SendRTCPPacket(int channel, const void *data, int len) = 0;
-
-protected:
-    virtual ~Transport() {}
-    Transport() {}
-};
-
-// ==================================================================
-// Voice specific types
-// ==================================================================
-
-// Each codec supported can be described by this structure.
-struct CodecInst
-{
-    int pltype;
-    char plname[32];
-    int plfreq;
-    int pacsize;
-    int channels;
-    int rate;
-};
-
-enum FrameType
-{
-    kFrameEmpty            = 0,
-    kAudioFrameSpeech      = 1,
-    kAudioFrameCN          = 2,
-    kVideoFrameKey         = 3,    // independent frame
-    kVideoFrameDelta       = 4,    // depends on the previus frame
-    kVideoFrameGolden      = 5,    // depends on a old known previus frame
-    kVideoFrameAltRef      = 6
-};
-
-// RTP
-enum {kRtpCsrcSize = 15}; // RFC 3550 page 13
-
-enum RTPDirections
-{
-    kRtpIncoming = 0,
-    kRtpOutgoing
-};
-
-enum PayloadFrequencies
-{
-    kFreq8000Hz = 8000,
-    kFreq16000Hz = 16000,
-    kFreq32000Hz = 32000
-};
-
-enum VadModes                 // degree of bandwidth reduction
-{
-    kVadConventional = 0,      // lowest reduction
-    kVadAggressiveLow,
-    kVadAggressiveMid,
-    kVadAggressiveHigh         // highest reduction
-};
-
-struct NetworkStatistics           // NETEQ statistics
-{
-    // current jitter buffer size in ms
-    WebRtc_UWord16 currentBufferSize;
-    // preferred (optimal) buffer size in ms
-    WebRtc_UWord16 preferredBufferSize;
-    // loss rate (network + late) in percent (in Q14)
-    WebRtc_UWord16 currentPacketLossRate;
-    // late loss rate in percent (in Q14)
-    WebRtc_UWord16 currentDiscardRate;
-    // fraction (of original stream) of synthesized speech inserted through
-    // expansion (in Q14)
-    WebRtc_UWord16 currentExpandRate;
-    // fraction of synthesized speech inserted through pre-emptive expansion
-    // (in Q14)
-    WebRtc_UWord16 currentPreemptiveRate;
-    // fraction of data removed through acceleration (in Q14)
-    WebRtc_UWord16 currentAccelerateRate;
-};
-
-struct JitterStatistics
-{
-    // smallest Jitter Buffer size during call in ms
-    WebRtc_UWord32 jbMinSize;
-    // largest Jitter Buffer size during call in ms
-    WebRtc_UWord32 jbMaxSize;
-    // the average JB size, measured over time - ms
-    WebRtc_UWord32 jbAvgSize;
-    // number of times the Jitter Buffer changed (using Accelerate or
-    // Pre-emptive Expand)
-    WebRtc_UWord32 jbChangeCount;
-    // amount (in ms) of audio data received late
-    WebRtc_UWord32 lateLossMs;
-    // milliseconds removed to reduce jitter buffer size
-    WebRtc_UWord32 accelerateMs;
-    // milliseconds discarded through buffer flushing
-    WebRtc_UWord32 flushedMs;
-    // milliseconds of generated silence
-    WebRtc_UWord32 generatedSilentMs;
-    // milliseconds of synthetic audio data (non-background noise)
-    WebRtc_UWord32 interpolatedVoiceMs;
-    // milliseconds of synthetic audio data (background noise level)
-    WebRtc_UWord32 interpolatedSilentMs;
-    // count of tiny expansions in output audio
-    WebRtc_UWord32 countExpandMoreThan120ms;
-    // count of small expansions in output audio
-    WebRtc_UWord32 countExpandMoreThan250ms;
-    // count of medium expansions in output audio
-    WebRtc_UWord32 countExpandMoreThan500ms;
-    // count of long expansions in output audio
-    WebRtc_UWord32 countExpandMoreThan2000ms;
-    // duration of longest audio drop-out
-    WebRtc_UWord32 longestExpandDurationMs;
-    // count of times we got small network outage (inter-arrival time in
-    // [500, 1000) ms)
-    WebRtc_UWord32 countIAT500ms;
-    // count of times we got medium network outage (inter-arrival time in
-    // [1000, 2000) ms)
-    WebRtc_UWord32 countIAT1000ms;
-    // count of times we got large network outage (inter-arrival time >=
-    // 2000 ms)
-    WebRtc_UWord32 countIAT2000ms;
-    // longest packet inter-arrival time in ms
-    WebRtc_UWord32 longestIATms;
-    // min time incoming Packet "waited" to be played
-    WebRtc_UWord32 minPacketDelayMs;
-    // max time incoming Packet "waited" to be played
-    WebRtc_UWord32 maxPacketDelayMs;
-    // avg time incoming Packet "waited" to be played
-    WebRtc_UWord32 avgPacketDelayMs;
-};
-
-typedef struct
-{
-    int min;              // minumum
-    int max;              // maximum
-    int average;          // average
-} StatVal;
-
-typedef struct           // All levels are reported in dBm0
-{
-    StatVal speech_rx;   // long-term speech levels on receiving side
-    StatVal speech_tx;   // long-term speech levels on transmitting side
-    StatVal noise_rx;    // long-term noise/silence levels on receiving side
-    StatVal noise_tx;    // long-term noise/silence levels on transmitting side
-} LevelStatistics;
-
-typedef struct        // All levels are reported in dB
-{
-    StatVal erl;      // Echo Return Loss
-    StatVal erle;     // Echo Return Loss Enhancement
-    StatVal rerl;     // RERL = ERL + ERLE
-    // Echo suppression inside EC at the point just before its NLP
-    StatVal a_nlp;
-} EchoStatistics;
-
-enum TelephoneEventDetectionMethods
-{
-    kInBand = 0,
-    kOutOfBand = 1,
-    kInAndOutOfBand = 2
-};
-
-enum NsModes    // type of Noise Suppression
-{
-    kNsUnchanged = 0,   // previously set mode
-    kNsDefault,         // platform default
-    kNsConference,      // conferencing default
-    kNsLowSuppression,  // lowest suppression
-    kNsModerateSuppression,
-    kNsHighSuppression,
-    kNsVeryHighSuppression,     // highest suppression
-};
-
-enum AgcModes                  // type of Automatic Gain Control
-{
-    kAgcUnchanged = 0,        // previously set mode
-    kAgcDefault,              // platform default
-    // adaptive mode for use when analog volume control exists (e.g. for
-    // PC softphone)
-    kAgcAdaptiveAnalog,
-    // scaling takes place in the digital domain (e.g. for conference servers
-    // and embedded devices)
-    kAgcAdaptiveDigital,
-    // can be used on embedded devices where the the capture signal is level
-    // is predictable
-    kAgcFixedDigital
-};
-
-// EC modes
-enum EcModes                   // type of Echo Control
-{
-    kEcUnchanged = 0,          // previously set mode
-    kEcDefault,                // platform default
-    kEcConference,             // conferencing default (aggressive AEC)
-    kEcAec,                    // Acoustic Echo Cancellation
-    kEcAecm,                   // AEC mobile
-};
-
-// AECM modes
-enum AecmModes                 // mode of AECM
-{
-    kAecmQuietEarpieceOrHeadset = 0,
-                               // Quiet earpiece or headset use
-    kAecmEarpiece,             // most earpiece use
-    kAecmLoudEarpiece,         // Loud earpiece or quiet speakerphone use
-    kAecmSpeakerphone,         // most speakerphone use (default)
-    kAecmLoudSpeakerphone      // Loud speakerphone
-};
-
-// AGC configuration
-typedef struct
-{
-    unsigned short targetLeveldBOv;
-    unsigned short digitalCompressionGaindB;
-    bool           limiterEnable;
-} AgcConfig;                  // AGC configuration parameters
-
-enum StereoChannel
-{
-    kStereoLeft = 0,
-    kStereoRight,
-    kStereoBoth
-};
-
-// Audio device layers
-enum AudioLayers
-{
-    kAudioPlatformDefault = 0,
-    kAudioWindowsWave = 1,
-    kAudioWindowsCore = 2,
-    kAudioLinuxAlsa = 3,
-    kAudioLinuxPulse = 4
-};
-
-enum NetEqModes             // NetEQ playout configurations
-{
-    // Optimized trade-off between low delay and jitter robustness for two-way
-    // communication.
-    kNetEqDefault = 0,
-    // Improved jitter robustness at the cost of increased delay. Can be
-    // used in one-way communication.
-    kNetEqStreaming = 1,
-    // Optimzed for decodability of fax signals rather than for perceived audio
-    // quality.
-    kNetEqFax = 2,
-};
-
-enum NetEqBgnModes          // NetEQ Background Noise (BGN) configurations
-{
-    // BGN is always on and will be generated when the incoming RTP stream
-    // stops (default).
-    kBgnOn = 0,
-    // The BGN is faded to zero (complete silence) after a few seconds.
-    kBgnFade = 1,
-    // BGN is not used at all. Silence is produced after speech extrapolation
-    // has faded.
-    kBgnOff = 2,
-};
-
-enum OnHoldModes            // On Hold direction
-{
-    kHoldSendAndPlay = 0,    // Put both sending and playing in on-hold state.
-    kHoldSendOnly,           // Put only sending in on-hold state.
-    kHoldPlayOnly            // Put only playing in on-hold state.
-};
-
-enum AmrMode
-{
-    kRfc3267BwEfficient = 0,
-    kRfc3267OctetAligned = 1,
-    kRfc3267FileStorage = 2,
-};
-
-// ==================================================================
-// Video specific types
-// ==================================================================
-
-// Raw video types
-enum RawVideoType
-{
-    kVideoI420     = 0,
-    kVideoYV12     = 1,
-    kVideoYUY2     = 2,
-    kVideoUYVY     = 3,
-    kVideoIYUV     = 4,
-    kVideoARGB     = 5,
-    kVideoRGB24    = 6,
-    kVideoRGB565   = 7,
-    kVideoARGB4444 = 8,
-    kVideoARGB1555 = 9,
-    kVideoMJPEG    = 10,
-    kVideoNV12     = 11,
-    kVideoNV21     = 12,
-    kVideoUnknown  = 99
-};
-
-// Video codec
-enum { kConfigParameterSize = 128};
-enum { kPayloadNameSize = 32};
-
-// H.263 specific
-struct VideoCodecH263
-{
-    char quality;
-};
-
-// H.264 specific
-enum H264Packetization
-{
-    kH264SingleMode         = 0,
-    kH264NonInterleavedMode = 1
-};
-
-enum VideoCodecComplexity
-{
-    kComplexityNormal = 0,
-    kComplexityHigh    = 1,
-    kComplexityHigher  = 2,
-    kComplexityMax     = 3
-};
-
-enum VideoCodecProfile
-{
-    kProfileBase = 0x00,
-    kProfileMain = 0x01
-};
-
-struct VideoCodecH264
-{
-    H264Packetization          packetization;
-    VideoCodecComplexity       complexity;
-    VideoCodecProfile          profile;
-    char                       level;
-    char                       quality;
-
-    bool                       useFMO;
-
-    unsigned char              configParameters[kConfigParameterSize];
-    unsigned char              configParametersSize;
-};
-
-// VP8 specific
-struct VideoCodecVP8
-{
-    bool                       pictureLossIndicationOn;
-    bool                       feedbackModeOn;
-    VideoCodecComplexity       complexity;
-};
-
-// MPEG-4 specific
-struct VideoCodecMPEG4
-{
-    unsigned char   configParameters[kConfigParameterSize];
-    unsigned char   configParametersSize;
-    char            level;
-};
-
-// Unknown specific
-struct VideoCodecGeneric
-{
-};
-
-// Video codec types
-enum VideoCodecType
-{
-    kVideoCodecH263,
-    kVideoCodecH264,
-    kVideoCodecVP8,
-    kVideoCodecMPEG4,
-    kVideoCodecI420,
-    kVideoCodecRED,
-    kVideoCodecULPFEC,
-    kVideoCodecUnknown
-};
-
-union VideoCodecUnion
-{
-    VideoCodecH263      H263;
-    VideoCodecH264      H264;
-    VideoCodecVP8       VP8;
-    VideoCodecMPEG4     MPEG4;
-    VideoCodecGeneric   Generic;
-};
-
-// Common video codec properties
-struct VideoCodec
-{
-    VideoCodecType      codecType;
-    char                plName[kPayloadNameSize];
-    unsigned char       plType;
-
-    unsigned short      width;
-    unsigned short      height;
-
-    unsigned int        startBitrate;
-    unsigned int        maxBitrate;
-    unsigned int        minBitrate;
-    unsigned char       maxFramerate;
-
-    VideoCodecUnion     codecSpecific;
-
-    unsigned int        qpMax;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_COMMON_TYPES_H
diff --git a/src/modules/audio_processing/OWNERS b/src/modules/audio_processing/OWNERS
deleted file mode 100644
index aecf56e..0000000
--- a/src/modules/audio_processing/OWNERS
+++ /dev/null
@@ -1,2 +0,0 @@
-ajm@google.com
-bjornv@google.com
diff --git a/src/modules/audio_processing/aec/main/interface/echo_cancellation.h b/src/modules/audio_processing/aec/main/interface/echo_cancellation.h
deleted file mode 100644
index 883357d..0000000
--- a/src/modules/audio_processing/aec/main/interface/echo_cancellation.h
+++ /dev/null
@@ -1,260 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_
-
-#include "typedefs.h"
-
-// Errors
-#define AEC_UNSPECIFIED_ERROR           12000
-#define AEC_UNSUPPORTED_FUNCTION_ERROR  12001
-#define AEC_UNINITIALIZED_ERROR         12002
-#define AEC_NULL_POINTER_ERROR          12003
-#define AEC_BAD_PARAMETER_ERROR         12004
-
-// Warnings
-#define AEC_BAD_PARAMETER_WARNING       12050
-
-enum {
-    kAecNlpConservative = 0,
-    kAecNlpModerate,
-    kAecNlpAggressive
-};
-
-enum {
-    kAecFalse = 0,
-    kAecTrue
-};
-
-typedef struct {
-    WebRtc_Word16 nlpMode;        // default kAecNlpModerate
-    WebRtc_Word16 skewMode;       // default kAecFalse
-    WebRtc_Word16 metricsMode;    // default kAecFalse
-    //float realSkew;
-} AecConfig;
-
-typedef struct {
-    WebRtc_Word16 instant;
-    WebRtc_Word16 average;
-    WebRtc_Word16 max;
-    WebRtc_Word16 min;
-} AecLevel;
-
-typedef struct {
-    AecLevel rerl;
-    AecLevel erl;
-    AecLevel erle;
-    AecLevel aNlp;
-} AecMetrics;
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * Allocates the memory needed by the AEC. The memory needs to be initialized
- * separately using the WebRtcAec_Init() function.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void **aecInst               Pointer to the AEC instance to be created
- *                              and initilized
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return          0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_Create(void **aecInst);
-
-/*
- * This function releases the memory allocated by WebRtcAec_Create().
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void         *aecInst        Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_Free(void *aecInst);
-
-/*
- * Initializes an AEC instance.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- * WebRtc_Word32  sampFreq      Sampling frequency of data
- * WebRtc_Word32  scSampFreq    Soundcard sampling frequency
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return          0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_Init(void *aecInst,
-                             WebRtc_Word32 sampFreq,
-                             WebRtc_Word32 scSampFreq);
-
-/*
- * Inserts an 80 or 160 sample block of data into the farend buffer.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- * WebRtc_Word16  *farend       In buffer containing one frame of
- *                              farend signal for L band
- * WebRtc_Word16  nrOfSamples   Number of samples in farend buffer
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst,
-                                     const WebRtc_Word16 *farend,
-                                     WebRtc_Word16 nrOfSamples);
-
-/*
- * Runs the echo canceller on an 80 or 160 sample blocks of data.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void          *aecInst       Pointer to the AEC instance
- * WebRtc_Word16 *nearend       In buffer containing one frame of
- *                              nearend+echo signal for L band
- * WebRtc_Word16 *nearendH      In buffer containing one frame of
- *                              nearend+echo signal for H band
- * WebRtc_Word16 nrOfSamples    Number of samples in nearend buffer
- * WebRtc_Word16 msInSndCardBuf Delay estimate for sound card and
- *                              system buffers
- * WebRtc_Word16 skew           Difference between number of samples played
- *                              and recorded at the soundcard (for clock skew
- *                              compensation)
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word16  *out          Out buffer, one frame of processed nearend
- *                              for L band
- * WebRtc_Word16  *outH         Out buffer, one frame of processed nearend
- *                              for H band
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_Process(void *aecInst,
-                                const WebRtc_Word16 *nearend,
-                                const WebRtc_Word16 *nearendH,
-                                WebRtc_Word16 *out,
-                                WebRtc_Word16 *outH,
-                                WebRtc_Word16 nrOfSamples,
-                                WebRtc_Word16 msInSndCardBuf,
-                                WebRtc_Word32 skew);
-
-/*
- * This function enables the user to set certain parameters on-the-fly.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- * AecConfig      config        Config instance that contains all
- *                              properties to be set
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config);
-
-/*
- * Gets the on-the-fly paramters.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * AecConfig      *config       Pointer to the config instance that
- *                              all properties will be written to
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config);
-
-/*
- * Gets the current echo status of the nearend signal.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word16  *status       0: Almost certainly nearend single-talk
- *                              1: Might not be neared single-talk
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status);
-
-/*
- * Gets the current echo metrics for the session.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * AecMetrics     *metrics      Struct which will be filled out with the
- *                              current echo metrics.
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics);
-
-/*
- * Gets the last error code.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecInst      Pointer to the AEC instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        11000-11100: error code
- */
-WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst);
-
-/*
- * Gets a version string.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * char           *versionStr   Pointer to a string array
- * WebRtc_Word16  len           The maximum length of the string
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word8   *versionStr   Pointer to a string array
- * WebRtc_Word32  return         0: OK
- *                              -1: error
- */
-WebRtc_Word32 WebRtcAec_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len);
-
-#ifdef __cplusplus
-}
-#endif
-#endif  /* WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_INTERFACE_ECHO_CANCELLATION_H_ */
diff --git a/src/modules/audio_processing/aec/main/matlab/fullaec.m b/src/modules/audio_processing/aec/main/matlab/fullaec.m
deleted file mode 100644
index 0f86a8c..0000000
--- a/src/modules/audio_processing/aec/main/matlab/fullaec.m
+++ /dev/null
@@ -1,953 +0,0 @@
-% Partitioned block frequency domain adaptive filtering NLMS and 
-% standard time-domain sample-based NLMS 
-%fid=fopen('aecFar-samsung.pcm', 'rb'); % Load far end
-fid=fopen('aecFar.pcm', 'rb'); % Load far end
-%fid=fopen(farFile, 'rb'); % Load far end
-rrin=fread(fid,inf,'int16');
-fclose(fid); 
-%rrin=loadsl('data/far_me2.pcm'); % Load far end
-%fid=fopen('aecNear-samsung.pcm', 'rb'); % Load near end
-fid=fopen('aecNear.pcm', 'rb'); % Load near end
-%fid=fopen(nearFile, 'rb'); % Load near end
-ssin=fread(fid,inf,'int16');
-%ssin = [zeros(1024,1) ; ssin(1:end-1024)];
-
-fclose(fid);
-rand('state',13);
-fs=16000;
-mult=fs/8000;
-%rrin=rrin(fs*0+1:round(fs*120));
-%ssin=ssin(fs*0+1:round(fs*120));
-if fs == 8000
-	cohRange = 2:3;
-elseif fs==16000
-	cohRange = 2;
-end
-
-% Flags
-NLPon=1;  % NLP
-CNon=1; % Comfort noise
-PLTon=1;  % Plotting
-
-M = 16; % Number of partitions
-N = 64; % Partition length
-L = M*N; % Filter length 
-if fs == 8000
-    mufb = 0.6;
-else
-    mufb = 0.5;  
-end
-%mufb=1;  
-VADtd=48;
-alp = 0.1; % Power estimation factor alc = 0.1; % Coherence estimation factor
-beta = 0.9; % Plotting factor 
-%% Changed a little %%
-step = 0.3;%0.1875; % Downward step size 
-%%
-if fs == 8000
-    threshold=2e-6;  % DTrob threshold
-else
-    %threshold=0.7e-6;
-    threshold=1.5e-6; end
-
-if fs == 8000
-    echoBandRange = ceil(300*2/fs*N):floor(1800*2/fs*N);
-    %echoBandRange = ceil(1500*2/fs*N):floor(2500*2/fs*N);
-else
-    echoBandRange = ceil(300*2/fs*N):floor(1800*2/fs*N);
-    %echoBandRange = ceil(300*2/fs*N):floor(1800*2/fs*N);
-end
-%echoBandRange = ceil(1600*2/fs*N):floor(1900*2/fs*N);
-%echoBandRange = ceil(2000*2/fs*N):floor(4000*2/fs*N);
-suppState = 1;
-transCtr = 0;
-
-Nt=1;
-vt=1;
-
-ramp = 1.0003; % Upward ramp
-rampd = 0.999; % Downward ramp
-cvt = 20; % Subband VAD threshold;
-nnthres = 20; % Noise threshold 
-
-shh=logspace(-1.3,-2.2,N+1)';
-sh=[shh;flipud(shh(2:end-1))]; % Suppression profile
-
-len=length(ssin);
-w=zeros(L,1); % Sample-based TD NLMS 
-WFb=zeros(N+1,M); % Block-based FD NLMS
-WFbOld=zeros(N+1,M); % Block-based FD NLMS
-YFb=zeros(N+1,M);
-erfb=zeros(len,1);
-erfb3=zeros(len,1);
-
-ercn=zeros(len,1);
-zm=zeros(N,1);
-XFm=zeros(N+1,M);
-YFm=zeros(N+1,M);
-pn0=10*ones(N+1,1);
-pn=zeros(N+1,1);
-NN=len;
-Nb=floor(NN/N)-M;
-erifb=zeros(Nb+1,1)+0.1;
-erifb3=zeros(Nb+1,1)+0.1;
-ericn=zeros(Nb+1,1)+0.1;
-dri=zeros(Nb+1,1)+0.1;
-start=1;
-xo=zeros(N,1);
-do=xo;
-eo=xo;
-
-echoBands=zeros(Nb+1,1);
-cohxdAvg=zeros(Nb+1,1);
-cohxdSlow=zeros(Nb+1,N+1);
-cohedSlow=zeros(Nb+1,N+1);
-%overdriveM=zeros(Nb+1,N+1);
-cohxdFastAvg=zeros(Nb+1,1);
-cohxdAvgBad=zeros(Nb+1,1);
-cohedAvg=zeros(Nb+1,1);
-cohedFastAvg=zeros(Nb+1,1);
-hnledAvg=zeros(Nb+1,1);
-hnlxdAvg=zeros(Nb+1,1);
-ovrdV=zeros(Nb+1,1);
-dIdxV=zeros(Nb+1,1);
-SLxV=zeros(Nb+1,1);
-hnlSortQV=zeros(Nb+1,1);
-hnlPrefAvgV=zeros(Nb+1,1);
-mutInfAvg=zeros(Nb+1,1);
-%overdrive=zeros(Nb+1,1);
-hnled = zeros(N+1, 1);
-weight=zeros(N+1,1);
-hnlMax = zeros(N+1, 1);
-hnl = zeros(N+1, 1);
-overdrive = ones(1, N+1);
-xfwm=zeros(N+1,M);
-dfm=zeros(N+1,M);
-WFbD=ones(N+1,1);
-
-fbSupp = 0;
-hnlLocalMin = 1;
-cohxdLocalMin = 1;
-hnlLocalMinV=zeros(Nb+1,1);
-cohxdLocalMinV=zeros(Nb+1,1);
-hnlMinV=zeros(Nb+1,1);
-dkEnV=zeros(Nb+1,1);
-ekEnV=zeros(Nb+1,1);
-ovrd = 2;
-ovrdPos = floor((N+1)/4);
-ovrdSm = 2;
-hnlMin = 1;
-minCtr = 0;
-SeMin = 0;
-SdMin = 0;
-SeLocalAvg = 0;
-SeMinSm = 0;
-divergeFact = 1;
-dIdx = 1;
-hnlMinCtr = 0;
-hnlNewMin = 0;
-divergeState = 0;
-
-Sy=ones(N+1,1);
-Sym=1e7*ones(N+1,1);
-
-wins=[0;sqrt(hanning(2*N-1))];
-ubufn=zeros(2*N,1);
-ebuf=zeros(2*N,1);
-ebuf2=zeros(2*N,1);
-ebuf4=zeros(2*N,1);
-mbuf=zeros(2*N,1);
-
-cohedFast = zeros(N+1,1);
-cohxdFast = zeros(N+1,1);
-cohxd = zeros(N+1,1);
-Se = zeros(N+1,1);
-Sd = zeros(N+1,1);
-Sx = zeros(N+1,1);
-SxBad = zeros(N+1,1);
-Sed = zeros(N+1,1);
-Sxd = zeros(N+1,1);
-SxdBad = zeros(N+1,1);
-hnledp=[];
-
-cohxdMax = 0;
-
-%hh=waitbar(0,'Please wait...');
-progressbar(0);
-
-%spaces = ' ';
-%spaces = repmat(spaces, 50, 1);
-%spaces = ['[' ; spaces ; ']'];
-%fprintf(1, spaces);
-%fprintf(1, '\n');
-
-for kk=1:Nb
-    pos = N * (kk-1) + start;
-    
-    % FD block method
-    % ----------------------   Organize data
-    xk = rrin(pos:pos+N-1);
-    dk = ssin(pos:pos+N-1);
-    
-    xx = [xo;xk];
-    xo = xk;
-    tmp = fft(xx); 
-	XX = tmp(1:N+1);
-
-	dd = [do;dk];  % Overlap
-	do = dk;
-	tmp = fft(dd); % Frequency domain 
-	DD = tmp(1:N+1);
-    
-    % ------------------------  Power estimation
-    pn0 = (1 - alp) * pn0 + alp * real(XX.* conj(XX));
-    pn = pn0;
-    %pn = (1 - alp) * pn + alp * M * pn0;
-	if (CNon)
-		Yp =  real(conj(DD).*DD); % Instantaneous power      
-		Sy =  (1 - alp) * Sy + alp * Yp; % Averaged power    
-		
-		mm = min(Sy,Sym);  
-		diff = Sym - mm;
-		if (kk>50)
-			Sym = (mm + step*diff) * ramp; % Estimated background noise power   
-		end
-	end
-    
-    % ----------------------   Filtering   
-    XFm(:,1) = XX;
-    for mm=0:(M-1)
-        m=mm+1;  
-        YFb(:,m) = XFm(:,m) .* WFb(:,m);
-    end
-    yfk = sum(YFb,2);
-	tmp = [yfk ; flipud(conj(yfk(2:N)))];
-    ykt = real(ifft(tmp));
-    ykfb = ykt(end-N+1:end); 
-    
-    % ----------------------   Error estimation 
-    ekfb = dk - ykfb; 
-    %if sum(abs(ekfb)) < sum(abs(dk))
-        %ekfb = dk - ykfb; 
-    %    erfb(pos:pos+N-1) = ekfb; 
-    %else
-        %ekfb = dk;
-    %    erfb(pos:pos+N-1) = dk; 
-    %end
-	%(kk-1)*(N*2)+1
-    erfb(pos:pos+N-1) = ekfb; 
-    tmp = fft([zm;ekfb]);      % FD version for cancelling part (overlap-save)
-	Ek = tmp(1:N+1);
-
-    % ------------------------  Adaptation  
-	Ek2 = Ek ./(M*pn + 0.001); % Normalized error
-	%Ek2 = Ek ./(pn + 0.001); % Normalized error
-	%Ek2 = Ek ./(100*pn + 0.001); % Normalized error
-	
-	absEf = max(abs(Ek2), threshold);
-	absEf = ones(N+1,1)*threshold./absEf;
-	Ek2 = Ek2.*absEf;
-	
-	mEk = mufb.*Ek2;
-	PP = conj(XFm).*(ones(M,1) * mEk')'; 
-	tmp = [PP ; flipud(conj(PP(2:N,:)))];
-	IFPP = real(ifft(tmp));
-	PH = IFPP(1:N,:);
-	tmp = fft([PH;zeros(N,M)]);
-	FPH = tmp(1:N+1,:);
-	WFb = WFb + FPH;
-
-    if mod(kk, 10*mult) == 0
-        WFbEn = sum(real(WFb.*conj(WFb)));
-        %WFbEn = sum(abs(WFb));
-        [tmp, dIdx] = max(WFbEn);
-
-        WFbD = sum(abs(WFb(:, dIdx)),2);
-        %WFbD = WFbD / (mean(WFbD) + 1e-10); 
-        WFbD = min(max(WFbD, 0.5), 4);
-    end
-    dIdxV(kk) = dIdx;
-    
-	% NLP
-	if (NLPon)
-        
-        ee = [eo;ekfb]; 
-        eo = ekfb;
-		window = wins;
-        if fs == 8000
-            %gamma = 0.88;
-            gamma = 0.9;
-        else
-		    %gamma = 0.92;
-		    gamma = 0.93;
-        end
-		%gamma = 0.9;
-
-		tmp = fft(xx.*window);
-		xf = tmp(1:N+1);
-		tmp = fft(dd.*window);
-		df = tmp(1:N+1);
-		tmp = fft(ee.*window);
-		ef = tmp(1:N+1);
-
-        xfwm(:,1) = xf;
-        xf = xfwm(:,dIdx);
-        %fprintf(1,'%d: %f\n', kk, xf(4));
-        dfm(:,1) = df;
-        
-        SxOld = Sx;
-
-		Se = gamma*Se + (1-gamma)*real(ef.*conj(ef));
-		Sd = gamma*Sd + (1-gamma)*real(df.*conj(df));
-		Sx = gamma*Sx + (1 - gamma)*real(xf.*conj(xf));
-
-        %xRatio = real(xfwm(:,1).*conj(xfwm(:,1))) ./ ...
-        %    (real(xfwm(:,2).*conj(xfwm(:,2))) + 1e-10);
-        %xRatio = Sx ./ (SxOld + 1e-10);
-        %SLx = log(1/(N+1)*sum(xRatio)) - 1/(N+1)*sum(log(xRatio));
-        %SLxV(kk) = SLx;
-
-        %freqSm = 0.9;
-        %Sx = filter(freqSm, [1 -(1-freqSm)], Sx);
-        %Sx(end:1) = filter(freqSm, [1 -(1-freqSm)], Sx(end:1));
-        %Se = filter(freqSm, [1 -(1-freqSm)], Se);
-        %Se(end:1) = filter(freqSm, [1 -(1-freqSm)], Se(end:1));
-        %Sd = filter(freqSm, [1 -(1-freqSm)], Sd);
-        %Sd(end:1) = filter(freqSm, [1 -(1-freqSm)], Sd(end:1));
-
-		%SeFast = ef.*conj(ef);
-		%SdFast = df.*conj(df);
-        %SxFast = xf.*conj(xf);
-        %cohedFast = 0.9*cohedFast + 0.1*SeFast ./ (SdFast + 1e-10);
-        %cohedFast(find(cohedFast > 1)) = 1;
-        %cohedFast(find(cohedFast > 1)) = 1 ./ cohedFast(find(cohedFast>1));
-        %cohedFastAvg(kk) = mean(cohedFast(echoBandRange));
-        %cohedFastAvg(kk) = min(cohedFast);
-
-        %cohxdFast = 0.8*cohxdFast + 0.2*log(SdFast ./ (SxFast + 1e-10));
-        %cohxdFastAvg(kk) = mean(cohxdFast(echoBandRange));
-
-		% coherence
-        Sxd = gamma*Sxd + (1 - gamma)*xf.*conj(df);
-		Sed = gamma*Sed + (1-gamma)*ef.*conj(df);
-
-        %Sxd = filter(freqSm, [1 -(1-freqSm)], Sxd);
-        %Sxd(end:1) = filter(freqSm, [1 -(1-freqSm)], Sxd(end:1));
-        %Sed = filter(freqSm, [1 -(1-freqSm)], Sed);
-        %Sed(end:1) = filter(freqSm, [1 -(1-freqSm)], Sed(end:1));
-
-		cohed = real(Sed.*conj(Sed))./(Se.*Sd + 1e-10);
-        %cohedAvg(kk) = mean(cohed(echoBandRange));
-        %cohedAvg(kk) = cohed(6);
-        %cohedAvg(kk) = min(cohed);
-
-		cohxd = real(Sxd.*conj(Sxd))./(Sx.*Sd + 1e-10);
-        %freqSm = 0.5;
-        %cohxd(3:end) = filter(freqSm, [1 -(1-freqSm)], cohxd(3:end));
-        %cohxd(end:3) = filter(freqSm, [1 -(1-freqSm)], cohxd(end:3));
-        %cohxdAvg(kk) = mean(cohxd(echoBandRange));
-        %cohxdAvg(kk) = (cohxd(32));
-        %cohxdAvg(kk) = max(cohxd);
-
-        %xf = xfm(:,dIdx);
-		%SxBad = gamma*SxBad + (1 - gamma)*real(xf.*conj(xf));
-        %SxdBad = gamma*SxdBad + (1 - gamma)*xf.*conj(df);
-		%cohxdBad = real(SxdBad.*conj(SxdBad))./(SxBad.*Sd + 0.01);
-        %cohxdAvgBad(kk) = mean(cohxdBad);
-
-        %for j=1:N+1
-        %    mutInf(j) = 0.9*mutInf(j) + 0.1*information(abs(xfm(j,:)), abs(dfm(j,:)));
-        %end
-        %mutInfAvg(kk) = mean(mutInf);
-
-        %hnled = cohedFast;
-        %xIdx = find(cohxd > 1 - cohed);
-        %hnled(xIdx) = 1 - cohxd(xIdx);
-        %hnled = 1 - max(cohxd, 1-cohedFast);
-        hnled = min(1 - cohxd, cohed);
-        %hnled = 1 - cohxd;
-        %hnled = max(1 - (cohxd + (1-cohedFast)), 0);
-        %hnled = 1 - max(cohxd, 1-cohed);
-
-        if kk > 1
-            cohxdSlow(kk,:) = 0.99*cohxdSlow(kk-1,:) + 0.01*cohxd';
-            cohedSlow(kk,:) = 0.99*cohedSlow(kk-1,:) + 0.01*(1-cohed)';
-        end
-
-
-        if 0
-        %if kk > 50
-            %idx = find(hnled > 0.3);
-            hnlMax = hnlMax*0.9999;
-            %hnlMax(idx) = max(hnlMax(idx), hnled(idx));
-            hnlMax = max(hnlMax, hnled);
-            %overdrive(idx) = max(log(hnlMax(idx))/log(0.99), 1); 
-            avgHnl = mean(hnlMax(echoBandRange));
-            if avgHnl > 0.3
-                overdrive = max(log(avgHnl)/log(0.99), 1);
-            end
-            weight(4:end) = max(hnlMax) - hnlMax(4:end);
-        end
-        
-        
-
-        %[hg, gidx] = max(hnled); 
-        %fnrg = Sx(gidx) / (Sd(gidx) + 1e-10);
-        
-        %[tmp, bidx] = find((Sx / Sd + 1e-10) > fnrg);
-        %hnled(bidx) = hg;
-
-
-		%cohed1 = mean(cohed(cohRange)); % range depends on bandwidth 
-		%cohed1 = cohed1^2;
-        %echoBands(kk) = length(find(cohed(echoBandRange) < 0.25))/length(echoBandRange);
-
-        %if (fbSupp == 0)
-        %    if (echoBands(kk) > 0.8)
-        %        fbSupp = 1;
-        %    end
-        %else
-        %    if (echoBands(kk) < 0.6)
-        %        fbSupp = 0;
-        %    end
-        %end
-        %overdrive(kk) = 7.5*echoBands(kk) + 0.5; 
-        
-		% Factor by which to weight other bands
-		%if (cohed1 < 0.1)
-		%	w = 0.8 - cohed1*10*0.4;
-		%else
-		%	w = 0.4;
-		%end
-			
-		% Weight coherence subbands
-		%hnled = w*cohed1 + (1 - w)*cohed;
-		%hnled = (hnled).^2;
-		%cohed(floor(N/2):end) = cohed(floor(N/2):end).^2;
-        %if fbSupp == 1
-        %    cohed = zeros(size(cohed));
-        %end
-        %cohed = cohed.^overdrive(kk);
-
-        %hnled = gamma*hnled + (1 - gamma)*cohed;
-		% Additional hf suppression
-		%hnledp = [hnledp ; mean(hnled)];
-		%hnled(floor(N/2):end) = hnled(floor(N/2):end).^2;
-		%ef = ef.*((weight*(min(1 - hnled)).^2 + (1 - weight).*(1 - hnled)).^2);
-
-        cohedMean = mean(cohed(echoBandRange));
-        %aggrFact = 4*(1-mean(hnled(echoBandRange))) + 1;
-        %[hnlSort, hnlSortIdx] = sort(hnled(echoBandRange));
-        [hnlSort, hnlSortIdx] = sort(1-cohxd(echoBandRange));
-        [xSort, xSortIdx] = sort(Sx);
-        %aggrFact = (1-mean(hnled(echoBandRange)));
-        %hnlSortQ = hnlSort(qIdx);
-        hnlSortQ = mean(1 - cohxd(echoBandRange));
-        %hnlSortQ = mean(1 - cohxd);
-
-        [hnlSort2, hnlSortIdx2] = sort(hnled(echoBandRange));
-        %[hnlSort2, hnlSortIdx2] = sort(hnled);
-        hnlQuant = 0.75;
-        hnlQuantLow = 0.5;
-        qIdx = floor(hnlQuant*length(hnlSort2));
-        qIdxLow = floor(hnlQuantLow*length(hnlSort2));
-        hnlPrefAvg = hnlSort2(qIdx);
-        hnlPrefAvgLow = hnlSort2(qIdxLow);
-        %hnlPrefAvgLow = mean(hnled);
-        %hnlPrefAvg = max(hnlSort2);
-        %hnlPrefAvgLow = min(hnlSort2);
-
-        %hnlPref = hnled(echoBandRange);
-        %hnlPrefAvg = mean(hnlPref(xSortIdx((0.5*length(xSortIdx)):end)));
-
-        %hnlPrefAvg = min(hnlPrefAvg, hnlSortQ);
-
-        %hnlSortQIdx = hnlSortIdx(qIdx);
-        %SeQ = Se(qIdx + echoBandRange(1) - 1); 
-        %SdQ = Sd(qIdx + echoBandRange(1) - 1); 
-        %SeQ = Se(qIdxLow + echoBandRange(1) - 1); 
-        %SdQ = Sd(qIdxLow + echoBandRange(1) - 1); 
-        %propLow = length(find(hnlSort < 0.1))/length(hnlSort);
-        %aggrFact = min((1 - hnlSortQ)/2, 0.5);
-        %aggrTerm = 1/aggrFact;
-
-        %hnlg = mean(hnled(echoBandRange));
-        %hnlg = hnlSortQ;
-        %if suppState == 0
-        %    if hnlg < 0.05
-        %        suppState = 2;
-        %        transCtr = 0;
-        %    elseif hnlg < 0.75
-        %        suppState = 1;
-        %        transCtr = 0;
-        %    end
-        %elseif suppState == 1
-        %    if hnlg > 0.8
-        %        suppState = 0;
-        %        transCtr = 0;
-        %    elseif hnlg < 0.05
-        %        suppState = 2;
-        %        transCtr = 0;
-        %    end
-        %else
-        %    if hnlg > 0.8
-        %        suppState = 0;
-        %        transCtr = 0;
-        %    elseif hnlg > 0.25
-        %        suppState = 1;
-        %        transCtr = 0;
-        %    end
-        %end
-        %if kk > 50
-
-            if cohedMean > 0.98 & hnlSortQ > 0.9
-                %if suppState == 1
-                %    hnled = 0.5*hnled + 0.5*cohed;
-                %    %hnlSortQ = 0.5*hnlSortQ + 0.5*cohedMean;
-                %    hnlPrefAvg = 0.5*hnlPrefAvg + 0.5*cohedMean;
-                %else
-                %    hnled = cohed;
-                %    %hnlSortQ = cohedMean;
-                %    hnlPrefAvg = cohedMean;
-                %end
-                suppState = 0;
-            elseif cohedMean < 0.95 | hnlSortQ < 0.8
-                %if suppState == 0
-                %    hnled = 0.5*hnled + 0.5*cohed;
-                %    %hnlSortQ = 0.5*hnlSortQ + 0.5*cohedMean;
-                %    hnlPrefAvg = 0.5*hnlPrefAvg + 0.5*cohedMean;
-                %end
-                suppState = 1;
-            end
-
-            if hnlSortQ < cohxdLocalMin & hnlSortQ < 0.75
-                cohxdLocalMin = hnlSortQ;
-            end
-
-            if cohxdLocalMin == 1
-                ovrd = 3;
-                hnled = 1-cohxd;
-                hnlPrefAvg = hnlSortQ;
-                hnlPrefAvgLow = hnlSortQ;
-            end
-
-            if suppState == 0
-                hnled = cohed;
-                hnlPrefAvg = cohedMean;
-                hnlPrefAvgLow = cohedMean;
-            end
-
-            %if hnlPrefAvg < hnlLocalMin & hnlPrefAvg < 0.6
-            if hnlPrefAvgLow < hnlLocalMin & hnlPrefAvgLow < 0.6
-                %hnlLocalMin = hnlPrefAvg;
-                %hnlMin = hnlPrefAvg;
-                hnlLocalMin = hnlPrefAvgLow;
-                hnlMin = hnlPrefAvgLow;
-                hnlNewMin = 1;
-                hnlMinCtr = 0;
-                %if hnlMinCtr == 0
-                %    hnlMinCtr = hnlMinCtr + 1;
-                %else
-                %    hnlMinCtr = 0;
-                %    hnlMin = hnlLocalMin;
-                    %SeLocalMin = SeQ;
-                    %SdLocalMin = SdQ;
-                    %SeLocalAvg = 0;
-                    %minCtr = 0;
-                %    ovrd = max(log(0.0001)/log(hnlMin), 2);
-                    %divergeFact = hnlLocalMin;
-            end
-
-            if hnlNewMin == 1
-                hnlMinCtr = hnlMinCtr + 1;
-            end
-            if hnlMinCtr == 2 
-                hnlNewMin = 0;
-                hnlMinCtr = 0;
-                %ovrd = max(log(0.0001)/log(hnlMin), 2);
-                ovrd = max(log(0.00001)/(log(hnlMin + 1e-10) + 1e-10), 3);
-                %ovrd = max(log(0.00000001)/(log(hnlMin + 1e-10) + 1e-10), 5);
-                %ovrd = max(log(0.0001)/log(hnlPrefAvg), 2);
-                %ovrd = max(log(0.001)/log(hnlMin), 2);
-            end
-            hnlLocalMin = min(hnlLocalMin + 0.0008/mult, 1);
-            cohxdLocalMin = min(cohxdLocalMin + 0.0004/mult, 1);
-            %divergeFact = hnlSortQ;
-
-
-            %if minCtr > 0 & hnlLocalMin < 1
-            %    hnlMin = hnlLocalMin;
-            %    %SeMin = 0.9*SeMin + 0.1*sqrt(SeLocalMin);
-            %    SdMin = sqrt(SdLocalMin);
-            %    %SeMin = sqrt(SeLocalMin)*hnlSortQ;
-            %    SeMin = sqrt(SeLocalMin);
-            %    %ovrd = log(100/SeMin)/log(hnlSortQ);
-            %    %ovrd = log(100/SeMin)/log(hnlSortQ);
-            %    ovrd = log(0.01)/log(hnlMin);
-            %    ovrd = max(ovrd, 2);
-            %    ovrdPos = hnlSortQIdx;
-            %    %ovrd = max(ovrd, 1);
-            %    %SeMin = sqrt(SeLocalAvg/5);
-            %    minCtr = 0;
-            %else
-            %    %SeLocalMin = 0.9*SeLocalMin +0.1*SeQ;
-            %    SeLocalAvg = SeLocalAvg + SeQ;
-            %    minCtr = minCtr + 1;
-            %end
-
-            if ovrd < ovrdSm
-                ovrdSm = 0.99*ovrdSm + 0.01*ovrd;
-            else
-                ovrdSm = 0.9*ovrdSm + 0.1*ovrd;
-            end
-        %end
-
-        %ekEn = sum(real(ekfb.^2));
-        %dkEn = sum(real(dk.^2));
-        ekEn = sum(Se);
-        dkEn = sum(Sd);
-
-        if divergeState == 0
-            if ekEn > dkEn
-                ef = df;
-                divergeState = 1;
-                %hnlPrefAvg = hnlSortQ;
-                %hnled = (1 - cohxd);
-            end
-        else
-            %if ekEn*1.1 < dkEn  
-            %if ekEn*1.26 < dkEn  
-            if ekEn*1.05 < dkEn  
-                divergeState = 0;
-            else
-                ef = df;
-            end
-        end
-
-        if ekEn > dkEn*19.95
-            WFb=zeros(N+1,M); % Block-based FD NLMS
-        end
-
-        ekEnV(kk) = ekEn;
-        dkEnV(kk) = dkEn;
-
-        hnlLocalMinV(kk) = hnlLocalMin;
-        cohxdLocalMinV(kk) = cohxdLocalMin;
-        hnlMinV(kk) = hnlMin;
-        %cohxdMaxLocal = max(cohxdSlow(kk,:));
-        %if kk > 50
-        %cohxdMaxLocal = 1-hnlSortQ;
-        %if cohxdMaxLocal > 0.5
-        %    %if cohxdMaxLocal > cohxdMax 
-        %        odScale = max(log(cohxdMaxLocal)/log(0.95), 1);
-        %        %overdrive(7:end) = max(log(cohxdSlow(kk,7:end))/log(0.9), 1);
-        %        cohxdMax = cohxdMaxLocal;
-        %    end
-        %end
-        %end
-        %cohxdMax = cohxdMax*0.999;
-
-        %overdriveM(kk,:) = max(overdrive, 1);
-        %aggrFact = 0.25;
-        aggrFact = 0.3;
-        %aggrFact = 0.5*propLow;
-        %if fs == 8000
-        %    wCurve = [0 ; 0 ; aggrFact*sqrt(linspace(0,1,N-1))' + 0.1];
-        %else
-        %    wCurve = [0; 0; 0; aggrFact*sqrt(linspace(0,1,N-2))' + 0.1];
-        %end
-        wCurve = [0; aggrFact*sqrt(linspace(0,1,N))' + 0.1];
-        % For sync with C
-        %if fs == 8000
-        %    wCurve = wCurve(2:end);
-        %else
-        %    wCurve = wCurve(1:end-1);
-        %end
-        %weight = aggrFact*(sqrt(linspace(0,1,N+1)'));
-        %weight = aggrFact*wCurve;
-        weight = wCurve;
-        %weight = aggrFact*ones(N+1,1);
-        %weight = zeros(N+1,1);
-        %hnled = weight.*min(hnled) + (1 - weight).*hnled;
-        %hnled = weight.*min(mean(hnled(echoBandRange)), hnled) + (1 - weight).*hnled;
-        %hnled = weight.*min(hnlSortQ, hnled) + (1 - weight).*hnled;
-
-        %hnlSortQV(kk) = mean(hnled);
-        %hnlPrefAvgV(kk) = mean(hnled(echoBandRange));
-
-        hnled = weight.*min(hnlPrefAvg, hnled) + (1 - weight).*hnled;
-
-        %od = aggrFact*(sqrt(linspace(0,1,N+1)') + aggrTerm);
-        %od = 4*(sqrt(linspace(0,1,N+1)') + 1/4);
-
-        %ovrdFact = (ovrdSm - 1) / sqrt(ovrdPos/(N+1));
-        %ovrdFact = ovrdSm / sqrt(echoBandRange(floor(length(echoBandRange)/2))/(N+1));
-        %od = ovrdFact*sqrt(linspace(0,1,N+1))' + 1;
-        %od = ovrdSm*ones(N+1,1).*abs(WFb(:,dIdx))/(max(abs(WFb(:,dIdx)))+1e-10);
-
-        %od = ovrdSm*ones(N+1,1);
-        %od = ovrdSm*WFbD.*(sqrt(linspace(0,1,N+1))' + 1);
-
-        od = ovrdSm*(sqrt(linspace(0,1,N+1))' + 1);
-        %od = 4*(sqrt(linspace(0,1,N+1))' + 1);
-
-        %od = 2*ones(N+1,1);
-        %od = 2*ones(N+1,1);
-        %sshift = ((1-hnled)*2-1).^3+1; 
-        sshift = ones(N+1,1);
-
-        hnled = hnled.^(od.*sshift);
-
-        %if hnlg > 0.75
-            %if (suppState ~= 0)
-            %    transCtr = 0;
-            %end
-        %    suppState = 0;
-        %elseif hnlg < 0.6 & hnlg > 0.2
-        %    suppState = 1;
-        %elseif hnlg < 0.1
-            %hnled = zeros(N+1, 1);
-            %if (suppState ~= 2)
-            %    transCtr = 0;
-            %end
-        %    suppState = 2;
-        %else
-        %    if (suppState ~= 2)
-        %        transCtr = 0;
-        %    end
-        %    suppState = 2;
-        %end
-        %if suppState == 0
-        %    hnled = ones(N+1, 1);
-        %elseif suppState == 2
-        %    hnled = zeros(N+1, 1);
-        %end
-        %hnled(find(hnled < 0.1)) = 0;
-        %hnled = hnled.^2;
-        %if transCtr < 5
-            %hnl = 0.75*hnl + 0.25*hnled;
-        %    transCtr = transCtr + 1;
-        %else
-            hnl = hnled;
-        %end
-        %hnled(find(hnled < 0.05)) = 0;
-        ef = ef.*(hnl);
-
-        %ef = ef.*(min(1 - cohxd, cohed).^2);
-        %ef = ef.*((1-cohxd).^2);
-        
-        ovrdV(kk) = ovrdSm;
-        %ovrdV(kk) = dIdx;
-        %ovrdV(kk) = divergeFact;
-        %hnledAvg(kk) = 1-mean(1-cohedFast(echoBandRange));
-        hnledAvg(kk) = 1-mean(1-cohed(echoBandRange));
-        hnlxdAvg(kk) = 1-mean(cohxd(echoBandRange));
-        %hnlxdAvg(kk) = cohxd(5);
-        %hnlSortQV(kk) = mean(hnled);
-        hnlSortQV(kk) = hnlPrefAvgLow;
-        hnlPrefAvgV(kk) = hnlPrefAvg;
-        %hnlAvg(kk) = propLow;
-        %ef(N/2:end) = 0;
-        %ner = (sum(Sd) ./ (sum(Se.*(hnl.^2)) + 1e-10));
-
-		% Comfort noise
-		if (CNon)
-			snn=sqrt(Sym);
-			snn(1)=0; % Reject LF noise
-			Un=snn.*exp(j*2*pi.*[0;rand(N-1,1);0]);
-
-			% Weight comfort noise by suppression
-			Un = sqrt(1-hnled.^2).*Un;
-			Fmix = ef + Un;
-		else
-			Fmix = ef;
-		end
-
-		% Overlap and add in time domain for smoothness 
-		tmp = [Fmix ; flipud(conj(Fmix(2:N)))];
-		mixw = wins.*real(ifft(tmp));
-		mola  = mbuf(end-N+1:end) + mixw(1:N);
-		mbuf = mixw;
-		ercn(pos:pos+N-1) = mola; 
-	end % NLPon
-
-    % Filter update
-	%Ek2 = Ek ./(12*pn + 0.001); % Normalized error
-    %Ek2 = Ek2 * divergeFact;
-	%Ek2 = Ek ./(pn + 0.001); % Normalized error
-	%Ek2 = Ek ./(100*pn + 0.001); % Normalized error
-
-    %divergeIdx = find(abs(Ek) > abs(DD));
-    %divergeIdx = find(Se > Sd);
-    %threshMod = threshold*ones(N+1,1); 
-    %if length(divergeIdx) > 0
-    %if sum(abs(Ek)) > sum(abs(DD))
-        %WFb(divergeIdx,:) = WFb(divergeIdx,:) .* repmat(sqrt(Sd(divergeIdx)./(Se(divergeIdx)+1e-10))),1,M);
-        %Ek2(divergeIdx) = Ek2(divergeIdx) .* sqrt(Sd(divergeIdx)./(Se(divergeIdx)+1e-10));
-        %Ek2(divergeIdx) = Ek2(divergeIdx) .* abs(DD(divergeIdx))./(abs(Ek(divergeIdx))+1e-10);
-        %WFb(divergeIdx,:) = WFbOld(divergeIdx,:);
-        %WFb = WFbOld;
-        %threshMod(divergeIdx) = threshMod(divergeIdx) .* abs(DD(divergeIdx))./(abs(Ek(divergeIdx))+1e-10);
-    %    threshMod(divergeIdx) = threshMod(divergeIdx) .* sqrt(Sd(divergeIdx)./(Se(divergeIdx)+1e-10));
-    %end
-	
-	%absEf = max(abs(Ek2), threshold);
-	%absEf = ones(N+1,1)*threshold./absEf;
-	%absEf = max(abs(Ek2), threshMod);
-	%absEf = threshMod./absEf;
-	%Ek2 = Ek2.*absEf;
-
-    %if sum(Se) <= sum(Sd)
-
-    %    mEk = mufb.*Ek2;
-    %    PP = conj(XFm).*(ones(M,1) * mEk')'; 
-    %    tmp = [PP ; flipud(conj(PP(2:N,:)))];
-    %    IFPP = real(ifft(tmp));
-    %    PH = IFPP(1:N,:);
-    %    tmp = fft([PH;zeros(N,M)]);
-    %    FPH = tmp(1:N+1,:);
-    %    %WFbOld = WFb;
-    %    WFb = WFb + FPH;
-
-    %else
-    %    WF = WFbOld;
-    %end
-
-	% Shift old FFTs
-	%for m=M:-1:2
-	%	XFm(:,m) = XFm(:,m-1);    
-	%	YFm(:,m) = YFm(:,m-1);    
-	%end
-    XFm(:,2:end) = XFm(:,1:end-1);
-    YFm(:,2:end) = YFm(:,1:end-1);
-    xfwm(:,2:end) = xfwm(:,1:end-1);
-    dfm(:,2:end) = dfm(:,1:end-1);
-
-	%if mod(kk, floor(Nb/50)) == 0
-    %    fprintf(1, '.');
-	%end
-
-	if mod(kk, floor(Nb/100)) == 0
-	%if mod(kk, floor(Nb/500)) == 0
-        progressbar(kk/Nb); 
-        %figure(5)
-        %plot(abs(WFb));
-        %legend('1','2','3','4','5','6','7','8','9','10','11','12');
-        %title(kk*N/fs);
-        %figure(6)
-        %plot(WFbD);
-        %figure(6)
-        %plot(threshMod)
-        %if length(divergeIdx) > 0
-        %    plot(abs(DD))
-        %    hold on
-        %    plot(abs(Ek), 'r')
-        %    hold off
-            %plot(min(sqrt(Sd./(Se+1e-10)),1))
-            %axis([0 N 0 1]);
-        %end
-        %figure(6)
-        %plot(cohedFast);
-        %axis([1 N+1 0 1]);
-        %plot(WFbEn);
-
-        %figure(7)
-        %plot(weight);
-        %plot([cohxd 1-cohed]);
-        %plot([cohxd 1-cohed 1-cohedFast hnled]);
-        %plot([cohxd cohxdFast/max(cohxdFast)]);
-        %legend('cohxd', '1-cohed', '1-cohedFast');
-        %axis([1 65 0 1]);
-        %pause(0.5);
-        %overdrive
-    end
-end
-progressbar(1);
-
-%figure(2);
-%plot([feat(:,1) feat(:,2)+1 feat(:,3)+2 mfeat+3]);
-%plot([feat(:,1) mfeat+1]);
-
-%figure(3);
-%plot(10*log10([dri erifb erifb3 ericn]));
-%legend('Near-end','Error','Post NLP','Final',4);
-% Compensate for delay
-%ercn=[ercn(N+1:end);zeros(N,1)];
-%ercn_=[ercn_(N+1:end);zeros(N,1)];
-
-%figure(11);
-%plot(cohxdSlow);
-
-%figure(12);
-%surf(cohxdSlow);
-%shading interp;
-
-%figure(13);
-%plot(overdriveM);
-
-%figure(14);
-%surf(overdriveM);
-%shading interp;
-
-figure(10);
-t = (0:Nb)*N/fs;
-rrinSubSamp = rrin(N*(1:(Nb+1)));
-plot(t, rrinSubSamp/max(abs(rrinSubSamp)),'b');
-hold on
-plot(t, hnledAvg, 'r');
-plot(t, hnlxdAvg, 'g');
-plot(t, hnlSortQV, 'y');
-plot(t, hnlLocalMinV, 'k');
-plot(t, cohxdLocalMinV, 'c');
-plot(t, hnlPrefAvgV, 'm');
-%plot(t, cohxdAvg, 'r');
-%plot(cohxdFastAvg, 'r');
-%plot(cohxdAvgBad, 'k');
-%plot(t, cohedAvg, 'k');
-%plot(t, 1-cohedFastAvg, 'k');
-%plot(ssin(N*(1:floor(length(ssin)/N)))/max(abs(ssin)));
-%plot(echoBands,'r');
-%plot(overdrive, 'g');
-%plot(erfb(N*(1:floor(length(erfb)/N)))/max(abs(erfb)));
-hold off
-tightx;
-
-figure(11)
-plot(t, ovrdV);
-tightx;
-%plot(mfeat,'r');
-%plot(1-cohxyp_,'r');
-%plot(Hnlxydp,'y');
-%plot(hnledp,'k');
-%plot(Hnlxydp, 'c');
-%plot(ccohpd_,'k');
-%plot(supplot_, 'g');
-%plot(ones(length(mfeat),1)*rr1_, 'k');
-%plot(ones(length(mfeat),1)*rr2_, 'k');
-%plot(N*(1:length(feat)), feat);
-%plot(Sep_,'r');
-%axis([1 floor(length(erfb)/N) -1 1])
-%hold off
-%plot(10*log10([Se_, Sx_, Seu_, real(sf_.*conj(sf_))]));
-%legend('Se','Sx','Seu','S');
-%figure(5)
-%plot([ercn ercn_]);
-
-figure(12)
-plot(t, dIdxV);
-%plot(t, SLxV);
-tightx;
-
-%figure(13)
-%plot(t, [ekEnV dkEnV]);
-%plot(t, dkEnV./(ekEnV+1e-10));
-%tightx;
-
-%close(hh);
-%spclab(fs,ssin,erfb,ercn,'outxd.pcm');
-%spclab(fs,rrin,ssin,erfb,1.78*ercn,'vqeOut-1.pcm');
-%spclab(fs,erfb,'aecOutLp.pcm');
-%spclab(fs,rrin,ssin,erfb,1.78*ercn,'aecOut25.pcm','vqeOut-1.pcm');
-%spclab(fs,rrin,ssin,erfb,ercn,'aecOut-mba.pcm');
-%spclab(fs,rrin,ssin,erfb,ercn,'aecOut.pcm');
-%spclab(fs, ssin, erfb, ercn, 'out0.pcm');
diff --git a/src/modules/audio_processing/aec/main/source/Android.mk b/src/modules/audio_processing/aec/main/source/Android.mk
deleted file mode 100644
index a0c1cd3..0000000
--- a/src/modules/audio_processing/aec/main/source/Android.mk
+++ /dev/null
@@ -1,54 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_aec
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := \
-    echo_cancellation.c \
-    aec_core.c \
-    aec_rdft.c \
-    resampler.c
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=  
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
-    '-DNO_HEAPCHECKER' \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_THREAD_RR' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' 
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../../.. \
-    $(LOCAL_PATH)/../interface \
-    $(LOCAL_PATH)/../../../utility \
-    $(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface 
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS := 
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
-    libdl \
-    libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/aec/main/source/aec.gyp b/src/modules/audio_processing/aec/main/source/aec.gyp
deleted file mode 100644
index 0427e00..0000000
--- a/src/modules/audio_processing/aec/main/source/aec.gyp
+++ /dev/null
@@ -1,49 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../../../../common_settings.gypi',
-  ],
-  'targets': [
-    {
-      'target_name': 'aec',
-      'type': '<(library)',
-      'dependencies': [
-        '../../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
-        '../../../utility/util.gyp:apm_util'
-      ],
-      'include_dirs': [
-        '../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-        ],
-      },
-      'sources': [
-        '../interface/echo_cancellation.h',
-        'echo_cancellation.c',
-        'aec_core.c',
-        'aec_core_sse2.c',
-        'aec_rdft.h',
-        'aec_rdft.c',
-        'aec_rdft_sse2.c',
-        'aec_core.h',
-        'resampler.c',
-        'resampler.h',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/aec/main/source/aec_core.c b/src/modules/audio_processing/aec/main/source/aec_core.c
deleted file mode 100644
index 81197ea..0000000
--- a/src/modules/audio_processing/aec/main/source/aec_core.c
+++ /dev/null
@@ -1,1456 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * The core AEC algorithm, which is presented with time-aligned signals.
- */
-
-#include <math.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "aec_core.h"
-#include "aec_rdft.h"
-#include "ring_buffer.h"
-#include "system_wrappers/interface/cpu_features_wrapper.h"
-
-// Noise suppression
-static const int converged = 250;
-
-// Metrics
-static const int subCountLen = 4;
-static const int countLen = 50;
-
-// Quantities to control H band scaling for SWB input
-static const int flagHbandCn = 1; // flag for adding comfort noise in H band
-static const float cnScaleHband = (float)0.4; // scale for comfort noise in H band
-// Initial bin for averaging nlp gain in low band
-static const int freqAvgIc = PART_LEN / 2;
-
-/* Matlab code to produce table:
-win = sqrt(hanning(63)); win = [0 ; win(1:32)];
-fprintf(1, '\t%.14f, %.14f, %.14f,\n', win);
-*/
-/*
-static const float sqrtHanning[33] = {
-    0.00000000000000, 0.04906767432742, 0.09801714032956,
-    0.14673047445536, 0.19509032201613, 0.24298017990326,
-    0.29028467725446, 0.33688985339222, 0.38268343236509,
-    0.42755509343028, 0.47139673682600, 0.51410274419322,
-    0.55557023301960, 0.59569930449243, 0.63439328416365,
-    0.67155895484702, 0.70710678118655, 0.74095112535496,
-    0.77301045336274, 0.80320753148064, 0.83146961230255,
-    0.85772861000027, 0.88192126434835, 0.90398929312344,
-    0.92387953251129, 0.94154406518302, 0.95694033573221,
-    0.97003125319454, 0.98078528040323, 0.98917650996478,
-    0.99518472667220, 0.99879545620517, 1.00000000000000
-};
-*/
-
-static const float sqrtHanning[65] = {
-    0.00000000000000f, 0.02454122852291f, 0.04906767432742f,
-    0.07356456359967f, 0.09801714032956f, 0.12241067519922f,
-    0.14673047445536f, 0.17096188876030f, 0.19509032201613f,
-    0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
-    0.29028467725446f, 0.31368174039889f, 0.33688985339222f,
-    0.35989503653499f, 0.38268343236509f, 0.40524131400499f,
-    0.42755509343028f, 0.44961132965461f, 0.47139673682600f,
-    0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
-    0.55557023301960f, 0.57580819141785f, 0.59569930449243f,
-    0.61523159058063f, 0.63439328416365f, 0.65317284295378f,
-    0.67155895484702f, 0.68954054473707f, 0.70710678118655f,
-    0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
-    0.77301045336274f, 0.78834642762661f, 0.80320753148064f,
-    0.81758481315158f, 0.83146961230255f, 0.84485356524971f,
-    0.85772861000027f, 0.87008699110871f, 0.88192126434835f,
-    0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
-    0.92387953251129f, 0.93299279883474f, 0.94154406518302f,
-    0.94952818059304f, 0.95694033573221f, 0.96377606579544f,
-    0.97003125319454f, 0.97570213003853f, 0.98078528040323f,
-    0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
-    0.99518472667220f, 0.99729045667869f, 0.99879545620517f,
-    0.99969881869620f, 1.00000000000000f
-};
-
-/* Matlab code to produce table:
-weightCurve = [0 ; 0.3 * sqrt(linspace(0,1,64))' + 0.1];
-fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', weightCurve);
-*/
-const float WebRtcAec_weightCurve[65] = {
-    0.0000f, 0.1000f, 0.1378f, 0.1535f, 0.1655f, 0.1756f,
-    0.1845f, 0.1926f, 0.2000f, 0.2069f, 0.2134f, 0.2195f,
-    0.2254f, 0.2309f, 0.2363f, 0.2414f, 0.2464f, 0.2512f,
-    0.2558f, 0.2604f, 0.2648f, 0.2690f, 0.2732f, 0.2773f,
-    0.2813f, 0.2852f, 0.2890f, 0.2927f, 0.2964f, 0.3000f,
-    0.3035f, 0.3070f, 0.3104f, 0.3138f, 0.3171f, 0.3204f,
-    0.3236f, 0.3268f, 0.3299f, 0.3330f, 0.3360f, 0.3390f,
-    0.3420f, 0.3449f, 0.3478f, 0.3507f, 0.3535f, 0.3563f,
-    0.3591f, 0.3619f, 0.3646f, 0.3673f, 0.3699f, 0.3726f,
-    0.3752f, 0.3777f, 0.3803f, 0.3828f, 0.3854f, 0.3878f,
-    0.3903f, 0.3928f, 0.3952f, 0.3976f, 0.4000f
-};
-
-/* Matlab code to produce table:
-overDriveCurve = [sqrt(linspace(0,1,65))' + 1];
-fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', overDriveCurve);
-*/
-const float WebRtcAec_overDriveCurve[65] = {
-    1.0000f, 1.1250f, 1.1768f, 1.2165f, 1.2500f, 1.2795f,
-    1.3062f, 1.3307f, 1.3536f, 1.3750f, 1.3953f, 1.4146f,
-    1.4330f, 1.4507f, 1.4677f, 1.4841f, 1.5000f, 1.5154f,
-    1.5303f, 1.5449f, 1.5590f, 1.5728f, 1.5863f, 1.5995f,
-    1.6124f, 1.6250f, 1.6374f, 1.6495f, 1.6614f, 1.6731f,
-    1.6847f, 1.6960f, 1.7071f, 1.7181f, 1.7289f, 1.7395f,
-    1.7500f, 1.7603f, 1.7706f, 1.7806f, 1.7906f, 1.8004f,
-    1.8101f, 1.8197f, 1.8292f, 1.8385f, 1.8478f, 1.8570f,
-    1.8660f, 1.8750f, 1.8839f, 1.8927f, 1.9014f, 1.9100f,
-    1.9186f, 1.9270f, 1.9354f, 1.9437f, 1.9520f, 1.9601f,
-    1.9682f, 1.9763f, 1.9843f, 1.9922f, 2.0000f
-};
-
-// "Private" function prototypes.
-static void ProcessBlock(aec_t *aec, const short *farend,
-                              const short *nearend, const short *nearendH,
-                              short *out, short *outH);
-
-static void BufferFar(aec_t *aec, const short *farend, int farLen);
-static void FetchFar(aec_t *aec, short *farend, int farLen, int knownDelay);
-
-static void NonLinearProcessing(aec_t *aec, short *output, short *outputH);
-
-static void GetHighbandGain(const float *lambda, float *nlpGainHband);
-
-// Comfort_noise also computes noise for H band returned in comfortNoiseHband
-static void ComfortNoise(aec_t *aec, float efw[2][PART_LEN1],
-                                  complex_t *comfortNoiseHband,
-                                  const float *noisePow, const float *lambda);
-
-static void WebRtcAec_InitLevel(power_level_t *level);
-static void WebRtcAec_InitStats(stats_t *stats);
-static void UpdateLevel(power_level_t *level, const short *in);
-static void UpdateMetrics(aec_t *aec);
-
-__inline static float MulRe(float aRe, float aIm, float bRe, float bIm)
-{
-    return aRe * bRe - aIm * bIm;
-}
-
-__inline static float MulIm(float aRe, float aIm, float bRe, float bIm)
-{
-    return aRe * bIm + aIm * bRe;
-}
-
-static int CmpFloat(const void *a, const void *b)
-{
-    const float *da = (const float *)a;
-    const float *db = (const float *)b;
-
-    return (*da > *db) - (*da < *db);
-}
-
-int WebRtcAec_CreateAec(aec_t **aecInst)
-{
-    aec_t *aec = malloc(sizeof(aec_t));
-    *aecInst = aec;
-    if (aec == NULL) {
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aec->farFrBuf, FRAME_LEN + PART_LEN) == -1) {
-        WebRtcAec_FreeAec(aec);
-        aec = NULL;
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aec->nearFrBuf, FRAME_LEN + PART_LEN) == -1) {
-        WebRtcAec_FreeAec(aec);
-        aec = NULL;
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aec->outFrBuf, FRAME_LEN + PART_LEN) == -1) {
-        WebRtcAec_FreeAec(aec);
-        aec = NULL;
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aec->nearFrBufH, FRAME_LEN + PART_LEN) == -1) {
-        WebRtcAec_FreeAec(aec);
-        aec = NULL;
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aec->outFrBufH, FRAME_LEN + PART_LEN) == -1) {
-        WebRtcAec_FreeAec(aec);
-        aec = NULL;
-        return -1;
-    }
-
-    return 0;
-}
-
-int WebRtcAec_FreeAec(aec_t *aec)
-{
-    if (aec == NULL) {
-        return -1;
-    }
-
-    WebRtcApm_FreeBuffer(aec->farFrBuf);
-    WebRtcApm_FreeBuffer(aec->nearFrBuf);
-    WebRtcApm_FreeBuffer(aec->outFrBuf);
-
-    WebRtcApm_FreeBuffer(aec->nearFrBufH);
-    WebRtcApm_FreeBuffer(aec->outFrBufH);
-
-    free(aec);
-    return 0;
-}
-
-static void FilterFar(aec_t *aec, float yf[2][PART_LEN1])
-{
-  int i;
-  for (i = 0; i < NR_PART; i++) {
-    int j;
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
-    int pos = i * PART_LEN1;
-    // Check for wrap
-    if (i + aec->xfBufBlockPos >= NR_PART) {
-      xPos -= NR_PART*(PART_LEN1);
-    }
-
-    for (j = 0; j < PART_LEN1; j++) {
-      yf[0][j] += MulRe(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
-      yf[1][j] += MulIm(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
-    }
-  }
-}
-
-static void ScaleErrorSignal(aec_t *aec, float ef[2][PART_LEN1])
-{
-  int i;
-  float absEf;
-  for (i = 0; i < (PART_LEN1); i++) {
-    ef[0][i] /= (aec->xPow[i] + 1e-10f);
-    ef[1][i] /= (aec->xPow[i] + 1e-10f);
-    absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
-
-    if (absEf > aec->errThresh) {
-      absEf = aec->errThresh / (absEf + 1e-10f);
-      ef[0][i] *= absEf;
-      ef[1][i] *= absEf;
-    }
-
-    // Stepsize factor
-    ef[0][i] *= aec->mu;
-    ef[1][i] *= aec->mu;
-  }
-}
-
-static void FilterAdaptation(aec_t *aec, float *fft, float ef[2][PART_LEN1]) {
-  int i, j;
-  for (i = 0; i < NR_PART; i++) {
-    int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
-    int pos;
-    // Check for wrap
-    if (i + aec->xfBufBlockPos >= NR_PART) {
-      xPos -= NR_PART * PART_LEN1;
-    }
-
-    pos = i * PART_LEN1;
-
-#ifdef UNCONSTR
-    for (j = 0; j < PART_LEN1; j++) {
-      aec->wfBuf[pos + j][0] += MulRe(aec->xfBuf[xPos + j][0],
-                                      -aec->xfBuf[xPos + j][1],
-                                      ef[j][0], ef[j][1]);
-      aec->wfBuf[pos + j][1] += MulIm(aec->xfBuf[xPos + j][0],
-                                      -aec->xfBuf[xPos + j][1],
-                                      ef[j][0], ef[j][1]);
-    }
-#else
-    for (j = 0; j < PART_LEN; j++) {
-
-      fft[2 * j] = MulRe(aec->xfBuf[0][xPos + j],
-                         -aec->xfBuf[1][xPos + j],
-                         ef[0][j], ef[1][j]);
-      fft[2 * j + 1] = MulIm(aec->xfBuf[0][xPos + j],
-                             -aec->xfBuf[1][xPos + j],
-                             ef[0][j], ef[1][j]);
-    }
-    fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
-                   -aec->xfBuf[1][xPos + PART_LEN],
-                   ef[0][PART_LEN], ef[1][PART_LEN]);
-
-    aec_rdft_inverse_128(fft);
-    memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
-
-    // fft scaling
-    {
-      float scale = 2.0f / PART_LEN2;
-      for (j = 0; j < PART_LEN; j++) {
-        fft[j] *= scale;
-      }
-    }
-    aec_rdft_forward_128(fft);
-
-    aec->wfBuf[0][pos] += fft[0];
-    aec->wfBuf[0][pos + PART_LEN] += fft[1];
-
-    for (j = 1; j < PART_LEN; j++) {
-      aec->wfBuf[0][pos + j] += fft[2 * j];
-      aec->wfBuf[1][pos + j] += fft[2 * j + 1];
-    }
-#endif // UNCONSTR
-  }
-}
-
-static void OverdriveAndSuppress(aec_t *aec, float hNl[PART_LEN1],
-                                 const float hNlFb,
-                                 float efw[2][PART_LEN1]) {
-  int i;
-  for (i = 0; i < PART_LEN1; i++) {
-    // Weight subbands
-    if (hNl[i] > hNlFb) {
-      hNl[i] = WebRtcAec_weightCurve[i] * hNlFb +
-          (1 - WebRtcAec_weightCurve[i]) * hNl[i];
-    }
-    hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
-
-    // Suppress error signal
-    efw[0][i] *= hNl[i];
-    efw[1][i] *= hNl[i];
-
-    // Ooura fft returns incorrect sign on imaginary component. It matters here
-    // because we are making an additive change with comfort noise.
-    efw[1][i] *= -1;
-  }
-}
-
-WebRtcAec_FilterFar_t WebRtcAec_FilterFar;
-WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal;
-WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation;
-WebRtcAec_OverdriveAndSuppress_t WebRtcAec_OverdriveAndSuppress;
-
-int WebRtcAec_InitAec(aec_t *aec, int sampFreq)
-{
-    int i;
-
-    aec->sampFreq = sampFreq;
-
-    if (sampFreq == 8000) {
-        aec->mu = 0.6f;
-        aec->errThresh = 2e-6f;
-    }
-    else {
-        aec->mu = 0.5f;
-        aec->errThresh = 1.5e-6f;
-    }
-
-    if (WebRtcApm_InitBuffer(aec->farFrBuf) == -1) {
-        return -1;
-    }
-
-    if (WebRtcApm_InitBuffer(aec->nearFrBuf) == -1) {
-        return -1;
-    }
-
-    if (WebRtcApm_InitBuffer(aec->outFrBuf) == -1) {
-        return -1;
-    }
-
-    if (WebRtcApm_InitBuffer(aec->nearFrBufH) == -1) {
-        return -1;
-    }
-
-    if (WebRtcApm_InitBuffer(aec->outFrBufH) == -1) {
-        return -1;
-    }
-
-    // Default target suppression level
-    aec->targetSupp = -11.5;
-    aec->minOverDrive = 2.0;
-
-    // Sampling frequency multiplier
-    // SWB is processed as 160 frame size
-    if (aec->sampFreq == 32000) {
-      aec->mult = (short)aec->sampFreq / 16000;
-    }
-    else {
-        aec->mult = (short)aec->sampFreq / 8000;
-    }
-
-    aec->farBufWritePos = 0;
-    aec->farBufReadPos = 0;
-
-    aec->inSamples = 0;
-    aec->outSamples = 0;
-    aec->knownDelay = 0;
-
-    // Initialize buffers
-    memset(aec->farBuf, 0, sizeof(aec->farBuf));
-    memset(aec->xBuf, 0, sizeof(aec->xBuf));
-    memset(aec->dBuf, 0, sizeof(aec->dBuf));
-    memset(aec->eBuf, 0, sizeof(aec->eBuf));
-    // For H band
-    memset(aec->dBufH, 0, sizeof(aec->dBufH));
-
-    memset(aec->xPow, 0, sizeof(aec->xPow));
-    memset(aec->dPow, 0, sizeof(aec->dPow));
-    memset(aec->dInitMinPow, 0, sizeof(aec->dInitMinPow));
-    aec->noisePow = aec->dInitMinPow;
-    aec->noiseEstCtr = 0;
-
-    // Initial comfort noise power
-    for (i = 0; i < PART_LEN1; i++) {
-        aec->dMinPow[i] = 1.0e6f;
-    }
-
-    // Holds the last block written to
-    aec->xfBufBlockPos = 0;
-    // TODO: Investigate need for these initializations. Deleting them doesn't
-    //       change the output at all and yields 0.4% overall speedup.
-    memset(aec->xfBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
-    memset(aec->wfBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
-    memset(aec->sde, 0, sizeof(complex_t) * PART_LEN1);
-    memset(aec->sxd, 0, sizeof(complex_t) * PART_LEN1);
-    memset(aec->xfwBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
-    memset(aec->se, 0, sizeof(float) * PART_LEN1);
-
-    // To prevent numerical instability in the first block.
-    for (i = 0; i < PART_LEN1; i++) {
-        aec->sd[i] = 1;
-    }
-    for (i = 0; i < PART_LEN1; i++) {
-        aec->sx[i] = 1;
-    }
-
-    memset(aec->hNs, 0, sizeof(aec->hNs));
-    memset(aec->outBuf, 0, sizeof(float) * PART_LEN);
-
-    aec->hNlFbMin = 1;
-    aec->hNlFbLocalMin = 1;
-    aec->hNlXdAvgMin = 1;
-    aec->hNlNewMin = 0;
-    aec->hNlMinCtr = 0;
-    aec->overDrive = 2;
-    aec->overDriveSm = 2;
-    aec->delayIdx = 0;
-    aec->stNearState = 0;
-    aec->echoState = 0;
-    aec->divergeState = 0;
-
-    aec->seed = 777;
-    aec->delayEstCtr = 0;
-
-    // Features on by default (G.167)
-#ifdef G167
-    aec->adaptToggle = 1;
-    aec->nlpToggle = 1;
-    aec->cnToggle = 1;
-#endif
-
-    // Metrics disabled by default
-    aec->metricsMode = 0;
-    WebRtcAec_InitMetrics(aec);
-
-    // Assembly optimization
-    WebRtcAec_FilterFar = FilterFar;
-    WebRtcAec_ScaleErrorSignal = ScaleErrorSignal;
-    WebRtcAec_FilterAdaptation = FilterAdaptation;
-    WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress;
-    if (WebRtc_GetCPUInfo(kSSE2)) {
-#if defined(__SSE2__)
-      WebRtcAec_InitAec_SSE2();
-#endif
-    }
-    aec_rdft_init();
-
-    return 0;
-}
-
-void WebRtcAec_InitMetrics(aec_t *aec)
-{
-    aec->stateCounter = 0;
-    WebRtcAec_InitLevel(&aec->farlevel);
-    WebRtcAec_InitLevel(&aec->nearlevel);
-    WebRtcAec_InitLevel(&aec->linoutlevel);
-    WebRtcAec_InitLevel(&aec->nlpoutlevel);
-
-    WebRtcAec_InitStats(&aec->erl);
-    WebRtcAec_InitStats(&aec->erle);
-    WebRtcAec_InitStats(&aec->aNlp);
-    WebRtcAec_InitStats(&aec->rerl);
-}
-
-
-void WebRtcAec_ProcessFrame(aec_t *aec, const short *farend,
-                       const short *nearend, const short *nearendH,
-                       short *out, short *outH,
-                       int knownDelay)
-{
-    short farBl[PART_LEN], nearBl[PART_LEN], outBl[PART_LEN];
-    short farFr[FRAME_LEN];
-    // For H band
-    short nearBlH[PART_LEN], outBlH[PART_LEN];
-
-    int size = 0;
-
-    // initialize: only used for SWB
-    memset(nearBlH, 0, sizeof(nearBlH));
-    memset(outBlH, 0, sizeof(outBlH));
-
-    // Buffer the current frame.
-    // Fetch an older one corresponding to the delay.
-    BufferFar(aec, farend, FRAME_LEN);
-    FetchFar(aec, farFr, FRAME_LEN, knownDelay);
-
-    // Buffer the synchronized far and near frames,
-    // to pass the smaller blocks individually.
-    WebRtcApm_WriteBuffer(aec->farFrBuf, farFr, FRAME_LEN);
-    WebRtcApm_WriteBuffer(aec->nearFrBuf, nearend, FRAME_LEN);
-    // For H band
-    if (aec->sampFreq == 32000) {
-        WebRtcApm_WriteBuffer(aec->nearFrBufH, nearendH, FRAME_LEN);
-    }
-
-    // Process as many blocks as possible.
-    while (WebRtcApm_get_buffer_size(aec->farFrBuf) >= PART_LEN) {
-
-        WebRtcApm_ReadBuffer(aec->farFrBuf, farBl, PART_LEN);
-        WebRtcApm_ReadBuffer(aec->nearFrBuf, nearBl, PART_LEN);
-
-        // For H band
-        if (aec->sampFreq == 32000) {
-            WebRtcApm_ReadBuffer(aec->nearFrBufH, nearBlH, PART_LEN);
-        }
-
-        ProcessBlock(aec, farBl, nearBl, nearBlH, outBl, outBlH);
-
-        WebRtcApm_WriteBuffer(aec->outFrBuf, outBl, PART_LEN);
-        // For H band
-        if (aec->sampFreq == 32000) {
-            WebRtcApm_WriteBuffer(aec->outFrBufH, outBlH, PART_LEN);
-        }
-    }
-
-    // Stuff the out buffer if we have less than a frame to output.
-    // This should only happen for the first frame.
-    size = WebRtcApm_get_buffer_size(aec->outFrBuf);
-    if (size < FRAME_LEN) {
-        WebRtcApm_StuffBuffer(aec->outFrBuf, FRAME_LEN - size);
-        if (aec->sampFreq == 32000) {
-            WebRtcApm_StuffBuffer(aec->outFrBufH, FRAME_LEN - size);
-        }
-    }
-
-    // Obtain an output frame.
-    WebRtcApm_ReadBuffer(aec->outFrBuf, out, FRAME_LEN);
-    // For H band
-    if (aec->sampFreq == 32000) {
-        WebRtcApm_ReadBuffer(aec->outFrBufH, outH, FRAME_LEN);
-    }
-}
-
-static void ProcessBlock(aec_t *aec, const short *farend,
-                              const short *nearend, const short *nearendH,
-                              short *output, short *outputH)
-{
-    int i;
-    float d[PART_LEN], y[PART_LEN], e[PART_LEN], dH[PART_LEN];
-    short eInt16[PART_LEN];
-    float scale;
-
-    float fft[PART_LEN2];
-    float xf[2][PART_LEN1], yf[2][PART_LEN1], ef[2][PART_LEN1];
-    complex_t df[PART_LEN1];
-
-    const float gPow[2] = {0.9f, 0.1f};
-
-    // Noise estimate constants.
-    const int noiseInitBlocks = 500 * aec->mult;
-    const float step = 0.1f;
-    const float ramp = 1.0002f;
-    const float gInitNoise[2] = {0.999f, 0.001f};
-
-#ifdef AEC_DEBUG
-    fwrite(farend, sizeof(short), PART_LEN, aec->farFile);
-    fwrite(nearend, sizeof(short), PART_LEN, aec->nearFile);
-#endif
-
-    memset(dH, 0, sizeof(dH));
-
-    // ---------- Ooura fft ----------
-    // Concatenate old and new farend blocks.
-    for (i = 0; i < PART_LEN; i++) {
-        aec->xBuf[i + PART_LEN] = (float)farend[i];
-        d[i] = (float)nearend[i];
-    }
-
-    if (aec->sampFreq == 32000) {
-        for (i = 0; i < PART_LEN; i++) {
-            dH[i] = (float)nearendH[i];
-        }
-    }
-
-
-    memcpy(fft, aec->xBuf, sizeof(float) * PART_LEN2);
-    memcpy(aec->dBuf + PART_LEN, d, sizeof(float) * PART_LEN);
-    // For H band
-    if (aec->sampFreq == 32000) {
-        memcpy(aec->dBufH + PART_LEN, dH, sizeof(float) * PART_LEN);
-    }
-
-    aec_rdft_forward_128(fft);
-
-    // Far fft
-    xf[1][0] = 0;
-    xf[1][PART_LEN] = 0;
-    xf[0][0] = fft[0];
-    xf[0][PART_LEN] = fft[1];
-
-    for (i = 1; i < PART_LEN; i++) {
-        xf[0][i] = fft[2 * i];
-        xf[1][i] = fft[2 * i + 1];
-    }
-
-    // Near fft
-    memcpy(fft, aec->dBuf, sizeof(float) * PART_LEN2);
-    aec_rdft_forward_128(fft);
-    df[0][1] = 0;
-    df[PART_LEN][1] = 0;
-    df[0][0] = fft[0];
-    df[PART_LEN][0] = fft[1];
-
-    for (i = 1; i < PART_LEN; i++) {
-        df[i][0] = fft[2 * i];
-        df[i][1] = fft[2 * i + 1];
-    }
-
-    // Power smoothing
-    for (i = 0; i < PART_LEN1; i++) {
-        aec->xPow[i] = gPow[0] * aec->xPow[i] + gPow[1] * NR_PART *
-            (xf[0][i] * xf[0][i] + xf[1][i] * xf[1][i]);
-        aec->dPow[i] = gPow[0] * aec->dPow[i] + gPow[1] *
-            (df[i][0] * df[i][0] + df[i][1] * df[i][1]);
-    }
-
-    // Estimate noise power. Wait until dPow is more stable.
-    if (aec->noiseEstCtr > 50) {
-        for (i = 0; i < PART_LEN1; i++) {
-            if (aec->dPow[i] < aec->dMinPow[i]) {
-                aec->dMinPow[i] = (aec->dPow[i] + step * (aec->dMinPow[i] -
-                    aec->dPow[i])) * ramp;
-            }
-            else {
-                aec->dMinPow[i] *= ramp;
-            }
-        }
-    }
-
-    // Smooth increasing noise power from zero at the start,
-    // to avoid a sudden burst of comfort noise.
-    if (aec->noiseEstCtr < noiseInitBlocks) {
-        aec->noiseEstCtr++;
-        for (i = 0; i < PART_LEN1; i++) {
-            if (aec->dMinPow[i] > aec->dInitMinPow[i]) {
-                aec->dInitMinPow[i] = gInitNoise[0] * aec->dInitMinPow[i] +
-                    gInitNoise[1] * aec->dMinPow[i];
-            }
-            else {
-                aec->dInitMinPow[i] = aec->dMinPow[i];
-            }
-        }
-        aec->noisePow = aec->dInitMinPow;
-    }
-    else {
-        aec->noisePow = aec->dMinPow;
-    }
-
-
-    // Update the xfBuf block position.
-    aec->xfBufBlockPos--;
-    if (aec->xfBufBlockPos == -1) {
-        aec->xfBufBlockPos = NR_PART - 1;
-    }
-
-    // Buffer xf
-    memcpy(aec->xfBuf[0] + aec->xfBufBlockPos * PART_LEN1, xf[0],
-           sizeof(float) * PART_LEN1);
-    memcpy(aec->xfBuf[1] + aec->xfBufBlockPos * PART_LEN1, xf[1],
-           sizeof(float) * PART_LEN1);
-
-    memset(yf[0], 0, sizeof(float) * (PART_LEN1 * 2));
-
-    // Filter far
-    WebRtcAec_FilterFar(aec, yf);
-
-    // Inverse fft to obtain echo estimate and error.
-    fft[0] = yf[0][0];
-    fft[1] = yf[0][PART_LEN];
-    for (i = 1; i < PART_LEN; i++) {
-        fft[2 * i] = yf[0][i];
-        fft[2 * i + 1] = yf[1][i];
-    }
-    aec_rdft_inverse_128(fft);
-
-    scale = 2.0f / PART_LEN2;
-    for (i = 0; i < PART_LEN; i++) {
-        y[i] = fft[PART_LEN + i] * scale; // fft scaling
-    }
-
-    for (i = 0; i < PART_LEN; i++) {
-        e[i] = d[i] - y[i];
-    }
-
-    // Error fft
-    memcpy(aec->eBuf + PART_LEN, e, sizeof(float) * PART_LEN);
-    memset(fft, 0, sizeof(float) * PART_LEN);
-    memcpy(fft + PART_LEN, e, sizeof(float) * PART_LEN);
-    aec_rdft_forward_128(fft);
-
-    ef[1][0] = 0;
-    ef[1][PART_LEN] = 0;
-    ef[0][0] = fft[0];
-    ef[0][PART_LEN] = fft[1];
-    for (i = 1; i < PART_LEN; i++) {
-        ef[0][i] = fft[2 * i];
-        ef[1][i] = fft[2 * i + 1];
-    }
-
-    // Scale error signal inversely with far power.
-    WebRtcAec_ScaleErrorSignal(aec, ef);
-#ifdef G167
-    if (aec->adaptToggle) {
-#endif
-        // Filter adaptation
-        WebRtcAec_FilterAdaptation(aec, fft, ef);
-#ifdef G167
-    }
-#endif
-
-    NonLinearProcessing(aec, output, outputH);
-
-#if defined(AEC_DEBUG) || defined(G167)
-    for (i = 0; i < PART_LEN; i++) {
-        eInt16[i] = (short)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, e[i],
-            WEBRTC_SPL_WORD16_MIN);
-    }
-#endif
-#ifdef G167
-    if (aec->nlpToggle == 0) {
-        memcpy(output, eInt16, sizeof(eInt16));
-    }
-#endif
-
-    if (aec->metricsMode == 1) {
-        for (i = 0; i < PART_LEN; i++) {
-            eInt16[i] = (short)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, e[i],
-                WEBRTC_SPL_WORD16_MIN);
-        }
-
-        // Update power levels and echo metrics
-        UpdateLevel(&aec->farlevel, farend);
-        UpdateLevel(&aec->nearlevel, nearend);
-        UpdateLevel(&aec->linoutlevel, eInt16);
-        UpdateLevel(&aec->nlpoutlevel, output);
-        UpdateMetrics(aec);
-    }
-
-#ifdef AEC_DEBUG
-    fwrite(eInt16, sizeof(short), PART_LEN, aec->outLpFile);
-    fwrite(output, sizeof(short), PART_LEN, aec->outFile);
-#endif
-}
-
-static void NonLinearProcessing(aec_t *aec, short *output, short *outputH)
-{
-    float efw[2][PART_LEN1], dfw[2][PART_LEN1];
-    complex_t xfw[PART_LEN1];
-    complex_t comfortNoiseHband[PART_LEN1];
-    float fft[PART_LEN2];
-    float scale, dtmp;
-    float nlpGainHband;
-    int i, j, pos;
-
-    // Coherence and non-linear filter
-    float cohde[PART_LEN1], cohxd[PART_LEN1];
-    float hNlDeAvg, hNlXdAvg;
-    float hNl[PART_LEN1];
-    float hNlPref[PREF_BAND_SIZE];
-    float hNlFb = 0, hNlFbLow = 0;
-    const float prefBandQuant = 0.75f, prefBandQuantLow = 0.5f;
-    const int prefBandSize = PREF_BAND_SIZE / aec->mult;
-    const int minPrefBand = 4 / aec->mult;
-
-    // Near and error power sums
-    float sdSum = 0, seSum = 0;
-
-    // Power estimate smoothing coefficients
-    const float gCoh[2][2] = {{0.9f, 0.1f}, {0.93f, 0.07f}};
-    const float *ptrGCoh = gCoh[aec->mult - 1];
-
-    // Filter energey
-    float wfEnMax = 0, wfEn = 0;
-    const int delayEstInterval = 10 * aec->mult;
-
-    aec->delayEstCtr++;
-    if (aec->delayEstCtr == delayEstInterval) {
-        aec->delayEstCtr = 0;
-    }
-
-    // initialize comfort noise for H band
-    memset(comfortNoiseHband, 0, sizeof(comfortNoiseHband));
-    nlpGainHband = (float)0.0;
-    dtmp = (float)0.0;
-
-    // Measure energy in each filter partition to determine delay.
-    // TODO: Spread by computing one partition per block?
-    if (aec->delayEstCtr == 0) {
-        wfEnMax = 0;
-        aec->delayIdx = 0;
-        for (i = 0; i < NR_PART; i++) {
-            pos = i * PART_LEN1;
-            wfEn = 0;
-            for (j = 0; j < PART_LEN1; j++) {
-                wfEn += aec->wfBuf[0][pos + j] * aec->wfBuf[0][pos + j] +
-                    aec->wfBuf[1][pos + j] * aec->wfBuf[1][pos + j];
-            }
-
-            if (wfEn > wfEnMax) {
-                wfEnMax = wfEn;
-                aec->delayIdx = i;
-            }
-        }
-    }
-
-    // NLP
-    // Windowed far fft
-    for (i = 0; i < PART_LEN; i++) {
-        fft[i] = aec->xBuf[i] * sqrtHanning[i];
-        fft[PART_LEN + i] = aec->xBuf[PART_LEN + i] * sqrtHanning[PART_LEN - i];
-    }
-    aec_rdft_forward_128(fft);
-
-    xfw[0][1] = 0;
-    xfw[PART_LEN][1] = 0;
-    xfw[0][0] = fft[0];
-    xfw[PART_LEN][0] = fft[1];
-    for (i = 1; i < PART_LEN; i++) {
-        xfw[i][0] = fft[2 * i];
-        xfw[i][1] = fft[2 * i + 1];
-    }
-
-    // Buffer far.
-    memcpy(aec->xfwBuf, xfw, sizeof(xfw));
-
-    // Use delayed far.
-    memcpy(xfw, aec->xfwBuf + aec->delayIdx * PART_LEN1, sizeof(xfw));
-
-    // Windowed near fft
-    for (i = 0; i < PART_LEN; i++) {
-        fft[i] = aec->dBuf[i] * sqrtHanning[i];
-        fft[PART_LEN + i] = aec->dBuf[PART_LEN + i] * sqrtHanning[PART_LEN - i];
-    }
-    aec_rdft_forward_128(fft);
-
-    dfw[1][0] = 0;
-    dfw[1][PART_LEN] = 0;
-    dfw[0][0] = fft[0];
-    dfw[0][PART_LEN] = fft[1];
-    for (i = 1; i < PART_LEN; i++) {
-        dfw[0][i] = fft[2 * i];
-        dfw[1][i] = fft[2 * i + 1];
-    }
-
-    // Windowed error fft
-    for (i = 0; i < PART_LEN; i++) {
-        fft[i] = aec->eBuf[i] * sqrtHanning[i];
-        fft[PART_LEN + i] = aec->eBuf[PART_LEN + i] * sqrtHanning[PART_LEN - i];
-    }
-    aec_rdft_forward_128(fft);
-    efw[1][0] = 0;
-    efw[1][PART_LEN] = 0;
-    efw[0][0] = fft[0];
-    efw[0][PART_LEN] = fft[1];
-    for (i = 1; i < PART_LEN; i++) {
-        efw[0][i] = fft[2 * i];
-        efw[1][i] = fft[2 * i + 1];
-    }
-
-    // Smoothed PSD
-    for (i = 0; i < PART_LEN1; i++) {
-        aec->sd[i] = ptrGCoh[0] * aec->sd[i] + ptrGCoh[1] *
-            (dfw[0][i] * dfw[0][i] + dfw[1][i] * dfw[1][i]);
-        aec->se[i] = ptrGCoh[0] * aec->se[i] + ptrGCoh[1] *
-            (efw[0][i] * efw[0][i] + efw[1][i] * efw[1][i]);
-        // We threshold here to protect against the ill-effects of a zero farend.
-        // The threshold is not arbitrarily chosen, but balances protection and
-        // adverse interaction with the algorithm's tuning.
-        // TODO: investigate further why this is so sensitive.
-        aec->sx[i] = ptrGCoh[0] * aec->sx[i] + ptrGCoh[1] *
-            WEBRTC_SPL_MAX(xfw[i][0] * xfw[i][0] + xfw[i][1] * xfw[i][1], 15);
-
-        aec->sde[i][0] = ptrGCoh[0] * aec->sde[i][0] + ptrGCoh[1] *
-            (dfw[0][i] * efw[0][i] + dfw[1][i] * efw[1][i]);
-        aec->sde[i][1] = ptrGCoh[0] * aec->sde[i][1] + ptrGCoh[1] *
-            (dfw[0][i] * efw[1][i] - dfw[1][i] * efw[0][i]);
-
-        aec->sxd[i][0] = ptrGCoh[0] * aec->sxd[i][0] + ptrGCoh[1] *
-            (dfw[0][i] * xfw[i][0] + dfw[1][i] * xfw[i][1]);
-        aec->sxd[i][1] = ptrGCoh[0] * aec->sxd[i][1] + ptrGCoh[1] *
-            (dfw[0][i] * xfw[i][1] - dfw[1][i] * xfw[i][0]);
-
-        sdSum += aec->sd[i];
-        seSum += aec->se[i];
-    }
-
-    // Divergent filter safeguard.
-    if (aec->divergeState == 0) {
-        if (seSum > sdSum) {
-            aec->divergeState = 1;
-        }
-    }
-    else {
-        if (seSum * 1.05f < sdSum) {
-            aec->divergeState = 0;
-        }
-    }
-
-    if (aec->divergeState == 1) {
-        memcpy(efw, dfw, sizeof(efw));
-    }
-
-    // Reset if error is significantly larger than nearend (13 dB).
-    if (seSum > (19.95f * sdSum)) {
-        memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
-    }
-
-    // Subband coherence
-    for (i = 0; i < PART_LEN1; i++) {
-        cohde[i] = (aec->sde[i][0] * aec->sde[i][0] + aec->sde[i][1] * aec->sde[i][1]) /
-            (aec->sd[i] * aec->se[i] + 1e-10f);
-        cohxd[i] = (aec->sxd[i][0] * aec->sxd[i][0] + aec->sxd[i][1] * aec->sxd[i][1]) /
-            (aec->sx[i] * aec->sd[i] + 1e-10f);
-    }
-
-    hNlXdAvg = 0;
-    for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
-        hNlXdAvg += cohxd[i];
-    }
-    hNlXdAvg /= prefBandSize;
-    hNlXdAvg = 1 - hNlXdAvg;
-
-    hNlDeAvg = 0;
-    for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
-        hNlDeAvg += cohde[i];
-    }
-    hNlDeAvg /= prefBandSize;
-
-    if (hNlXdAvg < 0.75f && hNlXdAvg < aec->hNlXdAvgMin) {
-        aec->hNlXdAvgMin = hNlXdAvg;
-    }
-
-    if (hNlDeAvg > 0.98f && hNlXdAvg > 0.9f) {
-        aec->stNearState = 1;
-    }
-    else if (hNlDeAvg < 0.95f || hNlXdAvg < 0.8f) {
-        aec->stNearState = 0;
-    }
-
-    if (aec->hNlXdAvgMin == 1) {
-        aec->echoState = 0;
-        aec->overDrive = aec->minOverDrive;
-
-        if (aec->stNearState == 1) {
-            memcpy(hNl, cohde, sizeof(hNl));
-            hNlFb = hNlDeAvg;
-            hNlFbLow = hNlDeAvg;
-        }
-        else {
-            for (i = 0; i < PART_LEN1; i++) {
-                hNl[i] = 1 - cohxd[i];
-            }
-            hNlFb = hNlXdAvg;
-            hNlFbLow = hNlXdAvg;
-        }
-    }
-    else {
-
-        if (aec->stNearState == 1) {
-            aec->echoState = 0;
-            memcpy(hNl, cohde, sizeof(hNl));
-            hNlFb = hNlDeAvg;
-            hNlFbLow = hNlDeAvg;
-        }
-        else {
-            aec->echoState = 1;
-            for (i = 0; i < PART_LEN1; i++) {
-                hNl[i] = WEBRTC_SPL_MIN(cohde[i], 1 - cohxd[i]);
-            }
-
-            // Select an order statistic from the preferred bands.
-            // TODO: Using quicksort now, but a selection algorithm may be preferred.
-            memcpy(hNlPref, &hNl[minPrefBand], sizeof(float) * prefBandSize);
-            qsort(hNlPref, prefBandSize, sizeof(float), CmpFloat);
-            hNlFb = hNlPref[(int)floor(prefBandQuant * (prefBandSize - 1))];
-            hNlFbLow = hNlPref[(int)floor(prefBandQuantLow * (prefBandSize - 1))];
-        }
-    }
-
-    // Track the local filter minimum to determine suppression overdrive.
-    if (hNlFbLow < 0.6f && hNlFbLow < aec->hNlFbLocalMin) {
-        aec->hNlFbLocalMin = hNlFbLow;
-        aec->hNlFbMin = hNlFbLow;
-        aec->hNlNewMin = 1;
-        aec->hNlMinCtr = 0;
-    }
-    aec->hNlFbLocalMin = WEBRTC_SPL_MIN(aec->hNlFbLocalMin + 0.0008f / aec->mult, 1);
-    aec->hNlXdAvgMin = WEBRTC_SPL_MIN(aec->hNlXdAvgMin + 0.0006f / aec->mult, 1);
-
-    if (aec->hNlNewMin == 1) {
-        aec->hNlMinCtr++;
-    }
-    if (aec->hNlMinCtr == 2) {
-        aec->hNlNewMin = 0;
-        aec->hNlMinCtr = 0;
-        aec->overDrive = WEBRTC_SPL_MAX(aec->targetSupp /
-            ((float)log(aec->hNlFbMin + 1e-10f) + 1e-10f), aec->minOverDrive);
-    }
-
-    // Smooth the overdrive.
-    if (aec->overDrive < aec->overDriveSm) {
-      aec->overDriveSm = 0.99f * aec->overDriveSm + 0.01f * aec->overDrive;
-    }
-    else {
-      aec->overDriveSm = 0.9f * aec->overDriveSm + 0.1f * aec->overDrive;
-    }
-
-    WebRtcAec_OverdriveAndSuppress(aec, hNl, hNlFb, efw);
-
-#ifdef G167
-    if (aec->cnToggle) {
-      ComfortNoise(aec, efw, comfortNoiseHband, aec->noisePow, hNl);
-    }
-#else
-    // Add comfort noise.
-    ComfortNoise(aec, efw, comfortNoiseHband, aec->noisePow, hNl);
-#endif
-
-    // Inverse error fft.
-    fft[0] = efw[0][0];
-    fft[1] = efw[0][PART_LEN];
-    for (i = 1; i < PART_LEN; i++) {
-        fft[2*i] = efw[0][i];
-        // Sign change required by Ooura fft.
-        fft[2*i + 1] = -efw[1][i];
-    }
-    aec_rdft_inverse_128(fft);
-
-    // Overlap and add to obtain output.
-    scale = 2.0f / PART_LEN2;
-    for (i = 0; i < PART_LEN; i++) {
-        fft[i] *= scale; // fft scaling
-        fft[i] = fft[i]*sqrtHanning[i] + aec->outBuf[i];
-
-        // Saturation protection
-        output[i] = (short)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, fft[i],
-            WEBRTC_SPL_WORD16_MIN);
-
-        fft[PART_LEN + i] *= scale; // fft scaling
-        aec->outBuf[i] = fft[PART_LEN + i] * sqrtHanning[PART_LEN - i];
-    }
-
-    // For H band
-    if (aec->sampFreq == 32000) {
-
-        // H band gain
-        // average nlp over low band: average over second half of freq spectrum
-        // (4->8khz)
-        GetHighbandGain(hNl, &nlpGainHband);
-
-        // Inverse comfort_noise
-        if (flagHbandCn == 1) {
-            fft[0] = comfortNoiseHband[0][0];
-            fft[1] = comfortNoiseHband[PART_LEN][0];
-            for (i = 1; i < PART_LEN; i++) {
-                fft[2*i] = comfortNoiseHband[i][0];
-                fft[2*i + 1] = comfortNoiseHband[i][1];
-            }
-            aec_rdft_inverse_128(fft);
-            scale = 2.0f / PART_LEN2;
-        }
-
-        // compute gain factor
-        for (i = 0; i < PART_LEN; i++) {
-            dtmp = (float)aec->dBufH[i];
-            dtmp = (float)dtmp * nlpGainHband; // for variable gain
-
-            // add some comfort noise where Hband is attenuated
-            if (flagHbandCn == 1) {
-                fft[i] *= scale; // fft scaling
-                dtmp += cnScaleHband * fft[i];
-            }
-
-            // Saturation protection
-            outputH[i] = (short)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, dtmp,
-                WEBRTC_SPL_WORD16_MIN);
-         }
-    }
-
-    // Copy the current block to the old position.
-    memcpy(aec->xBuf, aec->xBuf + PART_LEN, sizeof(float) * PART_LEN);
-    memcpy(aec->dBuf, aec->dBuf + PART_LEN, sizeof(float) * PART_LEN);
-    memcpy(aec->eBuf, aec->eBuf + PART_LEN, sizeof(float) * PART_LEN);
-
-    // Copy the current block to the old position for H band
-    if (aec->sampFreq == 32000) {
-        memcpy(aec->dBufH, aec->dBufH + PART_LEN, sizeof(float) * PART_LEN);
-    }
-
-    memmove(aec->xfwBuf + PART_LEN1, aec->xfwBuf, sizeof(aec->xfwBuf) -
-        sizeof(complex_t) * PART_LEN1);
-}
-
-static void GetHighbandGain(const float *lambda, float *nlpGainHband)
-{
-    int i;
-
-    nlpGainHband[0] = (float)0.0;
-    for (i = freqAvgIc; i < PART_LEN1 - 1; i++) {
-        nlpGainHband[0] += lambda[i];
-    }
-    nlpGainHband[0] /= (float)(PART_LEN1 - 1 - freqAvgIc);
-}
-
-static void ComfortNoise(aec_t *aec, float efw[2][PART_LEN1],
-    complex_t *comfortNoiseHband, const float *noisePow, const float *lambda)
-{
-    int i, num;
-    float rand[PART_LEN];
-    float noise, noiseAvg, tmp, tmpAvg;
-    WebRtc_Word16 randW16[PART_LEN];
-    complex_t u[PART_LEN1];
-
-    const float pi2 = 6.28318530717959f;
-
-    // Generate a uniform random array on [0 1]
-    WebRtcSpl_RandUArray(randW16, PART_LEN, &aec->seed);
-    for (i = 0; i < PART_LEN; i++) {
-        rand[i] = ((float)randW16[i]) / 32768;
-    }
-
-    // Reject LF noise
-    u[0][0] = 0;
-    u[0][1] = 0;
-    for (i = 1; i < PART_LEN1; i++) {
-        tmp = pi2 * rand[i - 1];
-
-        noise = sqrtf(noisePow[i]);
-        u[i][0] = noise * (float)cos(tmp);
-        u[i][1] = -noise * (float)sin(tmp);
-    }
-    u[PART_LEN][1] = 0;
-
-    for (i = 0; i < PART_LEN1; i++) {
-        // This is the proper weighting to match the background noise power
-        tmp = sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
-        //tmp = 1 - lambda[i];
-        efw[0][i] += tmp * u[i][0];
-        efw[1][i] += tmp * u[i][1];
-    }
-
-    // For H band comfort noise
-    // TODO: don't compute noise and "tmp" twice. Use the previous results.
-    noiseAvg = 0.0;
-    tmpAvg = 0.0;
-    num = 0;
-    if (aec->sampFreq == 32000 && flagHbandCn == 1) {
-
-        // average noise scale
-        // average over second half of freq spectrum (i.e., 4->8khz)
-        // TODO: we shouldn't need num. We know how many elements we're summing.
-        for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
-            num++;
-            noiseAvg += sqrtf(noisePow[i]);
-        }
-        noiseAvg /= (float)num;
-
-        // average nlp scale
-        // average over second half of freq spectrum (i.e., 4->8khz)
-        // TODO: we shouldn't need num. We know how many elements we're summing.
-        num = 0;
-        for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
-            num++;
-            tmpAvg += sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
-        }
-        tmpAvg /= (float)num;
-
-        // Use average noise for H band
-        // TODO: we should probably have a new random vector here.
-        // Reject LF noise
-        u[0][0] = 0;
-        u[0][1] = 0;
-        for (i = 1; i < PART_LEN1; i++) {
-            tmp = pi2 * rand[i - 1];
-
-            // Use average noise for H band
-            u[i][0] = noiseAvg * (float)cos(tmp);
-            u[i][1] = -noiseAvg * (float)sin(tmp);
-        }
-        u[PART_LEN][1] = 0;
-
-        for (i = 0; i < PART_LEN1; i++) {
-            // Use average NLP weight for H band
-            comfortNoiseHband[i][0] = tmpAvg * u[i][0];
-            comfortNoiseHband[i][1] = tmpAvg * u[i][1];
-        }
-    }
-}
-
-// Buffer the farend to account for knownDelay
-static void BufferFar(aec_t *aec, const short *farend, int farLen)
-{
-    int writeLen = farLen, writePos = 0;
-
-    // Check if the write position must be wrapped.
-    while (aec->farBufWritePos + writeLen > FAR_BUF_LEN) {
-
-        // Write to remaining buffer space before wrapping.
-        writeLen = FAR_BUF_LEN - aec->farBufWritePos;
-        memcpy(aec->farBuf + aec->farBufWritePos, farend + writePos,
-            sizeof(short) * writeLen);
-        aec->farBufWritePos = 0;
-        writePos = writeLen;
-        writeLen = farLen - writeLen;
-    }
-
-    memcpy(aec->farBuf + aec->farBufWritePos, farend + writePos,
-        sizeof(short) * writeLen);
-    aec->farBufWritePos +=  writeLen;
-}
-
-static void FetchFar(aec_t *aec, short *farend, int farLen, int knownDelay)
-{
-    int readLen = farLen, readPos = 0, delayChange = knownDelay - aec->knownDelay;
-
-    aec->farBufReadPos -= delayChange;
-
-    // Check if delay forces a read position wrap.
-    while(aec->farBufReadPos < 0) {
-        aec->farBufReadPos += FAR_BUF_LEN;
-    }
-    while(aec->farBufReadPos > FAR_BUF_LEN - 1) {
-        aec->farBufReadPos -= FAR_BUF_LEN;
-    }
-
-    aec->knownDelay = knownDelay;
-
-    // Check if read position must be wrapped.
-    while (aec->farBufReadPos + readLen > FAR_BUF_LEN) {
-
-        // Read from remaining buffer space before wrapping.
-        readLen = FAR_BUF_LEN - aec->farBufReadPos;
-        memcpy(farend + readPos, aec->farBuf + aec->farBufReadPos,
-            sizeof(short) * readLen);
-        aec->farBufReadPos = 0;
-        readPos = readLen;
-        readLen = farLen - readLen;
-    }
-    memcpy(farend + readPos, aec->farBuf + aec->farBufReadPos,
-        sizeof(short) * readLen);
-    aec->farBufReadPos += readLen;
-}
-
-static void WebRtcAec_InitLevel(power_level_t *level)
-{
-    const float bigFloat = 1E17f;
-
-    level->averagelevel = 0;
-    level->framelevel = 0;
-    level->minlevel = bigFloat;
-    level->frsum = 0;
-    level->sfrsum = 0;
-    level->frcounter = 0;
-    level->sfrcounter = 0;
-}
-
-static void WebRtcAec_InitStats(stats_t *stats)
-{
-    stats->instant = offsetLevel;
-    stats->average = offsetLevel;
-    stats->max = offsetLevel;
-    stats->min = offsetLevel * (-1);
-    stats->sum = 0;
-    stats->hisum = 0;
-    stats->himean = offsetLevel;
-    stats->counter = 0;
-    stats->hicounter = 0;
-}
-
-static void UpdateLevel(power_level_t *level, const short *in)
-{
-    int k;
-
-    for (k = 0; k < PART_LEN; k++) {
-        level->sfrsum += in[k] * in[k];
-    }
-    level->sfrcounter++;
-
-    if (level->sfrcounter > subCountLen) {
-        level->framelevel = level->sfrsum / (subCountLen * PART_LEN);
-        level->sfrsum = 0;
-        level->sfrcounter = 0;
-
-        if (level->framelevel > 0) {
-            if (level->framelevel < level->minlevel) {
-                level->minlevel = level->framelevel;     // New minimum
-            } else {
-                level->minlevel *= (1 + 0.001f);   // Small increase
-            }
-        }
-        level->frcounter++;
-        level->frsum += level->framelevel;
-
-        if (level->frcounter > countLen) {
-            level->averagelevel =  level->frsum / countLen;
-            level->frsum = 0;
-            level->frcounter = 0;
-        }
-
-    }
-}
-
-static void UpdateMetrics(aec_t *aec)
-{
-    float dtmp, dtmp2, dtmp3;
-
-    const float actThresholdNoisy = 8.0f;
-    const float actThresholdClean = 40.0f;
-    const float safety = 0.99995f;
-    const float noisyPower = 300000.0f;
-
-    float actThreshold;
-    float echo, suppressedEcho;
-
-    if (aec->echoState) {   // Check if echo is likely present
-        aec->stateCounter++;
-    }
-
-    if (aec->farlevel.frcounter == countLen) {
-
-        if (aec->farlevel.minlevel < noisyPower) {
-            actThreshold = actThresholdClean;
-        }
-        else {
-            actThreshold = actThresholdNoisy;
-        }
-
-        if ((aec->stateCounter > (0.5f * countLen * subCountLen))
-            && (aec->farlevel.sfrcounter == 0)
-
-            // Estimate in active far-end segments only
-            && (aec->farlevel.averagelevel > (actThreshold * aec->farlevel.minlevel))
-            ) {
-
-            // Subtract noise power
-            echo = aec->nearlevel.averagelevel - safety * aec->nearlevel.minlevel;
-
-            // ERL
-            dtmp = 10 * (float)log10(aec->farlevel.averagelevel /
-                aec->nearlevel.averagelevel + 1e-10f);
-            dtmp2 = 10 * (float)log10(aec->farlevel.averagelevel / echo + 1e-10f);
-
-            aec->erl.instant = dtmp;
-            if (dtmp > aec->erl.max) {
-                aec->erl.max = dtmp;
-            }
-
-            if (dtmp < aec->erl.min) {
-                aec->erl.min = dtmp;
-            }
-
-            aec->erl.counter++;
-            aec->erl.sum += dtmp;
-            aec->erl.average = aec->erl.sum / aec->erl.counter;
-
-            // Upper mean
-            if (dtmp > aec->erl.average) {
-                aec->erl.hicounter++;
-                aec->erl.hisum += dtmp;
-                aec->erl.himean = aec->erl.hisum / aec->erl.hicounter;
-            }
-
-            // A_NLP
-            dtmp = 10 * (float)log10(aec->nearlevel.averagelevel /
-                aec->linoutlevel.averagelevel + 1e-10f);
-
-            // subtract noise power
-            suppressedEcho = aec->linoutlevel.averagelevel - safety * aec->linoutlevel.minlevel;
-
-            dtmp2 = 10 * (float)log10(echo / suppressedEcho + 1e-10f);
-            dtmp3 = 10 * (float)log10(aec->nearlevel.averagelevel / suppressedEcho + 1e-10f);
-
-            aec->aNlp.instant = dtmp2;
-            if (dtmp > aec->aNlp.max) {
-                aec->aNlp.max = dtmp;
-            }
-
-            if (dtmp < aec->aNlp.min) {
-                aec->aNlp.min = dtmp;
-            }
-
-            aec->aNlp.counter++;
-            aec->aNlp.sum += dtmp;
-            aec->aNlp.average = aec->aNlp.sum / aec->aNlp.counter;
-
-            // Upper mean
-            if (dtmp > aec->aNlp.average) {
-                aec->aNlp.hicounter++;
-                aec->aNlp.hisum += dtmp;
-                aec->aNlp.himean = aec->aNlp.hisum / aec->aNlp.hicounter;
-            }
-
-            // ERLE
-
-            // subtract noise power
-            suppressedEcho = aec->nlpoutlevel.averagelevel - safety * aec->nlpoutlevel.minlevel;
-
-            dtmp = 10 * (float)log10(aec->nearlevel.averagelevel /
-                aec->nlpoutlevel.averagelevel + 1e-10f);
-            dtmp2 = 10 * (float)log10(echo / suppressedEcho + 1e-10f);
-
-            dtmp = dtmp2;
-            aec->erle.instant = dtmp;
-            if (dtmp > aec->erle.max) {
-                aec->erle.max = dtmp;
-            }
-
-            if (dtmp < aec->erle.min) {
-                aec->erle.min = dtmp;
-            }
-
-            aec->erle.counter++;
-            aec->erle.sum += dtmp;
-            aec->erle.average = aec->erle.sum / aec->erle.counter;
-
-            // Upper mean
-            if (dtmp > aec->erle.average) {
-                aec->erle.hicounter++;
-                aec->erle.hisum += dtmp;
-                aec->erle.himean = aec->erle.hisum / aec->erle.hicounter;
-            }
-        }
-
-        aec->stateCounter = 0;
-    }
-}
-
diff --git a/src/modules/audio_processing/aec/main/source/aec_core.h b/src/modules/audio_processing/aec/main/source/aec_core.h
deleted file mode 100644
index 3386b92..0000000
--- a/src/modules/audio_processing/aec/main/source/aec_core.h
+++ /dev/null
@@ -1,193 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Specifies the interface for the AEC core.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
-
-#include <stdio.h>
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-//#define G167 // for running G167 tests
-//#define UNCONSTR // time-unconstrained filter
-//#define AEC_DEBUG // for recording files
-
-#define FRAME_LEN 80
-#define PART_LEN 64 // Length of partition
-#define PART_LEN1 (PART_LEN + 1) // Unique fft coefficients
-#define PART_LEN2 (PART_LEN * 2) // Length of partition * 2
-#define NR_PART 12 // Number of partitions
-#define FILT_LEN (PART_LEN * NR_PART) // Filter length
-#define FILT_LEN2 (FILT_LEN * 2) // Double filter length
-#define FAR_BUF_LEN (FILT_LEN2 * 2)
-#define PREF_BAND_SIZE 24
-
-#define BLOCKL_MAX FRAME_LEN
-
-typedef float complex_t[2];
-// For performance reasons, some arrays of complex numbers are replaced by twice
-// as long arrays of float, all the real parts followed by all the imaginary
-// ones (complex_t[SIZE] -> float[2][SIZE]). This allows SIMD optimizations and
-// is better than two arrays (one for the real parts and one for the imaginary
-// parts) as this other way would require two pointers instead of one and cause
-// extra register spilling. This also allows the offsets to be calculated at
-// compile time.
-
-// Metrics
-enum {offsetLevel = -100};
-
-typedef struct {
-    float sfrsum;
-    int sfrcounter;
-    float framelevel;
-    float frsum;
-    int frcounter;
-    float minlevel;
-    float averagelevel;
-} power_level_t;
-
-typedef struct {
-    float instant;
-    float average;
-    float min;
-    float max;
-    float sum;
-    float hisum;
-    float himean;
-    int counter;
-    int hicounter;
-} stats_t;
-
-typedef struct {
-    int farBufWritePos, farBufReadPos;
-
-    int knownDelay;
-    int inSamples, outSamples;
-    int delayEstCtr;
-
-    void *farFrBuf, *nearFrBuf, *outFrBuf;
-
-    void *nearFrBufH;
-    void *outFrBufH;
-
-    float xBuf[PART_LEN2]; // farend
-    float dBuf[PART_LEN2]; // nearend
-    float eBuf[PART_LEN2]; // error
-
-    float dBufH[PART_LEN2]; // nearend
-
-    float xPow[PART_LEN1];
-    float dPow[PART_LEN1];
-    float dMinPow[PART_LEN1];
-    float dInitMinPow[PART_LEN1];
-    float *noisePow;
-#ifdef FFTW
-    float fftR[PART_LEN2];
-    fftw_complex fftC[PART_LEN2];
-    fftw_plan fftPlan, ifftPlan;
-
-    fftw_complex xfBuf[NR_PART * PART_LEN1];
-    fftw_complex wfBuf[NR_PART * PART_LEN1];
-    fftw_complex sde[PART_LEN1];
-#else
-    float xfBuf[2][NR_PART * PART_LEN1]; // farend fft buffer
-    float wfBuf[2][NR_PART * PART_LEN1]; // filter fft
-    complex_t sde[PART_LEN1]; // cross-psd of nearend and error
-    complex_t sxd[PART_LEN1]; // cross-psd of farend and nearend
-    complex_t xfwBuf[NR_PART * PART_LEN1]; // farend windowed fft buffer
-#endif
-    float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1]; // far, near and error psd
-    float hNs[PART_LEN1];
-    float hNlFbMin, hNlFbLocalMin;
-    float hNlXdAvgMin;
-    int hNlNewMin, hNlMinCtr;
-    float overDrive, overDriveSm;
-    float targetSupp, minOverDrive;
-    float outBuf[PART_LEN];
-    int delayIdx;
-
-    short stNearState, echoState;
-    short divergeState;
-
-    int xfBufBlockPos;
-
-    short farBuf[FILT_LEN2 * 2];
-
-    short mult; // sampling frequency multiple
-    int sampFreq;
-    WebRtc_UWord32 seed;
-
-    float mu; // stepsize
-    float errThresh; // error threshold
-
-    int noiseEstCtr;
-
-    // Toggles for G.167 testing
-#ifdef G167
-    short adaptToggle;  // Filter adaptation
-    short nlpToggle;    // Nonlinear processing
-    short cnToggle;     // Comfort noise
-#endif
-
-    power_level_t farlevel;
-    power_level_t nearlevel;
-    power_level_t linoutlevel;
-    power_level_t nlpoutlevel;
-
-    int metricsMode;
-    int stateCounter;
-    stats_t erl;
-    stats_t erle;
-    stats_t aNlp;
-    stats_t rerl;
-
-    // Quantities to control H band scaling for SWB input
-    int freq_avg_ic;         //initial bin for averaging nlp gain
-    int flag_Hband_cn;      //for comfort noise
-    float cn_scale_Hband;   //scale for comfort noise in H band
-
-#ifdef AEC_DEBUG
-    FILE *farFile;
-    FILE *nearFile;
-    FILE *outFile;
-    FILE *outLpFile;
-#endif
-} aec_t;
-
-typedef void (*WebRtcAec_FilterFar_t)(aec_t *aec, float yf[2][PART_LEN1]);
-extern WebRtcAec_FilterFar_t WebRtcAec_FilterFar;
-typedef void (*WebRtcAec_ScaleErrorSignal_t)(aec_t *aec, float ef[2][PART_LEN1]);
-extern WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal;
-#define IP_LEN PART_LEN // this must be at least ceil(2 + sqrt(PART_LEN))
-#define W_LEN PART_LEN
-typedef void (*WebRtcAec_FilterAdaptation_t)
-  (aec_t *aec, float *fft, float ef[2][PART_LEN1]);
-extern WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation;
-typedef void (*WebRtcAec_OverdriveAndSuppress_t)
-  (aec_t *aec, float hNl[PART_LEN1], const float hNlFb, float efw[2][PART_LEN1]);
-extern WebRtcAec_OverdriveAndSuppress_t WebRtcAec_OverdriveAndSuppress;
-
-int WebRtcAec_CreateAec(aec_t **aec);
-int WebRtcAec_FreeAec(aec_t *aec);
-int WebRtcAec_InitAec(aec_t *aec, int sampFreq);
-void WebRtcAec_InitAec_SSE2(void);
-
-void WebRtcAec_InitMetrics(aec_t *aec);
-void WebRtcAec_ProcessFrame(aec_t *aec, const short *farend,
-                       const short *nearend, const short *nearendH,
-                       short *out, short *outH,
-                       int knownDelay);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
-
diff --git a/src/modules/audio_processing/aec/main/source/aec_core_sse2.c b/src/modules/audio_processing/aec/main/source/aec_core_sse2.c
deleted file mode 100644
index 524669f..0000000
--- a/src/modules/audio_processing/aec/main/source/aec_core_sse2.c
+++ /dev/null
@@ -1,435 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * The core AEC algorithm, SSE2 version of speed-critical functions.
- */
-
-#if defined(__SSE2__)
-#include <emmintrin.h>
-#include <math.h>
-
-#include "aec_core.h"
-#include "aec_rdft.h"
-
-__inline static float MulRe(float aRe, float aIm, float bRe, float bIm)
-{
-  return aRe * bRe - aIm * bIm;
-}
-
-__inline static float MulIm(float aRe, float aIm, float bRe, float bIm)
-{
-  return aRe * bIm + aIm * bRe;
-}
-
-static void FilterFarSSE2(aec_t *aec, float yf[2][PART_LEN1])
-{
-  int i;
-  for (i = 0; i < NR_PART; i++) {
-    int j;
-    int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
-    int pos = i * PART_LEN1;
-    // Check for wrap
-    if (i + aec->xfBufBlockPos >= NR_PART) {
-      xPos -= NR_PART*(PART_LEN1);
-    }
-
-    // vectorized code (four at once)
-    for (j = 0; j + 3 < PART_LEN1; j += 4) {
-      const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
-      const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
-      const __m128 wfBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
-      const __m128 wfBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
-      const __m128 yf_re = _mm_loadu_ps(&yf[0][j]);
-      const __m128 yf_im = _mm_loadu_ps(&yf[1][j]);
-      const __m128 a = _mm_mul_ps(xfBuf_re, wfBuf_re);
-      const __m128 b = _mm_mul_ps(xfBuf_im, wfBuf_im);
-      const __m128 c = _mm_mul_ps(xfBuf_re, wfBuf_im);
-      const __m128 d = _mm_mul_ps(xfBuf_im, wfBuf_re);
-      const __m128 e = _mm_sub_ps(a, b);
-      const __m128 f = _mm_add_ps(c, d);
-      const __m128 g = _mm_add_ps(yf_re, e);
-      const __m128 h = _mm_add_ps(yf_im, f);
-      _mm_storeu_ps(&yf[0][j], g);
-      _mm_storeu_ps(&yf[1][j], h);
-    }
-    // scalar code for the remaining items.
-    for (; j < PART_LEN1; j++) {
-      yf[0][j] += MulRe(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
-      yf[1][j] += MulIm(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
-                        aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
-    }
-  }
-}
-
-static void ScaleErrorSignalSSE2(aec_t *aec, float ef[2][PART_LEN1])
-{
-  const __m128 k1e_10f = _mm_set1_ps(1e-10f);
-  const __m128 kThresh = _mm_set1_ps(aec->errThresh);
-  const __m128 kMu = _mm_set1_ps(aec->mu);
-
-  int i;
-  // vectorized code (four at once)
-  for (i = 0; i + 3 < PART_LEN1; i += 4) {
-    const __m128 xPow = _mm_loadu_ps(&aec->xPow[i]);
-    const __m128 ef_re_base = _mm_loadu_ps(&ef[0][i]);
-    const __m128 ef_im_base = _mm_loadu_ps(&ef[1][i]);
-
-    const __m128 xPowPlus = _mm_add_ps(xPow, k1e_10f);
-    __m128 ef_re = _mm_div_ps(ef_re_base, xPowPlus);
-    __m128 ef_im = _mm_div_ps(ef_im_base, xPowPlus);
-    const __m128 ef_re2 = _mm_mul_ps(ef_re, ef_re);
-    const __m128 ef_im2 = _mm_mul_ps(ef_im, ef_im);
-    const __m128 ef_sum2 = _mm_add_ps(ef_re2, ef_im2);
-    const __m128 absEf = _mm_sqrt_ps(ef_sum2);
-    const __m128 bigger = _mm_cmpgt_ps(absEf, kThresh);
-    __m128 absEfPlus = _mm_add_ps(absEf, k1e_10f);
-    const __m128 absEfInv = _mm_div_ps(kThresh, absEfPlus);
-    __m128 ef_re_if = _mm_mul_ps(ef_re, absEfInv);
-    __m128 ef_im_if = _mm_mul_ps(ef_im, absEfInv);
-    ef_re_if = _mm_and_ps(bigger, ef_re_if);
-    ef_im_if = _mm_and_ps(bigger, ef_im_if);
-    ef_re = _mm_andnot_ps(bigger, ef_re);
-    ef_im = _mm_andnot_ps(bigger, ef_im);
-    ef_re = _mm_or_ps(ef_re, ef_re_if);
-    ef_im = _mm_or_ps(ef_im, ef_im_if);
-    ef_re = _mm_mul_ps(ef_re, kMu);
-    ef_im = _mm_mul_ps(ef_im, kMu);
-
-    _mm_storeu_ps(&ef[0][i], ef_re);
-    _mm_storeu_ps(&ef[1][i], ef_im);
-  }
-  // scalar code for the remaining items.
-  for (; i < (PART_LEN1); i++) {
-    float absEf;
-    ef[0][i] /= (aec->xPow[i] + 1e-10f);
-    ef[1][i] /= (aec->xPow[i] + 1e-10f);
-    absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
-
-    if (absEf > aec->errThresh) {
-      absEf = aec->errThresh / (absEf + 1e-10f);
-      ef[0][i] *= absEf;
-      ef[1][i] *= absEf;
-    }
-
-    // Stepsize factor
-    ef[0][i] *= aec->mu;
-    ef[1][i] *= aec->mu;
-  }
-}
-
-static void FilterAdaptationSSE2(aec_t *aec, float *fft, float ef[2][PART_LEN1]) {
-  int i, j;
-  for (i = 0; i < NR_PART; i++) {
-    int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
-    int pos = i * PART_LEN1;
-    // Check for wrap
-    if (i + aec->xfBufBlockPos >= NR_PART) {
-      xPos -= NR_PART * PART_LEN1;
-    }
-
-#ifdef UNCONSTR
-    for (j = 0; j < PART_LEN1; j++) {
-      aec->wfBuf[pos + j][0] += MulRe(aec->xfBuf[xPos + j][0],
-                                      -aec->xfBuf[xPos + j][1],
-                                      ef[j][0], ef[j][1]);
-      aec->wfBuf[pos + j][1] += MulIm(aec->xfBuf[xPos + j][0],
-                                      -aec->xfBuf[xPos + j][1],
-                                      ef[j][0], ef[j][1]);
-    }
-#else
-    // Process the whole array...
-    for (j = 0; j < PART_LEN; j+= 4) {
-      // Load xfBuf and ef.
-      const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
-      const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
-      const __m128 ef_re = _mm_loadu_ps(&ef[0][j]);
-      const __m128 ef_im = _mm_loadu_ps(&ef[1][j]);
-      // Calculate the product of conjugate(xfBuf) by ef.
-      //   re(conjugate(a) * b) = aRe * bRe + aIm * bIm
-      //   im(conjugate(a) * b)=  aRe * bIm - aIm * bRe
-      const __m128 a = _mm_mul_ps(xfBuf_re, ef_re);
-      const __m128 b = _mm_mul_ps(xfBuf_im, ef_im);
-      const __m128 c = _mm_mul_ps(xfBuf_re, ef_im);
-      const __m128 d = _mm_mul_ps(xfBuf_im, ef_re);
-      const __m128 e = _mm_add_ps(a, b);
-      const __m128 f = _mm_sub_ps(c, d);
-      // Interleave real and imaginary parts.
-      const __m128 g = _mm_unpacklo_ps(e, f);
-      const __m128 h = _mm_unpackhi_ps(e, f);
-      // Store
-      _mm_storeu_ps(&fft[2*j + 0], g);
-      _mm_storeu_ps(&fft[2*j + 4], h);
-    }
-    // ... and fixup the first imaginary entry.
-    fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
-                   -aec->xfBuf[1][xPos + PART_LEN],
-                   ef[0][PART_LEN], ef[1][PART_LEN]);
-
-    aec_rdft_inverse_128(fft);
-    memset(fft + PART_LEN, 0, sizeof(float)*PART_LEN);
-
-    // fft scaling
-    {
-      float scale = 2.0f / PART_LEN2;
-      const __m128 scale_ps = _mm_load_ps1(&scale);
-      for (j = 0; j < PART_LEN; j+=4) {
-        const __m128 fft_ps = _mm_loadu_ps(&fft[j]);
-        const __m128 fft_scale = _mm_mul_ps(fft_ps, scale_ps);
-        _mm_storeu_ps(&fft[j], fft_scale);
-      }
-    }
-    aec_rdft_forward_128(fft);
-
-    {
-      float wt1 = aec->wfBuf[1][pos];
-      aec->wfBuf[0][pos + PART_LEN] += fft[1];
-      for (j = 0; j < PART_LEN; j+= 4) {
-        __m128 wtBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
-        __m128 wtBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
-        const __m128 fft0 = _mm_loadu_ps(&fft[2 * j + 0]);
-        const __m128 fft4 = _mm_loadu_ps(&fft[2 * j + 4]);
-        const __m128 fft_re = _mm_shuffle_ps(fft0, fft4, _MM_SHUFFLE(2, 0, 2 ,0));
-        const __m128 fft_im = _mm_shuffle_ps(fft0, fft4, _MM_SHUFFLE(3, 1, 3 ,1));
-        wtBuf_re = _mm_add_ps(wtBuf_re, fft_re);
-        wtBuf_im = _mm_add_ps(wtBuf_im, fft_im);
-        _mm_storeu_ps(&aec->wfBuf[0][pos + j], wtBuf_re);
-        _mm_storeu_ps(&aec->wfBuf[1][pos + j], wtBuf_im);
-      }
-      aec->wfBuf[1][pos] = wt1;
-    }
-#endif // UNCONSTR
-  }
-}
-
-#ifdef _MSC_VER /* visual c++ */
-# define ALIGN16_BEG __declspec(align(16))
-# define ALIGN16_END
-#else /* gcc or icc */
-# define ALIGN16_BEG
-# define ALIGN16_END __attribute__((aligned(16)))
-#endif
-
-static __m128 mm_pow_ps(__m128 a, __m128 b)
-{
-  // a^b = exp2(b * log2(a))
-  //   exp2(x) and log2(x) are calculated using polynomial approximations.
-  __m128 log2_a, b_log2_a, a_exp_b;
-
-  // Calculate log2(x), x = a.
-  {
-    // To calculate log2(x), we decompose x like this:
-    //   x = y * 2^n
-    //     n is an integer
-    //     y is in the [1.0, 2.0) range
-    //
-    //   log2(x) = log2(y) + n
-    //     n       can be evaluated by playing with float representation.
-    //     log2(y) in a small range can be approximated, this code uses an order
-    //             five polynomial approximation. The coefficients have been
-    //             estimated with the Remez algorithm and the resulting
-    //             polynomial has a maximum relative error of 0.00086%.
-
-    // Compute n.
-    //    This is done by masking the exponent, shifting it into the top bit of
-    //    the mantissa, putting eight into the biased exponent (to shift/
-    //    compensate the fact that the exponent has been shifted in the top/
-    //    fractional part and finally getting rid of the implicit leading one
-    //    from the mantissa by substracting it out.
-    static const ALIGN16_BEG int float_exponent_mask[4] ALIGN16_END =
-        {0x7F800000, 0x7F800000, 0x7F800000, 0x7F800000};
-    static const ALIGN16_BEG int eight_biased_exponent[4] ALIGN16_END =
-        {0x43800000, 0x43800000, 0x43800000, 0x43800000};
-    static const ALIGN16_BEG int implicit_leading_one[4] ALIGN16_END =
-        {0x43BF8000, 0x43BF8000, 0x43BF8000, 0x43BF8000};
-    static const int shift_exponent_into_top_mantissa = 8;
-    const __m128 two_n = _mm_and_ps(a, *((__m128 *)float_exponent_mask));
-    const __m128 n_1 = (__m128)_mm_srli_epi32((__m128i)two_n,
-        shift_exponent_into_top_mantissa);
-    const __m128 n_0 = _mm_or_ps(
-        (__m128)n_1, *((__m128 *)eight_biased_exponent));
-    const __m128 n   = _mm_sub_ps(n_0,  *((__m128 *)implicit_leading_one));
-
-    // Compute y.
-    static const ALIGN16_BEG int mantissa_mask[4] ALIGN16_END =
-        {0x007FFFFF, 0x007FFFFF, 0x007FFFFF, 0x007FFFFF};
-    static const ALIGN16_BEG int zero_biased_exponent_is_one[4] ALIGN16_END =
-        {0x3F800000, 0x3F800000, 0x3F800000, 0x3F800000};
-    const __m128 mantissa = _mm_and_ps(a, *((__m128 *)mantissa_mask));
-    const __m128 y        = _mm_or_ps(
-        mantissa,  *((__m128 *)zero_biased_exponent_is_one));
-
-    // Approximate log2(y) ~= (y - 1) * pol5(y).
-    //    pol5(y) = C5 * y^5 + C4 * y^4 + C3 * y^3 + C2 * y^2 + C1 * y + C0
-    static const ALIGN16_BEG float ALIGN16_END C5[4] =
-        {-3.4436006e-2f, -3.4436006e-2f, -3.4436006e-2f, -3.4436006e-2f};
-    static const ALIGN16_BEG float ALIGN16_END C4[4] =
-        {3.1821337e-1f, 3.1821337e-1f, 3.1821337e-1f, 3.1821337e-1f};
-    static const ALIGN16_BEG float ALIGN16_END C3[4] =
-        {-1.2315303f, -1.2315303f, -1.2315303f, -1.2315303f};
-    static const ALIGN16_BEG float ALIGN16_END C2[4] =
-        {2.5988452f, 2.5988452f, 2.5988452f, 2.5988452f};
-    static const ALIGN16_BEG float ALIGN16_END C1[4] =
-        {-3.3241990f, -3.3241990f, -3.3241990f, -3.3241990f};
-    static const ALIGN16_BEG float ALIGN16_END C0[4] =
-        {3.1157899f, 3.1157899f, 3.1157899f, 3.1157899f};
-    const __m128 pol5_y_0 = _mm_mul_ps(y,        *((__m128 *)C5));
-    const __m128 pol5_y_1 = _mm_add_ps(pol5_y_0, *((__m128 *)C4));
-    const __m128 pol5_y_2 = _mm_mul_ps(pol5_y_1, y);
-    const __m128 pol5_y_3 = _mm_add_ps(pol5_y_2, *((__m128 *)C3));
-    const __m128 pol5_y_4 = _mm_mul_ps(pol5_y_3, y);
-    const __m128 pol5_y_5 = _mm_add_ps(pol5_y_4, *((__m128 *)C2));
-    const __m128 pol5_y_6 = _mm_mul_ps(pol5_y_5, y);
-    const __m128 pol5_y_7 = _mm_add_ps(pol5_y_6, *((__m128 *)C1));
-    const __m128 pol5_y_8 = _mm_mul_ps(pol5_y_7, y);
-    const __m128 pol5_y   = _mm_add_ps(pol5_y_8, *((__m128 *)C0));
-    const __m128 y_minus_one = _mm_sub_ps(
-        y, *((__m128 *)zero_biased_exponent_is_one));
-    const __m128 log2_y = _mm_mul_ps(y_minus_one ,  pol5_y);
-
-    // Combine parts.
-    log2_a = _mm_add_ps(n, log2_y);
-  }
-
-  // b * log2(a)
-  b_log2_a = _mm_mul_ps(b, log2_a);
-
-  // Calculate exp2(x), x = b * log2(a).
-  {
-    // To calculate 2^x, we decompose x like this:
-    //   x = n + y
-    //     n is an integer, the value of x - 0.5 rounded down, therefore
-    //     y is in the [0.5, 1.5) range
-    //
-    //   2^x = 2^n * 2^y
-    //     2^n can be evaluated by playing with float representation.
-    //     2^y in a small range can be approximated, this code uses an order two
-    //         polynomial approximation. The coefficients have been estimated
-    //         with the Remez algorithm and the resulting polynomial has a
-    //         maximum relative error of 0.17%.
-
-    // To avoid over/underflow, we reduce the range of input to ]-127, 129].
-    static const ALIGN16_BEG float max_input[4] ALIGN16_END =
-        {129.f, 129.f, 129.f, 129.f};
-    static const ALIGN16_BEG float min_input[4] ALIGN16_END =
-        {-126.99999f, -126.99999f, -126.99999f, -126.99999f};
-    const __m128 x_min = _mm_min_ps(b_log2_a, *((__m128 *)max_input));
-    const __m128 x_max = _mm_max_ps(x_min,    *((__m128 *)min_input));
-    // Compute n.
-    static const ALIGN16_BEG float half[4] ALIGN16_END =
-        {0.5f, 0.5f, 0.5f, 0.5f};
-    const __m128  x_minus_half = _mm_sub_ps(x_max, *((__m128 *)half));
-    const __m128i x_minus_half_floor = _mm_cvtps_epi32(x_minus_half);
-    // Compute 2^n.
-    static const ALIGN16_BEG int float_exponent_bias[4] ALIGN16_END =
-        {127, 127, 127, 127};
-    static const int float_exponent_shift = 23;
-    const __m128i two_n_exponent = _mm_add_epi32(
-        x_minus_half_floor, *((__m128i *)float_exponent_bias));
-    const __m128  two_n = (__m128)_mm_slli_epi32(
-        two_n_exponent, float_exponent_shift);
-    // Compute y.
-    const __m128 y = _mm_sub_ps(x_max, _mm_cvtepi32_ps(x_minus_half_floor));
-    // Approximate 2^y ~= C2 * y^2 + C1 * y + C0.
-    static const ALIGN16_BEG float C2[4] ALIGN16_END =
-        {3.3718944e-1f, 3.3718944e-1f, 3.3718944e-1f, 3.3718944e-1f};
-    static const ALIGN16_BEG float C1[4] ALIGN16_END =
-        {6.5763628e-1f, 6.5763628e-1f, 6.5763628e-1f, 6.5763628e-1f};
-    static const ALIGN16_BEG float C0[4] ALIGN16_END =
-        {1.0017247f, 1.0017247f, 1.0017247f, 1.0017247f};
-    const __m128 exp2_y_0 = _mm_mul_ps(y,        *((__m128 *)C2));
-    const __m128 exp2_y_1 = _mm_add_ps(exp2_y_0, *((__m128 *)C1));
-    const __m128 exp2_y_2 = _mm_mul_ps(exp2_y_1, y);
-    const __m128 exp2_y   = _mm_add_ps(exp2_y_2, *((__m128 *)C0));
-
-    // Combine parts.
-    a_exp_b = _mm_mul_ps(exp2_y, two_n);
-  }
-  return a_exp_b;
-}
-
-extern const float WebRtcAec_weightCurve[65];
-extern const float WebRtcAec_overDriveCurve[65];
-
-static void OverdriveAndSuppressSSE2(aec_t *aec, float hNl[PART_LEN1],
-                                     const float hNlFb,
-                                     float efw[2][PART_LEN1]) {
-  int i;
-  const __m128 vec_hNlFb = _mm_set1_ps(hNlFb);
-  const __m128 vec_one = _mm_set1_ps(1.0f);
-  const __m128 vec_minus_one = _mm_set1_ps(-1.0f);
-  const __m128 vec_overDriveSm = _mm_set1_ps(aec->overDriveSm);
-  // vectorized code (four at once)
-  for (i = 0; i + 3 < PART_LEN1; i+=4) {
-    // Weight subbands
-    __m128 vec_hNl = _mm_loadu_ps(&hNl[i]);
-    const __m128 vec_weightCurve = _mm_loadu_ps(&WebRtcAec_weightCurve[i]);
-    const __m128 bigger = _mm_cmpgt_ps(vec_hNl, vec_hNlFb);
-    const __m128 vec_weightCurve_hNlFb = _mm_mul_ps(
-        vec_weightCurve, vec_hNlFb);
-    const __m128 vec_one_weightCurve = _mm_sub_ps(vec_one, vec_weightCurve);
-    const __m128 vec_one_weightCurve_hNl = _mm_mul_ps(
-        vec_one_weightCurve, vec_hNl);
-    const __m128 vec_if0 = _mm_andnot_ps(bigger, vec_hNl);
-    const __m128 vec_if1 = _mm_and_ps(
-        bigger, _mm_add_ps(vec_weightCurve_hNlFb, vec_one_weightCurve_hNl));
-    vec_hNl = _mm_or_ps(vec_if0, vec_if1);
-
-    {
-      const __m128 vec_overDriveCurve = _mm_loadu_ps(
-          &WebRtcAec_overDriveCurve[i]);
-      const __m128 vec_overDriveSm_overDriveCurve = _mm_mul_ps(
-          vec_overDriveSm, vec_overDriveCurve);
-      vec_hNl = mm_pow_ps(vec_hNl, vec_overDriveSm_overDriveCurve);
-      _mm_storeu_ps(&hNl[i], vec_hNl);
-    }
-
-    // Suppress error signal
-    {
-      __m128 vec_efw_re = _mm_loadu_ps(&efw[0][i]);
-      __m128 vec_efw_im = _mm_loadu_ps(&efw[1][i]);
-      vec_efw_re = _mm_mul_ps(vec_efw_re, vec_hNl);
-      vec_efw_im = _mm_mul_ps(vec_efw_im, vec_hNl);
-
-      // Ooura fft returns incorrect sign on imaginary component. It matters
-      // here because we are making an additive change with comfort noise.
-      vec_efw_im = _mm_mul_ps(vec_efw_im, vec_minus_one);
-      _mm_storeu_ps(&efw[0][i], vec_efw_re);
-      _mm_storeu_ps(&efw[1][i], vec_efw_im);
-    }
-  }
-  // scalar code for the remaining items.
-  for (; i < PART_LEN1; i++) {
-    // Weight subbands
-    if (hNl[i] > hNlFb) {
-      hNl[i] = WebRtcAec_weightCurve[i] * hNlFb +
-          (1 - WebRtcAec_weightCurve[i]) * hNl[i];
-    }
-    hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
-
-    // Suppress error signal
-    efw[0][i] *= hNl[i];
-    efw[1][i] *= hNl[i];
-
-    // Ooura fft returns incorrect sign on imaginary component. It matters
-    // here because we are making an additive change with comfort noise.
-    efw[1][i] *= -1;
-  }
-}
-
-void WebRtcAec_InitAec_SSE2(void) {
-  WebRtcAec_FilterFar = FilterFarSSE2;
-  WebRtcAec_ScaleErrorSignal = ScaleErrorSignalSSE2;
-  WebRtcAec_FilterAdaptation = FilterAdaptationSSE2;
-  WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
-}
-
-#endif   //__SSE2__
diff --git a/src/modules/audio_processing/aec/main/source/aec_rdft.c b/src/modules/audio_processing/aec/main/source/aec_rdft.c
deleted file mode 100644
index 072a1c4..0000000
--- a/src/modules/audio_processing/aec/main/source/aec_rdft.c
+++ /dev/null
@@ -1,522 +0,0 @@
-/*
- * http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
- * Copyright Takuya OOURA, 1996-2001
- *
- * You may use, copy, modify and distribute this code for any purpose (include
- * commercial use) and without fee. Please refer to this package when you modify
- * this code.
- *
- * Changes by the WebRTC authors:
- *    - Trivial type modifications.
- *    - Minimal code subset to do rdft of length 128.
- *    - Optimizations because of known length.
- *
- *  All changes are covered by the WebRTC license and IP grant:
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <math.h>
-
-#include "aec_rdft.h"
-#include "system_wrappers/interface/cpu_features_wrapper.h"
-
-float rdft_w[64];
-static int ip[16];
-
-static void bitrv2_32or128(int n, int *ip, float *a) {
-  // n is 32 or 128
-  int j, j1, k, k1, m, m2;
-  float xr, xi, yr, yi;
-
-  ip[0] = 0;
-  {
-    int l = n;
-    m = 1;
-    while ((m << 3) < l) {
-      l >>= 1;
-      for (j = 0; j < m; j++) {
-        ip[m + j] = ip[j] + l;
-      }
-      m <<= 1;
-    }
-  }
-  m2 = 2 * m;
-  for (k = 0; k < m; k++) {
-    for (j = 0; j < k; j++) {
-      j1 = 2 * j + ip[k];
-      k1 = 2 * k + ip[j];
-      xr = a[j1];
-      xi = a[j1 + 1];
-      yr = a[k1];
-      yi = a[k1 + 1];
-      a[j1] = yr;
-      a[j1 + 1] = yi;
-      a[k1] = xr;
-      a[k1 + 1] = xi;
-      j1 += m2;
-      k1 += 2 * m2;
-      xr = a[j1];
-      xi = a[j1 + 1];
-      yr = a[k1];
-      yi = a[k1 + 1];
-      a[j1] = yr;
-      a[j1 + 1] = yi;
-      a[k1] = xr;
-      a[k1 + 1] = xi;
-      j1 += m2;
-      k1 -= m2;
-      xr = a[j1];
-      xi = a[j1 + 1];
-      yr = a[k1];
-      yi = a[k1 + 1];
-      a[j1] = yr;
-      a[j1 + 1] = yi;
-      a[k1] = xr;
-      a[k1 + 1] = xi;
-      j1 += m2;
-      k1 += 2 * m2;
-      xr = a[j1];
-      xi = a[j1 + 1];
-      yr = a[k1];
-      yi = a[k1 + 1];
-      a[j1] = yr;
-      a[j1 + 1] = yi;
-      a[k1] = xr;
-      a[k1 + 1] = xi;
-    }
-    j1 = 2 * k + m2 + ip[k];
-    k1 = j1 + m2;
-    xr = a[j1];
-    xi = a[j1 + 1];
-    yr = a[k1];
-    yi = a[k1 + 1];
-    a[j1] = yr;
-    a[j1 + 1] = yi;
-    a[k1] = xr;
-    a[k1 + 1] = xi;
-  }
-}
-
-static void makewt_32() {
-  const int nw = 32;
-  int j, nwh;
-  float delta, x, y;
-
-  ip[0] = nw;
-  ip[1] = 1;
-  nwh = nw >> 1;
-  delta = atanf(1.0f) / nwh;
-  rdft_w[0] = 1;
-  rdft_w[1] = 0;
-  rdft_w[nwh] = cosf(delta * nwh);
-  rdft_w[nwh + 1] = rdft_w[nwh];
-  for (j = 2; j < nwh; j += 2) {
-    x = cosf(delta * j);
-    y = sinf(delta * j);
-    rdft_w[j] = x;
-    rdft_w[j + 1] = y;
-    rdft_w[nw - j] = y;
-    rdft_w[nw - j + 1] = x;
-  }
-  bitrv2_32or128(nw, ip + 2, rdft_w);
-}
-
-static void makect_32() {
-  float *c = rdft_w + 32;
-  const int nc = 32;
-  int j, nch;
-  float delta;
-
-  ip[1] = nc;
-  nch = nc >> 1;
-  delta = atanf(1.0f) / nch;
-  c[0] = cosf(delta * nch);
-  c[nch] = 0.5f * c[0];
-  for (j = 1; j < nch; j++) {
-    c[j] = 0.5f * cosf(delta * j);
-    c[nc - j] = 0.5f * sinf(delta * j);
-  }
-}
-
-static void cft1st_128(float *a) {
-  const int n = 128;
-  int j, k1, k2;
-  float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
-  float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
-  x0r = a[0] + a[2];
-  x0i = a[1] + a[3];
-  x1r = a[0] - a[2];
-  x1i = a[1] - a[3];
-  x2r = a[4] + a[6];
-  x2i = a[5] + a[7];
-  x3r = a[4] - a[6];
-  x3i = a[5] - a[7];
-  a[0] = x0r + x2r;
-  a[1] = x0i + x2i;
-  a[4] = x0r - x2r;
-  a[5] = x0i - x2i;
-  a[2] = x1r - x3i;
-  a[3] = x1i + x3r;
-  a[6] = x1r + x3i;
-  a[7] = x1i - x3r;
-  wk1r = rdft_w[2];
-  x0r = a[8] + a[10];
-  x0i = a[9] + a[11];
-  x1r = a[8] - a[10];
-  x1i = a[9] - a[11];
-  x2r = a[12] + a[14];
-  x2i = a[13] + a[15];
-  x3r = a[12] - a[14];
-  x3i = a[13] - a[15];
-  a[8] = x0r + x2r;
-  a[9] = x0i + x2i;
-  a[12] = x2i - x0i;
-  a[13] = x0r - x2r;
-  x0r = x1r - x3i;
-  x0i = x1i + x3r;
-  a[10] = wk1r * (x0r - x0i);
-  a[11] = wk1r * (x0r + x0i);
-  x0r = x3i + x1r;
-  x0i = x3r - x1i;
-  a[14] = wk1r * (x0i - x0r);
-  a[15] = wk1r * (x0i + x0r);
-  k1 = 0;
-  for (j = 16; j < n; j += 16) {
-    k1 += 2;
-    k2 = 2 * k1;
-    wk2r = rdft_w[k1];
-    wk2i = rdft_w[k1 + 1];
-    wk1r = rdft_w[k2];
-    wk1i = rdft_w[k2 + 1];
-    wk3r = wk1r - 2 * wk2i * wk1i;
-    wk3i = 2 * wk2i * wk1r - wk1i;
-    x0r = a[j] + a[j + 2];
-    x0i = a[j + 1] + a[j + 3];
-    x1r = a[j] - a[j + 2];
-    x1i = a[j + 1] - a[j + 3];
-    x2r = a[j + 4] + a[j + 6];
-    x2i = a[j + 5] + a[j + 7];
-    x3r = a[j + 4] - a[j + 6];
-    x3i = a[j + 5] - a[j + 7];
-    a[j] = x0r + x2r;
-    a[j + 1] = x0i + x2i;
-    x0r -= x2r;
-    x0i -= x2i;
-    a[j + 4] = wk2r * x0r - wk2i * x0i;
-    a[j + 5] = wk2r * x0i + wk2i * x0r;
-    x0r = x1r - x3i;
-    x0i = x1i + x3r;
-    a[j + 2] = wk1r * x0r - wk1i * x0i;
-    a[j + 3] = wk1r * x0i + wk1i * x0r;
-    x0r = x1r + x3i;
-    x0i = x1i - x3r;
-    a[j + 6] = wk3r * x0r - wk3i * x0i;
-    a[j + 7] = wk3r * x0i + wk3i * x0r;
-    wk1r = rdft_w[k2 + 2];
-    wk1i = rdft_w[k2 + 3];
-    wk3r = wk1r - 2 * wk2r * wk1i;
-    wk3i = 2 * wk2r * wk1r - wk1i;
-    x0r = a[j + 8] + a[j + 10];
-    x0i = a[j + 9] + a[j + 11];
-    x1r = a[j + 8] - a[j + 10];
-    x1i = a[j + 9] - a[j + 11];
-    x2r = a[j + 12] + a[j + 14];
-    x2i = a[j + 13] + a[j + 15];
-    x3r = a[j + 12] - a[j + 14];
-    x3i = a[j + 13] - a[j + 15];
-    a[j + 8] = x0r + x2r;
-    a[j + 9] = x0i + x2i;
-    x0r -= x2r;
-    x0i -= x2i;
-    a[j + 12] = -wk2i * x0r - wk2r * x0i;
-    a[j + 13] = -wk2i * x0i + wk2r * x0r;
-    x0r = x1r - x3i;
-    x0i = x1i + x3r;
-    a[j + 10] = wk1r * x0r - wk1i * x0i;
-    a[j + 11] = wk1r * x0i + wk1i * x0r;
-    x0r = x1r + x3i;
-    x0i = x1i - x3r;
-    a[j + 14] = wk3r * x0r - wk3i * x0i;
-    a[j + 15] = wk3r * x0i + wk3i * x0r;
-  }
-}
-
-static void cftmdl_128(int l, float *a) {
-  const int n = 128;
-  int j, j1, j2, j3, k, k1, k2, m, m2;
-  float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
-  float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
-  m = l << 2;
-  for (j = 0; j < l; j += 2) {
-    j1 = j + l;
-    j2 = j1 + l;
-    j3 = j2 + l;
-    x0r = a[j] + a[j1];
-    x0i = a[j + 1] + a[j1 + 1];
-    x1r = a[j] - a[j1];
-    x1i = a[j + 1] - a[j1 + 1];
-    x2r = a[j2] + a[j3];
-    x2i = a[j2 + 1] + a[j3 + 1];
-    x3r = a[j2] - a[j3];
-    x3i = a[j2 + 1] - a[j3 + 1];
-    a[j] = x0r + x2r;
-    a[j + 1] = x0i + x2i;
-    a[j2] = x0r - x2r;
-    a[j2 + 1] = x0i - x2i;
-    a[j1] = x1r - x3i;
-    a[j1 + 1] = x1i + x3r;
-    a[j3] = x1r + x3i;
-    a[j3 + 1] = x1i - x3r;
-  }
-  wk1r = rdft_w[2];
-  for (j = m; j < l + m; j += 2) {
-    j1 = j + l;
-    j2 = j1 + l;
-    j3 = j2 + l;
-    x0r = a[j] + a[j1];
-    x0i = a[j + 1] + a[j1 + 1];
-    x1r = a[j] - a[j1];
-    x1i = a[j + 1] - a[j1 + 1];
-    x2r = a[j2] + a[j3];
-    x2i = a[j2 + 1] + a[j3 + 1];
-    x3r = a[j2] - a[j3];
-    x3i = a[j2 + 1] - a[j3 + 1];
-    a[j] = x0r + x2r;
-    a[j + 1] = x0i + x2i;
-    a[j2] = x2i - x0i;
-    a[j2 + 1] = x0r - x2r;
-    x0r = x1r - x3i;
-    x0i = x1i + x3r;
-    a[j1] = wk1r * (x0r - x0i);
-    a[j1 + 1] = wk1r * (x0r + x0i);
-    x0r = x3i + x1r;
-    x0i = x3r - x1i;
-    a[j3] = wk1r * (x0i - x0r);
-    a[j3 + 1] = wk1r * (x0i + x0r);
-  }
-  k1 = 0;
-  m2 = 2 * m;
-  for (k = m2; k < n; k += m2) {
-    k1 += 2;
-    k2 = 2 * k1;
-    wk2r = rdft_w[k1];
-    wk2i = rdft_w[k1 + 1];
-    wk1r = rdft_w[k2];
-    wk1i = rdft_w[k2 + 1];
-    wk3r = wk1r - 2 * wk2i * wk1i;
-    wk3i = 2 * wk2i * wk1r - wk1i;
-    for (j = k; j < l + k; j += 2) {
-      j1 = j + l;
-      j2 = j1 + l;
-      j3 = j2 + l;
-      x0r = a[j] + a[j1];
-      x0i = a[j + 1] + a[j1 + 1];
-      x1r = a[j] - a[j1];
-      x1i = a[j + 1] - a[j1 + 1];
-      x2r = a[j2] + a[j3];
-      x2i = a[j2 + 1] + a[j3 + 1];
-      x3r = a[j2] - a[j3];
-      x3i = a[j2 + 1] - a[j3 + 1];
-      a[j] = x0r + x2r;
-      a[j + 1] = x0i + x2i;
-      x0r -= x2r;
-      x0i -= x2i;
-      a[j2] = wk2r * x0r - wk2i * x0i;
-      a[j2 + 1] = wk2r * x0i + wk2i * x0r;
-      x0r = x1r - x3i;
-      x0i = x1i + x3r;
-      a[j1] = wk1r * x0r - wk1i * x0i;
-      a[j1 + 1] = wk1r * x0i + wk1i * x0r;
-      x0r = x1r + x3i;
-      x0i = x1i - x3r;
-      a[j3] = wk3r * x0r - wk3i * x0i;
-      a[j3 + 1] = wk3r * x0i + wk3i * x0r;
-    }
-    wk1r = rdft_w[k2 + 2];
-    wk1i = rdft_w[k2 + 3];
-    wk3r = wk1r - 2 * wk2r * wk1i;
-    wk3i = 2 * wk2r * wk1r - wk1i;
-    for (j = k + m; j < l + (k + m); j += 2) {
-      j1 = j + l;
-      j2 = j1 + l;
-      j3 = j2 + l;
-      x0r = a[j] + a[j1];
-      x0i = a[j + 1] + a[j1 + 1];
-      x1r = a[j] - a[j1];
-      x1i = a[j + 1] - a[j1 + 1];
-      x2r = a[j2] + a[j3];
-      x2i = a[j2 + 1] + a[j3 + 1];
-      x3r = a[j2] - a[j3];
-      x3i = a[j2 + 1] - a[j3 + 1];
-      a[j] = x0r + x2r;
-      a[j + 1] = x0i + x2i;
-      x0r -= x2r;
-      x0i -= x2i;
-      a[j2] = -wk2i * x0r - wk2r * x0i;
-      a[j2 + 1] = -wk2i * x0i + wk2r * x0r;
-      x0r = x1r - x3i;
-      x0i = x1i + x3r;
-      a[j1] = wk1r * x0r - wk1i * x0i;
-      a[j1 + 1] = wk1r * x0i + wk1i * x0r;
-      x0r = x1r + x3i;
-      x0i = x1i - x3r;
-      a[j3] = wk3r * x0r - wk3i * x0i;
-      a[j3 + 1] = wk3r * x0i + wk3i * x0r;
-    }
-  }
-}
-
-static void cftfsub_128(float *a) {
-  int j, j1, j2, j3, l;
-  float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
-  cft1st_128(a);
-  cftmdl_128(8, a);
-  l = 32;
-  for (j = 0; j < l; j += 2) {
-    j1 = j + l;
-    j2 = j1 + l;
-    j3 = j2 + l;
-    x0r = a[j] + a[j1];
-    x0i = a[j + 1] + a[j1 + 1];
-    x1r = a[j] - a[j1];
-    x1i = a[j + 1] - a[j1 + 1];
-    x2r = a[j2] + a[j3];
-    x2i = a[j2 + 1] + a[j3 + 1];
-    x3r = a[j2] - a[j3];
-    x3i = a[j2 + 1] - a[j3 + 1];
-    a[j] = x0r + x2r;
-    a[j + 1] = x0i + x2i;
-    a[j2] = x0r - x2r;
-    a[j2 + 1] = x0i - x2i;
-    a[j1] = x1r - x3i;
-    a[j1 + 1] = x1i + x3r;
-    a[j3] = x1r + x3i;
-    a[j3 + 1] = x1i - x3r;
-  }
-}
-
-static void cftbsub_128(float *a) {
-  int j, j1, j2, j3, l;
-  float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
-  cft1st_128(a);
-  cftmdl_128(8, a);
-  l = 32;
-
-  for (j = 0; j < l; j += 2) {
-    j1 = j + l;
-    j2 = j1 + l;
-    j3 = j2 + l;
-    x0r = a[j] + a[j1];
-    x0i = -a[j + 1] - a[j1 + 1];
-    x1r = a[j] - a[j1];
-    x1i = -a[j + 1] + a[j1 + 1];
-    x2r = a[j2] + a[j3];
-    x2i = a[j2 + 1] + a[j3 + 1];
-    x3r = a[j2] - a[j3];
-    x3i = a[j2 + 1] - a[j3 + 1];
-    a[j] = x0r + x2r;
-    a[j + 1] = x0i - x2i;
-    a[j2] = x0r - x2r;
-    a[j2 + 1] = x0i + x2i;
-    a[j1] = x1r - x3i;
-    a[j1 + 1] = x1i - x3r;
-    a[j3] = x1r + x3i;
-    a[j3 + 1] = x1i + x3r;
-  }
-}
-
-static void rftfsub_128_C(float *a) {
-  const float *c = rdft_w + 32;
-  int j1, j2, k1, k2;
-  float wkr, wki, xr, xi, yr, yi;
-
-  for (j1 = 1, j2 = 2; j2 < 64; j1 += 1, j2 += 2) {
-    k2 = 128 - j2;
-    k1 =  32 - j1;
-    wkr = 0.5f - c[k1];
-    wki = c[j1];
-    xr = a[j2 + 0] - a[k2 + 0];
-    xi = a[j2 + 1] + a[k2 + 1];
-    yr = wkr * xr - wki * xi;
-    yi = wkr * xi + wki * xr;
-    a[j2 + 0] -= yr;
-    a[j2 + 1] -= yi;
-    a[k2 + 0] += yr;
-    a[k2 + 1] -= yi;
-  }
-}
-
-static void rftbsub_128_C(float *a) {
-  const float *c = rdft_w + 32;
-  int j1, j2, k1, k2;
-  float wkr, wki, xr, xi, yr, yi;
-
-  a[1] = -a[1];
-  for (j1 = 1, j2 = 2; j2 < 64; j1 += 1, j2 += 2) {
-    k2 = 128 - j2;
-    k1 =  32 - j1;
-    wkr = 0.5f - c[k1];
-    wki = c[j1];
-    xr = a[j2 + 0] - a[k2 + 0];
-    xi = a[j2 + 1] + a[k2 + 1];
-    yr = wkr * xr + wki * xi;
-    yi = wkr * xi - wki * xr;
-    a[j2 + 0] = a[j2 + 0] - yr;
-    a[j2 + 1] = yi - a[j2 + 1];
-    a[k2 + 0] = yr + a[k2 + 0];
-    a[k2 + 1] = yi - a[k2 + 1];
-  }
-  a[65] = -a[65];
-}
-
-void aec_rdft_forward_128(float *a) {
-  const int n = 128;
-  int nw;
-  float xi;
-
-  nw = ip[0];
-  bitrv2_32or128(n, ip + 2, a);
-  cftfsub_128(a);
-  rftfsub_128(a);
-  xi = a[0] - a[1];
-  a[0] += a[1];
-  a[1] = xi;
-}
-
-void aec_rdft_inverse_128(float *a) {
-  const int n = 128;
-  int nw;
-  float xi;
-
-  nw = ip[0];
-  a[1] = 0.5f * (a[0] - a[1]);
-  a[0] -= a[1];
-  rftbsub_128(a);
-  bitrv2_32or128(n, ip + 2, a);
-  cftbsub_128(a);
-}
-
-// code path selection
-rft_sub_128_t rftfsub_128;
-rft_sub_128_t rftbsub_128;
-
-void aec_rdft_init(void) {
-  rftfsub_128 = rftfsub_128_C;
-  rftbsub_128 = rftbsub_128_C;
-  if (WebRtc_GetCPUInfo(kSSE2)) {
-#if defined(__SSE2__)
-    aec_rdft_init_sse2();
-#endif
-  }
-  // init library constants.
-  makewt_32();
-  makect_32();
-}
diff --git a/src/modules/audio_processing/aec/main/source/aec_rdft.h b/src/modules/audio_processing/aec/main/source/aec_rdft.h
deleted file mode 100644
index cf90882..0000000
--- a/src/modules/audio_processing/aec/main/source/aec_rdft.h
+++ /dev/null
@@ -1,23 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// constants shared by all paths (C, SSE2).
-extern float rdft_w[64];
-
-// code path selection function pointers
-typedef void (*rft_sub_128_t)(float *a);
-extern rft_sub_128_t rftfsub_128;
-extern rft_sub_128_t rftbsub_128;
-
-// entry points
-void aec_rdft_init(void);
-void aec_rdft_init_sse2(void);
-void aec_rdft_forward_128(float *a);
-void aec_rdft_inverse_128(float *a);
diff --git a/src/modules/audio_processing/aec/main/source/aec_rdft_sse2.c b/src/modules/audio_processing/aec/main/source/aec_rdft_sse2.c
deleted file mode 100644
index 901a1b1..0000000
--- a/src/modules/audio_processing/aec/main/source/aec_rdft_sse2.c
+++ /dev/null
@@ -1,209 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <emmintrin.h>
-
-#include "aec_rdft.h"
-
-#ifdef _MSC_VER /* visual c++ */
-# define ALIGN16_BEG __declspec(align(16))
-# define ALIGN16_END
-#else /* gcc or icc */
-# define ALIGN16_BEG
-# define ALIGN16_END __attribute__((aligned(16)))
-#endif
-
-static void rftfsub_128_SSE2(float *a) {
-  const float *c = rdft_w + 32;
-  int j1, j2, k1, k2;
-  float wkr, wki, xr, xi, yr, yi;
-
-  static const ALIGN16_BEG float ALIGN16_END k_half[4] =
-    {0.5f, 0.5f, 0.5f, 0.5f};
-  const __m128 mm_half = _mm_load_ps(k_half);
-
-  // Vectorized code (four at once).
-  //    Note: commented number are indexes for the first iteration of the loop.
-  for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
-    // Load 'wk'.
-    const __m128 c_j1 = _mm_loadu_ps(&c[     j1]);         //  1,  2,  3,  4,
-    const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]);         // 28, 29, 30, 31,
-    const __m128 wkrt = _mm_sub_ps(mm_half, c_k1);         // 28, 29, 30, 31,
-    const __m128 wkr_ =
-      _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
-    const __m128 wki_ = c_j1;                              //  1,  2,  3,  4,
-    // Load and shuffle 'a'.
-    const __m128 a_j2_0 = _mm_loadu_ps(&a[0   + j2]);  //   2,   3,   4,   5,
-    const __m128 a_j2_4 = _mm_loadu_ps(&a[4   + j2]);  //   6,   7,   8,   9,
-    const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]);  // 120, 121, 122, 123,
-    const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]);  // 124, 125, 126, 127,
-    const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
-                            _MM_SHUFFLE(2, 0, 2 ,0));  //   2,   4,   6,   8,
-    const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
-                            _MM_SHUFFLE(3, 1, 3 ,1));  //   3,   5,   7,   9,
-    const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
-                            _MM_SHUFFLE(0, 2, 0 ,2));  // 126, 124, 122, 120,
-    const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
-                            _MM_SHUFFLE(1, 3, 1 ,3));  // 127, 125, 123, 121,
-    // Calculate 'x'.
-    const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
-                                               // 2-126, 4-124, 6-122, 8-120,
-    const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
-                                               // 3-127, 5-125, 7-123, 9-121,
-    // Calculate product into 'y'.
-    //    yr = wkr * xr - wki * xi;
-    //    yi = wkr * xi + wki * xr;
-    const __m128 a_ = _mm_mul_ps(wkr_, xr_);
-    const __m128 b_ = _mm_mul_ps(wki_, xi_);
-    const __m128 c_ = _mm_mul_ps(wkr_, xi_);
-    const __m128 d_ = _mm_mul_ps(wki_, xr_);
-    const __m128 yr_ = _mm_sub_ps(a_, b_);     // 2-126, 4-124, 6-122, 8-120,
-    const __m128 yi_ = _mm_add_ps(c_, d_);     // 3-127, 5-125, 7-123, 9-121,
-    // Update 'a'.
-    //    a[j2 + 0] -= yr;
-    //    a[j2 + 1] -= yi;
-    //    a[k2 + 0] += yr;
-    //    a[k2 + 1] -= yi;
-    const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_);  //   2,   4,   6,   8,
-    const __m128 a_j2_p1n = _mm_sub_ps(a_j2_p1, yi_);  //   3,   5,   7,   9,
-    const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_);  // 126, 124, 122, 120,
-    const __m128 a_k2_p1n = _mm_sub_ps(a_k2_p1, yi_);  // 127, 125, 123, 121,
-    // Shuffle in right order and store.
-    const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
-                                                       //   2,   3,   4,   5,
-    const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
-                                                       //   6,   7,   8,   9,
-    const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
-                                                       // 122, 123, 120, 121,
-    const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
-                                                       // 126, 127, 124, 125,
-    const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
-                            _MM_SHUFFLE(1, 0, 3 ,2));  // 120, 121, 122, 123,
-    const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
-                            _MM_SHUFFLE(1, 0, 3 ,2));  // 124, 125, 126, 127,
-    _mm_storeu_ps(&a[0   + j2], a_j2_0n);
-    _mm_storeu_ps(&a[4   + j2], a_j2_4n);
-    _mm_storeu_ps(&a[122 - j2], a_k2_0n);
-    _mm_storeu_ps(&a[126 - j2], a_k2_4n);
-  }
-  // Scalar code for the remaining items.
-  for (; j2 < 64; j1 += 1, j2 += 2) {
-    k2 = 128 - j2;
-    k1 =  32 - j1;
-    wkr = 0.5f - c[k1];
-    wki = c[j1];
-    xr = a[j2 + 0] - a[k2 + 0];
-    xi = a[j2 + 1] + a[k2 + 1];
-    yr = wkr * xr - wki * xi;
-    yi = wkr * xi + wki * xr;
-    a[j2 + 0] -= yr;
-    a[j2 + 1] -= yi;
-    a[k2 + 0] += yr;
-    a[k2 + 1] -= yi;
-  }
-}
-
-static void rftbsub_128_SSE2(float *a) {
-  const float *c = rdft_w + 32;
-  int j1, j2, k1, k2;
-  float wkr, wki, xr, xi, yr, yi;
-
-  static const ALIGN16_BEG float ALIGN16_END k_half[4] =
-    {0.5f, 0.5f, 0.5f, 0.5f};
-  const __m128 mm_half = _mm_load_ps(k_half);
-
-  a[1] = -a[1];
-  // Vectorized code (four at once).
-  //    Note: commented number are indexes for the first iteration of the loop.
-  for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
-    // Load 'wk'.
-    const __m128 c_j1 = _mm_loadu_ps(&c[     j1]);         //  1,  2,  3,  4,
-    const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]);         // 28, 29, 30, 31,
-    const __m128 wkrt = _mm_sub_ps(mm_half, c_k1);         // 28, 29, 30, 31,
-    const __m128 wkr_ =
-      _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
-    const __m128 wki_ = c_j1;                              //  1,  2,  3,  4,
-    // Load and shuffle 'a'.
-    const __m128 a_j2_0 = _mm_loadu_ps(&a[0   + j2]);  //   2,   3,   4,   5,
-    const __m128 a_j2_4 = _mm_loadu_ps(&a[4   + j2]);  //   6,   7,   8,   9,
-    const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]);  // 120, 121, 122, 123,
-    const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]);  // 124, 125, 126, 127,
-    const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
-                            _MM_SHUFFLE(2, 0, 2 ,0));  //   2,   4,   6,   8,
-    const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
-                            _MM_SHUFFLE(3, 1, 3 ,1));  //   3,   5,   7,   9,
-    const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
-                            _MM_SHUFFLE(0, 2, 0 ,2));  // 126, 124, 122, 120,
-    const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
-                            _MM_SHUFFLE(1, 3, 1 ,3));  // 127, 125, 123, 121,
-    // Calculate 'x'.
-    const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
-                                               // 2-126, 4-124, 6-122, 8-120,
-    const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
-                                               // 3-127, 5-125, 7-123, 9-121,
-    // Calculate product into 'y'.
-    //    yr = wkr * xr + wki * xi;
-    //    yi = wkr * xi - wki * xr;
-    const __m128 a_ = _mm_mul_ps(wkr_, xr_);
-    const __m128 b_ = _mm_mul_ps(wki_, xi_);
-    const __m128 c_ = _mm_mul_ps(wkr_, xi_);
-    const __m128 d_ = _mm_mul_ps(wki_, xr_);
-    const __m128 yr_ = _mm_add_ps(a_, b_);     // 2-126, 4-124, 6-122, 8-120,
-    const __m128 yi_ = _mm_sub_ps(c_, d_);     // 3-127, 5-125, 7-123, 9-121,
-    // Update 'a'.
-    //    a[j2 + 0] = a[j2 + 0] - yr;
-    //    a[j2 + 1] = yi - a[j2 + 1];
-    //    a[k2 + 0] = yr + a[k2 + 0];
-    //    a[k2 + 1] = yi - a[k2 + 1];
-    const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_);  //   2,   4,   6,   8,
-    const __m128 a_j2_p1n = _mm_sub_ps(yi_, a_j2_p1);  //   3,   5,   7,   9,
-    const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_);  // 126, 124, 122, 120,
-    const __m128 a_k2_p1n = _mm_sub_ps(yi_, a_k2_p1);  // 127, 125, 123, 121,
-    // Shuffle in right order and store.
-    // Shuffle in right order and store.
-    const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
-                                                       //   2,   3,   4,   5,
-    const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
-                                                       //   6,   7,   8,   9,
-    const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
-                                                       // 122, 123, 120, 121,
-    const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
-                                                       // 126, 127, 124, 125,
-    const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
-                            _MM_SHUFFLE(1, 0, 3 ,2));  // 120, 121, 122, 123,
-    const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
-                            _MM_SHUFFLE(1, 0, 3 ,2));  // 124, 125, 126, 127,
-    _mm_storeu_ps(&a[0   + j2], a_j2_0n);
-    _mm_storeu_ps(&a[4   + j2], a_j2_4n);
-    _mm_storeu_ps(&a[122 - j2], a_k2_0n);
-    _mm_storeu_ps(&a[126 - j2], a_k2_4n);
-  }
-  // Scalar code for the remaining items.
-  for (; j2 < 64; j1 += 1, j2 += 2) {
-    k2 = 128 - j2;
-    k1 =  32 - j1;
-    wkr = 0.5f - c[k1];
-    wki = c[j1];
-    xr = a[j2 + 0] - a[k2 + 0];
-    xi = a[j2 + 1] + a[k2 + 1];
-    yr = wkr * xr + wki * xi;
-    yi = wkr * xi - wki * xr;
-    a[j2 + 0] = a[j2 + 0] - yr;
-    a[j2 + 1] = yi - a[j2 + 1];
-    a[k2 + 0] = yr + a[k2 + 0];
-    a[k2 + 1] = yi - a[k2 + 1];
-  }
-  a[65] = -a[65];
-}
-
-void aec_rdft_init_sse2(void) {
-  rftfsub_128 = rftfsub_128_SSE2;
-  rftbsub_128 = rftbsub_128_SSE2;
-}
diff --git a/src/modules/audio_processing/aec/main/source/echo_cancellation.c b/src/modules/audio_processing/aec/main/source/echo_cancellation.c
deleted file mode 100644
index 1313e35..0000000
--- a/src/modules/audio_processing/aec/main/source/echo_cancellation.c
+++ /dev/null
@@ -1,821 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Contains the API functions for the AEC.
- */
-#include <stdlib.h>
-#include <string.h>
-
-#include "echo_cancellation.h"
-#include "aec_core.h"
-#include "ring_buffer.h"
-#include "resampler.h"
-#ifdef AEC_DEBUG
-    #include <stdio.h>
-#endif
-
-#define BUF_SIZE_FRAMES 50 // buffer size (frames)
-// Maximum length of resampled signal. Must be an integer multiple of frames
-// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
-// The factor of 2 handles wb, and the + 1 is as a safety margin
-#define MAX_RESAMP_LEN (5 * FRAME_LEN)
-
-static const int bufSizeSamp = BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
-static const int sampMsNb = 8; // samples per ms in nb
-// Target suppression levels for nlp modes
-// log{0.001, 0.00001, 0.00000001}
-static const float targetSupp[3] = {-6.9f, -11.5f, -18.4f};
-static const float minOverDrive[3] = {1.0f, 2.0f, 5.0f};
-static const int initCheck = 42;
-
-typedef struct {
-    int delayCtr;
-    int sampFreq;
-    int splitSampFreq;
-    int scSampFreq;
-    float sampFactor; // scSampRate / sampFreq
-    short nlpMode;
-    short autoOnOff;
-    short activity;
-    short skewMode;
-    short bufSizeStart;
-    //short bufResetCtr;  // counts number of noncausal frames
-    int knownDelay;
-
-    // Stores the last frame added to the farend buffer
-    short farendOld[2][FRAME_LEN];
-    short initFlag; // indicates if AEC has been initialized
-
-    // Variables used for averaging far end buffer size
-    short counter;
-    short sum;
-    short firstVal;
-    short checkBufSizeCtr;
-
-    // Variables used for delay shifts
-    short msInSndCardBuf;
-    short filtDelay;
-    int timeForDelayChange;
-    int ECstartup;
-    int checkBuffSize;
-    int delayChange;
-    short lastDelayDiff;
-
-#ifdef AEC_DEBUG
-    FILE *bufFile;
-    FILE *delayFile;
-    FILE *skewFile;
-    FILE *preCompFile;
-    FILE *postCompFile;
-#endif // AEC_DEBUG
-
-    // Structures
-    void *farendBuf;
-    void *resampler;
-
-    int skewFrCtr;
-    int resample; // if the skew is small enough we don't resample
-    int highSkewCtr;
-    float skew;
-
-    int lastError;
-
-    aec_t *aec;
-} aecpc_t;
-
-// Estimates delay to set the position of the farend buffer read pointer
-// (controlled by knownDelay)
-static int EstBufDelay(aecpc_t *aecInst, short msInSndCardBuf);
-
-// Stuffs the farend buffer if the estimated delay is too large
-static int DelayComp(aecpc_t *aecInst);
-
-WebRtc_Word32 WebRtcAec_Create(void **aecInst)
-{
-    aecpc_t *aecpc;
-    if (aecInst == NULL) {
-        return -1;
-    }
-
-    aecpc = malloc(sizeof(aecpc_t));
-    *aecInst = aecpc;
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (WebRtcAec_CreateAec(&aecpc->aec) == -1) {
-        WebRtcAec_Free(aecpc);
-        aecpc = NULL;
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aecpc->farendBuf, bufSizeSamp) == -1) {
-        WebRtcAec_Free(aecpc);
-        aecpc = NULL;
-        return -1;
-    }
-
-    if (WebRtcAec_CreateResampler(&aecpc->resampler) == -1) {
-        WebRtcAec_Free(aecpc);
-        aecpc = NULL;
-        return -1;
-    }
-
-    aecpc->initFlag = 0;
-    aecpc->lastError = 0;
-
-#ifdef AEC_DEBUG
-    aecpc->aec->farFile = fopen("aecFar.pcm","wb");
-    aecpc->aec->nearFile = fopen("aecNear.pcm","wb");
-    aecpc->aec->outFile = fopen("aecOut.pcm","wb");
-    aecpc->aec->outLpFile = fopen("aecOutLp.pcm","wb");
-
-    aecpc->bufFile = fopen("aecBuf.dat", "wb");
-    aecpc->skewFile = fopen("aecSkew.dat", "wb");
-    aecpc->delayFile = fopen("aecDelay.dat", "wb");
-    aecpc->preCompFile = fopen("preComp.pcm", "wb");
-    aecpc->postCompFile = fopen("postComp.pcm", "wb");
-#endif // AEC_DEBUG
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_Free(void *aecInst)
-{
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-#ifdef AEC_DEBUG
-    fclose(aecpc->aec->farFile);
-    fclose(aecpc->aec->nearFile);
-    fclose(aecpc->aec->outFile);
-    fclose(aecpc->aec->outLpFile);
-
-    fclose(aecpc->bufFile);
-    fclose(aecpc->skewFile);
-    fclose(aecpc->delayFile);
-    fclose(aecpc->preCompFile);
-    fclose(aecpc->postCompFile);
-#endif // AEC_DEBUG
-
-    WebRtcAec_FreeAec(aecpc->aec);
-    WebRtcApm_FreeBuffer(aecpc->farendBuf);
-    WebRtcAec_FreeResampler(aecpc->resampler);
-    free(aecpc);
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_Init(void *aecInst, WebRtc_Word32 sampFreq, WebRtc_Word32 scSampFreq)
-{
-    aecpc_t *aecpc = aecInst;
-    AecConfig aecConfig;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (sampFreq != 8000 && sampFreq != 16000  && sampFreq != 32000) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecpc->sampFreq = sampFreq;
-
-    if (scSampFreq < 1 || scSampFreq > 96000) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecpc->scSampFreq = scSampFreq;
-
-    // Initialize echo canceller core
-    if (WebRtcAec_InitAec(aecpc->aec, aecpc->sampFreq) == -1) {
-        aecpc->lastError = AEC_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    // Initialize farend buffer
-    if (WebRtcApm_InitBuffer(aecpc->farendBuf) == -1) {
-        aecpc->lastError = AEC_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) {
-        aecpc->lastError = AEC_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    aecpc->initFlag = initCheck;  // indicates that initilisation has been done
-
-    if (aecpc->sampFreq == 32000) {
-        aecpc->splitSampFreq = 16000;
-    }
-    else {
-        aecpc->splitSampFreq = sampFreq;
-    }
-
-    aecpc->skewFrCtr = 0;
-    aecpc->activity = 0;
-
-    aecpc->delayChange = 1;
-    aecpc->delayCtr = 0;
-
-    aecpc->sum = 0;
-    aecpc->counter = 0;
-    aecpc->checkBuffSize = 1;
-    aecpc->firstVal = 0;
-
-    aecpc->ECstartup = 1;
-    aecpc->bufSizeStart = 0;
-    aecpc->checkBufSizeCtr = 0;
-    aecpc->filtDelay = 0;
-    aecpc->timeForDelayChange =0;
-    aecpc->knownDelay = 0;
-    aecpc->lastDelayDiff = 0;
-
-    aecpc->skew = 0;
-    aecpc->resample = kAecFalse;
-    aecpc->highSkewCtr = 0;
-    aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq;
-
-    memset(&aecpc->farendOld[0][0], 0, 160);
-
-    // Default settings.
-    aecConfig.nlpMode = kAecNlpModerate;
-    aecConfig.skewMode = kAecFalse;
-    aecConfig.metricsMode = kAecFalse;
-
-    if (WebRtcAec_set_config(aecpc, aecConfig) == -1) {
-        aecpc->lastError = AEC_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    return 0;
-}
-
-// only buffer L band for farend
-WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst, const WebRtc_Word16 *farend,
-    WebRtc_Word16 nrOfSamples)
-{
-    aecpc_t *aecpc = aecInst;
-    WebRtc_Word32 retVal = 0;
-    short newNrOfSamples;
-    short newFarend[MAX_RESAMP_LEN];
-    float skew;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (farend == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    // number of samples == 160 for SWB input
-    if (nrOfSamples != 80 && nrOfSamples != 160) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-
-    skew = aecpc->skew;
-
-    // TODO: Is this really a good idea?
-    if (!aecpc->ECstartup) {
-        DelayComp(aecpc);
-    }
-
-    if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
-        // Resample and get a new number of samples
-        newNrOfSamples = WebRtcAec_ResampleLinear(aecpc->resampler,
-                                                  farend,
-                                                  nrOfSamples,
-                                                  skew,
-                                                  newFarend);
-        WebRtcApm_WriteBuffer(aecpc->farendBuf, newFarend, newNrOfSamples);
-
-#ifdef AEC_DEBUG
-        fwrite(farend, 2, nrOfSamples, aecpc->preCompFile);
-        fwrite(newFarend, 2, newNrOfSamples, aecpc->postCompFile);
-#endif
-    }
-    else {
-        WebRtcApm_WriteBuffer(aecpc->farendBuf, farend, nrOfSamples);
-    }
-
-    return retVal;
-}
-
-WebRtc_Word32 WebRtcAec_Process(void *aecInst, const WebRtc_Word16 *nearend,
-    const WebRtc_Word16 *nearendH, WebRtc_Word16 *out, WebRtc_Word16 *outH,
-    WebRtc_Word16 nrOfSamples, WebRtc_Word16 msInSndCardBuf, WebRtc_Word32 skew)
-{
-    aecpc_t *aecpc = aecInst;
-    WebRtc_Word32 retVal = 0;
-    short i;
-    short farend[FRAME_LEN];
-    short nmbrOfFilledBuffers;
-    short nBlocks10ms;
-    short nFrames;
-#ifdef AEC_DEBUG
-    short msInAECBuf;
-#endif
-    // Limit resampling to doubling/halving of signal
-    const float minSkewEst = -0.5f;
-    const float maxSkewEst = 1.0f;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (nearend == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (out == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    // number of samples == 160 for SWB input
-    if (nrOfSamples != 80 && nrOfSamples != 160) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-
-    // Check for valid pointers based on sampling rate
-    if (aecpc->sampFreq == 32000 && nearendH == NULL) {
-       aecpc->lastError = AEC_NULL_POINTER_ERROR;
-       return -1;
-    }
-
-    if (msInSndCardBuf < 0) {
-        msInSndCardBuf = 0;
-        aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
-        retVal = -1;
-    }
-    else if (msInSndCardBuf > 500) {
-        msInSndCardBuf = 500;
-        aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
-        retVal = -1;
-    }
-    msInSndCardBuf += 10;
-    aecpc->msInSndCardBuf = msInSndCardBuf;
-
-    if (aecpc->skewMode == kAecTrue) {
-        if (aecpc->skewFrCtr < 25) {
-            aecpc->skewFrCtr++;
-        }
-        else {
-            retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
-            if (retVal == -1) {
-                aecpc->skew = 0;
-                aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
-            }
-
-            aecpc->skew /= aecpc->sampFactor*nrOfSamples;
-
-            if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) {
-                aecpc->resample = kAecFalse;
-            }
-            else {
-                aecpc->resample = kAecTrue;
-            }
-
-            if (aecpc->skew < minSkewEst) {
-                aecpc->skew = minSkewEst;
-            }
-            else if (aecpc->skew > maxSkewEst) {
-                aecpc->skew = maxSkewEst;
-            }
-
-#ifdef AEC_DEBUG
-            fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
-#endif
-        }
-    }
-
-    nFrames = nrOfSamples / FRAME_LEN;
-    nBlocks10ms = nFrames / aecpc->aec->mult;
-
-    if (aecpc->ECstartup) {
-        memcpy(out, nearend, sizeof(short) * nrOfSamples);
-        nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecpc->farendBuf) / FRAME_LEN;
-
-        // The AEC is in the start up mode
-        // AEC is disabled until the soundcard buffer and farend buffers are OK
-
-        // Mechanism to ensure that the soundcard buffer is reasonably stable.
-        if (aecpc->checkBuffSize) {
-
-            aecpc->checkBufSizeCtr++;
-            // Before we fill up the far end buffer we require the amount of data on the
-            // sound card to be stable (+/-8 ms) compared to the first value. This
-            // comparison is made during the following 4 consecutive frames. If it seems
-            // to be stable then we start to fill up the far end buffer.
-
-            if (aecpc->counter == 0) {
-                aecpc->firstVal = aecpc->msInSndCardBuf;
-                aecpc->sum = 0;
-            }
-
-            if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) <
-                WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) {
-                aecpc->sum += aecpc->msInSndCardBuf;
-                aecpc->counter++;
-            }
-            else {
-                aecpc->counter = 0;
-            }
-
-            if (aecpc->counter*nBlocks10ms >= 6) {
-                // The farend buffer size is determined in blocks of 80 samples
-                // Use 75% of the average value of the soundcard buffer
-                aecpc->bufSizeStart = WEBRTC_SPL_MIN((int) (0.75 * (aecpc->sum *
-                    aecpc->aec->mult) / (aecpc->counter * 10)), BUF_SIZE_FRAMES);
-                // buffersize has now been determined
-                aecpc->checkBuffSize = 0;
-            }
-
-            if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) {
-                // for really bad sound cards, don't disable echocanceller for more than 0.5 sec
-                aecpc->bufSizeStart = WEBRTC_SPL_MIN((int) (0.75 * (aecpc->msInSndCardBuf *
-                    aecpc->aec->mult) / 10), BUF_SIZE_FRAMES);
-                aecpc->checkBuffSize = 0;
-            }
-        }
-
-        // if checkBuffSize changed in the if-statement above
-        if (!aecpc->checkBuffSize) {
-            // soundcard buffer is now reasonably stable
-            // When the far end buffer is filled with approximately the same amount of
-            // data as the amount on the sound card we end the start up phase and start
-            // to cancel echoes.
-
-            if (nmbrOfFilledBuffers == aecpc->bufSizeStart) {
-                aecpc->ECstartup = 0;  // Enable the AEC
-            }
-            else if (nmbrOfFilledBuffers > aecpc->bufSizeStart) {
-                WebRtcApm_FlushBuffer(aecpc->farendBuf, WebRtcApm_get_buffer_size(aecpc->farendBuf) -
-                    aecpc->bufSizeStart * FRAME_LEN);
-                aecpc->ECstartup = 0;
-            }
-        }
-
-    }
-    else {
-        // AEC is enabled
-
-        // Note only 1 block supported for nb and 2 blocks for wb
-        for (i = 0; i < nFrames; i++) {
-            nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecpc->farendBuf) / FRAME_LEN;
-
-            // Check that there is data in the far end buffer
-            if (nmbrOfFilledBuffers > 0) {
-                // Get the next 80 samples from the farend buffer
-                WebRtcApm_ReadBuffer(aecpc->farendBuf, farend, FRAME_LEN);
-
-                // Always store the last frame for use when we run out of data
-                memcpy(&(aecpc->farendOld[i][0]), farend, FRAME_LEN * sizeof(short));
-            }
-            else {
-                // We have no data so we use the last played frame
-                memcpy(farend, &(aecpc->farendOld[i][0]), FRAME_LEN * sizeof(short));
-            }
-
-            // Call buffer delay estimator when all data is extracted,
-            // i.e. i = 0 for NB and i = 1 for WB or SWB
-            if ((i == 0 && aecpc->splitSampFreq == 8000) ||
-                    (i == 1 && (aecpc->splitSampFreq == 16000))) {
-                EstBufDelay(aecpc, aecpc->msInSndCardBuf);
-            }
-
-            // Call the AEC
-           WebRtcAec_ProcessFrame(aecpc->aec, farend, &nearend[FRAME_LEN * i], &nearendH[FRAME_LEN * i],
-               &out[FRAME_LEN * i], &outH[FRAME_LEN * i], aecpc->knownDelay);
-        }
-    }
-
-#ifdef AEC_DEBUG
-    msInAECBuf = WebRtcApm_get_buffer_size(aecpc->farendBuf) / (sampMsNb*aecpc->aec->mult);
-    fwrite(&msInAECBuf, 2, 1, aecpc->bufFile);
-    fwrite(&(aecpc->knownDelay), sizeof(aecpc->knownDelay), 1, aecpc->delayFile);
-#endif
-
-    return retVal;
-}
-
-WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config)
-{
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    if (config.skewMode != kAecFalse && config.skewMode != kAecTrue) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecpc->skewMode = config.skewMode;
-
-    if (config.nlpMode != kAecNlpConservative && config.nlpMode !=
-            kAecNlpModerate && config.nlpMode != kAecNlpAggressive) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecpc->nlpMode = config.nlpMode;
-    aecpc->aec->targetSupp = targetSupp[aecpc->nlpMode];
-    aecpc->aec->minOverDrive = minOverDrive[aecpc->nlpMode];
-
-    if (config.metricsMode != kAecFalse && config.metricsMode != kAecTrue) {
-        aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecpc->aec->metricsMode = config.metricsMode;
-    if (aecpc->aec->metricsMode == kAecTrue) {
-        WebRtcAec_InitMetrics(aecpc->aec);
-    }
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config)
-{
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (config == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    config->nlpMode = aecpc->nlpMode;
-    config->skewMode = aecpc->skewMode;
-    config->metricsMode = aecpc->aec->metricsMode;
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status)
-{
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (status == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    *status = aecpc->aec->echoState;
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics)
-{
-    const float upweight = 0.7f;
-    float dtmp;
-    short stmp;
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    if (metrics == NULL) {
-        aecpc->lastError = AEC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecpc->initFlag != initCheck) {
-        aecpc->lastError = AEC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    // ERL
-    metrics->erl.instant = (short) aecpc->aec->erl.instant;
-
-    if ((aecpc->aec->erl.himean > offsetLevel) && (aecpc->aec->erl.average > offsetLevel)) {
-    // Use a mix between regular average and upper part average
-        dtmp = upweight * aecpc->aec->erl.himean + (1 - upweight) * aecpc->aec->erl.average;
-        metrics->erl.average = (short) dtmp;
-    }
-    else {
-        metrics->erl.average = offsetLevel;
-    }
-
-    metrics->erl.max = (short) aecpc->aec->erl.max;
-
-    if (aecpc->aec->erl.min < (offsetLevel * (-1))) {
-        metrics->erl.min = (short) aecpc->aec->erl.min;
-    }
-    else {
-        metrics->erl.min = offsetLevel;
-    }
-
-    // ERLE
-    metrics->erle.instant = (short) aecpc->aec->erle.instant;
-
-    if ((aecpc->aec->erle.himean > offsetLevel) && (aecpc->aec->erle.average > offsetLevel)) {
-        // Use a mix between regular average and upper part average
-        dtmp =  upweight * aecpc->aec->erle.himean + (1 - upweight) * aecpc->aec->erle.average;
-        metrics->erle.average = (short) dtmp;
-    }
-    else {
-        metrics->erle.average = offsetLevel;
-    }
-
-    metrics->erle.max = (short) aecpc->aec->erle.max;
-
-    if (aecpc->aec->erle.min < (offsetLevel * (-1))) {
-        metrics->erle.min = (short) aecpc->aec->erle.min;
-    } else {
-        metrics->erle.min = offsetLevel;
-    }
-
-    // RERL
-    if ((metrics->erl.average > offsetLevel) && (metrics->erle.average > offsetLevel)) {
-        stmp = metrics->erl.average + metrics->erle.average;
-    }
-    else {
-        stmp = offsetLevel;
-    }
-    metrics->rerl.average = stmp;
-
-    // No other statistics needed, but returned for completeness
-    metrics->rerl.instant = stmp;
-    metrics->rerl.max = stmp;
-    metrics->rerl.min = stmp;
-
-    // A_NLP
-    metrics->aNlp.instant = (short) aecpc->aec->aNlp.instant;
-
-    if ((aecpc->aec->aNlp.himean > offsetLevel) && (aecpc->aec->aNlp.average > offsetLevel)) {
-        // Use a mix between regular average and upper part average
-        dtmp =  upweight * aecpc->aec->aNlp.himean + (1 - upweight) * aecpc->aec->aNlp.average;
-        metrics->aNlp.average = (short) dtmp;
-    }
-    else {
-        metrics->aNlp.average = offsetLevel;
-    }
-
-    metrics->aNlp.max = (short) aecpc->aec->aNlp.max;
-
-    if (aecpc->aec->aNlp.min < (offsetLevel * (-1))) {
-        metrics->aNlp.min = (short) aecpc->aec->aNlp.min;
-    }
-    else {
-        metrics->aNlp.min = offsetLevel;
-    }
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len)
-{
-    const char version[] = "AEC 2.5.0";
-    const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
-
-    if (versionStr == NULL) {
-        return -1;
-    }
-
-    if (versionLen > len) {
-        return -1;
-    }
-
-    strncpy(versionStr, version, versionLen);
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst)
-{
-    aecpc_t *aecpc = aecInst;
-
-    if (aecpc == NULL) {
-        return -1;
-    }
-
-    return aecpc->lastError;
-}
-
-static int EstBufDelay(aecpc_t *aecpc, short msInSndCardBuf)
-{
-    short delayNew, nSampFar, nSampSndCard;
-    short diff;
-
-    nSampFar = WebRtcApm_get_buffer_size(aecpc->farendBuf);
-    nSampSndCard = msInSndCardBuf * sampMsNb * aecpc->aec->mult;
-
-    delayNew = nSampSndCard - nSampFar;
-
-    // Account for resampling frame delay
-    if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
-        delayNew -= kResamplingDelay;
-    }
-
-    if (delayNew < FRAME_LEN) {
-        WebRtcApm_FlushBuffer(aecpc->farendBuf, FRAME_LEN);
-        delayNew += FRAME_LEN;
-    }
-
-    aecpc->filtDelay = WEBRTC_SPL_MAX(0, (short)(0.8*aecpc->filtDelay + 0.2*delayNew));
-
-    diff = aecpc->filtDelay - aecpc->knownDelay;
-    if (diff > 224) {
-        if (aecpc->lastDelayDiff < 96) {
-            aecpc->timeForDelayChange = 0;
-        }
-        else {
-            aecpc->timeForDelayChange++;
-        }
-    }
-    else if (diff < 96 && aecpc->knownDelay > 0) {
-        if (aecpc->lastDelayDiff > 224) {
-            aecpc->timeForDelayChange = 0;
-        }
-        else {
-            aecpc->timeForDelayChange++;
-        }
-    }
-    else {
-        aecpc->timeForDelayChange = 0;
-    }
-    aecpc->lastDelayDiff = diff;
-
-    if (aecpc->timeForDelayChange > 25) {
-        aecpc->knownDelay = WEBRTC_SPL_MAX((int)aecpc->filtDelay - 160, 0);
-    }
-    return 0;
-}
-
-static int DelayComp(aecpc_t *aecpc)
-{
-    int nSampFar, nSampSndCard, delayNew, nSampAdd;
-    const int maxStuffSamp = 10 * FRAME_LEN;
-
-    nSampFar = WebRtcApm_get_buffer_size(aecpc->farendBuf);
-    nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->aec->mult;
-    delayNew = nSampSndCard - nSampFar;
-
-    // Account for resampling frame delay
-    if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
-        delayNew -= kResamplingDelay;
-    }
-
-    if (delayNew > FAR_BUF_LEN - FRAME_LEN*aecpc->aec->mult) {
-        // The difference of the buffersizes is larger than the maximum
-        // allowed known delay. Compensate by stuffing the buffer.
-        nSampAdd = (int)(WEBRTC_SPL_MAX((int)(0.5 * nSampSndCard - nSampFar),
-                    FRAME_LEN));
-        nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
-
-        WebRtcApm_StuffBuffer(aecpc->farendBuf, nSampAdd);
-        aecpc->delayChange = 1; // the delay needs to be updated
-    }
-
-    return 0;
-}
diff --git a/src/modules/audio_processing/aec/main/source/resampler.c b/src/modules/audio_processing/aec/main/source/resampler.c
deleted file mode 100644
index 4caa6f4..0000000
--- a/src/modules/audio_processing/aec/main/source/resampler.c
+++ /dev/null
@@ -1,235 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/* Resamples a signal to an arbitrary rate. Used by the AEC to compensate for clock
- * skew by resampling the farend signal.
- */
-
-#include <assert.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-
-#include "resampler.h"
-#include "aec_core.h"
-
-enum { kFrameBufferSize = FRAME_LEN * 4 };
-enum { kEstimateLengthFrames = 400 };
-
-typedef struct {
-    short buffer[kFrameBufferSize];
-    float position;
-
-    int deviceSampleRateHz;
-    int skewData[kEstimateLengthFrames];
-    int skewDataIndex;
-    float skewEstimate;
-} resampler_t;
-
-static int EstimateSkew(const int* rawSkew,
-                        int size,
-                        int absLimit,
-                        float *skewEst);
-
-int WebRtcAec_CreateResampler(void **resampInst)
-{
-    resampler_t *obj = malloc(sizeof(resampler_t));
-    *resampInst = obj;
-    if (obj == NULL) {
-        return -1;
-    }
-
-    return 0;
-}
-
-int WebRtcAec_InitResampler(void *resampInst, int deviceSampleRateHz)
-{
-    resampler_t *obj = (resampler_t*) resampInst;
-    memset(obj->buffer, 0, sizeof(obj->buffer));
-    obj->position = 0.0;
-
-    obj->deviceSampleRateHz = deviceSampleRateHz;
-    memset(obj->skewData, 0, sizeof(obj->skewData));
-    obj->skewDataIndex = 0;
-    obj->skewEstimate = 0.0;
-
-    return 0;
-}
-
-int WebRtcAec_FreeResampler(void *resampInst)
-{
-    resampler_t *obj = (resampler_t*) resampInst;
-    free(obj);
-
-    return 0;
-}
-
-int WebRtcAec_ResampleLinear(void *resampInst,
-                             const short *inspeech,
-                             int size,
-                             float skew,
-                             short *outspeech)
-{
-    resampler_t *obj = (resampler_t*) resampInst;
-
-    short *y;
-    float be, tnew, interp;
-    int tn, outsize, mm;
-
-    if (size < 0 || size > 2 * FRAME_LEN) {
-        return -1;
-    }
-
-    // Add new frame data in lookahead
-    memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay],
-           inspeech,
-           size * sizeof(short));
-
-    // Sample rate ratio
-    be = 1 + skew;
-
-    // Loop over input frame
-    mm = 0;
-    y = &obj->buffer[FRAME_LEN]; // Point at current frame
-
-    tnew = be * mm + obj->position;
-    tn = (int) tnew;
-
-    while (tn < size) {
-
-        // Interpolation
-        interp = y[tn] + (tnew - tn) * (y[tn+1] - y[tn]);
-
-        if (interp > 32767) {
-            interp = 32767;
-        }
-        else if (interp < -32768) {
-            interp = -32768;
-        }
-
-        outspeech[mm] = (short) interp;
-        mm++;
-
-        tnew = be * mm + obj->position;
-        tn = (int) tnew;
-    }
-
-    outsize = mm;
-    obj->position += outsize * be - size;
-
-    // Shift buffer
-    memmove(obj->buffer,
-            &obj->buffer[size],
-            (kFrameBufferSize - size) * sizeof(short));
-
-    return outsize;
-}
-
-int WebRtcAec_GetSkew(void *resampInst, int rawSkew, float *skewEst)
-{
-    resampler_t *obj = (resampler_t*)resampInst;
-    int err = 0;
-
-    if (obj->skewDataIndex < kEstimateLengthFrames) {
-        obj->skewData[obj->skewDataIndex] = rawSkew;
-        obj->skewDataIndex++;
-    }
-    else if (obj->skewDataIndex == kEstimateLengthFrames) {
-        err = EstimateSkew(obj->skewData,
-                           kEstimateLengthFrames,
-                           obj->deviceSampleRateHz,
-                           skewEst);
-        obj->skewEstimate = *skewEst;
-        obj->skewDataIndex++;
-    }
-    else {
-        *skewEst = obj->skewEstimate;
-    }
-
-    return err;
-}
-
-int EstimateSkew(const int* rawSkew,
-                 const int size,
-                 const int deviceSampleRateHz,
-                 float *skewEst)
-{
-    const int absLimitOuter = (int)(0.04f * deviceSampleRateHz);
-    const int absLimitInner = (int)(0.0025f * deviceSampleRateHz);
-    int i = 0;
-    int n = 0;
-    float rawAvg = 0;
-    float err = 0;
-    float rawAbsDev = 0;
-    int upperLimit = 0;
-    int lowerLimit = 0;
-    float cumSum = 0;
-    float x = 0;
-    float x2 = 0;
-    float y = 0;
-    float xy = 0;
-    float xAvg = 0;
-    float yAvg = 0;
-    float denom = 0;
-    float skew = 0;
-
-    *skewEst = 0; // Set in case of error below.
-    for (i = 0; i < size; i++) {
-      if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
-        n++;
-        rawAvg += rawSkew[i];
-      }
-    }
-
-    if (n == 0) {
-      return -1;
-    }
-    assert(n > 0);
-    rawAvg /= n;
-
-    for (i = 0; i < size; i++) {
-      if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
-        err = rawSkew[i] - rawAvg;
-        rawAbsDev += err >= 0 ? err : -err;
-      }
-    }
-    assert(n > 0);
-    rawAbsDev /= n;
-    upperLimit = (int)(rawAvg + 5 * rawAbsDev + 1); // +1 for ceiling.
-    lowerLimit = (int)(rawAvg - 5 * rawAbsDev - 1); // -1 for floor.
-
-    n = 0;
-    for (i = 0; i < size; i++) {
-        if ((rawSkew[i] < absLimitInner && rawSkew[i] > -absLimitInner) ||
-            (rawSkew[i] < upperLimit && rawSkew[i] > lowerLimit)) {
-            n++;
-            cumSum += rawSkew[i];
-            x += n;
-            x2 += n*n;
-            y += cumSum;
-            xy += n * cumSum;
-        }
-    }
-
-    if (n == 0) {
-      return -1;
-    }
-    assert(n > 0);
-    xAvg = x / n;
-    yAvg = y / n;
-    denom = x2 - xAvg*x;
-
-    if (denom != 0) {
-        skew = (xy - xAvg*y) / denom;
-    }
-
-    *skewEst = skew;
-    return 0;
-}
diff --git a/src/modules/audio_processing/aec/main/source/resampler.h b/src/modules/audio_processing/aec/main/source/resampler.h
deleted file mode 100644
index 9cb2837..0000000
--- a/src/modules/audio_processing/aec/main/source/resampler.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_RESAMPLER_H_
-
-enum { kResamplingDelay = 1 };
-
-// Unless otherwise specified, functions return 0 on success and -1 on error
-int WebRtcAec_CreateResampler(void **resampInst);
-int WebRtcAec_InitResampler(void *resampInst, int deviceSampleRateHz);
-int WebRtcAec_FreeResampler(void *resampInst);
-
-// Estimates skew from raw measurement.
-int WebRtcAec_GetSkew(void *resampInst, int rawSkew, float *skewEst);
-
-// Resamples input using linear interpolation.
-// Returns size of resampled array.
-int WebRtcAec_ResampleLinear(void *resampInst,
-                             const short *inspeech,
-                             int size,
-                             float skew,
-                             short *outspeech);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_RESAMPLER_H_
diff --git a/src/modules/audio_processing/aecm/main/interface/echo_control_mobile.h b/src/modules/audio_processing/aecm/main/interface/echo_control_mobile.h
deleted file mode 100644
index 26b1172..0000000
--- a/src/modules/audio_processing/aecm/main/interface/echo_control_mobile.h
+++ /dev/null
@@ -1,206 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_INTERFACE_ECHO_CONTROL_MOBILE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_INTERFACE_ECHO_CONTROL_MOBILE_H_
-
-#include "typedefs.h"
-
-enum {
-    AecmFalse = 0,
-    AecmTrue
-};
-
-// Errors
-#define AECM_UNSPECIFIED_ERROR           12000
-#define AECM_UNSUPPORTED_FUNCTION_ERROR  12001
-#define AECM_UNINITIALIZED_ERROR         12002
-#define AECM_NULL_POINTER_ERROR          12003
-#define AECM_BAD_PARAMETER_ERROR         12004
-
-// Warnings
-#define AECM_BAD_PARAMETER_WARNING       12100
-
-typedef struct {
-    WebRtc_Word16 cngMode;            // AECM_FALSE, AECM_TRUE (default)
-    WebRtc_Word16 echoMode;           // 0, 1, 2, 3 (default), 4
-} AecmConfig;
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * Allocates the memory needed by the AECM. The memory needs to be
- * initialized separately using the WebRtcAecm_Init() function.
- *
- * Inputs                           Description
- * -------------------------------------------------------------------
- * void **aecmInst                  Pointer to the AECM instance to be
- *                                  created and initialized
- *
- * Outputs                          Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return             0: OK
- *                                 -1: error
- */
-WebRtc_Word32 WebRtcAecm_Create(void **aecmInst);
-
-/*
- * This function releases the memory allocated by WebRtcAecm_Create()
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void *aecmInst               Pointer to the AECM instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_Free(void *aecmInst);
-
-/*
- * Initializes an AECM instance.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecmInst     Pointer to the AECM instance
- * WebRtc_Word32  sampFreq      Sampling frequency of data
- * WebRtc_Word32  scSampFreq    Soundcard sampling frequency
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return          0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_Init(void* aecmInst,
-                              WebRtc_Word32 sampFreq,
-                              WebRtc_Word32 scSampFreq);
-
-/*
- * Inserts an 80 or 160 sample block of data into the farend buffer.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecmInst     Pointer to the AECM instance
- * WebRtc_Word16  *farend       In buffer containing one frame of
- *                              farend signal
- * WebRtc_Word16  nrOfSamples   Number of samples in farend buffer
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_BufferFarend(void* aecmInst,
-                                      const WebRtc_Word16* farend,
-                                      WebRtc_Word16 nrOfSamples);
-
-/*
- * Runs the AECM on an 80 or 160 sample blocks of data.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void           *aecmInst      Pointer to the AECM instance
- * WebRtc_Word16  *nearendNoisy  In buffer containing one frame of
- *                               reference nearend+echo signal. If
- *                               noise reduction is active, provide
- *                               the noisy signal here.
- * WebRtc_Word16  *nearendClean  In buffer containing one frame of
- *                               nearend+echo signal. If noise
- *                               reduction is active, provide the
- *                               clean signal here. Otherwise pass a
- *                               NULL pointer.
- * WebRtc_Word16  nrOfSamples    Number of samples in nearend buffer
- * WebRtc_Word16  msInSndCardBuf Delay estimate for sound card and
- *                               system buffers
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word16  *out          Out buffer, one frame of processed nearend
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_Process(void* aecmInst,
-                                 const WebRtc_Word16* nearendNoisy,
-                                 const WebRtc_Word16* nearendClean,
-                                 WebRtc_Word16* out,
-                                 WebRtc_Word16 nrOfSamples,
-                                 WebRtc_Word16 msInSndCardBuf);
-
-/*
- * This function enables the user to set certain parameters on-the-fly
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void     *aecmInst           Pointer to the AECM instance
- * AecmConfig config            Config instance that contains all
- *                              properties to be set
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_set_config(void* aecmInst,
-                                    AecmConfig config);
-
-/*
- * This function enables the user to set certain parameters on-the-fly
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void *aecmInst               Pointer to the AECM instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * AecmConfig  *config          Pointer to the config instance that
- *                              all properties will be written to
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_get_config(void *aecmInst,
-                                    AecmConfig *config);
-
-/*
- * Gets the last error code.
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * void         *aecmInst       Pointer to the AECM instance
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word32  return        11000-11100: error code
- */
-WebRtc_Word32 WebRtcAecm_get_error_code(void *aecmInst);
-
-/*
- * Gets a version string
- *
- * Inputs                       Description
- * -------------------------------------------------------------------
- * char           *versionStr   Pointer to a string array
- * WebRtc_Word16  len           The maximum length of the string
- *
- * Outputs                      Description
- * -------------------------------------------------------------------
- * WebRtc_Word8   *versionStr   Pointer to a string array
- * WebRtc_Word32  return        0: OK
- *                             -1: error
- */
-WebRtc_Word32 WebRtcAecm_get_version(WebRtc_Word8 *versionStr,
-                                     WebRtc_Word16 len);
-
-#ifdef __cplusplus
-}
-#endif
-#endif /* WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_INTERFACE_ECHO_CONTROL_MOBILE_H_ */
diff --git a/src/modules/audio_processing/aecm/main/matlab/compsup.m b/src/modules/audio_processing/aecm/main/matlab/compsup.m
deleted file mode 100644
index 9575ec4..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/compsup.m
+++ /dev/null
@@ -1,447 +0,0 @@
-function [emicrophone,aaa]=compsup(microphone,TheFarEnd,avtime,samplingfreq);
-% microphone = microphone signal
-% aaa = nonlinearity input variable
-% TheFarEnd = far end signal
-% avtime = interval to compute suppression from (seconds)
-% samplingfreq = sampling frequency
-
-%if(nargin==6)
-%    fprintf(1,'suppress has received a delay sequence\n');
-%end
-
-
-Ap500=[  1.00, -4.95, 9.801, -9.70299, 4.80298005, -0.9509900499];
-Bp500=[  0.662743088639636, -2.5841655608125, 3.77668102146288, -2.45182477425154, 0.596566274575251, 0.0];
-
-
-Ap200=[ 1.00, -4.875, 9.50625, -9.26859375, 4.518439453125, -0.881095693359375];
-Bp200=[ 0.862545460994275, -3.2832804496114, 4.67892032308828, -2.95798023879133, 0.699796870041299, 0.0];
-
-maxDelay=0.4; %[s]
-histLen=1; %[s]
-
-
-% CONSTANTS THAT YOU CAN EXPERIMENT WITH
-A_GAIN=10.0;	 	% for the suppress case
-oversampling = 2;	% must be power of 2; minimum is 2; 4 works
-% fine for support=64, but for support=128,
-% 8 gives better results.
-support=64; %512	% fft support (frequency resolution; at low
-% settings you can hear more distortion
-% (e.g. pitch that is left-over from far-end))
-% 128 works well, 64 is ok)
-
-lowlevel = mean(abs(microphone))*0.0001;
-
-G_ol = 0;  % Use overlapping sets of estimates
-
-% ECHO SUPPRESSION SPECIFIC PARAMETERS
-suppress_overdrive=1.0;   % overdrive factor for suppression 1.4 is good
-gamma_echo=1.0;           % same as suppress_overdrive but at different place
-de_echo_bound=0.0;
-mLim=10;                  % rank of matrix G
-%limBW = 1;		  % use bandwidth-limited response for G
-if mLim > (support/2+1)
-    error('mLim in suppress.m too large\n');
-end
-
-
-dynrange=1.0000e-004;
-
-% other, constants
-hsupport = support/2;
-hsupport1 = hsupport+1;
-factor =  2 / oversampling;
-updatel = support/oversampling;
-win=sqrt(designwindow(0,support));
-estLen = round(avtime * samplingfreq/updatel)
-
-runningfmean =0.0;
-
-mLim = floor(hsupport1/2);
-V = sqrt(2/hsupport1)*cos(pi/hsupport1*(repmat((0:hsupport1-1) + 0.5, mLim, 1).* ...
-    repmat((0:mLim-1)' + 0.5, 1, hsupport1)));
-
-fprintf(1,'updatel is %5.3f s\n', updatel/samplingfreq);
-
-
-
-bandfirst=8; bandlast=25;
-dosmooth=0;  % to get rid of wavy bin counts (can be worse or better)
-
-% compute some constants
-blockLen = support/oversampling;
-maxDelayb = floor(samplingfreq*maxDelay/updatel); % in blocks
-histLenb = floor(samplingfreq*histLen/updatel); % in blocks
-
-x0=TheFarEnd;
-y0=microphone;
-
-
-%input
-tlength=min([length(microphone),length(TheFarEnd)]);
-updateno=floor(tlength/updatel);
-tlength=updatel*updateno;
-updateno = updateno - oversampling + 1;
-
-TheFarEnd =TheFarEnd(1:tlength);
-microphone =microphone(1:tlength);
-
-TheFarEnd =[zeros(hsupport,1);TheFarEnd(1:tlength)];
-microphone =[zeros(hsupport,1);microphone(1:tlength)];
-
-
-% signal length
-n = min([floor(length(x0)/support)*support,floor(length(y0)/support)*support]);
-nb = n/blockLen - oversampling + 1; % in blocks
-
-% initialize space
-win = sqrt([0 ; hanning(support-1)]);
-sxAll2 = zeros(hsupport1,nb);
-syAll2 = zeros(hsupport1,nb);
-
-z500=zeros(5,maxDelayb+1);
-z200=zeros(5,hsupport1);
-
-bxspectrum=uint32(zeros(nb,1));
-bxhist=uint32(zeros(maxDelayb+1,1));
-byspectrum=uint32(zeros(nb,1));
-bcount=zeros(1+maxDelayb,nb);
-fcount=zeros(1+maxDelayb,nb);
-fout=zeros(1+maxDelayb,nb);
-delay=zeros(nb,1);
-tdelay=zeros(nb,1);
-nlgains=zeros(nb,1);
-
-% create space (mainly for debugging)
-emicrophone=zeros(tlength,1);
-femicrophone=complex(zeros(hsupport1,updateno));
-thefilter=zeros(hsupport1,updateno);
-thelimiter=ones(hsupport1,updateno);
-fTheFarEnd=complex(zeros(hsupport1,updateno));
-afTheFarEnd=zeros(hsupport1,updateno);
-fmicrophone=complex(zeros(hsupport1,updateno));
-afmicrophone=zeros(hsupport1,updateno);
-
-G = zeros(hsupport1, hsupport1);
-zerovec = zeros(hsupport1,1);
-zeromat = zeros(hsupport1);
-
-% Reset sums
-mmxs_a = zerovec;
-mmys_a = zerovec;
-s2xs_a = zerovec;
-s2ys_a = zerovec;
-Rxxs_a = zeromat;
-Ryxs_a = zeromat;
-count_a = 1;
-
-mmxs_b = zerovec;
-mmys_b = zerovec;
-s2xs_b = zerovec;
-s2ys_b = zerovec;
-Rxxs_b = zeromat;
-Ryxs_b = zeromat;
-count_b = 1;
-
-nog=0;
-
-aaa=zeros(size(TheFarEnd));
-
-% loop over signal blocks
-fprintf(1,'.. Suppression; averaging G over %5.1f seconds; file length %5.1f seconds ..\n',avtime, length(microphone)/samplingfreq);
-fprintf(1,'.. SUPPRESSING ONLY AFTER %5.1f SECONDS! ..\n',avtime);
-fprintf(1,'.. 20 seconds is good ..\n');
-hh = waitbar_j(0,'Please wait...');
-
-
-for i=1:updateno
-
-    sb = (i-1)*updatel + 1;
-    se=sb+support-1;
-    
-    % analysis FFTs
-    temp=fft(win .* TheFarEnd(sb:se));
-    fTheFarEnd(:,i)=temp(1:hsupport1);
-    xf=fTheFarEnd(:,i);
-    afTheFarEnd(:,i)= abs(fTheFarEnd(:,i));
-    
-    temp=win .* microphone(sb:se);
-    
-    temp=fft(win .* microphone(sb:se));
-    fmicrophone(:,i)=temp(1:hsupport1);
-    yf=fmicrophone(:,i);
-    afmicrophone(:,i)= abs(fmicrophone(:,i));
-
-    
-    ener_orig = afmicrophone(:,i)'*afmicrophone(:,i);
-    if( ener_orig == 0)
-        afmicrophone(:,i)=lowlevel*ones(size(afmicrophone(:,i)));
-    end
-    
-    
-    	% use log domain (showed improved performance)
-xxf= sqrt(real(xf.*conj(xf))+1e-20);
-yyf= sqrt(real(yf.*conj(yf))+1e-20);
-        sxAll2(:,i) = 20*log10(xxf);
-	syAll2(:,i) = 20*log10(yyf);
-
-       mD=min(i-1,maxDelayb);
-      xthreshold = sum(sxAll2(:,i-mD:i),2)/(maxDelayb+1);
-
-      [yout, z200] = filter(Bp200,Ap200,syAll2(:,i),z200,2);
-      yout=yout/(maxDelayb+1);
-      ythreshold = mean(syAll2(:,i-mD:i),2);
-      
-
-  bxspectrum(i)=getBspectrum(sxAll2(:,i),xthreshold,bandfirst,bandlast);
-  byspectrum(i)=getBspectrum(syAll2(:,i),yout,bandfirst,bandlast);
-
-  bxhist(end-mD:end)=bxspectrum(i-mD:i);
-  
-  bcount(:,i)=hisser2( ...
-     byspectrum(i),flipud(bxhist),bandfirst,bandlast);
- 
- 
-  [fout(:,i), z500] = filter(Bp500,Ap500,bcount(:,i),z500,2);
-  fcount(:,i)=sum(bcount(:,max(1,i-histLenb+1):i),2); % using the history range
- fout(:,i)=round(fout(:,i)); 
-  [value,delay(i)]=min(fout(:,i),[],1);
-  tdelay(i)=(delay(i)-1)*support/(samplingfreq*oversampling);
-
-    % compensate
-
-    idel =  max(i - delay(i) + 1,1);
-    
-  
-    % echo suppression
-    
-    noisyspec = afmicrophone(:,i);
-    
-    % Estimate G using covariance matrices
-    
-    % Cumulative estimates    
-    xx = afTheFarEnd(:,idel);
-    yy = afmicrophone(:,i);
-    
-    % Means
-    mmxs_a = mmxs_a + xx;
-    mmys_a = mmys_a + yy;
-    if (G_ol)
-        mmxs_b = mmxs_b + xx;  
-        mmys_b = mmys_b + yy;
-        mmy = mean([mmys_a/count_a mmys_b/count_b],2);
-        mmx = mean([mmxs_a/count_a mmxs_b/count_b],2);
-    else
-        mmx = mmxs_a/count_a;   
-        mmy = mmys_a/count_a;   
-    end
-    count_a = count_a + 1;
-    count_b = count_b + 1;
-    
-    % Mean removal
-    xxm = xx - mmx;
-    yym = yy - mmy;
-    
-    % Variances
-    s2xs_a = s2xs_a +  xxm .* xxm;
-    s2ys_a = s2ys_a +  yym .* yym;
-    s2xs_b = s2xs_b +  xxm .* xxm;
-    s2ys_b = s2ys_b +  yym .* yym;
-    
-    % Correlation matrices  
-    Rxxs_a = Rxxs_a + xxm * xxm';
-    Ryxs_a = Ryxs_a + yym * xxm';
-    Rxxs_b = Rxxs_b + xxm * xxm';
-    Ryxs_b = Ryxs_b + yym * xxm';
-    
-    
-    % Gain matrix A
-    
-    if mod(i, estLen) == 0
-        
-        
-        % Cumulative based estimates
-        Rxxf = Rxxs_a / (estLen - 1);
-        Ryxf = Ryxs_a / (estLen - 1);
-        
-        % Variance normalization
-        s2x2 = s2xs_a / (estLen - 1);
-        s2x2 = sqrt(s2x2);
-       % Sx = diag(max(s2x2,dynrange*max(s2x2)));
-        Sx = diag(s2x2);
-        if (sum(s2x2) > 0)
-          iSx = inv(Sx);
-         else
-                 iSx= Sx + 0.01;
-         end
-             
-        s2y2 = s2ys_a / (estLen - 1);
-        s2y2 = sqrt(s2y2);
-       % Sy = diag(max(s2y2,dynrange*max(s2y2)));
-        Sy = diag(s2y2);
-        iSy = inv(Sy);        
-        rx = iSx * Rxxf * iSx;
-        ryx = iSy * Ryxf * iSx;
-        
-     
-        
-        dbd= 7; % Us less than the full matrix
-        
-        % k x m
-        % Bandlimited structure on G
-        LSEon = 0; % Default is using MMSE
-        if (LSEon)
-            ryx = ryx*rx;
-            rx = rx*rx;
-        end
-        p = dbd-1;
-        gaj = min(min(hsupport1,2*p+1),min([p+(1:hsupport1); hsupport1+p+1-(1:hsupport1)]));
-        cgaj = [0 cumsum(gaj)];
-        
-        G3 = zeros(hsupport1);
-        for kk=1:hsupport1
-            ki = max(0,kk-p-1);
-            if (sum(sum(rx(ki+1:ki+gaj(kk),ki+1:ki+gaj(kk))))>0)
-               G3(kk,ki+1:ki+gaj(kk)) = ryx(kk,ki+1:ki+gaj(kk))/rx(ki+1:ki+gaj(kk),ki+1:ki+gaj(kk));
-           else
-               G3(kk,ki+1:ki+gaj(kk)) = ryx(kk,ki+1:ki+gaj(kk));
-           end
-        end
-        % End Bandlimited structure
-        
-        G = G3;
-        G(abs(G)<0.01)=0;
-        G = suppress_overdrive * Sy * G * iSx;
-        
-        if 1
-            figure(32); mi=2;
-            surf(max(min(G,mi),-mi)); view(2)
-            title('Unscaled Masked Limited-bandwidth G');
-        end
-        pause(0.05);
-        
-        % Reset sums
-        mmxs_a = zerovec;
-        mmys_a = zerovec;
-        s2xs_a = zerovec;
-        s2ys_a = zerovec;
-        Rxxs_a = zeromat;
-        Ryxs_a = zeromat;
-        count_a = 1;
-        
-    end
-    
-    if (G_ol)    
-        % Gain matrix B
-        
-        if ((mod((i-estLen/2), estLen) == 0) & i>estLen)
-            
-            
-            % Cumulative based estimates
-            Rxxf = Rxxs_b / (estLen - 1);
-            Ryxf = Ryxs_b / (estLen - 1);
-            
-            % Variance normalization
-            s2x2 = s2xs_b / (estLen - 1);
-            s2x2 = sqrt(s2x2);
-            Sx = diag(max(s2x2,dynrange*max(s2x2)));
-            iSx = inv(Sx);
-            s2y2 = s2ys_b / (estLen - 1);
-            s2y2 = sqrt(s2y2);
-            Sy = diag(max(s2y2,dynrange*max(s2y2)));
-            iSy = inv(Sy);        
-            rx = iSx * Rxxf * iSx;
-            ryx = iSy * Ryxf * iSx;
-            
-            
-            % Bandlimited structure on G
-            LSEon = 0; % Default is using MMSE
-            if (LSEon)
-                ryx = ryx*rx;
-                rx = rx*rx;
-            end
-            p = dbd-1;
-            gaj = min(min(hsupport1,2*p+1),min([p+(1:hsupport1); hsupport1+p+1-(1:hsupport1)]));
-            cgaj = [0 cumsum(gaj)];
-            
-            G3 = zeros(hsupport1);
-            for kk=1:hsupport1
-                ki = max(0,kk-p-1);
-                G3(kk,ki+1:ki+gaj(kk)) = ryx(kk,ki+1:ki+gaj(kk))/rx(ki+1:ki+gaj(kk),ki+1:ki+gaj(kk));
-            end
-            % End Bandlimited structure
-            
-            G = G3;
-            G(abs(G)<0.01)=0;
-            G = suppress_overdrive * Sy * G * iSx;
-            
-            if 1
-                figure(32); mi=2;
-                surf(max(min(G,mi),-mi)); view(2)
-                title('Unscaled Masked Limited-bandwidth G');
-            end
-            pause(0.05);
-            
-            
-            % Reset sums
-            mmxs_b = zerovec;
-            mmys_b = zerovec;
-            s2xs_b = zerovec;
-            s2ys_b = zerovec;
-            Rxxs_b = zeromat;
-            Ryxs_b = zeromat;
-            count_b = 1;
-            
-        end
-        
-    end
-    
-    FECestimate2 = G*afTheFarEnd(:,idel);
-    
-    % compute Wiener filter and suppressor function
-    thefilter(:,i) = (noisyspec - gamma_echo*FECestimate2) ./ noisyspec;
-    ix0 = find(thefilter(:,i)<de_echo_bound);   % bounding trick 1
-    thefilter(ix0,i) = de_echo_bound;     % bounding trick 2
-    ix0 = find(thefilter(:,i)>1);   % bounding in reasonable range
-    thefilter(ix0,i) = 1;
-    
-    % NONLINEARITY
-    nl_alpha=0.8;    % memory; seems not very critical
-    nlSeverity=0.3;  % nonlinearity severity: 0 does nothing; 1 suppresses all
-    thefmean=mean(thefilter(8:16,i));
-    if (thefmean<1)
-        disp('');
-    end
-    runningfmean = nl_alpha*runningfmean + (1-nl_alpha)*thefmean;
-    aaa(sb+20+1:sb+20+updatel)=10000*runningfmean* ones(updatel,1); % debug
-    slope0=1.0/(1.0-nlSeverity); %
-    thegain = max(0.0,min(1.0,slope0*(runningfmean-nlSeverity)));
-    % END NONLINEARITY
-    thefilter(:,i) = thegain*thefilter(:,i);
-    
-    
-    % Wiener filtering
-    femicrophone(:,i) = fmicrophone(:,i) .* thefilter(:,i);
-    thelimiter(:,i) = (noisyspec - A_GAIN*FECestimate2) ./ noisyspec;
-    index = find(thelimiter(:,i)>1.0);
-    thelimiter(index,i) = 1.0;
-    index = find(thelimiter(:,i)<0.0);
-    thelimiter(index,i) = 0.0;
-    
-    if (rem(i,floor(updateno/20))==0)
-        fprintf(1,'.');
-    end
-    if mod(i,50)==0
-        waitbar_j(i/updateno,hh); 
-    end
-    
-    
-    % reconstruction; first make spectrum odd
-    temp=[femicrophone(:,i);flipud(conj(femicrophone(2:hsupport,i)))];
-    emicrophone(sb:se) = emicrophone(sb:se) + factor * win .* real(ifft(temp));
-
-end
-fprintf(1,'\n');
-
-close(hh);
\ No newline at end of file
diff --git a/src/modules/audio_processing/aecm/main/matlab/getBspectrum.m b/src/modules/audio_processing/aecm/main/matlab/getBspectrum.m
deleted file mode 100644
index a4a533d..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/getBspectrum.m
+++ /dev/null
@@ -1,22 +0,0 @@
-function bspectrum=getBspectrum(ps,threshold,bandfirst,bandlast)
-% function bspectrum=getBspectrum(ps,threshold,bandfirst,bandlast)
-% compute binary spectrum using threshold spectrum as pivot
-% bspectrum = binary spectrum (binary)
-% ps=current power spectrum (float)
-% threshold=threshold spectrum (float)
-% bandfirst = first band considered
-% bandlast = last band considered
-  
-% initialization stuff
-  if( length(ps)<bandlast | bandlast>32 | length(ps)~=length(threshold)) 
-  error('BinDelayEst:spectrum:invalid','Dimensionality error');
-end
-
-% get current binary spectrum
-diff = ps - threshold;
-bspectrum=uint32(0);
-for(i=bandfirst:bandlast)
-  if( diff(i)>0 ) 
-    bspectrum = bitset(bspectrum,i);
-  end
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/hisser2.m b/src/modules/audio_processing/aecm/main/matlab/hisser2.m
deleted file mode 100644
index 5a414f9..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/hisser2.m
+++ /dev/null
@@ -1,21 +0,0 @@
-function  bcount=hisser2(bs,bsr,bandfirst,bandlast)
-% function  bcount=hisser(bspectrum,bandfirst,bandlast)
-% histogram for the binary spectra
-% bcount= array of bit counts 
-% bs=binary spectrum (one int32 number each)  
-% bsr=reference binary spectra (one int32 number each)
-% blockSize = histogram over blocksize blocks
-% bandfirst = first band considered
-% bandlast = last band considered
-
-% weight all delays equally
-maxDelay = length(bsr);
-
-% compute counts (two methods; the first works better and is operational)
-bcount=zeros(maxDelay,1);
-for(i=1:maxDelay)
- % the delay should have low count for low-near&high-far and high-near&low-far
- bcount(i)= sum(bitget(bitxor(bs,bsr(i)),bandfirst:bandlast));  
- % the delay should have low count for low-near&high-far (works less well)
-% bcount(i)= sum(bitget(bitand(bsr(i),bitxor(bs,bsr(i))),bandfirst:bandlast));
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/main2.m b/src/modules/audio_processing/aecm/main/matlab/main2.m
deleted file mode 100644
index 7e24c69..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/main2.m
+++ /dev/null
@@ -1,19 +0,0 @@
-
-fid=fopen('aecfar.pcm'); far=fread(fid,'short'); fclose(fid);
-fid=fopen('aecnear.pcm'); mic=fread(fid,'short'); fclose(fid);
-
-%fid=fopen('QA1far.pcm'); far=fread(fid,'short'); fclose(fid);
-%fid=fopen('QA1near.pcm'); mic=fread(fid,'short'); fclose(fid);
-
-start=0 * 8000+1;
-stop= 30 * 8000;
-microphone=mic(start:stop);
-TheFarEnd=far(start:stop);
-avtime=1;
-
-% 16000 to make it compatible with the C-version
-[emicrophone,tdel]=compsup(microphone,TheFarEnd,avtime,16000); 
-
-spclab(8000,TheFarEnd,microphone,emicrophone);
-
-
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/AECMobile.m b/src/modules/audio_processing/aecm/main/matlab/matlab/AECMobile.m
deleted file mode 100644
index 2d3e686..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/AECMobile.m
+++ /dev/null
@@ -1,269 +0,0 @@
-function [femicrophone, aecmStructNew, enerNear, enerFar] = AECMobile(fmicrophone, afTheFarEnd, setupStruct, aecmStruct)
-global NEARENDFFT;
-global F;
-
-aecmStructNew = aecmStruct;
-
-% Magnitude spectrum of near end signal
-afmicrophone = abs(fmicrophone);
-%afmicrophone = NEARENDFFT(setupStruct.currentBlock,:)'/2^F(setupStruct.currentBlock,end);
-% Near end energy level
-ener_orig = afmicrophone'*afmicrophone;
-if( ener_orig == 0)
-    lowlevel = 0.01;
-    afmicrophone = lowlevel*ones(size(afmicrophone));
-end
-%adiff = max(abs(afmicrophone - afTheFarEnd));
-%if (adiff > 0)
-%    disp([setupStruct.currentBlock adiff])
-%end
-
-% Store the near end energy
-%aecmStructNew.enerNear(setupStruct.currentBlock) = log(afmicrophone'*afmicrophone);
-aecmStructNew.enerNear(setupStruct.currentBlock) = log(sum(afmicrophone));
-% Store the far end energy
-%aecmStructNew.enerFar(setupStruct.currentBlock) = log(afTheFarEnd'*afTheFarEnd);
-aecmStructNew.enerFar(setupStruct.currentBlock) = log(sum(afTheFarEnd));
-
-% Update subbands (We currently use all frequency bins, hence .useSubBand is turned off)
-if aecmStructNew.useSubBand
-    internalIndex = 1;
-    for kk=1:setupStruct.subBandLength+1
-        ySubBand(kk) = mean(afmicrophone(internalIndex:internalIndex+setupStruct.numInBand(kk)-1).^aecmStructNew.bandFactor);
-        xSubBand(kk) = mean(afTheFarEnd(internalIndex:internalIndex+setupStruct.numInBand(kk)-1).^aecmStructNew.bandFactor);
-        internalIndex = internalIndex + setupStruct.numInBand(kk);
-    end
-else
-    ySubBand = afmicrophone.^aecmStructNew.bandFactor;
-    xSubBand = afTheFarEnd.^aecmStructNew.bandFactor;
-end
-
-% Estimated echo energy
-if (aecmStructNew.bandFactor == 1)
-    %aecmStructNew.enerEcho(setupStruct.currentBlock) = log((aecmStructNew.H.*xSubBand)'*(aecmStructNew.H.*xSubBand));
-    %aecmStructNew.enerEchoStored(setupStruct.currentBlock) = log((aecmStructNew.HStored.*xSubBand)'*(aecmStructNew.HStored.*xSubBand));
-    aecmStructNew.enerEcho(setupStruct.currentBlock) = log(sum(aecmStructNew.H.*xSubBand));
-    aecmStructNew.enerEchoStored(setupStruct.currentBlock) = log(sum(aecmStructNew.HStored.*xSubBand));
-elseif (aecmStructNew.bandFactor == 2)
-    aecmStructNew.enerEcho(setupStruct.currentBlock) = log(aecmStructNew.H'*xSubBand);
-    aecmStructNew.enerEchoStored(setupStruct.currentBlock) = log(aecmStructNew.HStored'*xSubBand);
-end
-
-% Last 100 blocks of data, used for plotting
-n100 = max(1,setupStruct.currentBlock-99):setupStruct.currentBlock;
-enerError = aecmStructNew.enerNear(n100)-aecmStructNew.enerEcho(n100);
-enerErrorStored = aecmStructNew.enerNear(n100)-aecmStructNew.enerEchoStored(n100);
-
-% Store the far end sub band. This is needed if we use LSE instead of NLMS
-aecmStructNew.X = [xSubBand aecmStructNew.X(:,1:end-1)];
-
-% Update energy levels, which control the VAD
-if ((aecmStructNew.enerFar(setupStruct.currentBlock) < aecmStructNew.energyMin) & (aecmStructNew.enerFar(setupStruct.currentBlock) >= aecmStruct.FAR_ENERGY_MIN))
-    aecmStructNew.energyMin = aecmStructNew.enerFar(setupStruct.currentBlock);
-    %aecmStructNew.energyMin = max(aecmStructNew.energyMin,12);
-    aecmStructNew.energyMin = max(aecmStructNew.energyMin,aecmStruct.FAR_ENERGY_MIN);
-    aecmStructNew.energyLevel = (aecmStructNew.energyMax-aecmStructNew.energyMin)*aecmStructNew.energyThres+aecmStructNew.energyMin;
-    aecmStructNew.energyLevelMSE = (aecmStructNew.energyMax-aecmStructNew.energyMin)*aecmStructNew.energyThresMSE+aecmStructNew.energyMin;
-end
-if (aecmStructNew.enerFar(setupStruct.currentBlock) > aecmStructNew.energyMax)
-    aecmStructNew.energyMax = aecmStructNew.enerFar(setupStruct.currentBlock);
-    aecmStructNew.energyLevel = (aecmStructNew.energyMax-aecmStructNew.energyMin)*aecmStructNew.energyThres+aecmStructNew.energyMin;
-    aecmStructNew.energyLevelMSE = (aecmStructNew.energyMax-aecmStructNew.energyMin)*aecmStructNew.energyThresMSE+aecmStructNew.energyMin;
-end
-
-% Calculate current energy error in near end (estimated echo vs. near end)
-dE = aecmStructNew.enerNear(setupStruct.currentBlock)-aecmStructNew.enerEcho(setupStruct.currentBlock);
-
-%%%%%%%%
-% Calculate step size used in LMS algorithm, based on current far end energy and near end energy error (dE)
-%%%%%%%%
-if setupStruct.stepSize_flag
-    [mu, aecmStructNew] = calcStepSize(aecmStructNew.enerFar(setupStruct.currentBlock), dE, aecmStructNew, setupStruct.currentBlock, 1);
-else
-    mu = 0.25;
-end
-aecmStructNew.muLog(setupStruct.currentBlock) = mu; % Store the step size
-
-% Estimate Echo Spectral Shape
-[U, aecmStructNew.H] = fallerEstimator(ySubBand,aecmStructNew.X,aecmStructNew.H,mu);
-
-%%%%%
-% Determine if we should store or restore the channel
-%%%%%
-if ((setupStruct.currentBlock <= aecmStructNew.convLength) | (~setupStruct.channelUpdate_flag))
-    aecmStructNew.HStored = aecmStructNew.H; % Store what you have after startup
-elseif ((setupStruct.currentBlock > aecmStructNew.convLength) & (setupStruct.channelUpdate_flag))
-    if ((aecmStructNew.enerFar(setupStruct.currentBlock) < aecmStructNew.energyLevelMSE) & (aecmStructNew.enerFar(setupStruct.currentBlock-1) >= aecmStructNew.energyLevelMSE))
-        xxx = aecmStructNew.countMseH;
-        if (xxx > 20)
-            mseStored = mean(abs(aecmStructNew.enerEchoStored(setupStruct.currentBlock-xxx:setupStruct.currentBlock-1)-aecmStructNew.enerNear(setupStruct.currentBlock-xxx:setupStruct.currentBlock-1)));
-            mseLatest = mean(abs(aecmStructNew.enerEcho(setupStruct.currentBlock-xxx:setupStruct.currentBlock-1)-aecmStructNew.enerNear(setupStruct.currentBlock-xxx:setupStruct.currentBlock-1)));
-            %fprintf('Stored: %4f Latest: %4f\n', mseStored, mseLatest) % Uncomment if you want to display the MSE values
-            if ((mseStored < 0.8*mseLatest) & (aecmStructNew.mseHStoredOld < 0.8*aecmStructNew.mseHLatestOld))
-                aecmStructNew.H = aecmStructNew.HStored;
-                fprintf('Restored H at block %d\n',setupStruct.currentBlock)
-            elseif (((0.8*mseStored > mseLatest) & (mseLatest < aecmStructNew.mseHThreshold) & (aecmStructNew.mseHLatestOld < aecmStructNew.mseHThreshold)) | (mseStored == Inf))
-                aecmStructNew.HStored = aecmStructNew.H;
-                fprintf('Stored new H at block %d\n',setupStruct.currentBlock)
-            end
-            aecmStructNew.mseHStoredOld = mseStored;
-            aecmStructNew.mseHLatestOld = mseLatest;
-        end
-    elseif ((aecmStructNew.enerFar(setupStruct.currentBlock) >= aecmStructNew.energyLevelMSE) & (aecmStructNew.enerFar(setupStruct.currentBlock-1) < aecmStructNew.energyLevelMSE))
-        aecmStructNew.countMseH = 1;
-    elseif (aecmStructNew.enerFar(setupStruct.currentBlock) >= aecmStructNew.energyLevelMSE)
-        aecmStructNew.countMseH = aecmStructNew.countMseH + 1;
-    end
-end
-
-%%%%%
-% Check delay (calculate the delay offset (if we can))
-% The algorithm is not tuned and should be used with care. It runs separately from Bastiaan's algorithm.
-%%%%%
-yyy = 31; % Correlation buffer length (currently unfortunately hard coded)
-dxxx = 25; % Maximum offset (currently unfortunately hard coded)
-if (setupStruct.currentBlock > aecmStructNew.convLength)
-    if (aecmStructNew.enerFar(setupStruct.currentBlock-(yyy+2*dxxx-1):setupStruct.currentBlock) > aecmStructNew.energyLevelMSE)
-        for xxx = -dxxx:dxxx
-            aecmStructNew.delayLatestS(xxx+dxxx+1) = sum(sign(aecmStructNew.enerEcho(setupStruct.currentBlock-(yyy+dxxx-xxx)+1:setupStruct.currentBlock+xxx-dxxx)-mean(aecmStructNew.enerEcho(setupStruct.currentBlock-(yyy++dxxx-xxx)+1:setupStruct.currentBlock+xxx-dxxx))).*sign(aecmStructNew.enerNear(setupStruct.currentBlock-yyy-dxxx+1:setupStruct.currentBlock-dxxx)-mean(aecmStructNew.enerNear(setupStruct.currentBlock-yyy-dxxx+1:setupStruct.currentBlock-dxxx))));
-        end
-        aecmStructNew.newDelayCurve = 1;
-    end
-end
-if ((setupStruct.currentBlock > 2*aecmStructNew.convLength) & ~rem(setupStruct.currentBlock,yyy*2) & aecmStructNew.newDelayCurve)
-    [maxV,maxP] = max(aecmStructNew.delayLatestS);
-    if ((maxP > 2) & (maxP < 2*dxxx))
-        maxVLeft = aecmStructNew.delayLatestS(max(1,maxP-4));
-        maxVRight = aecmStructNew.delayLatestS(min(2*dxxx+1,maxP+4));
-        %fprintf('Max %d, Left %d, Right %d\n',maxV,maxVLeft,maxVRight) % Uncomment if you want to see max value
-        if ((maxV > 24) & (maxVLeft < maxV - 10)  & (maxVRight < maxV - 10))
-            aecmStructNew.feedbackDelay = maxP-dxxx-1;
-            aecmStructNew.newDelayCurve = 0;
-            aecmStructNew.feedbackDelayUpdate = 1;
-            fprintf('Feedback Update at block %d\n',setupStruct.currentBlock)
-        end
-    end
-end
-% End of "Check delay"
-%%%%%%%%
-
-%%%%%
-% Calculate suppression gain, based on far end energy and near end energy error (dE)
-if (setupStruct.supGain_flag)
-    [gamma_echo, aecmStructNew.cntIn, aecmStructNew.cntOut] = calcFilterGain(aecmStructNew.enerFar(setupStruct.currentBlock), dE, aecmStructNew, setupStruct.currentBlock, aecmStructNew.convLength, aecmStructNew.cntIn, aecmStructNew.cntOut);
-else
-    gamma_echo = 1;
-end
-aecmStructNew.gammaLog(setupStruct.currentBlock) = gamma_echo; % Store the gain
-gamma_use = gamma_echo;
-
-% Use the stored channel
-U = aecmStructNew.HStored.*xSubBand;
-
-% compute Wiener filter and suppressor function
-Iy = find(ySubBand);
-subBandFilter = zeros(size(ySubBand));
-if (aecmStructNew.bandFactor == 2)
-    subBandFilter(Iy) = (1 - gamma_use*sqrt(U(Iy)./ySubBand(Iy))); % For Faller
-else
-    subBandFilter(Iy) = (1 - gamma_use*(U(Iy)./ySubBand(Iy))); % For COV
-end
-ix0 = find(subBandFilter < 0);   % bounding trick 1
-subBandFilter(ix0) = 0;
-ix0 = find(subBandFilter > 1);   % bounding trick 1
-subBandFilter(ix0) = 1;
-
-% Interpolate back to normal frequency bins if we use sub bands
-if aecmStructNew.useSubBand
-    thefilter = interp1(setupStruct.centerFreq,subBandFilter,linspace(0,setupStruct.samplingfreq/2,setupStruct.hsupport1)','nearest');
-    testfilter = interp1(setupStruct.centerFreq,subBandFilter,linspace(0,setupStruct.samplingfreq/2,1000),'nearest');
-    thefilter(end) = subBandFilter(end);
-    
-    internalIndex = 1;
-    for kk=1:setupStruct.subBandLength+1
-        internalIndex:internalIndex+setupStruct.numInBand(kk)-1;
-        thefilter(internalIndex:internalIndex+setupStruct.numInBand(kk)-1) = subBandFilter(kk);
-        internalIndex = internalIndex + setupStruct.numInBand(kk);
-    end
-else
-    thefilter = subBandFilter;
-    testfilter = subBandFilter;
-end
-
-% Bound the filter
-ix0 = find(thefilter < setupStruct.de_echo_bound);   % bounding trick 1
-thefilter(ix0) = setupStruct.de_echo_bound;     % bounding trick 2
-ix0 = find(thefilter > 1);   % bounding in reasonable range
-thefilter(ix0) = 1;
-
-%%%%
-% NLP
-%%%%
-thefmean = mean(thefilter(8:16));
-if (thefmean < 1)
-    disp('');
-end
-aecmStructNew.runningfmean = setupStruct.nl_alpha*aecmStructNew.runningfmean + (1-setupStruct.nl_alpha)*thefmean;
-slope0 = 1.0/(1.0 - setupStruct.nlSeverity); %
-thegain = max(0.0, min(1.0, slope0*(aecmStructNew.runningfmean - setupStruct.nlSeverity)));
-if ~setupStruct.nlp_flag
-    thegain = 1;
-end
-% END NONLINEARITY
-thefilter = thegain*thefilter;
-
-%%%%
-% The suppression
-%%%%
-femicrophone = fmicrophone .* thefilter;
-% Store the output energy (used for plotting)
-%aecmStructNew.enerOut(setupStruct.currentBlock) = log(abs(femicrophone)'*abs(femicrophone));
-aecmStructNew.enerOut(setupStruct.currentBlock) = log(sum(abs(femicrophone)));
-
-if aecmStructNew.plotIt
-    figure(13)
-    subplot(311)
-    %plot(n100,enerFar(n100),'b-',n100,enerNear(n100),'k--',n100,enerEcho(n100),'r-',[n100(1) n100(end)],[1 1]*vadThNew,'b:',[n100(1) n100(end)],[1 1]*((energyMax-energyMin)/4+energyMin),'r-.',[n100(1) n100(end)],[1 1]*vadNearThNew,'g:',[n100(1) n100(end)],[1 1]*energyMax,'r-.',[n100(1) n100(end)],[1 1]*energyMin,'r-.','LineWidth',2)
-    plot(n100,aecmStructNew.enerFar(n100),'b-',n100,aecmStructNew.enerNear(n100),'k--',n100,aecmStructNew.enerOut(n100),'r-.',n100,aecmStructNew.enerEcho(n100),'r-',n100,aecmStructNew.enerEchoStored(n100),'c-',[n100(1) n100(end)],[1 1]*((aecmStructNew.energyMax-aecmStructNew.energyMin)/4+aecmStructNew.energyMin),'g-.',[n100(1) n100(end)],[1 1]*aecmStructNew.energyMax,'g-.',[n100(1) n100(end)],[1 1]*aecmStructNew.energyMin,'g-.','LineWidth',2)
-    %title(['Frame ',int2str(i),' av ',int2str(setupStruct.updateno),' State = ',int2str(speechState),' \mu = ',num2str(mu)])
-    title(['\gamma = ',num2str(gamma_echo),' \mu = ',num2str(mu)])
-    subplot(312)
-    %plot(n100,enerError,'b-',[n100(1) n100(end)],[1 1]*vadNearTh,'r:',[n100(1) n100(end)],[-1.5 -1.5]*vadNearTh,'r:','LineWidth',2)
-    %plot(n100,enerError,'b-',[n100(1) n100(end)],[1 1],'r:',[n100(1) n100(end)],[-2 -2],'r:','LineWidth',2)
-    plot(n100,enerError,'b-',n100,enerErrorStored,'c-',[n100(1) n100(end)],[1 1]*aecmStructNew.varMean,'k--',[n100(1) n100(end)],[1 1],'r:',[n100(1) n100(end)],[-2 -2],'r:','LineWidth',2)
-    % Plot mu
-    %plot(n100,log2(aecmStructNew.muLog(n100)),'b-','LineWidth',2)
-    %plot(n100,log2(aecmStructNew.HGain(n100)),'b-',[n100(1) n100(end)],[1 1]*log2(sum(aecmStructNew.HStored)),'r:','LineWidth',2)
-    title(['Block ',int2str(setupStruct.currentBlock),' av ',int2str(setupStruct.updateno)])
-    subplot(313)
-    %plot(n100,enerVar(n100),'b-',[n100(1) n100(end)],[1 1],'r:',[n100(1) n100(end)],[-2 -2],'r:','LineWidth',2)
-    %plot(n100,enerVar(n100),'b-','LineWidth',2)
-    % Plot correlation curve
-
-    %plot(-25:25,aecmStructNew.delayStored/max(aecmStructNew.delayStored),'c-',-25:25,aecmStructNew.delayLatest/max(aecmStructNew.delayLatest),'r-',-25:25,(max(aecmStructNew.delayStoredS)-aecmStructNew.delayStoredS)/(max(aecmStructNew.delayStoredS)-min(aecmStructNew.delayStoredS)),'c:',-25:25,(max(aecmStructNew.delayLatestS)-aecmStructNew.delayLatestS)/(max(aecmStructNew.delayLatestS)-min(aecmStructNew.delayLatestS)),'r:','LineWidth',2)
-    %plot(-25:25,aecmStructNew.delayStored,'c-',-25:25,aecmStructNew.delayLatest,'r-',-25:25,(max(aecmStructNew.delayStoredS)-aecmStructNew.delayStoredS)/(max(aecmStructNew.delayStoredS)-min(aecmStructNew.delayStoredS)),'c:',-25:25,(max(aecmStructNew.delayLatestS)-aecmStructNew.delayLatestS)/(max(aecmStructNew.delayLatestS)-min(aecmStructNew.delayLatestS)),'r:','LineWidth',2)
-    %plot(-25:25,aecmStructNew.delayLatest,'r-',-25:25,(50-aecmStructNew.delayLatestS)/100,'r:','LineWidth',2)
-    plot(-25:25,aecmStructNew.delayLatestS,'r:','LineWidth',2)
-    %plot(-25:25,aecmStructNew.delayStored,'c-',-25:25,aecmStructNew.delayLatest,'r-','LineWidth',2)
-    plot(0:32,aecmStruct.HStored,'bo-','LineWidth',2)
-    %title(['\gamma | In = ',int2str(aecmStructNew.muStruct.countInInterval),' | Out High = ',int2str(aecmStructNew.muStruct.countOutHighInterval),' | Out Low = ',int2str(aecmStructNew.muStruct.countOutLowInterval)])
-    pause(1)
-    %if ((setupStruct.currentBlock == 860) | (setupStruct.currentBlock == 420) | (setupStruct.currentBlock == 960))
-    if 0%(setupStruct.currentBlock == 960)
-        figure(60)
-        plot(n100,aecmStructNew.enerNear(n100),'k--',n100,aecmStructNew.enerEcho(n100),'k:','LineWidth',2)
-        legend('Near End','Estimated Echo')
-        title('Signal Energy witH offset compensation')
-        figure(61)
-        subplot(211)
-        stem(sign(aecmStructNew.enerNear(n100)-mean(aecmStructNew.enerNear(n100))))
-        title('Near End Energy Pattern (around mean value)')
-        subplot(212)
-        stem(sign(aecmStructNew.enerEcho(n100)-mean(aecmStructNew.enerEcho(n100))))
-        title('Estimated Echo Energy Pattern (around mean value)')
-        pause
-    end
-    drawnow%,pause
-elseif ~rem(setupStruct.currentBlock,100)
-    fprintf('Block %d of %d\n',setupStruct.currentBlock,setupStruct.updateno)
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/align.m b/src/modules/audio_processing/aecm/main/matlab/matlab/align.m
deleted file mode 100644
index 9b9c0ba..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/align.m
+++ /dev/null
@@ -1,98 +0,0 @@
-function [delayStructNew] = align(xf, yf, delayStruct, i, trueDelay);
-
-%%%%%%%
-% Bastiaan's algorithm copied
-%%%%%%%
-Ap500 = [1.00, -4.95, 9.801, -9.70299, 4.80298005, -0.9509900499];
-Bp500 = [0.662743088639636, -2.5841655608125, 3.77668102146288, -2.45182477425154, 0.596566274575251, 0.0];
-Ap200 = [1.00, -4.875, 9.50625, -9.26859375, 4.518439453125, -0.881095693359375];
-Bp200 = [0.862545460994275, -3.2832804496114, 4.67892032308828, -2.95798023879133, 0.699796870041299, 0.0];
-
-oldMethod = 1; % Turn on or off the old method. The new one is Bastiaan's August 2008 updates
-THReSHoLD = 2.0; % ADJUSTABLE threshold factor; 4.0 seems good
-%%%%%%%%%%%%%%%%%%%
-% use log domain (showed improved performance)
-xxf = sqrt(real(xf.*conj(xf))+1e-20);
-yyf = sqrt(real(yf.*conj(yf))+1e-20);
-delayStruct.sxAll2(:,i) = 20*log10(xxf);
-delayStruct.syAll2(:,i) = 20*log10(yyf);
-
-mD = min(i-1,delayStruct.maxDelayb);
-if oldMethod
-    factor = 1.0;
-    histLenb = 250;
-    xthreshold = factor*median(delayStruct.sxAll2(:,i-mD:i),2);
-    ythreshold = factor*median(delayStruct.syAll2(:,i-mD:i),2);
-else
-    xthreshold = sum(delayStruct.sxAll2(:,i-mD:i),2)/(delayStruct.maxDelayb+1);
-    
-    [yout, delayStruct.z200] = filter(Bp200, Ap200, delayStruct.syAll2(:,i), delayStruct.z200, 2);
-    yout = yout/(delayStruct.maxDelayb+1);
-    ythreshold = mean(delayStruct.syAll2(:,i-mD:i),2);
-    ythreshold = yout;
-end
-
-delayStruct.bxspectrum(i) = getBspectrum(delayStruct.sxAll2(:,i), xthreshold, delayStruct.bandfirst, delayStruct.bandlast);
-delayStruct.byspectrum(i) = getBspectrum(delayStruct.syAll2(:,i), ythreshold, delayStruct.bandfirst, delayStruct.bandlast);
-
-delayStruct.bxhist(end-mD:end) = delayStruct.bxspectrum(i-mD:i);
-
-delayStruct.bcount(:,i) = hisser2(delayStruct.byspectrum(i), flipud(delayStruct.bxhist), delayStruct.bandfirst, delayStruct.bandlast);
-[delayStruct.fout(:,i), delayStruct.z500] = filter(Bp500, Ap500, delayStruct.bcount(:,i), delayStruct.z500, 2);
-if oldMethod
-    %delayStruct.new(:,i) = sum(delayStruct.bcount(:,max(1,i-histLenb+1):i),2); % using the history range
-    tmpVec = [delayStruct.fout(1,i)*ones(2,1); delayStruct.fout(:,i); delayStruct.fout(end,i)*ones(2,1)]; % using the history range
-    tmpVec = filter(ones(1,5), 1, tmpVec);
-    delayStruct.new(:,i) = tmpVec(5:end);
-    %delayStruct.new(:,i) = delayStruct.fout(:,i); % using the history range
-else
-    [delayStruct.fout(:,i), delayStruct.z500] = filter(Bp500, Ap500, delayStruct.bcount(:,i), delayStruct.z500, 2);
-    % NEW CODE
-    delayStruct.new(:,i) = filter([-1,-2,1,4,1,-2,-1], 1, delayStruct.fout(:,i)); %remv smth component
-    delayStruct.new(1:end-3,i) = delayStruct.new(1+3:end,i);
-    delayStruct.new(1:6,i) = 0.0;
-    delayStruct.new(end-6:end,i) = 0.0;  % ends are no good
-end
-[valuen, tempdelay] = min(delayStruct.new(:,i));  % find minimum
-if oldMethod
-    threshold = valuen + (max(delayStruct.new(:,i)) - valuen)/4;
-    thIndex = find(delayStruct.new(:,i) <= threshold);
-    if (i > 1)
-        delayDiff = abs(delayStruct.delay(i-1)-tempdelay+1);
-        if (delayStruct.oneGoodEstimate & (max(diff(thIndex)) > 1) & (delayDiff < 10))
-            % We consider this minimum to be significant, hence update the delay
-            delayStruct.delay(i) = tempdelay;
-        elseif (~delayStruct.oneGoodEstimate & (max(diff(thIndex)) > 1))
-            delayStruct.delay(i) = tempdelay;
-            if (i > histLenb)
-                delayStruct.oneGoodEstimate = 1;
-            end
-        else
-            delayStruct.delay(i) = delayStruct.delay(i-1);
-        end
-    else
-        delayStruct.delay(i) = tempdelay;
-    end
-else
-    threshold = THReSHoLD*std(delayStruct.new(:,i));   % set updata threshold 
-    if ((-valuen > threshold) | (i < delayStruct.smlength)) % see if you want to update delay
-        delayStruct.delay(i) = tempdelay;
-    else
-        delayStruct.delay(i) = delayStruct.delay(i-1);
-    end
-    % END NEW CODE
-end
-delayStructNew = delayStruct;
-
-% administrative and plotting stuff
-if( 0)
-    figure(10);
-    plot([1:length(delayStructNew.new(:,i))],delayStructNew.new(:,i),trueDelay*[1 1],[min(delayStructNew.new(:,i)),max(delayStructNew.new(:,i))],'r',[1 length(delayStructNew.new(:,i))],threshold*[1 1],'r:', 'LineWidth',2);
-    %plot([1:length(delayStructNew.bcount(:,i))],delayStructNew.bcount(:,i),trueDelay*[1 1],[min(delayStructNew.bcount(:,i)),max(delayStructNew.bcount(:,i))],'r','LineWidth',2);
-    %plot([thedelay,thedelay],[min(fcount(:,i)),max(fcount(:,i))],'r');
-    %title(sprintf('bin count and known delay at time %5.1f s\n',(i-1)*(support/(fs*oversampling))));
-    title(delayStructNew.oneGoodEstimate)
-    xlabel('delay in frames');
-    %hold off;
-    drawnow
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/calcFilterGain.m b/src/modules/audio_processing/aecm/main/matlab/matlab/calcFilterGain.m
deleted file mode 100644
index a09a7f2..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/calcFilterGain.m
+++ /dev/null
@@ -1,88 +0,0 @@
-function [gam, cntIn2, cntOut2] = calcFilterGain(energy, dE, aecmStruct, t, T, cntIn, cntOut)
-
-defaultLevel = 1.2;
-cntIn2 = cntIn;
-cntOut2 = cntOut;
-if (t < T)
-    gam = 1;
-else
-    dE1 = -5;
-    dE2 = 1;
-    gamMid = 0.2;
-    gam = max(0,min((energy - aecmStruct.energyMin)/(aecmStruct.energyLevel - aecmStruct.energyMin), 1-(1-gamMid)*(aecmStruct.energyMax-energy)/(aecmStruct.energyMax-aecmStruct.energyLevel)));
-    
-    dEOffset = -0.5;
-    dEWidth = 1.5;
-    %gam2 = max(1,2-((dE-dEOffset)/(dE2-dEOffset)).^2);
-    gam2 = 1+(abs(dE-dEOffset)<(dE2-dEOffset));
-    
-    gam = gam*gam2;
-    
-    
-    if (energy < aecmStruct.energyLevel)
-        gam = 0;
-    else
-        gam = defaultLevel;
-    end
-    dEVec = aecmStruct.enerNear(t-63:t)-aecmStruct.enerEcho(t-63:t);
-    %dEVec = aecmStruct.enerNear(t-20:t)-aecmStruct.enerEcho(t-20:t);
-    numCross = 0;
-    currentState = 0;
-    for ii=1:64
-        if (currentState == 0)
-            currentState = (dEVec(ii) > dE2) - (dEVec(ii) < -2);
-        elseif ((currentState == 1) & (dEVec(ii) < -2))
-            numCross = numCross + 1;
-            currentState = -1;
-        elseif ((currentState == -1) & (dEVec(ii) > dE2))
-            numCross = numCross + 1;
-            currentState = 1;
-        end
-    end
-    gam = max(0, gam - numCross/25);
-    gam = 1;
-    
-    ener_A = 1;
-    ener_B = 0.8;
-    ener_C = aecmStruct.energyLevel + (aecmStruct.energyMax-aecmStruct.energyLevel)/5;
-    dE_A = 4;%2;
-    dE_B = 3.6;%1.8;
-    dE_C = 0.9*dEWidth;
-    dE_D = 1;
-    timeFactorLength = 10;
-    ddE = abs(dE-dEOffset);
-    if (energy < aecmStruct.energyLevel)
-        gam = 0;
-    else
-        gam = 1;
-        gam2 = max(0, min(ener_B*(energy-aecmStruct.energyLevel)/(ener_C-aecmStruct.energyLevel), ener_B+(ener_A-ener_B)*(energy-ener_C)/(aecmStruct.energyMax-ener_C)));
-        if (ddE < dEWidth)
-            % Update counters
-            cntIn2 = cntIn2 + 1;
-            if (cntIn2 > 2)
-                cntOut2 = 0;
-            end
-            gam3 = max(dE_D, min(dE_A-(dE_A-dE_B)*(ddE/dE_C), dE_D+(dE_B-dE_D)*(dEWidth-ddE)/(dEWidth-dE_C)));
-            gam3 = dE_A;
-        else
-            % Update counters
-            cntOut2 = cntOut2 + 1;
-            if (cntOut2 > 2)
-                cntIn2 = 0;
-            end
-            %gam2 = 1;
-            gam3 = dE_D;
-        end
-        timeFactor = min(1, cntIn2/timeFactorLength);
-        gam = gam*(1-timeFactor) + timeFactor*gam2*gam3;
-    end
-    %gam = gam/floor(numCross/2+1);
-end
-if isempty(gam)
-    numCross
-    timeFactor
-    cntIn2
-    cntOut2
-    gam2
-    gam3
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/calcStepSize.m b/src/modules/audio_processing/aecm/main/matlab/matlab/calcStepSize.m
deleted file mode 100644
index ae1365f..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/calcStepSize.m
+++ /dev/null
@@ -1,105 +0,0 @@
-function [mu, aecmStructNew] = calcStepSize(energy, dE, aecmStruct, t, logscale)
-
-if (nargin < 4)
-    t = 1;
-    logscale = 1;
-elseif (nargin == 4)
-    logscale = 1;
-end
-T = aecmStruct.convLength;
-
-if logscale
-    currentMuMax = aecmStruct.MU_MIN + (aecmStruct.MU_MAX-aecmStruct.MU_MIN)*min(t,T)/T;
-    if (aecmStruct.energyMin >= aecmStruct.energyMax)
-        mu = aecmStruct.MU_MIN;
-    else
-        mu = (energy - aecmStruct.energyMin)/(aecmStruct.energyMax - aecmStruct.energyMin)*(currentMuMax-aecmStruct.MU_MIN) + aecmStruct.MU_MIN;
-    end
-    mu = 2^mu;
-    if (energy < aecmStruct.energyLevel)
-        mu = 0;
-    end
-else
-    muMin = 0;
-    muMax = 0.5;
-    currentMuMax = muMin + (muMax-muMin)*min(t,T)/T;
-    if (aecmStruct.energyMin >= aecmStruct.energyMax)
-        mu = muMin;
-    else
-        mu = (energy - aecmStruct.energyMin)/(aecmStruct.energyMax - aecmStruct.energyMin)*(currentMuMax-muMin) + muMin;
-    end
-end
-dE2 = 1;
-dEOffset = -0.5;
-offBoost = 5;
-if (mu > 0)
-    if (abs(dE-aecmStruct.ENERGY_DEV_OFFSET) > aecmStruct.ENERGY_DEV_TOL)
-        aecmStruct.muStruct.countInInterval = 0;
-    else
-        aecmStruct.muStruct.countInInterval = aecmStruct.muStruct.countInInterval + 1;
-    end
-    if (dE < aecmStruct.ENERGY_DEV_OFFSET - aecmStruct.ENERGY_DEV_TOL)
-        aecmStruct.muStruct.countOutLowInterval = aecmStruct.muStruct.countOutLowInterval + 1;
-    else
-        aecmStruct.muStruct.countOutLowInterval = 0;
-    end
-    if (dE > aecmStruct.ENERGY_DEV_OFFSET + aecmStruct.ENERGY_DEV_TOL)
-        aecmStruct.muStruct.countOutHighInterval = aecmStruct.muStruct.countOutHighInterval + 1;
-    else
-        aecmStruct.muStruct.countOutHighInterval = 0;
-    end
-end
-muVar = 2^min(-3,5/50*aecmStruct.muStruct.countInInterval-3);
-muOff = 2^max(offBoost,min(0,offBoost*(aecmStruct.muStruct.countOutLowInterval-aecmStruct.muStruct.minOutLowInterval)/(aecmStruct.muStruct.maxOutLowInterval-aecmStruct.muStruct.minOutLowInterval)));
-
-muLow = 1/64;
-muVar = 1;
-if (t < 2*T)
-    muDT = 1;
-    muVar = 1;
-    mdEVec = 0;
-    numCross = 0;
-else
-    muDT = min(1,max(muLow,1-(1-muLow)*(dE-aecmStruct.ENERGY_DEV_OFFSET)/aecmStruct.ENERGY_DEV_TOL));
-    dEVec = aecmStruct.enerNear(t-63:t)-aecmStruct.enerEcho(t-63:t);
-    %dEVec = aecmStruct.enerNear(t-20:t)-aecmStruct.enerEcho(t-20:t);
-    numCross = 0;
-    currentState = 0;
-    for ii=1:64
-        if (currentState == 0)
-            currentState = (dEVec(ii) > dE2) - (dEVec(ii) < -2);
-        elseif ((currentState == 1) & (dEVec(ii) < -2))
-            numCross = numCross + 1;
-            currentState = -1;
-        elseif ((currentState == -1) & (dEVec(ii) > dE2))
-            numCross = numCross + 1;
-            currentState = 1;
-        end
-    end
-            
-    %logicDEVec = (dEVec > dE2) - (dEVec < -2);
-    %numCross = sum(abs(diff(logicDEVec)));
-    %mdEVec = mean(abs(dEVec-dEOffset));
-    %mdEVec = mean(abs(dEVec-mean(dEVec)));
-    %mdEVec = max(dEVec)-min(dEVec);
-    %if (mdEVec > 4)%1.5)
-    %    muVar = 0;
-    %end
-    muVar = 2^(-floor(numCross/2));
-    muVar = 2^(-numCross);
-end
-%muVar = 1;
-
-
-% if (eStd > (dE2-dEOffset))
-%     muVar = 1/8;
-% else
-%     muVar = 1;
-% end
-
-%mu = mu*muDT*muVar*muOff;
-mu = mu*muDT*muVar;
-mu = min(mu,0.25);
-aecmStructNew = aecmStruct;
-%aecmStructNew.varMean = mdEVec;
-aecmStructNew.varMean = numCross;
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/fallerEstimator.m b/src/modules/audio_processing/aecm/main/matlab/matlab/fallerEstimator.m
deleted file mode 100644
index d038b51..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/fallerEstimator.m
+++ /dev/null
@@ -1,42 +0,0 @@
-function [U, Hnew] = fallerEstimator(Y, X, H, mu)
-
-% Near end signal is stacked frame by frame columnwise in matrix Y and far end in X
-%
-% Possible estimation procedures are
-% 1) LSE
-% 2) NLMS
-% 3) Separated numerator and denomerator filters
-regParam = 1;
-[numFreqs, numFrames] = size(Y);
-[numFreqs, Q] = size(X);
-U = zeros(numFreqs, 1);
-
-if ((nargin == 3) | (nargin == 5))
-    dtd = 0;
-end
-if (nargin == 4)
-    dtd = H;
-end
-Emax = 7;
-dEH = Emax-sum(sum(H));
-nu = 2*mu;
-% if (nargin < 5)
-%     H = zeros(numFreqs, Q);
-%     for kk = 1:numFreqs
-%         Xmatrix = hankel(X(kk,1:Q),X(kk,Q:end));
-%         y = Y(kk,1:end-Q+1)';
-%         H(kk,:) = (y'*Xmatrix')*inv(Xmatrix*Xmatrix'+regParam);
-%         U(kk,1) = H(kk,:)*Xmatrix(:,1);
-%     end
-% else
-    for kk = 1:numFreqs
-        x = X(kk,1:Q)';
-        y = Y(kk,1);
-        Htmp = mu*(y-H(kk,:)*x)/(x'*x+regParam)*x;
-        %Htmp = (mu*(y-H(kk,:)*x)/(x'*x+regParam) - nu/dEH)*x;
-        H(kk,:) = H(kk,:) + Htmp';
-        U(kk,1) = H(kk,:)*x;
-    end
-% end
-
-Hnew = H;
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/getBspectrum.m b/src/modules/audio_processing/aecm/main/matlab/matlab/getBspectrum.m
deleted file mode 100644
index a4a533d..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/getBspectrum.m
+++ /dev/null
@@ -1,22 +0,0 @@
-function bspectrum=getBspectrum(ps,threshold,bandfirst,bandlast)
-% function bspectrum=getBspectrum(ps,threshold,bandfirst,bandlast)
-% compute binary spectrum using threshold spectrum as pivot
-% bspectrum = binary spectrum (binary)
-% ps=current power spectrum (float)
-% threshold=threshold spectrum (float)
-% bandfirst = first band considered
-% bandlast = last band considered
-  
-% initialization stuff
-  if( length(ps)<bandlast | bandlast>32 | length(ps)~=length(threshold)) 
-  error('BinDelayEst:spectrum:invalid','Dimensionality error');
-end
-
-% get current binary spectrum
-diff = ps - threshold;
-bspectrum=uint32(0);
-for(i=bandfirst:bandlast)
-  if( diff(i)>0 ) 
-    bspectrum = bitset(bspectrum,i);
-  end
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/hisser2.m b/src/modules/audio_processing/aecm/main/matlab/matlab/hisser2.m
deleted file mode 100644
index 5a414f9..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/hisser2.m
+++ /dev/null
@@ -1,21 +0,0 @@
-function  bcount=hisser2(bs,bsr,bandfirst,bandlast)
-% function  bcount=hisser(bspectrum,bandfirst,bandlast)
-% histogram for the binary spectra
-% bcount= array of bit counts 
-% bs=binary spectrum (one int32 number each)  
-% bsr=reference binary spectra (one int32 number each)
-% blockSize = histogram over blocksize blocks
-% bandfirst = first band considered
-% bandlast = last band considered
-
-% weight all delays equally
-maxDelay = length(bsr);
-
-% compute counts (two methods; the first works better and is operational)
-bcount=zeros(maxDelay,1);
-for(i=1:maxDelay)
- % the delay should have low count for low-near&high-far and high-near&low-far
- bcount(i)= sum(bitget(bitxor(bs,bsr(i)),bandfirst:bandlast));  
- % the delay should have low count for low-near&high-far (works less well)
-% bcount(i)= sum(bitget(bitand(bsr(i),bitxor(bs,bsr(i))),bandfirst:bandlast));
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/mainProgram.m b/src/modules/audio_processing/aecm/main/matlab/matlab/mainProgram.m
deleted file mode 100644
index eeb2aaa..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/mainProgram.m
+++ /dev/null
@@ -1,283 +0,0 @@
-useHTC = 1; % Set this if you want to run a single file and set file names below. Otherwise use simEnvironment to run from several scenarios in a row
-delayCompensation_flag = 0; % Set this flag to one if you want to turn on the delay compensation/enhancement
-global FARENDFFT;
-global NEARENDFFT;
-global F;
-
-if useHTC
-%    fid=fopen('./htcTouchHd/nb/aecFar.pcm'); xFar=fread(fid,'short'); fclose(fid);
-%    fid=fopen('./htcTouchHd/nb/aecNear.pcm'); yNear=fread(fid,'short'); fclose(fid);
-%    fid=fopen('./samsungBlackjack/nb/aecFar.pcm'); xFar=fread(fid,'short'); fclose(fid);
-%    fid=fopen('./samsungBlackjack/nb/aecNear.pcm'); yNear=fread(fid,'short'); fclose(fid);
-%     fid=fopen('aecFarPoor.pcm'); xFar=fread(fid,'short'); fclose(fid);
-%     fid=fopen('aecNearPoor.pcm'); yNear=fread(fid,'short'); fclose(fid);
-%     fid=fopen('out_aes.pcm'); outAES=fread(fid,'short'); fclose(fid);
-   fid=fopen('aecFar4.pcm'); xFar=fread(fid,'short'); fclose(fid);
-   fid=fopen('aecNear4.pcm'); yNear=fread(fid,'short'); fclose(fid);
-    yNearSpeech = zeros(size(xFar));
-     fs = 8000;
-     frameSize = 64;
-%     frameSize = 128;
-     fs = 16000;
-%     frameSize = 256;
-%F = load('fftValues.txt');
-%FARENDFFT = F(:,1:33);
-%NEARENDFFT = F(:,34:66);
-
-else
-    loadFileFar = [speakerType, '_s_',scenario,'_far_b.wav'];
-    [xFar,fs,nbits] = wavread(loadFileFar);
-    xFar = xFar*2^(nbits-1);
-    loadFileNear = [speakerType, '_s_',scenario,'_near_b.wav'];
-    [yNear,fs,nbits] = wavread(loadFileNear);
-    yNear = yNear*2^(nbits-1);
-    loadFileNearSpeech = [speakerType, '_s_',scenario,'_nearSpeech_b.wav'];
-    [yNearSpeech,fs,nbits] = wavread(loadFileNearSpeech);
-    yNearSpeech = yNearSpeech*2^(nbits-1);
-    frameSize = 256;
-end
-
-dtRegions = [];
-
-% General settings for the AECM
-setupStruct = struct(...
-    'stepSize_flag', 1,...      % This flag turns on the step size calculation. If turned off, mu = 0.25.
-    'supGain_flag', 0,...       % This flag turns on the suppression gain calculation. If turned off, gam = 1.
-    'channelUpdate_flag', 0,... % This flag turns on the channel update. If turned off, H is updated for convLength and then kept constant.
-    'nlp_flag', 0,...           % Turn on/off NLP
-    'withVAD_flag', 0,...           % Turn on/off NLP
-    'useSubBand', 0,...         % Set to 1 if to use subBands
-    'useDelayEstimation', 1,... % Set to 1 if to use delay estimation
-    'support', frameSize,...    % # of samples per frame
-    'samplingfreq',fs,...       % Sampling frequency
-    'oversampling', 2,...       % Overlap between blocks/frames
-    'updatel', 0,...            % # of samples between blocks
-    'hsupport1', 0,...          % # of bins in frequency domain
-    'factor', 0,...             % synthesis window amplification
-    'tlength', 0,...            % # of samples of entire file
-    'updateno', 0,...           % # of updates
-    'nb', 1,...                 % # of blocks
-    'currentBlock', 0,...       %
-    'win', zeros(frameSize,1),...% Window to apply for fft and synthesis
-    'avtime', 1,...             % Time (in sec.) to perform averaging
-    'estLen', 0,...             % Averaging in # of blocks
-    'A_GAIN', 10.0,...          % 
-    'suppress_overdrive', 1.0,...   % overdrive factor for suppression 1.4 is good
-    'gamma_echo', 1.0,...       % same as suppress_overdrive but at different place
-    'de_echo_bound', 0.0,...    %
-    'nl_alpha', 0.4,...         % memory; seems not very critical
-    'nlSeverity', 0.2,...         % nonlinearity severity: 0 does nothing; 1 suppresses all
-    'numInBand', [],...         % # of frequency bins in resp. subBand
-    'centerFreq', [],...        % Center frequency of resp. subBand
-    'dtRegions', dtRegions,...  % Regions where we have DT
-    'subBandLength', frameSize/2);%All bins
-    %'subBandLength', 11);       %Something's wrong when subBandLength even
-    %'nl_alpha', 0.8,...         % memory; seems not very critical
-
-delayStruct = struct(...
-    'bandfirst', 8,...
-    'bandlast', 25,...
-    'smlength', 600,...
-    'maxDelay', 0.4,...
-    'oneGoodEstimate', 0,...
-    'delayAdjust', 0,...
-    'maxDelayb', 0);
-% More parameters in delayStruct are constructed in "updateSettings" below
-
-% Make struct settings
-[setupStruct, delayStruct] = updateSettings(yNear, xFar, setupStruct, delayStruct);
-setupStruct.numInBand = ones(setupStruct.hsupport1,1);
-
-Q = 1; % Time diversity in channel
-% General settings for the step size calculation
-muStruct = struct(...
-    'countInInterval', 0,...
-    'countOutHighInterval', 0,...
-    'countOutLowInterval', 0,...
-    'minInInterval', 50,...
-    'minOutHighInterval', 10,...
-    'minOutLowInterval', 10,...
-    'maxOutLowInterval', 50);
-% General settings for the AECM
-aecmStruct = struct(...
-    'plotIt', 0,... % Set to 0 to turn off plotting
-    'useSubBand', 0,...
-    'bandFactor', 1,...
-    'H', zeros(setupStruct.subBandLength+1,Q),...
-    'HStored', zeros(setupStruct.subBandLength+1,Q),...
-    'X', zeros(setupStruct.subBandLength+1,Q),...
-    'energyThres', 0.28,...
-    'energyThresMSE', 0.4,...
-    'energyMin', inf,...
-    'energyMax', -inf,...
-    'energyLevel', 0,...
-    'energyLevelMSE', 0,...
-    'convLength', 100,...
-    'gammaLog', ones(setupStruct.updateno,1),...
-    'muLog', ones(setupStruct.updateno,1),...
-    'enerFar', zeros(setupStruct.updateno,1),...
-    'enerNear', zeros(setupStruct.updateno,1),...
-    'enerEcho', zeros(setupStruct.updateno,1),...
-    'enerEchoStored', zeros(setupStruct.updateno,1),...
-    'enerOut', zeros(setupStruct.updateno,1),...
-    'runningfmean', 0,...
-    'muStruct', muStruct,...
-    'varMean', 0,...
-    'countMseH', 0,...
-    'mseHThreshold', 1.1,...
-    'mseHStoredOld', inf,...
-    'mseHLatestOld', inf,...
-    'delayLatestS', zeros(1,51),...
-    'feedbackDelay', 0,...
-    'feedbackDelayUpdate', 0,...
-    'cntIn', 0,...
-    'cntOut', 0,...
-    'FAR_ENERGY_MIN', 1,...
-    'ENERGY_DEV_OFFSET', 0.5,...
-    'ENERGY_DEV_TOL', 1.5,...
-    'MU_MIN', -16,...
-    'MU_MAX', -2,...
-    'newDelayCurve', 0);
-
-% Adjust speech signals
-xFar = [zeros(setupStruct.hsupport1-1,1);xFar(1:setupStruct.tlength)];
-yNear = [zeros(setupStruct.hsupport1-1,1);yNear(1:setupStruct.tlength)];
-yNearSpeech = [zeros(setupStruct.hsupport1-1,1);yNearSpeech(1:setupStruct.tlength)];
-xFar = xFar(1:setupStruct.tlength);
-yNear = yNear(1:setupStruct.tlength);
-
-% Set figure settings
-if aecmStruct.plotIt
-    figure(13)
-    set(gcf,'doublebuffer','on')
-end
-%%%%%%%%%%
-% Here starts the algorithm
-% Dividing into frames and then estimating the near end speech
-%%%%%%%%%%
-fTheFarEnd      = complex(zeros(setupStruct.hsupport1,1));
-afTheFarEnd     = zeros(setupStruct.hsupport1,setupStruct.updateno+1);
-fFar            = zeros(setupStruct.hsupport1,setupStruct.updateno+1);
-fmicrophone     = complex(zeros(setupStruct.hsupport1,1));
-afmicrophone    = zeros(setupStruct.hsupport1,setupStruct.updateno+1);
-fNear           = zeros(setupStruct.hsupport1,setupStruct.updateno+1);
-femicrophone    = complex(zeros(setupStruct.hsupport1,1));
-emicrophone     = zeros(setupStruct.tlength,1);
-
-if (setupStruct.useDelayEstimation == 2)
-    delSamples = [1641 1895 2032 1895 2311 2000 2350 2222 NaN 2332 2330 2290 2401 2415 NaN 2393 2305 2381 2398];
-    delBlocks = round(delSamples/setupStruct.updatel);
-    delStarts = floor([25138 46844 105991 169901 195739 218536 241803 333905 347703 362660 373753 745135 765887 788078 806257 823835 842443 860139 881869]/setupStruct.updatel);
-else
-    delStarts = [];
-end
-
-for i=1:setupStruct.updateno
-    setupStruct.currentBlock = i;
-    
-    sb = (i-1)*setupStruct.updatel + 1;
-    se = sb + setupStruct.support - 1;
-    
-    %%%%%%%
-    % Analysis FFTs
-    %%%%%%%
-    % Far end signal
-    temp = fft(setupStruct.win .* xFar(sb:se))/frameSize;
-    fTheFarEnd = temp(1:setupStruct.hsupport1);
-    afTheFarEnd(:,i) = abs(fTheFarEnd);
-    fFar(:,i) = fTheFarEnd;
-    % Near end signal
-    temp = fft(setupStruct.win .* yNear(sb:se))/frameSize;%,pause
-    fmicrophone = temp(1:setupStruct.hsupport1);
-    afmicrophone(:,i) = abs(fmicrophone);
-    fNear(:,i) = fmicrophone;
-    %abs(fmicrophone),pause
-    % The true near end speaker (if we have such info)
-    temp = fft(setupStruct.win .* yNearSpeech(sb:se));
-    aftrueSpeech = abs(temp(1:setupStruct.hsupport1));
-    
-    if(i == 1000)
-        %break;
-    end
-    
-    % Perform delay estimation
-    if (setupStruct.useDelayEstimation == 1)
-        % Delay Estimation
-        delayStruct = align(fTheFarEnd, fmicrophone, delayStruct, i);
-        %delayStruct.delay(i) = 39;%19;
-        idel =  max(i - delayStruct.delay(i) + 1,1);
-        
-        if delayCompensation_flag
-            % If we have a new delay estimate from Bastiaan's alg. update the offset
-            if (delayStruct.delay(i) ~= delayStruct.delay(max(1,i-1)))
-                delayStruct.delayAdjust = delayStruct.delayAdjust + delayStruct.delay(i) - delayStruct.delay(i-1);
-            end
-            % Store the compensated delay
-            delayStruct.delayNew(i) = delayStruct.delay(i) - delayStruct.delayAdjust;
-            if (delayStruct.delayNew(i) < 1)
-                % Something's wrong
-                pause,break
-            end
-            % Compensate with the offset estimate
-            idel = idel + delayStruct.delayAdjust;
-        end
-        if 0%aecmStruct.plotIt
-            figure(1)
-            plot(1:i,delayStruct.delay(1:i),'k:',1:i,delayStruct.delayNew(1:i),'k--','LineWidth',2),drawnow
-        end
-    elseif (setupStruct.useDelayEstimation == 2)
-        % Use "manual delay"
-        delIndex = find(delStarts<i);
-        if isempty(delIndex)
-            idel = i;
-        else
-            idel = i - delBlocks(max(delIndex));
-            if isnan(idel)
-                idel = i - delBlocks(max(delIndex)-1);
-            end
-        end
-    else
-        % No delay estimation
-        %idel = max(i - 18, 1);
-        idel = max(i - 50, 1);
-    end
-
-    %%%%%%%%
-    % This is the AECM algorithm
-    %
-    % Output is the new frequency domain signal (hopefully) echo compensated
-    %%%%%%%%
-    [femicrophone, aecmStruct] = AECMobile(fmicrophone, afTheFarEnd(:,idel), setupStruct, aecmStruct);
-    %[femicrophone, aecmStruct] = AECMobile(fmicrophone, FARENDFFT(idel,:)'/2^F(idel,end-1), setupStruct, aecmStruct);
-    
-    if aecmStruct.feedbackDelayUpdate
-        % If the feedback tells us there is a new offset out there update the enhancement
-        delayStruct.delayAdjust = delayStruct.delayAdjust + aecmStruct.feedbackDelay;
-        aecmStruct.feedbackDelayUpdate = 0;
-    end
-    
-    % reconstruction; first make spectrum odd
-    temp = [femicrophone; flipud(conj(femicrophone(2:(setupStruct.hsupport1-1))))];
-    emicrophone(sb:se) = emicrophone(sb:se) + setupStruct.factor * setupStruct.win .* real(ifft(temp))*frameSize;
-    if max(isnan(emicrophone(sb:se)))
-        % Something's wrong with the output at block i
-        i
-        break
-    end
-end
-
-
-if useHTC
-    fid=fopen('aecOutMatlabC.pcm','w');fwrite(fid,int16(emicrophone),'short');fclose(fid);
-    %fid=fopen('farendFFT.txt','w');fwrite(fid,int16(afTheFarEnd(:)),'short');fclose(fid);
-    %fid=fopen('farendFFTreal.txt','w');fwrite(fid,int16(imag(fFar(:))),'short');fclose(fid);
-    %fid=fopen('farendFFTimag.txt','w');fwrite(fid,int16(real(fFar(:))),'short');fclose(fid);
-    %fid=fopen('nearendFFT.txt','w');fwrite(fid,int16(afmicrophone(:)),'short');fclose(fid);
-    %fid=fopen('nearendFFTreal.txt','w');fwrite(fid,int16(real(fNear(:))),'short');fclose(fid);
-    %fid=fopen('nearendFFTimag.txt','w');fwrite(fid,int16(imag(fNear(:))),'short');fclose(fid);
-end
-if useHTC
-    %spclab(setupStruct.samplingfreq,xFar,yNear,emicrophone)
-else
-    spclab(setupStruct.samplingfreq,xFar,yNear,emicrophone,yNearSpeech)
-end    
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/simEnvironment.m b/src/modules/audio_processing/aecm/main/matlab/matlab/simEnvironment.m
deleted file mode 100644
index 3ebe701..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/simEnvironment.m
+++ /dev/null
@@ -1,15 +0,0 @@
-speakerType = 'fm';
-%for k=2:5
-%for k=[2 4 5]
-for k=3
-    scenario = int2str(k);
-    fprintf('Current scenario: %d\n',k)
-    mainProgram
-    %saveFile = [speakerType, '_s_',scenario,'_delayEst_v2_vad_man.wav'];
-    %wavwrite(emic,fs,nbits,saveFile);
-    %saveFile = ['P:\Engineering_share\BjornV\AECM\',speakerType, '_s_',scenario,'_delayEst_v2_vad_man.pcm'];
-    %saveFile = [speakerType, '_s_',scenario,'_adaptMu_adaptGamma_withVar_gammFilt_HSt.pcm'];
-    saveFile = ['scenario_',scenario,'_090417_backupH_nlp.pcm'];
-    fid=fopen(saveFile,'w');fwrite(fid,int16(emicrophone),'short');fclose(fid);
-    %pause
-end
diff --git a/src/modules/audio_processing/aecm/main/matlab/matlab/updateSettings.m b/src/modules/audio_processing/aecm/main/matlab/matlab/updateSettings.m
deleted file mode 100644
index c805f1d..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/matlab/updateSettings.m
+++ /dev/null
@@ -1,94 +0,0 @@
-function [setupStructNew, delayStructNew] = updateSettings(microphone, TheFarEnd, setupStruct, delayStruct);
-
-% other, constants
-setupStruct.hsupport1 = setupStruct.support/2 + 1;
-setupStruct.factor =  2 / setupStruct.oversampling;
-setupStruct.updatel = setupStruct.support/setupStruct.oversampling;
-setupStruct.estLen = round(setupStruct.avtime * setupStruct.samplingfreq/setupStruct.updatel);
-
-% compute some constants
-blockLen = setupStruct.support/setupStruct.oversampling;
-delayStruct.maxDelayb = floor(setupStruct.samplingfreq*delayStruct.maxDelay/setupStruct.updatel); % in blocks
-
-%input
-tlength = min([length(microphone),length(TheFarEnd)]);
-updateno = floor(tlength/setupStruct.updatel);
-setupStruct.tlength = setupStruct.updatel*updateno;
-setupStruct.updateno = updateno - setupStruct.oversampling + 1;
-
-% signal length
-n = floor(min([length(TheFarEnd), length(microphone)])/setupStruct.support)*setupStruct.support;
-setupStruct.nb = n/blockLen - setupStruct.oversampling + 1; % in blocks
-
-setupStruct.win = sqrt([0 ; hanning(setupStruct.support-1)]);
-
-% Construct filterbank in Bark-scale
-
-K = setupStruct.subBandLength; %Something's wrong when K even
-erbs = 21.4*log10(0.00437*setupStruct.samplingfreq/2+1);
-fe = (10.^((0:K)'*erbs/K/21.4)-1)/0.00437;
-setupStruct.centerFreq = fe;
-H = diag(ones(1,K-1))+diag(ones(1,K-2),-1);
-Hinv = inv(H);
-aty = 2*Hinv(end,:)*fe(2:end-1);
-boundary = aty - (setupStruct.samplingfreq/2 + fe(end-1))/2;
-if rem(K,2)
-    x1 = min([fe(2)/2, -boundary]);
-else
-    x1 = max([0, boundary]);
-end
-%x1
-g = fe(2:end-1);
-g(1) = g(1) - x1/2;
-x = 2*Hinv*g;
-x = [x1;x];
-%figure(42), clf
-xy = zeros((K+1)*4,1);
-yy = zeros((K+1)*4,1);
-xy(1:4) = [fe(1) fe(1) x(1) x(1)]';
-yy(1:4) = [0 1 1 0]'/x(1);
-for kk=2:K
-    xy((kk-1)*4+(1:4)) = [x(kk-1) x(kk-1) x(kk) x(kk)]';
-    yy((kk-1)*4+(1:4)) = [0 1 1 0]'/(x(kk)-x(kk-1));
-end
-xy(end-3:end) = [x(K) x(K) fe(end) fe(end)]';
-yy(end-3:end) = [0 1 1 0]'/(fe(end)*2-2*x(K));
-%plot(xy,yy,'LineWidth',2)
-%fill(xy,yy,'y')
-
-x = [0;x];
-xk = x*setupStruct.hsupport1/setupStruct.samplingfreq*2;
-%setupStruct.erbBoundaries = xk;
-numInBand = zeros(length(xk),1);
-xh = (0:setupStruct.hsupport1-1);
-
-for kk=1:length(xk)
-    if (kk==length(xk))
-        numInBand(kk) = length(find(xh>=xk(kk)));
-    else
-        numInBand(kk) = length(intersect(find(xh>=xk(kk)),find(xh<xk(kk+1))));
-    end
-end
-setupStruct.numInBand = numInBand;
-
-setupStructNew = setupStruct;
-
-delayStructNew = struct(...
-    'sxAll2',zeros(setupStructNew.hsupport1,setupStructNew.nb),...
-    'syAll2',zeros(setupStructNew.hsupport1,setupStructNew.nb),...
-    'z200',zeros(5,setupStructNew.hsupport1),...
-    'z500',zeros(5,delayStruct.maxDelayb+1),...
-    'bxspectrum',uint32(zeros(setupStructNew.nb,1)),...
-    'byspectrum',uint32(zeros(setupStructNew.nb,1)),...
-    'bandfirst',delayStruct.bandfirst,'bandlast',delayStruct.bandlast,...
-    'bxhist',uint32(zeros(delayStruct.maxDelayb+1,1)),...
-    'bcount',zeros(1+delayStruct.maxDelayb,setupStructNew.nb),...
-    'fout',zeros(1+delayStruct.maxDelayb,setupStructNew.nb),...
-    'new',zeros(1+delayStruct.maxDelayb,setupStructNew.nb),...
-    'smlength',delayStruct.smlength,...
-    'maxDelay', delayStruct.maxDelay,...
-    'maxDelayb', delayStruct.maxDelayb,...
-    'oneGoodEstimate', 0,...
-    'delayAdjust', 0,...
-    'delayNew',zeros(setupStructNew.nb,1),...
-    'delay',zeros(setupStructNew.nb,1));
diff --git a/src/modules/audio_processing/aecm/main/matlab/waitbar_j.m b/src/modules/audio_processing/aecm/main/matlab/waitbar_j.m
deleted file mode 100644
index 50b9ccf..0000000
--- a/src/modules/audio_processing/aecm/main/matlab/waitbar_j.m
+++ /dev/null
@@ -1,234 +0,0 @@
-function fout = waitbar_j(x,whichbar, varargin)
-%WAITBAR Display wait bar.
-%   H = WAITBAR(X,'title', property, value, property, value, ...) 
-%   creates and displays a waitbar of fractional length X.  The 
-%   handle to the waitbar figure is returned in H.
-%   X should be between 0 and 1.  Optional arguments property and 
-%   value allow to set corresponding waitbar figure properties.
-%   Property can also be an action keyword 'CreateCancelBtn', in 
-%   which case a cancel button will be added to the figure, and 
-%   the passed value string will be executed upon clicking on the 
-%   cancel button or the close figure button.
-%
-%   WAITBAR(X) will set the length of the bar in the most recently
-%   created waitbar window to the fractional length X.
-%
-%   WAITBAR(X,H) will set the length of the bar in waitbar H
-%   to the fractional length X.
-%
-%   WAITBAR(X,H,'updated title') will update the title text in
-%   the waitbar figure, in addition to setting the fractional
-%   length to X.
-%
-%   WAITBAR is typically used inside a FOR loop that performs a 
-%   lengthy computation.  A sample usage is shown below:
-%
-%       h = waitbar(0,'Please wait...');
-%       for i=1:100,
-%           % computation here %
-%           waitbar(i/100,h)
-%       end
-%       close(h)
-
-%   Clay M. Thompson 11-9-92
-%   Vlad Kolesnikov  06-7-99
-%   Copyright 1984-2001 The MathWorks, Inc.
-%   $Revision: 1.22 $  $Date: 2001/04/15 12:03:29 $
-
-if nargin>=2
-    if ischar(whichbar)
-        type=2; %we are initializing
-        name=whichbar;
-    elseif isnumeric(whichbar)
-        type=1; %we are updating, given a handle
-        f=whichbar;
-    else
-        error(['Input arguments of type ' class(whichbar) ' not valid.'])
-    end
-elseif nargin==1
-    f = findobj(allchild(0),'flat','Tag','TMWWaitbar');
-    
-    if isempty(f)
-        type=2;
-        name='Waitbar';
-    else
-        type=1;
-        f=f(1);
-    end   
-else
-    error('Input arguments not valid.');
-end
-
-x = max(0,min(100*x,100));
-
-switch type
- case 1,  % waitbar(x)    update
-  p = findobj(f,'Type','patch');
-  l = findobj(f,'Type','line');
-  if isempty(f) | isempty(p) | isempty(l), 
-      error('Couldn''t find waitbar handles.'); 
-  end
-  xpatch = get(p,'XData');
-  xpatch = [0 x x 0];
-  set(p,'XData',xpatch)
-  xline = get(l,'XData');
-  set(l,'XData',xline);
-  
-  if nargin>2,
-      % Update waitbar title:
-      hAxes = findobj(f,'type','axes');
-      hTitle = get(hAxes,'title');
-      set(hTitle,'string',varargin{1});
-  end
-  
- case 2,  % waitbar(x,name)  initialize
-  vertMargin = 0;
-  if nargin > 2,
-      % we have optional arguments: property-value pairs
-      if rem (nargin, 2 ) ~= 0
-          error( 'Optional initialization arguments must be passed in pairs' );
-      end
-  end
-  
-  oldRootUnits = get(0,'Units');
-
-  set(0, 'Units', 'points');
-  screenSize = get(0,'ScreenSize');
-  
-  axFontSize=get(0,'FactoryAxesFontSize');
-  
-  pointsPerPixel = 72/get(0,'ScreenPixelsPerInch');
-  
-  width = 360 * pointsPerPixel;
-  height = 75 * pointsPerPixel;
-  pos = [screenSize(3)/2-width/2 screenSize(4)/2-height/2 width height];
-
-%pos=  [501.75 589.5 393.75 52.5];
-  f = figure(...
-      'Units', 'points', ...
-      'BusyAction', 'queue', ...
-      'Position', pos, ...
-      'Resize','on', ...
-      'CreateFcn','', ...
-      'NumberTitle','off', ...
-      'IntegerHandle','off', ...
-      'MenuBar', 'none', ...
-      'Tag','TMWWaitbar',...
-      'Interruptible', 'off', ...
-      'Visible','on');
-  
-  %%%%%%%%%%%%%%%%%%%%%
-  % set figure properties as passed to the fcn
-  % pay special attention to the 'cancel' request
-  %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
-  if nargin > 2,
-      propList = varargin(1:2:end);
-      valueList = varargin(2:2:end);
-      cancelBtnCreated = 0;
-      for ii = 1:length( propList )
-          try
-              if strcmp(lower(propList{ii}), 'createcancelbtn' ) & ~cancelBtnCreated
-                  cancelBtnHeight = 23 * pointsPerPixel;
-                  cancelBtnWidth = 60 * pointsPerPixel;
-                  newPos = pos;
-                  vertMargin = vertMargin + cancelBtnHeight;
-                  newPos(4) = newPos(4)+vertMargin;
-                  callbackFcn = [valueList{ii}];
-                  set( f, 'Position', newPos, 'CloseRequestFcn', callbackFcn );
-                  cancelButt = uicontrol('Parent',f, ...
-                                         'Units','points', ...
-                                         'Callback',callbackFcn, ...
-                                         'ButtonDownFcn', callbackFcn, ...
-                                         'Enable','on', ...
-                                         'Interruptible','off', ...
-                                         'Position', [pos(3)-cancelBtnWidth*1.4, 7,  ...
-                    cancelBtnWidth, cancelBtnHeight], ...
-                                         'String','Cancel', ...
-                                         'Tag','TMWWaitbarCancelButton');
-                  cancelBtnCreated = 1;
-              else
-                  % simply set the prop/value pair of the figure
-                  set( f, propList{ii}, valueList{ii});
-              end
-          catch
-              disp ( ['Warning: could not set property ''' propList{ii} ''' with value ''' num2str(valueList{ii}) '''' ] );
-          end
-      end
-  end  
-  
-  %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%  
-  
-  
-  colormap([]);
-  
-  axNorm=[.05 .3 .9 .2];
- % axNorm=[1 1 1 1];
-  axPos=axNorm.*[pos(3:4),pos(3:4)] + [0 vertMargin 0 0];
-  
-  h = axes('XLim',[0 100],...
-           'YLim',[0 1],...
-           'Box','on', ...
-           'Units','Points',...
-           'FontSize', axFontSize,...
-           'Position',axPos,...
-           'XTickMode','manual',...
-           'YTickMode','manual',...
-           'XTick',[],...
-           'YTick',[],...
-           'XTickLabelMode','manual',...
-           'XTickLabel',[],...
-           'YTickLabelMode','manual',...
-           'YTickLabel',[]);
-  
-  tHandle=title(name);
-  tHandle=get(h,'title');
-  oldTitleUnits=get(tHandle,'Units');
-  set(tHandle,...
-      'Units',      'points',...
-      'String',     name);
-  
-  tExtent=get(tHandle,'Extent');
-  set(tHandle,'Units',oldTitleUnits);
-  
-  titleHeight=tExtent(4)+axPos(2)+axPos(4)+5;
-  if titleHeight>pos(4)
-      pos(4)=titleHeight;
-      pos(2)=screenSize(4)/2-pos(4)/2;
-      figPosDirty=logical(1);
-  else
-      figPosDirty=logical(0);
-  end
-  
-  if tExtent(3)>pos(3)*1.10;
-      pos(3)=min(tExtent(3)*1.10,screenSize(3));
-      pos(1)=screenSize(3)/2-pos(3)/2;
-      
-      axPos([1,3])=axNorm([1,3])*pos(3);
-      set(h,'Position',axPos);
-      
-      figPosDirty=logical(1);
-  end
-  
-  if figPosDirty
-      set(f,'Position',pos);
-  end
-
-  xpatch = [0 x x 0];
-  ypatch = [0 0 1 1];
-   xline = [100 0 0 100 100];
-   yline = [0 0 1 1 0];
-  
-  p = patch(xpatch,ypatch,'r','EdgeColor','r','EraseMode','none');
-  l = line(xline,yline,'EraseMode','none');
-  set(l,'Color',get(gca,'XColor'));
-  
-  
-  set(f,'HandleVisibility','callback','visible','on', 'resize','off');
-  
-  set(0, 'Units', oldRootUnits);
-end  % case
-drawnow;
-
-if nargout==1,
-    fout = f;
-end
diff --git a/src/modules/audio_processing/aecm/main/source/Android.mk b/src/modules/audio_processing/aecm/main/source/Android.mk
deleted file mode 100644
index ec18258..0000000
--- a/src/modules/audio_processing/aecm/main/source/Android.mk
+++ /dev/null
@@ -1,52 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_aecm
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := echo_control_mobile.c \
-    aecm_core.c 
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=  
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
-    '-DNO_HEAPCHECKER' \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_THREAD_RR' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' 
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../../.. \
-    $(LOCAL_PATH)/../interface \
-    $(LOCAL_PATH)/../../../utility \
-    $(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface 
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS := 
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
-    libdl \
-    libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/aecm/main/source/aecm.gyp b/src/modules/audio_processing/aecm/main/source/aecm.gyp
deleted file mode 100644
index a535d2b..0000000
--- a/src/modules/audio_processing/aecm/main/source/aecm.gyp
+++ /dev/null
@@ -1,43 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../../../../common_settings.gypi',
-  ],
-  'targets': [
-    {
-      'target_name': 'aecm',
-      'type': '<(library)',
-      'dependencies': [
-        '../../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
-        '../../../utility/util.gyp:apm_util'
-      ],
-      'include_dirs': [
-        '../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-        ],
-      },
-      'sources': [
-        '../interface/echo_control_mobile.h',
-        'echo_control_mobile.c',
-        'aecm_core.c',
-        'aecm_core.h',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/aecm/main/source/aecm_core.c b/src/modules/audio_processing/aecm/main/source/aecm_core.c
deleted file mode 100644
index f17f1bf..0000000
--- a/src/modules/audio_processing/aecm/main/source/aecm_core.c
+++ /dev/null
@@ -1,2534 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-
-#include "aecm_core.h"
-#include "ring_buffer.h"
-#include "echo_control_mobile.h"
-#include "typedefs.h"
-
-// TODO(bjornv): Will be removed in final version.
-//#include <stdio.h>
-
-#ifdef ARM_WINM_LOG
-#include <stdio.h>
-#include <windows.h>
-#endif
-
-// BANDLAST - BANDFIRST must be < 32
-#define BANDFIRST                   12   // Only bit BANDFIRST through bit BANDLAST are processed
-#define BANDLAST                    43
-
-#ifdef ARM_WINM
-#define WebRtcSpl_AddSatW32(a,b)  _AddSatInt(a,b)
-#define WebRtcSpl_SubSatW32(a,b)  _SubSatInt(a,b)
-#endif
-// 16 instructions on most risc machines for 32-bit bitcount !
-
-#ifdef AEC_DEBUG
-FILE *dfile;
-FILE *testfile;
-#endif
-
-#ifdef AECM_SHORT
-
-// Square root of Hanning window in Q14
-static const WebRtc_Word16 kSqrtHanning[] =
-{
-    0, 804, 1606, 2404, 3196, 3981, 4756, 5520,
-    6270, 7005, 7723, 8423, 9102, 9760, 10394, 11003,
-    11585, 12140, 12665, 13160, 13623, 14053, 14449, 14811,
-    15137, 15426, 15679, 15893, 16069, 16207, 16305, 16364,
-    16384
-};
-
-#else
-
-// Square root of Hanning window in Q14
-static const WebRtc_Word16 kSqrtHanning[] = {0, 399, 798, 1196, 1594, 1990, 2386, 2780, 3172,
-        3562, 3951, 4337, 4720, 5101, 5478, 5853, 6224, 6591, 6954, 7313, 7668, 8019, 8364,
-        8705, 9040, 9370, 9695, 10013, 10326, 10633, 10933, 11227, 11514, 11795, 12068, 12335,
-        12594, 12845, 13089, 13325, 13553, 13773, 13985, 14189, 14384, 14571, 14749, 14918,
-        15079, 15231, 15373, 15506, 15631, 15746, 15851, 15947, 16034, 16111, 16179, 16237,
-        16286, 16325, 16354, 16373, 16384};
-
-#endif
-
-//Q15 alpha = 0.99439986968132  const Factor for magnitude approximation
-static const WebRtc_UWord16 kAlpha1 = 32584;
-//Q15 beta = 0.12967166976970   const Factor for magnitude approximation
-static const WebRtc_UWord16 kBeta1 = 4249;
-//Q15 alpha = 0.94234827210087  const Factor for magnitude approximation
-static const WebRtc_UWord16 kAlpha2 = 30879;
-//Q15 beta = 0.33787806009150   const Factor for magnitude approximation
-static const WebRtc_UWord16 kBeta2 = 11072;
-//Q15 alpha = 0.82247698684306  const Factor for magnitude approximation
-static const WebRtc_UWord16 kAlpha3 = 26951;
-//Q15 beta = 0.57762063060713   const Factor for magnitude approximation
-static const WebRtc_UWord16 kBeta3 = 18927;
-
-// Initialization table for echo channel in 8 kHz
-static const WebRtc_Word16 kChannelStored8kHz[PART_LEN1] = {
-    2040,   1815,   1590,   1498,   1405,   1395,   1385,   1418,
-    1451,   1506,   1562,   1644,   1726,   1804,   1882,   1918,
-    1953,   1982,   2010,   2025,   2040,   2034,   2027,   2021,
-    2014,   1997,   1980,   1925,   1869,   1800,   1732,   1683,
-    1635,   1604,   1572,   1545,   1517,   1481,   1444,   1405,
-    1367,   1331,   1294,   1270,   1245,   1239,   1233,   1247,
-    1260,   1282,   1303,   1338,   1373,   1407,   1441,   1470,
-    1499,   1524,   1549,   1565,   1582,   1601,   1621,   1649,
-    1676
-};
-
-// Initialization table for echo channel in 16 kHz
-static const WebRtc_Word16 kChannelStored16kHz[PART_LEN1] = {
-    2040,   1590,   1405,   1385,   1451,   1562,   1726,   1882,
-    1953,   2010,   2040,   2027,   2014,   1980,   1869,   1732,
-    1635,   1572,   1517,   1444,   1367,   1294,   1245,   1233,
-    1260,   1303,   1373,   1441,   1499,   1549,   1582,   1621,
-    1676,   1741,   1802,   1861,   1921,   1983,   2040,   2102,
-    2170,   2265,   2375,   2515,   2651,   2781,   2922,   3075,
-    3253,   3471,   3738,   3976,   4151,   4258,   4308,   4288,
-    4270,   4253,   4237,   4179,   4086,   3947,   3757,   3484,
-    3153
-};
-
-#ifdef ARM_WINM_LOG
-HANDLE logFile = NULL;
-#endif
-
-static void WebRtcAecm_ComfortNoise(AecmCore_t* const aecm, const WebRtc_UWord16 * const dfa,
-                                    WebRtc_Word16 * const outReal,
-                                    WebRtc_Word16 * const outImag,
-                                    const WebRtc_Word16 * const lambda);
-
-static __inline WebRtc_UWord32 WebRtcAecm_SetBit(WebRtc_UWord32 in, WebRtc_Word32 pos)
-{
-    WebRtc_UWord32 mask, out;
-
-    mask = WEBRTC_SPL_SHIFT_W32(1, pos);
-    out = (in | mask);
-
-    return out;
-}
-
-// WebRtcAecm_Hisser(...)
-//
-// This function compares the binary vector specvec with all rows of the binary matrix specmat
-// and counts per row the number of times they have the same value.
-// Input:
-//       - specvec   : binary "vector"  that is stored in a long
-//       - specmat   : binary "matrix"  that is stored as a vector of long
-// Output:
-//       - bcount    : "Vector" stored as a long, containing for each row the number of times
-//                      the matrix row and the input vector have the same value
-//
-//
-void WebRtcAecm_Hisser(const WebRtc_UWord32 specvec, const WebRtc_UWord32 * const specmat,
-                       WebRtc_UWord32 * const bcount)
-{
-    int n;
-    WebRtc_UWord32 a, b;
-    register WebRtc_UWord32 tmp;
-
-    a = specvec;
-    // compare binary vector specvec with all rows of the binary matrix specmat
-    for (n = 0; n < MAX_DELAY; n++)
-    {
-        b = specmat[n];
-        a = (specvec ^ b);
-        // Returns bit counts in tmp
-        tmp = a - ((a >> 1) & 033333333333) - ((a >> 2) & 011111111111);
-        tmp = ((tmp + (tmp >> 3)) & 030707070707);
-        tmp = (tmp + (tmp >> 6));
-        tmp = (tmp + (tmp >> 12) + (tmp >> 24)) & 077;
-
-        bcount[n] = tmp;
-    }
-}
-
-// WebRtcAecm_BSpectrum(...)
-//
-// Computes the binary spectrum by comparing the input spectrum with a threshold spectrum.
-//
-// Input:
-//       - spectrum  : Spectrum of which the binary spectrum should be calculated.
-//       - thresvec  : Threshold spectrum with which the input spectrum is compared.
-// Return:
-//       - out       : Binary spectrum
-//
-WebRtc_UWord32 WebRtcAecm_BSpectrum(const WebRtc_UWord16 * const spectrum,
-                                    const WebRtc_UWord16 * const thresvec)
-{
-    int k;
-    WebRtc_UWord32 out;
-
-    out = 0;
-    for (k = BANDFIRST; k <= BANDLAST; k++)
-    {
-        if (spectrum[k] > thresvec[k])
-        {
-            out = WebRtcAecm_SetBit(out, k - BANDFIRST);
-        }
-    }
-
-    return out;
-}
-
-//   WebRtcAecm_MedianEstimator(...)
-//
-//   Calculates the median recursively.
-//
-//   Input:
-//           - newVal            :   new additional value
-//           - medianVec         :   vector with current medians
-//           - factor            :   factor for smoothing
-//
-//   Output:
-//           - medianVec         :   vector with updated median
-//
-int WebRtcAecm_MedianEstimator(const WebRtc_UWord16 newVal, WebRtc_UWord16 * const medianVec,
-                               const int factor)
-{
-    WebRtc_Word32 median;
-    WebRtc_Word32 diff;
-
-    median = (WebRtc_Word32)medianVec[0];
-
-    //median = median + ((newVal-median)>>factor);
-    diff = (WebRtc_Word32)newVal - median;
-    diff = WEBRTC_SPL_SHIFT_W32(diff, -factor);
-    median = median + diff;
-
-    medianVec[0] = (WebRtc_UWord16)median;
-
-    return 0;
-}
-
-int WebRtcAecm_CreateCore(AecmCore_t **aecmInst)
-{
-    AecmCore_t *aecm = malloc(sizeof(AecmCore_t));
-    *aecmInst = aecm;
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aecm->farFrameBuf, FRAME_LEN + PART_LEN) == -1)
-    {
-        WebRtcAecm_FreeCore(aecm);
-        aecm = NULL;
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aecm->nearNoisyFrameBuf, FRAME_LEN + PART_LEN) == -1)
-    {
-        WebRtcAecm_FreeCore(aecm);
-        aecm = NULL;
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aecm->nearCleanFrameBuf, FRAME_LEN + PART_LEN) == -1)
-    {
-        WebRtcAecm_FreeCore(aecm);
-        aecm = NULL;
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aecm->outFrameBuf, FRAME_LEN + PART_LEN) == -1)
-    {
-        WebRtcAecm_FreeCore(aecm);
-        aecm = NULL;
-        return -1;
-    }
-
-    return 0;
-}
-
-// WebRtcAecm_InitCore(...)
-//
-// This function initializes the AECM instant created with WebRtcAecm_CreateCore(...)
-// Input:
-//      - aecm            : Pointer to the Echo Suppression instance
-//      - samplingFreq   : Sampling Frequency
-//
-// Output:
-//      - aecm            : Initialized instance
-//
-// Return value         :  0 - Ok
-//                        -1 - Error
-//
-int WebRtcAecm_InitCore(AecmCore_t * const aecm, int samplingFreq)
-{
-    int retVal = 0;
-    WebRtc_Word16 i;
-    WebRtc_Word16 tmp16;
-
-    if (samplingFreq != 8000 && samplingFreq != 16000)
-    {
-        samplingFreq = 8000;
-        retVal = -1;
-    }
-    // sanity check of sampling frequency
-    aecm->mult = (WebRtc_Word16)samplingFreq / 8000;
-
-    aecm->farBufWritePos = 0;
-    aecm->farBufReadPos = 0;
-    aecm->knownDelay = 0;
-    aecm->lastKnownDelay = 0;
-
-    WebRtcApm_InitBuffer(aecm->farFrameBuf);
-    WebRtcApm_InitBuffer(aecm->nearNoisyFrameBuf);
-    WebRtcApm_InitBuffer(aecm->nearCleanFrameBuf);
-    WebRtcApm_InitBuffer(aecm->outFrameBuf);
-
-    memset(aecm->xBuf, 0, sizeof(aecm->xBuf));
-    memset(aecm->dBufClean, 0, sizeof(aecm->dBufClean));
-    memset(aecm->dBufNoisy, 0, sizeof(aecm->dBufNoisy));
-    memset(aecm->outBuf, 0, sizeof(WebRtc_Word16) * PART_LEN);
-
-    aecm->seed = 666;
-    aecm->totCount = 0;
-
-    memset(aecm->xfaHistory, 0, sizeof(WebRtc_UWord16) * (PART_LEN1) * MAX_DELAY);
-
-    aecm->delHistoryPos = MAX_DELAY;
-
-    memset(aecm->medianYlogspec, 0, sizeof(WebRtc_UWord16) * PART_LEN1);
-    memset(aecm->medianXlogspec, 0, sizeof(WebRtc_UWord16) * PART_LEN1);
-    memset(aecm->medianBCount, 0, sizeof(WebRtc_UWord16) * MAX_DELAY);
-    memset(aecm->bxHistory, 0, sizeof(aecm->bxHistory));
-
-    // Initialize to reasonable values
-    aecm->currentDelay = 8;
-    aecm->previousDelay = 8;
-    aecm->delayAdjust = 0;
-
-    aecm->nlpFlag = 1;
-    aecm->fixedDelay = -1;
-
-    memset(aecm->xfaQDomainBuf, 0, sizeof(WebRtc_Word16) * MAX_DELAY);
-    aecm->dfaCleanQDomain = 0;
-    aecm->dfaCleanQDomainOld = 0;
-    aecm->dfaNoisyQDomain = 0;
-    aecm->dfaNoisyQDomainOld = 0;
-
-    memset(aecm->nearLogEnergy, 0, sizeof(WebRtc_Word16) * MAX_BUF_LEN);
-    memset(aecm->farLogEnergy, 0, sizeof(WebRtc_Word16) * MAX_BUF_LEN);
-    memset(aecm->echoAdaptLogEnergy, 0, sizeof(WebRtc_Word16) * MAX_BUF_LEN);
-    memset(aecm->echoStoredLogEnergy, 0, sizeof(WebRtc_Word16) * MAX_BUF_LEN);
-
-    // Initialize the echo channels with a stored shape.
-    if (samplingFreq == 8000)
-    {
-        memcpy(aecm->channelAdapt16, kChannelStored8kHz, sizeof(WebRtc_Word16) * PART_LEN1);
-    }
-    else
-    {
-        memcpy(aecm->channelAdapt16, kChannelStored16kHz, sizeof(WebRtc_Word16) * PART_LEN1);
-    }
-    memcpy(aecm->channelStored, aecm->channelAdapt16, sizeof(WebRtc_Word16) * PART_LEN1);
-    for (i = 0; i < PART_LEN1; i++)
-    {
-        aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
-            (WebRtc_Word32)(aecm->channelAdapt16[i]), 16);
-    }
-
-    memset(aecm->echoFilt, 0, sizeof(WebRtc_Word32) * PART_LEN1);
-    memset(aecm->nearFilt, 0, sizeof(WebRtc_Word16) * PART_LEN1);
-    aecm->noiseEstCtr = 0;
-
-    aecm->cngMode = AecmTrue;
-
-    // Increase the noise Q domain with increasing frequency, to correspond to the
-    // expected energy levels.
-    // Also shape the initial noise level with this consideration.
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
-    for (i = 0; i < PART_LEN1; i++)
-    {
-        if (i < PART_LEN1 >> 2)
-        {
-            aecm->noiseEstQDomain[i] = 10;
-            tmp16 = PART_LEN1 - i;
-            aecm->noiseEst[i] = (tmp16 * tmp16) << 4;
-        } else if (i < PART_LEN1 >> 1)
-        {
-            aecm->noiseEstQDomain[i] = 11;
-            tmp16 = PART_LEN1 - i;
-            aecm->noiseEst[i] = ((tmp16 * tmp16) << 4) << 1;
-        } else
-        {
-            aecm->noiseEstQDomain[i] = 12;
-            aecm->noiseEst[i] = aecm->noiseEst[(PART_LEN1 >> 1) - 1] << 1;
-        }
-    }
-#else
-    for (i = 0; i < PART_LEN1 >> 2; i++)
-    {
-        aecm->noiseEstQDomain[i] = 10;
-        tmp16 = PART_LEN1 - i;
-        aecm->noiseEst[i] = (tmp16 * tmp16) << 4;
-    }
-    for (; i < PART_LEN1 >> 1; i++)
-    {
-        aecm->noiseEstQDomain[i] = 11;
-        tmp16 = PART_LEN1 - i;
-        aecm->noiseEst[i] = ((tmp16 * tmp16) << 4) << 1;
-    }
-    for (; i < PART_LEN1; i++)
-    {
-        aecm->noiseEstQDomain[i] = 12;
-        aecm->noiseEst[i] = aecm->noiseEst[(PART_LEN1 >> 1) - 1] << 1;
-    }
-#endif
-
-    aecm->mseAdaptOld = 1000;
-    aecm->mseStoredOld = 1000;
-    aecm->mseThreshold = WEBRTC_SPL_WORD32_MAX;
-
-    aecm->farEnergyMin = WEBRTC_SPL_WORD16_MAX;
-    aecm->farEnergyMax = WEBRTC_SPL_WORD16_MIN;
-    aecm->farEnergyMaxMin = 0;
-    aecm->farEnergyVAD = FAR_ENERGY_MIN; // This prevents false speech detection at the
-                                         // beginning.
-    aecm->farEnergyMSE = 0;
-    aecm->currentVADValue = 0;
-    aecm->vadUpdateCount = 0;
-    aecm->firstVAD = 1;
-
-    aecm->delayCount = 0;
-    aecm->newDelayCorrData = 0;
-    aecm->lastDelayUpdateCount = 0;
-    memset(aecm->delayCorrelation, 0, sizeof(WebRtc_Word16) * ((CORR_MAX << 1) + 1));
-
-    aecm->startupState = 0;
-    aecm->mseChannelCount = 0;
-    aecm->supGain = SUPGAIN_DEFAULT;
-    aecm->supGainOld = SUPGAIN_DEFAULT;
-    aecm->delayOffsetFlag = 0;
-
-    memset(aecm->delayHistogram, 0, sizeof(aecm->delayHistogram));
-    aecm->delayVadCount = 0;
-    aecm->maxDelayHistIdx = 0;
-    aecm->lastMinPos = 0;
-
-    aecm->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
-    aecm->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
-    aecm->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
-    aecm->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
-
-    return 0;
-}
-
-int WebRtcAecm_Control(AecmCore_t *aecm, int delay, int nlpFlag, int delayOffsetFlag)
-{
-    aecm->nlpFlag = nlpFlag;
-    aecm->fixedDelay = delay;
-    aecm->delayOffsetFlag = delayOffsetFlag;
-
-    return 0;
-}
-
-// WebRtcAecm_GetNewDelPos(...)
-//
-// Moves the pointer to the next entry. Returns to zero if max position reached.
-//
-// Input:
-//       - aecm     : Pointer to the AECM instance
-// Return:
-//       - pos      : New position in the history.
-//
-//
-WebRtc_Word16 WebRtcAecm_GetNewDelPos(AecmCore_t * const aecm)
-{
-    WebRtc_Word16 pos;
-
-    pos = aecm->delHistoryPos;
-    pos++;
-    if (pos >= MAX_DELAY)
-    {
-        pos = 0;
-    }
-    aecm->delHistoryPos = pos;
-
-    return pos;
-}
-
-// WebRtcAecm_EstimateDelay(...)
-//
-// Estimate the delay of the echo signal.
-//
-// Inputs:
-//      - aecm          : Pointer to the AECM instance
-//      - farSpec       : Delayed farend magnitude spectrum
-//      - nearSpec      : Nearend magnitude spectrum
-//      - stages        : Q-domain of xxFIX and yyFIX (without dynamic Q-domain)
-//      - xfaQ          : normalization factor, i.e., Q-domain before FFT
-// Return:
-//      - delay         : Estimated delay
-//
-WebRtc_Word16 WebRtcAecm_EstimateDelay(AecmCore_t * const aecm,
-                                       const WebRtc_UWord16 * const farSpec,
-                                       const WebRtc_UWord16 * const nearSpec,
-                                       const WebRtc_Word16 xfaQ)
-{
-    WebRtc_UWord32 bxspectrum, byspectrum;
-    WebRtc_UWord32 bcount[MAX_DELAY];
-
-    int i, res;
-
-    WebRtc_UWord16 xmean[PART_LEN1], ymean[PART_LEN1];
-    WebRtc_UWord16 dtmp1;
-    WebRtc_Word16 fcount[MAX_DELAY];
-
-    //WebRtc_Word16 res;
-    WebRtc_Word16 histpos;
-    WebRtc_Word16 maxHistLvl;
-    WebRtc_UWord16 *state;
-    WebRtc_Word16 minpos = -1;
-
-    enum
-    {
-        kVadCountThreshold = 25
-    };
-    enum
-    {
-        kMaxHistogram = 600
-    };
-
-    histpos = WebRtcAecm_GetNewDelPos(aecm);
-
-    for (i = 0; i < PART_LEN1; i++)
-    {
-        aecm->xfaHistory[i][histpos] = farSpec[i];
-
-        state = &(aecm->medianXlogspec[i]);
-        res = WebRtcAecm_MedianEstimator(farSpec[i], state, 6);
-
-        state = &(aecm->medianYlogspec[i]);
-        res = WebRtcAecm_MedianEstimator(nearSpec[i], state, 6);
-
-        //  Mean:
-        //  FLOAT:
-        //  ymean = dtmp2/MAX_DELAY
-        //
-        //  FIX:
-        //  input: dtmp2FIX in Q0
-        //  output: ymeanFIX in Q8
-        //  20 = 1/MAX_DELAY in Q13 = 1/MAX_DELAY * 2^13
-        xmean[i] = (aecm->medianXlogspec[i]);
-        ymean[i] = (aecm->medianYlogspec[i]);
-
-    }
-    // Update Q-domain buffer
-    aecm->xfaQDomainBuf[histpos] = xfaQ;
-
-    // Get binary spectra
-    //  FLOAT:
-    //  bxspectrum = bspectrum(xlogspec, xmean);
-    //
-    //  FIX:
-    //  input:  xlogspecFIX,ylogspecFIX in Q8
-    //          xmeanFIX, ymeanFIX in Q8
-    //  output: unsigned long bxspectrum, byspectrum in Q0
-    bxspectrum = WebRtcAecm_BSpectrum(farSpec, xmean);
-    byspectrum = WebRtcAecm_BSpectrum(nearSpec, ymean);
-
-    // Shift binary spectrum history
-    memmove(&(aecm->bxHistory[1]), &(aecm->bxHistory[0]),
-            (MAX_DELAY - 1) * sizeof(WebRtc_UWord32));
-
-    aecm->bxHistory[0] = bxspectrum;
-
-    // Compare with delayed spectra
-    WebRtcAecm_Hisser(byspectrum, aecm->bxHistory, bcount);
-
-    for (i = 0; i < MAX_DELAY; i++)
-    {
-        // Update sum
-        // bcount is constrained to [0, 32], meaning we can smooth with a factor up to 2^11.
-        dtmp1 = (WebRtc_UWord16)bcount[i];
-        dtmp1 = WEBRTC_SPL_LSHIFT_W16(dtmp1, 9);
-        state = &(aecm->medianBCount[i]);
-        res = WebRtcAecm_MedianEstimator(dtmp1, state, 9);
-        fcount[i] = (aecm->medianBCount[i]);
-    }
-
-    // Find minimum
-    minpos = WebRtcSpl_MinIndexW16(fcount, MAX_DELAY);
-
-    // If the farend has been active sufficiently long, begin accumulating a histogram
-    // of the minimum positions. Search for the maximum bin to determine the delay.
-    if (aecm->currentVADValue == 1)
-    {
-        if (aecm->delayVadCount >= kVadCountThreshold)
-        {
-            // Increment the histogram at the current minimum position.
-            if (aecm->delayHistogram[minpos] < kMaxHistogram)
-            {
-                aecm->delayHistogram[minpos] += 3;
-            }
-
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
-            // Decrement the entire histogram.
-            for (i = 0; i < MAX_DELAY; i++)
-            {
-                if (aecm->delayHistogram[i] > 0)
-                {
-                    aecm->delayHistogram[i]--;
-                }
-            }
-
-            // Select the histogram index corresponding to the maximum bin as the delay.
-            maxHistLvl = 0;
-            aecm->maxDelayHistIdx = 0;
-            for (i = 0; i < MAX_DELAY; i++)
-            {
-                if (aecm->delayHistogram[i] > maxHistLvl)
-                {
-                    maxHistLvl = aecm->delayHistogram[i];
-                    aecm->maxDelayHistIdx = i;
-                }
-            }
-#else
-            maxHistLvl = 0;
-            aecm->maxDelayHistIdx = 0;
-
-            for (i = 0; i < MAX_DELAY; i++)
-            {
-                WebRtc_Word16 tempVar = aecm->delayHistogram[i];
-
-                // Decrement the entire histogram.
-                if (tempVar > 0)
-                {
-                    tempVar--;
-                    aecm->delayHistogram[i] = tempVar;
-
-                    // Select the histogram index corresponding to the maximum bin as the delay.
-                    if (tempVar > maxHistLvl)
-                    {
-                        maxHistLvl = tempVar;
-                        aecm->maxDelayHistIdx = i;
-                    }
-                }
-            }
-#endif
-        } else
-        {
-            aecm->delayVadCount++;
-        }
-    } else
-    {
-        aecm->delayVadCount = 0;
-    }
-
-    return aecm->maxDelayHistIdx;
-}
-
-int WebRtcAecm_FreeCore(AecmCore_t *aecm)
-{
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    WebRtcApm_FreeBuffer(aecm->farFrameBuf);
-    WebRtcApm_FreeBuffer(aecm->nearNoisyFrameBuf);
-    WebRtcApm_FreeBuffer(aecm->nearCleanFrameBuf);
-    WebRtcApm_FreeBuffer(aecm->outFrameBuf);
-
-    free(aecm);
-
-    return 0;
-}
-
-void WebRtcAecm_ProcessFrame(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
-                             const WebRtc_Word16 * const nearendNoisy,
-                             const WebRtc_Word16 * const nearendClean,
-                             WebRtc_Word16 * const out)
-{
-    WebRtc_Word16 farBlock[PART_LEN];
-    WebRtc_Word16 nearNoisyBlock[PART_LEN];
-    WebRtc_Word16 nearCleanBlock[PART_LEN];
-    WebRtc_Word16 outBlock[PART_LEN];
-    WebRtc_Word16 farFrame[FRAME_LEN];
-    int size = 0;
-
-    // Buffer the current frame.
-    // Fetch an older one corresponding to the delay.
-    WebRtcAecm_BufferFarFrame(aecm, farend, FRAME_LEN);
-    WebRtcAecm_FetchFarFrame(aecm, farFrame, FRAME_LEN, aecm->knownDelay);
-
-    // Buffer the synchronized far and near frames,
-    // to pass the smaller blocks individually.
-    WebRtcApm_WriteBuffer(aecm->farFrameBuf, farFrame, FRAME_LEN);
-    WebRtcApm_WriteBuffer(aecm->nearNoisyFrameBuf, nearendNoisy, FRAME_LEN);
-    if (nearendClean != NULL)
-    {
-        WebRtcApm_WriteBuffer(aecm->nearCleanFrameBuf, nearendClean, FRAME_LEN);
-    }
-
-    // Process as many blocks as possible.
-    while (WebRtcApm_get_buffer_size(aecm->farFrameBuf) >= PART_LEN)
-    {
-        WebRtcApm_ReadBuffer(aecm->farFrameBuf, farBlock, PART_LEN);
-        WebRtcApm_ReadBuffer(aecm->nearNoisyFrameBuf, nearNoisyBlock, PART_LEN);
-        if (nearendClean != NULL)
-        {
-            WebRtcApm_ReadBuffer(aecm->nearCleanFrameBuf, nearCleanBlock, PART_LEN);
-            WebRtcAecm_ProcessBlock(aecm, farBlock, nearNoisyBlock, nearCleanBlock, outBlock);
-        } else
-        {
-            WebRtcAecm_ProcessBlock(aecm, farBlock, nearNoisyBlock, NULL, outBlock);
-        }
-
-        WebRtcApm_WriteBuffer(aecm->outFrameBuf, outBlock, PART_LEN);
-    }
-
-    // Stuff the out buffer if we have less than a frame to output.
-    // This should only happen for the first frame.
-    size = WebRtcApm_get_buffer_size(aecm->outFrameBuf);
-    if (size < FRAME_LEN)
-    {
-        WebRtcApm_StuffBuffer(aecm->outFrameBuf, FRAME_LEN - size);
-    }
-
-    // Obtain an output frame.
-    WebRtcApm_ReadBuffer(aecm->outFrameBuf, out, FRAME_LEN);
-}
-
-// WebRtcAecm_AsymFilt(...)
-//
-// Performs asymmetric filtering.
-//
-// Inputs:
-//      - filtOld       : Previous filtered value.
-//      - inVal         : New input value.
-//      - stepSizePos   : Step size when we have a positive contribution.
-//      - stepSizeNeg   : Step size when we have a negative contribution.
-//
-// Output:
-//
-// Return: - Filtered value.
-//
-WebRtc_Word16 WebRtcAecm_AsymFilt(const WebRtc_Word16 filtOld, const WebRtc_Word16 inVal,
-                                  const WebRtc_Word16 stepSizePos,
-                                  const WebRtc_Word16 stepSizeNeg)
-{
-    WebRtc_Word16 retVal;
-
-    if ((filtOld == WEBRTC_SPL_WORD16_MAX) | (filtOld == WEBRTC_SPL_WORD16_MIN))
-    {
-        return inVal;
-    }
-    retVal = filtOld;
-    if (filtOld > inVal)
-    {
-        retVal -= WEBRTC_SPL_RSHIFT_W16(filtOld - inVal, stepSizeNeg);
-    } else
-    {
-        retVal += WEBRTC_SPL_RSHIFT_W16(inVal - filtOld, stepSizePos);
-    }
-
-    return retVal;
-}
-
-// WebRtcAecm_CalcEnergies(...)
-//
-// This function calculates the log of energies for nearend, farend and estimated
-// echoes. There is also an update of energy decision levels, i.e. internl VAD.
-//
-//
-// @param  aecm         [i/o]   Handle of the AECM instance.
-// @param  delayDiff    [in]    Delay position in farend buffer.
-// @param  nearEner     [in]    Near end energy for current block (Q[aecm->dfaQDomain]).
-// @param  echoEst      [i/o]   Estimated echo
-//                              (Q[aecm->xfaQDomain[delayDiff]+RESOLUTION_CHANNEL16]).
-//
-void WebRtcAecm_CalcEnergies(AecmCore_t * const aecm, const WebRtc_Word16 delayDiff,
-                             const WebRtc_UWord32 nearEner, WebRtc_Word32 * const echoEst)
-{
-    // Local variables
-    WebRtc_UWord32 tmpAdapt, tmpStored, tmpFar;
-
-    int i;
-
-    WebRtc_Word16 zeros, frac;
-    WebRtc_Word16 tmp16;
-    WebRtc_Word16 increase_max_shifts = 4;
-    WebRtc_Word16 decrease_max_shifts = 11;
-    WebRtc_Word16 increase_min_shifts = 11;
-    WebRtc_Word16 decrease_min_shifts = 3;
-
-    // Get log of near end energy and store in buffer
-
-    // Shift buffer
-    memmove(aecm->nearLogEnergy + 1, aecm->nearLogEnergy,
-            sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
-
-    // Logarithm of integrated magnitude spectrum (nearEner)
-    if (nearEner)
-    {
-        zeros = WebRtcSpl_NormU32(nearEner);
-        frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(
-                              (WEBRTC_SPL_LSHIFT_U32(nearEner, zeros) & 0x7FFFFFFF),
-                              23);
-        // log2 in Q8
-        aecm->nearLogEnergy[0] = WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
-        aecm->nearLogEnergy[0] -= WEBRTC_SPL_LSHIFT_W16(aecm->dfaNoisyQDomain, 8);
-    } else
-    {
-        aecm->nearLogEnergy[0] = 0;
-    }
-    aecm->nearLogEnergy[0] += WEBRTC_SPL_LSHIFT_W16(PART_LEN_SHIFT, 7);
-    // END: Get log of near end energy
-
-    // Get energy for the delayed far end signal and estimated
-    // echo using both stored and adapted channels.
-    tmpAdapt = 0;
-    tmpStored = 0;
-    tmpFar = 0;
-
-    for (i = 0; i < PART_LEN1; i++)
-    {
-        // Get estimated echo energies for adaptive channel and stored channel
-        echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
-                aecm->xfaHistory[i][delayDiff]);
-        tmpFar += (WebRtc_UWord32)(aecm->xfaHistory[i][delayDiff]);
-        tmpAdapt += WEBRTC_SPL_UMUL_16_16(aecm->channelAdapt16[i],
-                aecm->xfaHistory[i][delayDiff]);
-        tmpStored += (WebRtc_UWord32)echoEst[i];
-    }
-    // Shift buffers
-    memmove(aecm->farLogEnergy + 1, aecm->farLogEnergy,
-            sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
-    memmove(aecm->echoAdaptLogEnergy + 1, aecm->echoAdaptLogEnergy,
-            sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
-    memmove(aecm->echoStoredLogEnergy + 1, aecm->echoStoredLogEnergy,
-            sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
-
-    // Logarithm of delayed far end energy
-    if (tmpFar)
-    {
-        zeros = WebRtcSpl_NormU32(tmpFar);
-        frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpFar, zeros)
-                        & 0x7FFFFFFF), 23);
-        // log2 in Q8
-        aecm->farLogEnergy[0] = WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
-        aecm->farLogEnergy[0] -= WEBRTC_SPL_LSHIFT_W16(aecm->xfaQDomainBuf[delayDiff], 8);
-    } else
-    {
-        aecm->farLogEnergy[0] = 0;
-    }
-    aecm->farLogEnergy[0] += WEBRTC_SPL_LSHIFT_W16(PART_LEN_SHIFT, 7);
-
-    // Logarithm of estimated echo energy through adapted channel
-    if (tmpAdapt)
-    {
-        zeros = WebRtcSpl_NormU32(tmpAdapt);
-        frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpAdapt, zeros)
-                        & 0x7FFFFFFF), 23);
-        //log2 in Q8
-        aecm->echoAdaptLogEnergy[0] = WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
-        aecm->echoAdaptLogEnergy[0]
-                -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + aecm->xfaQDomainBuf[delayDiff], 8);
-    } else
-    {
-        aecm->echoAdaptLogEnergy[0] = 0;
-    }
-    aecm->echoAdaptLogEnergy[0] += WEBRTC_SPL_LSHIFT_W16(PART_LEN_SHIFT, 7);
-
-    // Logarithm of estimated echo energy through stored channel
-    if (tmpStored)
-    {
-        zeros = WebRtcSpl_NormU32(tmpStored);
-        frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpStored, zeros)
-                        & 0x7FFFFFFF), 23);
-        //log2 in Q8
-        aecm->echoStoredLogEnergy[0] = WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
-        aecm->echoStoredLogEnergy[0]
-                -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + aecm->xfaQDomainBuf[delayDiff], 8);
-    } else
-    {
-        aecm->echoStoredLogEnergy[0] = 0;
-    }
-    aecm->echoStoredLogEnergy[0] += WEBRTC_SPL_LSHIFT_W16(PART_LEN_SHIFT, 7);
-
-    // Update farend energy levels (min, max, vad, mse)
-    if (aecm->farLogEnergy[0] > FAR_ENERGY_MIN)
-    {
-        if (aecm->startupState == 0)
-        {
-            increase_max_shifts = 2;
-            decrease_min_shifts = 2;
-            increase_min_shifts = 8;
-        }
-
-        aecm->farEnergyMin = WebRtcAecm_AsymFilt(aecm->farEnergyMin, aecm->farLogEnergy[0],
-                                                 increase_min_shifts, decrease_min_shifts);
-        aecm->farEnergyMax = WebRtcAecm_AsymFilt(aecm->farEnergyMax, aecm->farLogEnergy[0],
-                                                 increase_max_shifts, decrease_max_shifts);
-        aecm->farEnergyMaxMin = (aecm->farEnergyMax - aecm->farEnergyMin);
-
-        // Dynamic VAD region size
-        tmp16 = 2560 - aecm->farEnergyMin;
-        if (tmp16 > 0)
-        {
-            tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, FAR_ENERGY_VAD_REGION, 9);
-        } else
-        {
-            tmp16 = 0;
-        }
-        tmp16 += FAR_ENERGY_VAD_REGION;
-
-        if ((aecm->startupState == 0) | (aecm->vadUpdateCount > 1024))
-        {
-            // In startup phase or VAD update halted
-            aecm->farEnergyVAD = aecm->farEnergyMin + tmp16;
-        } else
-        {
-            if (aecm->farEnergyVAD > aecm->farLogEnergy[0])
-            {
-                aecm->farEnergyVAD += WEBRTC_SPL_RSHIFT_W16(aecm->farLogEnergy[0] + tmp16
-                        - aecm->farEnergyVAD, 6);
-                aecm->vadUpdateCount = 0;
-            } else
-            {
-                aecm->vadUpdateCount++;
-            }
-        }
-        // Put MSE threshold higher than VAD
-        aecm->farEnergyMSE = aecm->farEnergyVAD + (1 << 8);
-    }
-
-    // Update VAD variables
-    if (aecm->farLogEnergy[0] > aecm->farEnergyVAD)
-    {
-        if ((aecm->startupState == 0) | (aecm->farEnergyMaxMin > FAR_ENERGY_DIFF))
-        {
-            // We are in startup or have significant dynamics in input speech level
-            aecm->currentVADValue = 1;
-        }
-    } else
-    {
-        aecm->currentVADValue = 0;
-    }
-    if ((aecm->currentVADValue) && (aecm->firstVAD))
-    {
-        aecm->firstVAD = 0;
-        if (aecm->echoAdaptLogEnergy[0] > aecm->nearLogEnergy[0])
-        {
-            // The estimated echo has higher energy than the near end signal. This means that
-            // the initialization was too aggressive. Scale down by a factor 8
-            for (i = 0; i < PART_LEN1; i++)
-            {
-                aecm->channelAdapt16[i] >>= 3;
-            }
-            // Compensate the adapted echo energy level accordingly.
-            aecm->echoAdaptLogEnergy[0] -= (3 << 8);
-            aecm->firstVAD = 1;
-        }
-    }
-    // END: Energies of delayed far, echo estimates
-    // TODO(bjornv): Will be removed in final version.
-#ifdef VAD_DATA
-    fwrite(&(aecm->currentVADValue), sizeof(WebRtc_Word16), 1, aecm->vad_file);
-    fwrite(&(aecm->currentDelay), sizeof(WebRtc_Word16), 1, aecm->delay_file);
-    fwrite(&(aecm->farLogEnergy[0]), sizeof(WebRtc_Word16), 1, aecm->far_cur_file);
-    fwrite(&(aecm->farEnergyMin), sizeof(WebRtc_Word16), 1, aecm->far_min_file);
-    fwrite(&(aecm->farEnergyMax), sizeof(WebRtc_Word16), 1, aecm->far_max_file);
-    fwrite(&(aecm->farEnergyVAD), sizeof(WebRtc_Word16), 1, aecm->far_vad_file);
-#endif
-}
-
-// WebRtcAecm_CalcStepSize(...)
-//
-// This function calculates the step size used in channel estimation
-//
-//
-// @param  aecm  [in]    Handle of the AECM instance.
-// @param  mu   [out]   (Return value) Stepsize in log2(), i.e. number of shifts.
-//
-//
-WebRtc_Word16 WebRtcAecm_CalcStepSize(AecmCore_t * const aecm)
-{
-
-    WebRtc_Word32 tmp32;
-    WebRtc_Word16 tmp16;
-    WebRtc_Word16 mu;
-
-    // Here we calculate the step size mu used in the
-    // following NLMS based Channel estimation algorithm
-    mu = MU_MAX;
-    if (!aecm->currentVADValue)
-    {
-        // Far end energy level too low, no channel update
-        mu = 0;
-    } else if (aecm->startupState > 0)
-    {
-        if (aecm->farEnergyMin >= aecm->farEnergyMax)
-        {
-            mu = MU_MIN;
-        } else
-        {
-            tmp16 = (aecm->farLogEnergy[0] - aecm->farEnergyMin);
-            tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, MU_DIFF);
-            tmp32 = WebRtcSpl_DivW32W16(tmp32, aecm->farEnergyMaxMin);
-            mu = MU_MIN - 1 - (WebRtc_Word16)(tmp32);
-            // The -1 is an alternative to rounding. This way we get a larger
-            // stepsize, so we in some sense compensate for truncation in NLMS
-        }
-        if (mu < MU_MAX)
-        {
-            mu = MU_MAX; // Equivalent with maximum step size of 2^-MU_MAX
-        }
-    }
-    // END: Update step size
-
-    return mu;
-}
-
-// WebRtcAecm_UpdateChannel(...)
-//
-// This function performs channel estimation. NLMS and decision on channel storage.
-//
-//
-// @param  aecm         [i/o]   Handle of the AECM instance.
-// @param  dfa          [in]    Absolute value of the nearend signal (Q[aecm->dfaQDomain])
-// @param  delayDiff    [in]    Delay position in farend buffer.
-// @param  mu           [in]    NLMS step size.
-// @param  echoEst      [i/o]   Estimated echo
-//                              (Q[aecm->xfaQDomain[delayDiff]+RESOLUTION_CHANNEL16]).
-//
-void WebRtcAecm_UpdateChannel(AecmCore_t * const aecm, const WebRtc_UWord16 * const dfa,
-                              const WebRtc_Word16 delayDiff, const WebRtc_Word16 mu,
-                              WebRtc_Word32 * const echoEst)
-{
-
-    WebRtc_UWord32 tmpU32no1, tmpU32no2;
-    WebRtc_Word32 tmp32no1, tmp32no2;
-    WebRtc_Word32 mseStored;
-    WebRtc_Word32 mseAdapt;
-
-    int i;
-
-    WebRtc_Word16 zerosFar, zerosNum, zerosCh, zerosDfa;
-    WebRtc_Word16 shiftChFar, shiftNum, shift2ResChan;
-    WebRtc_Word16 tmp16no1;
-    WebRtc_Word16 xfaQ, dfaQ;
-
-    // This is the channel estimation algorithm. It is base on NLMS but has a variable step
-    // length, which was calculated above.
-    if (mu)
-    {
-        for (i = 0; i < PART_LEN1; i++)
-        {
-            // Determine norm of channel and farend to make sure we don't get overflow in
-            // multiplication
-            zerosCh = WebRtcSpl_NormU32(aecm->channelAdapt32[i]);
-            zerosFar = WebRtcSpl_NormU32((WebRtc_UWord32)aecm->xfaHistory[i][delayDiff]);
-            if (zerosCh + zerosFar > 31)
-            {
-                // Multiplication is safe
-                tmpU32no1 = WEBRTC_SPL_UMUL_32_16(aecm->channelAdapt32[i],
-                        aecm->xfaHistory[i][delayDiff]);
-                shiftChFar = 0;
-            } else
-            {
-                // We need to shift down before multiplication
-                shiftChFar = 32 - zerosCh - zerosFar;
-                tmpU32no1
-                        = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_W32(aecm->channelAdapt32[i],
-                                        shiftChFar),
-                                aecm->xfaHistory[i][delayDiff]);
-            }
-            // Determine Q-domain of numerator
-            zerosNum = WebRtcSpl_NormU32(tmpU32no1);
-            if (dfa[i])
-            {
-                zerosDfa = WebRtcSpl_NormU32((WebRtc_UWord32)dfa[i]);
-            } else
-            {
-                zerosDfa = 32;
-            }
-            tmp16no1 = zerosDfa - 2 + aecm->dfaNoisyQDomain - RESOLUTION_CHANNEL32
-                    - aecm->xfaQDomainBuf[delayDiff] + shiftChFar;
-            if (zerosNum > tmp16no1 + 1)
-            {
-                xfaQ = tmp16no1;
-                dfaQ = zerosDfa - 2;
-            } else
-            {
-                xfaQ = zerosNum - 2;
-                dfaQ = RESOLUTION_CHANNEL32 + aecm->xfaQDomainBuf[delayDiff]
-                        - aecm->dfaNoisyQDomain - shiftChFar + xfaQ;
-            }
-            // Add in the same Q-domain
-            tmpU32no1 = WEBRTC_SPL_SHIFT_W32(tmpU32no1, xfaQ);
-            tmpU32no2 = WEBRTC_SPL_SHIFT_W32((WebRtc_UWord32)dfa[i], dfaQ);
-            tmp32no1 = (WebRtc_Word32)tmpU32no2 - (WebRtc_Word32)tmpU32no1;
-            zerosNum = WebRtcSpl_NormW32(tmp32no1);
-            if ((tmp32no1) && (aecm->xfaHistory[i][delayDiff] > (CHANNEL_VAD
-                    << aecm->xfaQDomainBuf[delayDiff])))
-            {
-                //
-                // Update is needed
-                //
-                // This is what we would like to compute
-                //
-                // tmp32no1 = dfa[i] - (aecm->channelAdapt[i] * aecm->xfaHistory[i][delayDiff])
-                // tmp32norm = (i + 1)
-                // aecm->channelAdapt[i] += (2^mu) * tmp32no1
-                //                        / (tmp32norm * aecm->xfaHistory[i][delayDiff])
-                //
-
-                // Make sure we don't get overflow in multiplication.
-                if (zerosNum + zerosFar > 31)
-                {
-                    if (tmp32no1 > 0)
-                    {
-                        tmp32no2 = (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(tmp32no1,
-                                aecm->xfaHistory[i][delayDiff]);
-                    } else
-                    {
-                        tmp32no2 = -(WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(-tmp32no1,
-                                aecm->xfaHistory[i][delayDiff]);
-                    }
-                    shiftNum = 0;
-                } else
-                {
-                    shiftNum = 32 - (zerosNum + zerosFar);
-                    if (tmp32no1 > 0)
-                    {
-                        tmp32no2 = (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(
-                                WEBRTC_SPL_RSHIFT_W32(tmp32no1, shiftNum),
-                                aecm->xfaHistory[i][delayDiff]);
-                    } else
-                    {
-                        tmp32no2 = -(WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(
-                                WEBRTC_SPL_RSHIFT_W32(-tmp32no1, shiftNum),
-                                aecm->xfaHistory[i][delayDiff]);
-                    }
-                }
-                // Normalize with respect to frequency bin
-                tmp32no2 = WebRtcSpl_DivW32W16(tmp32no2, i + 1);
-                // Make sure we are in the right Q-domain
-                shift2ResChan = shiftNum + shiftChFar - xfaQ - mu - ((30 - zerosFar) << 1);
-                if (WebRtcSpl_NormW32(tmp32no2) < shift2ResChan)
-                {
-                    tmp32no2 = WEBRTC_SPL_WORD32_MAX;
-                } else
-                {
-                    tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, shift2ResChan);
-                }
-                aecm->channelAdapt32[i] = WEBRTC_SPL_ADD_SAT_W32(aecm->channelAdapt32[i],
-                        tmp32no2);
-                if (aecm->channelAdapt32[i] < 0)
-                {
-                    // We can never have negative channel gain
-                    aecm->channelAdapt32[i] = 0;
-                }
-                aecm->channelAdapt16[i]
-                        = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(aecm->channelAdapt32[i], 16);
-            }
-        }
-    }
-    // END: Adaptive channel update
-
-    // Determine if we should store or restore the channel
-    if ((aecm->startupState == 0) & (aecm->currentVADValue))
-    {
-        // During startup we store the channel every block.
-        memcpy(aecm->channelStored, aecm->channelAdapt16, sizeof(WebRtc_Word16) * PART_LEN1);
-        // TODO(bjornv): Will be removed in final version.
-#ifdef STORE_CHANNEL_DATA
-        fwrite(aecm->channelStored, sizeof(WebRtc_Word16), PART_LEN1, aecm->channel_file_init);
-#endif
-        // Recalculate echo estimate
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
-        for (i = 0; i < PART_LEN1; i++)
-        {
-            echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
-                    aecm->xfaHistory[i][delayDiff]);
-        }
-#else
-        for (i = 0; i < PART_LEN; ) //assume PART_LEN is 4's multiples
-
-        {
-            echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
-                    aecm->xfaHistory[i][delayDiff]);
-            i++;
-            echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
-                    aecm->xfaHistory[i][delayDiff]);
-            i++;
-            echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
-                    aecm->xfaHistory[i][delayDiff]);
-            i++;
-            echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
-                    aecm->xfaHistory[i][delayDiff]);
-            i++;
-        }
-        echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
-                aecm->xfaHistory[i][delayDiff]);
-#endif
-    } else
-    {
-        if (aecm->farLogEnergy[0] < aecm->farEnergyMSE)
-        {
-            aecm->mseChannelCount = 0;
-            aecm->delayCount = 0;
-        } else
-        {
-            aecm->mseChannelCount++;
-            aecm->delayCount++;
-        }
-        // Enough data for validation. Store channel if we can.
-        if (aecm->mseChannelCount >= (MIN_MSE_COUNT + 10))
-        {
-            // We have enough data.
-            // Calculate MSE of "Adapt" and "Stored" versions.
-            // It is actually not MSE, but average absolute error.
-            mseStored = 0;
-            mseAdapt = 0;
-            for (i = 0; i < MIN_MSE_COUNT; i++)
-            {
-                tmp32no1 = ((WebRtc_Word32)aecm->echoStoredLogEnergy[i]
-                        - (WebRtc_Word32)aecm->nearLogEnergy[i]);
-                tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
-                mseStored += tmp32no2;
-
-                tmp32no1 = ((WebRtc_Word32)aecm->echoAdaptLogEnergy[i]
-                        - (WebRtc_Word32)aecm->nearLogEnergy[i]);
-                tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
-                mseAdapt += tmp32no2;
-            }
-            if (((mseStored << MSE_RESOLUTION) < (MIN_MSE_DIFF * mseAdapt))
-                    & ((aecm->mseStoredOld << MSE_RESOLUTION) < (MIN_MSE_DIFF
-                            * aecm->mseAdaptOld)))
-            {
-                // The stored channel has a significantly lower MSE than the adaptive one for
-                // two consecutive calculations. Reset the adaptive channel.
-                memcpy(aecm->channelAdapt16, aecm->channelStored,
-                       sizeof(WebRtc_Word16) * PART_LEN1);
-                // Restore the W32 channel
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
-                for (i = 0; i < PART_LEN1; i++)
-                {
-                    aecm->channelAdapt32[i]
-                            = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
-                }
-#else
-                for (i = 0; i < PART_LEN; ) //assume PART_LEN is 4's multiples
-
-                {
-                    aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
-                    i++;
-                    aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
-                    i++;
-                    aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
-                    i++;
-                    aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
-                    i++;
-                }
-                aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
-#endif
-
-            } else if (((MIN_MSE_DIFF * mseStored) > (mseAdapt << MSE_RESOLUTION)) & (mseAdapt
-                    < aecm->mseThreshold) & (aecm->mseAdaptOld < aecm->mseThreshold))
-            {
-                // The adaptive channel has a significantly lower MSE than the stored one.
-                // The MSE for the adaptive channel has also been low for two consecutive
-                // calculations. Store the adaptive channel.
-                memcpy(aecm->channelStored, aecm->channelAdapt16,
-                       sizeof(WebRtc_Word16) * PART_LEN1);
-                // TODO(bjornv): Will be removed in final version.
-#ifdef STORE_CHANNEL_DATA
-                fwrite(aecm->channelStored, sizeof(WebRtc_Word16), PART_LEN1,
-                       aecm->channel_file);
-#endif
-// Recalculate echo estimate
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
-                for (i = 0; i < PART_LEN1; i++)
-                {
-                    echoEst[i]
-                            = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
-                }
-#else
-                for (i = 0; i < PART_LEN; ) //assume PART_LEN is 4's multiples
-
-                {
-                    echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
-                    i++;
-                    echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
-                    i++;
-                    echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
-                    i++;
-                    echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
-                    i++;
-                }
-                echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], aecm->xfaHistory[i][delayDiff]);
-#endif
-                // Update threshold
-                if (aecm->mseThreshold == WEBRTC_SPL_WORD32_MAX)
-                {
-                    aecm->mseThreshold = (mseAdapt + aecm->mseAdaptOld);
-                } else
-                {
-                    aecm->mseThreshold += WEBRTC_SPL_MUL_16_16_RSFT(mseAdapt
-                            - WEBRTC_SPL_MUL_16_16_RSFT(aecm->mseThreshold, 5, 3), 205, 8);
-                }
-
-            }
-
-            // Reset counter
-            aecm->mseChannelCount = 0;
-
-            // Store the MSE values.
-            aecm->mseStoredOld = mseStored;
-            aecm->mseAdaptOld = mseAdapt;
-        }
-    }
-    // END: Determine if we should store or reset channel estimate.
-}
-
-// WebRtcAecm_CalcSuppressionGain(...)
-//
-// This function calculates the suppression gain that is used in the Wiener filter.
-//
-//
-// @param  aecm     [i/n]   Handle of the AECM instance.
-// @param  supGain  [out]   (Return value) Suppression gain with which to scale the noise
-//                          level (Q14).
-//
-//
-WebRtc_Word16 WebRtcAecm_CalcSuppressionGain(AecmCore_t * const aecm)
-{
-    WebRtc_Word32 tmp32no1;
-
-    WebRtc_Word16 supGain;
-    WebRtc_Word16 tmp16no1;
-    WebRtc_Word16 dE = 0;
-
-    // Determine suppression gain used in the Wiener filter. The gain is based on a mix of far
-    // end energy and echo estimation error.
-    supGain = SUPGAIN_DEFAULT;
-    // Adjust for the far end signal level. A low signal level indicates no far end signal,
-    // hence we set the suppression gain to 0
-    if (!aecm->currentVADValue)
-    {
-        supGain = 0;
-    } else
-    {
-        // Adjust for possible double talk. If we have large variations in estimation error we
-        // likely have double talk (or poor channel).
-        tmp16no1 = (aecm->nearLogEnergy[0] - aecm->echoStoredLogEnergy[0] - ENERGY_DEV_OFFSET);
-        dE = WEBRTC_SPL_ABS_W16(tmp16no1);
-
-        if (dE < ENERGY_DEV_TOL)
-        {
-            // Likely no double talk. The better estimation, the more we can suppress signal.
-            // Update counters
-            if (dE < SUPGAIN_EPC_DT)
-            {
-                tmp32no1 = WEBRTC_SPL_MUL_16_16(aecm->supGainErrParamDiffAB, dE);
-                tmp32no1 += (SUPGAIN_EPC_DT >> 1);
-                tmp16no1 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32no1, SUPGAIN_EPC_DT);
-                supGain = aecm->supGainErrParamA - tmp16no1;
-            } else
-            {
-                tmp32no1 = WEBRTC_SPL_MUL_16_16(aecm->supGainErrParamDiffBD,
-                                                (ENERGY_DEV_TOL - dE));
-                tmp32no1 += ((ENERGY_DEV_TOL - SUPGAIN_EPC_DT) >> 1);
-                tmp16no1 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32no1, (ENERGY_DEV_TOL
-                        - SUPGAIN_EPC_DT));
-                supGain = aecm->supGainErrParamD + tmp16no1;
-            }
-        } else
-        {
-            // Likely in double talk. Use default value
-            supGain = aecm->supGainErrParamD;
-        }
-    }
-
-    if (supGain > aecm->supGainOld)
-    {
-        tmp16no1 = supGain;
-    } else
-    {
-        tmp16no1 = aecm->supGainOld;
-    }
-    aecm->supGainOld = supGain;
-    if (tmp16no1 < aecm->supGain)
-    {
-        aecm->supGain += (WebRtc_Word16)((tmp16no1 - aecm->supGain) >> 4);
-    } else
-    {
-        aecm->supGain += (WebRtc_Word16)((tmp16no1 - aecm->supGain) >> 4);
-    }
-
-    // END: Update suppression gain
-
-    return aecm->supGain;
-}
-
-// WebRtcAecm_DelayCompensation(...)
-//
-// Secondary delay estimation that can be used as a backup or for validation. This function is
-// still under construction and not activated in current version.
-//
-//
-// @param  aecm  [i/o]   Handle of the AECM instance.
-//
-//
-void WebRtcAecm_DelayCompensation(AecmCore_t * const aecm)
-{
-    int i, j;
-    WebRtc_Word32 delayMeanEcho[CORR_BUF_LEN];
-    WebRtc_Word32 delayMeanNear[CORR_BUF_LEN];
-    WebRtc_Word16 sumBitPattern, bitPatternEcho, bitPatternNear, maxPos, maxValue,
-            maxValueLeft, maxValueRight;
-
-    // Check delay (calculate the delay offset (if we can)).
-    if ((aecm->startupState > 0) & (aecm->delayCount >= CORR_MAX_BUF) & aecm->delayOffsetFlag)
-    {
-        // Calculate mean values
-        for (i = 0; i < CORR_BUF_LEN; i++)
-        {
-            delayMeanEcho[i] = 0;
-            delayMeanNear[i] = 0;
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
-            for (j = 0; j < CORR_WIDTH; j++)
-            {
-                delayMeanEcho[i] += (WebRtc_Word32)aecm->echoStoredLogEnergy[i + j];
-                delayMeanNear[i] += (WebRtc_Word32)aecm->nearLogEnergy[i + j];
-            }
-#else
-            for (j = 0; j < CORR_WIDTH -1; )
-            {
-                delayMeanEcho[i] += (WebRtc_Word32)aecm->echoStoredLogEnergy[i + j];
-                delayMeanNear[i] += (WebRtc_Word32)aecm->nearLogEnergy[i + j];
-                j++;
-                delayMeanEcho[i] += (WebRtc_Word32)aecm->echoStoredLogEnergy[i + j];
-                delayMeanNear[i] += (WebRtc_Word32)aecm->nearLogEnergy[i + j];
-                j++;
-            }
-            delayMeanEcho[i] += (WebRtc_Word32)aecm->echoStoredLogEnergy[i + j];
-            delayMeanNear[i] += (WebRtc_Word32)aecm->nearLogEnergy[i + j];
-#endif
-        }
-        // Calculate correlation values
-        for (i = 0; i < CORR_BUF_LEN; i++)
-        {
-            sumBitPattern = 0;
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
-            for (j = 0; j < CORR_WIDTH; j++)
-            {
-                bitPatternEcho = (WebRtc_Word16)((WebRtc_Word32)aecm->echoStoredLogEnergy[i
-                        + j] * CORR_WIDTH > delayMeanEcho[i]);
-                bitPatternNear = (WebRtc_Word16)((WebRtc_Word32)aecm->nearLogEnergy[CORR_MAX
-                        + j] * CORR_WIDTH > delayMeanNear[CORR_MAX]);
-                sumBitPattern += !(bitPatternEcho ^ bitPatternNear);
-            }
-#else
-            for (j = 0; j < CORR_WIDTH -1; )
-            {
-                bitPatternEcho = (WebRtc_Word16)((WebRtc_Word32)aecm->echoStoredLogEnergy[i
-                    + j] * CORR_WIDTH > delayMeanEcho[i]);
-                bitPatternNear = (WebRtc_Word16)((WebRtc_Word32)aecm->nearLogEnergy[CORR_MAX
-                    + j] * CORR_WIDTH > delayMeanNear[CORR_MAX]);
-                sumBitPattern += !(bitPatternEcho ^ bitPatternNear);
-                j++;
-                bitPatternEcho = (WebRtc_Word16)((WebRtc_Word32)aecm->echoStoredLogEnergy[i
-                    + j] * CORR_WIDTH > delayMeanEcho[i]);
-                bitPatternNear = (WebRtc_Word16)((WebRtc_Word32)aecm->nearLogEnergy[CORR_MAX
-                    + j] * CORR_WIDTH > delayMeanNear[CORR_MAX]);
-                sumBitPattern += !(bitPatternEcho ^ bitPatternNear);
-                j++;
-            }
-            bitPatternEcho = (WebRtc_Word16)((WebRtc_Word32)aecm->echoStoredLogEnergy[i + j]
-                    * CORR_WIDTH > delayMeanEcho[i]);
-            bitPatternNear = (WebRtc_Word16)((WebRtc_Word32)aecm->nearLogEnergy[CORR_MAX + j]
-                    * CORR_WIDTH > delayMeanNear[CORR_MAX]);
-            sumBitPattern += !(bitPatternEcho ^ bitPatternNear);
-#endif
-            aecm->delayCorrelation[i] = sumBitPattern;
-        }
-        aecm->newDelayCorrData = 1; // Indicate we have new correlation data to evaluate
-    }
-    if ((aecm->startupState == 2) & (aecm->lastDelayUpdateCount > (CORR_WIDTH << 1))
-            & aecm->newDelayCorrData)
-    {
-        // Find maximum value and maximum position as well as values on the sides.
-        maxPos = 0;
-        maxValue = aecm->delayCorrelation[0];
-        maxValueLeft = maxValue;
-        maxValueRight = aecm->delayCorrelation[CORR_DEV];
-        for (i = 1; i < CORR_BUF_LEN; i++)
-        {
-            if (aecm->delayCorrelation[i] > maxValue)
-            {
-                maxValue = aecm->delayCorrelation[i];
-                maxPos = i;
-                if (maxPos < CORR_DEV)
-                {
-                    maxValueLeft = aecm->delayCorrelation[0];
-                    maxValueRight = aecm->delayCorrelation[i + CORR_DEV];
-                } else if (maxPos > (CORR_MAX << 1) - CORR_DEV)
-                {
-                    maxValueLeft = aecm->delayCorrelation[i - CORR_DEV];
-                    maxValueRight = aecm->delayCorrelation[(CORR_MAX << 1)];
-                } else
-                {
-                    maxValueLeft = aecm->delayCorrelation[i - CORR_DEV];
-                    maxValueRight = aecm->delayCorrelation[i + CORR_DEV];
-                }
-            }
-        }
-        if ((maxPos > 0) & (maxPos < (CORR_MAX << 1)))
-        {
-            // Avoid maximum at boundaries. The maximum peak has to be higher than
-            // CORR_MAX_LEVEL. It also has to be sharp, i.e. the value CORR_DEV bins off should
-            // be CORR_MAX_LOW lower than the maximum.
-            if ((maxValue > CORR_MAX_LEVEL) & (maxValueLeft < maxValue - CORR_MAX_LOW)
-                    & (maxValueRight < maxValue - CORR_MAX_LOW))
-            {
-                aecm->delayAdjust += CORR_MAX - maxPos;
-                aecm->newDelayCorrData = 0;
-                aecm->lastDelayUpdateCount = 0;
-            }
-        }
-    }
-    // END: "Check delay"
-}
-
-void WebRtcAecm_ProcessBlock(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
-                             const WebRtc_Word16 * const nearendNoisy,
-                             const WebRtc_Word16 * const nearendClean,
-                             WebRtc_Word16 * const output)
-{
-    int i, j;
-
-    WebRtc_UWord32 xfaSum;
-    WebRtc_UWord32 dfaNoisySum;
-    WebRtc_UWord32 echoEst32Gained;
-    WebRtc_UWord32 tmpU32;
-
-    WebRtc_Word32 tmp32no1;
-    WebRtc_Word32 tmp32no2;
-    WebRtc_Word32 echoEst32[PART_LEN1];
-
-    WebRtc_UWord16 xfa[PART_LEN1];
-    WebRtc_UWord16 dfaNoisy[PART_LEN1];
-    WebRtc_UWord16 dfaClean[PART_LEN1];
-    WebRtc_UWord16* ptrDfaClean = dfaClean;
-
-    int outCFFT;
-
-    WebRtc_Word16 fft[PART_LEN4];
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
-    WebRtc_Word16 postFft[PART_LEN4];
-#else
-    WebRtc_Word16 postFft[PART_LEN2];
-#endif
-    WebRtc_Word16 dfwReal[PART_LEN1];
-    WebRtc_Word16 dfwImag[PART_LEN1];
-    WebRtc_Word16 xfwReal[PART_LEN1];
-    WebRtc_Word16 xfwImag[PART_LEN1];
-    WebRtc_Word16 efwReal[PART_LEN1];
-    WebRtc_Word16 efwImag[PART_LEN1];
-    WebRtc_Word16 hnl[PART_LEN1];
-    WebRtc_Word16 numPosCoef;
-    WebRtc_Word16 nlpGain;
-    WebRtc_Word16 delay, diff, diffMinusOne;
-    WebRtc_Word16 tmp16no1;
-    WebRtc_Word16 tmp16no2;
-#ifdef AECM_WITH_ABS_APPROX
-    WebRtc_Word16 maxValue;
-    WebRtc_Word16 minValue;
-#endif
-    WebRtc_Word16 mu;
-    WebRtc_Word16 supGain;
-    WebRtc_Word16 zeros32, zeros16;
-    WebRtc_Word16 zerosDBufNoisy, zerosDBufClean, zerosXBuf;
-    WebRtc_Word16 resolutionDiff, qDomainDiff;
-
-#ifdef ARM_WINM_LOG_
-    DWORD temp;
-    static int flag0 = 0;
-    __int64 freq, start, end, diff__;
-    unsigned int milliseconds;
-#endif
-
-#ifdef AECM_WITH_ABS_APPROX
-    WebRtc_UWord16 alpha, beta;
-#endif
-
-    // Determine startup state. There are three states:
-    // (0) the first CONV_LEN blocks
-    // (1) another CONV_LEN blocks
-    // (2) the rest
-
-    if (aecm->startupState < 2)
-    {
-        aecm->startupState = (aecm->totCount >= CONV_LEN) + (aecm->totCount >= CONV_LEN2);
-    }
-    // END: Determine startup state
-
-    // Buffer near and far end signals
-    memcpy(aecm->xBuf + PART_LEN, farend, sizeof(WebRtc_Word16) * PART_LEN);
-    memcpy(aecm->dBufNoisy + PART_LEN, nearendNoisy, sizeof(WebRtc_Word16) * PART_LEN);
-    if (nearendClean != NULL)
-    {
-        memcpy(aecm->dBufClean + PART_LEN, nearendClean, sizeof(WebRtc_Word16) * PART_LEN);
-    }
-    // TODO(bjornv): Will be removed in final version.
-#ifdef VAD_DATA
-    fwrite(aecm->xBuf, sizeof(WebRtc_Word16), PART_LEN, aecm->far_file);
-#endif
-
-#ifdef AECM_DYNAMIC_Q
-    tmp16no1 = WebRtcSpl_MaxAbsValueW16(aecm->dBufNoisy, PART_LEN2);
-    tmp16no2 = WebRtcSpl_MaxAbsValueW16(aecm->xBuf, PART_LEN2);
-    zerosDBufNoisy = WebRtcSpl_NormW16(tmp16no1);
-    zerosXBuf = WebRtcSpl_NormW16(tmp16no2);
-#else
-    zerosDBufNoisy = 0;
-    zerosXBuf = 0;
-#endif
-    aecm->dfaNoisyQDomainOld = aecm->dfaNoisyQDomain;
-    aecm->dfaNoisyQDomain = zerosDBufNoisy;
-
-    if (nearendClean != NULL)
-    {
-#ifdef AECM_DYNAMIC_Q
-        tmp16no1 = WebRtcSpl_MaxAbsValueW16(aecm->dBufClean, PART_LEN2);
-        zerosDBufClean = WebRtcSpl_NormW16(tmp16no1);
-#else
-        zerosDBufClean = 0;
-#endif
-        aecm->dfaCleanQDomainOld = aecm->dfaCleanQDomain;
-        aecm->dfaCleanQDomain = zerosDBufClean;
-    } else
-    {
-        zerosDBufClean = zerosDBufNoisy;
-        aecm->dfaCleanQDomainOld = aecm->dfaNoisyQDomainOld;
-        aecm->dfaCleanQDomain = aecm->dfaNoisyQDomain;
-    }
-
-#ifdef ARM_WINM_LOG_
-    // measure tick start
-    QueryPerformanceFrequency((LARGE_INTEGER*)&freq);
-    QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
-    // FFT of noisy near end signal
-    for (i = 0; i < PART_LEN; i++)
-    {
-        j = WEBRTC_SPL_LSHIFT_W32(i, 1);
-        // Window near end
-        fft[j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((aecm->dBufNoisy[i]
-                        << zerosDBufNoisy), kSqrtHanning[i], 14);
-        fft[PART_LEN2 + j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
-                (aecm->dBufNoisy[PART_LEN + i] << zerosDBufNoisy),
-                kSqrtHanning[PART_LEN - i], 14);
-        // Inserting zeros in imaginary parts
-        fft[j + 1] = 0;
-        fft[PART_LEN2 + j + 1] = 0;
-    }
-
-    // Fourier transformation of near end signal.
-    // The result is scaled with 1/PART_LEN2, that is, the result is in Q(-6) for PART_LEN = 32
-
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
-    outCFFT = WebRtcSpl_ComplexFFT2(fft, postFft, PART_LEN_SHIFT, 1);
-
-    // The imaginary part has to switch sign
-    for(i = 1; i < PART_LEN2-1;)
-    {
-        postFft[i] = -postFft[i];
-        i += 2;
-        postFft[i] = -postFft[i];
-        i += 2;
-    }
-#else
-    WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
-    outCFFT = WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
-
-    // Take only the first PART_LEN2 samples
-    for (i = 0; i < PART_LEN2; i++)
-    {
-        postFft[i] = fft[i];
-    }
-    // The imaginary part has to switch sign
-    for (i = 1; i < PART_LEN2;)
-    {
-        postFft[i] = -postFft[i];
-        i += 2;
-    }
-#endif
-
-    // Extract imaginary and real part, calculate the magnitude for all frequency bins
-    dfwImag[0] = 0;
-    dfwImag[PART_LEN] = 0;
-    dfwReal[0] = postFft[0];
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
-    dfwReal[PART_LEN] = postFft[PART_LEN2];
-#else
-    dfwReal[PART_LEN] = fft[PART_LEN2];
-#endif
-    dfaNoisy[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[0]);
-    dfaNoisy[PART_LEN] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[PART_LEN]);
-    dfaNoisySum = (WebRtc_UWord32)(dfaNoisy[0]);
-    dfaNoisySum += (WebRtc_UWord32)(dfaNoisy[PART_LEN]);
-
-    for (i = 1; i < PART_LEN; i++)
-    {
-        j = WEBRTC_SPL_LSHIFT_W32(i, 1);
-        dfwReal[i] = postFft[j];
-        dfwImag[i] = postFft[j + 1];
-
-        if (dfwReal[i] == 0 || dfwImag[i] == 0)
-        {
-            dfaNoisy[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[i] + dfwImag[i]);
-        } else
-        {
-            // Approximation for magnitude of complex fft output
-            // magn = sqrt(real^2 + imag^2)
-            // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
-            //
-            // The parameters alpha and beta are stored in Q15
-
-            tmp16no1 = WEBRTC_SPL_ABS_W16(postFft[j]);
-            tmp16no2 = WEBRTC_SPL_ABS_W16(postFft[j + 1]);
-
-#ifdef AECM_WITH_ABS_APPROX
-            if(tmp16no1 > tmp16no2)
-            {
-                maxValue = tmp16no1;
-                minValue = tmp16no2;
-            } else
-            {
-                maxValue = tmp16no2;
-                minValue = tmp16no1;
-            }
-
-            // Magnitude in Q-6
-            if ((maxValue >> 2) > minValue)
-            {
-                alpha = kAlpha1;
-                beta = kBeta1;
-            } else if ((maxValue >> 1) > minValue)
-            {
-                alpha = kAlpha2;
-                beta = kBeta2;
-            } else
-            {
-                alpha = kAlpha3;
-                beta = kBeta3;
-            }
-            tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(maxValue, alpha, 15);
-            tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(minValue, beta, 15);
-            dfaNoisy[i] = (WebRtc_UWord16)tmp16no1 + (WebRtc_UWord16)tmp16no2;
-#else
-            tmp32no1 = WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1);
-            tmp32no2 = WEBRTC_SPL_MUL_16_16(tmp16no2, tmp16no2);
-            tmp32no2 = WEBRTC_SPL_ADD_SAT_W32(tmp32no1, tmp32no2);
-            tmp32no1 = WebRtcSpl_Sqrt(tmp32no2);
-            dfaNoisy[i] = (WebRtc_UWord16)tmp32no1;
-#endif
-        }
-        dfaNoisySum += (WebRtc_UWord32)dfaNoisy[i];
-    }
-    // END: FFT of noisy near end signal
-
-    if (nearendClean == NULL)
-    {
-        ptrDfaClean = dfaNoisy;
-    } else
-    {
-        // FFT of clean near end signal
-        for (i = 0; i < PART_LEN; i++)
-        {
-            j = WEBRTC_SPL_LSHIFT_W32(i, 1);
-            // Window near end
-            fft[j]
-                    = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((aecm->dBufClean[i] << zerosDBufClean), kSqrtHanning[i], 14);
-            fft[PART_LEN2 + j]
-                    = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((aecm->dBufClean[PART_LEN + i] << zerosDBufClean), kSqrtHanning[PART_LEN - i], 14);
-            // Inserting zeros in imaginary parts
-            fft[j + 1] = 0;
-            fft[PART_LEN2 + j + 1] = 0;
-        }
-
-        // Fourier transformation of near end signal.
-        // The result is scaled with 1/PART_LEN2, that is, in Q(-6) for PART_LEN = 32
-
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
-        outCFFT = WebRtcSpl_ComplexFFT2(fft, postFft, PART_LEN_SHIFT, 1);
-
-        // The imaginary part has to switch sign
-        for(i = 1; i < PART_LEN2-1;)
-        {
-            postFft[i] = -postFft[i];
-            i += 2;
-            postFft[i] = -postFft[i];
-            i += 2;
-        }
-#else
-        WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
-        outCFFT = WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
-
-        // Take only the first PART_LEN2 samples
-        for (i = 0; i < PART_LEN2; i++)
-        {
-            postFft[i] = fft[i];
-        }
-        // The imaginary part has to switch sign
-        for (i = 1; i < PART_LEN2;)
-        {
-            postFft[i] = -postFft[i];
-            i += 2;
-        }
-#endif
-
-        // Extract imaginary and real part, calculate the magnitude for all frequency bins
-        dfwImag[0] = 0;
-        dfwImag[PART_LEN] = 0;
-        dfwReal[0] = postFft[0];
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
-        dfwReal[PART_LEN] = postFft[PART_LEN2];
-#else
-        dfwReal[PART_LEN] = fft[PART_LEN2];
-#endif
-        dfaClean[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[0]);
-        dfaClean[PART_LEN] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[PART_LEN]);
-
-        for (i = 1; i < PART_LEN; i++)
-        {
-            j = WEBRTC_SPL_LSHIFT_W32(i, 1);
-            dfwReal[i] = postFft[j];
-            dfwImag[i] = postFft[j + 1];
-
-            if (dfwReal[i] == 0 || dfwImag[i] == 0)
-            {
-                dfaClean[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(dfwReal[i] + dfwImag[i]);
-            } else
-            {
-                // Approximation for magnitude of complex fft output
-                // magn = sqrt(real^2 + imag^2)
-                // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
-                //
-                // The parameters alpha and beta are stored in Q15
-
-                tmp16no1 = WEBRTC_SPL_ABS_W16(postFft[j]);
-                tmp16no2 = WEBRTC_SPL_ABS_W16(postFft[j + 1]);
-
-#ifdef AECM_WITH_ABS_APPROX
-                if(tmp16no1 > tmp16no2)
-                {
-                    maxValue = tmp16no1;
-                    minValue = tmp16no2;
-                } else
-                {
-                    maxValue = tmp16no2;
-                    minValue = tmp16no1;
-                }
-
-                // Magnitude in Q-6
-                if ((maxValue >> 2) > minValue)
-                {
-                    alpha = kAlpha1;
-                    beta = kBeta1;
-                } else if ((maxValue >> 1) > minValue)
-                {
-                    alpha = kAlpha2;
-                    beta = kBeta2;
-                } else
-                {
-                    alpha = kAlpha3;
-                    beta = kBeta3;
-                }
-                tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(maxValue, alpha, 15);
-                tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(minValue, beta, 15);
-                dfaClean[i] = (WebRtc_UWord16)tmp16no1 + (WebRtc_UWord16)tmp16no2;
-#else
-                tmp32no1 = WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1);
-                tmp32no2 = WEBRTC_SPL_MUL_16_16(tmp16no2, tmp16no2);
-                tmp32no2 = WEBRTC_SPL_ADD_SAT_W32(tmp32no1, tmp32no2);
-                tmp32no1 = WebRtcSpl_Sqrt(tmp32no2);
-                dfaClean[i] = (WebRtc_UWord16)tmp32no1;
-#endif
-            }
-        }
-    }
-    // END: FFT of clean near end signal
-
-    // FFT of far end signal
-    for (i = 0; i < PART_LEN; i++)
-    {
-        j = WEBRTC_SPL_LSHIFT_W32(i, 1);
-        // Window farend
-        fft[j]
-                = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((aecm->xBuf[i] << zerosXBuf), kSqrtHanning[i], 14);
-        fft[PART_LEN2 + j]
-                = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((aecm->xBuf[PART_LEN + i] << zerosXBuf), kSqrtHanning[PART_LEN - i], 14);
-        // Inserting zeros in imaginary parts
-        fft[j + 1] = 0;
-        fft[PART_LEN2 + j + 1] = 0;
-    }
-    // Fourier transformation of far end signal.
-    // The result is scaled with 1/PART_LEN2, that is the result is in Q(-6) for PART_LEN = 32
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
-    outCFFT = WebRtcSpl_ComplexFFT2(fft, postFft, PART_LEN_SHIFT, 1);
-
-    // The imaginary part has to switch sign
-    for(i = 1; i < PART_LEN2-1;)
-    {
-        postFft[i] = -postFft[i];
-        i += 2;
-        postFft[i] = -postFft[i];
-        i += 2;
-    }
-#else
-    WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
-    outCFFT = WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
-
-    // Take only the first PART_LEN2 samples
-    for (i = 0; i < PART_LEN2; i++)
-    {
-        postFft[i] = fft[i];
-    }
-    // The imaginary part has to switch sign
-    for (i = 1; i < PART_LEN2;)
-    {
-        postFft[i] = -postFft[i];
-        i += 2;
-    }
-#endif
-
-    // Extract imaginary and real part, calculate the magnitude for all frequency bins
-    xfwImag[0] = 0;
-    xfwImag[PART_LEN] = 0;
-    xfwReal[0] = postFft[0];
-#if (defined ARM_WINM) || (defined ARM9E_GCC) || (defined ANDROID_AECOPT)
-    xfwReal[PART_LEN] = postFft[PART_LEN2];
-#else
-    xfwReal[PART_LEN] = fft[PART_LEN2];
-#endif
-    xfa[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(xfwReal[0]);
-    xfa[PART_LEN] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(xfwReal[PART_LEN]);
-    xfaSum = (WebRtc_UWord32)(xfa[0]) + (WebRtc_UWord32)(xfa[PART_LEN]);
-
-    for (i = 1; i < PART_LEN; i++)
-    {
-        j = WEBRTC_SPL_LSHIFT_W32(i,1);
-        xfwReal[i] = postFft[j];
-        xfwImag[i] = postFft[j + 1];
-
-        if (xfwReal[i] == 0 || xfwImag[i] == 0)
-        {
-            xfa[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(xfwReal[i] + xfwImag[i]);
-        } else
-        {
-            // Approximation for magnitude of complex fft output
-            // magn = sqrt(real^2 + imag^2)
-            // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
-            //
-            // The parameters alpha and beta are stored in Q15
-
-            tmp16no1 = WEBRTC_SPL_ABS_W16(postFft[j]);
-            tmp16no2 = WEBRTC_SPL_ABS_W16(postFft[j + 1]);
-
-#ifdef AECM_WITH_ABS_APPROX
-            if(tmp16no1 > xfwImag[i])
-            {
-                maxValue = tmp16no1;
-                minValue = tmp16no2;
-            } else
-            {
-                maxValue = tmp16no2;
-                minValue = tmp16no1;
-            }
-            // Magnitude in Q-6
-            if ((maxValue >> 2) > minValue)
-            {
-                alpha = kAlpha1;
-                beta = kBeta1;
-            } else if ((maxValue >> 1) > minValue)
-            {
-                alpha = kAlpha2;
-                beta = kBeta2;
-            } else
-            {
-                alpha = kAlpha3;
-                beta = kBeta3;
-            }
-            tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(maxValue, alpha, 15);
-            tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(minValue, beta, 15);
-            xfa[i] = (WebRtc_UWord16)tmp16no1 + (WebRtc_UWord16)tmp16no2;
-#else
-            tmp32no1 = WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1);
-            tmp32no2 = WEBRTC_SPL_MUL_16_16(tmp16no2, tmp16no2);
-            tmp32no2 = WEBRTC_SPL_ADD_SAT_W32(tmp32no1, tmp32no2);
-            tmp32no1 = WebRtcSpl_Sqrt(tmp32no2);
-            xfa[i] = (WebRtc_UWord16)tmp32no1;
-#endif
-        }
-        xfaSum += (WebRtc_UWord32)xfa[i];
-    }
-
-#ifdef ARM_WINM_LOG_
-    // measure tick end
-    QueryPerformanceCounter((LARGE_INTEGER*)&end);
-    diff__ = ((end - start) * 1000) / (freq/1000);
-    milliseconds = (unsigned int)(diff__ & 0xffffffff);
-    WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
-    // END: FFT of far end signal
-
-    // Get the delay
-
-    // Fixed delay estimation
-    // input: dfaFIX, xfaFIX in Q-stages
-    // output: delay in Q0
-    //
-    // comment on the fixed point accuracy of estimate_delayFIX
-    // -> due to rounding the fixed point variables xfa and dfa contain a lot more zeros
-    // than the corresponding floating point variables this results in big differences
-    // between the floating point and the fixed point logarithmic spectra for small values
-#ifdef ARM_WINM_LOG_
-    // measure tick start
-    QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
-    // Save far-end history and estimate delay
-    delay = WebRtcAecm_EstimateDelay(aecm, xfa, dfaNoisy, zerosXBuf);
-
-    if (aecm->fixedDelay >= 0)
-    {
-        // Use fixed delay
-        delay = aecm->fixedDelay;
-    }
-
-    aecm->currentDelay = delay;
-
-    if ((aecm->delayOffsetFlag) & (aecm->startupState > 0)) // If delay compensation is on
-    {
-        // If the delay estimate changed from previous block, update the offset
-        if ((aecm->currentDelay != aecm->previousDelay) & !aecm->currentDelay
-                & !aecm->previousDelay)
-        {
-            aecm->delayAdjust += (aecm->currentDelay - aecm->previousDelay);
-        }
-        // Compensate with the offset estimate
-        aecm->currentDelay -= aecm->delayAdjust;
-        aecm->previousDelay = delay;
-    }
-
-    diff = aecm->delHistoryPos - aecm->currentDelay;
-    if (diff < 0)
-    {
-        diff = diff + MAX_DELAY;
-    }
-
-#ifdef ARM_WINM_LOG_
-    // measure tick end
-    QueryPerformanceCounter((LARGE_INTEGER*)&end);
-    diff__ = ((end - start) * 1000) / (freq/1000);
-    milliseconds = (unsigned int)(diff__ & 0xffffffff);
-    WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
-
-    // END: Get the delay
-
-#ifdef ARM_WINM_LOG_
-    // measure tick start
-    QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-    // Calculate log(energy) and update energy threshold levels
-    WebRtcAecm_CalcEnergies(aecm, diff, dfaNoisySum, echoEst32);
-
-    // Calculate stepsize
-    mu = WebRtcAecm_CalcStepSize(aecm);
-
-    // Update counters
-    aecm->totCount++;
-    aecm->lastDelayUpdateCount++;
-
-    // This is the channel estimation algorithm.
-    // It is base on NLMS but has a variable step length, which was calculated above.
-    WebRtcAecm_UpdateChannel(aecm, dfaNoisy, diff, mu, echoEst32);
-    WebRtcAecm_DelayCompensation(aecm);
-    supGain = WebRtcAecm_CalcSuppressionGain(aecm);
-
-#ifdef ARM_WINM_LOG_
-    // measure tick end
-    QueryPerformanceCounter((LARGE_INTEGER*)&end);
-    diff__ = ((end - start) * 1000) / (freq/1000);
-    milliseconds = (unsigned int)(diff__ & 0xffffffff);
-    WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
-
-#ifdef ARM_WINM_LOG_
-    // measure tick start
-    QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
-    // Calculate Wiener filter hnl[]
-    numPosCoef = 0;
-    diffMinusOne = diff - 1;
-    if (diff == 0)
-    {
-        diffMinusOne = MAX_DELAY;
-    }
-    for (i = 0; i < PART_LEN1; i++)
-    {
-        // Far end signal through channel estimate in Q8
-        // How much can we shift right to preserve resolution
-        tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
-        aecm->echoFilt[i] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32no1, 50), 8);
-
-        zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
-        zeros16 = WebRtcSpl_NormW16(supGain) + 1;
-        if (zeros32 + zeros16 > 16)
-        {
-            // Multiplication is safe
-            // Result in Q(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN+aecm->xfaQDomainBuf[diff])
-            echoEst32Gained = WEBRTC_SPL_UMUL_32_16((WebRtc_UWord32)aecm->echoFilt[i],
-                                                    (WebRtc_UWord16)supGain);
-            resolutionDiff = 14 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
-            resolutionDiff += (aecm->dfaCleanQDomain - aecm->xfaQDomainBuf[diff]);
-        } else
-        {
-            tmp16no1 = 17 - zeros32 - zeros16;
-            resolutionDiff = 14 + tmp16no1 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
-            resolutionDiff += (aecm->dfaCleanQDomain - aecm->xfaQDomainBuf[diff]);
-            if (zeros32 > tmp16no1)
-            {
-                echoEst32Gained = WEBRTC_SPL_UMUL_32_16((WebRtc_UWord32)aecm->echoFilt[i],
-                        (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W16(supGain,
-                                tmp16no1)); // Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
-            } else
-            {
-                // Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
-                echoEst32Gained = WEBRTC_SPL_UMUL_32_16(
-                        (WebRtc_UWord32)WEBRTC_SPL_RSHIFT_W32(aecm->echoFilt[i], tmp16no1),
-                        (WebRtc_UWord16)supGain);
-            }
-        }
-
-        zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]);
-        if ((zeros16 < (aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld))
-                & (aecm->nearFilt[i]))
-        {
-            tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i], zeros16);
-            qDomainDiff = zeros16 - aecm->dfaCleanQDomain + aecm->dfaCleanQDomainOld;
-        } else
-        {
-            tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i], aecm->dfaCleanQDomain
-                                            - aecm->dfaCleanQDomainOld);
-            qDomainDiff = 0;
-        }
-        tmp16no2 = WEBRTC_SPL_SHIFT_W16(ptrDfaClean[i], qDomainDiff);
-        tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no2 - tmp16no1, 1, 4);
-        tmp16no2 += tmp16no1;
-        zeros16 = WebRtcSpl_NormW16(tmp16no2);
-        if ((tmp16no2) & (-qDomainDiff > zeros16))
-        {
-            aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX;
-        } else
-        {
-            aecm->nearFilt[i] = WEBRTC_SPL_SHIFT_W16(tmp16no2, -qDomainDiff);
-        }
-
-        // Wiener filter coefficients, resulting hnl in Q14
-        if (echoEst32Gained == 0)
-        {
-            hnl[i] = ONE_Q14;
-        } else if (aecm->nearFilt[i] == 0)
-        {
-            hnl[i] = 0;
-        } else
-        {
-            // Multiply the suppression gain
-            // Rounding
-            echoEst32Gained += (WebRtc_UWord32)(aecm->nearFilt[i] >> 1);
-            tmpU32 = WebRtcSpl_DivU32U16(echoEst32Gained, (WebRtc_UWord16)aecm->nearFilt[i]);
-
-            // Current resolution is
-            // Q-(RESOLUTION_CHANNEL + RESOLUTION_SUPGAIN - max(0, 17 - zeros16 - zeros32))
-            // Make sure we are in Q14
-            tmp32no1 = (WebRtc_Word32)WEBRTC_SPL_SHIFT_W32(tmpU32, resolutionDiff);
-            if (tmp32no1 > ONE_Q14)
-            {
-                hnl[i] = 0;
-            } else if (tmp32no1 < 0)
-            {
-                hnl[i] = ONE_Q14;
-            } else
-            {
-                // 1-echoEst/dfa
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
-                hnl[i] = ONE_Q14 - (WebRtc_Word16)tmp32no1;
-                if (hnl[i] < 0)
-                {
-                    hnl[i] = 0;
-                }
-#else
-                hnl[i] = ((ONE_Q14 - (WebRtc_Word16)tmp32no1) > 0) ? (ONE_Q14 - (WebRtc_Word16)tmp32no1) : 0;
-#endif
-            }
-        }
-        if (hnl[i])
-        {
-            numPosCoef++;
-        }
-    }
-
-#ifdef ARM_WINM_LOG_
-    // measure tick end
-    QueryPerformanceCounter((LARGE_INTEGER*)&end);
-    diff__ = ((end - start) * 1000) / (freq/1000);
-    milliseconds = (unsigned int)(diff__ & 0xffffffff);
-    WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
-
-#ifdef ARM_WINM_LOG_
-    // measure tick start
-    QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
-    // Calculate NLP gain, result is in Q14
-    for (i = 0; i < PART_LEN1; i++)
-    {
-        if (aecm->nlpFlag)
-        {
-            // Truncate values close to zero and one.
-            if (hnl[i] > NLP_COMP_HIGH)
-            {
-                hnl[i] = ONE_Q14;
-            } else if (hnl[i] < NLP_COMP_LOW)
-            {
-                hnl[i] = 0;
-            }
-
-            // Remove outliers
-            if (numPosCoef < 3)
-            {
-                nlpGain = 0;
-            } else
-            {
-                nlpGain = ONE_Q14;
-            }
-            // NLP
-            if ((hnl[i] == ONE_Q14) && (nlpGain == ONE_Q14))
-            {
-                hnl[i] = ONE_Q14;
-            } else
-            {
-                hnl[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], nlpGain, 14);
-            }
-        }
-
-        // multiply with Wiener coefficients
-        efwReal[i] = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfwReal[i], hnl[i],
-                                                                          14));
-        efwImag[i] = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfwImag[i], hnl[i],
-                                                                          14));
-    }
-
-    if (aecm->cngMode == AecmTrue)
-    {
-        WebRtcAecm_ComfortNoise(aecm, ptrDfaClean, efwReal, efwImag, hnl);
-    }
-
-#ifdef ARM_WINM_LOG_
-    // measure tick end
-    QueryPerformanceCounter((LARGE_INTEGER*)&end);
-    diff__ = ((end - start) * 1000) / (freq/1000);
-    milliseconds = (unsigned int)(diff__ & 0xffffffff);
-    WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
-
-#ifdef ARM_WINM_LOG_
-    // measure tick start
-    QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
-    // Synthesis
-    for (i = 1; i < PART_LEN; i++)
-    {
-        j = WEBRTC_SPL_LSHIFT_W32(i, 1);
-        fft[j] = efwReal[i];
-
-        // mirrored data, even
-        fft[PART_LEN4 - j] = efwReal[i];
-        fft[j + 1] = -efwImag[i];
-
-        //mirrored data, odd
-        fft[PART_LEN4 - (j - 1)] = efwImag[i];
-    }
-    fft[0] = efwReal[0];
-    fft[1] = -efwImag[0];
-
-    fft[PART_LEN2] = efwReal[PART_LEN];
-    fft[PART_LEN2 + 1] = -efwImag[PART_LEN];
-
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
-    // inverse FFT, result should be scaled with outCFFT
-    WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
-    outCFFT = WebRtcSpl_ComplexIFFT(fft, PART_LEN_SHIFT, 1);
-
-    //take only the real values and scale with outCFFT
-    for (i = 0; i < PART_LEN2; i++)
-    {
-        j = WEBRTC_SPL_LSHIFT_W32(i, 1);
-        fft[i] = fft[j];
-    }
-#else
-    outCFFT = WebRtcSpl_ComplexIFFT2(fft, postFft, PART_LEN_SHIFT, 1);
-
-    //take only the real values and scale with outCFFT
-    for(i = 0, j = 0; i < PART_LEN2;)
-    {
-        fft[i] = postFft[j];
-        i += 1;
-        j += 2;
-        fft[i] = postFft[j];
-        i += 1;
-        j += 2;
-    }
-#endif
-
-    for (i = 0; i < PART_LEN; i++)
-    {
-        fft[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
-                fft[i],
-                kSqrtHanning[i],
-                14);
-        tmp32no1 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)fft[i],
-                outCFFT - aecm->dfaCleanQDomain);
-        fft[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
-                tmp32no1 + aecm->outBuf[i],
-                WEBRTC_SPL_WORD16_MIN);
-        output[i] = fft[i];
-
-        tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT(
-                fft[PART_LEN + i],
-                kSqrtHanning[PART_LEN - i],
-                14);
-        tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1,
-                outCFFT - aecm->dfaCleanQDomain);
-        aecm->outBuf[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(
-                WEBRTC_SPL_WORD16_MAX,
-                tmp32no1,
-                WEBRTC_SPL_WORD16_MIN);
-    }
-
-#ifdef ARM_WINM_LOG_
-    // measure tick end
-    QueryPerformanceCounter((LARGE_INTEGER*)&end);
-    diff__ = ((end - start) * 1000) / (freq/1000);
-    milliseconds = (unsigned int)(diff__ & 0xffffffff);
-    WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
-    // Copy the current block to the old position (outBuf is shifted elsewhere)
-    memcpy(aecm->xBuf, aecm->xBuf + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
-    memcpy(aecm->dBufNoisy, aecm->dBufNoisy + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
-    if (nearendClean != NULL)
-    {
-        memcpy(aecm->dBufClean, aecm->dBufClean + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
-    }
-}
-
-// Generate comfort noise and add to output signal.
-//
-// \param[in]     aecm     Handle of the AECM instance.
-// \param[in]     dfa     Absolute value of the nearend signal (Q[aecm->dfaQDomain]).
-// \param[in,out] outReal Real part of the output signal (Q[aecm->dfaQDomain]).
-// \param[in,out] outImag Imaginary part of the output signal (Q[aecm->dfaQDomain]).
-// \param[in]     lambda  Suppression gain with which to scale the noise level (Q14).
-//
-static void WebRtcAecm_ComfortNoise(AecmCore_t * const aecm, const WebRtc_UWord16 * const dfa,
-                                    WebRtc_Word16 * const outReal,
-                                    WebRtc_Word16 * const outImag,
-                                    const WebRtc_Word16 * const lambda)
-{
-    WebRtc_Word16 i;
-    WebRtc_Word16 tmp16;
-    WebRtc_Word32 tmp32;
-
-    WebRtc_Word16 randW16[PART_LEN];
-    WebRtc_Word16 uReal[PART_LEN1];
-    WebRtc_Word16 uImag[PART_LEN1];
-    WebRtc_Word32 outLShift32[PART_LEN1];
-    WebRtc_Word16 noiseRShift16[PART_LEN1];
-
-    WebRtc_Word16 shiftFromNearToNoise[PART_LEN1];
-    WebRtc_Word16 minTrackShift;
-    WebRtc_Word32 upper32;
-    WebRtc_Word32 lower32;
-
-    if (aecm->noiseEstCtr < 100)
-    {
-        // Track the minimum more quickly initially.
-        aecm->noiseEstCtr++;
-        minTrackShift = 7;
-    } else
-    {
-        minTrackShift = 9;
-    }
-
-    // Estimate noise power.
-    for (i = 0; i < PART_LEN1; i++)
-    {
-        shiftFromNearToNoise[i] = aecm->noiseEstQDomain[i] - aecm->dfaCleanQDomain;
-
-        // Shift to the noise domain.
-        tmp32 = (WebRtc_Word32)dfa[i];
-        outLShift32[i] = WEBRTC_SPL_SHIFT_W32(tmp32, shiftFromNearToNoise[i]);
-
-        if (outLShift32[i] < aecm->noiseEst[i])
-        {
-            // Track the minimum.
-            aecm->noiseEst[i] += ((outLShift32[i] - aecm->noiseEst[i]) >> minTrackShift);
-        } else
-        {
-            // Ramp slowly upwards until we hit the minimum again.
-
-            // Avoid overflow.
-            if (aecm->noiseEst[i] < 2146435583)
-            {
-                // Store the fractional portion.
-                upper32 = (aecm->noiseEst[i] & 0xffff0000) >> 16;
-                lower32 = aecm->noiseEst[i] & 0x0000ffff;
-                upper32 = ((upper32 * 2049) >> 11);
-                lower32 = ((lower32 * 2049) >> 11);
-                aecm->noiseEst[i] = WEBRTC_SPL_ADD_SAT_W32(upper32 << 16, lower32);
-            }
-        }
-    }
-
-    for (i = 0; i < PART_LEN1; i++)
-    {
-        tmp32 = WEBRTC_SPL_SHIFT_W32(aecm->noiseEst[i], -shiftFromNearToNoise[i]);
-        if (tmp32 > 32767)
-        {
-            tmp32 = 32767;
-            aecm->noiseEst[i] = WEBRTC_SPL_SHIFT_W32(tmp32, shiftFromNearToNoise[i]);
-        }
-        noiseRShift16[i] = (WebRtc_Word16)tmp32;
-
-        tmp16 = ONE_Q14 - lambda[i];
-        noiseRShift16[i]
-                = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, noiseRShift16[i], 14);
-    }
-
-    // Generate a uniform random array on [0 2^15-1].
-    WebRtcSpl_RandUArray(randW16, PART_LEN, &aecm->seed);
-
-    // Generate noise according to estimated energy.
-    uReal[0] = 0; // Reject LF noise.
-    uImag[0] = 0;
-    for (i = 1; i < PART_LEN1; i++)
-    {
-        // Get a random index for the cos and sin tables over [0 359].
-        tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(359, randW16[i - 1], 15);
-
-        // Tables are in Q13.
-        uReal[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(noiseRShift16[i],
-                WebRtcSpl_kCosTable[tmp16], 13);
-        uImag[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(-noiseRShift16[i],
-                WebRtcSpl_kSinTable[tmp16], 13);
-    }
-    uImag[PART_LEN] = 0;
-
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
-    for (i = 0; i < PART_LEN1; i++)
-    {
-        outReal[i] = WEBRTC_SPL_ADD_SAT_W16(outReal[i], uReal[i]);
-        outImag[i] = WEBRTC_SPL_ADD_SAT_W16(outImag[i], uImag[i]);
-    }
-#else
-    for (i = 0; i < PART_LEN1 -1; )
-    {
-        outReal[i] = WEBRTC_SPL_ADD_SAT_W16(outReal[i], uReal[i]);
-        outImag[i] = WEBRTC_SPL_ADD_SAT_W16(outImag[i], uImag[i]);
-        i++;
-
-        outReal[i] = WEBRTC_SPL_ADD_SAT_W16(outReal[i], uReal[i]);
-        outImag[i] = WEBRTC_SPL_ADD_SAT_W16(outImag[i], uImag[i]);
-        i++;
-    }
-    outReal[i] = WEBRTC_SPL_ADD_SAT_W16(outReal[i], uReal[i]);
-    outImag[i] = WEBRTC_SPL_ADD_SAT_W16(outImag[i], uImag[i]);
-#endif
-}
-
-void WebRtcAecm_BufferFarFrame(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
-                               const int farLen)
-{
-    int writeLen = farLen, writePos = 0;
-
-    // Check if the write position must be wrapped
-    while (aecm->farBufWritePos + writeLen > FAR_BUF_LEN)
-    {
-        // Write to remaining buffer space before wrapping
-        writeLen = FAR_BUF_LEN - aecm->farBufWritePos;
-        memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
-               sizeof(WebRtc_Word16) * writeLen);
-        aecm->farBufWritePos = 0;
-        writePos = writeLen;
-        writeLen = farLen - writeLen;
-    }
-
-    memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
-           sizeof(WebRtc_Word16) * writeLen);
-    aecm->farBufWritePos += writeLen;
-}
-
-void WebRtcAecm_FetchFarFrame(AecmCore_t * const aecm, WebRtc_Word16 * const farend,
-                              const int farLen, const int knownDelay)
-{
-    int readLen = farLen;
-    int readPos = 0;
-    int delayChange = knownDelay - aecm->lastKnownDelay;
-
-    aecm->farBufReadPos -= delayChange;
-
-    // Check if delay forces a read position wrap
-    while (aecm->farBufReadPos < 0)
-    {
-        aecm->farBufReadPos += FAR_BUF_LEN;
-    }
-    while (aecm->farBufReadPos > FAR_BUF_LEN - 1)
-    {
-        aecm->farBufReadPos -= FAR_BUF_LEN;
-    }
-
-    aecm->lastKnownDelay = knownDelay;
-
-    // Check if read position must be wrapped
-    while (aecm->farBufReadPos + readLen > FAR_BUF_LEN)
-    {
-
-        // Read from remaining buffer space before wrapping
-        readLen = FAR_BUF_LEN - aecm->farBufReadPos;
-        memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
-               sizeof(WebRtc_Word16) * readLen);
-        aecm->farBufReadPos = 0;
-        readPos = readLen;
-        readLen = farLen - readLen;
-    }
-    memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
-           sizeof(WebRtc_Word16) * readLen);
-    aecm->farBufReadPos += readLen;
-}
diff --git a/src/modules/audio_processing/aecm/main/source/aecm_core.h b/src/modules/audio_processing/aecm/main/source/aecm_core.h
deleted file mode 100644
index 5defbe4..0000000
--- a/src/modules/audio_processing/aecm/main/source/aecm_core.h
+++ /dev/null
@@ -1,338 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Performs echo control (suppression) with fft routines in fixed-point
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_SOURCE_AECM_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_SOURCE_AECM_CORE_H_
-
-#define AECM_DYNAMIC_Q // turn on/off dynamic Q-domain
-//#define AECM_WITH_ABS_APPROX
-//#define AECM_SHORT                // for 32 sample partition length (otherwise 64)
-
-// TODO(bjornv): These defines will be removed in final version.
-//#define STORE_CHANNEL_DATA
-//#define VAD_DATA
-
-#include "typedefs.h"
-#include "signal_processing_library.h"
-// TODO(bjornv): Will be removed in final version.
-//#include <stdio.h>
-
-// Algorithm parameters
-
-#define FRAME_LEN       80              // Total frame length, 10 ms
-#ifdef AECM_SHORT
-
-#define PART_LEN        32              // Length of partition
-#define PART_LEN_SHIFT  6               // Length of (PART_LEN * 2) in base 2
-
-#else
-
-#define PART_LEN        64              // Length of partition
-#define PART_LEN_SHIFT  7               // Length of (PART_LEN * 2) in base 2
-
-#endif
-
-#define PART_LEN1       (PART_LEN + 1)  // Unique fft coefficients
-#define PART_LEN2       (PART_LEN << 1) // Length of partition * 2
-#define PART_LEN4       (PART_LEN << 2) // Length of partition * 4
-#define FAR_BUF_LEN     PART_LEN4       // Length of buffers
-#define MAX_DELAY 100
-
-// Counter parameters
-#ifdef AECM_SHORT
-
-#define CONV_LEN        1024            // Convergence length used at startup
-#else
-
-#define CONV_LEN        512             // Convergence length used at startup
-#endif
-
-#define CONV_LEN2       (CONV_LEN << 1) // Convergence length * 2 used at startup
-// Energy parameters
-#define MAX_BUF_LEN     64              // History length of energy signals
-
-#define FAR_ENERGY_MIN  1025            // Lowest Far energy level: At least 2 in energy
-#define FAR_ENERGY_DIFF 929             // Allowed difference between max and min
-
-#define ENERGY_DEV_OFFSET       0       // The energy error offset in Q8
-#define ENERGY_DEV_TOL  400             // The energy estimation tolerance in Q8
-#define FAR_ENERGY_VAD_REGION   230     // Far VAD tolerance region
-// Stepsize parameters
-#define MU_MIN          10              // Min stepsize 2^-MU_MIN (far end energy dependent)
-#define MU_MAX          1               // Max stepsize 2^-MU_MAX (far end energy dependent)
-#define MU_DIFF         9               // MU_MIN - MU_MAX
-// Channel parameters
-#define MIN_MSE_COUNT   20              // Min number of consecutive blocks with enough far end
-                                        // energy to compare channel estimates
-#define MIN_MSE_DIFF    29              // The ratio between adapted and stored channel to
-                                        // accept a new storage (0.8 in Q-MSE_RESOLUTION)
-#define MSE_RESOLUTION  5               // MSE parameter resolution
-#define RESOLUTION_CHANNEL16    12      // W16 Channel in Q-RESOLUTION_CHANNEL16
-#define RESOLUTION_CHANNEL32    28      // W32 Channel in Q-RESOLUTION_CHANNEL
-#define CHANNEL_VAD     16              // Minimum energy in frequency band to update channel
-// Suppression gain parameters: SUPGAIN_ parameters in Q-(RESOLUTION_SUPGAIN)
-#define RESOLUTION_SUPGAIN      8       // Channel in Q-(RESOLUTION_SUPGAIN)
-#define SUPGAIN_DEFAULT (1 << RESOLUTION_SUPGAIN)   // Default suppression gain
-#define SUPGAIN_ERROR_PARAM_A   3072    // Estimation error parameter (Maximum gain) (8 in Q8)
-#define SUPGAIN_ERROR_PARAM_B   1536    // Estimation error parameter (Gain before going down)
-#define SUPGAIN_ERROR_PARAM_D   SUPGAIN_DEFAULT // Estimation error parameter
-                                                // (Should be the same as Default) (1 in Q8)
-#define SUPGAIN_EPC_DT  200             // = SUPGAIN_ERROR_PARAM_C * ENERGY_DEV_TOL
-// Defines for "check delay estimation"
-#define CORR_WIDTH      31              // Number of samples to correlate over.
-#define CORR_MAX        16              // Maximum correlation offset
-#define CORR_MAX_BUF    63
-#define CORR_DEV        4
-#define CORR_MAX_LEVEL  20
-#define CORR_MAX_LOW    4
-#define CORR_BUF_LEN    (CORR_MAX << 1) + 1
-// Note that CORR_WIDTH + 2*CORR_MAX <= MAX_BUF_LEN
-
-#define ONE_Q14         (1 << 14)
-
-// NLP defines
-#define NLP_COMP_LOW    3277            // 0.2 in Q14
-#define NLP_COMP_HIGH   ONE_Q14         // 1 in Q14
-
-typedef struct
-{
-    int farBufWritePos;
-    int farBufReadPos;
-    int knownDelay;
-    int lastKnownDelay;
-    int firstVAD; // Parameter to control poorly initialized channels
-
-    void *farFrameBuf;
-    void *nearNoisyFrameBuf;
-    void *nearCleanFrameBuf;
-    void *outFrameBuf;
-
-    WebRtc_Word16 xBuf[PART_LEN2]; // farend
-    WebRtc_Word16 dBufClean[PART_LEN2]; // nearend
-    WebRtc_Word16 dBufNoisy[PART_LEN2]; // nearend
-    WebRtc_Word16 outBuf[PART_LEN];
-
-    WebRtc_Word16 farBuf[FAR_BUF_LEN];
-
-    WebRtc_Word16 mult;
-    WebRtc_UWord32 seed;
-
-    // Delay estimation variables
-    WebRtc_UWord16 medianYlogspec[PART_LEN1];
-    WebRtc_UWord16 medianXlogspec[PART_LEN1];
-    WebRtc_UWord16 medianBCount[MAX_DELAY];
-    WebRtc_UWord16 xfaHistory[PART_LEN1][MAX_DELAY];
-    WebRtc_Word16 delHistoryPos;
-    WebRtc_UWord32 bxHistory[MAX_DELAY];
-    WebRtc_UWord16 currentDelay;
-    WebRtc_UWord16 previousDelay;
-    WebRtc_Word16 delayAdjust;
-
-    WebRtc_Word16 nlpFlag;
-    WebRtc_Word16 fixedDelay;
-
-    WebRtc_UWord32 totCount;
-
-    WebRtc_Word16 xfaQDomainBuf[MAX_DELAY];
-    WebRtc_Word16 dfaCleanQDomain;
-    WebRtc_Word16 dfaCleanQDomainOld;
-    WebRtc_Word16 dfaNoisyQDomain;
-    WebRtc_Word16 dfaNoisyQDomainOld;
-
-    WebRtc_Word16 nearLogEnergy[MAX_BUF_LEN];
-    WebRtc_Word16 farLogEnergy[MAX_BUF_LEN];
-    WebRtc_Word16 echoAdaptLogEnergy[MAX_BUF_LEN];
-    WebRtc_Word16 echoStoredLogEnergy[MAX_BUF_LEN];
-
-    WebRtc_Word16 channelAdapt16[PART_LEN1];
-    WebRtc_Word32 channelAdapt32[PART_LEN1];
-    WebRtc_Word16 channelStored[PART_LEN1];
-    WebRtc_Word32 echoFilt[PART_LEN1];
-    WebRtc_Word16 nearFilt[PART_LEN1];
-    WebRtc_Word32 noiseEst[PART_LEN1];
-    WebRtc_Word16 noiseEstQDomain[PART_LEN1];
-    WebRtc_Word16 noiseEstCtr;
-    WebRtc_Word16 cngMode;
-
-    WebRtc_Word32 mseAdaptOld;
-    WebRtc_Word32 mseStoredOld;
-    WebRtc_Word32 mseThreshold;
-
-    WebRtc_Word16 farEnergyMin;
-    WebRtc_Word16 farEnergyMax;
-    WebRtc_Word16 farEnergyMaxMin;
-    WebRtc_Word16 farEnergyVAD;
-    WebRtc_Word16 farEnergyMSE;
-    WebRtc_Word16 currentVADValue;
-    WebRtc_Word16 vadUpdateCount;
-
-    WebRtc_Word16 delayHistogram[MAX_DELAY];
-    WebRtc_Word16 delayVadCount;
-    WebRtc_Word16 maxDelayHistIdx;
-    WebRtc_Word16 lastMinPos;
-
-    WebRtc_Word16 startupState;
-    WebRtc_Word16 mseChannelCount;
-    WebRtc_Word16 delayCount;
-    WebRtc_Word16 newDelayCorrData;
-    WebRtc_Word16 lastDelayUpdateCount;
-    WebRtc_Word16 delayCorrelation[CORR_BUF_LEN];
-    WebRtc_Word16 supGain;
-    WebRtc_Word16 supGainOld;
-    WebRtc_Word16 delayOffsetFlag;
-
-    WebRtc_Word16 supGainErrParamA;
-    WebRtc_Word16 supGainErrParamD;
-    WebRtc_Word16 supGainErrParamDiffAB;
-    WebRtc_Word16 supGainErrParamDiffBD;
-
-    // TODO(bjornv): Will be removed after final version has been committed.
-#ifdef VAD_DATA
-    FILE *vad_file;
-    FILE *delay_file;
-    FILE *far_file;
-    FILE *far_cur_file;
-    FILE *far_min_file;
-    FILE *far_max_file;
-    FILE *far_vad_file;
-#endif
-
-    // TODO(bjornv): Will be removed after final version has been committed.
-#ifdef STORE_CHANNEL_DATA
-    FILE *channel_file;
-    FILE *channel_file_init;
-#endif
-
-#ifdef AEC_DEBUG
-    FILE *farFile;
-    FILE *nearFile;
-    FILE *outFile;
-#endif
-} AecmCore_t;
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_CreateCore(...)
-//
-// Allocates the memory needed by the AECM. The memory needs to be
-// initialized separately using the WebRtcAecm_InitCore() function.
-//
-// Input:
-//      - aecm          : Instance that should be created
-//
-// Output:
-//      - aecm          : Created instance
-//
-// Return value         :  0 - Ok
-//                        -1 - Error
-//
-int WebRtcAecm_CreateCore(AecmCore_t **aecm);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_InitCore(...)
-//
-// This function initializes the AECM instant created with WebRtcAecm_CreateCore(...)
-// Input:
-//      - aecm          : Pointer to the AECM instance
-//      - samplingFreq  : Sampling Frequency
-//
-// Output:
-//      - aecm          : Initialized instance
-//
-// Return value         :  0 - Ok
-//                        -1 - Error
-//
-int WebRtcAecm_InitCore(AecmCore_t * const aecm, int samplingFreq);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_FreeCore(...)
-//
-// This function releases the memory allocated by WebRtcAecm_CreateCore()
-// Input:
-//      - aecm          : Pointer to the AECM instance
-//
-// Return value         :  0 - Ok
-//                        -1 - Error
-//           11001-11016: Error
-//
-int WebRtcAecm_FreeCore(AecmCore_t *aecm);
-
-int WebRtcAecm_Control(AecmCore_t *aecm, int delay, int nlpFlag, int delayOffsetFlag);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_ProcessFrame(...)
-//
-// This function processes frames and sends blocks to WebRtcAecm_ProcessBlock(...)
-//
-// Inputs:
-//      - aecm          : Pointer to the AECM instance
-//      - farend        : In buffer containing one frame of echo signal
-//      - nearendNoisy  : In buffer containing one frame of nearend+echo signal without NS
-//      - nearendClean  : In buffer containing one frame of nearend+echo signal with NS
-//
-// Output:
-//      - out           : Out buffer, one frame of nearend signal          :
-//
-//
-void WebRtcAecm_ProcessFrame(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
-                             const WebRtc_Word16 * const nearendNoisy,
-                             const WebRtc_Word16 * const nearendClean,
-                             WebRtc_Word16 * const out);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_ProcessBlock(...)
-//
-// This function is called for every block within one frame
-// This function is called by WebRtcAecm_ProcessFrame(...)
-//
-// Inputs:
-//      - aecm          : Pointer to the AECM instance
-//      - farend        : In buffer containing one block of echo signal
-//      - nearendNoisy  : In buffer containing one frame of nearend+echo signal without NS
-//      - nearendClean  : In buffer containing one frame of nearend+echo signal with NS
-//
-// Output:
-//      - out           : Out buffer, one block of nearend signal          :
-//
-//
-void WebRtcAecm_ProcessBlock(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
-                                const WebRtc_Word16 * const nearendNoisy,
-                                const WebRtc_Word16 * const noisyClean,
-                                WebRtc_Word16 * const out);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_BufferFarFrame()
-//
-// Inserts a frame of data into farend buffer.
-//
-// Inputs:
-//      - aecm          : Pointer to the AECM instance
-//      - farend        : In buffer containing one frame of farend signal
-//      - farLen        : Length of frame
-//
-void WebRtcAecm_BufferFarFrame(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
-                               const int farLen);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_FetchFarFrame()
-//
-// Read the farend buffer to account for known delay
-//
-// Inputs:
-//      - aecm          : Pointer to the AECM instance
-//      - farend        : In buffer containing one frame of farend signal
-//      - farLen        : Length of frame
-//      - knownDelay    : known delay
-//
-void WebRtcAecm_FetchFarFrame(AecmCore_t * const aecm, WebRtc_Word16 * const farend,
-                              const int farLen, const int knownDelay);
-
-#endif
diff --git a/src/modules/audio_processing/aecm/main/source/echo_control_mobile.c b/src/modules/audio_processing/aecm/main/source/echo_control_mobile.c
deleted file mode 100644
index f9d84f0..0000000
--- a/src/modules/audio_processing/aecm/main/source/echo_control_mobile.c
+++ /dev/null
@@ -1,733 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-//#include <string.h>
-
-#include "echo_control_mobile.h"
-#include "aecm_core.h"
-#include "ring_buffer.h"
-#ifdef AEC_DEBUG
-#include <stdio.h>
-#endif
-#ifdef MAC_IPHONE_PRINT
-#include <time.h>
-#include <stdio.h>
-#elif defined ARM_WINM_LOG
-#include "windows.h"
-extern HANDLE logFile;
-#endif
-
-#define BUF_SIZE_FRAMES 50 // buffer size (frames)
-// Maximum length of resampled signal. Must be an integer multiple of frames
-// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
-// The factor of 2 handles wb, and the + 1 is as a safety margin
-#define MAX_RESAMP_LEN (5 * FRAME_LEN)
-
-static const int kBufSizeSamp = BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
-static const int kSampMsNb = 8; // samples per ms in nb
-// Target suppression levels for nlp modes
-// log{0.001, 0.00001, 0.00000001}
-static const int kInitCheck = 42;
-
-typedef struct
-{
-    int sampFreq;
-    int scSampFreq;
-    short bufSizeStart;
-    int knownDelay;
-
-    // Stores the last frame added to the farend buffer
-    short farendOld[2][FRAME_LEN];
-    short initFlag; // indicates if AEC has been initialized
-
-    // Variables used for averaging far end buffer size
-    short counter;
-    short sum;
-    short firstVal;
-    short checkBufSizeCtr;
-
-    // Variables used for delay shifts
-    short msInSndCardBuf;
-    short filtDelay;
-    int timeForDelayChange;
-    int ECstartup;
-    int checkBuffSize;
-    int delayChange;
-    short lastDelayDiff;
-
-    WebRtc_Word16 echoMode;
-
-#ifdef AEC_DEBUG
-    FILE *bufFile;
-    FILE *delayFile;
-    FILE *preCompFile;
-    FILE *postCompFile;
-#endif // AEC_DEBUG
-    // Structures
-    void *farendBuf;
-
-    int lastError;
-
-    AecmCore_t *aecmCore;
-} aecmob_t;
-
-// Estimates delay to set the position of the farend buffer read pointer
-// (controlled by knownDelay)
-static int WebRtcAecm_EstBufDelay(aecmob_t *aecmInst, short msInSndCardBuf);
-
-// Stuffs the farend buffer if the estimated delay is too large
-static int WebRtcAecm_DelayComp(aecmob_t *aecmInst);
-
-WebRtc_Word32 WebRtcAecm_Create(void **aecmInst)
-{
-    aecmob_t *aecm;
-    if (aecmInst == NULL)
-    {
-        return -1;
-    }
-
-    aecm = malloc(sizeof(aecmob_t));
-    *aecmInst = aecm;
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (WebRtcAecm_CreateCore(&aecm->aecmCore) == -1)
-    {
-        WebRtcAecm_Free(aecm);
-        aecm = NULL;
-        return -1;
-    }
-
-    if (WebRtcApm_CreateBuffer(&aecm->farendBuf, kBufSizeSamp) == -1)
-    {
-        WebRtcAecm_Free(aecm);
-        aecm = NULL;
-        return -1;
-    }
-
-    aecm->initFlag = 0;
-    aecm->lastError = 0;
-
-#ifdef AEC_DEBUG
-    aecm->aecmCore->farFile = fopen("aecFar.pcm","wb");
-    aecm->aecmCore->nearFile = fopen("aecNear.pcm","wb");
-    aecm->aecmCore->outFile = fopen("aecOut.pcm","wb");
-    //aecm->aecmCore->outLpFile = fopen("aecOutLp.pcm","wb");
-
-    aecm->bufFile = fopen("aecBuf.dat", "wb");
-    aecm->delayFile = fopen("aecDelay.dat", "wb");
-    aecm->preCompFile = fopen("preComp.pcm", "wb");
-    aecm->postCompFile = fopen("postComp.pcm", "wb");
-#endif // AEC_DEBUG
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_Free(void *aecmInst)
-{
-    aecmob_t *aecm = aecmInst;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-#ifdef AEC_DEBUG
-    fclose(aecm->aecmCore->farFile);
-    fclose(aecm->aecmCore->nearFile);
-    fclose(aecm->aecmCore->outFile);
-    //fclose(aecm->aecmCore->outLpFile);
-
-    fclose(aecm->bufFile);
-    fclose(aecm->delayFile);
-    fclose(aecm->preCompFile);
-    fclose(aecm->postCompFile);
-#endif // AEC_DEBUG
-    WebRtcAecm_FreeCore(aecm->aecmCore);
-    WebRtcApm_FreeBuffer(aecm->farendBuf);
-    free(aecm);
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_Init(void *aecmInst, WebRtc_Word32 sampFreq, WebRtc_Word32 scSampFreq)
-{
-    aecmob_t *aecm = aecmInst;
-    AecmConfig aecConfig;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (sampFreq != 8000 && sampFreq != 16000)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecm->sampFreq = sampFreq;
-
-    if (scSampFreq < 1 || scSampFreq > 96000)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecm->scSampFreq = scSampFreq;
-
-    // Initialize AECM core
-    if (WebRtcAecm_InitCore(aecm->aecmCore, aecm->sampFreq) == -1)
-    {
-        aecm->lastError = AECM_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    // Initialize farend buffer
-    if (WebRtcApm_InitBuffer(aecm->farendBuf) == -1)
-    {
-        aecm->lastError = AECM_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    aecm->initFlag = kInitCheck; // indicates that initialization has been done
-
-    aecm->delayChange = 1;
-
-    aecm->sum = 0;
-    aecm->counter = 0;
-    aecm->checkBuffSize = 1;
-    aecm->firstVal = 0;
-
-    aecm->ECstartup = 1;
-    aecm->bufSizeStart = 0;
-    aecm->checkBufSizeCtr = 0;
-    aecm->filtDelay = 0;
-    aecm->timeForDelayChange = 0;
-    aecm->knownDelay = 0;
-    aecm->lastDelayDiff = 0;
-
-    memset(&aecm->farendOld[0][0], 0, 160);
-
-    // Default settings.
-    aecConfig.cngMode = AecmTrue;
-    aecConfig.echoMode = 3;
-
-    if (WebRtcAecm_set_config(aecm, aecConfig) == -1)
-    {
-        aecm->lastError = AECM_UNSPECIFIED_ERROR;
-        return -1;
-    }
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_BufferFarend(void *aecmInst, const WebRtc_Word16 *farend,
-                                      WebRtc_Word16 nrOfSamples)
-{
-    aecmob_t *aecm = aecmInst;
-    WebRtc_Word32 retVal = 0;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (farend == NULL)
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    if (nrOfSamples != 80 && nrOfSamples != 160)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-
-    // TODO: Is this really a good idea?
-    if (!aecm->ECstartup)
-    {
-        WebRtcAecm_DelayComp(aecm);
-    }
-
-    WebRtcApm_WriteBuffer(aecm->farendBuf, farend, nrOfSamples);
-
-    return retVal;
-}
-
-WebRtc_Word32 WebRtcAecm_Process(void *aecmInst, const WebRtc_Word16 *nearendNoisy,
-                                 const WebRtc_Word16 *nearendClean, WebRtc_Word16 *out,
-                                 WebRtc_Word16 nrOfSamples, WebRtc_Word16 msInSndCardBuf)
-{
-    aecmob_t *aecm = aecmInst;
-    WebRtc_Word32 retVal = 0;
-    short i;
-    short farend[FRAME_LEN];
-    short nmbrOfFilledBuffers;
-    short nBlocks10ms;
-    short nFrames;
-#ifdef AEC_DEBUG
-    short msInAECBuf;
-#endif
-
-#ifdef ARM_WINM_LOG
-    __int64 freq, start, end, diff;
-    unsigned int milliseconds;
-    DWORD temp;
-#elif defined MAC_IPHONE_PRINT
-    //       double endtime = 0, starttime = 0;
-    struct timeval starttime;
-    struct timeval endtime;
-    static long int timeused = 0;
-    static int timecount = 0;
-#endif
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (nearendNoisy == NULL)
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (out == NULL)
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    if (nrOfSamples != 80 && nrOfSamples != 160)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-
-    if (msInSndCardBuf < 0)
-    {
-        msInSndCardBuf = 0;
-        aecm->lastError = AECM_BAD_PARAMETER_WARNING;
-        retVal = -1;
-    } else if (msInSndCardBuf > 500)
-    {
-        msInSndCardBuf = 500;
-        aecm->lastError = AECM_BAD_PARAMETER_WARNING;
-        retVal = -1;
-    }
-    msInSndCardBuf += 10;
-    aecm->msInSndCardBuf = msInSndCardBuf;
-
-    nFrames = nrOfSamples / FRAME_LEN;
-    nBlocks10ms = nFrames / aecm->aecmCore->mult;
-
-    if (aecm->ECstartup)
-    {
-        if (nearendClean == NULL)
-        {
-            memcpy(out, nearendNoisy, sizeof(short) * nrOfSamples);
-        } else
-        {
-            memcpy(out, nearendClean, sizeof(short) * nrOfSamples);
-        }
-
-        nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecm->farendBuf) / FRAME_LEN;
-        // The AECM is in the start up mode
-        // AECM is disabled until the soundcard buffer and farend buffers are OK
-
-        // Mechanism to ensure that the soundcard buffer is reasonably stable.
-        if (aecm->checkBuffSize)
-        {
-            aecm->checkBufSizeCtr++;
-            // Before we fill up the far end buffer we require the amount of data on the
-            // sound card to be stable (+/-8 ms) compared to the first value. This
-            // comparison is made during the following 4 consecutive frames. If it seems
-            // to be stable then we start to fill up the far end buffer.
-
-            if (aecm->counter == 0)
-            {
-                aecm->firstVal = aecm->msInSndCardBuf;
-                aecm->sum = 0;
-            }
-
-            if (abs(aecm->firstVal - aecm->msInSndCardBuf)
-                    < WEBRTC_SPL_MAX(0.2 * aecm->msInSndCardBuf, kSampMsNb))
-            {
-                aecm->sum += aecm->msInSndCardBuf;
-                aecm->counter++;
-            } else
-            {
-                aecm->counter = 0;
-            }
-
-            if (aecm->counter * nBlocks10ms >= 6)
-            {
-                // The farend buffer size is determined in blocks of 80 samples
-                // Use 75% of the average value of the soundcard buffer
-                aecm->bufSizeStart
-                        = WEBRTC_SPL_MIN((3 * aecm->sum
-                                        * aecm->aecmCore->mult) / (aecm->counter * 40), BUF_SIZE_FRAMES);
-                // buffersize has now been determined
-                aecm->checkBuffSize = 0;
-            }
-
-            if (aecm->checkBufSizeCtr * nBlocks10ms > 50)
-            {
-                // for really bad sound cards, don't disable echocanceller for more than 0.5 sec
-                aecm->bufSizeStart = WEBRTC_SPL_MIN((3 * aecm->msInSndCardBuf
-                                * aecm->aecmCore->mult) / 40, BUF_SIZE_FRAMES);
-                aecm->checkBuffSize = 0;
-            }
-        }
-
-        // if checkBuffSize changed in the if-statement above
-        if (!aecm->checkBuffSize)
-        {
-            // soundcard buffer is now reasonably stable
-            // When the far end buffer is filled with approximately the same amount of
-            // data as the amount on the sound card we end the start up phase and start
-            // to cancel echoes.
-
-            if (nmbrOfFilledBuffers == aecm->bufSizeStart)
-            {
-                aecm->ECstartup = 0; // Enable the AECM
-            } else if (nmbrOfFilledBuffers > aecm->bufSizeStart)
-            {
-                WebRtcApm_FlushBuffer(
-                                       aecm->farendBuf,
-                                       WebRtcApm_get_buffer_size(aecm->farendBuf)
-                                               - aecm->bufSizeStart * FRAME_LEN);
-                aecm->ECstartup = 0;
-            }
-        }
-
-    } else
-    {
-        // AECM is enabled
-
-        // Note only 1 block supported for nb and 2 blocks for wb
-        for (i = 0; i < nFrames; i++)
-        {
-            nmbrOfFilledBuffers = WebRtcApm_get_buffer_size(aecm->farendBuf) / FRAME_LEN;
-
-            // Check that there is data in the far end buffer
-            if (nmbrOfFilledBuffers > 0)
-            {
-                // Get the next 80 samples from the farend buffer
-                WebRtcApm_ReadBuffer(aecm->farendBuf, farend, FRAME_LEN);
-
-                // Always store the last frame for use when we run out of data
-                memcpy(&(aecm->farendOld[i][0]), farend, FRAME_LEN * sizeof(short));
-            } else
-            {
-                // We have no data so we use the last played frame
-                memcpy(farend, &(aecm->farendOld[i][0]), FRAME_LEN * sizeof(short));
-            }
-
-            // Call buffer delay estimator when all data is extracted,
-            // i,e. i = 0 for NB and i = 1 for WB
-            if ((i == 0 && aecm->sampFreq == 8000) || (i == 1 && aecm->sampFreq == 16000))
-            {
-                WebRtcAecm_EstBufDelay(aecm, aecm->msInSndCardBuf);
-            }
-
-#ifdef ARM_WINM_LOG
-            // measure tick start
-            QueryPerformanceFrequency((LARGE_INTEGER*)&freq);
-            QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#elif defined MAC_IPHONE_PRINT
-            //            starttime = clock()/(double)CLOCKS_PER_SEC;
-            gettimeofday(&starttime, NULL);
-#endif
-            // Call the AECM
-            /*WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearend[FRAME_LEN * i],
-             &out[FRAME_LEN * i], aecm->knownDelay);*/
-            if (nearendClean == NULL)
-            {
-                WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearendNoisy[FRAME_LEN * i],
-                                        NULL, &out[FRAME_LEN * i]);
-            } else
-            {
-                WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearendNoisy[FRAME_LEN * i],
-                                        &nearendClean[FRAME_LEN * i], &out[FRAME_LEN * i]);
-            }
-
-#ifdef ARM_WINM_LOG
-
-            // measure tick end
-            QueryPerformanceCounter((LARGE_INTEGER*)&end);
-
-            if(end > start)
-            {
-                diff = ((end - start) * 1000) / (freq/1000);
-                milliseconds = (unsigned int)(diff & 0xffffffff);
-                WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-            }
-#elif defined MAC_IPHONE_PRINT
-            //            endtime = clock()/(double)CLOCKS_PER_SEC;
-            //            printf("%f\n", endtime - starttime);
-
-            gettimeofday(&endtime, NULL);
-
-            if( endtime.tv_usec > starttime.tv_usec)
-            {
-                timeused += endtime.tv_usec - starttime.tv_usec;
-            } else
-            {
-                timeused += endtime.tv_usec + 1000000 - starttime.tv_usec;
-            }
-
-            if(++timecount == 1000)
-            {
-                timecount = 0;
-                printf("AEC: %ld\n", timeused);
-                timeused = 0;
-            }
-#endif
-
-        }
-    }
-
-#ifdef AEC_DEBUG
-    msInAECBuf = WebRtcApm_get_buffer_size(aecm->farendBuf) / (kSampMsNb*aecm->aecmCore->mult);
-    fwrite(&msInAECBuf, 2, 1, aecm->bufFile);
-    fwrite(&(aecm->knownDelay), sizeof(aecm->knownDelay), 1, aecm->delayFile);
-#endif
-
-    return retVal;
-}
-
-WebRtc_Word32 WebRtcAecm_set_config(void *aecmInst, AecmConfig config)
-{
-    aecmob_t *aecm = aecmInst;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    if (config.cngMode != AecmFalse && config.cngMode != AecmTrue)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecm->aecmCore->cngMode = config.cngMode;
-
-    if (config.echoMode < 0 || config.echoMode > 4)
-    {
-        aecm->lastError = AECM_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    aecm->echoMode = config.echoMode;
-
-    if (aecm->echoMode == 0)
-    {
-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 3;
-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 3;
-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 3;
-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 3;
-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 3)
-                - (SUPGAIN_ERROR_PARAM_B >> 3);
-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 3)
-                - (SUPGAIN_ERROR_PARAM_D >> 3);
-    } else if (aecm->echoMode == 1)
-    {
-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 2;
-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 2;
-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 2;
-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 2;
-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 2)
-                - (SUPGAIN_ERROR_PARAM_B >> 2);
-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 2)
-                - (SUPGAIN_ERROR_PARAM_D >> 2);
-    } else if (aecm->echoMode == 2)
-    {
-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 1;
-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 1;
-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 1;
-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 1;
-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 1)
-                - (SUPGAIN_ERROR_PARAM_B >> 1);
-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 1)
-                - (SUPGAIN_ERROR_PARAM_D >> 1);
-    } else if (aecm->echoMode == 3)
-    {
-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT;
-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT;
-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
-        aecm->aecmCore->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
-        aecm->aecmCore->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
-    } else if (aecm->echoMode == 4)
-    {
-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT << 1;
-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT << 1;
-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A << 1;
-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D << 1;
-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A << 1)
-                - (SUPGAIN_ERROR_PARAM_B << 1);
-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B << 1)
-                - (SUPGAIN_ERROR_PARAM_D << 1);
-    }
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_get_config(void *aecmInst, AecmConfig *config)
-{
-    aecmob_t *aecm = aecmInst;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    if (config == NULL)
-    {
-        aecm->lastError = AECM_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (aecm->initFlag != kInitCheck)
-    {
-        aecm->lastError = AECM_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    config->cngMode = aecm->aecmCore->cngMode;
-    config->echoMode = aecm->echoMode;
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_get_version(WebRtc_Word8 *versionStr, WebRtc_Word16 len)
-{
-    const char version[] = "AECM 1.2.0";
-    const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
-
-    if (versionStr == NULL)
-    {
-        return -1;
-    }
-
-    if (versionLen > len)
-    {
-        return -1;
-    }
-
-    strncpy(versionStr, version, versionLen);
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_get_error_code(void *aecmInst)
-{
-    aecmob_t *aecm = aecmInst;
-
-    if (aecm == NULL)
-    {
-        return -1;
-    }
-
-    return aecm->lastError;
-}
-
-static int WebRtcAecm_EstBufDelay(aecmob_t *aecm, short msInSndCardBuf)
-{
-    short delayNew, nSampFar, nSampSndCard;
-    short diff;
-
-    nSampFar = WebRtcApm_get_buffer_size(aecm->farendBuf);
-    nSampSndCard = msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
-
-    delayNew = nSampSndCard - nSampFar;
-
-    if (delayNew < FRAME_LEN)
-    {
-        WebRtcApm_FlushBuffer(aecm->farendBuf, FRAME_LEN);
-        delayNew += FRAME_LEN;
-    }
-
-    aecm->filtDelay = WEBRTC_SPL_MAX(0, (8 * aecm->filtDelay + 2 * delayNew) / 10);
-
-    diff = aecm->filtDelay - aecm->knownDelay;
-    if (diff > 224)
-    {
-        if (aecm->lastDelayDiff < 96)
-        {
-            aecm->timeForDelayChange = 0;
-        } else
-        {
-            aecm->timeForDelayChange++;
-        }
-    } else if (diff < 96 && aecm->knownDelay > 0)
-    {
-        if (aecm->lastDelayDiff > 224)
-        {
-            aecm->timeForDelayChange = 0;
-        } else
-        {
-            aecm->timeForDelayChange++;
-        }
-    } else
-    {
-        aecm->timeForDelayChange = 0;
-    }
-    aecm->lastDelayDiff = diff;
-
-    if (aecm->timeForDelayChange > 25)
-    {
-        aecm->knownDelay = WEBRTC_SPL_MAX((int)aecm->filtDelay - 160, 0);
-    }
-    return 0;
-}
-
-static int WebRtcAecm_DelayComp(aecmob_t *aecm)
-{
-    int nSampFar, nSampSndCard, delayNew, nSampAdd;
-    const int maxStuffSamp = 10 * FRAME_LEN;
-
-    nSampFar = WebRtcApm_get_buffer_size(aecm->farendBuf);
-    nSampSndCard = aecm->msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
-    delayNew = nSampSndCard - nSampFar;
-
-    if (delayNew > FAR_BUF_LEN - FRAME_LEN * aecm->aecmCore->mult)
-    {
-        // The difference of the buffer sizes is larger than the maximum
-        // allowed known delay. Compensate by stuffing the buffer.
-        nSampAdd = (int)(WEBRTC_SPL_MAX(((nSampSndCard >> 1) - nSampFar),
-                FRAME_LEN));
-        nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
-
-        WebRtcApm_StuffBuffer(aecm->farendBuf, nSampAdd);
-        aecm->delayChange = 1; // the delay needs to be updated
-    }
-
-    return 0;
-}
diff --git a/src/modules/audio_processing/agc/main/interface/gain_control.h b/src/modules/audio_processing/agc/main/interface/gain_control.h
deleted file mode 100644
index 2893331..0000000
--- a/src/modules/audio_processing/agc/main/interface/gain_control.h
+++ /dev/null
@@ -1,273 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_INTERFACE_GAIN_CONTROL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_INTERFACE_GAIN_CONTROL_H_
-
-#include "typedefs.h"
-
-// Errors
-#define AGC_UNSPECIFIED_ERROR           18000
-#define AGC_UNSUPPORTED_FUNCTION_ERROR  18001
-#define AGC_UNINITIALIZED_ERROR         18002
-#define AGC_NULL_POINTER_ERROR          18003
-#define AGC_BAD_PARAMETER_ERROR         18004
-
-// Warnings
-#define AGC_BAD_PARAMETER_WARNING       18050
-
-enum
-{
-    kAgcModeUnchanged,
-    kAgcModeAdaptiveAnalog,
-    kAgcModeAdaptiveDigital,
-    kAgcModeFixedDigital
-};
-
-enum
-{
-    kAgcFalse = 0,
-    kAgcTrue
-};
-
-typedef struct
-{
-    WebRtc_Word16 targetLevelDbfs;   // default 3 (-3 dBOv)
-    WebRtc_Word16 compressionGaindB; // default 9 dB
-    WebRtc_UWord8 limiterEnable;     // default kAgcTrue (on)
-} WebRtcAgc_config_t;
-
-#if defined(__cplusplus)
-extern "C"
-{
-#endif
-
-/*
- * This function processes a 10/20ms frame of far-end speech to determine
- * if there is active speech. Far-end speech length can be either 10ms or
- * 20ms. The length of the input speech vector must be given in samples
- * (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000).
- *
- * Input:
- *      - agcInst           : AGC instance.
- *      - inFar             : Far-end input speech vector (10 or 20ms)
- *      - samples           : Number of samples in input vector
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_AddFarend(void* agcInst,
-                        const WebRtc_Word16* inFar,
-                        WebRtc_Word16 samples);
-
-/*
- * This function processes a 10/20ms frame of microphone speech to determine
- * if there is active speech. Microphone speech length can be either 10ms or
- * 20ms. The length of the input speech vector must be given in samples
- * (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000). For very low
- * input levels, the input signal is increased in level by multiplying and
- * overwriting the samples in inMic[].
- *
- * This function should be called before any further processing of the
- * near-end microphone signal.
- *
- * Input:
- *      - agcInst           : AGC instance.
- *      - inMic             : Microphone input speech vector (10 or 20 ms) for
- *                            L band
- *      - inMic_H           : Microphone input speech vector (10 or 20 ms) for
- *                            H band
- *      - samples           : Number of samples in input vector
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_AddMic(void* agcInst,
-                     WebRtc_Word16* inMic,
-                     WebRtc_Word16* inMic_H,
-                     WebRtc_Word16 samples);
-
-/*
- * This function replaces the analog microphone with a virtual one.
- * It is a digital gain applied to the input signal and is used in the
- * agcAdaptiveDigital mode where no microphone level is adjustable.
- * Microphone speech length can be either 10ms or 20ms. The length of the
- * input speech vector must be given in samples (80/160 when FS=8000, and
- * 160/320 when FS=16000 or FS=32000).
- *
- * Input:
- *      - agcInst           : AGC instance.
- *      - inMic             : Microphone input speech vector for (10 or 20 ms)
- *                            L band
- *      - inMic_H           : Microphone input speech vector for (10 or 20 ms)
- *                            H band
- *      - samples           : Number of samples in input vector
- *      - micLevelIn        : Input level of microphone (static)
- *
- * Output:
- *      - inMic             : Microphone output after processing (L band)
- *      - inMic_H           : Microphone output after processing (H band)
- *      - micLevelOut       : Adjusted microphone level after processing
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_VirtualMic(void* agcInst,
-                         WebRtc_Word16* inMic,
-                         WebRtc_Word16* inMic_H,
-                         WebRtc_Word16 samples,
-                         WebRtc_Word32 micLevelIn,
-                         WebRtc_Word32* micLevelOut);
-
-/*
- * This function processes a 10/20ms frame and adjusts (normalizes) the gain
- * both analog and digitally. The gain adjustments are done only during
- * active periods of speech. The input speech length can be either 10ms or
- * 20ms and the output is of the same length. The length of the speech
- * vectors must be given in samples (80/160 when FS=8000, and 160/320 when
- * FS=16000 or FS=32000). The echo parameter can be used to ensure the AGC will
- * not adjust upward in the presence of echo.
- *
- * This function should be called after processing the near-end microphone
- * signal, in any case after any echo cancellation.
- *
- * Input:
- *      - agcInst           : AGC instance
- *      - inNear            : Near-end input speech vector (10 or 20 ms) for
- *                            L band
- *      - inNear_H          : Near-end input speech vector (10 or 20 ms) for
- *                            H band
- *      - samples           : Number of samples in input/output vector
- *      - inMicLevel        : Current microphone volume level
- *      - echo              : Set to 0 if the signal passed to add_mic is
- *                            almost certainly free of echo; otherwise set
- *                            to 1. If you have no information regarding echo
- *                            set to 0.
- *
- * Output:
- *      - outMicLevel       : Adjusted microphone volume level
- *      - out               : Gain-adjusted near-end speech vector (L band)
- *                          : May be the same vector as the input.
- *      - out_H             : Gain-adjusted near-end speech vector (H band)
- *      - saturationWarning : A returned value of 1 indicates a saturation event
- *                            has occurred and the volume cannot be further
- *                            reduced. Otherwise will be set to 0.
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_Process(void* agcInst,
-                      const WebRtc_Word16* inNear,
-                      const WebRtc_Word16* inNear_H,
-                      WebRtc_Word16 samples,
-                      WebRtc_Word16* out,
-                      WebRtc_Word16* out_H,
-                      WebRtc_Word32 inMicLevel,
-                      WebRtc_Word32* outMicLevel,
-                      WebRtc_Word16 echo,
-                      WebRtc_UWord8* saturationWarning);
-
-/*
- * This function sets the config parameters (targetLevelDbfs,
- * compressionGaindB and limiterEnable).
- *
- * Input:
- *      - agcInst           : AGC instance
- *      - config            : config struct
- *
- * Output:
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_set_config(void* agcInst, WebRtcAgc_config_t config);
-
-/*
- * This function returns the config parameters (targetLevelDbfs,
- * compressionGaindB and limiterEnable).
- *
- * Input:
- *      - agcInst           : AGC instance
- *
- * Output:
- *      - config            : config struct
- *
- * Return value:
- *                          :  0 - Normal operation.
- *                          : -1 - Error
- */
-int WebRtcAgc_get_config(void* agcInst, WebRtcAgc_config_t* config);
-
-/*
- * This function creates an AGC instance, which will contain the state
- * information for one (duplex) channel.
- *
- * Return value             : AGC instance if successful
- *                          : 0 (i.e., a NULL pointer) if unsuccessful
- */
-int WebRtcAgc_Create(void **agcInst);
-
-/*
- * This function frees the AGC instance created at the beginning.
- *
- * Input:
- *      - agcInst           : AGC instance.
- *
- * Return value             :  0 - Ok
- *                            -1 - Error
- */
-int WebRtcAgc_Free(void *agcInst);
-
-/*
- * This function initializes an AGC instance.
- *
- * Input:
- *      - agcInst           : AGC instance.
- *      - minLevel          : Minimum possible mic level
- *      - maxLevel          : Maximum possible mic level
- *      - agcMode           : 0 - Unchanged
- *                          : 1 - Adaptive Analog Automatic Gain Control -3dBOv
- *                          : 2 - Adaptive Digital Automatic Gain Control -3dBOv
- *                          : 3 - Fixed Digital Gain 0dB
- *      - fs                : Sampling frequency
- *
- * Return value             :  0 - Ok
- *                            -1 - Error
- */
-int WebRtcAgc_Init(void *agcInst,
-                   WebRtc_Word32 minLevel,
-                   WebRtc_Word32 maxLevel,
-                   WebRtc_Word16 agcMode,
-                   WebRtc_UWord32 fs);
-
-/*
- * This function returns a text string containing the version.
- *
- * Input:
- *      - length            : Length of the char array pointed to by version
- * Output:
- *      - version           : Pointer to a char array of to which the version
- *                          : string will be copied.
- *
- * Return value             :  0 - OK
- *                            -1 - Error
- */
-int WebRtcAgc_Version(WebRtc_Word8 *versionStr, WebRtc_Word16 length);
-
-#if defined(__cplusplus)
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_INTERFACE_GAIN_CONTROL_H_
diff --git a/src/modules/audio_processing/agc/main/matlab/getGains.m b/src/modules/audio_processing/agc/main/matlab/getGains.m
deleted file mode 100644
index e0234b8..0000000
--- a/src/modules/audio_processing/agc/main/matlab/getGains.m
+++ /dev/null
@@ -1,32 +0,0 @@
-% Outputs a file for testing purposes. 
-%
-% Adjust the following parameters to suit. Their purpose becomes more clear on
-% viewing the gain plots.
-% MaxGain: Max gain in dB
-% MinGain: Min gain at overload (0 dBov) in dB
-% CompRatio: Compression ratio, essentially determines the slope of the gain
-%            function between the max and min gains
-% Knee: The smoothness of the transition to max gain (smaller is smoother)
-MaxGain = 5; MinGain = 0; CompRatio = 3; Knee = 1;
-
-% Compute gains
-zeros = 0:31; lvl = 2.^(1-zeros); 
-A = -10*log10(lvl) * (CompRatio - 1) / CompRatio;
-B = MaxGain - MinGain;
-gains = round(2^16*10.^(0.05 * (MinGain + B * ( log(exp(-Knee*A)+exp(-Knee*B)) - log(1+exp(-Knee*B)) ) / log(1/(1+exp(Knee*B))))));
-fprintf(1, '\t%i, %i, %i, %i,\n', gains);
-
-% Save gains to file
-fid = fopen('gains', 'wb');
-if fid == -1
-	error(sprintf('Unable to open file %s', filename));
-	return
-end
-fwrite(fid, gains, 'int32');
-fclose(fid);
-
-% Plotting
-in = 10*log10(lvl); out = 20*log10(gains/65536);
-subplot(121); plot(in, out); axis([-60, 0, -5, 30]); grid on; xlabel('Input (dB)'); ylabel('Gain (dB)');
-subplot(122); plot(in, in+out); axis([-60, 0, -60, 10]); grid on; xlabel('Input (dB)'); ylabel('Output (dB)');
-zoom on;
diff --git a/src/modules/audio_processing/agc/main/source/Android.mk b/src/modules/audio_processing/agc/main/source/Android.mk
deleted file mode 100644
index 2fd97bd..0000000
--- a/src/modules/audio_processing/agc/main/source/Android.mk
+++ /dev/null
@@ -1,46 +0,0 @@
-# This file is generated by gyp; do not edit. This means you!
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_agc
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := analog_agc.c \
-    digital_agc.c
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=  
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
-    '-DNO_HEAPCHECKER' \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_THREAD_RR' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' 
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../../.. \
-    $(LOCAL_PATH)/../interface \
-    $(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface 
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS := 
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-# Duplicate the static libraries to fix circular references
-LOCAL_STATIC_LIBRARIES += $(LOCAL_STATIC_LIBRARIES)
-
-LOCAL_SHARED_LIBRARIES := libcutils \
-    libdl \
-    libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/agc/main/source/agc.gyp b/src/modules/audio_processing/agc/main/source/agc.gyp
deleted file mode 100644
index e28a4c8..0000000
--- a/src/modules/audio_processing/agc/main/source/agc.gyp
+++ /dev/null
@@ -1,43 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../../../../common_settings.gypi', # Common settings
-  ],
-  'targets': [
-    {
-      'target_name': 'agc',
-      'type': '<(library)',
-      'dependencies': [
-        '../../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
-      ],
-      'include_dirs': [
-        '../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-        ],
-      },
-      'sources': [
-        '../interface/gain_control.h',
-        'analog_agc.c',
-        'analog_agc.h',
-        'digital_agc.c',
-        'digital_agc.h',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/agc/main/source/analog_agc.c b/src/modules/audio_processing/agc/main/source/analog_agc.c
deleted file mode 100644
index dfb7adc..0000000
--- a/src/modules/audio_processing/agc/main/source/analog_agc.c
+++ /dev/null
@@ -1,1700 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/* analog_agc.c
- *
- * Using a feedback system, determines an appropriate analog volume level
- * given an input signal and current volume level. Targets a conservative
- * signal level and is intended for use with a digital AGC to apply
- * additional gain.
- *
- */
-
-#include <assert.h>
-#include <stdlib.h>
-#ifdef AGC_DEBUG //test log
-#include <stdio.h>
-#endif
-#include "analog_agc.h"
-
-/* The slope of in Q13*/
-static const WebRtc_Word16 kSlope1[8] = {21793, 12517, 7189, 4129, 2372, 1362, 472, 78};
-
-/* The offset in Q14 */
-static const WebRtc_Word16 kOffset1[8] = {25395, 23911, 22206, 20737, 19612, 18805, 17951,
-        17367};
-
-/* The slope of in Q13*/
-static const WebRtc_Word16 kSlope2[8] = {2063, 1731, 1452, 1218, 1021, 857, 597, 337};
-
-/* The offset in Q14 */
-static const WebRtc_Word16 kOffset2[8] = {18432, 18379, 18290, 18177, 18052, 17920, 17670,
-        17286};
-
-static const WebRtc_Word16 kMuteGuardTimeMs = 8000;
-static const WebRtc_Word16 kInitCheck = 42;
-
-/* Default settings if config is not used */
-#define AGC_DEFAULT_TARGET_LEVEL 3
-#define AGC_DEFAULT_COMP_GAIN 9
-/* This is the target level for the analog part in ENV scale. To convert to RMS scale you
- * have to add OFFSET_ENV_TO_RMS.
- */
-#define ANALOG_TARGET_LEVEL 11
-#define ANALOG_TARGET_LEVEL_2 5 // ANALOG_TARGET_LEVEL / 2
-/* Offset between RMS scale (analog part) and ENV scale (digital part). This value actually
- * varies with the FIXED_ANALOG_TARGET_LEVEL, hence we should in the future replace it with
- * a table.
- */
-#define OFFSET_ENV_TO_RMS 9
-/* The reference input level at which the digital part gives an output of targetLevelDbfs
- * (desired level) if we have no compression gain. This level should be set high enough not
- * to compress the peaks due to the dynamics.
- */
-#define DIGITAL_REF_AT_0_COMP_GAIN 4
-/* Speed of reference level decrease.
- */
-#define DIFF_REF_TO_ANALOG 5
-
-#ifdef MIC_LEVEL_FEEDBACK
-#define NUM_BLOCKS_IN_SAT_BEFORE_CHANGE_TARGET 7
-#endif
-/* Size of analog gain table */
-#define GAIN_TBL_LEN 32
-/* Matlab code:
- * fprintf(1, '\t%i, %i, %i, %i,\n', round(10.^(linspace(0,10,32)/20) * 2^12));
- */
-/* Q12 */
-static const WebRtc_UWord16 kGainTableAnalog[GAIN_TBL_LEN] = {4096, 4251, 4412, 4579, 4752,
-        4932, 5118, 5312, 5513, 5722, 5938, 6163, 6396, 6638, 6889, 7150, 7420, 7701, 7992,
-        8295, 8609, 8934, 9273, 9623, 9987, 10365, 10758, 11165, 11587, 12025, 12480, 12953};
-
-/* Gain/Suppression tables for virtual Mic (in Q10) */
-static const WebRtc_UWord16 kGainTableVirtualMic[128] = {1052, 1081, 1110, 1141, 1172, 1204,
-        1237, 1271, 1305, 1341, 1378, 1416, 1454, 1494, 1535, 1577, 1620, 1664, 1710, 1757,
-        1805, 1854, 1905, 1957, 2010, 2065, 2122, 2180, 2239, 2301, 2364, 2428, 2495, 2563,
-        2633, 2705, 2779, 2855, 2933, 3013, 3096, 3180, 3267, 3357, 3449, 3543, 3640, 3739,
-        3842, 3947, 4055, 4166, 4280, 4397, 4517, 4640, 4767, 4898, 5032, 5169, 5311, 5456,
-        5605, 5758, 5916, 6078, 6244, 6415, 6590, 6770, 6956, 7146, 7341, 7542, 7748, 7960,
-        8178, 8402, 8631, 8867, 9110, 9359, 9615, 9878, 10148, 10426, 10711, 11004, 11305,
-        11614, 11932, 12258, 12593, 12938, 13292, 13655, 14029, 14412, 14807, 15212, 15628,
-        16055, 16494, 16945, 17409, 17885, 18374, 18877, 19393, 19923, 20468, 21028, 21603,
-        22194, 22801, 23425, 24065, 24724, 25400, 26095, 26808, 27541, 28295, 29069, 29864,
-        30681, 31520, 32382};
-static const WebRtc_UWord16 kSuppressionTableVirtualMic[128] = {1024, 1006, 988, 970, 952,
-        935, 918, 902, 886, 870, 854, 839, 824, 809, 794, 780, 766, 752, 739, 726, 713, 700,
-        687, 675, 663, 651, 639, 628, 616, 605, 594, 584, 573, 563, 553, 543, 533, 524, 514,
-        505, 496, 487, 478, 470, 461, 453, 445, 437, 429, 421, 414, 406, 399, 392, 385, 378,
-        371, 364, 358, 351, 345, 339, 333, 327, 321, 315, 309, 304, 298, 293, 288, 283, 278,
-        273, 268, 263, 258, 254, 249, 244, 240, 236, 232, 227, 223, 219, 215, 211, 208, 204,
-        200, 197, 193, 190, 186, 183, 180, 176, 173, 170, 167, 164, 161, 158, 155, 153, 150,
-        147, 145, 142, 139, 137, 134, 132, 130, 127, 125, 123, 121, 118, 116, 114, 112, 110,
-        108, 106, 104, 102};
-
-/* Table for target energy levels. Values in Q(-7)
- * Matlab code
- * targetLevelTable = fprintf('%d,\t%d,\t%d,\t%d,\n', round((32767*10.^(-(0:63)'/20)).^2*16/2^7) */
-
-static const WebRtc_Word32 kTargetLevelTable[64] = {134209536, 106606424, 84680493, 67264106,
-        53429779, 42440782, 33711911, 26778323, 21270778, 16895980, 13420954, 10660642,
-        8468049, 6726411, 5342978, 4244078, 3371191, 2677832, 2127078, 1689598, 1342095,
-        1066064, 846805, 672641, 534298, 424408, 337119, 267783, 212708, 168960, 134210,
-        106606, 84680, 67264, 53430, 42441, 33712, 26778, 21271, 16896, 13421, 10661, 8468,
-        6726, 5343, 4244, 3371, 2678, 2127, 1690, 1342, 1066, 847, 673, 534, 424, 337, 268,
-        213, 169, 134, 107, 85, 67};
-
-int WebRtcAgc_AddMic(void *state, WebRtc_Word16 *in_mic, WebRtc_Word16 *in_mic_H,
-                     WebRtc_Word16 samples)
-{
-    WebRtc_Word32 nrg, max_nrg, sample, tmp32;
-    WebRtc_Word32 *ptr;
-    WebRtc_UWord16 targetGainIdx, gain;
-    WebRtc_Word16 i, n, L, M, subFrames, tmp16, tmp_speech[16];
-    Agc_t *stt;
-    stt = (Agc_t *)state;
-
-    //default/initial values corresponding to 10ms for wb and swb
-    M = 10;
-    L = 16;
-    subFrames = 160;
-
-    if (stt->fs == 8000)
-    {
-        if (samples == 80)
-        {
-            subFrames = 80;
-            M = 10;
-            L = 8;
-        } else if (samples == 160)
-        {
-            subFrames = 80;
-            M = 20;
-            L = 8;
-        } else
-        {
-#ifdef AGC_DEBUG //test log
-            fprintf(stt->fpt,
-                    "AGC->add_mic, frame %d: Invalid number of samples\n\n",
-                    (stt->fcount + 1));
-#endif
-            return -1;
-        }
-    } else if (stt->fs == 16000)
-    {
-        if (samples == 160)
-        {
-            subFrames = 160;
-            M = 10;
-            L = 16;
-        } else if (samples == 320)
-        {
-            subFrames = 160;
-            M = 20;
-            L = 16;
-        } else
-        {
-#ifdef AGC_DEBUG //test log
-            fprintf(stt->fpt,
-                    "AGC->add_mic, frame %d: Invalid number of samples\n\n",
-                    (stt->fcount + 1));
-#endif
-            return -1;
-        }
-    } else if (stt->fs == 32000)
-    {
-        /* SWB is processed as 160 sample for L and H bands */
-        if (samples == 160)
-        {
-            subFrames = 160;
-            M = 10;
-            L = 16;
-        } else
-        {
-#ifdef AGC_DEBUG
-            fprintf(stt->fpt,
-                    "AGC->add_mic, frame %d: Invalid sample rate\n\n",
-                    (stt->fcount + 1));
-#endif
-            return -1;
-        }
-    }
-
-    /* Check for valid pointers based on sampling rate */
-    if ((stt->fs == 32000) && (in_mic_H == NULL))
-    {
-        return -1;
-    }
-    /* Check for valid pointer for low band */
-    if (in_mic == NULL)
-    {
-        return -1;
-    }
-
-    /* apply slowly varying digital gain */
-    if (stt->micVol > stt->maxAnalog)
-    {
-        /* Q1 */
-        tmp16 = (WebRtc_Word16)(stt->micVol - stt->maxAnalog);
-        tmp32 = WEBRTC_SPL_MUL_16_16(GAIN_TBL_LEN - 1, tmp16);
-        tmp16 = (WebRtc_Word16)(stt->maxLevel - stt->maxAnalog);
-        targetGainIdx = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32, tmp16);
-        assert(targetGainIdx < GAIN_TBL_LEN);
-
-        /* Increment through the table towards the target gain.
-         * If micVol drops below maxAnalog, we allow the gain
-         * to be dropped immediately. */
-        if (stt->gainTableIdx < targetGainIdx)
-        {
-            stt->gainTableIdx++;
-        } else if (stt->gainTableIdx > targetGainIdx)
-        {
-            stt->gainTableIdx--;
-        }
-
-        /* Q12 */
-        gain = kGainTableAnalog[stt->gainTableIdx];
-
-        for (i = 0; i < samples; i++)
-        {
-            // For lower band
-            tmp32 = WEBRTC_SPL_MUL_16_U16(in_mic[i], gain);
-            sample = WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
-            if (sample > 32767)
-            {
-                in_mic[i] = 32767;
-            } else if (sample < -32768)
-            {
-                in_mic[i] = -32768;
-            } else
-            {
-                in_mic[i] = (WebRtc_Word16)sample;
-            }
-
-            // For higher band
-            if (stt->fs == 32000)
-            {
-                tmp32 = WEBRTC_SPL_MUL_16_U16(in_mic_H[i], gain);
-                sample = WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
-                if (sample > 32767)
-                {
-                    in_mic_H[i] = 32767;
-                } else if (sample < -32768)
-                {
-                    in_mic_H[i] = -32768;
-                } else
-                {
-                    in_mic_H[i] = (WebRtc_Word16)sample;
-                }
-            }
-        }
-    } else
-    {
-        stt->gainTableIdx = 0;
-    }
-
-    /* compute envelope */
-    if ((M == 10) && (stt->inQueue > 0))
-    {
-        ptr = stt->env[1];
-    } else
-    {
-        ptr = stt->env[0];
-    }
-
-    for (i = 0; i < M; i++)
-    {
-        /* iterate over samples */
-        max_nrg = 0;
-        for (n = 0; n < L; n++)
-        {
-            nrg = WEBRTC_SPL_MUL_16_16(in_mic[i * L + n], in_mic[i * L + n]);
-            if (nrg > max_nrg)
-            {
-                max_nrg = nrg;
-            }
-        }
-        ptr[i] = max_nrg;
-    }
-
-    /* compute energy */
-    if ((M == 10) && (stt->inQueue > 0))
-    {
-        ptr = stt->Rxx16w32_array[1];
-    } else
-    {
-        ptr = stt->Rxx16w32_array[0];
-    }
-
-    for (i = 0; i < WEBRTC_SPL_RSHIFT_W16(M, 1); i++)
-    {
-        if (stt->fs == 16000)
-        {
-            WebRtcSpl_DownsampleBy2(&in_mic[i * 32], 32, tmp_speech, stt->filterState);
-        } else
-        {
-            memcpy(tmp_speech, &in_mic[i * 16], 16 * sizeof(short));
-        }
-        /* Compute energy in blocks of 16 samples */
-        ptr[i] = WebRtcSpl_DotProductWithScale(tmp_speech, tmp_speech, 16, 4);
-    }
-
-    /* update queue information */
-    if ((stt->inQueue == 0) && (M == 10))
-    {
-        stt->inQueue = 1;
-    } else
-    {
-        stt->inQueue = 2;
-    }
-
-    /* call VAD (use low band only) */
-    for (i = 0; i < samples; i += subFrames)
-    {
-        WebRtcAgc_ProcessVad(&stt->vadMic, &in_mic[i], subFrames);
-    }
-
-    return 0;
-}
-
-int WebRtcAgc_AddFarend(void *state, const WebRtc_Word16 *in_far, WebRtc_Word16 samples)
-{
-    WebRtc_Word32 errHandle = 0;
-    WebRtc_Word16 i, subFrames;
-    Agc_t *stt;
-    stt = (Agc_t *)state;
-
-    if (stt == NULL)
-    {
-        return -1;
-    }
-
-    if (stt->fs == 8000)
-    {
-        if ((samples != 80) && (samples != 160))
-        {
-#ifdef AGC_DEBUG //test log
-            fprintf(stt->fpt,
-                    "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
-                    stt->fcount);
-#endif
-            return -1;
-        }
-        subFrames = 80;
-    } else if (stt->fs == 16000)
-    {
-        if ((samples != 160) && (samples != 320))
-        {
-#ifdef AGC_DEBUG //test log
-            fprintf(stt->fpt,
-                    "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
-                    stt->fcount);
-#endif
-            return -1;
-        }
-        subFrames = 160;
-    } else if (stt->fs == 32000)
-    {
-        if ((samples != 160) && (samples != 320))
-        {
-#ifdef AGC_DEBUG //test log
-            fprintf(stt->fpt,
-                    "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
-                    stt->fcount);
-#endif
-            return -1;
-        }
-        subFrames = 160;
-    } else
-    {
-#ifdef AGC_DEBUG //test log
-        fprintf(stt->fpt,
-                "AGC->add_far_end, frame %d: Invalid sample rate\n\n",
-                stt->fcount + 1);
-#endif
-        return -1;
-    }
-
-    for (i = 0; i < samples; i += subFrames)
-    {
-        errHandle += WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, &in_far[i], subFrames);
-    }
-
-    return errHandle;
-}
-
-int WebRtcAgc_VirtualMic(void *agcInst, WebRtc_Word16 *in_near, WebRtc_Word16 *in_near_H,
-                         WebRtc_Word16 samples, WebRtc_Word32 micLevelIn,
-                         WebRtc_Word32 *micLevelOut)
-{
-    WebRtc_Word32 tmpFlt, micLevelTmp, gainIdx;
-    WebRtc_UWord16 gain;
-    WebRtc_Word16 ii;
-    Agc_t *stt;
-
-    WebRtc_UWord32 nrg;
-    WebRtc_Word16 sampleCntr;
-    WebRtc_UWord32 frameNrg = 0;
-    WebRtc_UWord32 frameNrgLimit = 5500;
-    WebRtc_Word16 numZeroCrossing = 0;
-    const WebRtc_Word16 kZeroCrossingLowLim = 15;
-    const WebRtc_Word16 kZeroCrossingHighLim = 20;
-
-    stt = (Agc_t *)agcInst;
-
-    /*
-     *  Before applying gain decide if this is a low-level signal.
-     *  The idea is that digital AGC will not adapt to low-level
-     *  signals.
-     */
-    if (stt->fs != 8000)
-    {
-        frameNrgLimit = frameNrgLimit << 1;
-    }
-
-    frameNrg = WEBRTC_SPL_MUL_16_16(in_near[0], in_near[0]);
-    for (sampleCntr = 1; sampleCntr < samples; sampleCntr++)
-    {
-
-        // increment frame energy if it is less than the limit
-        // the correct value of the energy is not important
-        if (frameNrg < frameNrgLimit)
-        {
-            nrg = WEBRTC_SPL_MUL_16_16(in_near[sampleCntr], in_near[sampleCntr]);
-            frameNrg += nrg;
-        }
-
-        // Count the zero crossings
-        numZeroCrossing += ((in_near[sampleCntr] ^ in_near[sampleCntr - 1]) < 0);
-    }
-
-    if ((frameNrg < 500) || (numZeroCrossing <= 5))
-    {
-        stt->lowLevelSignal = 1;
-    } else if (numZeroCrossing <= kZeroCrossingLowLim)
-    {
-        stt->lowLevelSignal = 0;
-    } else if (frameNrg <= frameNrgLimit)
-    {
-        stt->lowLevelSignal = 1;
-    } else if (numZeroCrossing >= kZeroCrossingHighLim)
-    {
-        stt->lowLevelSignal = 1;
-    } else
-    {
-        stt->lowLevelSignal = 0;
-    }
-
-    micLevelTmp = WEBRTC_SPL_LSHIFT_W32(micLevelIn, stt->scale);
-    /* Set desired level */
-    gainIdx = stt->micVol;
-    if (stt->micVol > stt->maxAnalog)
-    {
-        gainIdx = stt->maxAnalog;
-    }
-    if (micLevelTmp != stt->micRef)
-    {
-        /* Something has happened with the physical level, restart. */
-        stt->micRef = micLevelTmp;
-        stt->micVol = 127;
-        *micLevelOut = 127;
-        stt->micGainIdx = 127;
-        gainIdx = 127;
-    }
-    /* Pre-process the signal to emulate the microphone level. */
-    /* Take one step at a time in the gain table. */
-    if (gainIdx > 127)
-    {
-        gain = kGainTableVirtualMic[gainIdx - 128];
-    } else
-    {
-        gain = kSuppressionTableVirtualMic[127 - gainIdx];
-    }
-    for (ii = 0; ii < samples; ii++)
-    {
-        tmpFlt = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_U16(in_near[ii], gain), 10);
-        if (tmpFlt > 32767)
-        {
-            tmpFlt = 32767;
-            gainIdx--;
-            if (gainIdx >= 127)
-            {
-                gain = kGainTableVirtualMic[gainIdx - 127];
-            } else
-            {
-                gain = kSuppressionTableVirtualMic[127 - gainIdx];
-            }
-        }
-        if (tmpFlt < -32768)
-        {
-            tmpFlt = -32768;
-            gainIdx--;
-            if (gainIdx >= 127)
-            {
-                gain = kGainTableVirtualMic[gainIdx - 127];
-            } else
-            {
-                gain = kSuppressionTableVirtualMic[127 - gainIdx];
-            }
-        }
-        in_near[ii] = (WebRtc_Word16)tmpFlt;
-        if (stt->fs == 32000)
-        {
-            tmpFlt = WEBRTC_SPL_MUL_16_U16(in_near_H[ii], gain);
-            tmpFlt = WEBRTC_SPL_RSHIFT_W32(tmpFlt, 10);
-            if (tmpFlt > 32767)
-            {
-                tmpFlt = 32767;
-            }
-            if (tmpFlt < -32768)
-            {
-                tmpFlt = -32768;
-            }
-            in_near_H[ii] = (WebRtc_Word16)tmpFlt;
-        }
-    }
-    /* Set the level we (finally) used */
-    stt->micGainIdx = gainIdx;
-//    *micLevelOut = stt->micGainIdx;
-    *micLevelOut = WEBRTC_SPL_RSHIFT_W32(stt->micGainIdx, stt->scale);
-    /* Add to Mic as if it was the output from a true microphone */
-    if (WebRtcAgc_AddMic(agcInst, in_near, in_near_H, samples) != 0)
-    {
-        return -1;
-    }
-    return 0;
-}
-
-void WebRtcAgc_UpdateAgcThresholds(Agc_t *stt)
-{
-
-    WebRtc_Word16 tmp16;
-#ifdef MIC_LEVEL_FEEDBACK
-    int zeros;
-
-    if (stt->micLvlSat)
-    {
-        /* Lower the analog target level since we have reached its maximum */
-        zeros = WebRtcSpl_NormW32(stt->Rxx160_LPw32);
-        stt->targetIdxOffset = WEBRTC_SPL_RSHIFT_W16((3 * zeros) - stt->targetIdx - 2, 2);
-    }
-#endif
-
-    /* Set analog target level in envelope dBOv scale */
-    tmp16 = (DIFF_REF_TO_ANALOG * stt->compressionGaindB) + ANALOG_TARGET_LEVEL_2;
-    tmp16 = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32)tmp16, ANALOG_TARGET_LEVEL);
-    stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN + tmp16;
-    if (stt->analogTarget < DIGITAL_REF_AT_0_COMP_GAIN)
-    {
-        stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN;
-    }
-    if (stt->agcMode == kAgcModeFixedDigital)
-    {
-        /* Adjust for different parameter interpretation in FixedDigital mode */
-        stt->analogTarget = stt->compressionGaindB;
-    }
-#ifdef MIC_LEVEL_FEEDBACK
-    stt->analogTarget += stt->targetIdxOffset;
-#endif
-    /* Since the offset between RMS and ENV is not constant, we should make this into a
-     * table, but for now, we'll stick with a constant, tuned for the chosen analog
-     * target level.
-     */
-    stt->targetIdx = ANALOG_TARGET_LEVEL + OFFSET_ENV_TO_RMS;
-#ifdef MIC_LEVEL_FEEDBACK
-    stt->targetIdx += stt->targetIdxOffset;
-#endif
-    /* Analog adaptation limits */
-    /* analogTargetLevel = round((32767*10^(-targetIdx/20))^2*16/2^7) */
-    stt->analogTargetLevel = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx]; /* ex. -20 dBov */
-    stt->startUpperLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 1];/* -19 dBov */
-    stt->startLowerLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 1];/* -21 dBov */
-    stt->upperPrimaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 2];/* -18 dBov */
-    stt->lowerPrimaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 2];/* -22 dBov */
-    stt->upperSecondaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 5];/* -15 dBov */
-    stt->lowerSecondaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 5];/* -25 dBov */
-    stt->upperLimit = stt->startUpperLimit;
-    stt->lowerLimit = stt->startLowerLimit;
-}
-
-void WebRtcAgc_SaturationCtrl(Agc_t *stt, WebRtc_UWord8 *saturated, WebRtc_Word32 *env)
-{
-    WebRtc_Word16 i, tmpW16;
-
-    /* Check if the signal is saturated */
-    for (i = 0; i < 10; i++)
-    {
-        tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(env[i], 20);
-        if (tmpW16 > 875)
-        {
-            stt->envSum += tmpW16;
-        }
-    }
-
-    if (stt->envSum > 25000)
-    {
-        *saturated = 1;
-        stt->envSum = 0;
-    }
-
-    /* stt->envSum *= 0.99; */
-    stt->envSum = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(stt->envSum,
-            (WebRtc_Word16)32440, 15);
-}
-
-void WebRtcAgc_ZeroCtrl(Agc_t *stt, WebRtc_Word32 *inMicLevel, WebRtc_Word32 *env)
-{
-    WebRtc_Word16 i;
-    WebRtc_Word32 tmp32 = 0;
-    WebRtc_Word32 midVal;
-
-    /* Is the input signal zero? */
-    for (i = 0; i < 10; i++)
-    {
-        tmp32 += env[i];
-    }
-
-    /* Each block is allowed to have a few non-zero
-     * samples.
-     */
-    if (tmp32 < 500)
-    {
-        stt->msZero += 10;
-    } else
-    {
-        stt->msZero = 0;
-    }
-
-    if (stt->muteGuardMs > 0)
-    {
-        stt->muteGuardMs -= 10;
-    }
-
-    if (stt->msZero > 500)
-    {
-        stt->msZero = 0;
-
-        /* Increase microphone level only if it's less than 50% */
-        midVal = WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog + stt->minLevel + 1, 1);
-        if (*inMicLevel < midVal)
-        {
-            /* *inMicLevel *= 1.1; */
-            tmp32 = WEBRTC_SPL_MUL(1126, *inMicLevel);
-            *inMicLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 10);
-            /* Reduces risk of a muted mic repeatedly triggering excessive levels due
-             * to zero signal detection. */
-            *inMicLevel = WEBRTC_SPL_MIN(*inMicLevel, stt->zeroCtrlMax);
-            stt->micVol = *inMicLevel;
-        }
-
-#ifdef AGC_DEBUG //test log
-        fprintf(stt->fpt,
-                "\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold, micVol:\n",
-                stt->fcount, stt->micVol);
-#endif
-
-        stt->activeSpeech = 0;
-        stt->Rxx16_LPw32Max = 0;
-
-        /* The AGC has a tendency (due to problems with the VAD parameters), to
-         * vastly increase the volume after a muting event. This timer prevents
-         * upwards adaptation for a short period. */
-        stt->muteGuardMs = kMuteGuardTimeMs;
-    }
-}
-
-void WebRtcAgc_SpeakerInactiveCtrl(Agc_t *stt)
-{
-    /* Check if the near end speaker is inactive.
-     * If that is the case the VAD threshold is
-     * increased since the VAD speech model gets
-     * more sensitive to any sound after a long
-     * silence.
-     */
-
-    WebRtc_Word32 tmp32;
-    WebRtc_Word16 vadThresh;
-
-    if (stt->vadMic.stdLongTerm < 2500)
-    {
-        stt->vadThreshold = 1500;
-    } else
-    {
-        vadThresh = kNormalVadThreshold;
-        if (stt->vadMic.stdLongTerm < 4500)
-        {
-            /* Scale between min and max threshold */
-            vadThresh += WEBRTC_SPL_RSHIFT_W16(4500 - stt->vadMic.stdLongTerm, 1);
-        }
-
-        /* stt->vadThreshold = (31 * stt->vadThreshold + vadThresh) / 32; */
-        tmp32 = (WebRtc_Word32)vadThresh;
-        tmp32 += WEBRTC_SPL_MUL_16_16((WebRtc_Word16)31, stt->vadThreshold);
-        stt->vadThreshold = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 5);
-    }
-}
-
-void WebRtcAgc_ExpCurve(WebRtc_Word16 volume, WebRtc_Word16 *index)
-{
-    // volume in Q14
-    // index in [0-7]
-    /* 8 different curves */
-    if (volume > 5243)
-    {
-        if (volume > 7864)
-        {
-            if (volume > 12124)
-            {
-                *index = 7;
-            } else
-            {
-                *index = 6;
-            }
-        } else
-        {
-            if (volume > 6554)
-            {
-                *index = 5;
-            } else
-            {
-                *index = 4;
-            }
-        }
-    } else
-    {
-        if (volume > 2621)
-        {
-            if (volume > 3932)
-            {
-                *index = 3;
-            } else
-            {
-                *index = 2;
-            }
-        } else
-        {
-            if (volume > 1311)
-            {
-                *index = 1;
-            } else
-            {
-                *index = 0;
-            }
-        }
-    }
-}
-
-WebRtc_Word32 WebRtcAgc_ProcessAnalog(void *state, WebRtc_Word32 inMicLevel,
-                                        WebRtc_Word32 *outMicLevel,
-                                        WebRtc_Word16 vadLogRatio,
-                                        WebRtc_Word16 echo, WebRtc_UWord8 *saturationWarning)
-{
-    WebRtc_UWord32 tmpU32;
-    WebRtc_Word32 Rxx16w32, tmp32;
-    WebRtc_Word32 inMicLevelTmp, lastMicVol;
-    WebRtc_Word16 i;
-    WebRtc_UWord8 saturated = 0;
-    Agc_t *stt;
-
-    stt = (Agc_t *)state;
-    inMicLevelTmp = WEBRTC_SPL_LSHIFT_W32(inMicLevel, stt->scale);
-
-    if (inMicLevelTmp > stt->maxAnalog)
-    {
-#ifdef AGC_DEBUG //test log
-        fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n", stt->fcount);
-#endif
-        return -1;
-    } else if (inMicLevelTmp < stt->minLevel)
-    {
-#ifdef AGC_DEBUG //test log
-        fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel\n", stt->fcount);
-#endif
-        return -1;
-    }
-
-    if (stt->firstCall == 0)
-    {
-        WebRtc_Word32 tmpVol;
-        stt->firstCall = 1;
-        tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (WebRtc_Word32)51, 9);
-        tmpVol = (stt->minLevel + tmp32);
-
-        /* If the mic level is very low at start, increase it! */
-        if ((inMicLevelTmp < tmpVol) && (stt->agcMode == kAgcModeAdaptiveAnalog))
-        {
-            inMicLevelTmp = tmpVol;
-        }
-        stt->micVol = inMicLevelTmp;
-    }
-
-    /* Set the mic level to the previous output value if there is digital input gain */
-    if ((inMicLevelTmp == stt->maxAnalog) && (stt->micVol > stt->maxAnalog))
-    {
-        inMicLevelTmp = stt->micVol;
-    }
-
-    /* If the mic level was manually changed to a very low value raise it! */
-    if ((inMicLevelTmp != stt->micVol) && (inMicLevelTmp < stt->minOutput))
-    {
-        tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (WebRtc_Word32)51, 9);
-        inMicLevelTmp = (stt->minLevel + tmp32);
-        stt->micVol = inMicLevelTmp;
-#ifdef MIC_LEVEL_FEEDBACK
-        //stt->numBlocksMicLvlSat = 0;
-#endif
-#ifdef AGC_DEBUG //test log
-        fprintf(stt->fpt,
-                "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel by manual decrease, raise vol\n",
-                stt->fcount);
-#endif
-    }
-
-    if (inMicLevelTmp != stt->micVol)
-    {
-        // Incoming level mismatch; update our level.
-        // This could be the case if the volume is changed manually, or if the
-        // sound device has a low volume resolution.
-        stt->micVol = inMicLevelTmp;
-    }
-
-    if (inMicLevelTmp > stt->maxLevel)
-    {
-        // Always allow the user to raise the volume above the maxLevel.
-        stt->maxLevel = inMicLevelTmp;
-    }
-
-    // Store last value here, after we've taken care of manual updates etc.
-    lastMicVol = stt->micVol;
-
-    /* Checks if the signal is saturated. Also a check if individual samples
-     * are larger than 12000 is done. If they are the counter for increasing
-     * the volume level is set to -100ms
-     */
-    WebRtcAgc_SaturationCtrl(stt, &saturated, stt->env[0]);
-
-    /* The AGC is always allowed to lower the level if the signal is saturated */
-    if (saturated == 1)
-    {
-        /* Lower the recording level
-         * Rxx160_LP is adjusted down because it is so slow it could
-         * cause the AGC to make wrong decisions. */
-        /* stt->Rxx160_LPw32 *= 0.875; */
-        stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 3), 7);
-
-        stt->zeroCtrlMax = stt->micVol;
-
-        /* stt->micVol *= 0.903; */
-        tmp32 = inMicLevelTmp - stt->minLevel;
-        tmpU32 = WEBRTC_SPL_UMUL(29591, (WebRtc_UWord32)(tmp32));
-        stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
-        if (stt->micVol > lastMicVol - 2)
-        {
-            stt->micVol = lastMicVol - 2;
-        }
-        inMicLevelTmp = stt->micVol;
-
-#ifdef AGC_DEBUG //test log
-        fprintf(stt->fpt,
-                "\tAGC->ProcessAnalog, frame %d: saturated, micVol = %d\n",
-                stt->fcount, stt->micVol);
-#endif
-
-        if (stt->micVol < stt->minOutput)
-        {
-            *saturationWarning = 1;
-        }
-
-        /* Reset counter for decrease of volume level to avoid
-         * decreasing too much. The saturation control can still
-         * lower the level if needed. */
-        stt->msTooHigh = -100;
-
-        /* Enable the control mechanism to ensure that our measure,
-         * Rxx160_LP, is in the correct range. This must be done since
-         * the measure is very slow. */
-        stt->activeSpeech = 0;
-        stt->Rxx16_LPw32Max = 0;
-
-        /* Reset to initial values */
-        stt->msecSpeechInnerChange = kMsecSpeechInner;
-        stt->msecSpeechOuterChange = kMsecSpeechOuter;
-        stt->changeToSlowMode = 0;
-
-        stt->muteGuardMs = 0;
-
-        stt->upperLimit = stt->startUpperLimit;
-        stt->lowerLimit = stt->startLowerLimit;
-#ifdef MIC_LEVEL_FEEDBACK
-        //stt->numBlocksMicLvlSat = 0;
-#endif
-    }
-
-    /* Check if the input speech is zero. If so the mic volume
-     * is increased. On some computers the input is zero up as high
-     * level as 17% */
-    WebRtcAgc_ZeroCtrl(stt, &inMicLevelTmp, stt->env[0]);
-
-    /* Check if the near end speaker is inactive.
-     * If that is the case the VAD threshold is
-     * increased since the VAD speech model gets
-     * more sensitive to any sound after a long
-     * silence.
-     */
-    WebRtcAgc_SpeakerInactiveCtrl(stt);
-
-    for (i = 0; i < 5; i++)
-    {
-        /* Computed on blocks of 16 samples */
-
-        Rxx16w32 = stt->Rxx16w32_array[0][i];
-
-        /* Rxx160w32 in Q(-7) */
-        tmp32 = WEBRTC_SPL_RSHIFT_W32(Rxx16w32 - stt->Rxx16_vectorw32[stt->Rxx16pos], 3);
-        stt->Rxx160w32 = stt->Rxx160w32 + tmp32;
-        stt->Rxx16_vectorw32[stt->Rxx16pos] = Rxx16w32;
-
-        /* Circular buffer */
-        stt->Rxx16pos = stt->Rxx16pos++;
-        if (stt->Rxx16pos == RXX_BUFFER_LEN)
-        {
-            stt->Rxx16pos = 0;
-        }
-
-        /* Rxx16_LPw32 in Q(-4) */
-        tmp32 = WEBRTC_SPL_RSHIFT_W32(Rxx16w32 - stt->Rxx16_LPw32, kAlphaShortTerm);
-        stt->Rxx16_LPw32 = (stt->Rxx16_LPw32) + tmp32;
-
-        if (vadLogRatio > stt->vadThreshold)
-        {
-            /* Speech detected! */
-
-            /* Check if Rxx160_LP is in the correct range. If
-             * it is too high/low then we set it to the maximum of
-             * Rxx16_LPw32 during the first 200ms of speech.
-             */
-            if (stt->activeSpeech < 250)
-            {
-                stt->activeSpeech += 2;
-
-                if (stt->Rxx16_LPw32 > stt->Rxx16_LPw32Max)
-                {
-                    stt->Rxx16_LPw32Max = stt->Rxx16_LPw32;
-                }
-            } else if (stt->activeSpeech == 250)
-            {
-                stt->activeSpeech += 2;
-                tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx16_LPw32Max, 3);
-                stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, RXX_BUFFER_LEN);
-            }
-
-            tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160w32 - stt->Rxx160_LPw32, kAlphaLongTerm);
-            stt->Rxx160_LPw32 = stt->Rxx160_LPw32 + tmp32;
-
-            if (stt->Rxx160_LPw32 > stt->upperSecondaryLimit)
-            {
-                stt->msTooHigh += 2;
-                stt->msTooLow = 0;
-                stt->changeToSlowMode = 0;
-
-                if (stt->msTooHigh > stt->msecSpeechOuterChange)
-                {
-                    stt->msTooHigh = 0;
-
-                    /* Lower the recording level */
-                    /* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
-                    tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
-                    stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 53);
-
-                    /* Reduce the max gain to avoid excessive oscillation
-                     * (but never drop below the maximum analog level).
-                     * stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
-                     */
-                    tmp32 = (15 * stt->maxLevel) + stt->micVol;
-                    stt->maxLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
-                    stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);
-
-                    stt->zeroCtrlMax = stt->micVol;
-
-                    /* 0.95 in Q15 */
-                    tmp32 = inMicLevelTmp - stt->minLevel;
-                    tmpU32 = WEBRTC_SPL_UMUL(31130, (WebRtc_UWord32)(tmp32));
-                    stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
-                    if (stt->micVol > lastMicVol - 1)
-                    {
-                        stt->micVol = lastMicVol - 1;
-                    }
-                    inMicLevelTmp = stt->micVol;
-
-                    /* Enable the control mechanism to ensure that our measure,
-                     * Rxx160_LP, is in the correct range.
-                     */
-                    stt->activeSpeech = 0;
-                    stt->Rxx16_LPw32Max = 0;
-#ifdef MIC_LEVEL_FEEDBACK
-                    //stt->numBlocksMicLvlSat = 0;
-#endif
-#ifdef AGC_DEBUG //test log
-                    fprintf(stt->fpt,
-                            "\tAGC->ProcessAnalog, frame %d: measure > 2ndUpperLim, micVol = %d, maxLevel = %d\n",
-                            stt->fcount, stt->micVol, stt->maxLevel);
-#endif
-                }
-            } else if (stt->Rxx160_LPw32 > stt->upperLimit)
-            {
-                stt->msTooHigh += 2;
-                stt->msTooLow = 0;
-                stt->changeToSlowMode = 0;
-
-                if (stt->msTooHigh > stt->msecSpeechInnerChange)
-                {
-                    /* Lower the recording level */
-                    stt->msTooHigh = 0;
-                    /* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
-                    tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
-                    stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 53);
-
-                    /* Reduce the max gain to avoid excessive oscillation
-                     * (but never drop below the maximum analog level).
-                     * stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
-                     */
-                    tmp32 = (15 * stt->maxLevel) + stt->micVol;
-                    stt->maxLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
-                    stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);
-
-                    stt->zeroCtrlMax = stt->micVol;
-
-                    /* 0.965 in Q15 */
-                    tmp32 = inMicLevelTmp - stt->minLevel;
-                    tmpU32 = WEBRTC_SPL_UMUL(31621, (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel));
-                    stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
-                    if (stt->micVol > lastMicVol - 1)
-                    {
-                        stt->micVol = lastMicVol - 1;
-                    }
-                    inMicLevelTmp = stt->micVol;
-
-#ifdef MIC_LEVEL_FEEDBACK
-                    //stt->numBlocksMicLvlSat = 0;
-#endif
-#ifdef AGC_DEBUG //test log
-                    fprintf(stt->fpt,
-                            "\tAGC->ProcessAnalog, frame %d: measure > UpperLim, micVol = %d, maxLevel = %d\n",
-                            stt->fcount, stt->micVol, stt->maxLevel);
-#endif
-                }
-            } else if (stt->Rxx160_LPw32 < stt->lowerSecondaryLimit)
-            {
-                stt->msTooHigh = 0;
-                stt->changeToSlowMode = 0;
-                stt->msTooLow += 2;
-
-                if (stt->msTooLow > stt->msecSpeechOuterChange)
-                {
-                    /* Raise the recording level */
-                    WebRtc_Word16 index, weightFIX;
-                    WebRtc_Word16 volNormFIX = 16384; // =1 in Q14.
-
-                    stt->msTooLow = 0;
-
-                    /* Normalize the volume level */
-                    tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
-                    if (stt->maxInit != stt->minLevel)
-                    {
-                        volNormFIX = (WebRtc_Word16)WEBRTC_SPL_DIV(tmp32,
-                                                              (stt->maxInit - stt->minLevel));
-                    }
-
-                    /* Find correct curve */
-                    WebRtcAgc_ExpCurve(volNormFIX, &index);
-
-                    /* Compute weighting factor for the volume increase, 32^(-2*X)/2+1.05 */
-                    weightFIX = kOffset1[index]
-                              - (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(kSlope1[index],
-                                                                         volNormFIX, 13);
-
-                    /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
-                    tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
-                    stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 67);
-
-                    tmp32 = inMicLevelTmp - stt->minLevel;
-                    tmpU32 = ((WebRtc_UWord32)weightFIX * (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel));
-                    stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 14) + stt->minLevel;
-                    if (stt->micVol < lastMicVol + 2)
-                    {
-                        stt->micVol = lastMicVol + 2;
-                    }
-
-                    inMicLevelTmp = stt->micVol;
-
-#ifdef MIC_LEVEL_FEEDBACK
-                    /* Count ms in level saturation */
-                    //if (stt->micVol > stt->maxAnalog) {
-                    if (stt->micVol > 150)
-                    {
-                        /* mic level is saturated */
-                        stt->numBlocksMicLvlSat++;
-                        fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
-                    }
-#endif
-#ifdef AGC_DEBUG //test log
-                    fprintf(stt->fpt,
-                            "\tAGC->ProcessAnalog, frame %d: measure < 2ndLowerLim, micVol = %d\n",
-                            stt->fcount, stt->micVol);
-#endif
-                }
-            } else if (stt->Rxx160_LPw32 < stt->lowerLimit)
-            {
-                stt->msTooHigh = 0;
-                stt->changeToSlowMode = 0;
-                stt->msTooLow += 2;
-
-                if (stt->msTooLow > stt->msecSpeechInnerChange)
-                {
-                    /* Raise the recording level */
-                    WebRtc_Word16 index, weightFIX;
-                    WebRtc_Word16 volNormFIX = 16384; // =1 in Q14.
-
-                    stt->msTooLow = 0;
-
-                    /* Normalize the volume level */
-                    tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
-                    if (stt->maxInit != stt->minLevel)
-                    {
-                        volNormFIX = (WebRtc_Word16)WEBRTC_SPL_DIV(tmp32,
-                                                              (stt->maxInit - stt->minLevel));
-                    }
-
-                    /* Find correct curve */
-                    WebRtcAgc_ExpCurve(volNormFIX, &index);
-
-                    /* Compute weighting factor for the volume increase, (3.^(-2.*X))/8+1 */
-                    weightFIX = kOffset2[index]
-                              - (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(kSlope2[index],
-                                                                         volNormFIX, 13);
-
-                    /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
-                    tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
-                    stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 67);
-
-                    tmp32 = inMicLevelTmp - stt->minLevel;
-                    tmpU32 = ((WebRtc_UWord32)weightFIX * (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel));
-                    stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 14) + stt->minLevel;
-                    if (stt->micVol < lastMicVol + 1)
-                    {
-                        stt->micVol = lastMicVol + 1;
-                    }
-
-                    inMicLevelTmp = stt->micVol;
-
-#ifdef MIC_LEVEL_FEEDBACK
-                    /* Count ms in level saturation */
-                    //if (stt->micVol > stt->maxAnalog) {
-                    if (stt->micVol > 150)
-                    {
-                        /* mic level is saturated */
-                        stt->numBlocksMicLvlSat++;
-                        fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
-                    }
-#endif
-#ifdef AGC_DEBUG //test log
-                    fprintf(stt->fpt,
-                            "\tAGC->ProcessAnalog, frame %d: measure < LowerLim, micVol = %d\n",
-                            stt->fcount, stt->micVol);
-#endif
-
-                }
-            } else
-            {
-                /* The signal is inside the desired range which is:
-                 * lowerLimit < Rxx160_LP/640 < upperLimit
-                 */
-                if (stt->changeToSlowMode > 4000)
-                {
-                    stt->msecSpeechInnerChange = 1000;
-                    stt->msecSpeechOuterChange = 500;
-                    stt->upperLimit = stt->upperPrimaryLimit;
-                    stt->lowerLimit = stt->lowerPrimaryLimit;
-                } else
-                {
-                    stt->changeToSlowMode += 2; // in milliseconds
-                }
-                stt->msTooLow = 0;
-                stt->msTooHigh = 0;
-
-                stt->micVol = inMicLevelTmp;
-
-            }
-#ifdef MIC_LEVEL_FEEDBACK
-            if (stt->numBlocksMicLvlSat > NUM_BLOCKS_IN_SAT_BEFORE_CHANGE_TARGET)
-            {
-                stt->micLvlSat = 1;
-                fprintf(stderr, "target before = %d (%d)\n", stt->analogTargetLevel, stt->targetIdx);
-                WebRtcAgc_UpdateAgcThresholds(stt);
-                WebRtcAgc_CalculateGainTable(&(stt->digitalAgc.gainTable[0]),
-                        stt->compressionGaindB, stt->targetLevelDbfs, stt->limiterEnable,
-                        stt->analogTarget);
-                stt->numBlocksMicLvlSat = 0;
-                stt->micLvlSat = 0;
-                fprintf(stderr, "target offset = %d\n", stt->targetIdxOffset);
-                fprintf(stderr, "target after  = %d (%d)\n", stt->analogTargetLevel, stt->targetIdx);
-            }
-#endif
-        }
-    }
-
-    /* Ensure gain is not increased in presence of echo or after a mute event
-     * (but allow the zeroCtrl() increase on the frame of a mute detection).
-     */
-    if (echo == 1 || (stt->muteGuardMs > 0 && stt->muteGuardMs < kMuteGuardTimeMs))
-    {
-        if (stt->micVol > lastMicVol)
-        {
-            stt->micVol = lastMicVol;
-        }
-    }
-
-    /* limit the gain */
-    if (stt->micVol > stt->maxLevel)
-    {
-        stt->micVol = stt->maxLevel;
-    } else if (stt->micVol < stt->minOutput)
-    {
-        stt->micVol = stt->minOutput;
-    }
-
-    *outMicLevel = WEBRTC_SPL_RSHIFT_W32(stt->micVol, stt->scale);
-    if (*outMicLevel > WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog, stt->scale))
-    {
-        *outMicLevel = WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog, stt->scale);
-    }
-
-    return 0;
-}
-
-int WebRtcAgc_Process(void *agcInst, const WebRtc_Word16 *in_near,
-                      const WebRtc_Word16 *in_near_H, WebRtc_Word16 samples,
-                      WebRtc_Word16 *out, WebRtc_Word16 *out_H, WebRtc_Word32 inMicLevel,
-                      WebRtc_Word32 *outMicLevel, WebRtc_Word16 echo,
-                      WebRtc_UWord8 *saturationWarning)
-{
-    Agc_t *stt;
-    WebRtc_Word32 inMicLevelTmp;
-    WebRtc_Word16 subFrames, i;
-    WebRtc_UWord8 satWarningTmp = 0;
-
-    stt = (Agc_t *)agcInst;
-
-    //
-    if (stt == NULL)
-    {
-        return -1;
-    }
-    //
-
-
-    if (stt->fs == 8000)
-    {
-        if ((samples != 80) && (samples != 160))
-        {
-#ifdef AGC_DEBUG //test log
-            fprintf(stt->fpt,
-                    "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
-#endif
-            return -1;
-        }
-        subFrames = 80;
-    } else if (stt->fs == 16000)
-    {
-        if ((samples != 160) && (samples != 320))
-        {
-#ifdef AGC_DEBUG //test log
-            fprintf(stt->fpt,
-                    "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
-#endif
-            return -1;
-        }
-        subFrames = 160;
-    } else if (stt->fs == 32000)
-    {
-        if ((samples != 160) && (samples != 320))
-        {
-#ifdef AGC_DEBUG //test log
-            fprintf(stt->fpt,
-                    "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
-#endif
-            return -1;
-        }
-        subFrames = 160;
-    } else
-    {
-#ifdef AGC_DEBUG// test log
-        fprintf(stt->fpt,
-                "AGC->Process, frame %d: Invalid sample rate\n\n", stt->fcount);
-#endif
-        return -1;
-    }
-
-    /* Check for valid pointers based on sampling rate */
-    if (stt->fs == 32000 && in_near_H == NULL)
-    {
-        return -1;
-    }
-    /* Check for valid pointers for low band */
-    if (in_near == NULL)
-    {
-        return -1;
-    }
-
-    *saturationWarning = 0;
-    //TODO: PUT IN RANGE CHECKING FOR INPUT LEVELS
-    *outMicLevel = inMicLevel;
-    inMicLevelTmp = inMicLevel;
-
-    memcpy(out, in_near, samples * sizeof(WebRtc_Word16));
-    if (stt->fs == 32000)
-    {
-        memcpy(out_H, in_near_H, samples * sizeof(WebRtc_Word16));
-    }
-
-#ifdef AGC_DEBUG//test log
-    stt->fcount++;
-#endif
-
-    for (i = 0; i < samples; i += subFrames)
-    {
-        if (WebRtcAgc_ProcessDigital(&stt->digitalAgc, &in_near[i], &in_near_H[i], &out[i], &out_H[i],
-                           stt->fs, stt->lowLevelSignal) == -1)
-        {
-#ifdef AGC_DEBUG//test log
-            fprintf(stt->fpt, "AGC->Process, frame %d: Error from DigAGC\n\n", stt->fcount);
-#endif
-            return -1;
-        }
-        if ((stt->agcMode < kAgcModeFixedDigital) && ((stt->lowLevelSignal == 0)
-                || (stt->agcMode != kAgcModeAdaptiveDigital)))
-        {
-            if (WebRtcAgc_ProcessAnalog(agcInst, inMicLevelTmp, outMicLevel,
-                                          stt->vadMic.logRatio, echo, saturationWarning) == -1)
-            {
-                return -1;
-            }
-        }
-#ifdef AGC_DEBUG//test log
-        fprintf(stt->agcLog, "%5d\t%d\t%d\t%d\n", stt->fcount, inMicLevelTmp, *outMicLevel, stt->maxLevel, stt->micVol);
-#endif
-
-        /* update queue */
-        if (stt->inQueue > 1)
-        {
-            memcpy(stt->env[0], stt->env[1], 10 * sizeof(WebRtc_Word32));
-            memcpy(stt->Rxx16w32_array[0], stt->Rxx16w32_array[1], 5 * sizeof(WebRtc_Word32));
-        }
-
-        if (stt->inQueue > 0)
-        {
-            stt->inQueue--;
-        }
-
-        /* If 20ms frames are used the input mic level must be updated so that
-         * the analog AGC does not think that there has been a manual volume
-         * change. */
-        inMicLevelTmp = *outMicLevel;
-
-        /* Store a positive saturation warning. */
-        if (*saturationWarning == 1)
-        {
-            satWarningTmp = 1;
-        }
-    }
-
-    /* Trigger the saturation warning if displayed by any of the frames. */
-    *saturationWarning = satWarningTmp;
-
-    return 0;
-}
-
-int WebRtcAgc_set_config(void *agcInst, WebRtcAgc_config_t agcConfig)
-{
-    Agc_t *stt;
-    stt = (Agc_t *)agcInst;
-
-    if (stt == NULL)
-    {
-        return -1;
-    }
-
-    if (stt->initFlag != kInitCheck)
-    {
-        stt->lastError = AGC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    if (agcConfig.limiterEnable != kAgcFalse && agcConfig.limiterEnable != kAgcTrue)
-    {
-        stt->lastError = AGC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    stt->limiterEnable = agcConfig.limiterEnable;
-    stt->compressionGaindB = agcConfig.compressionGaindB;
-    if ((agcConfig.targetLevelDbfs < 0) || (agcConfig.targetLevelDbfs > 31))
-    {
-        stt->lastError = AGC_BAD_PARAMETER_ERROR;
-        return -1;
-    }
-    stt->targetLevelDbfs = agcConfig.targetLevelDbfs;
-
-    if (stt->agcMode == kAgcModeFixedDigital)
-    {
-        /* Adjust for different parameter interpretation in FixedDigital mode */
-        stt->compressionGaindB += agcConfig.targetLevelDbfs;
-    }
-
-    /* Update threshold levels for analog adaptation */
-    WebRtcAgc_UpdateAgcThresholds(stt);
-
-    /* Recalculate gain table */
-    if (WebRtcAgc_CalculateGainTable(&(stt->digitalAgc.gainTable[0]), stt->compressionGaindB,
-                           stt->targetLevelDbfs, stt->limiterEnable, stt->analogTarget) == -1)
-    {
-#ifdef AGC_DEBUG//test log
-        fprintf(stt->fpt, "AGC->set_config, frame %d: Error from calcGainTable\n\n", stt->fcount);
-#endif
-        return -1;
-    }
-    /* Store the config in a WebRtcAgc_config_t */
-    stt->usedConfig.compressionGaindB = agcConfig.compressionGaindB;
-    stt->usedConfig.limiterEnable = agcConfig.limiterEnable;
-    stt->usedConfig.targetLevelDbfs = agcConfig.targetLevelDbfs;
-
-    return 0;
-}
-
-int WebRtcAgc_get_config(void *agcInst, WebRtcAgc_config_t *config)
-{
-    Agc_t *stt;
-    stt = (Agc_t *)agcInst;
-
-    if (stt == NULL)
-    {
-        return -1;
-    }
-
-    if (config == NULL)
-    {
-        stt->lastError = AGC_NULL_POINTER_ERROR;
-        return -1;
-    }
-
-    if (stt->initFlag != kInitCheck)
-    {
-        stt->lastError = AGC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    config->limiterEnable = stt->usedConfig.limiterEnable;
-    config->targetLevelDbfs = stt->usedConfig.targetLevelDbfs;
-    config->compressionGaindB = stt->usedConfig.compressionGaindB;
-
-    return 0;
-}
-
-int WebRtcAgc_Create(void **agcInst)
-{
-    Agc_t *stt;
-    if (agcInst == NULL)
-    {
-        return -1;
-    }
-    stt = (Agc_t *)malloc(sizeof(Agc_t));
-
-    *agcInst = stt;
-    if (stt == NULL)
-    {
-        return -1;
-    }
-
-#ifdef AGC_DEBUG
-    stt->fpt = fopen("./agc_test_log.txt", "wt");
-    stt->agcLog = fopen("./agc_debug_log.txt", "wt");
-    stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt");
-#endif
-
-    stt->initFlag = 0;
-    stt->lastError = 0;
-
-    return 0;
-}
-
-int WebRtcAgc_Free(void *state)
-{
-    Agc_t *stt;
-
-    stt = (Agc_t *)state;
-#ifdef AGC_DEBUG
-    fclose(stt->fpt);
-    fclose(stt->agcLog);
-    fclose(stt->digitalAgc.logFile);
-#endif
-    free(stt);
-
-    return 0;
-}
-
-/* minLevel     - Minimum volume level
- * maxLevel     - Maximum volume level
- */
-int WebRtcAgc_Init(void *agcInst, WebRtc_Word32 minLevel, WebRtc_Word32 maxLevel,
-                             WebRtc_Word16 agcMode, WebRtc_UWord32 fs)
-{
-    WebRtc_Word32 max_add, tmp32;
-    WebRtc_Word16 i;
-    int tmpNorm;
-    Agc_t *stt;
-
-    /* typecast state pointer */
-    stt = (Agc_t *)agcInst;
-
-    if (WebRtcAgc_InitDigital(&stt->digitalAgc, agcMode) != 0)
-    {
-        stt->lastError = AGC_UNINITIALIZED_ERROR;
-        return -1;
-    }
-
-    /* Analog AGC variables */
-    stt->envSum = 0;
-
-    /* mode     = 0 - Only saturation protection
-     *            1 - Analog Automatic Gain Control [-targetLevelDbfs (default -3 dBOv)]
-     *            2 - Digital Automatic Gain Control [-targetLevelDbfs (default -3 dBOv)]
-     *            3 - Fixed Digital Gain [compressionGaindB (default 8 dB)]
-     */
-#ifdef AGC_DEBUG//test log
-    stt->fcount = 0;
-    fprintf(stt->fpt, "AGC->Init\n");
-#endif
-    if (agcMode < kAgcModeUnchanged || agcMode > kAgcModeFixedDigital)
-    {
-#ifdef AGC_DEBUG//test log
-        fprintf(stt->fpt, "AGC->Init: error, incorrect mode\n\n");
-#endif
-        return -1;
-    }
-    stt->agcMode = agcMode;
-    stt->fs = fs;
-
-    /* initialize input VAD */
-    WebRtcAgc_InitVad(&stt->vadMic);
-
-    /* If the volume range is smaller than 0-256 then
-     * the levels are shifted up to Q8-domain */
-    tmpNorm = WebRtcSpl_NormU32((WebRtc_UWord32)maxLevel);
-    stt->scale = tmpNorm - 23;
-    if (stt->scale < 0)
-    {
-        stt->scale = 0;
-    }
-    // TODO(bjornv): Investigate if we really need to scale up a small range now when we have
-    // a guard against zero-increments. For now, we do not support scale up (scale = 0).
-    stt->scale = 0;
-    maxLevel = WEBRTC_SPL_LSHIFT_W32(maxLevel, stt->scale);
-    minLevel = WEBRTC_SPL_LSHIFT_W32(minLevel, stt->scale);
-
-    /* Make minLevel and maxLevel static in AdaptiveDigital */
-    if (stt->agcMode == kAgcModeAdaptiveDigital)
-    {
-        minLevel = 0;
-        maxLevel = 255;
-        stt->scale = 0;
-    }
-    /* The maximum supplemental volume range is based on a vague idea
-     * of how much lower the gain will be than the real analog gain. */
-    max_add = WEBRTC_SPL_RSHIFT_W32(maxLevel - minLevel, 2);
-
-    /* Minimum/maximum volume level that can be set */
-    stt->minLevel = minLevel;
-    stt->maxAnalog = maxLevel;
-    stt->maxLevel = maxLevel + max_add;
-    stt->maxInit = stt->maxLevel;
-
-    stt->zeroCtrlMax = stt->maxAnalog;
-
-    /* Initialize micVol parameter */
-    stt->micVol = stt->maxAnalog;
-    if (stt->agcMode == kAgcModeAdaptiveDigital)
-    {
-        stt->micVol = 127; /* Mid-point of mic level */
-    }
-    stt->micRef = stt->micVol;
-    stt->micGainIdx = 127;
-#ifdef MIC_LEVEL_FEEDBACK
-    stt->numBlocksMicLvlSat = 0;
-    stt->micLvlSat = 0;
-#endif
-#ifdef AGC_DEBUG//test log
-    fprintf(stt->fpt,
-            "AGC->Init: minLevel = %d, maxAnalog = %d, maxLevel = %d\n",
-            stt->minLevel, stt->maxAnalog, stt->maxLevel);
-#endif
-
-    /* Minimum output volume is 4% higher than the available lowest volume level */
-    tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (WebRtc_Word32)10, 8);
-    stt->minOutput = (stt->minLevel + tmp32);
-
-    stt->msTooLow = 0;
-    stt->msTooHigh = 0;
-    stt->changeToSlowMode = 0;
-    stt->firstCall = 0;
-    stt->msZero = 0;
-    stt->muteGuardMs = 0;
-    stt->gainTableIdx = 0;
-
-    stt->msecSpeechInnerChange = kMsecSpeechInner;
-    stt->msecSpeechOuterChange = kMsecSpeechOuter;
-
-    stt->activeSpeech = 0;
-    stt->Rxx16_LPw32Max = 0;
-
-    stt->vadThreshold = kNormalVadThreshold;
-    stt->inActive = 0;
-
-    for (i = 0; i < RXX_BUFFER_LEN; i++)
-    {
-        stt->Rxx16_vectorw32[i] = (WebRtc_Word32)1000; /* -54dBm0 */
-    }
-    stt->Rxx160w32 = 125 * RXX_BUFFER_LEN; /* (stt->Rxx16_vectorw32[0]>>3) = 125 */
-
-    stt->Rxx16pos = 0;
-    stt->Rxx16_LPw32 = (WebRtc_Word32)16284; /* Q(-4) */
-
-    for (i = 0; i < 5; i++)
-    {
-        stt->Rxx16w32_array[0][i] = 0;
-    }
-    for (i = 0; i < 20; i++)
-    {
-        stt->env[0][i] = 0;
-    }
-    stt->inQueue = 0;
-
-#ifdef MIC_LEVEL_FEEDBACK
-    stt->targetIdxOffset = 0;
-#endif
-
-    WebRtcSpl_MemSetW32(stt->filterState, 0, 8);
-
-    stt->initFlag = kInitCheck;
-    // Default config settings.
-    stt->defaultConfig.limiterEnable = kAgcTrue;
-    stt->defaultConfig.targetLevelDbfs = AGC_DEFAULT_TARGET_LEVEL;
-    stt->defaultConfig.compressionGaindB = AGC_DEFAULT_COMP_GAIN;
-
-    if (WebRtcAgc_set_config(stt, stt->defaultConfig) == -1)
-    {
-        stt->lastError = AGC_UNSPECIFIED_ERROR;
-        return -1;
-    }
-    stt->Rxx160_LPw32 = stt->analogTargetLevel; // Initialize rms value
-
-    stt->lowLevelSignal = 0;
-
-    /* Only positive values are allowed that are not too large */
-    if ((minLevel >= maxLevel) || (maxLevel & 0xFC000000))
-    {
-#ifdef AGC_DEBUG//test log
-        fprintf(stt->fpt, "minLevel, maxLevel value(s) are invalid\n\n");
-#endif
-        return -1;
-    } else
-    {
-#ifdef AGC_DEBUG//test log
-        fprintf(stt->fpt, "\n");
-#endif
-        return 0;
-    }
-}
-
-int WebRtcAgc_Version(WebRtc_Word8 *versionStr, WebRtc_Word16 length)
-{
-    const WebRtc_Word8 version[] = "AGC 1.7.0";
-    const WebRtc_Word16 versionLen = (WebRtc_Word16)strlen(version) + 1;
-
-    if (versionStr == NULL)
-    {
-        return -1;
-    }
-
-    if (versionLen > length)
-    {
-        return -1;
-    }
-
-    strncpy(versionStr, version, versionLen);
-    return 0;
-}
diff --git a/src/modules/audio_processing/agc/main/source/analog_agc.h b/src/modules/audio_processing/agc/main/source/analog_agc.h
deleted file mode 100644
index b32ac65..0000000
--- a/src/modules/audio_processing/agc/main/source/analog_agc.h
+++ /dev/null
@@ -1,133 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
-
-#include "typedefs.h"
-#include "gain_control.h"
-#include "digital_agc.h"
-
-//#define AGC_DEBUG
-//#define MIC_LEVEL_FEEDBACK
-#ifdef AGC_DEBUG
-#include <stdio.h>
-#endif
-
-/* Analog Automatic Gain Control variables:
- * Constant declarations (inner limits inside which no changes are done)
- * In the beginning the range is narrower to widen as soon as the measure
- * 'Rxx160_LP' is inside it. Currently the starting limits are -22.2+/-1dBm0
- * and the final limits -22.2+/-2.5dBm0. These levels makes the speech signal
- * go towards -25.4dBm0 (-31.4dBov). Tuned with wbfile-31.4dBov.pcm
- * The limits are created by running the AGC with a file having the desired
- * signal level and thereafter plotting Rxx160_LP in the dBm0-domain defined
- * by out=10*log10(in/260537279.7); Set the target level to the average level
- * of our measure Rxx160_LP. Remember that the levels are in blocks of 16 in
- * Q(-7). (Example matlab code: round(db2pow(-21.2)*16/2^7) )
- */
-#define RXX_BUFFER_LEN  10
-
-static const WebRtc_Word16 kMsecSpeechInner = 520;
-static const WebRtc_Word16 kMsecSpeechOuter = 340;
-
-static const WebRtc_Word16 kNormalVadThreshold = 400;
-
-static const WebRtc_Word16 kAlphaShortTerm = 6; // 1 >> 6 = 0.0156
-static const WebRtc_Word16 kAlphaLongTerm = 10; // 1 >> 10 = 0.000977
-
-typedef struct
-{
-    // Configurable parameters/variables
-    WebRtc_UWord32      fs;                 // Sampling frequency
-    WebRtc_Word16       compressionGaindB;  // Fixed gain level in dB
-    WebRtc_Word16       targetLevelDbfs;    // Target level in -dBfs of envelope (default -3)
-    WebRtc_Word16       agcMode;            // Hard coded mode (adaptAna/adaptDig/fixedDig)
-    WebRtc_UWord8       limiterEnable;      // Enabling limiter (on/off (default off))
-    WebRtcAgc_config_t  defaultConfig;
-    WebRtcAgc_config_t  usedConfig;
-
-    // General variables
-    WebRtc_Word16       initFlag;
-    WebRtc_Word16       lastError;
-
-    // Target level parameters
-    // Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7)
-    WebRtc_Word32       analogTargetLevel;  // = RXX_BUFFER_LEN * 846805;       -22 dBfs
-    WebRtc_Word32       startUpperLimit;    // = RXX_BUFFER_LEN * 1066064;      -21 dBfs
-    WebRtc_Word32       startLowerLimit;    // = RXX_BUFFER_LEN * 672641;       -23 dBfs
-    WebRtc_Word32       upperPrimaryLimit;  // = RXX_BUFFER_LEN * 1342095;      -20 dBfs
-    WebRtc_Word32       lowerPrimaryLimit;  // = RXX_BUFFER_LEN * 534298;       -24 dBfs
-    WebRtc_Word32       upperSecondaryLimit;// = RXX_BUFFER_LEN * 2677832;      -17 dBfs
-    WebRtc_Word32       lowerSecondaryLimit;// = RXX_BUFFER_LEN * 267783;       -27 dBfs
-    WebRtc_UWord16      targetIdx;          // Table index for corresponding target level
-#ifdef MIC_LEVEL_FEEDBACK
-    WebRtc_UWord16      targetIdxOffset;    // Table index offset for level compensation
-#endif
-    WebRtc_Word16       analogTarget;       // Digital reference level in ENV scale
-
-    // Analog AGC specific variables
-    WebRtc_Word32       filterState[8];     // For downsampling wb to nb
-    WebRtc_Word32       upperLimit;         // Upper limit for mic energy
-    WebRtc_Word32       lowerLimit;         // Lower limit for mic energy
-    WebRtc_Word32       Rxx160w32;          // Average energy for one frame
-    WebRtc_Word32       Rxx16_LPw32;        // Low pass filtered subframe energies
-    WebRtc_Word32       Rxx160_LPw32;       // Low pass filtered frame energies
-    WebRtc_Word32       Rxx16_LPw32Max;     // Keeps track of largest energy subframe
-    WebRtc_Word32       Rxx16_vectorw32[RXX_BUFFER_LEN];// Array with subframe energies
-    WebRtc_Word32       Rxx16w32_array[2][5];// Energy values of microphone signal
-    WebRtc_Word32       env[2][10];         // Envelope values of subframes
-
-    WebRtc_Word16       Rxx16pos;           // Current position in the Rxx16_vectorw32
-    WebRtc_Word16       envSum;             // Filtered scaled envelope in subframes
-    WebRtc_Word16       vadThreshold;       // Threshold for VAD decision
-    WebRtc_Word16       inActive;           // Inactive time in milliseconds
-    WebRtc_Word16       msTooLow;           // Milliseconds of speech at a too low level
-    WebRtc_Word16       msTooHigh;          // Milliseconds of speech at a too high level
-    WebRtc_Word16       changeToSlowMode;   // Change to slow mode after some time at target
-    WebRtc_Word16       firstCall;          // First call to the process-function
-    WebRtc_Word16       msZero;             // Milliseconds of zero input
-    WebRtc_Word16       msecSpeechOuterChange;// Min ms of speech between volume changes
-    WebRtc_Word16       msecSpeechInnerChange;// Min ms of speech between volume changes
-    WebRtc_Word16       activeSpeech;       // Milliseconds of active speech
-    WebRtc_Word16       muteGuardMs;        // Counter to prevent mute action
-    WebRtc_Word16       inQueue;            // 10 ms batch indicator
-
-    // Microphone level variables
-    WebRtc_Word32       micRef;             // Remember ref. mic level for virtual mic
-    WebRtc_UWord16      gainTableIdx;       // Current position in virtual gain table
-    WebRtc_Word32       micGainIdx;         // Gain index of mic level to increase slowly
-    WebRtc_Word32       micVol;             // Remember volume between frames
-    WebRtc_Word32       maxLevel;           // Max possible vol level, incl dig gain
-    WebRtc_Word32       maxAnalog;          // Maximum possible analog volume level
-    WebRtc_Word32       maxInit;            // Initial value of "max"
-    WebRtc_Word32       minLevel;           // Minimum possible volume level
-    WebRtc_Word32       minOutput;          // Minimum output volume level
-    WebRtc_Word32       zeroCtrlMax;        // Remember max gain => don't amp low input
-
-    WebRtc_Word16       scale;              // Scale factor for internal volume levels
-#ifdef MIC_LEVEL_FEEDBACK
-    WebRtc_Word16       numBlocksMicLvlSat;
-    WebRtc_UWord8 micLvlSat;
-#endif
-    // Structs for VAD and digital_agc
-    AgcVad_t            vadMic;
-    DigitalAgc_t        digitalAgc;
-
-#ifdef AGC_DEBUG
-    FILE*               fpt;
-    FILE*               agcLog;
-    WebRtc_Word32       fcount;
-#endif
-
-    WebRtc_Word16       lowLevelSignal;
-} Agc_t;
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
diff --git a/src/modules/audio_processing/agc/main/source/digital_agc.c b/src/modules/audio_processing/agc/main/source/digital_agc.c
deleted file mode 100644
index 2966586..0000000
--- a/src/modules/audio_processing/agc/main/source/digital_agc.c
+++ /dev/null
@@ -1,780 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/* digital_agc.c
- *
- */
-
-#include <string.h>
-#ifdef AGC_DEBUG
-#include <stdio.h>
-#endif
-#include "digital_agc.h"
-#include "gain_control.h"
-
-// To generate the gaintable, copy&paste the following lines to a Matlab window:
-// MaxGain = 6; MinGain = 0; CompRatio = 3; Knee = 1;
-// zeros = 0:31; lvl = 2.^(1-zeros);
-// A = -10*log10(lvl) * (CompRatio - 1) / CompRatio;
-// B = MaxGain - MinGain;
-// gains = round(2^16*10.^(0.05 * (MinGain + B * ( log(exp(-Knee*A)+exp(-Knee*B)) - log(1+exp(-Knee*B)) ) / log(1/(1+exp(Knee*B))))));
-// fprintf(1, '\t%i, %i, %i, %i,\n', gains);
-// % Matlab code for plotting the gain and input/output level characteristic (copy/paste the following 3 lines):
-// in = 10*log10(lvl); out = 20*log10(gains/65536);
-// subplot(121); plot(in, out); axis([-30, 0, -5, 20]); grid on; xlabel('Input (dB)'); ylabel('Gain (dB)');
-// subplot(122); plot(in, in+out); axis([-30, 0, -30, 5]); grid on; xlabel('Input (dB)'); ylabel('Output (dB)');
-// zoom on;
-
-// Generator table for y=log2(1+e^x) in Q8.
-static const WebRtc_UWord16 kGenFuncTable[128] = {
-          256,   485,   786,  1126,  1484,  1849,  2217,  2586,
-         2955,  3324,  3693,  4063,  4432,  4801,  5171,  5540,
-         5909,  6279,  6648,  7017,  7387,  7756,  8125,  8495,
-         8864,  9233,  9603,  9972, 10341, 10711, 11080, 11449,
-        11819, 12188, 12557, 12927, 13296, 13665, 14035, 14404,
-        14773, 15143, 15512, 15881, 16251, 16620, 16989, 17359,
-        17728, 18097, 18466, 18836, 19205, 19574, 19944, 20313,
-        20682, 21052, 21421, 21790, 22160, 22529, 22898, 23268,
-        23637, 24006, 24376, 24745, 25114, 25484, 25853, 26222,
-        26592, 26961, 27330, 27700, 28069, 28438, 28808, 29177,
-        29546, 29916, 30285, 30654, 31024, 31393, 31762, 32132,
-        32501, 32870, 33240, 33609, 33978, 34348, 34717, 35086,
-        35456, 35825, 36194, 36564, 36933, 37302, 37672, 38041,
-        38410, 38780, 39149, 39518, 39888, 40257, 40626, 40996,
-        41365, 41734, 42104, 42473, 42842, 43212, 43581, 43950,
-        44320, 44689, 45058, 45428, 45797, 46166, 46536, 46905
-};
-
-static const WebRtc_Word16 kAvgDecayTime = 250; // frames; < 3000
-
-WebRtc_Word32 WebRtcAgc_CalculateGainTable(WebRtc_Word32 *gainTable, // Q16
-                                           WebRtc_Word16 digCompGaindB, // Q0
-                                           WebRtc_Word16 targetLevelDbfs,// Q0
-                                           WebRtc_UWord8 limiterEnable,
-                                           WebRtc_Word16 analogTarget) // Q0
-{
-    // This function generates the compressor gain table used in the fixed digital part.
-    WebRtc_UWord32 tmpU32no1, tmpU32no2, absInLevel, logApprox;
-    WebRtc_Word32 inLevel, limiterLvl;
-    WebRtc_Word32 tmp32, tmp32no1, tmp32no2, numFIX, den, y32;
-    const WebRtc_UWord16 kLog10 = 54426; // log2(10)     in Q14
-    const WebRtc_UWord16 kLog10_2 = 49321; // 10*log10(2)  in Q14
-    const WebRtc_UWord16 kLogE_1 = 23637; // log2(e)      in Q14
-    WebRtc_UWord16 constMaxGain;
-    WebRtc_UWord16 tmpU16, intPart, fracPart;
-    const WebRtc_Word16 kCompRatio = 3;
-    const WebRtc_Word16 kSoftLimiterLeft = 1;
-    WebRtc_Word16 limiterOffset = 0; // Limiter offset
-    WebRtc_Word16 limiterIdx, limiterLvlX;
-    WebRtc_Word16 constLinApprox, zeroGainLvl, maxGain, diffGain;
-    WebRtc_Word16 i, tmp16, tmp16no1;
-    int zeros, zerosScale;
-
-    // Constants
-//    kLogE_1 = 23637; // log2(e)      in Q14
-//    kLog10 = 54426; // log2(10)     in Q14
-//    kLog10_2 = 49321; // 10*log10(2)  in Q14
-
-    // Calculate maximum digital gain and zero gain level
-    tmp32no1 = WEBRTC_SPL_MUL_16_16(digCompGaindB - analogTarget, kCompRatio - 1);
-    tmp16no1 = analogTarget - targetLevelDbfs;
-    tmp16no1 += WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
-    maxGain = WEBRTC_SPL_MAX(tmp16no1, (analogTarget - targetLevelDbfs));
-    tmp32no1 = WEBRTC_SPL_MUL_16_16(maxGain, kCompRatio);
-    zeroGainLvl = digCompGaindB;
-    zeroGainLvl -= WebRtcSpl_DivW32W16ResW16(tmp32no1 + ((kCompRatio - 1) >> 1),
-                                             kCompRatio - 1);
-    if ((digCompGaindB <= analogTarget) && (limiterEnable))
-    {
-        zeroGainLvl += (analogTarget - digCompGaindB + kSoftLimiterLeft);
-        limiterOffset = 0;
-    }
-
-    // Calculate the difference between maximum gain and gain at 0dB0v:
-    //  diffGain = maxGain + (compRatio-1)*zeroGainLvl/compRatio
-    //           = (compRatio-1)*digCompGaindB/compRatio
-    tmp32no1 = WEBRTC_SPL_MUL_16_16(digCompGaindB, kCompRatio - 1);
-    diffGain = WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
-    if (diffGain < 0)
-    {
-        return -1;
-    }
-
-    // Calculate the limiter level and index:
-    //  limiterLvlX = analogTarget - limiterOffset
-    //  limiterLvl  = targetLevelDbfs + limiterOffset/compRatio
-    limiterLvlX = analogTarget - limiterOffset;
-    limiterIdx = 2
-            + WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)limiterLvlX, 13),
-                                        WEBRTC_SPL_RSHIFT_U16(kLog10_2, 1));
-    tmp16no1 = WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio);
-    limiterLvl = targetLevelDbfs + tmp16no1;
-
-    // Calculate (through table lookup):
-    //  constMaxGain = log2(1+2^(log2(e)*diffGain)); (in Q8)
-    constMaxGain = kGenFuncTable[diffGain]; // in Q8
-
-    // Calculate a parameter used to approximate the fractional part of 2^x with a
-    // piecewise linear function in Q14:
-    //  constLinApprox = round(3/2*(4*(3-2*sqrt(2))/(log(2)^2)-0.5)*2^14);
-    constLinApprox = 22817; // in Q14
-
-    // Calculate a denominator used in the exponential part to convert from dB to linear scale:
-    //  den = 20*constMaxGain (in Q8)
-    den = WEBRTC_SPL_MUL_16_U16(20, constMaxGain); // in Q8
-
-    for (i = 0; i < 32; i++)
-    {
-        // Calculate scaled input level (compressor):
-        //  inLevel = fix((-constLog10_2*(compRatio-1)*(1-i)+fix(compRatio/2))/compRatio)
-        tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(kCompRatio - 1, i - 1); // Q0
-        tmp32 = WEBRTC_SPL_MUL_16_U16(tmp16, kLog10_2) + 1; // Q14
-        inLevel = WebRtcSpl_DivW32W16(tmp32, kCompRatio); // Q14
-
-        // Calculate diffGain-inLevel, to map using the genFuncTable
-        inLevel = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)diffGain, 14) - inLevel; // Q14
-
-        // Make calculations on abs(inLevel) and compensate for the sign afterwards.
-        absInLevel = (WebRtc_UWord32)WEBRTC_SPL_ABS_W32(inLevel); // Q14
-
-        // LUT with interpolation
-        intPart = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(absInLevel, 14);
-        fracPart = (WebRtc_UWord16)(absInLevel & 0x00003FFF); // extract the fractional part
-        tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8
-        tmpU32no1 = WEBRTC_SPL_UMUL_16_16(tmpU16, fracPart); // Q22
-        tmpU32no1 += WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)kGenFuncTable[intPart], 14); // Q22
-        logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 8); // Q14
-        // Compensate for negative exponent using the relation:
-        //  log2(1 + 2^-x) = log2(1 + 2^x) - x
-        if (inLevel < 0)
-        {
-            zeros = WebRtcSpl_NormU32(absInLevel);
-            zerosScale = 0;
-            if (zeros < 15)
-            {
-                // Not enough space for multiplication
-                tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(absInLevel, 15 - zeros); // Q(zeros-1)
-                tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no2, kLogE_1); // Q(zeros+13)
-                if (zeros < 9)
-                {
-                    tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 9 - zeros); // Q(zeros+13)
-                    zerosScale = 9 - zeros;
-                } else
-                {
-                    tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, zeros - 9); // Q22
-                }
-            } else
-            {
-                tmpU32no2 = WEBRTC_SPL_UMUL_32_16(absInLevel, kLogE_1); // Q28
-                tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6); // Q22
-            }
-            logApprox = 0;
-            if (tmpU32no2 < tmpU32no1)
-            {
-                logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1 - tmpU32no2, 8 - zerosScale); //Q14
-            }
-        }
-        numFIX = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_U16(maxGain, constMaxGain), 6); // Q14
-        numFIX -= WEBRTC_SPL_MUL_32_16((WebRtc_Word32)logApprox, diffGain); // Q14
-
-        // Calculate ratio
-        // Shift numFIX as much as possible
-        zeros = WebRtcSpl_NormW32(numFIX);
-        numFIX = WEBRTC_SPL_LSHIFT_W32(numFIX, zeros); // Q(14+zeros)
-
-        // Shift den so we end up in Qy1
-        tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 8); // Q(zeros)
-        if (numFIX < 0)
-        {
-            numFIX -= WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
-        } else
-        {
-            numFIX += WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
-        }
-        y32 = WEBRTC_SPL_DIV(numFIX, tmp32no1); // in Q14
-        if (limiterEnable && (i < limiterIdx))
-        {
-            tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14
-            tmp32 -= WEBRTC_SPL_LSHIFT_W32(limiterLvl, 14); // Q14
-            y32 = WebRtcSpl_DivW32W16(tmp32 + 10, 20);
-        }
-        if (y32 > 39000)
-        {
-            tmp32 = WEBRTC_SPL_MUL(y32 >> 1, kLog10) + 4096; // in Q27
-            tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 13); // in Q14
-        } else
-        {
-            tmp32 = WEBRTC_SPL_MUL(y32, kLog10) + 8192; // in Q28
-            tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 14); // in Q14
-        }
-        tmp32 += WEBRTC_SPL_LSHIFT_W32(16, 14); // in Q14 (Make sure final output is in Q16)
-
-        // Calculate power
-        if (tmp32 > 0)
-        {
-            intPart = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 14);
-            fracPart = (WebRtc_UWord16)(tmp32 & 0x00003FFF); // in Q14
-            if (WEBRTC_SPL_RSHIFT_W32(fracPart, 13))
-            {
-                tmp16 = WEBRTC_SPL_LSHIFT_W16(2, 14) - constLinApprox;
-                tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - fracPart;
-                tmp32no2 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16);
-                tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
-                tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - tmp32no2;
-            } else
-            {
-                tmp16 = constLinApprox - WEBRTC_SPL_LSHIFT_W16(1, 14);
-                tmp32no2 = WEBRTC_SPL_MUL_32_16(fracPart, tmp16);
-                tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
-            }
-            fracPart = (WebRtc_UWord16)tmp32no2;
-            gainTable[i] = WEBRTC_SPL_LSHIFT_W32(1, intPart)
-                    + WEBRTC_SPL_SHIFT_W32(fracPart, intPart - 14);
-        } else
-        {
-            gainTable[i] = 0;
-        }
-    }
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAgc_InitDigital(DigitalAgc_t *stt, WebRtc_Word16 agcMode)
-{
-
-    if (agcMode == kAgcModeFixedDigital)
-    {
-        // start at minimum to find correct gain faster
-        stt->capacitorSlow = 0;
-    } else
-    {
-        // start out with 0 dB gain
-        stt->capacitorSlow = 134217728; // (WebRtc_Word32)(0.125f * 32768.0f * 32768.0f);
-    }
-    stt->capacitorFast = 0;
-    stt->gain = 65536;
-    stt->gatePrevious = 0;
-    stt->agcMode = agcMode;
-#ifdef AGC_DEBUG
-    stt->frameCounter = 0;
-#endif
-
-    // initialize VADs
-    WebRtcAgc_InitVad(&stt->vadNearend);
-    WebRtcAgc_InitVad(&stt->vadFarend);
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAgc_AddFarendToDigital(DigitalAgc_t *stt, const WebRtc_Word16 *in_far,
-                                           WebRtc_Word16 nrSamples)
-{
-    // Check for valid pointer
-    if (&stt->vadFarend == NULL)
-    {
-        return -1;
-    }
-
-    // VAD for far end
-    WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples);
-
-    return 0;
-}
-
-WebRtc_Word32 WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const WebRtc_Word16 *in_near,
-                                       const WebRtc_Word16 *in_near_H, WebRtc_Word16 *out,
-                                       WebRtc_Word16 *out_H, WebRtc_UWord32 FS,
-                                       WebRtc_Word16 lowlevelSignal)
-{
-    // array for gains (one value per ms, incl start & end)
-    WebRtc_Word32 gains[11];
-
-    WebRtc_Word32 out_tmp, tmp32;
-    WebRtc_Word32 env[10];
-    WebRtc_Word32 nrg, max_nrg;
-    WebRtc_Word32 cur_level;
-    WebRtc_Word32 gain32, delta;
-    WebRtc_Word16 logratio;
-    WebRtc_Word16 lower_thr, upper_thr;
-    WebRtc_Word16 zeros, zeros_fast, frac;
-    WebRtc_Word16 decay;
-    WebRtc_Word16 gate, gain_adj;
-    WebRtc_Word16 k, n;
-    WebRtc_Word16 L, L2; // samples/subframe
-
-    // determine number of samples per ms
-    if (FS == 8000)
-    {
-        L = 8;
-        L2 = 3;
-    } else if (FS == 16000)
-    {
-        L = 16;
-        L2 = 4;
-    } else if (FS == 32000)
-    {
-        L = 16;
-        L2 = 4;
-    } else
-    {
-        return -1;
-    }
-
-    memcpy(out, in_near, 10 * L * sizeof(WebRtc_Word16));
-    if (FS == 32000)
-    {
-        memcpy(out_H, in_near_H, 10 * L * sizeof(WebRtc_Word16));
-    }
-    // VAD for near end
-    logratio = WebRtcAgc_ProcessVad(&stt->vadNearend, out, L * 10);
-
-    // Account for far end VAD
-    if (stt->vadFarend.counter > 10)
-    {
-        tmp32 = WEBRTC_SPL_MUL_16_16(3, logratio);
-        logratio = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 - stt->vadFarend.logRatio, 2);
-    }
-
-    // Determine decay factor depending on VAD
-    //  upper_thr = 1.0f;
-    //  lower_thr = 0.25f;
-    upper_thr = 1024; // Q10
-    lower_thr = 0; // Q10
-    if (logratio > upper_thr)
-    {
-        // decay = -2^17 / DecayTime;  ->  -65
-        decay = -65;
-    } else if (logratio < lower_thr)
-    {
-        decay = 0;
-    } else
-    {
-        // decay = (WebRtc_Word16)(((lower_thr - logratio)
-        //       * (2^27/(DecayTime*(upper_thr-lower_thr)))) >> 10);
-        // SUBSTITUTED: 2^27/(DecayTime*(upper_thr-lower_thr))  ->  65
-        tmp32 = WEBRTC_SPL_MUL_16_16((lower_thr - logratio), 65);
-        decay = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 10);
-    }
-
-    // adjust decay factor for long silence (detected as low standard deviation)
-    // This is only done in the adaptive modes
-    if (stt->agcMode != kAgcModeFixedDigital)
-    {
-        if (stt->vadNearend.stdLongTerm < 4000)
-        {
-            decay = 0;
-        } else if (stt->vadNearend.stdLongTerm < 8096)
-        {
-            // decay = (WebRtc_Word16)(((stt->vadNearend.stdLongTerm - 4000) * decay) >> 12);
-            tmp32 = WEBRTC_SPL_MUL_16_16((stt->vadNearend.stdLongTerm - 4000), decay);
-            decay = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
-        }
-
-        if (lowlevelSignal != 0)
-        {
-            decay = 0;
-        }
-    }
-#ifdef AGC_DEBUG
-    stt->frameCounter++;
-    fprintf(stt->logFile, "%5.2f\t%d\t%d\t%d\t", (float)(stt->frameCounter) / 100, logratio, decay, stt->vadNearend.stdLongTerm);
-#endif
-    // Find max amplitude per sub frame
-    // iterate over sub frames
-    for (k = 0; k < 10; k++)
-    {
-        // iterate over samples
-        max_nrg = 0;
-        for (n = 0; n < L; n++)
-        {
-            nrg = WEBRTC_SPL_MUL_16_16(out[k * L + n], out[k * L + n]);
-            if (nrg > max_nrg)
-            {
-                max_nrg = nrg;
-            }
-        }
-        env[k] = max_nrg;
-    }
-
-    // Calculate gain per sub frame
-    gains[0] = stt->gain;
-    for (k = 0; k < 10; k++)
-    {
-        // Fast envelope follower
-        //  decay time = -131000 / -1000 = 131 (ms)
-        stt->capacitorFast = AGC_SCALEDIFF32(-1000, stt->capacitorFast, stt->capacitorFast);
-        if (env[k] > stt->capacitorFast)
-        {
-            stt->capacitorFast = env[k];
-        }
-        // Slow envelope follower
-        if (env[k] > stt->capacitorSlow)
-        {
-            // increase capacitorSlow
-            stt->capacitorSlow
-                    = AGC_SCALEDIFF32(500, (env[k] - stt->capacitorSlow), stt->capacitorSlow);
-        } else
-        {
-            // decrease capacitorSlow
-            stt->capacitorSlow
-                    = AGC_SCALEDIFF32(decay, stt->capacitorSlow, stt->capacitorSlow);
-        }
-
-        // use maximum of both capacitors as current level
-        if (stt->capacitorFast > stt->capacitorSlow)
-        {
-            cur_level = stt->capacitorFast;
-        } else
-        {
-            cur_level = stt->capacitorSlow;
-        }
-        // Translate signal level into gain, using a piecewise linear approximation
-        // find number of leading zeros
-        zeros = WebRtcSpl_NormU32((WebRtc_UWord32)cur_level);
-        if (cur_level == 0)
-        {
-            zeros = 31;
-        }
-        tmp32 = (WEBRTC_SPL_LSHIFT_W32(cur_level, zeros) & 0x7FFFFFFF);
-        frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 19); // Q12
-        tmp32 = WEBRTC_SPL_MUL((stt->gainTable[zeros-1] - stt->gainTable[zeros]), frac);
-        gains[k + 1] = stt->gainTable[zeros] + WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
-#ifdef AGC_DEBUG
-        if (k == 0)
-        {
-            fprintf(stt->logFile, "%d\t%d\t%d\t%d\t%d\n", env[0], cur_level, stt->capacitorFast, stt->capacitorSlow, zeros);
-        }
-#endif
-    }
-
-    // Gate processing (lower gain during absence of speech)
-    zeros = WEBRTC_SPL_LSHIFT_W16(zeros, 9) - WEBRTC_SPL_RSHIFT_W16(frac, 3);
-    // find number of leading zeros
-    zeros_fast = WebRtcSpl_NormU32((WebRtc_UWord32)stt->capacitorFast);
-    if (stt->capacitorFast == 0)
-    {
-        zeros_fast = 31;
-    }
-    tmp32 = (WEBRTC_SPL_LSHIFT_W32(stt->capacitorFast, zeros_fast) & 0x7FFFFFFF);
-    zeros_fast = WEBRTC_SPL_LSHIFT_W16(zeros_fast, 9);
-    zeros_fast -= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 22);
-
-    gate = 1000 + zeros_fast - zeros - stt->vadNearend.stdShortTerm;
-
-    if (gate < 0)
-    {
-        stt->gatePrevious = 0;
-    } else
-    {
-        tmp32 = WEBRTC_SPL_MUL_16_16(stt->gatePrevious, 7);
-        gate = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)gate + tmp32, 3);
-        stt->gatePrevious = gate;
-    }
-    // gate < 0     -> no gate
-    // gate > 2500  -> max gate
-    if (gate > 0)
-    {
-        if (gate < 2500)
-        {
-            gain_adj = WEBRTC_SPL_RSHIFT_W16(2500 - gate, 5);
-        } else
-        {
-            gain_adj = 0;
-        }
-        for (k = 0; k < 10; k++)
-        {
-            if ((gains[k + 1] - stt->gainTable[0]) > 8388608)
-            {
-                // To prevent wraparound
-                tmp32 = WEBRTC_SPL_RSHIFT_W32((gains[k+1] - stt->gainTable[0]), 8);
-                tmp32 = WEBRTC_SPL_MUL(tmp32, (178 + gain_adj));
-            } else
-            {
-                tmp32 = WEBRTC_SPL_MUL((gains[k+1] - stt->gainTable[0]), (178 + gain_adj));
-                tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 8);
-            }
-            gains[k + 1] = stt->gainTable[0] + tmp32;
-        }
-    }
-
-    // Limit gain to avoid overload distortion
-    for (k = 0; k < 10; k++)
-    {
-        // To prevent wrap around
-        zeros = 10;
-        if (gains[k + 1] > 47453132)
-        {
-            zeros = 16 - WebRtcSpl_NormW32(gains[k + 1]);
-        }
-        gain32 = WEBRTC_SPL_RSHIFT_W32(gains[k+1], zeros) + 1;
-        gain32 = WEBRTC_SPL_MUL(gain32, gain32);
-        // check for overflow
-        while (AGC_MUL32(WEBRTC_SPL_RSHIFT_W32(env[k], 12) + 1, gain32)
-                > WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)32767, 2 * (1 - zeros + 10)))
-        {
-            // multiply by 253/256 ==> -0.1 dB
-            if (gains[k + 1] > 8388607)
-            {
-                // Prevent wrap around
-                gains[k + 1] = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(gains[k+1], 8), 253);
-            } else
-            {
-                gains[k + 1] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(gains[k+1], 253), 8);
-            }
-            gain32 = WEBRTC_SPL_RSHIFT_W32(gains[k+1], zeros) + 1;
-            gain32 = WEBRTC_SPL_MUL(gain32, gain32);
-        }
-    }
-    // gain reductions should be done 1 ms earlier than gain increases
-    for (k = 1; k < 10; k++)
-    {
-        if (gains[k] > gains[k + 1])
-        {
-            gains[k] = gains[k + 1];
-        }
-    }
-    // save start gain for next frame
-    stt->gain = gains[10];
-
-    // Apply gain
-    // handle first sub frame separately
-    delta = WEBRTC_SPL_LSHIFT_W32(gains[1] - gains[0], (4 - L2));
-    gain32 = WEBRTC_SPL_LSHIFT_W32(gains[0], 4);
-    // iterate over samples
-    for (n = 0; n < L; n++)
-    {
-        // For lower band
-        tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[n], WEBRTC_SPL_RSHIFT_W32(gain32 + 127, 7));
-        out_tmp = WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-        if (out_tmp > 4095)
-        {
-            out[n] = (WebRtc_Word16)32767;
-        } else if (out_tmp < -4096)
-        {
-            out[n] = (WebRtc_Word16)-32768;
-        } else
-        {
-            tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[n], WEBRTC_SPL_RSHIFT_W32(gain32, 4));
-            out[n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-        }
-        // For higher band
-        if (FS == 32000)
-        {
-            tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[n],
-                                   WEBRTC_SPL_RSHIFT_W32(gain32 + 127, 7));
-            out_tmp = WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-            if (out_tmp > 4095)
-            {
-                out_H[n] = (WebRtc_Word16)32767;
-            } else if (out_tmp < -4096)
-            {
-                out_H[n] = (WebRtc_Word16)-32768;
-            } else
-            {
-                tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[n],
-                                       WEBRTC_SPL_RSHIFT_W32(gain32, 4));
-                out_H[n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-            }
-        }
-        //
-
-        gain32 += delta;
-    }
-    // iterate over subframes
-    for (k = 1; k < 10; k++)
-    {
-        delta = WEBRTC_SPL_LSHIFT_W32(gains[k+1] - gains[k], (4 - L2));
-        gain32 = WEBRTC_SPL_LSHIFT_W32(gains[k], 4);
-        // iterate over samples
-        for (n = 0; n < L; n++)
-        {
-            // For lower band
-            tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[k * L + n],
-                                   WEBRTC_SPL_RSHIFT_W32(gain32, 4));
-            out[k * L + n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-            // For higher band
-            if (FS == 32000)
-            {
-                tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[k * L + n],
-                                       WEBRTC_SPL_RSHIFT_W32(gain32, 4));
-                out_H[k * L + n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
-            }
-            gain32 += delta;
-        }
-    }
-
-    return 0;
-}
-
-void WebRtcAgc_InitVad(AgcVad_t *state)
-{
-    WebRtc_Word16 k;
-
-    state->HPstate = 0; // state of high pass filter
-    state->logRatio = 0; // log( P(active) / P(inactive) )
-    // average input level (Q10)
-    state->meanLongTerm = WEBRTC_SPL_LSHIFT_W16(15, 10);
-
-    // variance of input level (Q8)
-    state->varianceLongTerm = WEBRTC_SPL_LSHIFT_W32(500, 8);
-
-    state->stdLongTerm = 0; // standard deviation of input level in dB
-    // short-term average input level (Q10)
-    state->meanShortTerm = WEBRTC_SPL_LSHIFT_W16(15, 10);
-
-    // short-term variance of input level (Q8)
-    state->varianceShortTerm = WEBRTC_SPL_LSHIFT_W32(500, 8);
-
-    state->stdShortTerm = 0; // short-term standard deviation of input level in dB
-    state->counter = 3; // counts updates
-    for (k = 0; k < 8; k++)
-    {
-        // downsampling filter
-        state->downState[k] = 0;
-    }
-}
-
-WebRtc_Word16 WebRtcAgc_ProcessVad(AgcVad_t *state, // (i) VAD state
-                                   const WebRtc_Word16 *in, // (i) Speech signal
-                                   WebRtc_Word16 nrSamples) // (i) number of samples
-{
-    WebRtc_Word32 out, nrg, tmp32, tmp32b;
-    WebRtc_UWord16 tmpU16;
-    WebRtc_Word16 k, subfr, tmp16;
-    WebRtc_Word16 buf1[8];
-    WebRtc_Word16 buf2[4];
-    WebRtc_Word16 HPstate;
-    WebRtc_Word16 zeros, dB;
-    WebRtc_Word16 *buf1_ptr;
-
-    // process in 10 sub frames of 1 ms (to save on memory)
-    nrg = 0;
-    buf1_ptr = &buf1[0];
-    HPstate = state->HPstate;
-    for (subfr = 0; subfr < 10; subfr++)
-    {
-        // downsample to 4 kHz
-        if (nrSamples == 160)
-        {
-            for (k = 0; k < 8; k++)
-            {
-                tmp32 = (WebRtc_Word32)in[2 * k] + (WebRtc_Word32)in[2 * k + 1];
-                tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
-                buf1[k] = (WebRtc_Word16)tmp32;
-            }
-            in += 16;
-
-            WebRtcSpl_DownsampleBy2(buf1, 8, buf2, state->downState);
-        } else
-        {
-            WebRtcSpl_DownsampleBy2(in, 8, buf2, state->downState);
-            in += 8;
-        }
-
-        // high pass filter and compute energy
-        for (k = 0; k < 4; k++)
-        {
-            out = buf2[k] + HPstate;
-            tmp32 = WEBRTC_SPL_MUL(600, out);
-            HPstate = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(tmp32, 10) - buf2[k]);
-            tmp32 = WEBRTC_SPL_MUL(out, out);
-            nrg += WEBRTC_SPL_RSHIFT_W32(tmp32, 6);
-        }
-    }
-    state->HPstate = HPstate;
-
-    // find number of leading zeros
-    if (!(0xFFFF0000 & nrg))
-    {
-        zeros = 16;
-    } else
-    {
-        zeros = 0;
-    }
-    if (!(0xFF000000 & (nrg << zeros)))
-    {
-        zeros += 8;
-    }
-    if (!(0xF0000000 & (nrg << zeros)))
-    {
-        zeros += 4;
-    }
-    if (!(0xC0000000 & (nrg << zeros)))
-    {
-        zeros += 2;
-    }
-    if (!(0x80000000 & (nrg << zeros)))
-    {
-        zeros += 1;
-    }
-
-    // energy level (range {-32..30}) (Q10)
-    dB = WEBRTC_SPL_LSHIFT_W16(15 - zeros, 11);
-
-    // Update statistics
-
-    if (state->counter < kAvgDecayTime)
-    {
-        // decay time = AvgDecTime * 10 ms
-        state->counter++;
-    }
-
-    // update short-term estimate of mean energy level (Q10)
-    tmp32 = (WEBRTC_SPL_MUL_16_16(state->meanShortTerm, 15) + (WebRtc_Word32)dB);
-    state->meanShortTerm = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
-
-    // update short-term estimate of variance in energy level (Q8)
-    tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(dB, dB), 12);
-    tmp32 += WEBRTC_SPL_MUL(state->varianceShortTerm, 15);
-    state->varianceShortTerm = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
-
-    // update short-term estimate of standard deviation in energy level (Q10)
-    tmp32 = WEBRTC_SPL_MUL_16_16(state->meanShortTerm, state->meanShortTerm);
-    tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceShortTerm, 12) - tmp32;
-    state->stdShortTerm = (WebRtc_Word16)WebRtcSpl_Sqrt(tmp32);
-
-    // update long-term estimate of mean energy level (Q10)
-    tmp32 = WEBRTC_SPL_MUL_16_16(state->meanLongTerm, state->counter) + (WebRtc_Word32)dB;
-    state->meanLongTerm = WebRtcSpl_DivW32W16ResW16(tmp32,
-                                                    WEBRTC_SPL_ADD_SAT_W16(state->counter, 1));
-
-    // update long-term estimate of variance in energy level (Q8)
-    tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(dB, dB), 12);
-    tmp32 += WEBRTC_SPL_MUL(state->varianceLongTerm, state->counter);
-    state->varianceLongTerm = WebRtcSpl_DivW32W16(tmp32,
-                                                  WEBRTC_SPL_ADD_SAT_W16(state->counter, 1));
-
-    // update long-term estimate of standard deviation in energy level (Q10)
-    tmp32 = WEBRTC_SPL_MUL_16_16(state->meanLongTerm, state->meanLongTerm);
-    tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceLongTerm, 12) - tmp32;
-    state->stdLongTerm = (WebRtc_Word16)WebRtcSpl_Sqrt(tmp32);
-
-    // update voice activity measure (Q10)
-    tmp16 = WEBRTC_SPL_LSHIFT_W16(3, 12);
-    tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, (dB - state->meanLongTerm));
-    tmp32 = WebRtcSpl_DivW32W16(tmp32, state->stdLongTerm);
-    tmpU16 = WEBRTC_SPL_LSHIFT_U16((WebRtc_UWord16)13, 12);
-    tmp32b = WEBRTC_SPL_MUL_16_U16(state->logRatio, tmpU16);
-    tmp32 += WEBRTC_SPL_RSHIFT_W32(tmp32b, 10);
-
-    state->logRatio = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 6);
-
-    // limit
-    if (state->logRatio > 2048)
-    {
-        state->logRatio = 2048;
-    }
-    if (state->logRatio < -2048)
-    {
-        state->logRatio = -2048;
-    }
-
-    return state->logRatio; // Q10
-}
diff --git a/src/modules/audio_processing/agc/main/source/digital_agc.h b/src/modules/audio_processing/agc/main/source/digital_agc.h
deleted file mode 100644
index 240b220..0000000
--- a/src/modules/audio_processing/agc/main/source/digital_agc.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
-
-#ifdef AGC_DEBUG
-#include <stdio.h>
-#endif
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-// the 32 most significant bits of A(19) * B(26) >> 13
-#define AGC_MUL32(A, B)             (((B)>>13)*(A) + ( ((0x00001FFF & (B))*(A)) >> 13 ))
-// C + the 32 most significant bits of A * B
-#define AGC_SCALEDIFF32(A, B, C)    ((C) + ((B)>>16)*(A) + ( ((0x0000FFFF & (B))*(A)) >> 16 ))
-
-typedef struct
-{
-    WebRtc_Word32 downState[8];
-    WebRtc_Word16 HPstate;
-    WebRtc_Word16 counter;
-    WebRtc_Word16 logRatio; // log( P(active) / P(inactive) ) (Q10)
-    WebRtc_Word16 meanLongTerm; // Q10
-    WebRtc_Word32 varianceLongTerm; // Q8
-    WebRtc_Word16 stdLongTerm; // Q10
-    WebRtc_Word16 meanShortTerm; // Q10
-    WebRtc_Word32 varianceShortTerm; // Q8
-    WebRtc_Word16 stdShortTerm; // Q10
-} AgcVad_t; // total = 54 bytes
-
-typedef struct
-{
-    WebRtc_Word32 capacitorSlow;
-    WebRtc_Word32 capacitorFast;
-    WebRtc_Word32 gain;
-    WebRtc_Word32 gainTable[32];
-    WebRtc_Word16 gatePrevious;
-    WebRtc_Word16 agcMode;
-    AgcVad_t      vadNearend;
-    AgcVad_t      vadFarend;
-#ifdef AGC_DEBUG
-    FILE*         logFile;
-    int           frameCounter;
-#endif
-} DigitalAgc_t;
-
-WebRtc_Word32 WebRtcAgc_InitDigital(DigitalAgc_t *digitalAgcInst, WebRtc_Word16 agcMode);
-
-WebRtc_Word32 WebRtcAgc_ProcessDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inNear,
-                             const WebRtc_Word16 *inNear_H, WebRtc_Word16 *out,
-                             WebRtc_Word16 *out_H, WebRtc_UWord32 FS,
-                             WebRtc_Word16 lowLevelSignal);
-
-WebRtc_Word32 WebRtcAgc_AddFarendToDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inFar,
-                                 WebRtc_Word16 nrSamples);
-
-void WebRtcAgc_InitVad(AgcVad_t *vadInst);
-
-WebRtc_Word16 WebRtcAgc_ProcessVad(AgcVad_t *vadInst, // (i) VAD state
-                            const WebRtc_Word16 *in, // (i) Speech signal
-                            WebRtc_Word16 nrSamples); // (i) number of samples
-
-WebRtc_Word32 WebRtcAgc_CalculateGainTable(WebRtc_Word32 *gainTable, // Q16
-                                 WebRtc_Word16 compressionGaindB, // Q0 (in dB)
-                                 WebRtc_Word16 targetLevelDbfs,// Q0 (in dB)
-                                 WebRtc_UWord8 limiterEnable, WebRtc_Word16 analogTarget);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
diff --git a/src/modules/audio_processing/main/apm_tests.gyp b/src/modules/audio_processing/main/apm_tests.gyp
deleted file mode 100644
index 441abeb..0000000
--- a/src/modules/audio_processing/main/apm_tests.gyp
+++ /dev/null
@@ -1,60 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../../common_settings.gypi',
-  ],
-  'targets': [
-    {
-      'target_name': 'unit_test',
-      'type': 'executable',
-      'dependencies': [
-        'source/apm.gyp:audio_processing',
-        '../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
-        '../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
-
-        '../../../../testing/gtest.gyp:gtest',
-        '../../../../testing/gtest.gyp:gtest_main',
-        '../../../../third_party/protobuf/protobuf.gyp:protobuf_lite',
-      ],
-      'include_dirs': [
-        '../../../../testing/gtest/include',
-      ],
-      'sources': [
-        'test/unit_test/unit_test.cc',
-        'test/unit_test/audio_processing_unittest.pb.cc',
-        'test/unit_test/audio_processing_unittest.pb.h',
-      ],
-    },
-    {
-      'target_name': 'process_test',
-      'type': 'executable',
-      'dependencies': [
-        'source/apm.gyp:audio_processing',
-        '../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
-
-        '../../../../testing/gtest.gyp:gtest',
-        '../../../../testing/gtest.gyp:gtest_main',
-      ],
-      'include_dirs': [
-        '../../../../testing/gtest/include',
-      ],
-      'sources': [
-        'test/process_test/process_test.cc',
-      ],
-    },
-
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/main/interface/audio_processing.h b/src/modules/audio_processing/main/interface/audio_processing.h
deleted file mode 100644
index dc9c232..0000000
--- a/src/modules/audio_processing/main/interface/audio_processing.h
+++ /dev/null
@@ -1,564 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_
-
-#include "typedefs.h"
-#include "module.h"
-
-namespace webrtc {
-
-class AudioFrame;
-class EchoCancellation;
-class EchoControlMobile;
-class GainControl;
-class HighPassFilter;
-class LevelEstimator;
-class NoiseSuppression;
-class VoiceDetection;
-
-// The Audio Processing Module (APM) provides a collection of voice processing
-// components designed for real-time communications software.
-//
-// APM operates on two audio streams on a frame-by-frame basis. Frames of the
-// primary stream, on which all processing is applied, are passed to
-// |ProcessStream()|. Frames of the reverse direction stream, which are used for
-// analysis by some components, are passed to |AnalyzeReverseStream()|. On the
-// client-side, this will typically be the near-end (capture) and far-end
-// (render) streams, respectively. APM should be placed in the signal chain as
-// close to the audio hardware abstraction layer (HAL) as possible.
-//
-// On the server-side, the reverse stream will normally not be used, with
-// processing occurring on each incoming stream.
-//
-// Component interfaces follow a similar pattern and are accessed through
-// corresponding getters in APM. All components are disabled at create-time,
-// with default settings that are recommended for most situations. New settings
-// can be applied without enabling a component. Enabling a component triggers
-// memory allocation and initialization to allow it to start processing the
-// streams.
-//
-// Thread safety is provided with the following assumptions to reduce locking
-// overhead:
-//   1. The stream getters and setters are called from the same thread as
-//      ProcessStream(). More precisely, stream functions are never called
-//      concurrently with ProcessStream().
-//   2. Parameter getters are never called concurrently with the corresponding
-//      setter.
-//
-// APM accepts only 16-bit linear PCM audio data in frames of 10 ms. Multiple
-// channels should be interleaved.
-//
-// Usage example, omitting error checking:
-// AudioProcessing* apm = AudioProcessing::Create(0);
-// apm->set_sample_rate_hz(32000); // Super-wideband processing.
-//
-// // Mono capture and stereo render.
-// apm->set_num_channels(1, 1);
-// apm->set_num_reverse_channels(2);
-//
-// apm->high_pass_filter()->Enable(true);
-//
-// apm->echo_cancellation()->enable_drift_compensation(false);
-// apm->echo_cancellation()->Enable(true);
-//
-// apm->noise_reduction()->set_level(kHighSuppression);
-// apm->noise_reduction()->Enable(true);
-//
-// apm->gain_control()->set_analog_level_limits(0, 255);
-// apm->gain_control()->set_mode(kAdaptiveAnalog);
-// apm->gain_control()->Enable(true);
-//
-// apm->voice_detection()->Enable(true);
-//
-// // Start a voice call...
-//
-// // ... Render frame arrives bound for the audio HAL ...
-// apm->AnalyzeReverseStream(render_frame);
-//
-// // ... Capture frame arrives from the audio HAL ...
-// // Call required set_stream_ functions.
-// apm->set_stream_delay_ms(delay_ms);
-// apm->gain_control()->set_stream_analog_level(analog_level);
-//
-// apm->ProcessStream(capture_frame);
-//
-// // Call required stream_ functions.
-// analog_level = apm->gain_control()->stream_analog_level();
-// has_voice = apm->stream_has_voice();
-//
-// // Repeate render and capture processing for the duration of the call...
-// // Start a new call...
-// apm->Initialize();
-//
-// // Close the application...
-// AudioProcessing::Destroy(apm);
-// apm = NULL;
-//
-class AudioProcessing : public Module {
- public:
-  // Creates a APM instance, with identifier |id|. Use one instance for every
-  // primary audio stream requiring processing. On the client-side, this would
-  // typically be one instance for the near-end stream, and additional instances
-  // for each far-end stream which requires processing. On the server-side,
-  // this would typically be one instance for every incoming stream.
-  static AudioProcessing* Create(int id);
-
-  // Destroys a |apm| instance.
-  static void Destroy(AudioProcessing* apm);
-
-  // Initializes internal states, while retaining all user settings. This
-  // should be called before beginning to process a new audio stream. However,
-  // it is not necessary to call before processing the first stream after
-  // creation.
-  virtual int Initialize() = 0;
-
-  // Sets the sample |rate| in Hz for both the primary and reverse audio
-  // streams. 8000, 16000 or 32000 Hz are permitted.
-  virtual int set_sample_rate_hz(int rate) = 0;
-  virtual int sample_rate_hz() const = 0;
-
-  // Sets the number of channels for the primary audio stream. Input frames must
-  // contain a number of channels given by |input_channels|, while output frames
-  // will be returned with number of channels given by |output_channels|.
-  virtual int set_num_channels(int input_channels, int output_channels) = 0;
-  virtual int num_input_channels() const = 0;
-  virtual int num_output_channels() const = 0;
-
-  // Sets the number of channels for the reverse audio stream. Input frames must
-  // contain a number of channels given by |channels|.
-  virtual int set_num_reverse_channels(int channels) = 0;
-  virtual int num_reverse_channels() const = 0;
-
-  // Processes a 10 ms |frame| of the primary audio stream. On the client-side,
-  // this is the near-end (or captured) audio.
-  //
-  // If needed for enabled functionality, any function with the set_stream_ tag
-  // must be called prior to processing the current frame. Any getter function
-  // with the stream_ tag which is needed should be called after processing.
-  //
-  // The |_frequencyInHz|, |_audioChannel|, and |_payloadDataLengthInSamples|
-  // members of |frame| must be valid, and correspond to settings supplied
-  // to APM.
-  virtual int ProcessStream(AudioFrame* frame) = 0;
-
-  // Analyzes a 10 ms |frame| of the reverse direction audio stream. The frame
-  // will not be modified. On the client-side, this is the far-end (or to be
-  // rendered) audio.
-  //
-  // It is only necessary to provide this if echo processing is enabled, as the
-  // reverse stream forms the echo reference signal. It is recommended, but not
-  // necessary, to provide if gain control is enabled. On the server-side this
-  // typically will not be used. If you're not sure what to pass in here,
-  // chances are you don't need to use it.
-  //
-  // The |_frequencyInHz|, |_audioChannel|, and |_payloadDataLengthInSamples|
-  // members of |frame| must be valid.
-  //
-  // TODO(ajm): add const to input; requires an implementation fix.
-  virtual int AnalyzeReverseStream(AudioFrame* frame) = 0;
-
-  // This must be called if and only if echo processing is enabled.
-  //
-  // Sets the |delay| in ms between AnalyzeReverseStream() receiving a far-end
-  // frame and ProcessStream() receiving a near-end frame containing the
-  // corresponding echo. On the client-side this can be expressed as
-  //   delay = (t_render - t_analyze) + (t_process - t_capture)
-  // where,
-  //   - t_analyze is the time a frame is passed to AnalyzeReverseStream() and
-  //     t_render is the time the first sample of the same frame is rendered by
-  //     the audio hardware.
-  //   - t_capture is the time the first sample of a frame is captured by the
-  //     audio hardware and t_pull is the time the same frame is passed to
-  //     ProcessStream().
-  virtual int set_stream_delay_ms(int delay) = 0;
-  virtual int stream_delay_ms() const = 0;
-
-  // Starts recording debugging information to a file specified by |filename|,
-  // a NULL-terminated string. If there is an ongoing recording, the old file
-  // will be closed, and recording will continue in the newly specified file.
-  // An already existing file will be overwritten without warning.
-  static const int kMaxFilenameSize = 1024;
-  virtual int StartDebugRecording(const char filename[kMaxFilenameSize]) = 0;
-
-  // Stops recording debugging information, and closes the file. Recording
-  // cannot be resumed in the same file (without overwriting it).
-  virtual int StopDebugRecording() = 0;
-
-  // These provide access to the component interfaces and should never return
-  // NULL. The pointers will be valid for the lifetime of the APM instance.
-  // The memory for these objects is entirely managed internally.
-  virtual EchoCancellation* echo_cancellation() const = 0;
-  virtual EchoControlMobile* echo_control_mobile() const = 0;
-  virtual GainControl* gain_control() const = 0;
-  virtual HighPassFilter* high_pass_filter() const = 0;
-  virtual LevelEstimator* level_estimator() const = 0;
-  virtual NoiseSuppression* noise_suppression() const = 0;
-  virtual VoiceDetection* voice_detection() const = 0;
-
-  struct Statistic {
-    int instant;  // Instantaneous value.
-    int average;  // Long-term average.
-    int maximum;  // Long-term maximum.
-    int minimum;  // Long-term minimum.
-  };
-
-  // Fatal errors.
-  enum Errors {
-    kNoError = 0,
-    kUnspecifiedError = -1,
-    kCreationFailedError = -2,
-    kUnsupportedComponentError = -3,
-    kUnsupportedFunctionError = -4,
-    kNullPointerError = -5,
-    kBadParameterError = -6,
-    kBadSampleRateError = -7,
-    kBadDataLengthError = -8,
-    kBadNumberChannelsError = -9,
-    kFileError = -10,
-    kStreamParameterNotSetError = -11,
-    kNotEnabledError = -12
-  };
-
-  // Warnings are non-fatal.
-  enum Warnings {
-    // This results when a set_stream_ parameter is out of range. Processing
-    // will continue, but the parameter may have been truncated.
-    kBadStreamParameterWarning = -13,
-  };
-
-  // Inherited from Module.
-  virtual WebRtc_Word32 TimeUntilNextProcess() { return -1; };
-  virtual WebRtc_Word32 Process() { return -1; };
-
- protected:
-  virtual ~AudioProcessing() {};
-};
-
-// The acoustic echo cancellation (AEC) component provides better performance
-// than AECM but also requires more processing power and is dependent on delay
-// stability and reporting accuracy. As such it is well-suited and recommended
-// for PC and IP phone applications.
-//
-// Not recommended to be enabled on the server-side.
-class EchoCancellation {
- public:
-  // EchoCancellation and EchoControlMobile may not be enabled simultaneously.
-  // Enabling one will disable the other.
-  virtual int Enable(bool enable) = 0;
-  virtual bool is_enabled() const = 0;
-
-  // Differences in clock speed on the primary and reverse streams can impact
-  // the AEC performance. On the client-side, this could be seen when different
-  // render and capture devices are used, particularly with webcams.
-  //
-  // This enables a compensation mechanism, and requires that
-  // |set_device_sample_rate_hz()| and |set_stream_drift_samples()| be called.
-  virtual int enable_drift_compensation(bool enable) = 0;
-  virtual bool is_drift_compensation_enabled() const = 0;
-
-  // Provides the sampling rate of the audio devices. It is assumed the render
-  // and capture devices use the same nominal sample rate. Required if and only
-  // if drift compensation is enabled.
-  virtual int set_device_sample_rate_hz(int rate) = 0;
-  virtual int device_sample_rate_hz() const = 0;
-
-  // Sets the difference between the number of samples rendered and captured by
-  // the audio devices since the last call to |ProcessStream()|. Must be called
-  // if and only if drift compensation is enabled, prior to |ProcessStream()|.
-  virtual int set_stream_drift_samples(int drift) = 0;
-  virtual int stream_drift_samples() const = 0;
-
-  enum SuppressionLevel {
-    kLowSuppression,
-    kModerateSuppression,
-    kHighSuppression
-  };
-
-  // Sets the aggressiveness of the suppressor. A higher level trades off
-  // double-talk performance for increased echo suppression.
-  virtual int set_suppression_level(SuppressionLevel level) = 0;
-  virtual SuppressionLevel suppression_level() const = 0;
-
-  // Returns false if the current frame almost certainly contains no echo
-  // and true if it _might_ contain echo.
-  virtual bool stream_has_echo() const = 0;
-
-  // Enables the computation of various echo metrics. These are obtained
-  // through |GetMetrics()|.
-  virtual int enable_metrics(bool enable) = 0;
-  virtual bool are_metrics_enabled() const = 0;
-
-  // Each statistic is reported in dB.
-  // P_far:  Far-end (render) signal power.
-  // P_echo: Near-end (capture) echo signal power.
-  // P_out:  Signal power at the output of the AEC.
-  // P_a:    Internal signal power at the point before the AEC's non-linear
-  //         processor.
-  struct Metrics {
-    // RERL = ERL + ERLE
-    AudioProcessing::Statistic residual_echo_return_loss;
-
-    // ERL = 10log_10(P_far / P_echo)
-    AudioProcessing::Statistic echo_return_loss;
-
-    // ERLE = 10log_10(P_echo / P_out)
-    AudioProcessing::Statistic echo_return_loss_enhancement;
-
-    // (Pre non-linear processing suppression) A_NLP = 10log_10(P_echo / P_a)
-    AudioProcessing::Statistic a_nlp;
-  };
-
-  // TODO(ajm): discuss the metrics update period.
-  virtual int GetMetrics(Metrics* metrics) = 0;
-
- protected:
-  virtual ~EchoCancellation() {};
-};
-
-// The acoustic echo control for mobile (AECM) component is a low complexity
-// robust option intended for use on mobile devices.
-//
-// Not recommended to be enabled on the server-side.
-class EchoControlMobile {
- public:
-  // EchoCancellation and EchoControlMobile may not be enabled simultaneously.
-  // Enabling one will disable the other.
-  virtual int Enable(bool enable) = 0;
-  virtual bool is_enabled() const = 0;
-
-  // Recommended settings for particular audio routes. In general, the louder
-  // the echo is expected to be, the higher this value should be set. The
-  // preferred setting may vary from device to device.
-  enum RoutingMode {
-    kQuietEarpieceOrHeadset,
-    kEarpiece,
-    kLoudEarpiece,
-    kSpeakerphone,
-    kLoudSpeakerphone
-  };
-
-  // Sets echo control appropriate for the audio routing |mode| on the device.
-  // It can and should be updated during a call if the audio routing changes.
-  virtual int set_routing_mode(RoutingMode mode) = 0;
-  virtual RoutingMode routing_mode() const = 0;
-
-  // Comfort noise replaces suppressed background noise to maintain a
-  // consistent signal level.
-  virtual int enable_comfort_noise(bool enable) = 0;
-  virtual bool is_comfort_noise_enabled() const = 0;
-
- protected:
-  virtual ~EchoControlMobile() {};
-};
-
-// The automatic gain control (AGC) component brings the signal to an
-// appropriate range. This is done by applying a digital gain directly and, in
-// the analog mode, prescribing an analog gain to be applied at the audio HAL.
-//
-// Recommended to be enabled on the client-side.
-class GainControl {
- public:
-  virtual int Enable(bool enable) = 0;
-  virtual bool is_enabled() const = 0;
-
-  // When an analog mode is set, this must be called prior to |ProcessStream()|
-  // to pass the current analog level from the audio HAL. Must be within the
-  // range provided to |set_analog_level_limits()|.
-  virtual int set_stream_analog_level(int level) = 0;
-
-  // When an analog mode is set, this should be called after |ProcessStream()|
-  // to obtain the recommended new analog level for the audio HAL. It is the
-  // users responsibility to apply this level.
-  virtual int stream_analog_level() = 0;
-
-  enum Mode {
-    // Adaptive mode intended for use if an analog volume control is available
-    // on the capture device. It will require the user to provide coupling
-    // between the OS mixer controls and AGC through the |stream_analog_level()|
-    // functions.
-    //
-    // It consists of an analog gain prescription for the audio device and a
-    // digital compression stage.
-    kAdaptiveAnalog,
-
-    // Adaptive mode intended for situations in which an analog volume control
-    // is unavailable. It operates in a similar fashion to the adaptive analog
-    // mode, but with scaling instead applied in the digital domain. As with
-    // the analog mode, it additionally uses a digital compression stage.
-    kAdaptiveDigital,
-
-    // Fixed mode which enables only the digital compression stage also used by
-    // the two adaptive modes.
-    //
-    // It is distinguished from the adaptive modes by considering only a
-    // short time-window of the input signal. It applies a fixed gain through
-    // most of the input level range, and compresses (gradually reduces gain
-    // with increasing level) the input signal at higher levels. This mode is
-    // preferred on embedded devices where the capture signal level is
-    // predictable, so that a known gain can be applied.
-    kFixedDigital
-  };
-
-  virtual int set_mode(Mode mode) = 0;
-  virtual Mode mode() const = 0;
-
-  // Sets the target peak |level| (or envelope) of the AGC in dBFs (decibels
-  // from digital full-scale). The convention is to use positive values. For
-  // instance, passing in a value of 3 corresponds to -3 dBFs, or a target
-  // level 3 dB below full-scale. Limited to [0, 31].
-  //
-  // TODO(ajm): use a negative value here instead, if/when VoE will similarly
-  //            update its interface.
-  virtual int set_target_level_dbfs(int level) = 0;
-  virtual int target_level_dbfs() const = 0;
-
-  // Sets the maximum |gain| the digital compression stage may apply, in dB. A
-  // higher number corresponds to greater compression, while a value of 0 will
-  // leave the signal uncompressed. Limited to [0, 90].
-  virtual int set_compression_gain_db(int gain) = 0;
-  virtual int compression_gain_db() const = 0;
-
-  // When enabled, the compression stage will hard limit the signal to the
-  // target level. Otherwise, the signal will be compressed but not limited
-  // above the target level.
-  virtual int enable_limiter(bool enable) = 0;
-  virtual bool is_limiter_enabled() const = 0;
-
-  // Sets the |minimum| and |maximum| analog levels of the audio capture device.
-  // Must be set if and only if an analog mode is used. Limited to [0, 65535].
-  virtual int set_analog_level_limits(int minimum,
-                                      int maximum) = 0;
-  virtual int analog_level_minimum() const = 0;
-  virtual int analog_level_maximum() const = 0;
-
-  // Returns true if the AGC has detected a saturation event (period where the
-  // signal reaches digital full-scale) in the current frame and the analog
-  // level cannot be reduced.
-  //
-  // This could be used as an indicator to reduce or disable analog mic gain at
-  // the audio HAL.
-  virtual bool stream_is_saturated() const = 0;
-
- protected:
-  virtual ~GainControl() {};
-};
-
-// A filtering component which removes DC offset and low-frequency noise.
-// Recommended to be enabled on the client-side.
-class HighPassFilter {
- public:
-  virtual int Enable(bool enable) = 0;
-  virtual bool is_enabled() const = 0;
-
- protected:
-  virtual ~HighPassFilter() {};
-};
-
-// An estimation component used to retrieve level metrics.
-class LevelEstimator {
- public:
-  virtual int Enable(bool enable) = 0;
-  virtual bool is_enabled() const = 0;
-
-  // The metrics are reported in dBFs calculated as:
-  //   Level = 10log_10(P_s / P_max) [dBFs], where
-  //   P_s is the signal power and P_max is the maximum possible (or peak)
-  //   power. With 16-bit signals, P_max = (2^15)^2.
-  struct Metrics {
-    AudioProcessing::Statistic signal;  // Overall signal level.
-    AudioProcessing::Statistic speech;  // Speech level.
-    AudioProcessing::Statistic noise;   // Noise level.
-  };
-
-  virtual int GetMetrics(Metrics* metrics, Metrics* reverse_metrics) = 0;
-
-  //virtual int enable_noise_warning(bool enable) = 0;
-  //bool is_noise_warning_enabled() const = 0;
-  //virtual bool stream_has_high_noise() const = 0;
-
- protected:
-  virtual ~LevelEstimator() {};
-};
-
-// The noise suppression (NS) component attempts to remove noise while
-// retaining speech. Recommended to be enabled on the client-side.
-//
-// Recommended to be enabled on the client-side.
-class NoiseSuppression {
- public:
-  virtual int Enable(bool enable) = 0;
-  virtual bool is_enabled() const = 0;
-
-  // Determines the aggressiveness of the suppression. Increasing the level
-  // will reduce the noise level at the expense of a higher speech distortion.
-  enum Level {
-    kLow,
-    kModerate,
-    kHigh,
-    kVeryHigh
-  };
-
-  virtual int set_level(Level level) = 0;
-  virtual Level level() const = 0;
-
- protected:
-  virtual ~NoiseSuppression() {};
-};
-
-// The voice activity detection (VAD) component analyzes the stream to
-// determine if voice is present. A facility is also provided to pass in an
-// external VAD decision.
-class VoiceDetection {
- public:
-  virtual int Enable(bool enable) = 0;
-  virtual bool is_enabled() const = 0;
-
-  // Returns true if voice is detected in the current frame. Should be called
-  // after |ProcessStream()|.
-  virtual bool stream_has_voice() const = 0;
-
-  // Some of the APM functionality requires a VAD decision. In the case that
-  // a decision is externally available for the current frame, it can be passed
-  // in here, before |ProcessStream()| is called.
-  //
-  // VoiceDetection does _not_ need to be enabled to use this. If it happens to
-  // be enabled, detection will be skipped for any frame in which an external
-  // VAD decision is provided.
-  virtual int set_stream_has_voice(bool has_voice) = 0;
-
-  // Specifies the likelihood that a frame will be declared to contain voice.
-  // A higher value makes it more likely that speech will not be clipped, at
-  // the expense of more noise being detected as voice.
-  enum Likelihood {
-    kVeryLowLikelihood,
-    kLowLikelihood,
-    kModerateLikelihood,
-    kHighLikelihood
-  };
-
-  virtual int set_likelihood(Likelihood likelihood) = 0;
-  virtual Likelihood likelihood() const = 0;
-
-  // Sets the |size| of the frames in ms on which the VAD will operate. Larger
-  // frames will improve detection accuracy, but reduce the frequency of
-  // updates.
-  //
-  // This does not impact the size of frames passed to |ProcessStream()|.
-  virtual int set_frame_size_ms(int size) = 0;
-  virtual int frame_size_ms() const = 0;
-
- protected:
-  virtual ~VoiceDetection() {};
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_
diff --git a/src/modules/audio_processing/main/source/Android.mk b/src/modules/audio_processing/main/source/Android.mk
deleted file mode 100644
index 12c7bc7..0000000
--- a/src/modules/audio_processing/main/source/Android.mk
+++ /dev/null
@@ -1,72 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc_apm
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := audio_buffer.cc \
-    audio_processing_impl.cc \
-    echo_cancellation_impl.cc \
-    echo_control_mobile_impl.cc \
-    gain_control_impl.cc \
-    high_pass_filter_impl.cc \
-    level_estimator_impl.cc \
-    noise_suppression_impl.cc \
-    splitting_filter.cc \
-    processing_component.cc \
-    voice_detection_impl.cc
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=  
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
-    '-DNO_HEAPCHECKER' \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_THREAD_RR' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' \
-    '-DWEBRTC_NS_FIXED'
-#   floating point
-#   -DWEBRTC_NS_FLOAT'
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../.. \
-    $(LOCAL_PATH)/../interface \
-    $(LOCAL_PATH)/../../../interface \
-    $(LOCAL_PATH)/../../../../system_wrappers/interface \
-    $(LOCAL_PATH)/../../aec/main/interface \
-    $(LOCAL_PATH)/../../aecm/main/interface \
-    $(LOCAL_PATH)/../../agc/main/interface \
-    $(LOCAL_PATH)/../../ns/main/interface \
-    $(LOCAL_PATH)/../../../../common_audio/signal_processing_library/main/interface \
-    $(LOCAL_PATH)/../../../../common_audio/vad/main/interface 
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS := 
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES := 
-
-LOCAL_SHARED_LIBRARIES := libcutils \
-    libdl \
-    libstlport 
-
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
-
diff --git a/src/modules/audio_processing/main/source/apm.gyp b/src/modules/audio_processing/main/source/apm.gyp
deleted file mode 100644
index 93811c7..0000000
--- a/src/modules/audio_processing/main/source/apm.gyp
+++ /dev/null
@@ -1,77 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../../../common_settings.gypi', # Common settings
-  ],
-  'targets': [
-    {
-      'target_name': 'audio_processing',
-      'type': '<(library)',
-      'conditions': [
-        ['prefer_fixed_point==1', {
-          'dependencies': ['../../ns/main/source/ns.gyp:ns_fix'],
-          'defines': ['WEBRTC_NS_FIXED'],
-        }, { # else: prefer_fixed_point==0
-          'dependencies': ['../../ns/main/source/ns.gyp:ns'],
-          'defines': ['WEBRTC_NS_FLOAT'],
-        }],
-      ],
-      'dependencies': [
-        '../../../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
-        '../../aec/main/source/aec.gyp:aec',
-        '../../aecm/main/source/aecm.gyp:aecm',
-        '../../agc/main/source/agc.gyp:agc',
-        '../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
-        '../../../../common_audio/vad/main/source/vad.gyp:vad',
-      ],
-      'include_dirs': [
-        '../interface',
-        '../../../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-          '../../../interface',
-        ],
-      },
-      'sources': [
-        '../interface/audio_processing.h',
-        'audio_buffer.cc',
-        'audio_buffer.h',
-        'audio_processing_impl.cc',
-        'audio_processing_impl.h',
-        'echo_cancellation_impl.cc',
-        'echo_cancellation_impl.h',
-        'echo_control_mobile_impl.cc',
-        'echo_control_mobile_impl.h',
-        'gain_control_impl.cc',
-        'gain_control_impl.h',
-        'high_pass_filter_impl.cc',
-        'high_pass_filter_impl.h',
-        'level_estimator_impl.cc',
-        'level_estimator_impl.h',
-        'noise_suppression_impl.cc',
-        'noise_suppression_impl.h',
-        'splitting_filter.cc',
-        'splitting_filter.h',
-        'processing_component.cc',
-        'processing_component.h',
-        'voice_detection_impl.cc',
-        'voice_detection_impl.h',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/main/source/audio_buffer.cc b/src/modules/audio_processing/main/source/audio_buffer.cc
deleted file mode 100644
index 6b20fce..0000000
--- a/src/modules/audio_processing/main/source/audio_buffer.cc
+++ /dev/null
@@ -1,278 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_buffer.h"
-
-#include "module_common_types.h"
-
-namespace webrtc {
-namespace {
-
-enum {
-  kSamplesPer8kHzChannel = 80,
-  kSamplesPer16kHzChannel = 160,
-  kSamplesPer32kHzChannel = 320
-};
-
-void StereoToMono(const WebRtc_Word16* left, const WebRtc_Word16* right,
-                  WebRtc_Word16* out, int samples_per_channel) {
-  WebRtc_Word32 data_int32 = 0;
-  for (int i = 0; i < samples_per_channel; i++) {
-    data_int32 = (left[i] + right[i]) >> 1;
-    if (data_int32 > 32767) {
-      data_int32 = 32767;
-    } else if (data_int32 < -32768) {
-      data_int32 = -32768;
-    }
-
-    out[i] = static_cast<WebRtc_Word16>(data_int32);
-  }
-}
-}  // namespace
-
-struct AudioChannel {
-  AudioChannel() {
-    memset(data, 0, sizeof(data));
-  }
-
-  WebRtc_Word16 data[kSamplesPer32kHzChannel];
-};
-
-struct SplitAudioChannel {
-  SplitAudioChannel() {
-    memset(low_pass_data, 0, sizeof(low_pass_data));
-    memset(high_pass_data, 0, sizeof(high_pass_data));
-    memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1));
-    memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2));
-    memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1));
-    memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2));
-  }
-
-  WebRtc_Word16 low_pass_data[kSamplesPer16kHzChannel];
-  WebRtc_Word16 high_pass_data[kSamplesPer16kHzChannel];
-
-  WebRtc_Word32 analysis_filter_state1[6];
-  WebRtc_Word32 analysis_filter_state2[6];
-  WebRtc_Word32 synthesis_filter_state1[6];
-  WebRtc_Word32 synthesis_filter_state2[6];
-};
-
-// TODO(am): check range of input parameters?
-AudioBuffer::AudioBuffer(WebRtc_Word32 max_num_channels,
-                         WebRtc_Word32 samples_per_channel)
-    : max_num_channels_(max_num_channels),
-      num_channels_(0),
-      num_mixed_channels_(0),
-      num_mixed_low_pass_channels_(0),
-      samples_per_channel_(samples_per_channel),
-      samples_per_split_channel_(samples_per_channel),
-      reference_copied_(false),
-      data_(NULL),
-      channels_(NULL),
-      split_channels_(NULL),
-      mixed_low_pass_channels_(NULL),
-      low_pass_reference_channels_(NULL) {
-  if (max_num_channels_ > 1) {
-    channels_ = new AudioChannel[max_num_channels_];
-    mixed_low_pass_channels_ = new AudioChannel[max_num_channels_];
-  }
-  low_pass_reference_channels_ = new AudioChannel[max_num_channels_];
-
-  if (samples_per_channel_ == kSamplesPer32kHzChannel) {
-    split_channels_ = new SplitAudioChannel[max_num_channels_];
-    samples_per_split_channel_ = kSamplesPer16kHzChannel;
-  }
-}
-
-AudioBuffer::~AudioBuffer() {
-  if (channels_ != NULL) {
-    delete [] channels_;
-  }
-
-  if (mixed_low_pass_channels_ != NULL) {
-    delete [] mixed_low_pass_channels_;
-  }
-
-  if (low_pass_reference_channels_ != NULL) {
-    delete [] low_pass_reference_channels_;
-  }
-
-  if (split_channels_ != NULL) {
-    delete [] split_channels_;
-  }
-}
-
-WebRtc_Word16* AudioBuffer::data(WebRtc_Word32 channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  if (data_ != NULL) {
-    return data_;
-  }
-
-  return channels_[channel].data;
-}
-
-WebRtc_Word16* AudioBuffer::low_pass_split_data(WebRtc_Word32 channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  if (split_channels_ == NULL) {
-    return data(channel);
-  }
-
-  return split_channels_[channel].low_pass_data;
-}
-
-WebRtc_Word16* AudioBuffer::high_pass_split_data(WebRtc_Word32 channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  if (split_channels_ == NULL) {
-    return NULL;
-  }
-
-  return split_channels_[channel].high_pass_data;
-}
-
-WebRtc_Word16* AudioBuffer::mixed_low_pass_data(WebRtc_Word32 channel) const {
-  assert(channel >= 0 && channel < num_mixed_low_pass_channels_);
-
-  return mixed_low_pass_channels_[channel].data;
-}
-
-WebRtc_Word16* AudioBuffer::low_pass_reference(WebRtc_Word32 channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  if (!reference_copied_) {
-    return NULL;
-  }
-
-  return low_pass_reference_channels_[channel].data;
-}
-
-WebRtc_Word32* AudioBuffer::analysis_filter_state1(WebRtc_Word32 channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  return split_channels_[channel].analysis_filter_state1;
-}
-
-WebRtc_Word32* AudioBuffer::analysis_filter_state2(WebRtc_Word32 channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  return split_channels_[channel].analysis_filter_state2;
-}
-
-WebRtc_Word32* AudioBuffer::synthesis_filter_state1(WebRtc_Word32 channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  return split_channels_[channel].synthesis_filter_state1;
-}
-
-WebRtc_Word32* AudioBuffer::synthesis_filter_state2(WebRtc_Word32 channel) const {
-  assert(channel >= 0 && channel < num_channels_);
-  return split_channels_[channel].synthesis_filter_state2;
-}
-
-WebRtc_Word32 AudioBuffer::num_channels() const {
-  return num_channels_;
-}
-
-WebRtc_Word32 AudioBuffer::samples_per_channel() const {
-  return samples_per_channel_;
-}
-
-WebRtc_Word32 AudioBuffer::samples_per_split_channel() const {
-  return samples_per_split_channel_;
-}
-
-// TODO(ajm): Do deinterleaving and mixing in one step?
-void AudioBuffer::DeinterleaveFrom(AudioFrame* audioFrame) {
-  assert(audioFrame->_audioChannel <= max_num_channels_);
-  assert(audioFrame->_payloadDataLengthInSamples ==  samples_per_channel_);
-
-  num_channels_ = audioFrame->_audioChannel;
-  num_mixed_channels_ = 0;
-  num_mixed_low_pass_channels_ = 0;
-  reference_copied_ = false;
-
-  if (num_channels_ == 1) {
-    // We can get away with a pointer assignment in this case.
-    data_ = audioFrame->_payloadData;
-    return;
-  }
-
-  for (int i = 0; i < num_channels_; i++) {
-    WebRtc_Word16* deinterleaved = channels_[i].data;
-    WebRtc_Word16* interleaved = audioFrame->_payloadData;
-    WebRtc_Word32 interleaved_idx = i;
-    for (int j = 0; j < samples_per_channel_; j++) {
-      deinterleaved[j] = interleaved[interleaved_idx];
-      interleaved_idx += num_channels_;
-    }
-  }
-}
-
-void AudioBuffer::InterleaveTo(AudioFrame* audioFrame) const {
-  assert(audioFrame->_audioChannel == num_channels_);
-  assert(audioFrame->_payloadDataLengthInSamples == samples_per_channel_);
-
-  if (num_channels_ == 1) {
-    if (num_mixed_channels_ == 1) {
-      memcpy(audioFrame->_payloadData,
-             channels_[0].data,
-             sizeof(WebRtc_Word16) * samples_per_channel_);
-    } else {
-      // These should point to the same buffer in this case.
-      assert(data_ == audioFrame->_payloadData);
-    }
-
-    return;
-  }
-
-  for (int i = 0; i < num_channels_; i++) {
-    WebRtc_Word16* deinterleaved = channels_[i].data;
-    WebRtc_Word16* interleaved = audioFrame->_payloadData;
-    WebRtc_Word32 interleaved_idx = i;
-    for (int j = 0; j < samples_per_channel_; j++) {
-      interleaved[interleaved_idx] = deinterleaved[j];
-      interleaved_idx += num_channels_;
-    }
-  }
-}
-
-// TODO(ajm): would be good to support the no-mix case with pointer assignment.
-// TODO(ajm): handle mixing to multiple channels?
-void AudioBuffer::Mix(WebRtc_Word32 num_mixed_channels) {
-  // We currently only support the stereo to mono case.
-  assert(num_channels_ == 2);
-  assert(num_mixed_channels == 1);
-
-  StereoToMono(channels_[0].data,
-               channels_[1].data,
-               channels_[0].data,
-               samples_per_channel_);
-
-  num_channels_ = num_mixed_channels;
-  num_mixed_channels_ = num_mixed_channels;
-}
-
-void AudioBuffer::CopyAndMixLowPass(WebRtc_Word32 num_mixed_channels) {
-  // We currently only support the stereo to mono case.
-  assert(num_channels_ == 2);
-  assert(num_mixed_channels == 1);
-
-  StereoToMono(low_pass_split_data(0),
-               low_pass_split_data(1),
-               mixed_low_pass_channels_[0].data,
-               samples_per_split_channel_);
-
-  num_mixed_low_pass_channels_ = num_mixed_channels;
-}
-
-void AudioBuffer::CopyLowPassToReference() {
-  reference_copied_ = true;
-  for (int i = 0; i < num_channels_; i++) {
-    memcpy(low_pass_reference_channels_[i].data,
-           low_pass_split_data(i),
-           sizeof(WebRtc_Word16) * samples_per_split_channel_);
-  }
-}
-}  // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/audio_buffer.h b/src/modules/audio_processing/main/source/audio_buffer.h
deleted file mode 100644
index 15f850b..0000000
--- a/src/modules/audio_processing/main/source/audio_buffer.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
-
-#include "typedefs.h"
-
-
-namespace webrtc {
-
-struct AudioChannel;
-struct SplitAudioChannel;
-class AudioFrame;
-
-class AudioBuffer {
- public:
-  AudioBuffer(WebRtc_Word32 max_num_channels, WebRtc_Word32 samples_per_channel);
-  virtual ~AudioBuffer();
-
-  WebRtc_Word32 num_channels() const;
-  WebRtc_Word32 samples_per_channel() const;
-  WebRtc_Word32 samples_per_split_channel() const;
-
-  WebRtc_Word16* data(WebRtc_Word32 channel) const;
-  WebRtc_Word16* low_pass_split_data(WebRtc_Word32 channel) const;
-  WebRtc_Word16* high_pass_split_data(WebRtc_Word32 channel) const;
-  WebRtc_Word16* mixed_low_pass_data(WebRtc_Word32 channel) const;
-  WebRtc_Word16* low_pass_reference(WebRtc_Word32 channel) const;
-
-  WebRtc_Word32* analysis_filter_state1(WebRtc_Word32 channel) const;
-  WebRtc_Word32* analysis_filter_state2(WebRtc_Word32 channel) const;
-  WebRtc_Word32* synthesis_filter_state1(WebRtc_Word32 channel) const;
-  WebRtc_Word32* synthesis_filter_state2(WebRtc_Word32 channel) const;
-
-  void DeinterleaveFrom(AudioFrame* audioFrame);
-  void InterleaveTo(AudioFrame* audioFrame) const;
-  void Mix(WebRtc_Word32 num_mixed_channels);
-  void CopyAndMixLowPass(WebRtc_Word32 num_mixed_channels);
-  void CopyLowPassToReference();
-
- private:
-  const WebRtc_Word32 max_num_channels_;
-  WebRtc_Word32 num_channels_;
-  WebRtc_Word32 num_mixed_channels_;
-  WebRtc_Word32 num_mixed_low_pass_channels_;
-  const WebRtc_Word32 samples_per_channel_;
-  WebRtc_Word32 samples_per_split_channel_;
-  bool reference_copied_;
-
-  WebRtc_Word16* data_;
-  // TODO(ajm): Prefer to make these vectors if permitted...
-  AudioChannel* channels_;
-  SplitAudioChannel* split_channels_;
-  // TODO(ajm): improve this, we don't need the full 32 kHz space here.
-  AudioChannel* mixed_low_pass_channels_;
-  AudioChannel* low_pass_reference_channels_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
diff --git a/src/modules/audio_processing/main/source/audio_processing_impl.cc b/src/modules/audio_processing/main/source/audio_processing_impl.cc
deleted file mode 100644
index 6440e36..0000000
--- a/src/modules/audio_processing/main/source/audio_processing_impl.cc
+++ /dev/null
@@ -1,636 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_processing_impl.h"
-
-#include <cassert>
-
-#include "module_common_types.h"
-
-#include "critical_section_wrapper.h"
-#include "file_wrapper.h"
-
-#include "audio_buffer.h"
-#include "echo_cancellation_impl.h"
-#include "echo_control_mobile_impl.h"
-#include "high_pass_filter_impl.h"
-#include "gain_control_impl.h"
-#include "level_estimator_impl.h"
-#include "noise_suppression_impl.h"
-#include "processing_component.h"
-#include "splitting_filter.h"
-#include "voice_detection_impl.h"
-
-namespace webrtc {
-namespace {
-
-enum Events {
-  kInitializeEvent,
-  kRenderEvent,
-  kCaptureEvent
-};
-
-const char kMagicNumber[] = "#!vqetrace1.2";
-}  // namespace
-
-AudioProcessing* AudioProcessing::Create(int id) {
-  /*WEBRTC_TRACE(webrtc::kTraceModuleCall,
-             webrtc::kTraceAudioProcessing,
-             id,
-             "AudioProcessing::Create()");*/
-
-  AudioProcessingImpl* apm = new AudioProcessingImpl(id);
-  if (apm->Initialize() != kNoError) {
-    delete apm;
-    apm = NULL;
-  }
-
-  return apm;
-}
-
-void AudioProcessing::Destroy(AudioProcessing* apm) {
-  delete static_cast<AudioProcessingImpl*>(apm);
-}
-
-AudioProcessingImpl::AudioProcessingImpl(int id)
-    : id_(id),
-      echo_cancellation_(NULL),
-      echo_control_mobile_(NULL),
-      gain_control_(NULL),
-      high_pass_filter_(NULL),
-      level_estimator_(NULL),
-      noise_suppression_(NULL),
-      voice_detection_(NULL),
-      debug_file_(FileWrapper::Create()),
-      crit_(CriticalSectionWrapper::CreateCriticalSection()),
-      render_audio_(NULL),
-      capture_audio_(NULL),
-      sample_rate_hz_(kSampleRate16kHz),
-      split_sample_rate_hz_(kSampleRate16kHz),
-      samples_per_channel_(sample_rate_hz_ / 100),
-      stream_delay_ms_(0),
-      was_stream_delay_set_(false),
-      num_render_input_channels_(1),
-      num_capture_input_channels_(1),
-      num_capture_output_channels_(1) {
-
-  echo_cancellation_ = new EchoCancellationImpl(this);
-  component_list_.push_back(echo_cancellation_);
-
-  echo_control_mobile_ = new EchoControlMobileImpl(this);
-  component_list_.push_back(echo_control_mobile_);
-
-  gain_control_ = new GainControlImpl(this);
-  component_list_.push_back(gain_control_);
-
-  high_pass_filter_ = new HighPassFilterImpl(this);
-  component_list_.push_back(high_pass_filter_);
-
-  level_estimator_ = new LevelEstimatorImpl(this);
-  component_list_.push_back(level_estimator_);
-
-  noise_suppression_ = new NoiseSuppressionImpl(this);
-  component_list_.push_back(noise_suppression_);
-
-  voice_detection_ = new VoiceDetectionImpl(this);
-  component_list_.push_back(voice_detection_);
-}
-
-AudioProcessingImpl::~AudioProcessingImpl() {
-  while (!component_list_.empty()) {
-    ProcessingComponent* component = component_list_.front();
-    component->Destroy();
-    delete component;
-    component_list_.pop_front();
-  }
-
-  if (debug_file_->Open()) {
-    debug_file_->CloseFile();
-  }
-  delete debug_file_;
-  debug_file_ = NULL;
-
-  delete crit_;
-  crit_ = NULL;
-
-  if (render_audio_ != NULL) {
-    delete render_audio_;
-    render_audio_ = NULL;
-  }
-
-  if (capture_audio_ != NULL) {
-    delete capture_audio_;
-    capture_audio_ = NULL;
-  }
-}
-
-CriticalSectionWrapper* AudioProcessingImpl::crit() const {
-  return crit_;
-}
-
-int AudioProcessingImpl::split_sample_rate_hz() const {
-  return split_sample_rate_hz_;
-}
-
-int AudioProcessingImpl::Initialize() {
-  CriticalSectionScoped crit_scoped(*crit_);
-  return InitializeLocked();
-}
-
-int AudioProcessingImpl::InitializeLocked() {
-  if (render_audio_ != NULL) {
-    delete render_audio_;
-    render_audio_ = NULL;
-  }
-
-  if (capture_audio_ != NULL) {
-    delete capture_audio_;
-    capture_audio_ = NULL;
-  }
-
-  render_audio_ = new AudioBuffer(num_render_input_channels_,
-                                  samples_per_channel_);
-  capture_audio_ = new AudioBuffer(num_capture_input_channels_,
-                                   samples_per_channel_);
-
-  was_stream_delay_set_ = false;
-
-  // Initialize all components.
-  std::list<ProcessingComponent*>::iterator it;
-  for (it = component_list_.begin(); it != component_list_.end(); it++) {
-    int err = (*it)->Initialize();
-    if (err != kNoError) {
-      return err;
-    }
-  }
-
-  return kNoError;
-}
-
-int AudioProcessingImpl::set_sample_rate_hz(int rate) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  if (rate != kSampleRate8kHz &&
-      rate != kSampleRate16kHz &&
-      rate != kSampleRate32kHz) {
-    return kBadParameterError;
-  }
-
-  sample_rate_hz_ = rate;
-  samples_per_channel_ = rate / 100;
-
-  if (sample_rate_hz_ == kSampleRate32kHz) {
-    split_sample_rate_hz_ = kSampleRate16kHz;
-  } else {
-    split_sample_rate_hz_ = sample_rate_hz_;
-  }
-
-  return InitializeLocked();
-}
-
-int AudioProcessingImpl::sample_rate_hz() const {
-  return sample_rate_hz_;
-}
-
-int AudioProcessingImpl::set_num_reverse_channels(int channels) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  // Only stereo supported currently.
-  if (channels > 2 || channels < 1) {
-    return kBadParameterError;
-  }
-
-  num_render_input_channels_ = channels;
-
-  return InitializeLocked();
-}
-
-int AudioProcessingImpl::num_reverse_channels() const {
-  return num_render_input_channels_;
-}
-
-int AudioProcessingImpl::set_num_channels(
-    int input_channels,
-    int output_channels) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  if (output_channels > input_channels) {
-    return kBadParameterError;
-  }
-
-  // Only stereo supported currently.
-  if (input_channels > 2 || input_channels < 1) {
-    return kBadParameterError;
-  }
-
-  if (output_channels > 2 || output_channels < 1) {
-    return kBadParameterError;
-  }
-
-  num_capture_input_channels_ = input_channels;
-  num_capture_output_channels_ = output_channels;
-
-  return InitializeLocked();
-}
-
-int AudioProcessingImpl::num_input_channels() const {
-  return num_capture_input_channels_;
-}
-
-int AudioProcessingImpl::num_output_channels() const {
-  return num_capture_output_channels_;
-}
-
-int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  int err = kNoError;
-
-  if (frame == NULL) {
-    return kNullPointerError;
-  }
-
-  if (frame->_frequencyInHz !=
-      static_cast<WebRtc_UWord32>(sample_rate_hz_)) {
-    return kBadSampleRateError;
-  }
-
-  if (frame->_audioChannel != num_capture_input_channels_) {
-    return kBadNumberChannelsError;
-  }
-
-  if (frame->_payloadDataLengthInSamples != samples_per_channel_) {
-    return kBadDataLengthError;
-  }
-
-  if (debug_file_->Open()) {
-    WebRtc_UWord8 event = kCaptureEvent;
-    if (!debug_file_->Write(&event, sizeof(event))) {
-      return kFileError;
-    }
-
-    if (!debug_file_->Write(&frame->_frequencyInHz,
-                                   sizeof(frame->_frequencyInHz))) {
-      return kFileError;
-    }
-
-    if (!debug_file_->Write(&frame->_audioChannel,
-                                   sizeof(frame->_audioChannel))) {
-      return kFileError;
-    }
-
-    if (!debug_file_->Write(&frame->_payloadDataLengthInSamples,
-        sizeof(frame->_payloadDataLengthInSamples))) {
-      return kFileError;
-    }
-
-    if (!debug_file_->Write(frame->_payloadData,
-        sizeof(WebRtc_Word16) * frame->_payloadDataLengthInSamples *
-        frame->_audioChannel)) {
-      return kFileError;
-    }
-  }
-
-  capture_audio_->DeinterleaveFrom(frame);
-
-  // TODO(ajm): experiment with mixing and AEC placement.
-  if (num_capture_output_channels_ < num_capture_input_channels_) {
-    capture_audio_->Mix(num_capture_output_channels_);
-
-    frame->_audioChannel = num_capture_output_channels_;
-  }
-
-  if (sample_rate_hz_ == kSampleRate32kHz) {
-    for (int i = 0; i < num_capture_input_channels_; i++) {
-      // Split into a low and high band.
-      SplittingFilterAnalysis(capture_audio_->data(i),
-                              capture_audio_->low_pass_split_data(i),
-                              capture_audio_->high_pass_split_data(i),
-                              capture_audio_->analysis_filter_state1(i),
-                              capture_audio_->analysis_filter_state2(i));
-    }
-  }
-
-  err = high_pass_filter_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = gain_control_->AnalyzeCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = echo_cancellation_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  if (echo_control_mobile_->is_enabled() &&
-      noise_suppression_->is_enabled()) {
-    capture_audio_->CopyLowPassToReference();
-  }
-
-  err = noise_suppression_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = echo_control_mobile_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = voice_detection_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = gain_control_->ProcessCaptureAudio(capture_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  //err = level_estimator_->ProcessCaptureAudio(capture_audio_);
-  //if (err != kNoError) {
-  //  return err;
-  //}
-
-  if (sample_rate_hz_ == kSampleRate32kHz) {
-    for (int i = 0; i < num_capture_output_channels_; i++) {
-      // Recombine low and high bands.
-      SplittingFilterSynthesis(capture_audio_->low_pass_split_data(i),
-                               capture_audio_->high_pass_split_data(i),
-                               capture_audio_->data(i),
-                               capture_audio_->synthesis_filter_state1(i),
-                               capture_audio_->synthesis_filter_state2(i));
-    }
-  }
-
-  capture_audio_->InterleaveTo(frame);
-
-  return kNoError;
-}
-
-int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  int err = kNoError;
-
-  if (frame == NULL) {
-    return kNullPointerError;
-  }
-
-  if (frame->_frequencyInHz !=
-      static_cast<WebRtc_UWord32>(sample_rate_hz_)) {
-    return kBadSampleRateError;
-  }
-
-  if (frame->_audioChannel != num_render_input_channels_) {
-    return kBadNumberChannelsError;
-  }
-
-  if (frame->_payloadDataLengthInSamples != samples_per_channel_) {
-    return kBadDataLengthError;
-  }
-
-  if (debug_file_->Open()) {
-    WebRtc_UWord8 event = kRenderEvent;
-    if (!debug_file_->Write(&event, sizeof(event))) {
-      return kFileError;
-    }
-
-    if (!debug_file_->Write(&frame->_frequencyInHz,
-                                   sizeof(frame->_frequencyInHz))) {
-      return kFileError;
-    }
-
-    if (!debug_file_->Write(&frame->_audioChannel,
-                                   sizeof(frame->_audioChannel))) {
-      return kFileError;
-    }
-
-    if (!debug_file_->Write(&frame->_payloadDataLengthInSamples,
-        sizeof(frame->_payloadDataLengthInSamples))) {
-      return kFileError;
-    }
-
-    if (!debug_file_->Write(frame->_payloadData,
-        sizeof(WebRtc_Word16) * frame->_payloadDataLengthInSamples *
-        frame->_audioChannel)) {
-      return kFileError;
-    }
-  }
-
-  render_audio_->DeinterleaveFrom(frame);
-
-  // TODO(ajm): turn the splitting filter into a component?
-  if (sample_rate_hz_ == kSampleRate32kHz) {
-    for (int i = 0; i < num_render_input_channels_; i++) {
-      // Split into low and high band.
-      SplittingFilterAnalysis(render_audio_->data(i),
-                              render_audio_->low_pass_split_data(i),
-                              render_audio_->high_pass_split_data(i),
-                              render_audio_->analysis_filter_state1(i),
-                              render_audio_->analysis_filter_state2(i));
-    }
-  }
-
-  // TODO(ajm): warnings possible from components?
-  err = echo_cancellation_->ProcessRenderAudio(render_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = echo_control_mobile_->ProcessRenderAudio(render_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  err = gain_control_->ProcessRenderAudio(render_audio_);
-  if (err != kNoError) {
-    return err;
-  }
-
-  //err = level_estimator_->AnalyzeReverseStream(render_audio_);
-  //if (err != kNoError) {
-  //  return err;
-  //}
-
-  was_stream_delay_set_ = false;
-  return err;  // TODO(ajm): this is for returning warnings; necessary?
-}
-
-int AudioProcessingImpl::set_stream_delay_ms(int delay) {
-  was_stream_delay_set_ = true;
-  if (delay < 0) {
-    return kBadParameterError;
-  }
-
-  // TODO(ajm): the max is rather arbitrarily chosen; investigate.
-  if (delay > 500) {
-    stream_delay_ms_ = 500;
-    return kBadStreamParameterWarning;
-  }
-
-  stream_delay_ms_ = delay;
-  return kNoError;
-}
-
-int AudioProcessingImpl::stream_delay_ms() const {
-  return stream_delay_ms_;
-}
-
-bool AudioProcessingImpl::was_stream_delay_set() const {
-  return was_stream_delay_set_;
-}
-
-int AudioProcessingImpl::StartDebugRecording(
-    const char filename[AudioProcessing::kMaxFilenameSize]) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize);
-
-  if (filename == NULL) {
-    return kNullPointerError;
-  }
-
-  // Stop any ongoing recording.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
-      return kFileError;
-    }
-  }
-
-  if (debug_file_->OpenFile(filename, false) == -1) {
-    debug_file_->CloseFile();
-    return kFileError;
-  }
-
-  if (debug_file_->WriteText("%s\n", kMagicNumber) == -1) {
-    debug_file_->CloseFile();
-    return kFileError;
-  }
-
-  // TODO(ajm): should we do this? If so, we need the number of channels etc.
-  // Record the default sample rate.
-  WebRtc_UWord8 event = kInitializeEvent;
-  if (!debug_file_->Write(&event, sizeof(event))) {
-    return kFileError;
-  }
-
-  if (!debug_file_->Write(&sample_rate_hz_, sizeof(sample_rate_hz_))) {
-    return kFileError;
-  }
-
-  return kNoError;
-}
-
-int AudioProcessingImpl::StopDebugRecording() {
-  CriticalSectionScoped crit_scoped(*crit_);
-  // We just return if recording hasn't started.
-  if (debug_file_->Open()) {
-    if (debug_file_->CloseFile() == -1) {
-      return kFileError;
-    }
-  }
-
-  return kNoError;
-}
-
-EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
-  return echo_cancellation_;
-}
-
-EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
-  return echo_control_mobile_;
-}
-
-GainControl* AudioProcessingImpl::gain_control() const {
-  return gain_control_;
-}
-
-HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
-  return high_pass_filter_;
-}
-
-LevelEstimator* AudioProcessingImpl::level_estimator() const {
-  return level_estimator_;
-}
-
-NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
-  return noise_suppression_;
-}
-
-VoiceDetection* AudioProcessingImpl::voice_detection() const {
-  return voice_detection_;
-}
-
-WebRtc_Word32 AudioProcessingImpl::Version(WebRtc_Word8* version,
-    WebRtc_UWord32& bytes_remaining, WebRtc_UWord32& position) const {
-  if (version == NULL) {
-    /*WEBRTC_TRACE(webrtc::kTraceError,
-               webrtc::kTraceAudioProcessing,
-               -1,
-               "Null version pointer");*/
-    return kNullPointerError;
-  }
-  memset(&version[position], 0, bytes_remaining);
-
-  WebRtc_Word8 my_version[] = "AudioProcessing 1.0.0";
-  // Includes null termination.
-  WebRtc_UWord32 length = static_cast<WebRtc_UWord32>(strlen(my_version));
-  if (bytes_remaining < length) {
-    /*WEBRTC_TRACE(webrtc::kTraceError,
-               webrtc::kTraceAudioProcessing,
-               -1,
-               "Buffer of insufficient length");*/
-    return kBadParameterError;
-  }
-  memcpy(&version[position], my_version, length);
-  bytes_remaining -= length;
-  position += length;
-
-  std::list<ProcessingComponent*>::const_iterator it;
-  for (it = component_list_.begin(); it != component_list_.end(); it++) {
-    char component_version[256];
-    strcpy(component_version, "\n");
-    int err = (*it)->get_version(&component_version[1],
-                                 sizeof(component_version) - 1);
-    if (err != kNoError) {
-      return err;
-    }
-    if (strncmp(&component_version[1], "\0", 1) == 0) {
-      // Assume empty if first byte is NULL.
-      continue;
-    }
-
-    length = static_cast<WebRtc_UWord32>(strlen(component_version));
-    if (bytes_remaining < length) {
-      /*WEBRTC_TRACE(webrtc::kTraceError,
-                 webrtc::kTraceAudioProcessing,
-                 -1,
-                 "Buffer of insufficient length");*/
-      return kBadParameterError;
-    }
-    memcpy(&version[position], component_version, length);
-    bytes_remaining -= length;
-    position += length;
-  }
-
-  return kNoError;
-}
-
-WebRtc_Word32 AudioProcessingImpl::ChangeUniqueId(const WebRtc_Word32 id) {
-  CriticalSectionScoped crit_scoped(*crit_);
-  /*WEBRTC_TRACE(webrtc::kTraceModuleCall,
-             webrtc::kTraceAudioProcessing,
-             id_,
-             "ChangeUniqueId(new id = %d)",
-             id);*/
-  id_ = id;
-
-  return kNoError;
-}
-}  // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/audio_processing_impl.h b/src/modules/audio_processing/main/source/audio_processing_impl.h
deleted file mode 100644
index 9707bde..0000000
--- a/src/modules/audio_processing/main/source/audio_processing_impl.h
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
-
-#include <list>
-
-#include "audio_processing.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class FileWrapper;
-
-class AudioBuffer;
-class EchoCancellationImpl;
-class EchoControlMobileImpl;
-class GainControlImpl;
-class HighPassFilterImpl;
-class LevelEstimatorImpl;
-class NoiseSuppressionImpl;
-class ProcessingComponent;
-class VoiceDetectionImpl;
-
-class AudioProcessingImpl : public AudioProcessing {
- public:
-  enum {
-    kSampleRate8kHz = 8000,
-    kSampleRate16kHz = 16000,
-    kSampleRate32kHz = 32000
-  };
-
-  explicit AudioProcessingImpl(int id);
-  virtual ~AudioProcessingImpl();
-
-  CriticalSectionWrapper* crit() const;
-
-  int split_sample_rate_hz() const;
-  bool was_stream_delay_set() const;
-
-  // AudioProcessing methods.
-  virtual int Initialize();
-  virtual int InitializeLocked();
-  virtual int set_sample_rate_hz(int rate);
-  virtual int sample_rate_hz() const;
-  virtual int set_num_channels(int input_channels, int output_channels);
-  virtual int num_input_channels() const;
-  virtual int num_output_channels() const;
-  virtual int set_num_reverse_channels(int channels);
-  virtual int num_reverse_channels() const;
-  virtual int ProcessStream(AudioFrame* frame);
-  virtual int AnalyzeReverseStream(AudioFrame* frame);
-  virtual int set_stream_delay_ms(int delay);
-  virtual int stream_delay_ms() const;
-  virtual int StartDebugRecording(const char filename[kMaxFilenameSize]);
-  virtual int StopDebugRecording();
-  virtual EchoCancellation* echo_cancellation() const;
-  virtual EchoControlMobile* echo_control_mobile() const;
-  virtual GainControl* gain_control() const;
-  virtual HighPassFilter* high_pass_filter() const;
-  virtual LevelEstimator* level_estimator() const;
-  virtual NoiseSuppression* noise_suppression() const;
-  virtual VoiceDetection* voice_detection() const;
-
-  // Module methods.
-  virtual WebRtc_Word32 Version(WebRtc_Word8* version,
-                              WebRtc_UWord32& remainingBufferInBytes,
-                              WebRtc_UWord32& position) const;
-  virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- private:
-  int id_;
-
-  EchoCancellationImpl* echo_cancellation_;
-  EchoControlMobileImpl* echo_control_mobile_;
-  GainControlImpl* gain_control_;
-  HighPassFilterImpl* high_pass_filter_;
-  LevelEstimatorImpl* level_estimator_;
-  NoiseSuppressionImpl* noise_suppression_;
-  VoiceDetectionImpl* voice_detection_;
-
-  std::list<ProcessingComponent*> component_list_;
-
-  FileWrapper* debug_file_;
-  CriticalSectionWrapper* crit_;
-
-  AudioBuffer* render_audio_;
-  AudioBuffer* capture_audio_;
-
-  int sample_rate_hz_;
-  int split_sample_rate_hz_;
-  int samples_per_channel_;
-  int stream_delay_ms_;
-  bool was_stream_delay_set_;
-
-  int num_render_input_channels_;
-  int num_capture_input_channels_;
-  int num_capture_output_channels_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
diff --git a/src/modules/audio_processing/main/source/echo_cancellation_impl.cc b/src/modules/audio_processing/main/source/echo_cancellation_impl.cc
deleted file mode 100644
index 886d5f1..0000000
--- a/src/modules/audio_processing/main/source/echo_cancellation_impl.cc
+++ /dev/null
@@ -1,348 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "echo_cancellation_impl.h"
-
-#include <cassert>
-#include <string.h>
-
-#include "critical_section_wrapper.h"
-#include "echo_cancellation.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-
-typedef void Handle;
-
-namespace {
-WebRtc_Word16 MapSetting(EchoCancellation::SuppressionLevel level) {
-  switch (level) {
-    case EchoCancellation::kLowSuppression:
-      return kAecNlpConservative;
-    case EchoCancellation::kModerateSuppression:
-      return kAecNlpModerate;
-    case EchoCancellation::kHighSuppression:
-      return kAecNlpAggressive;
-    default:
-      return -1;
-  }
-}
-
-int MapError(int err) {
-  switch (err) {
-    case AEC_UNSUPPORTED_FUNCTION_ERROR:
-      return AudioProcessing::kUnsupportedFunctionError;
-      break;
-    case AEC_BAD_PARAMETER_ERROR:
-      return AudioProcessing::kBadParameterError;
-      break;
-    case AEC_BAD_PARAMETER_WARNING:
-      return AudioProcessing::kBadStreamParameterWarning;
-      break;
-    default:
-      // AEC_UNSPECIFIED_ERROR
-      // AEC_UNINITIALIZED_ERROR
-      // AEC_NULL_POINTER_ERROR
-      return AudioProcessing::kUnspecifiedError;
-  }
-}
-}  // namespace
-
-EchoCancellationImpl::EchoCancellationImpl(const AudioProcessingImpl* apm)
-  : ProcessingComponent(apm),
-    apm_(apm),
-    drift_compensation_enabled_(false),
-    metrics_enabled_(false),
-    suppression_level_(kModerateSuppression),
-    device_sample_rate_hz_(48000),
-    stream_drift_samples_(0),
-    was_stream_drift_set_(false),
-    stream_has_echo_(false) {}
-
-EchoCancellationImpl::~EchoCancellationImpl() {}
-
-int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  assert(audio->samples_per_split_channel() <= 160);
-  assert(audio->num_channels() == apm_->num_reverse_channels());
-
-  int err = apm_->kNoError;
-
-  // The ordering convention must be followed to pass to the correct AEC.
-  size_t handle_index = 0;
-  for (int i = 0; i < apm_->num_output_channels(); i++) {
-    for (int j = 0; j < audio->num_channels(); j++) {
-      Handle* my_handle = static_cast<Handle*>(handle(handle_index));
-      err = WebRtcAec_BufferFarend(
-          my_handle,
-          audio->low_pass_split_data(j),
-          static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
-
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);  // TODO(ajm): warning possible?
-      }
-
-      handle_index++;
-    }
-  }
-
-  return apm_->kNoError;
-}
-
-int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  if (!apm_->was_stream_delay_set()) {
-    return apm_->kStreamParameterNotSetError;
-  }
-
-  if (drift_compensation_enabled_ && !was_stream_drift_set_) {
-    return apm_->kStreamParameterNotSetError;
-  }
-
-  assert(audio->samples_per_split_channel() <= 160);
-  assert(audio->num_channels() == apm_->num_output_channels());
-
-  int err = apm_->kNoError;
-
-  // The ordering convention must be followed to pass to the correct AEC.
-  size_t handle_index = 0;
-  stream_has_echo_ = false;
-  for (int i = 0; i < audio->num_channels(); i++) {
-    for (int j = 0; j < apm_->num_reverse_channels(); j++) {
-      Handle* my_handle = handle(handle_index);
-      err = WebRtcAec_Process(
-          my_handle,
-          audio->low_pass_split_data(i),
-          audio->high_pass_split_data(i),
-          audio->low_pass_split_data(i),
-          audio->high_pass_split_data(i),
-          static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
-          apm_->stream_delay_ms(),
-          stream_drift_samples_);
-
-      if (err != apm_->kNoError) {
-        err = GetHandleError(my_handle);
-        // TODO(ajm): Figure out how to return warnings properly.
-        if (err != apm_->kBadStreamParameterWarning) {
-          return err;
-        }
-      }
-
-      WebRtc_Word16 status = 0;
-      err = WebRtcAec_get_echo_status(my_handle, &status);
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);
-      }
-
-      if (status == 1) {
-        stream_has_echo_ = true;
-      }
-
-      handle_index++;
-    }
-  }
-
-  was_stream_drift_set_ = false;
-  return apm_->kNoError;
-}
-
-int EchoCancellationImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  // Ensure AEC and AECM are not both enabled.
-  if (enable && apm_->echo_control_mobile()->is_enabled()) {
-    return apm_->kBadParameterError;
-  }
-
-  return EnableComponent(enable);
-}
-
-bool EchoCancellationImpl::is_enabled() const {
-  return is_component_enabled();
-}
-
-int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (MapSetting(level) == -1) {
-    return apm_->kBadParameterError;
-  }
-
-  suppression_level_ = level;
-  return Configure();
-}
-
-EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level()
-    const {
-  return suppression_level_;
-}
-
-int EchoCancellationImpl::enable_drift_compensation(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  drift_compensation_enabled_ = enable;
-  return Configure();
-}
-
-bool EchoCancellationImpl::is_drift_compensation_enabled() const {
-  return drift_compensation_enabled_;
-}
-
-int EchoCancellationImpl::set_device_sample_rate_hz(int rate) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (rate < 8000 || rate > 96000) {
-    return apm_->kBadParameterError;
-  }
-
-  device_sample_rate_hz_ = rate;
-  return Initialize();
-}
-
-int EchoCancellationImpl::device_sample_rate_hz() const {
-  return device_sample_rate_hz_;
-}
-
-int EchoCancellationImpl::set_stream_drift_samples(int drift) {
-  was_stream_drift_set_ = true;
-  stream_drift_samples_ = drift;
-  return apm_->kNoError;
-}
-
-int EchoCancellationImpl::stream_drift_samples() const {
-  return stream_drift_samples_;
-}
-
-int EchoCancellationImpl::enable_metrics(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  metrics_enabled_ = enable;
-  return Configure();
-}
-
-bool EchoCancellationImpl::are_metrics_enabled() const {
-  return metrics_enabled_;
-}
-
-// TODO(ajm): we currently just use the metrics from the first AEC. Think more
-//            aboue the best way to extend this to multi-channel.
-int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (metrics == NULL) {
-    return apm_->kNullPointerError;
-  }
-
-  if (!is_component_enabled() || !metrics_enabled_) {
-    return apm_->kNotEnabledError;
-  }
-
-  AecMetrics my_metrics;
-  memset(&my_metrics, 0, sizeof(my_metrics));
-  memset(metrics, 0, sizeof(Metrics));
-
-  Handle* my_handle = static_cast<Handle*>(handle(0));
-  int err = WebRtcAec_GetMetrics(my_handle, &my_metrics);
-  if (err != apm_->kNoError) {
-    return GetHandleError(my_handle);
-  }
-
-  metrics->residual_echo_return_loss.instant = my_metrics.rerl.instant;
-  metrics->residual_echo_return_loss.average = my_metrics.rerl.average;
-  metrics->residual_echo_return_loss.maximum = my_metrics.rerl.max;
-  metrics->residual_echo_return_loss.minimum = my_metrics.rerl.min;
-
-  metrics->echo_return_loss.instant = my_metrics.erl.instant;
-  metrics->echo_return_loss.average = my_metrics.erl.average;
-  metrics->echo_return_loss.maximum = my_metrics.erl.max;
-  metrics->echo_return_loss.minimum = my_metrics.erl.min;
-
-  metrics->echo_return_loss_enhancement.instant = my_metrics.erle.instant;
-  metrics->echo_return_loss_enhancement.average = my_metrics.erle.average;
-  metrics->echo_return_loss_enhancement.maximum = my_metrics.erle.max;
-  metrics->echo_return_loss_enhancement.minimum = my_metrics.erle.min;
-
-  metrics->a_nlp.instant = my_metrics.aNlp.instant;
-  metrics->a_nlp.average = my_metrics.aNlp.average;
-  metrics->a_nlp.maximum = my_metrics.aNlp.max;
-  metrics->a_nlp.minimum = my_metrics.aNlp.min;
-
-  return apm_->kNoError;
-}
-
-bool EchoCancellationImpl::stream_has_echo() const {
-  return stream_has_echo_;
-}
-
-int EchoCancellationImpl::Initialize() {
-  int err = ProcessingComponent::Initialize();
-  if (err != apm_->kNoError || !is_component_enabled()) {
-    return err;
-  }
-
-  was_stream_drift_set_ = false;
-
-  return apm_->kNoError;
-}
-
-int EchoCancellationImpl::get_version(char* version,
-                                      int version_len_bytes) const {
-  if (WebRtcAec_get_version(version, version_len_bytes) != 0) {
-      return apm_->kBadParameterError;
-  }
-
-  return apm_->kNoError;
-}
-
-void* EchoCancellationImpl::CreateHandle() const {
-  Handle* handle = NULL;
-  if (WebRtcAec_Create(&handle) != apm_->kNoError) {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
-}
-
-int EchoCancellationImpl::DestroyHandle(void* handle) const {
-  assert(handle != NULL);
-  return WebRtcAec_Free(static_cast<Handle*>(handle));
-}
-
-int EchoCancellationImpl::InitializeHandle(void* handle) const {
-  assert(handle != NULL);
-  return WebRtcAec_Init(static_cast<Handle*>(handle),
-                       apm_->sample_rate_hz(),
-                       device_sample_rate_hz_);
-}
-
-int EchoCancellationImpl::ConfigureHandle(void* handle) const {
-  assert(handle != NULL);
-  AecConfig config;
-  config.metricsMode = metrics_enabled_;
-  config.nlpMode = MapSetting(suppression_level_);
-  config.skewMode = drift_compensation_enabled_;
-
-  return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
-}
-
-int EchoCancellationImpl::num_handles_required() const {
-  return apm_->num_output_channels() *
-         apm_->num_reverse_channels();
-}
-
-int EchoCancellationImpl::GetHandleError(void* handle) const {
-  assert(handle != NULL);
-  return MapError(WebRtcAec_get_error_code(static_cast<Handle*>(handle)));
-}
-}  // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/echo_cancellation_impl.h b/src/modules/audio_processing/main/source/echo_cancellation_impl.h
deleted file mode 100644
index 380a698..0000000
--- a/src/modules/audio_processing/main/source/echo_cancellation_impl.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CANCELLATION_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CANCELLATION_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class EchoCancellationImpl : public EchoCancellation,
-                             public ProcessingComponent {
- public:
-  explicit EchoCancellationImpl(const AudioProcessingImpl* apm);
-  virtual ~EchoCancellationImpl();
-
-  int ProcessRenderAudio(const AudioBuffer* audio);
-  int ProcessCaptureAudio(AudioBuffer* audio);
-
-  // EchoCancellation implementation.
-  virtual bool is_enabled() const;
-
-  // ProcessingComponent implementation.
-  virtual int Initialize();
-  virtual int get_version(char* version, int version_len_bytes) const;
-
- private:
-  // EchoCancellation implementation.
-  virtual int Enable(bool enable);
-  virtual int enable_drift_compensation(bool enable);
-  virtual bool is_drift_compensation_enabled() const;
-  virtual int set_device_sample_rate_hz(int rate);
-  virtual int device_sample_rate_hz() const;
-  virtual int set_stream_drift_samples(int drift);
-  virtual int stream_drift_samples() const;
-  virtual int set_suppression_level(SuppressionLevel level);
-  virtual SuppressionLevel suppression_level() const;
-  virtual int enable_metrics(bool enable);
-  virtual bool are_metrics_enabled() const;
-  virtual bool stream_has_echo() const;
-  virtual int GetMetrics(Metrics* metrics);
-
-  // ProcessingComponent implementation.
-  virtual void* CreateHandle() const;
-  virtual int InitializeHandle(void* handle) const;
-  virtual int ConfigureHandle(void* handle) const;
-  virtual int DestroyHandle(void* handle) const;
-  virtual int num_handles_required() const;
-  virtual int GetHandleError(void* handle) const;
-
-  const AudioProcessingImpl* apm_;
-  bool drift_compensation_enabled_;
-  bool metrics_enabled_;
-  SuppressionLevel suppression_level_;
-  int device_sample_rate_hz_;
-  int stream_drift_samples_;
-  bool was_stream_drift_set_;
-  bool stream_has_echo_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CANCELLATION_IMPL_H_
diff --git a/src/modules/audio_processing/main/source/echo_control_mobile_impl.cc b/src/modules/audio_processing/main/source/echo_control_mobile_impl.cc
deleted file mode 100644
index 1cd2502..0000000
--- a/src/modules/audio_processing/main/source/echo_control_mobile_impl.cc
+++ /dev/null
@@ -1,245 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "echo_control_mobile_impl.h"
-
-#include <cassert>
-
-#include "critical_section_wrapper.h"
-#include "echo_control_mobile.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-
-typedef void Handle;
-
-namespace {
-WebRtc_Word16 MapSetting(EchoControlMobile::RoutingMode mode) {
-  switch (mode) {
-    case EchoControlMobile::kQuietEarpieceOrHeadset:
-      return 0;
-    case EchoControlMobile::kEarpiece:
-      return 1;
-    case EchoControlMobile::kLoudEarpiece:
-      return 2;
-    case EchoControlMobile::kSpeakerphone:
-      return 3;
-    case EchoControlMobile::kLoudSpeakerphone:
-      return 4;
-    default:
-      return -1;
-  }
-}
-
-int MapError(int err) {
-  switch (err) {
-    case AECM_UNSUPPORTED_FUNCTION_ERROR:
-      return AudioProcessing::kUnsupportedFunctionError;
-    case AECM_BAD_PARAMETER_ERROR:
-      return AudioProcessing::kBadParameterError;
-    case AECM_BAD_PARAMETER_WARNING:
-      return AudioProcessing::kBadStreamParameterWarning;
-    default:
-      // AECMOBFIX_UNSPECIFIED_ERROR
-      // AECMOBFIX_UNINITIALIZED_ERROR
-      // AECMOBFIX_NULL_POINTER_ERROR
-      return AudioProcessing::kUnspecifiedError;
-  }
-}
-}  // namespace
-
-EchoControlMobileImpl::EchoControlMobileImpl(const AudioProcessingImpl* apm)
-  : ProcessingComponent(apm),
-    apm_(apm),
-    routing_mode_(kSpeakerphone),
-    comfort_noise_enabled_(true) {}
-
-EchoControlMobileImpl::~EchoControlMobileImpl() {}
-
-int EchoControlMobileImpl::ProcessRenderAudio(const AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  assert(audio->samples_per_split_channel() <= 160);
-  assert(audio->num_channels() == apm_->num_reverse_channels());
-
-  int err = apm_->kNoError;
-
-  // The ordering convention must be followed to pass to the correct AECM.
-  size_t handle_index = 0;
-  for (int i = 0; i < apm_->num_output_channels(); i++) {
-    for (int j = 0; j < audio->num_channels(); j++) {
-      Handle* my_handle = static_cast<Handle*>(handle(handle_index));
-      err = WebRtcAecm_BufferFarend(
-          my_handle,
-          audio->low_pass_split_data(j),
-          static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
-
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);  // TODO(ajm): warning possible?
-      }
-
-      handle_index++;
-    }
-  }
-
-  return apm_->kNoError;
-}
-
-int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  if (!apm_->was_stream_delay_set()) {
-    return apm_->kStreamParameterNotSetError;
-  }
-
-  assert(audio->samples_per_split_channel() <= 160);
-  assert(audio->num_channels() == apm_->num_output_channels());
-
-  int err = apm_->kNoError;
-
-  // The ordering convention must be followed to pass to the correct AECM.
-  size_t handle_index = 0;
-  for (int i = 0; i < audio->num_channels(); i++) {
-    // TODO(ajm): improve how this works, possibly inside AECM.
-    //            This is kind of hacked up.
-    WebRtc_Word16* noisy = audio->low_pass_reference(i);
-    WebRtc_Word16* clean = audio->low_pass_split_data(i);
-    if (noisy == NULL) {
-      noisy = clean;
-      clean = NULL;
-    }
-    for (int j = 0; j < apm_->num_reverse_channels(); j++) {
-      Handle* my_handle = static_cast<Handle*>(handle(handle_index));
-      err = WebRtcAecm_Process(
-          my_handle,
-          noisy,
-          clean,
-          audio->low_pass_split_data(i),
-          static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
-          apm_->stream_delay_ms());
-
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);  // TODO(ajm): warning possible?
-      }
-
-      handle_index++;
-    }
-  }
-
-  return apm_->kNoError;
-}
-
-int EchoControlMobileImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  // Ensure AEC and AECM are not both enabled.
-  if (enable && apm_->echo_cancellation()->is_enabled()) {
-    return apm_->kBadParameterError;
-  }
-
-  return EnableComponent(enable);
-}
-
-bool EchoControlMobileImpl::is_enabled() const {
-  return is_component_enabled();
-}
-
-int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (MapSetting(mode) == -1) {
-    return apm_->kBadParameterError;
-  }
-
-  routing_mode_ = mode;
-  return Configure();
-}
-
-EchoControlMobile::RoutingMode EchoControlMobileImpl::routing_mode()
-    const {
-  return routing_mode_;
-}
-
-int EchoControlMobileImpl::enable_comfort_noise(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  comfort_noise_enabled_ = enable;
-  return Configure();
-}
-
-bool EchoControlMobileImpl::is_comfort_noise_enabled() const {
-  return comfort_noise_enabled_;
-}
-
-int EchoControlMobileImpl::Initialize() {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  if (apm_->sample_rate_hz() == apm_->kSampleRate32kHz) {
-    // AECM doesn't support super-wideband.
-    return apm_->kBadSampleRateError;
-  }
-
-  return ProcessingComponent::Initialize();
-}
-
-int EchoControlMobileImpl::get_version(char* version,
-                                       int version_len_bytes) const {
-  if (WebRtcAecm_get_version(version, version_len_bytes) != 0) {
-      return apm_->kBadParameterError;
-  }
-
-  return apm_->kNoError;
-}
-
-void* EchoControlMobileImpl::CreateHandle() const {
-  Handle* handle = NULL;
-  if (WebRtcAecm_Create(&handle) != apm_->kNoError) {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
-}
-
-int EchoControlMobileImpl::DestroyHandle(void* handle) const {
-  return WebRtcAecm_Free(static_cast<Handle*>(handle));
-}
-
-int EchoControlMobileImpl::InitializeHandle(void* handle) const {
-  return WebRtcAecm_Init(static_cast<Handle*>(handle),
-                         apm_->sample_rate_hz(),
-                         48000); // Dummy value. This isn't actually
-                                 // required by AECM.
-}
-
-int EchoControlMobileImpl::ConfigureHandle(void* handle) const {
-  AecmConfig config;
-  config.cngMode = comfort_noise_enabled_;
-  config.echoMode = MapSetting(routing_mode_);
-
-  return WebRtcAecm_set_config(static_cast<Handle*>(handle), config);
-}
-
-int EchoControlMobileImpl::num_handles_required() const {
-  return apm_->num_output_channels() *
-         apm_->num_reverse_channels();
-}
-
-int EchoControlMobileImpl::GetHandleError(void* handle) const {
-  assert(handle != NULL);
-  return MapError(WebRtcAecm_get_error_code(static_cast<Handle*>(handle)));
-}
-}  // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/echo_control_mobile_impl.h b/src/modules/audio_processing/main/source/echo_control_mobile_impl.h
deleted file mode 100644
index 2fd6248..0000000
--- a/src/modules/audio_processing/main/source/echo_control_mobile_impl.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CONTROL_MOBILE_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CONTROL_MOBILE_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class EchoControlMobileImpl : public EchoControlMobile,
-                              public ProcessingComponent {
- public:
-  explicit EchoControlMobileImpl(const AudioProcessingImpl* apm);
-  virtual ~EchoControlMobileImpl();
-
-  int ProcessRenderAudio(const AudioBuffer* audio);
-  int ProcessCaptureAudio(AudioBuffer* audio);
-
-  // EchoControlMobile implementation.
-  virtual bool is_enabled() const;
-
-  // ProcessingComponent implementation.
-  virtual int Initialize();
-  virtual int get_version(char* version, int version_len_bytes) const;
-
- private:
-  // EchoControlMobile implementation.
-  virtual int Enable(bool enable);
-  virtual int set_routing_mode(RoutingMode mode);
-  virtual RoutingMode routing_mode() const;
-  virtual int enable_comfort_noise(bool enable);
-  virtual bool is_comfort_noise_enabled() const;
-
-  // ProcessingComponent implementation.
-  virtual void* CreateHandle() const;
-  virtual int InitializeHandle(void* handle) const;
-  virtual int ConfigureHandle(void* handle) const;
-  virtual int DestroyHandle(void* handle) const;
-  virtual int num_handles_required() const;
-  virtual int GetHandleError(void* handle) const;
-
-  const AudioProcessingImpl* apm_;
-  RoutingMode routing_mode_;
-  bool comfort_noise_enabled_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_ECHO_CONTROL_MOBILE_IMPL_H_
diff --git a/src/modules/audio_processing/main/source/gain_control_impl.cc b/src/modules/audio_processing/main/source/gain_control_impl.cc
deleted file mode 100644
index dc3e565..0000000
--- a/src/modules/audio_processing/main/source/gain_control_impl.cc
+++ /dev/null
@@ -1,391 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gain_control_impl.h"
-
-#include <cassert>
-
-#include "critical_section_wrapper.h"
-#include "gain_control.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-
-typedef void Handle;
-
-/*template <class T>
-class GainControlHandle : public ComponentHandle<T> {
-  public:
-    GainControlHandle();
-    virtual ~GainControlHandle();
-
-    virtual int Create();
-    virtual T* ptr() const;
-
-  private:
-    T* handle;
-};*/
-
-namespace {
-WebRtc_Word16 MapSetting(GainControl::Mode mode) {
-  switch (mode) {
-    case GainControl::kAdaptiveAnalog:
-      return kAgcModeAdaptiveAnalog;
-      break;
-    case GainControl::kAdaptiveDigital:
-      return kAgcModeAdaptiveDigital;
-      break;
-    case GainControl::kFixedDigital:
-      return kAgcModeFixedDigital;
-      break;
-    default:
-      return -1;
-  }
-}
-}  // namespace
-
-GainControlImpl::GainControlImpl(const AudioProcessingImpl* apm)
-  : ProcessingComponent(apm),
-    apm_(apm),
-    mode_(kAdaptiveAnalog),
-    minimum_capture_level_(0),
-    maximum_capture_level_(255),
-    limiter_enabled_(true),
-    target_level_dbfs_(3),
-    compression_gain_db_(9),
-    analog_capture_level_(0),
-    was_analog_level_set_(false),
-    stream_is_saturated_(false) {}
-
-GainControlImpl::~GainControlImpl() {}
-
-int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  assert(audio->samples_per_split_channel() <= 160);
-
-  WebRtc_Word16* mixed_data = audio->low_pass_split_data(0);
-  if (audio->num_channels() > 1) {
-    audio->CopyAndMixLowPass(1);
-    mixed_data = audio->mixed_low_pass_data(0);
-  }
-
-  for (int i = 0; i < num_handles(); i++) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
-    int err = WebRtcAgc_AddFarend(
-        my_handle,
-        mixed_data,
-        static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
-
-    if (err != apm_->kNoError) {
-      return GetHandleError(my_handle);
-    }
-  }
-
-  return apm_->kNoError;
-}
-
-int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  assert(audio->samples_per_split_channel() <= 160);
-  assert(audio->num_channels() == num_handles());
-
-  int err = apm_->kNoError;
-
-  if (mode_ == kAdaptiveAnalog) {
-    for (int i = 0; i < num_handles(); i++) {
-      Handle* my_handle = static_cast<Handle*>(handle(i));
-      err = WebRtcAgc_AddMic(
-          my_handle,
-          audio->low_pass_split_data(i),
-          audio->high_pass_split_data(i),
-          static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
-
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);
-      }
-    }
-  } else if (mode_ == kAdaptiveDigital) {
-
-    for (int i = 0; i < num_handles(); i++) {
-      Handle* my_handle = static_cast<Handle*>(handle(i));
-      WebRtc_Word32 capture_level_out = 0;
-
-      err = WebRtcAgc_VirtualMic(
-          my_handle,
-          audio->low_pass_split_data(i),
-          audio->high_pass_split_data(i),
-          static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
-          //capture_levels_[i],
-          analog_capture_level_,
-          &capture_level_out);
-
-      capture_levels_[i] = capture_level_out;
-
-      if (err != apm_->kNoError) {
-        return GetHandleError(my_handle);
-      }
-
-    }
-  }
-
-  return apm_->kNoError;
-}
-
-int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
-    return apm_->kStreamParameterNotSetError;
-  }
-
-  assert(audio->samples_per_split_channel() <= 160);
-  assert(audio->num_channels() == num_handles());
-
-  stream_is_saturated_ = false;
-  for (int i = 0; i < num_handles(); i++) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
-    WebRtc_Word32 capture_level_out = 0;
-    WebRtc_UWord8 saturation_warning = 0;
-
-    int err = WebRtcAgc_Process(
-        my_handle,
-        audio->low_pass_split_data(i),
-        audio->high_pass_split_data(i),
-        static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
-        audio->low_pass_split_data(i),
-        audio->high_pass_split_data(i),
-        capture_levels_[i],
-        &capture_level_out,
-        apm_->echo_cancellation()->stream_has_echo(),
-        &saturation_warning);
-
-    if (err != apm_->kNoError) {
-      return GetHandleError(my_handle);
-    }
-
-    capture_levels_[i] = capture_level_out;
-    if (saturation_warning == 1) {
-      stream_is_saturated_ = true;
-    }
-  }
-
-  if (mode_ == kAdaptiveAnalog) {
-    // Take the analog level to be the average across the handles.
-    analog_capture_level_ = 0;
-    for (int i = 0; i < num_handles(); i++) {
-      analog_capture_level_ += capture_levels_[i];
-    }
-
-    analog_capture_level_ /= num_handles();
-  }
-
-  was_analog_level_set_ = false;
-  return apm_->kNoError;
-}
-
-// TODO(ajm): ensure this is called under kAdaptiveAnalog.
-int GainControlImpl::set_stream_analog_level(int level) {
-  was_analog_level_set_ = true;
-  if (level < minimum_capture_level_ || level > maximum_capture_level_) {
-    return apm_->kBadParameterError;
-  }
-
-  if (mode_ == kAdaptiveAnalog) {
-    if (level != analog_capture_level_) {
-      // The analog level has been changed; update our internal levels.
-      capture_levels_.assign(num_handles(), level);
-    }
-  }
-  analog_capture_level_ = level;
-
-  return apm_->kNoError;
-}
-
-int GainControlImpl::stream_analog_level() {
-  // TODO(ajm): enable this assertion?
-  //assert(mode_ == kAdaptiveAnalog);
-
-  return analog_capture_level_;
-}
-
-int GainControlImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  return EnableComponent(enable);
-}
-
-bool GainControlImpl::is_enabled() const {
-  return is_component_enabled();
-}
-
-int GainControlImpl::set_mode(Mode mode) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (MapSetting(mode) == -1) {
-    return apm_->kBadParameterError;
-  }
-
-  mode_ = mode;
-  return Initialize();
-}
-
-GainControl::Mode GainControlImpl::mode() const {
-  return mode_;
-}
-
-int GainControlImpl::set_analog_level_limits(int minimum,
-                                             int maximum) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (minimum < 0) {
-    return apm_->kBadParameterError;
-  }
-
-  if (maximum > 65535) {
-    return apm_->kBadParameterError;
-  }
-
-  if (maximum < minimum) {
-    return apm_->kBadParameterError;
-  }
-
-  minimum_capture_level_ = minimum;
-  maximum_capture_level_ = maximum;
-
-  return Initialize();
-}
-
-int GainControlImpl::analog_level_minimum() const {
-  return minimum_capture_level_;
-}
-
-int GainControlImpl::analog_level_maximum() const {
-  return maximum_capture_level_;
-}
-
-bool GainControlImpl::stream_is_saturated() const {
-  return stream_is_saturated_;
-}
-
-int GainControlImpl::set_target_level_dbfs(int level) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (level > 31 || level < 0) {
-    return apm_->kBadParameterError;
-  }
-
-  target_level_dbfs_ = level;
-  return Configure();
-}
-
-int GainControlImpl::target_level_dbfs() const {
-  return target_level_dbfs_;
-}
-
-int GainControlImpl::set_compression_gain_db(int gain) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (gain < 0 || gain > 90) {
-    return apm_->kBadParameterError;
-  }
-
-  compression_gain_db_ = gain;
-  return Configure();
-}
-
-int GainControlImpl::compression_gain_db() const {
-  return compression_gain_db_;
-}
-
-int GainControlImpl::enable_limiter(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  limiter_enabled_ = enable;
-  return Configure();
-}
-
-bool GainControlImpl::is_limiter_enabled() const {
-  return limiter_enabled_;
-}
-
-int GainControlImpl::Initialize() {
-  int err = ProcessingComponent::Initialize();
-  if (err != apm_->kNoError || !is_component_enabled()) {
-    return err;
-  }
-
-  analog_capture_level_ =
-      (maximum_capture_level_ - minimum_capture_level_) >> 1;
-  capture_levels_.assign(num_handles(), analog_capture_level_);
-  was_analog_level_set_ = false;
-
-  return apm_->kNoError;
-}
-
-int GainControlImpl::get_version(char* version, int version_len_bytes) const {
-  if (WebRtcAgc_Version(version, version_len_bytes) != 0) {
-      return apm_->kBadParameterError;
-  }
-
-  return apm_->kNoError;
-}
-
-void* GainControlImpl::CreateHandle() const {
-  Handle* handle = NULL;
-  if (WebRtcAgc_Create(&handle) != apm_->kNoError) {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
-}
-
-int GainControlImpl::DestroyHandle(void* handle) const {
-  return WebRtcAgc_Free(static_cast<Handle*>(handle));
-}
-
-int GainControlImpl::InitializeHandle(void* handle) const {
-  return WebRtcAgc_Init(static_cast<Handle*>(handle),
-                          minimum_capture_level_,
-                          maximum_capture_level_,
-                          MapSetting(mode_),
-                          apm_->sample_rate_hz());
-}
-
-int GainControlImpl::ConfigureHandle(void* handle) const {
-  WebRtcAgc_config_t config;
-  // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
-  //            change the interface.
-  //assert(target_level_dbfs_ <= 0);
-  //config.targetLevelDbfs = static_cast<WebRtc_Word16>(-target_level_dbfs_);
-  config.targetLevelDbfs = static_cast<WebRtc_Word16>(target_level_dbfs_);
-  config.compressionGaindB =
-      static_cast<WebRtc_Word16>(compression_gain_db_);
-  config.limiterEnable = limiter_enabled_;
-
-  return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
-}
-
-int GainControlImpl::num_handles_required() const {
-  return apm_->num_output_channels();
-}
-
-int GainControlImpl::GetHandleError(void* handle) const {
-  // The AGC has no get_error() function.
-  // (Despite listing errors in its interface...)
-  assert(handle != NULL);
-  return apm_->kUnspecifiedError;
-}
-}  // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/gain_control_impl.h b/src/modules/audio_processing/main/source/gain_control_impl.h
deleted file mode 100644
index a11d606..0000000
--- a/src/modules/audio_processing/main/source/gain_control_impl.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
-
-#include <vector>
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class GainControlImpl : public GainControl,
-                        public ProcessingComponent {
- public:
-  explicit GainControlImpl(const AudioProcessingImpl* apm);
-  virtual ~GainControlImpl();
-
-  int ProcessRenderAudio(AudioBuffer* audio);
-  int AnalyzeCaptureAudio(AudioBuffer* audio);
-  int ProcessCaptureAudio(AudioBuffer* audio);
-
-  // ProcessingComponent implementation.
-  virtual int Initialize();
-  virtual int get_version(char* version, int version_len_bytes) const;
-
-  // GainControl implementation.
-  virtual bool is_enabled() const;
-
- private:
-  // GainControl implementation.
-  virtual int Enable(bool enable);
-  virtual int set_stream_analog_level(int level);
-  virtual int stream_analog_level();
-  virtual int set_mode(Mode mode);
-  virtual Mode mode() const;
-  virtual int set_target_level_dbfs(int level);
-  virtual int target_level_dbfs() const;
-  virtual int set_compression_gain_db(int gain);
-  virtual int compression_gain_db() const;
-  virtual int enable_limiter(bool enable);
-  virtual bool is_limiter_enabled() const;
-  virtual int set_analog_level_limits(int minimum, int maximum);
-  virtual int analog_level_minimum() const;
-  virtual int analog_level_maximum() const;
-  virtual bool stream_is_saturated() const;
-
-  // ProcessingComponent implementation.
-  virtual void* CreateHandle() const;
-  virtual int InitializeHandle(void* handle) const;
-  virtual int ConfigureHandle(void* handle) const;
-  virtual int DestroyHandle(void* handle) const;
-  virtual int num_handles_required() const;
-  virtual int GetHandleError(void* handle) const;
-
-  const AudioProcessingImpl* apm_;
-  Mode mode_;
-  int minimum_capture_level_;
-  int maximum_capture_level_;
-  bool limiter_enabled_;
-  int target_level_dbfs_;
-  int compression_gain_db_;
-  std::vector<int> capture_levels_;
-  int analog_capture_level_;
-  bool was_analog_level_set_;
-  bool stream_is_saturated_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_GAIN_CONTROL_IMPL_H_
diff --git a/src/modules/audio_processing/main/source/high_pass_filter_impl.cc b/src/modules/audio_processing/main/source/high_pass_filter_impl.cc
deleted file mode 100644
index fa6d5d5..0000000
--- a/src/modules/audio_processing/main/source/high_pass_filter_impl.cc
+++ /dev/null
@@ -1,180 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "high_pass_filter_impl.h"
-
-#include <cassert>
-
-#include "critical_section_wrapper.h"
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-namespace {
-const WebRtc_Word16 kFilterCoefficients8kHz[5] =
-    {3798, -7596, 3798, 7807, -3733};
-
-const WebRtc_Word16 kFilterCoefficients[5] =
-    {4012, -8024, 4012, 8002, -3913};
-
-struct FilterState {
-  WebRtc_Word16 y[4];
-  WebRtc_Word16 x[2];
-  const WebRtc_Word16* ba;
-};
-
-int InitializeFilter(FilterState* hpf, int sample_rate_hz) {
-  assert(hpf != NULL);
-
-  if (sample_rate_hz == AudioProcessingImpl::kSampleRate8kHz) {
-    hpf->ba = kFilterCoefficients8kHz;
-  } else {
-    hpf->ba = kFilterCoefficients;
-  }
-
-  WebRtcSpl_MemSetW16(hpf->x, 0, 2);
-  WebRtcSpl_MemSetW16(hpf->y, 0, 4);
-
-  return AudioProcessing::kNoError;
-}
-
-int Filter(FilterState* hpf, WebRtc_Word16* data, int length) {
-  assert(hpf != NULL);
-
-  WebRtc_Word32 tmp_int32 = 0;
-  WebRtc_Word16* y = hpf->y;
-  WebRtc_Word16* x = hpf->x;
-  const WebRtc_Word16* ba = hpf->ba;
-
-  for (int i = 0; i < length; i++) {
-    //  y[i] = b[0] * x[i] + b[1] * x[i-1] + b[2] * x[i-2]
-    //         + -a[1] * y[i-1] + -a[2] * y[i-2];
-
-    tmp_int32 =
-        WEBRTC_SPL_MUL_16_16(y[1], ba[3]); // -a[1] * y[i-1] (low part)
-    tmp_int32 +=
-        WEBRTC_SPL_MUL_16_16(y[3], ba[4]); // -a[2] * y[i-2] (low part)
-    tmp_int32 = (tmp_int32 >> 15);
-    tmp_int32 +=
-        WEBRTC_SPL_MUL_16_16(y[0], ba[3]); // -a[1] * y[i-1] (high part)
-    tmp_int32 +=
-        WEBRTC_SPL_MUL_16_16(y[2], ba[4]); // -a[2] * y[i-2] (high part)
-    tmp_int32 = (tmp_int32 << 1);
-
-    tmp_int32 += WEBRTC_SPL_MUL_16_16(data[i], ba[0]); // b[0]*x[0]
-    tmp_int32 += WEBRTC_SPL_MUL_16_16(x[0], ba[1]);    // b[1]*x[i-1]
-    tmp_int32 += WEBRTC_SPL_MUL_16_16(x[1], ba[2]);    // b[2]*x[i-2]
-
-    // Update state (input part)
-    x[1] = x[0];
-    x[0] = data[i];
-
-    // Update state (filtered part)
-    y[2] = y[0];
-    y[3] = y[1];
-    y[0] = static_cast<WebRtc_Word16>(tmp_int32 >> 13);
-    y[1] = static_cast<WebRtc_Word16>((tmp_int32 -
-        WEBRTC_SPL_LSHIFT_W32(static_cast<WebRtc_Word32>(y[0]), 13)) << 2);
-
-    // Rounding in Q12, i.e. add 2^11
-    tmp_int32 += 2048;
-
-    // Saturate (to 2^27) so that the HP filtered signal does not overflow
-    tmp_int32 = WEBRTC_SPL_SAT(static_cast<WebRtc_Word32>(134217727),
-                               tmp_int32,
-                               static_cast<WebRtc_Word32>(-134217728));
-
-    // Convert back to Q0 and use rounding
-    data[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp_int32, 12);
-
-  }
-
-  return AudioProcessing::kNoError;
-}
-}  // namespace
-
-typedef FilterState Handle;
-
-HighPassFilterImpl::HighPassFilterImpl(const AudioProcessingImpl* apm)
-  : ProcessingComponent(apm),
-    apm_(apm) {}
-
-HighPassFilterImpl::~HighPassFilterImpl() {}
-
-int HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  int err = apm_->kNoError;
-
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  assert(audio->samples_per_split_channel() <= 160);
-
-  for (int i = 0; i < num_handles(); i++) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
-    err = Filter(my_handle,
-                 audio->low_pass_split_data(i),
-                 audio->samples_per_split_channel());
-
-    if (err != apm_->kNoError) {
-      return GetHandleError(my_handle);
-    }
-  }
-
-  return apm_->kNoError;
-}
-
-int HighPassFilterImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  return EnableComponent(enable);
-}
-
-bool HighPassFilterImpl::is_enabled() const {
-  return is_component_enabled();
-}
-
-int HighPassFilterImpl::get_version(char* version,
-                                    int version_len_bytes) const {
-  // An empty string is used to indicate no version information.
-  memset(version, 0, version_len_bytes);
-  return apm_->kNoError;
-}
-
-void* HighPassFilterImpl::CreateHandle() const {
-  return new FilterState;
-}
-
-int HighPassFilterImpl::DestroyHandle(void* handle) const {
-  delete static_cast<Handle*>(handle);
-  return apm_->kNoError;
-}
-
-int HighPassFilterImpl::InitializeHandle(void* handle) const {
-  return InitializeFilter(static_cast<Handle*>(handle),
-                          apm_->sample_rate_hz());
-}
-
-int HighPassFilterImpl::ConfigureHandle(void* /*handle*/) const {
-  return apm_->kNoError; // Not configurable.
-}
-
-int HighPassFilterImpl::num_handles_required() const {
-  return apm_->num_output_channels();
-}
-
-int HighPassFilterImpl::GetHandleError(void* handle) const {
-  // The component has no detailed errors.
-  assert(handle != NULL);
-  return apm_->kUnspecifiedError;
-}
-}  // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/high_pass_filter_impl.h b/src/modules/audio_processing/main/source/high_pass_filter_impl.h
deleted file mode 100644
index 4c23754..0000000
--- a/src/modules/audio_processing/main/source/high_pass_filter_impl.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_HIGH_PASS_FILTER_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_HIGH_PASS_FILTER_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class HighPassFilterImpl : public HighPassFilter,
-                           public ProcessingComponent {
- public:
-  explicit HighPassFilterImpl(const AudioProcessingImpl* apm);
-  virtual ~HighPassFilterImpl();
-
-  int ProcessCaptureAudio(AudioBuffer* audio);
-
-  // HighPassFilter implementation.
-  virtual bool is_enabled() const;
-
-  // ProcessingComponent implementation.
-  virtual int get_version(char* version, int version_len_bytes) const;
-
- private:
-  // HighPassFilter implementation.
-  virtual int Enable(bool enable);
-
-  // ProcessingComponent implementation.
-  virtual void* CreateHandle() const;
-  virtual int InitializeHandle(void* handle) const;
-  virtual int ConfigureHandle(void* handle) const;
-  virtual int DestroyHandle(void* handle) const;
-  virtual int num_handles_required() const;
-  virtual int GetHandleError(void* handle) const;
-
-  const AudioProcessingImpl* apm_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_HIGH_PASS_FILTER_IMPL_H_
diff --git a/src/modules/audio_processing/main/source/level_estimator_impl.cc b/src/modules/audio_processing/main/source/level_estimator_impl.cc
deleted file mode 100644
index 799a962..0000000
--- a/src/modules/audio_processing/main/source/level_estimator_impl.cc
+++ /dev/null
@@ -1,182 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "level_estimator_impl.h"
-
-#include <cassert>
-#include <cstring>
-
-#include "critical_section_wrapper.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-// TODO(ajm): implement the underlying level estimator component.
-
-namespace webrtc {
-
-typedef void Handle;
-
-namespace {
-/*int EstimateLevel(AudioBuffer* audio, Handle* my_handle) {
-  assert(audio->samples_per_split_channel() <= 160);
-
-  WebRtc_Word16* mixed_data = audio->low_pass_split_data(0);
-  if (audio->num_channels() > 1) {
-    audio->CopyAndMixLowPass(1);
-    mixed_data = audio->mixed_low_pass_data(0);
-  }
-
-  int err = UpdateLvlEst(my_handle,
-                         mixed_data,
-                         audio->samples_per_split_channel());
-  if (err != AudioProcessing::kNoError) {
-    return GetHandleError(my_handle);
-  }
-
-  return AudioProcessing::kNoError;
-}
-
-int GetMetricsLocal(Handle* my_handle, LevelEstimator::Metrics* metrics) {
-  level_t levels;
-  memset(&levels, 0, sizeof(levels));
-
-  int err = ExportLevels(my_handle, &levels, 2);
-  if (err != AudioProcessing::kNoError) {
-    return err;
-  }
-  metrics->signal.instant = levels.instant;
-  metrics->signal.average = levels.average;
-  metrics->signal.maximum = levels.max;
-  metrics->signal.minimum = levels.min;
-
-  err = ExportLevels(my_handle, &levels, 1);
-  if (err != AudioProcessing::kNoError) {
-    return err;
-  }
-  metrics->speech.instant = levels.instant;
-  metrics->speech.average = levels.average;
-  metrics->speech.maximum = levels.max;
-  metrics->speech.minimum = levels.min;
-
-  err = ExportLevels(my_handle, &levels, 0);
-  if (err != AudioProcessing::kNoError) {
-    return err;
-  }
-  metrics->noise.instant = levels.instant;
-  metrics->noise.average = levels.average;
-  metrics->noise.maximum = levels.max;
-  metrics->noise.minimum = levels.min;
-
-  return AudioProcessing::kNoError;
-}*/
-}  // namespace
-
-LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessingImpl* apm)
-  : ProcessingComponent(apm),
-    apm_(apm) {}
-
-LevelEstimatorImpl::~LevelEstimatorImpl() {}
-
-int LevelEstimatorImpl::AnalyzeReverseStream(AudioBuffer* /*audio*/) {
-  return apm_->kUnsupportedComponentError;
-  /*if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  return EstimateLevel(audio, static_cast<Handle*>(handle(1)));*/
-}
-
-int LevelEstimatorImpl::ProcessCaptureAudio(AudioBuffer* /*audio*/) {
-  return apm_->kUnsupportedComponentError;
-  /*if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  return EstimateLevel(audio, static_cast<Handle*>(handle(0)));*/
-}
-
-int LevelEstimatorImpl::Enable(bool /*enable*/) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  return apm_->kUnsupportedComponentError;
-  //return EnableComponent(enable);
-}
-
-bool LevelEstimatorImpl::is_enabled() const {
-  return is_component_enabled();
-}
-
-int LevelEstimatorImpl::GetMetrics(LevelEstimator::Metrics* /*metrics*/,
-                                   LevelEstimator::Metrics* /*reverse_metrics*/) {
-  return apm_->kUnsupportedComponentError;
-  /*if (!is_component_enabled()) {
-    return apm_->kNotEnabledError;
-  }
-
-  int err = GetMetricsLocal(static_cast<Handle*>(handle(0)), metrics);
-  if (err != apm_->kNoError) {
-    return err;
-  }
-
-  err = GetMetricsLocal(static_cast<Handle*>(handle(1)), reverse_metrics);
-  if (err != apm_->kNoError) {
-    return err;
-  }
-
-  return apm_->kNoError;*/
-}
-
-int LevelEstimatorImpl::get_version(char* version,
-                                    int version_len_bytes) const {
-  // An empty string is used to indicate no version information.
-  memset(version, 0, version_len_bytes);
-  return apm_->kNoError;
-}
-
-void* LevelEstimatorImpl::CreateHandle() const {
-  Handle* handle = NULL;
-  /*if (CreateLvlEst(&handle) != apm_->kNoError) {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }*/
-
-  return handle;
-}
-
-int LevelEstimatorImpl::DestroyHandle(void* /*handle*/) const {
-  return apm_->kUnsupportedComponentError;
-  //return FreeLvlEst(static_cast<Handle*>(handle));
-}
-
-int LevelEstimatorImpl::InitializeHandle(void* /*handle*/) const {
-  return apm_->kUnsupportedComponentError;
-  /*const double kIntervalSeconds = 1.5;
-  return InitLvlEst(static_cast<Handle*>(handle),
-                    apm_->sample_rate_hz(),
-                    kIntervalSeconds);*/
-}
-
-int LevelEstimatorImpl::ConfigureHandle(void* /*handle*/) const {
-  return apm_->kUnsupportedComponentError;
-  //return apm_->kNoError;
-}
-
-int LevelEstimatorImpl::num_handles_required() const {
-  return apm_->kUnsupportedComponentError;
-  //return 2;
-}
-
-int LevelEstimatorImpl::GetHandleError(void* handle) const {
-  // The component has no detailed errors.
-  assert(handle != NULL);
-  return apm_->kUnspecifiedError;
-}
-}  // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/level_estimator_impl.h b/src/modules/audio_processing/main/source/level_estimator_impl.h
deleted file mode 100644
index 1515722..0000000
--- a/src/modules/audio_processing/main/source/level_estimator_impl.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class LevelEstimatorImpl : public LevelEstimator,
-                           public ProcessingComponent {
- public:
-  explicit LevelEstimatorImpl(const AudioProcessingImpl* apm);
-  virtual ~LevelEstimatorImpl();
-
-  int AnalyzeReverseStream(AudioBuffer* audio);
-  int ProcessCaptureAudio(AudioBuffer* audio);
-
-  // LevelEstimator implementation.
-  virtual bool is_enabled() const;
-
-  // ProcessingComponent implementation.
-  virtual int get_version(char* version, int version_len_bytes) const;
-
- private:
-  // LevelEstimator implementation.
-  virtual int Enable(bool enable);
-  virtual int GetMetrics(Metrics* metrics, Metrics* reverse_metrics);
-
-  // ProcessingComponent implementation.
-  virtual void* CreateHandle() const;
-  virtual int InitializeHandle(void* handle) const;
-  virtual int ConfigureHandle(void* handle) const;
-  virtual int DestroyHandle(void* handle) const;
-  virtual int num_handles_required() const;
-  virtual int GetHandleError(void* handle) const;
-
-  const AudioProcessingImpl* apm_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_LEVEL_ESTIMATOR_IMPL_H_
diff --git a/src/modules/audio_processing/main/source/noise_suppression_impl.cc b/src/modules/audio_processing/main/source/noise_suppression_impl.cc
deleted file mode 100644
index f899f35..0000000
--- a/src/modules/audio_processing/main/source/noise_suppression_impl.cc
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "noise_suppression_impl.h"
-
-#include <cassert>
-
-#include "critical_section_wrapper.h"
-#if defined(WEBRTC_NS_FLOAT)
-#include "noise_suppression.h"
-#elif defined(WEBRTC_NS_FIXED)
-#include "noise_suppression_x.h"
-#endif
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-
-#if defined(WEBRTC_NS_FLOAT)
-typedef NsHandle Handle;
-#elif defined(WEBRTC_NS_FIXED)
-typedef NsxHandle Handle;
-#endif
-
-namespace {
-int MapSetting(NoiseSuppression::Level level) {
-  switch (level) {
-    case NoiseSuppression::kLow:
-      return 0;
-    case NoiseSuppression::kModerate:
-      return 1;
-    case NoiseSuppression::kHigh:
-      return 2;
-    case NoiseSuppression::kVeryHigh:
-      return 3;
-    default:
-      return -1;
-  }
-}
-}  // namespace
-
-NoiseSuppressionImpl::NoiseSuppressionImpl(const AudioProcessingImpl* apm)
-  : ProcessingComponent(apm),
-    apm_(apm),
-    level_(kModerate) {}
-
-NoiseSuppressionImpl::~NoiseSuppressionImpl() {}
-
-int NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  int err = apm_->kNoError;
-
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-  assert(audio->samples_per_split_channel() <= 160);
-  assert(audio->num_channels() == num_handles());
-
-  for (int i = 0; i < num_handles(); i++) {
-    Handle* my_handle = static_cast<Handle*>(handle(i));
-#if defined(WEBRTC_NS_FLOAT)
-    err = WebRtcNs_Process(static_cast<Handle*>(handle(i)),
-                           audio->low_pass_split_data(i),
-                           audio->high_pass_split_data(i),
-                           audio->low_pass_split_data(i),
-                           audio->high_pass_split_data(i));
-#elif defined(WEBRTC_NS_FIXED)
-    err = WebRtcNsx_Process(static_cast<Handle*>(handle(i)),
-                            audio->low_pass_split_data(i),
-                            audio->high_pass_split_data(i),
-                            audio->low_pass_split_data(i),
-                            audio->high_pass_split_data(i));
-#endif
-
-    if (err != apm_->kNoError) {
-      return GetHandleError(my_handle);
-    }
-  }
-
-  return apm_->kNoError;
-}
-
-int NoiseSuppressionImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  return EnableComponent(enable);
-}
-
-bool NoiseSuppressionImpl::is_enabled() const {
-  return is_component_enabled();
-}
-
-int NoiseSuppressionImpl::set_level(Level level) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (MapSetting(level) == -1) {
-    return apm_->kBadParameterError;
-  }
-
-  level_ = level;
-  return Configure();
-}
-
-NoiseSuppression::Level NoiseSuppressionImpl::level() const {
-  return level_;
-}
-
-int NoiseSuppressionImpl::get_version(char* version,
-                                      int version_len_bytes) const {
-#if defined(WEBRTC_NS_FLOAT)
-  if (WebRtcNs_get_version(version, version_len_bytes) != 0)
-#elif defined(WEBRTC_NS_FIXED)
-  if (WebRtcNsx_get_version(version, version_len_bytes) != 0)
-#endif
-  {
-      return apm_->kBadParameterError;
-  }
-
-  return apm_->kNoError;
-}
-
-void* NoiseSuppressionImpl::CreateHandle() const {
-  Handle* handle = NULL;
-#if defined(WEBRTC_NS_FLOAT)
-  if (WebRtcNs_Create(&handle) != apm_->kNoError)
-#elif defined(WEBRTC_NS_FIXED)
-  if (WebRtcNsx_Create(&handle) != apm_->kNoError)
-#endif
-  {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
-}
-
-int NoiseSuppressionImpl::DestroyHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
-  return WebRtcNs_Free(static_cast<Handle*>(handle));
-#elif defined(WEBRTC_NS_FIXED)
-  return WebRtcNsx_Free(static_cast<Handle*>(handle));
-#endif
-}
-
-int NoiseSuppressionImpl::InitializeHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
-  return WebRtcNs_Init(static_cast<Handle*>(handle), apm_->sample_rate_hz());
-#elif defined(WEBRTC_NS_FIXED)
-  return WebRtcNsx_Init(static_cast<Handle*>(handle), apm_->sample_rate_hz());
-#endif
-}
-
-int NoiseSuppressionImpl::ConfigureHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
-  return WebRtcNs_set_policy(static_cast<Handle*>(handle),
-                             MapSetting(level_));
-#elif defined(WEBRTC_NS_FIXED)
-  return WebRtcNsx_set_policy(static_cast<Handle*>(handle),
-                              MapSetting(level_));
-#endif
-}
-
-int NoiseSuppressionImpl::num_handles_required() const {
-  return apm_->num_output_channels();
-}
-
-int NoiseSuppressionImpl::GetHandleError(void* handle) const {
-  // The NS has no get_error() function.
-  assert(handle != NULL);
-  return apm_->kUnspecifiedError;
-}
-}  // namespace webrtc
-
diff --git a/src/modules/audio_processing/main/source/noise_suppression_impl.h b/src/modules/audio_processing/main/source/noise_suppression_impl.h
deleted file mode 100644
index c9ff9b3..0000000
--- a/src/modules/audio_processing/main/source/noise_suppression_impl.h
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_NOISE_SUPPRESSION_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_NOISE_SUPPRESSION_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class NoiseSuppressionImpl : public NoiseSuppression,
-                             public ProcessingComponent {
- public:
-  explicit NoiseSuppressionImpl(const AudioProcessingImpl* apm);
-  virtual ~NoiseSuppressionImpl();
-
-  int ProcessCaptureAudio(AudioBuffer* audio);
-
-  // NoiseSuppression implementation.
-  virtual bool is_enabled() const;
-
-  // ProcessingComponent implementation.
-  virtual int get_version(char* version, int version_len_bytes) const;
-
- private:
-  // NoiseSuppression implementation.
-  virtual int Enable(bool enable);
-  virtual int set_level(Level level);
-  virtual Level level() const;
-
-  // ProcessingComponent implementation.
-  virtual void* CreateHandle() const;
-  virtual int InitializeHandle(void* handle) const;
-  virtual int ConfigureHandle(void* handle) const;
-  virtual int DestroyHandle(void* handle) const;
-  virtual int num_handles_required() const;
-  virtual int GetHandleError(void* handle) const;
-
-  const AudioProcessingImpl* apm_;
-  Level level_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_NOISE_SUPPRESSION_IMPL_H_
diff --git a/src/modules/audio_processing/main/source/processing_component.cc b/src/modules/audio_processing/main/source/processing_component.cc
deleted file mode 100644
index 9ac1257..0000000
--- a/src/modules/audio_processing/main/source/processing_component.cc
+++ /dev/null
@@ -1,112 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "processing_component.h"
-
-#include <cassert>
-
-#include "audio_processing_impl.h"
-
-namespace webrtc {
-
-ProcessingComponent::ProcessingComponent(const AudioProcessingImpl* apm)
-  : apm_(apm),
-    initialized_(false),
-    enabled_(false),
-    num_handles_(0) {}
-
-ProcessingComponent::~ProcessingComponent() {
-  assert(initialized_ == false);
-}
-
-int ProcessingComponent::Destroy() {
-  while (!handles_.empty()) {
-    DestroyHandle(handles_.back());
-    handles_.pop_back();
-  }
-  initialized_ = false;
-
-  return apm_->kNoError;
-}
-
-int ProcessingComponent::EnableComponent(bool enable) {
-  if (enable && !enabled_) {
-    enabled_ = enable; // Must be set before Initialize() is called.
-
-    int err = Initialize();
-    if (err != apm_->kNoError) {
-      enabled_ = false;
-      return err;
-    }
-  } else {
-    enabled_ = enable;
-  }
-
-  return apm_->kNoError;
-}
-
-bool ProcessingComponent::is_component_enabled() const {
-  return enabled_;
-}
-
-void* ProcessingComponent::handle(int index) const {
-  assert(index < num_handles_);
-  return handles_[index];
-}
-
-int ProcessingComponent::num_handles() const {
-  return num_handles_;
-}
-
-int ProcessingComponent::Initialize() {
-  if (!enabled_) {
-    return apm_->kNoError;
-  }
-
-  num_handles_ = num_handles_required();
-  if (num_handles_ > static_cast<int>(handles_.size())) {
-    handles_.resize(num_handles_, NULL);
-  }
-
-  assert(static_cast<int>(handles_.size()) >= num_handles_);
-  for (int i = 0; i < num_handles_; i++) {
-    if (handles_[i] == NULL) {
-      handles_[i] = CreateHandle();
-      if (handles_[i] == NULL) {
-        return apm_->kCreationFailedError;
-      }
-    }
-
-    int err = InitializeHandle(handles_[i]);
-    if (err != apm_->kNoError) {
-      return GetHandleError(handles_[i]);
-    }
-  }
-
-  initialized_ = true;
-  return Configure();
-}
-
-int ProcessingComponent::Configure() {
-  if (!initialized_) {
-    return apm_->kNoError;
-  }
-
-  assert(static_cast<int>(handles_.size()) >= num_handles_);
-  for (int i = 0; i < num_handles_; i++) {
-    int err = ConfigureHandle(handles_[i]);
-    if (err != apm_->kNoError) {
-      return GetHandleError(handles_[i]);
-    }
-  }
-
-  return apm_->kNoError;
-}
-}  // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/processing_component.h b/src/modules/audio_processing/main/source/processing_component.h
deleted file mode 100644
index 3d8a02b..0000000
--- a/src/modules/audio_processing/main/source/processing_component.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_PROCESSING_COMPONENT_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_PROCESSING_COMPONENT_H_
-
-#include <vector>
-
-#include "audio_processing.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-
-/*template <class T>
-class ComponentHandle {
-  public:
-    ComponentHandle();
-    virtual ~ComponentHandle();
-
-    virtual int Create() = 0;
-    virtual T* ptr() const = 0;
-};*/
-
-class ProcessingComponent {
- public:
-  explicit ProcessingComponent(const AudioProcessingImpl* apm);
-  virtual ~ProcessingComponent();
-
-  virtual int Initialize();
-  virtual int Destroy();
-  virtual int get_version(char* version, int version_len_bytes) const = 0;
-
- protected:
-  virtual int Configure();
-  int EnableComponent(bool enable);
-  bool is_component_enabled() const;
-  void* handle(int index) const;
-  int num_handles() const;
-
- private:
-  virtual void* CreateHandle() const = 0;
-  virtual int InitializeHandle(void* handle) const = 0;
-  virtual int ConfigureHandle(void* handle) const = 0;
-  virtual int DestroyHandle(void* handle) const = 0;
-  virtual int num_handles_required() const = 0;
-  virtual int GetHandleError(void* handle) const = 0;
-
-  const AudioProcessingImpl* apm_;
-  std::vector<void*> handles_;
-  bool initialized_;
-  bool enabled_;
-  int num_handles_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_PROCESSING_COMPONENT_H__
diff --git a/src/modules/audio_processing/main/source/splitting_filter.cc b/src/modules/audio_processing/main/source/splitting_filter.cc
deleted file mode 100644
index 1526141..0000000
--- a/src/modules/audio_processing/main/source/splitting_filter.cc
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "splitting_filter.h"
-#include "signal_processing_library.h"
-
-namespace webrtc {
-
-void SplittingFilterAnalysis(const WebRtc_Word16* in_data,
-                             WebRtc_Word16* low_band,
-                             WebRtc_Word16* high_band,
-                             WebRtc_Word32* filter_state1,
-                             WebRtc_Word32* filter_state2)
-{
-    WebRtcSpl_AnalysisQMF(in_data, low_band, high_band, filter_state1, filter_state2);
-}
-
-void SplittingFilterSynthesis(const WebRtc_Word16* low_band,
-                              const WebRtc_Word16* high_band,
-                              WebRtc_Word16* out_data,
-                              WebRtc_Word32* filt_state1,
-                              WebRtc_Word32* filt_state2)
-{
-    WebRtcSpl_SynthesisQMF(low_band, high_band, out_data, filt_state1, filt_state2);
-}
-}  // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/splitting_filter.h b/src/modules/audio_processing/main/source/splitting_filter.h
deleted file mode 100644
index 661bfb2..0000000
--- a/src/modules/audio_processing/main/source/splitting_filter.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_
-
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-namespace webrtc {
-/*
- * SplittingFilterbank_analysisQMF(...)
- *
- * Splits a super-wb signal into two subbands: 0-8 kHz and 8-16 kHz.
- *
- * Input:
- *    - in_data  : super-wb audio signal
- *
- * Input & Output:
- *    - filt_state1: Filter state for first all-pass filter
- *    - filt_state2: Filter state for second all-pass filter
- *
- * Output:
- *    - low_band : The signal from the 0-4 kHz band
- *    - high_band  : The signal from the 4-8 kHz band
- */
-void SplittingFilterAnalysis(const WebRtc_Word16* in_data,
-                             WebRtc_Word16* low_band,
-                             WebRtc_Word16* high_band,
-                             WebRtc_Word32* filt_state1,
-                             WebRtc_Word32* filt_state2);
-
-/*
- * SplittingFilterbank_synthesisQMF(...)
- *
- * Combines the two subbands (0-8 and 8-16 kHz) into a super-wb signal.
- *
- * Input:
- *    - low_band : The signal with the 0-8 kHz band
- *    - high_band  : The signal with the 8-16 kHz band
- *
- * Input & Output:
- *    - filt_state1: Filter state for first all-pass filter
- *    - filt_state2: Filter state for second all-pass filter
- *
- * Output:
- *    - out_data : super-wb speech signal
- */
-void SplittingFilterSynthesis(const WebRtc_Word16* low_band,
-                              const WebRtc_Word16* high_band,
-                              WebRtc_Word16* out_data,
-                              WebRtc_Word32* filt_state1,
-                              WebRtc_Word32* filt_state2);
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_
diff --git a/src/modules/audio_processing/main/source/voice_detection_impl.cc b/src/modules/audio_processing/main/source/voice_detection_impl.cc
deleted file mode 100644
index 3eb446e..0000000
--- a/src/modules/audio_processing/main/source/voice_detection_impl.cc
+++ /dev/null
@@ -1,202 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voice_detection_impl.h"
-
-#include <cassert>
-
-#include "critical_section_wrapper.h"
-#include "webrtc_vad.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-
-typedef VadInst Handle;
-
-namespace {
-WebRtc_Word16 MapSetting(VoiceDetection::Likelihood likelihood) {
-  switch (likelihood) {
-    case VoiceDetection::kVeryLowLikelihood:
-      return 3;
-      break;
-    case VoiceDetection::kLowLikelihood:
-      return 2;
-      break;
-    case VoiceDetection::kModerateLikelihood:
-      return 1;
-      break;
-    case VoiceDetection::kHighLikelihood:
-      return 0;
-      break;
-    default:
-      return -1;
-  }
-}
-}  // namespace
-
-
-VoiceDetectionImpl::VoiceDetectionImpl(const AudioProcessingImpl* apm)
-  : ProcessingComponent(apm),
-    apm_(apm),
-    stream_has_voice_(false),
-    using_external_vad_(false),
-    likelihood_(kLowLikelihood),
-    frame_size_ms_(10),
-    frame_size_samples_(0) {}
-
-VoiceDetectionImpl::~VoiceDetectionImpl() {}
-
-int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
-  if (!is_component_enabled()) {
-    return apm_->kNoError;
-  }
-
-  if (using_external_vad_) {
-    using_external_vad_ = false;
-    return apm_->kNoError;
-  }
-  assert(audio->samples_per_split_channel() <= 160);
-
-  WebRtc_Word16* mixed_data = audio->low_pass_split_data(0);
-  if (audio->num_channels() > 1) {
-    audio->CopyAndMixLowPass(1);
-    mixed_data = audio->mixed_low_pass_data(0);
-  }
-
-  // TODO(ajm): concatenate data in frame buffer here.
-
-  int vad_ret_val;
-  vad_ret_val = WebRtcVad_Process(static_cast<Handle*>(handle(0)),
-                      apm_->split_sample_rate_hz(),
-                      mixed_data,
-                      frame_size_samples_);
-
-  if (vad_ret_val == 0) {
-    stream_has_voice_ = false;
-  } else if (vad_ret_val == 1) {
-    stream_has_voice_ = true;
-  } else {
-    return apm_->kUnspecifiedError;
-  }
-
-  return apm_->kNoError;
-}
-
-int VoiceDetectionImpl::Enable(bool enable) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  return EnableComponent(enable);
-}
-
-bool VoiceDetectionImpl::is_enabled() const {
-  return is_component_enabled();
-}
-
-int VoiceDetectionImpl::set_stream_has_voice(bool has_voice) {
-  using_external_vad_ = true;
-  stream_has_voice_ = has_voice;
-  return apm_->kNoError;
-}
-
-bool VoiceDetectionImpl::stream_has_voice() const {
-  // TODO(ajm): enable this assertion?
-  //assert(using_external_vad_ || is_component_enabled());
-  return stream_has_voice_;
-}
-
-int VoiceDetectionImpl::set_likelihood(VoiceDetection::Likelihood likelihood) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  if (MapSetting(likelihood) == -1) {
-    return apm_->kBadParameterError;
-  }
-
-  likelihood_ = likelihood;
-  return Configure();
-}
-
-VoiceDetection::Likelihood VoiceDetectionImpl::likelihood() const {
-  return likelihood_;
-}
-
-int VoiceDetectionImpl::set_frame_size_ms(int size) {
-  CriticalSectionScoped crit_scoped(*apm_->crit());
-  assert(size == 10); // TODO(ajm): remove when supported.
-  if (size != 10 &&
-      size != 20 &&
-      size != 30) {
-    return apm_->kBadParameterError;
-  }
-
-  frame_size_ms_ = size;
-
-  return Initialize();
-}
-
-int VoiceDetectionImpl::frame_size_ms() const {
-  return frame_size_ms_;
-}
-
-int VoiceDetectionImpl::Initialize() {
-  int err = ProcessingComponent::Initialize();
-  if (err != apm_->kNoError || !is_component_enabled()) {
-    return err;
-  }
-
-  using_external_vad_ = false;
-  frame_size_samples_ = frame_size_ms_ * (apm_->split_sample_rate_hz() / 1000);
-  // TODO(ajm): intialize frame buffer here.
-
-  return apm_->kNoError;
-}
-
-int VoiceDetectionImpl::get_version(char* version,
-                                    int version_len_bytes) const {
-  if (WebRtcVad_get_version(version, version_len_bytes) != 0) {
-    return apm_->kBadParameterError;
-  }
-
-  return apm_->kNoError;
-}
-
-void* VoiceDetectionImpl::CreateHandle() const {
-  Handle* handle = NULL;
-  if (WebRtcVad_Create(&handle) != apm_->kNoError) {
-    handle = NULL;
-  } else {
-    assert(handle != NULL);
-  }
-
-  return handle;
-}
-
-int VoiceDetectionImpl::DestroyHandle(void* handle) const {
-  return WebRtcVad_Free(static_cast<Handle*>(handle));
-}
-
-int VoiceDetectionImpl::InitializeHandle(void* handle) const {
-  return WebRtcVad_Init(static_cast<Handle*>(handle));
-}
-
-int VoiceDetectionImpl::ConfigureHandle(void* handle) const {
-  return WebRtcVad_set_mode(static_cast<Handle*>(handle),
-                            MapSetting(likelihood_));
-}
-
-int VoiceDetectionImpl::num_handles_required() const {
-  return 1;
-}
-
-int VoiceDetectionImpl::GetHandleError(void* handle) const {
-  // The VAD has no get_error() function.
-  assert(handle != NULL);
-  return apm_->kUnspecifiedError;
-}
-}  // namespace webrtc
diff --git a/src/modules/audio_processing/main/source/voice_detection_impl.h b/src/modules/audio_processing/main/source/voice_detection_impl.h
deleted file mode 100644
index ef212d1..0000000
--- a/src/modules/audio_processing/main/source/voice_detection_impl.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class VoiceDetectionImpl : public VoiceDetection,
-                           public ProcessingComponent {
- public:
-  explicit VoiceDetectionImpl(const AudioProcessingImpl* apm);
-  virtual ~VoiceDetectionImpl();
-
-  int ProcessCaptureAudio(AudioBuffer* audio);
-
-  // VoiceDetection implementation.
-  virtual bool is_enabled() const;
-
-  // ProcessingComponent implementation.
-  virtual int Initialize();
-  virtual int get_version(char* version, int version_len_bytes) const;
-
- private:
-  // VoiceDetection implementation.
-  virtual int Enable(bool enable);
-  virtual int set_stream_has_voice(bool has_voice);
-  virtual bool stream_has_voice() const;
-  virtual int set_likelihood(Likelihood likelihood);
-  virtual Likelihood likelihood() const;
-  virtual int set_frame_size_ms(int size);
-  virtual int frame_size_ms() const;
-
-  // ProcessingComponent implementation.
-  virtual void* CreateHandle() const;
-  virtual int InitializeHandle(void* handle) const;
-  virtual int ConfigureHandle(void* handle) const;
-  virtual int DestroyHandle(void* handle) const;
-  virtual int num_handles_required() const;
-  virtual int GetHandleError(void* handle) const;
-
-  const AudioProcessingImpl* apm_;
-  bool stream_has_voice_;
-  bool using_external_vad_;
-  Likelihood likelihood_;
-  int frame_size_ms_;
-  int frame_size_samples_;
-};
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_VOICE_DETECTION_IMPL_H_
diff --git a/src/modules/audio_processing/main/test/android/apmtest/AndroidManifest.xml b/src/modules/audio_processing/main/test/android/apmtest/AndroidManifest.xml
deleted file mode 100644
index c6063b3..0000000
--- a/src/modules/audio_processing/main/test/android/apmtest/AndroidManifest.xml
+++ /dev/null
@@ -1,30 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<!-- BEGIN_INCLUDE(manifest) -->
-<manifest xmlns:android="http://schemas.android.com/apk/res/android"
-        package="com.example.native_activity"
-        android:versionCode="1"
-        android:versionName="1.0">
-
-    <!-- This is the platform API where NativeActivity was introduced. -->
-    <uses-sdk android:minSdkVersion="8" />
-
-    <!-- This .apk has no Java code itself, so set hasCode to false. -->
-    <application android:label="@string/app_name" android:hasCode="false" android:debuggable="true">
-
-        <!-- Our activity is the built-in NativeActivity framework class.
-             This will take care of integrating with our NDK code. -->
-        <activity android:name="android.app.NativeActivity"
-                android:label="@string/app_name"
-                android:configChanges="orientation|keyboardHidden">
-            <!-- Tell NativeActivity the name of or .so -->
-            <meta-data android:name="android.app.lib_name"
-                    android:value="apmtest-activity" />
-            <intent-filter>
-                <action android:name="android.intent.action.MAIN" />
-                <category android:name="android.intent.category.LAUNCHER" />
-            </intent-filter>
-        </activity>
-    </application>
-
-</manifest> 
-<!-- END_INCLUDE(manifest) -->
diff --git a/src/modules/audio_processing/main/test/android/apmtest/default.properties b/src/modules/audio_processing/main/test/android/apmtest/default.properties
deleted file mode 100644
index 9a2c9f6..0000000
--- a/src/modules/audio_processing/main/test/android/apmtest/default.properties
+++ /dev/null
@@ -1,11 +0,0 @@
-# This file is automatically generated by Android Tools.
-# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
-# 
-# This file must be checked in Version Control Systems.
-# 
-# To customize properties used by the Ant build system use,
-# "build.properties", and override values to adapt the script to your
-# project structure.
-
-# Project target.
-target=android-9
diff --git a/src/modules/audio_processing/main/test/android/apmtest/jni/Android.mk b/src/modules/audio_processing/main/test/android/apmtest/jni/Android.mk
deleted file mode 100644
index eaf3c9d..0000000
--- a/src/modules/audio_processing/main/test/android/apmtest/jni/Android.mk
+++ /dev/null
@@ -1,26 +0,0 @@
-# Copyright (C) 2010 The Android Open Source Project
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-#      http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-#
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE    := apmtest-activity
-LOCAL_SRC_FILES := main.c
-LOCAL_LDLIBS    := -llog -landroid -lEGL -lGLESv1_CM
-LOCAL_STATIC_LIBRARIES := android_native_app_glue
-
-include $(BUILD_SHARED_LIBRARY)
-
-$(call import-module,android/native_app_glue)
diff --git a/src/modules/audio_processing/main/test/android/apmtest/jni/Application.mk b/src/modules/audio_processing/main/test/android/apmtest/jni/Application.mk
deleted file mode 100644
index 22d188e..0000000
--- a/src/modules/audio_processing/main/test/android/apmtest/jni/Application.mk
+++ /dev/null
@@ -1 +0,0 @@
-APP_PLATFORM := android-9
diff --git a/src/modules/audio_processing/main/test/android/apmtest/jni/main.c b/src/modules/audio_processing/main/test/android/apmtest/jni/main.c
deleted file mode 100644
index 2e19635..0000000
--- a/src/modules/audio_processing/main/test/android/apmtest/jni/main.c
+++ /dev/null
@@ -1,307 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- *
- */
-
-//BEGIN_INCLUDE(all)
-#include <jni.h>
-#include <errno.h>
-
-#include <EGL/egl.h>
-#include <GLES/gl.h>
-
-#include <android/sensor.h>
-#include <android/log.h>
-#include <android_native_app_glue.h>
-
-#define LOGI(...) ((void)__android_log_print(ANDROID_LOG_INFO, "native-activity", __VA_ARGS__))
-#define LOGW(...) ((void)__android_log_print(ANDROID_LOG_WARN, "native-activity", __VA_ARGS__))
-
-/**
- * Our saved state data.
- */
-struct saved_state {
-    float angle;
-    int32_t x;
-    int32_t y;
-};
-
-/**
- * Shared state for our app.
- */
-struct engine {
-    struct android_app* app;
-
-    ASensorManager* sensorManager;
-    const ASensor* accelerometerSensor;
-    ASensorEventQueue* sensorEventQueue;
-
-    int animating;
-    EGLDisplay display;
-    EGLSurface surface;
-    EGLContext context;
-    int32_t width;
-    int32_t height;
-    struct saved_state state;
-};
-
-/**
- * Initialize an EGL context for the current display.
- */
-static int engine_init_display(struct engine* engine) {
-    // initialize OpenGL ES and EGL
-
-    /*
-     * Here specify the attributes of the desired configuration.
-     * Below, we select an EGLConfig with at least 8 bits per color
-     * component compatible with on-screen windows
-     */
-    const EGLint attribs[] = {
-            EGL_SURFACE_TYPE, EGL_WINDOW_BIT,
-            EGL_BLUE_SIZE, 8,
-            EGL_GREEN_SIZE, 8,
-            EGL_RED_SIZE, 8,
-            EGL_NONE
-    };
-    EGLint w, h, dummy, format;
-    EGLint numConfigs;
-    EGLConfig config;
-    EGLSurface surface;
-    EGLContext context;
-
-    EGLDisplay display = eglGetDisplay(EGL_DEFAULT_DISPLAY);
-
-    eglInitialize(display, 0, 0);
-
-    /* Here, the application chooses the configuration it desires. In this
-     * sample, we have a very simplified selection process, where we pick
-     * the first EGLConfig that matches our criteria */
-    eglChooseConfig(display, attribs, &config, 1, &numConfigs);
-
-    /* EGL_NATIVE_VISUAL_ID is an attribute of the EGLConfig that is
-     * guaranteed to be accepted by ANativeWindow_setBuffersGeometry().
-     * As soon as we picked a EGLConfig, we can safely reconfigure the
-     * ANativeWindow buffers to match, using EGL_NATIVE_VISUAL_ID. */
-    eglGetConfigAttrib(display, config, EGL_NATIVE_VISUAL_ID, &format);
-
-    ANativeWindow_setBuffersGeometry(engine->app->window, 0, 0, format);
-
-    surface = eglCreateWindowSurface(display, config, engine->app->window, NULL);
-    context = eglCreateContext(display, config, NULL, NULL);
-
-    if (eglMakeCurrent(display, surface, surface, context) == EGL_FALSE) {
-        LOGW("Unable to eglMakeCurrent");
-        return -1;
-    }
-
-    eglQuerySurface(display, surface, EGL_WIDTH, &w);
-    eglQuerySurface(display, surface, EGL_HEIGHT, &h);
-
-    engine->display = display;
-    engine->context = context;
-    engine->surface = surface;
-    engine->width = w;
-    engine->height = h;
-    engine->state.angle = 0;
-
-    // Initialize GL state.
-    glHint(GL_PERSPECTIVE_CORRECTION_HINT, GL_FASTEST);
-    glEnable(GL_CULL_FACE);
-    glShadeModel(GL_SMOOTH);
-    glDisable(GL_DEPTH_TEST);
-
-    return 0;
-}
-
-/**
- * Just the current frame in the display.
- */
-static void engine_draw_frame(struct engine* engine) {
-    if (engine->display == NULL) {
-        // No display.
-        return;
-    }
-
-    // Just fill the screen with a color.
-    glClearColor(((float)engine->state.x)/engine->width, engine->state.angle,
-            ((float)engine->state.y)/engine->height, 1);
-    glClear(GL_COLOR_BUFFER_BIT);
-
-    eglSwapBuffers(engine->display, engine->surface);
-}
-
-/**
- * Tear down the EGL context currently associated with the display.
- */
-static void engine_term_display(struct engine* engine) {
-    if (engine->display != EGL_NO_DISPLAY) {
-        eglMakeCurrent(engine->display, EGL_NO_SURFACE, EGL_NO_SURFACE, EGL_NO_CONTEXT);
-        if (engine->context != EGL_NO_CONTEXT) {
-            eglDestroyContext(engine->display, engine->context);
-        }
-        if (engine->surface != EGL_NO_SURFACE) {
-            eglDestroySurface(engine->display, engine->surface);
-        }
-        eglTerminate(engine->display);
-    }
-    engine->animating = 0;
-    engine->display = EGL_NO_DISPLAY;
-    engine->context = EGL_NO_CONTEXT;
-    engine->surface = EGL_NO_SURFACE;
-}
-
-/**
- * Process the next input event.
- */
-static int32_t engine_handle_input(struct android_app* app, AInputEvent* event) {
-    struct engine* engine = (struct engine*)app->userData;
-    if (AInputEvent_getType(event) == AINPUT_EVENT_TYPE_MOTION) {
-        engine->animating = 1;
-        engine->state.x = AMotionEvent_getX(event, 0);
-        engine->state.y = AMotionEvent_getY(event, 0);
-        return 1;
-    }
-    return 0;
-}
-
-/**
- * Process the next main command.
- */
-static void engine_handle_cmd(struct android_app* app, int32_t cmd) {
-    struct engine* engine = (struct engine*)app->userData;
-    switch (cmd) {
-        case APP_CMD_SAVE_STATE:
-            // The system has asked us to save our current state.  Do so.
-            engine->app->savedState = malloc(sizeof(struct saved_state));
-            *((struct saved_state*)engine->app->savedState) = engine->state;
-            engine->app->savedStateSize = sizeof(struct saved_state);
-            break;
-        case APP_CMD_INIT_WINDOW:
-            // The window is being shown, get it ready.
-            if (engine->app->window != NULL) {
-                engine_init_display(engine);
-                engine_draw_frame(engine);
-            }
-            break;
-        case APP_CMD_TERM_WINDOW:
-            // The window is being hidden or closed, clean it up.
-            engine_term_display(engine);
-            break;
-        case APP_CMD_GAINED_FOCUS:
-            // When our app gains focus, we start monitoring the accelerometer.
-            if (engine->accelerometerSensor != NULL) {
-                ASensorEventQueue_enableSensor(engine->sensorEventQueue,
-                        engine->accelerometerSensor);
-                // We'd like to get 60 events per second (in us).
-                ASensorEventQueue_setEventRate(engine->sensorEventQueue,
-                        engine->accelerometerSensor, (1000L/60)*1000);
-            }
-            break;
-        case APP_CMD_LOST_FOCUS:
-            // When our app loses focus, we stop monitoring the accelerometer.
-            // This is to avoid consuming battery while not being used.
-            if (engine->accelerometerSensor != NULL) {
-                ASensorEventQueue_disableSensor(engine->sensorEventQueue,
-                        engine->accelerometerSensor);
-            }
-            // Also stop animating.
-            engine->animating = 0;
-            engine_draw_frame(engine);
-            break;
-    }
-}
-
-/**
- * This is the main entry point of a native application that is using
- * android_native_app_glue.  It runs in its own thread, with its own
- * event loop for receiving input events and doing other things.
- */
-void android_main(struct android_app* state) {
-    struct engine engine;
-
-    // Make sure glue isn't stripped.
-    app_dummy();
-
-    memset(&engine, 0, sizeof(engine));
-    state->userData = &engine;
-    state->onAppCmd = engine_handle_cmd;
-    state->onInputEvent = engine_handle_input;
-    engine.app = state;
-
-    // Prepare to monitor accelerometer
-    engine.sensorManager = ASensorManager_getInstance();
-    engine.accelerometerSensor = ASensorManager_getDefaultSensor(engine.sensorManager,
-            ASENSOR_TYPE_ACCELEROMETER);
-    engine.sensorEventQueue = ASensorManager_createEventQueue(engine.sensorManager,
-            state->looper, LOOPER_ID_USER, NULL, NULL);
-
-    if (state->savedState != NULL) {
-        // We are starting with a previous saved state; restore from it.
-        engine.state = *(struct saved_state*)state->savedState;
-    }
-
-    // loop waiting for stuff to do.
-
-    while (1) {
-        // Read all pending events.
-        int ident;
-        int events;
-        struct android_poll_source* source;
-
-        // If not animating, we will block forever waiting for events.
-        // If animating, we loop until all events are read, then continue
-        // to draw the next frame of animation.
-        while ((ident=ALooper_pollAll(engine.animating ? 0 : -1, NULL, &events,
-                (void**)&source)) >= 0) {
-
-            // Process this event.
-            if (source != NULL) {
-                source->process(state, source);
-            }
-
-            // If a sensor has data, process it now.
-            if (ident == LOOPER_ID_USER) {
-                if (engine.accelerometerSensor != NULL) {
-                    ASensorEvent event;
-                    while (ASensorEventQueue_getEvents(engine.sensorEventQueue,
-                            &event, 1) > 0) {
-                        LOGI("accelerometer: x=%f y=%f z=%f",
-                                event.acceleration.x, event.acceleration.y,
-                                event.acceleration.z);
-                    }
-                }
-            }
-
-            // Check if we are exiting.
-            if (state->destroyRequested != 0) {
-                engine_term_display(&engine);
-                return;
-            }
-        }
-
-        if (engine.animating) {
-            // Done with events; draw next animation frame.
-            engine.state.angle += .01f;
-            if (engine.state.angle > 1) {
-                engine.state.angle = 0;
-            }
-
-            // Drawing is throttled to the screen update rate, so there
-            // is no need to do timing here.
-            engine_draw_frame(&engine);
-        }
-    }
-}
-//END_INCLUDE(all)
diff --git a/src/modules/audio_processing/main/test/android/apmtest/res/values/strings.xml b/src/modules/audio_processing/main/test/android/apmtest/res/values/strings.xml
deleted file mode 100644
index d0bd0f3..0000000
--- a/src/modules/audio_processing/main/test/android/apmtest/res/values/strings.xml
+++ /dev/null
@@ -1,4 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<resources>
-    <string name="app_name">apmtest</string>
-</resources>
diff --git a/src/modules/audio_processing/main/test/process_test/Android.mk b/src/modules/audio_processing/main/test/process_test/Android.mk
deleted file mode 100644
index 23080aa..0000000
--- a/src/modules/audio_processing/main/test/process_test/Android.mk
+++ /dev/null
@@ -1,48 +0,0 @@
-#  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-#  Use of this source code is governed by a BSD-style license
-#  that can be found in the LICENSE file in the root of the source
-#  tree. An additional intellectual property rights grant can be found
-#  in the file PATENTS.  All contributing project authors may
-#  be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH:= $(call my-dir)
-
-# apm test app
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= \
-    process_test.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_THREAD_RR' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' 
-
-LOCAL_CPPFLAGS := 
-LOCAL_LDFLAGS :=
-LOCAL_C_INCLUDES := \
-    external/gtest/include \
-    $(LOCAL_PATH)/../../../../../system_wrappers/interface \
-    $(LOCAL_PATH)/../../interface \
-    $(LOCAL_PATH)/../../../../interface \
-    $(LOCAL_PATH)/../../../../..
-
-LOCAL_STATIC_LIBRARIES := \
-    libgtest 
-
-LOCAL_SHARED_LIBRARIES := \
-    libutils \
-    libstlport \
-    libwebrtc_audio_preprocessing 
-
-LOCAL_MODULE:= webrtc_apm_process_test
-
-include external/stlport/libstlport.mk
-include $(BUILD_EXECUTABLE)
diff --git a/src/modules/audio_processing/main/test/process_test/apmtest.m b/src/modules/audio_processing/main/test/process_test/apmtest.m
deleted file mode 100644
index 6152bb5..0000000
--- a/src/modules/audio_processing/main/test/process_test/apmtest.m
+++ /dev/null
@@ -1,360 +0,0 @@
-function apmtest(task, testname, casenumber, legacy)
-%APMTEST is a tool to process APM file sets and easily display the output.
-%   APMTEST(TASK, TESTNAME, CASENUMBER) performs one of several TASKs:
-%     'test'  Processes the files to produce test output.
-%     'list'  Prints a list of cases in the test set, preceded by their
-%             CASENUMBERs.
-%     'show'  Uses spclab to show the test case specified by the
-%             CASENUMBER parameter.
-%
-%   using a set of test files determined by TESTNAME:
-%     'all'   All tests.
-%     'apm'   The standard APM test set (default).
-%     'apmm'  The mobile APM test set.
-%     'aec'   The AEC test set.
-%     'aecm'  The AECM test set.
-%     'agc'   The AGC test set.
-%     'ns'    The NS test set.
-%     'vad'   The VAD test set.
-%
-%   CASENUMBER can be used to select a single test case. Omit CASENUMBER,
-%   or set to zero, to use all test cases.
-%
-
-if nargin < 4
-  % Set to true to run old VQE recordings.
-  legacy = false;
-end
-
-if nargin < 3
-  casenumber = 0;
-end
-
-if nargin < 2
-  task = 'test';
-end
-
-if nargin < 1
-  testname = 'all';
-end
-
-if ~strcmp(task, 'test') && ~strcmp(task, 'list') && ~strcmp(task, 'show')
-  error(['TASK ' task ' is not recognized']);
-end
-
-if casenumber == 0 && strcmp(task, 'show')
-  error(['CASENUMBER must be specified for TASK ' task]);
-end
-
-filepath = 'data/';
-inpath = [filepath 'input/'];
-outpath = [filepath 'output/'];
-refpath = [filepath 'reference/'];
-
-% Temporary
-if legacy
-  refpath = [filepath 'output/'];
-  outpath = [filepath 'reference/'];
-end
-
-if strcmp(testname, 'all')
-  tests = {'apm','apmm','aec','aecm','agc','ns','vad'};
-else
-  tests = {testname};
-end
-
-if legacy
-  progname = '/usr/local/google/p4/dev/depot/test';
-else
-  progname = './process_test';
-end
-
-global farFile;
-global nearFile;
-global eventFile;
-global delayFile;
-global driftFile;
-
-if legacy
-  farFile = 'vqeFar.pcm';
-  nearFile = 'vqeNear.pcm';
-  eventFile = 'vqeEvent.dat';
-  delayFile = 'vqeBuf.dat';
-  driftFile = 'vqeDrift.dat';
-else
-  farFile = 'apm_far.pcm';
-  nearFile = 'apm_near.pcm';
-  eventFile = 'apm_event.dat';
-  delayFile = 'apm_delay.dat';
-  driftFile = 'apm_drift.dat';
-end
-
-simulateMode = false;
-nErr = 0;
-nCases = 0;
-for i=1:length(tests)
-  simulateMode = false;
-
-  if strcmp(tests{i}, 'apm')
-    testdir = ['apm/'];
-    outfile = ['out'];
-    if legacy
-      opt = ['-ec 1 -agc 2 -nc 2 -vad 3'];
-    else
-      opt = ['--no_progress -hpf' ...
-          ' -aec --drift_compensation -agc --fixed_digital' ...
-          ' -ns --ns_moderate -vad'];
-    end
-
-  elseif strcmp(tests{i}, 'apm-swb')
-    simulateMode = true;
-    testdir = ['apm-swb/'];
-    outfile = ['out'];
-    if legacy
-      opt = ['-fs 32000 -ec 1 -agc 2 -nc 2'];
-    else
-      opt = ['--no_progress -fs 32000 -hpf' ...
-          ' -aec --drift_compensation -agc --adaptive_digital' ...
-          ' -ns --ns_moderate -vad'];
-    end
-  elseif strcmp(tests{i}, 'apmm')
-    testdir = ['apmm/'];
-    outfile = ['out'];
-    opt = ['-aec --drift_compensation -agc --fixed_digital -hpf -ns ' ...
-        '--ns_moderate'];
-
-  else
-    error(['TESTNAME ' tests{i} ' is not recognized']);
-  end
-
-  inpath = [inpath testdir];
-  outpath = [outpath testdir];
-  refpath = [refpath testdir];
-
-  if ~exist(inpath,'dir')
-    error(['Input directory ' inpath ' does not exist']);
-  end
-
-  if ~exist(refpath,'dir')
-    warning(['Reference directory ' refpath ' does not exist']);
-  end
-
-  [status, errMsg] = mkdir(outpath);
-  if (status == 0)
-    error(errMsg);
-  end
-
-  [nErr, nCases] = recurseDir(inpath, outpath, refpath, outfile, ...
-      progname, opt, simulateMode, nErr, nCases, task, casenumber, legacy);
-
-  if strcmp(task, 'test') || strcmp(task, 'show')
-    system(['rm ' farFile]);
-    system(['rm ' nearFile]);
-    if simulateMode == false
-      system(['rm ' eventFile]);
-      system(['rm ' delayFile]);
-      system(['rm ' driftFile]);
-    end
-  end
-end
-
-if ~strcmp(task, 'list')
-  if nErr == 0
-    fprintf(1, '\nAll files are bit-exact to reference\n', nErr);
-  else
-    fprintf(1, '\n%d files are NOT bit-exact to reference\n', nErr);
-  end
-end
-
-
-function [nErrOut, nCases] = recurseDir(inpath, outpath, refpath, ...
-    outfile, progname, opt, simulateMode, nErr, nCases, task, casenumber, ...
-    legacy)
-
-global farFile;
-global nearFile;
-global eventFile;
-global delayFile;
-global driftFile;
-
-dirs = dir(inpath);
-nDirs = 0;
-nErrOut = nErr;
-for i=3:length(dirs) % skip . and ..
-  nDirs = nDirs + dirs(i).isdir;
-end
-
-
-if nDirs == 0
-  nCases = nCases + 1;
-
-  if casenumber == nCases || casenumber == 0
-
-    if strcmp(task, 'list')
-      fprintf([num2str(nCases) '. ' outfile '\n'])
-    else
-      vadoutfile = ['vad_' outfile '.dat'];
-      outfile = [outfile '.pcm'];
-
-      % Check for VAD test
-      vadTest = 0;
-      if ~isempty(findstr(opt, '-vad'))
-        vadTest = 1;
-        if legacy
-          opt = [opt ' ' outpath vadoutfile];
-        else
-          opt = [opt ' --vad_out_file ' outpath vadoutfile];
-        end
-      end
-
-      if exist([inpath 'vqeFar.pcm'])
-        system(['ln -s -f ' inpath 'vqeFar.pcm ' farFile]);
-      elseif exist([inpath 'apm_far.pcm'])
-        system(['ln -s -f ' inpath 'apm_far.pcm ' farFile]);
-      end
-
-      if exist([inpath 'vqeNear.pcm'])
-        system(['ln -s -f ' inpath 'vqeNear.pcm ' nearFile]);
-      elseif exist([inpath 'apm_near.pcm'])
-        system(['ln -s -f ' inpath 'apm_near.pcm ' nearFile]);
-      end
-
-      if exist([inpath 'vqeEvent.dat'])
-        system(['ln -s -f ' inpath 'vqeEvent.dat ' eventFile]);
-      elseif exist([inpath 'apm_event.day'])
-        system(['ln -s -f ' inpath 'apm_event.dat ' eventFile]);
-      end
-
-      if exist([inpath 'vqeBuf.dat'])
-        system(['ln -s -f ' inpath 'vqeBuf.dat ' delayFile]);
-      elseif exist([inpath 'apm_delay.day'])
-        system(['ln -s -f ' inpath 'apm_delay.dat ' delayFile]);
-      end
-
-      if exist([inpath 'vqeSkew.dat'])
-        system(['ln -s -f ' inpath 'vqeSkew.dat ' driftFile]);
-      elseif exist([inpath 'vqeDrift.dat'])
-        system(['ln -s -f ' inpath 'vqeDrift.dat ' driftFile]);
-      elseif exist([inpath 'apm_drift.dat'])
-        system(['ln -s -f ' inpath 'apm_drift.dat ' driftFile]);
-      end
-
-      if simulateMode == false
-        command = [progname ' -o ' outpath outfile ' ' opt];
-      else
-        if legacy
-          inputCmd = [' -in ' nearFile];
-        else
-          inputCmd = [' -i ' nearFile];
-        end
-
-        if exist([farFile])
-          if legacy
-            inputCmd = [' -if ' farFile inputCmd];
-          else
-            inputCmd = [' -ir ' farFile inputCmd];
-          end
-        end
-        command = [progname inputCmd ' -o ' outpath outfile ' ' opt];
-      end
-      % This prevents MATLAB from using its own C libraries.
-      shellcmd = ['bash -c "unset LD_LIBRARY_PATH;'];
-      fprintf([command '\n']);
-      [status, result] = system([shellcmd command '"']);
-      fprintf(result);
-
-      fprintf(['Reference file: ' refpath outfile '\n']);
-
-      if vadTest == 1
-        equal_to_ref = are_files_equal([outpath vadoutfile], ...
-                                       [refpath vadoutfile], ...
-                                       'int8');
-        if ~equal_to_ref
-          nErr = nErr + 1;
-        end
-      end
-
-      [equal_to_ref, diffvector] = are_files_equal([outpath outfile], ...
-                                                   [refpath outfile], ...
-                                                   'int16');
-      if ~equal_to_ref
-        nErr = nErr + 1;
-      end
-
-      if strcmp(task, 'show')
-        % Assume the last init gives the sample rate of interest.
-        str_idx = strfind(result, 'Sample rate:');
-        fs = str2num(result(str_idx(end) + 13:str_idx(end) + 17));
-        fprintf('Using %d Hz\n', fs);
-
-        if exist([farFile])
-          spclab(fs, farFile, nearFile, [refpath outfile], ...
-              [outpath outfile], diffvector);
-          %spclab(fs, diffvector);
-        else
-          spclab(fs, nearFile, [refpath outfile], [outpath outfile], ...
-              diffvector);
-          %spclab(fs, diffvector);
-        end
-
-        if vadTest == 1
-          spclab([refpath vadoutfile], [outpath vadoutfile]);
-        end
-      end
-    end
-  end
-else
-
-  for i=3:length(dirs)
-    if dirs(i).isdir
-      [nErr, nCases] = recurseDir([inpath dirs(i).name '/'], outpath, ...
-          refpath,[outfile '_' dirs(i).name], progname, opt, ...
-          simulateMode, nErr, nCases, task, casenumber, legacy);
-    end
-  end
-end
-nErrOut = nErr;
-
-function [are_equal, diffvector] = ...
-    are_files_equal(newfile, reffile, precision, diffvector)
-
-are_equal = false;
-diffvector = 0;
-if ~exist(newfile,'file')
-  warning(['Output file ' newfile ' does not exist']);  
-  return
-end
-
-if ~exist(reffile,'file')
-  warning(['Reference file ' reffile ' does not exist']);  
-  return
-end
-
-fid = fopen(newfile,'rb');
-new = fread(fid,inf,precision);
-fclose(fid);
-
-fid = fopen(reffile,'rb');
-ref = fread(fid,inf,precision);
-fclose(fid);
-
-if length(new) ~= length(ref)
-  warning('Reference is not the same length as output');
-  minlength = min(length(new), length(ref));
-  new = new(1:minlength);
-  ref = ref(1:minlength);
-end
-diffvector = new - ref;
-
-if isequal(new, ref)
-  fprintf([newfile ' is bit-exact to reference\n']);
-  are_equal = true;
-else
-  if isempty(new)
-    warning([newfile ' is empty']);
-    return
-  end
-  snr = snrseg(new,ref,80);
-  fprintf('\n');
-  are_equal = false;
-end
diff --git a/src/modules/audio_processing/main/test/process_test/process_test.cc b/src/modules/audio_processing/main/test/process_test/process_test.cc
deleted file mode 100644
index c62345f..0000000
--- a/src/modules/audio_processing/main/test/process_test/process_test.cc
+++ /dev/null
@@ -1,628 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-#ifdef WEBRTC_ANDROID
-#include <sys/stat.h>
-#endif
-
-#include "tick_util.h"
-#include "gtest/gtest.h"
-#include "module_common_types.h"
-
-#include "audio_processing.h"
-
-#include "cpu_features_wrapper.h"
-
-using webrtc::AudioFrame;
-using webrtc::TickInterval;
-using webrtc::TickTime;
-
-using webrtc::AudioProcessing;
-using webrtc::GainControl;
-using webrtc::NoiseSuppression;
-
-void usage() {
-  printf(
-  "Usage: process_test [options] [-ir REVERSE_FILE] [-i PRIMARY_FILE]\n");
-  printf(
-  "  [-o OUT_FILE]\n");
-  printf(
-  "process_test is a test application for AudioProcessing.\n\n"
-  "When -ir or -i is specified the files will be processed directly in a\n"
-  "simulation mode. Otherwise the full set of test files is expected to be\n"
-  "present in the working directory.\n");
-  printf("\n");
-  printf("Options\n");
-  printf("General configuration:\n");
-  printf("  -fs SAMPLE_RATE_HZ\n");
-  printf("  -ch CHANNELS_IN CHANNELS_OUT\n");
-  printf("  -rch REVERSE_CHANNELS\n");
-  printf("\n");
-  printf("Component configuration:\n");
-  printf(
-  "All components are disabled by default. Each block below begins with a\n"
-  "flag to enable the component with default settings. The subsequent flags\n"
-  "in the block are used to provide configuration settings.\n");
-  printf("\n  -aec     Echo cancellation\n");
-  printf("  --drift_compensation\n");
-  printf("  --no_drift_compensation\n");
-  printf("\n  -aecm    Echo control mobile\n");
-  printf("\n  -agc     Gain control\n");
-  printf("  --analog\n");
-  printf("  --adaptive_digital\n");
-  printf("  --fixed_digital\n");
-  printf("  --target_level LEVEL\n");
-  printf("  --compression_gain GAIN\n");
-  printf("  --limiter\n");
-  printf("  --no_limiter\n");
-  printf("\n  -hpf     High pass filter\n");
-  printf("\n  -ns      Noise suppression\n");
-  printf("  --ns_low\n");
-  printf("  --ns_moderate\n");
-  printf("  --ns_high\n");
-  printf("  --ns_very_high\n");
-  printf("\n  -vad     Voice activity detection\n");
-  printf("  --vad_out_file FILE");
-  printf("\n");
-  printf("Modifiers:\n");
-  printf("  --perf          Measure performance.\n");
-  printf("  --quiet         Suppress text output.\n");
-  printf("  --no_progress   Suppress progress.\n");
-  printf("  --version       Print version information and exit.\n");
-}
-
-// void function for gtest.
-void void_main(int argc, char* argv[]) {
-  if (argc > 1 && strcmp(argv[1], "--help") == 0) {
-    usage();
-    return;
-  }
-
-  if (argc < 2) {
-    printf("Did you mean to run without arguments?\n");
-    printf("Try `process_test --help' for more information.\n\n");
-  }
-
-  AudioProcessing* apm = AudioProcessing::Create(0);
-  ASSERT_TRUE(apm != NULL);
-
-  WebRtc_Word8 version[1024];
-  WebRtc_UWord32 version_bytes_remaining = sizeof(version);
-  WebRtc_UWord32 version_position = 0;
-
-  const char* far_filename = NULL;
-  const char* near_filename = NULL;
-  const char* out_filename = NULL;
-  const char* vad_out_filename = NULL;
-
-  int32_t sample_rate_hz = 16000;
-  int32_t device_sample_rate_hz = 16000;
-
-  int num_capture_input_channels = 1;
-  int num_capture_output_channels = 1;
-  int num_render_channels = 1;
-
-  int samples_per_channel = sample_rate_hz / 100;
-
-  bool simulating = false;
-  bool perf_testing = false;
-  bool verbose = true;
-  bool progress = true;
-  //bool interleaved = true;
-
-  for (int i = 1; i < argc; i++) {
-     if (strcmp(argv[i], "-ir") == 0) {
-      i++;
-      ASSERT_LT(i, argc) << "Specify filename after -ir";
-      far_filename = argv[i];
-      simulating = true;
-
-    } else if (strcmp(argv[i], "-i") == 0) {
-      i++;
-      ASSERT_LT(i, argc) << "Specify filename after -i";
-      near_filename = argv[i];
-      simulating = true;
-
-    } else if (strcmp(argv[i], "-o") == 0) {
-      i++;
-      ASSERT_LT(i, argc) << "Specify filename after -o";
-      out_filename = argv[i];
-
-    } else if (strcmp(argv[i], "-fs") == 0) {
-      i++;
-      ASSERT_LT(i, argc) << "Specify sample rate after -fs";
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &sample_rate_hz));
-      samples_per_channel = sample_rate_hz / 100;
-
-      ASSERT_EQ(apm->kNoError,
-                apm->set_sample_rate_hz(sample_rate_hz));
-
-    } else if (strcmp(argv[i], "-ch") == 0) {
-      i++;
-      ASSERT_LT(i + 1, argc) << "Specify number of channels after -ch";
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_input_channels));
-      i++;
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_output_channels));
-
-      ASSERT_EQ(apm->kNoError,
-                apm->set_num_channels(num_capture_input_channels,
-                                      num_capture_output_channels));
-
-    } else if (strcmp(argv[i], "-rch") == 0) {
-      i++;
-      ASSERT_LT(i, argc) << "Specify number of channels after -rch";
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &num_render_channels));
-
-      ASSERT_EQ(apm->kNoError,
-                apm->set_num_reverse_channels(num_render_channels));
-
-    } else if (strcmp(argv[i], "-aec") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
-
-    } else if (strcmp(argv[i], "-noasm") == 0) {
-      WebRtc_GetCPUInfo = WebRtc_GetCPUInfoNoASM;
-
-    } else if (strcmp(argv[i], "--drift_compensation") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
-      // TODO(ajm): this is enabled in the VQE test app by default. Investigate
-      //            why it can give better performance despite passing zeros.
-      ASSERT_EQ(apm->kNoError,
-                apm->echo_cancellation()->enable_drift_compensation(true));
-    } else if (strcmp(argv[i], "--no_drift_compensation") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-                apm->echo_cancellation()->enable_drift_compensation(false));
-
-    } else if (strcmp(argv[i], "-aecm") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(true));
-
-    } else if (strcmp(argv[i], "-agc") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
-
-    } else if (strcmp(argv[i], "--analog") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-                apm->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
-
-    } else if (strcmp(argv[i], "--adaptive_digital") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-                apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
-
-    } else if (strcmp(argv[i], "--fixed_digital") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-                apm->gain_control()->set_mode(GainControl::kFixedDigital));
-
-    } else if (strcmp(argv[i], "--target_level") == 0) {
-      i++;
-      int level;
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &level));
-
-      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-                apm->gain_control()->set_target_level_dbfs(level));
-
-    } else if (strcmp(argv[i], "--compression_gain") == 0) {
-      i++;
-      int gain;
-      ASSERT_EQ(1, sscanf(argv[i], "%d", &gain));
-
-      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-                apm->gain_control()->set_compression_gain_db(gain));
-
-    } else if (strcmp(argv[i], "--limiter") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-                apm->gain_control()->enable_limiter(true));
-
-    } else if (strcmp(argv[i], "--no_limiter") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-                apm->gain_control()->enable_limiter(false));
-
-    } else if (strcmp(argv[i], "-hpf") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->high_pass_filter()->Enable(true));
-
-    } else if (strcmp(argv[i], "-ns") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
-
-    } else if (strcmp(argv[i], "--ns_low") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-          apm->noise_suppression()->set_level(NoiseSuppression::kLow));
-
-    } else if (strcmp(argv[i], "--ns_moderate") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-          apm->noise_suppression()->set_level(NoiseSuppression::kModerate));
-
-    } else if (strcmp(argv[i], "--ns_high") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-          apm->noise_suppression()->set_level(NoiseSuppression::kHigh));
-
-    } else if (strcmp(argv[i], "--ns_very_high") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
-      ASSERT_EQ(apm->kNoError,
-          apm->noise_suppression()->set_level(NoiseSuppression::kVeryHigh));
-
-    } else if (strcmp(argv[i], "-vad") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
-
-    } else if (strcmp(argv[i], "--vad_out_file") == 0) {
-      i++;
-      ASSERT_LT(i, argc) << "Specify filename after --vad_out_file";
-      vad_out_filename = argv[i];
-
-    } else if (strcmp(argv[i], "--perf") == 0) {
-      perf_testing = true;
-
-    } else if (strcmp(argv[i], "--quiet") == 0) {
-      verbose = false;
-      progress = false;
-
-    } else if (strcmp(argv[i], "--no_progress") == 0) {
-      progress = false;
-
-    } else if (strcmp(argv[i], "--version") == 0) {
-      ASSERT_EQ(apm->kNoError, apm->Version(version,
-                                            version_bytes_remaining,
-                                            version_position));
-      printf("%s\n", version);
-      return;
-
-    } else {
-      FAIL() << "Unrecognized argument " << argv[i];
-    }
-  }
-
-  if (verbose) {
-    printf("Sample rate: %d Hz\n", sample_rate_hz);
-    printf("Primary channels: %d (in), %d (out)\n",
-           num_capture_input_channels,
-           num_capture_output_channels);
-    printf("Reverse channels: %d \n", num_render_channels);
-  }
-
-  const char far_file_default[] = "apm_far.pcm";
-  const char near_file_default[] = "apm_near.pcm";
-  const char out_file_default[] = "out.pcm";
-  const char event_filename[] = "apm_event.dat";
-  const char delay_filename[] = "apm_delay.dat";
-  const char drift_filename[] = "apm_drift.dat";
-  const char vad_file_default[] = "vad_out.dat";
-
-  if (!simulating) {
-    far_filename = far_file_default;
-    near_filename = near_file_default;
-  }
-
-  if (out_filename == NULL) {
-    out_filename = out_file_default;
-  }
-
-  if (vad_out_filename == NULL) {
-    vad_out_filename = vad_file_default;
-  }
-
-  FILE* far_file = NULL;
-  FILE* near_file = NULL;
-  FILE* out_file = NULL;
-  FILE* event_file = NULL;
-  FILE* delay_file = NULL;
-  FILE* drift_file = NULL;
-  FILE* vad_out_file = NULL;
-
-  if (far_filename != NULL) {
-    far_file = fopen(far_filename, "rb");
-    ASSERT_TRUE(NULL != far_file) << "Unable to open far-end audio file "
-                                  << far_filename;
-  }
-
-  near_file = fopen(near_filename, "rb");
-  ASSERT_TRUE(NULL != near_file) << "Unable to open near-end audio file "
-                                 << near_filename;
-  struct stat st;
-  stat(near_filename, &st);
-  int near_size_samples = st.st_size / sizeof(int16_t);
-
-  out_file = fopen(out_filename, "wb");
-  ASSERT_TRUE(NULL != out_file) << "Unable to open output audio file "
-                                << out_filename;
-
-  if (!simulating) {
-    event_file = fopen(event_filename, "rb");
-    ASSERT_TRUE(NULL != event_file) << "Unable to open event file "
-                                    << event_filename;
-
-    delay_file = fopen(delay_filename, "rb");
-    ASSERT_TRUE(NULL != delay_file) << "Unable to open buffer file "
-                                    << delay_filename;
-
-    drift_file = fopen(drift_filename, "rb");
-    ASSERT_TRUE(NULL != drift_file) << "Unable to open drift file "
-                                    << drift_filename;
-  }
-
-  if (apm->voice_detection()->is_enabled()) {
-    vad_out_file = fopen(vad_out_filename, "wb");
-    ASSERT_TRUE(NULL != vad_out_file) << "Unable to open VAD output file "
-                                      << vad_out_file;
-  }
-
-  enum Events {
-    kInitializeEvent,
-    kRenderEvent,
-    kCaptureEvent,
-    kResetEventDeprecated
-  };
-  int16_t event = 0;
-  size_t read_count = 0;
-  int reverse_count = 0;
-  int primary_count = 0;
-  int near_read_samples = 0;
-  TickInterval acc_ticks;
-
-  AudioFrame far_frame;
-  far_frame._frequencyInHz = sample_rate_hz;
-
-  AudioFrame near_frame;
-  near_frame._frequencyInHz = sample_rate_hz;
-
-  int delay_ms = 0;
-  int drift_samples = 0;
-  int capture_level = 127;
-  int8_t stream_has_voice = 0;
-
-  TickTime t0 = TickTime::Now();
-  TickTime t1 = t0;
-  WebRtc_Word64 max_time_us = 0;
-  WebRtc_Word64 max_time_reverse_us = 0;
-  WebRtc_Word64 min_time_us = 1e6;
-  WebRtc_Word64 min_time_reverse_us = 1e6;
-
-  while (simulating || feof(event_file) == 0) {
-    std::ostringstream trace_stream;
-    trace_stream << "Processed frames: " << reverse_count << " (reverse), "
-                 << primary_count << " (primary)";
-    SCOPED_TRACE(trace_stream.str());
-
-
-    if (simulating) {
-      if (far_file == NULL) {
-        event = kCaptureEvent;
-      } else {
-        if (event == kRenderEvent) {
-          event = kCaptureEvent;
-        } else {
-          event = kRenderEvent;
-        }
-      }
-    } else {
-      read_count = fread(&event, sizeof(event), 1, event_file);
-      if (read_count != 1) {
-        break;
-      }
-      //if (fread(&event, sizeof(event), 1, event_file) != 1) {
-      //  break; // This is expected.
-      //}
-    }
-
-    if (event == kInitializeEvent || event == kResetEventDeprecated) {
-      ASSERT_EQ(1u,
-          fread(&sample_rate_hz, sizeof(sample_rate_hz), 1, event_file));
-      samples_per_channel = sample_rate_hz / 100;
-
-      ASSERT_EQ(1u,
-          fread(&device_sample_rate_hz,
-                sizeof(device_sample_rate_hz),
-                1,
-                event_file));
-
-      ASSERT_EQ(apm->kNoError,
-          apm->set_sample_rate_hz(sample_rate_hz));
-
-      ASSERT_EQ(apm->kNoError,
-                apm->echo_cancellation()->set_device_sample_rate_hz(
-                    device_sample_rate_hz));
-
-      far_frame._frequencyInHz = sample_rate_hz;
-      near_frame._frequencyInHz = sample_rate_hz;
-
-      if (verbose) {
-        printf("Init at frame: %d (primary), %d (reverse)\n",
-            primary_count, reverse_count);
-        printf("  Sample rate: %d Hz\n", sample_rate_hz);
-      }
-
-    } else if (event == kRenderEvent) {
-      reverse_count++;
-      far_frame._audioChannel = num_render_channels;
-      far_frame._payloadDataLengthInSamples =
-          num_render_channels * samples_per_channel;
-
-      read_count = fread(far_frame._payloadData,
-                         sizeof(WebRtc_Word16),
-                         far_frame._payloadDataLengthInSamples,
-                         far_file);
-
-      if (simulating) {
-        if (read_count != far_frame._payloadDataLengthInSamples) {
-          break; // This is expected.
-        }
-      } else {
-        ASSERT_EQ(read_count,
-            far_frame._payloadDataLengthInSamples);
-      }
-
-      if (perf_testing) {
-        t0 = TickTime::Now();
-      }
-
-      ASSERT_EQ(apm->kNoError,
-                apm->AnalyzeReverseStream(&far_frame));
-
-      if (perf_testing) {
-        t1 = TickTime::Now();
-        TickInterval tick_diff = t1 - t0;
-        acc_ticks += tick_diff;
-        if (tick_diff.Microseconds() > max_time_reverse_us) {
-          max_time_reverse_us = tick_diff.Microseconds();
-        }
-        if (tick_diff.Microseconds() < min_time_reverse_us) {
-          min_time_reverse_us = tick_diff.Microseconds();
-        }
-      }
-
-    } else if (event == kCaptureEvent) {
-      primary_count++;
-      near_frame._audioChannel = num_capture_input_channels;
-      near_frame._payloadDataLengthInSamples =
-          num_capture_input_channels * samples_per_channel;
-
-      read_count = fread(near_frame._payloadData,
-                         sizeof(WebRtc_Word16),
-                         near_frame._payloadDataLengthInSamples,
-                         near_file);
-
-      near_read_samples += read_count;
-      if (progress && primary_count % 100 == 0) {
-        printf("%.0f%% complete\r",
-            (near_read_samples * 100.0) / near_size_samples);
-        fflush(stdout);
-      }
-      if (simulating) {
-        if (read_count != near_frame._payloadDataLengthInSamples) {
-          break; // This is expected.
-        }
-
-        delay_ms = 0;
-        drift_samples = 0;
-      } else {
-        ASSERT_EQ(read_count,
-            near_frame._payloadDataLengthInSamples);
-
-        // TODO(ajm): sizeof(delay_ms) for current files?
-        ASSERT_EQ(1u,
-            fread(&delay_ms, 2, 1, delay_file));
-        ASSERT_EQ(1u,
-            fread(&drift_samples, sizeof(drift_samples), 1, drift_file));
-      }
-
-      if (perf_testing) {
-        t0 = TickTime::Now();
-      }
-
-      // TODO(ajm): fake an analog gain while simulating.
-
-      int capture_level_in = capture_level;
-      ASSERT_EQ(apm->kNoError,
-                apm->gain_control()->set_stream_analog_level(capture_level));
-      ASSERT_EQ(apm->kNoError,
-                apm->set_stream_delay_ms(delay_ms));
-      ASSERT_EQ(apm->kNoError,
-          apm->echo_cancellation()->set_stream_drift_samples(drift_samples));
-
-      int err = apm->ProcessStream(&near_frame);
-      if (err == apm->kBadStreamParameterWarning) {
-        printf("Bad parameter warning. %s\n", trace_stream.str().c_str());
-      }
-      ASSERT_TRUE(err == apm->kNoError ||
-                  err == apm->kBadStreamParameterWarning);
-
-      capture_level = apm->gain_control()->stream_analog_level();
-
-      stream_has_voice =
-          static_cast<int8_t>(apm->voice_detection()->stream_has_voice());
-      if (vad_out_file != NULL) {
-        ASSERT_EQ(1u, fwrite(&stream_has_voice,
-                             sizeof(stream_has_voice),
-                             1,
-                             vad_out_file));
-      }
-
-      if (apm->gain_control()->mode() != GainControl::kAdaptiveAnalog) {
-        ASSERT_EQ(capture_level_in, capture_level);
-      }
-
-      if (perf_testing) {
-        t1 = TickTime::Now();
-        TickInterval tick_diff = t1 - t0;
-        acc_ticks += tick_diff;
-        if (tick_diff.Microseconds() > max_time_us) {
-          max_time_us = tick_diff.Microseconds();
-        }
-        if (tick_diff.Microseconds() < min_time_us) {
-          min_time_us = tick_diff.Microseconds();
-        }
-      }
-
-      ASSERT_EQ(near_frame._payloadDataLengthInSamples,
-                fwrite(near_frame._payloadData,
-                       sizeof(WebRtc_Word16),
-                       near_frame._payloadDataLengthInSamples,
-                       out_file));
-    }
-    else {
-      FAIL() << "Event " << event << " is unrecognized";
-    }
-  }
-
-  if (verbose) {
-    printf("\nProcessed frames: %d (primary), %d (reverse)\n",
-        primary_count, reverse_count);
-  }
-
-  int8_t temp_int8;
-  if (far_file != NULL) {
-    read_count = fread(&temp_int8, sizeof(temp_int8), 1, far_file);
-    EXPECT_NE(0, feof(far_file)) << "Far-end file not fully processed";
-  }
-  read_count = fread(&temp_int8, sizeof(temp_int8), 1, near_file);
-  EXPECT_NE(0, feof(near_file)) << "Near-end file not fully processed";
-
-  if (!simulating) {
-    read_count = fread(&temp_int8, sizeof(temp_int8), 1, event_file);
-    EXPECT_NE(0, feof(event_file)) << "Event file not fully processed";
-    read_count = fread(&temp_int8, sizeof(temp_int8), 1, delay_file);
-    EXPECT_NE(0, feof(delay_file)) << "Delay file not fully processed";
-    read_count = fread(&temp_int8, sizeof(temp_int8), 1, drift_file);
-    EXPECT_NE(0, feof(drift_file)) << "Drift file not fully processed";
-  }
-
-  if (perf_testing) {
-    if (primary_count > 0) {
-      WebRtc_Word64 exec_time = acc_ticks.Milliseconds();
-      printf("\nTotal time: %.3f s, file time: %.2f s\n",
-        exec_time * 0.001, primary_count * 0.01);
-      printf("Time per frame: %.3f ms (average), %.3f ms (max),"
-             " %.3f ms (min)\n",
-          (exec_time * 1.0) / primary_count,
-          (max_time_us + max_time_reverse_us) / 1000.0,
-          (min_time_us + min_time_reverse_us) / 1000.0);
-    } else {
-      printf("Warning: no capture frames\n");
-    }
-  }
-
-  AudioProcessing::Destroy(apm);
-  apm = NULL;
-}
-
-int main(int argc, char* argv[])
-{
-  void_main(argc, argv);
-
-  return 0;
-}
diff --git a/src/modules/audio_processing/main/test/unit_test/Android.mk b/src/modules/audio_processing/main/test/unit_test/Android.mk
deleted file mode 100644
index b2029cf..0000000
--- a/src/modules/audio_processing/main/test/unit_test/Android.mk
+++ /dev/null
@@ -1,49 +0,0 @@
-#  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-#  Use of this source code is governed by a BSD-style license
-#  that can be found in the LICENSE file in the root of the source
-#  tree. An additional intellectual property rights grant can be found
-#  in the file PATENTS.  All contributing project authors may
-#  be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH:= $(call my-dir)
-
-# apm test app
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= \
-    unit_test.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_THREAD_RR' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' 
-
-LOCAL_CPPFLAGS := 
-LOCAL_LDFLAGS :=
-LOCAL_C_INCLUDES := \
-    external/gtest/include \
-    $(LOCAL_PATH)/../../../../../system_wrappers/interface \
-    $(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface \
-    $(LOCAL_PATH)/../../interface \
-    $(LOCAL_PATH)/../../../../interface \
-    $(LOCAL_PATH)/../../../../..
-
-LOCAL_STATIC_LIBRARIES := \
-    libgtest 
-
-LOCAL_SHARED_LIBRARIES := \
-    libutils \
-    libstlport \
-    libwebrtc_audio_preprocessing 
-
-LOCAL_MODULE:= webrtc_apm_unit_test
-
-include external/stlport/libstlport.mk
-include $(BUILD_EXECUTABLE)
diff --git a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.cc b/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.cc
deleted file mode 100644
index c82ffdb..0000000
--- a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.cc
+++ /dev/null
@@ -1,1111 +0,0 @@
-// Generated by the protocol buffer compiler.  DO NOT EDIT!
-
-#define INTERNAL_SUPPRESS_PROTOBUF_FIELD_DEPRECATION
-#include "audio_processing_unittest.pb.h"
-
-#include <algorithm>
-
-#include <google/protobuf/stubs/once.h>
-#include <google/protobuf/io/coded_stream.h>
-#include <google/protobuf/wire_format_lite_inl.h>
-// @@protoc_insertion_point(includes)
-
-namespace audio_processing_unittest {
-
-void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto() {
-  delete Test::default_instance_;
-  delete Test_Statistic::default_instance_;
-  delete Test_EchoMetrics::default_instance_;
-  delete OutputData::default_instance_;
-}
-
-void protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto() {
-  static bool already_here = false;
-  if (already_here) return;
-  already_here = true;
-  GOOGLE_PROTOBUF_VERIFY_VERSION;
-
-  Test::default_instance_ = new Test();
-  Test_Statistic::default_instance_ = new Test_Statistic();
-  Test_EchoMetrics::default_instance_ = new Test_EchoMetrics();
-  OutputData::default_instance_ = new OutputData();
-  Test::default_instance_->InitAsDefaultInstance();
-  Test_Statistic::default_instance_->InitAsDefaultInstance();
-  Test_EchoMetrics::default_instance_->InitAsDefaultInstance();
-  OutputData::default_instance_->InitAsDefaultInstance();
-  ::google::protobuf::internal::OnShutdown(&protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto);
-}
-
-// Force AddDescriptors() to be called at static initialization time.
-struct StaticDescriptorInitializer_audio_5fprocessing_5funittest_2eproto {
-  StaticDescriptorInitializer_audio_5fprocessing_5funittest_2eproto() {
-    protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
-  }
-} static_descriptor_initializer_audio_5fprocessing_5funittest_2eproto_;
-
-
-// ===================================================================
-
-#ifndef _MSC_VER
-const int Test_Statistic::kInstantFieldNumber;
-const int Test_Statistic::kAverageFieldNumber;
-const int Test_Statistic::kMaximumFieldNumber;
-const int Test_Statistic::kMinimumFieldNumber;
-#endif  // !_MSC_VER
-
-Test_Statistic::Test_Statistic()
-  : ::google::protobuf::MessageLite() {
-  SharedCtor();
-}
-
-void Test_Statistic::InitAsDefaultInstance() {
-}
-
-Test_Statistic::Test_Statistic(const Test_Statistic& from)
-  : ::google::protobuf::MessageLite() {
-  SharedCtor();
-  MergeFrom(from);
-}
-
-void Test_Statistic::SharedCtor() {
-  _cached_size_ = 0;
-  instant_ = 0;
-  average_ = 0;
-  maximum_ = 0;
-  minimum_ = 0;
-  ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-Test_Statistic::~Test_Statistic() {
-  SharedDtor();
-}
-
-void Test_Statistic::SharedDtor() {
-  if (this != default_instance_) {
-  }
-}
-
-void Test_Statistic::SetCachedSize(int size) const {
-  GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
-  _cached_size_ = size;
-  GOOGLE_SAFE_CONCURRENT_WRITES_END();
-}
-const Test_Statistic& Test_Statistic::default_instance() {
-  if (default_instance_ == NULL) protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();  return *default_instance_;
-}
-
-Test_Statistic* Test_Statistic::default_instance_ = NULL;
-
-Test_Statistic* Test_Statistic::New() const {
-  return new Test_Statistic;
-}
-
-void Test_Statistic::Clear() {
-  if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
-    instant_ = 0;
-    average_ = 0;
-    maximum_ = 0;
-    minimum_ = 0;
-  }
-  ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-bool Test_Statistic::MergePartialFromCodedStream(
-    ::google::protobuf::io::CodedInputStream* input) {
-#define DO_(EXPRESSION) if (!(EXPRESSION)) return false
-  ::google::protobuf::uint32 tag;
-  while ((tag = input->ReadTag()) != 0) {
-    switch (::google::protobuf::internal::WireFormatLite::GetTagFieldNumber(tag)) {
-      // optional int32 instant = 1;
-      case 1: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
-          DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
-                   ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
-                 input, &instant_)));
-          set_has_instant();
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(16)) goto parse_average;
-        break;
-      }
-      
-      // optional int32 average = 2;
-      case 2: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
-         parse_average:
-          DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
-                   ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
-                 input, &average_)));
-          set_has_average();
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(24)) goto parse_maximum;
-        break;
-      }
-      
-      // optional int32 maximum = 3;
-      case 3: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
-         parse_maximum:
-          DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
-                   ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
-                 input, &maximum_)));
-          set_has_maximum();
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(32)) goto parse_minimum;
-        break;
-      }
-      
-      // optional int32 minimum = 4;
-      case 4: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
-         parse_minimum:
-          DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
-                   ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
-                 input, &minimum_)));
-          set_has_minimum();
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectAtEnd()) return true;
-        break;
-      }
-      
-      default: {
-      handle_uninterpreted:
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_END_GROUP) {
-          return true;
-        }
-        DO_(::google::protobuf::internal::WireFormatLite::SkipField(input, tag, NULL));
-        break;
-      }
-    }
-  }
-  return true;
-#undef DO_
-}
-
-void Test_Statistic::SerializeWithCachedSizes(
-    ::google::protobuf::io::CodedOutputStream* output) const {
-  // optional int32 instant = 1;
-  if (has_instant()) {
-    ::google::protobuf::internal::WireFormatLite::WriteInt32(1, this->instant(), output);
-  }
-  
-  // optional int32 average = 2;
-  if (has_average()) {
-    ::google::protobuf::internal::WireFormatLite::WriteInt32(2, this->average(), output);
-  }
-  
-  // optional int32 maximum = 3;
-  if (has_maximum()) {
-    ::google::protobuf::internal::WireFormatLite::WriteInt32(3, this->maximum(), output);
-  }
-  
-  // optional int32 minimum = 4;
-  if (has_minimum()) {
-    ::google::protobuf::internal::WireFormatLite::WriteInt32(4, this->minimum(), output);
-  }
-  
-}
-
-int Test_Statistic::ByteSize() const {
-  int total_size = 0;
-  
-  if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
-    // optional int32 instant = 1;
-    if (has_instant()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::Int32Size(
-          this->instant());
-    }
-    
-    // optional int32 average = 2;
-    if (has_average()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::Int32Size(
-          this->average());
-    }
-    
-    // optional int32 maximum = 3;
-    if (has_maximum()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::Int32Size(
-          this->maximum());
-    }
-    
-    // optional int32 minimum = 4;
-    if (has_minimum()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::Int32Size(
-          this->minimum());
-    }
-    
-  }
-  GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
-  _cached_size_ = total_size;
-  GOOGLE_SAFE_CONCURRENT_WRITES_END();
-  return total_size;
-}
-
-void Test_Statistic::CheckTypeAndMergeFrom(
-    const ::google::protobuf::MessageLite& from) {
-  MergeFrom(*::google::protobuf::down_cast<const Test_Statistic*>(&from));
-}
-
-void Test_Statistic::MergeFrom(const Test_Statistic& from) {
-  GOOGLE_CHECK_NE(&from, this);
-  if (from._has_bits_[0 / 32] & (0xffu << (0 % 32))) {
-    if (from.has_instant()) {
-      set_instant(from.instant());
-    }
-    if (from.has_average()) {
-      set_average(from.average());
-    }
-    if (from.has_maximum()) {
-      set_maximum(from.maximum());
-    }
-    if (from.has_minimum()) {
-      set_minimum(from.minimum());
-    }
-  }
-}
-
-void Test_Statistic::CopyFrom(const Test_Statistic& from) {
-  if (&from == this) return;
-  Clear();
-  MergeFrom(from);
-}
-
-bool Test_Statistic::IsInitialized() const {
-  
-  return true;
-}
-
-void Test_Statistic::Swap(Test_Statistic* other) {
-  if (other != this) {
-    std::swap(instant_, other->instant_);
-    std::swap(average_, other->average_);
-    std::swap(maximum_, other->maximum_);
-    std::swap(minimum_, other->minimum_);
-    std::swap(_has_bits_[0], other->_has_bits_[0]);
-    std::swap(_cached_size_, other->_cached_size_);
-  }
-}
-
-::std::string Test_Statistic::GetTypeName() const {
-  return "audio_processing_unittest.Test.Statistic";
-}
-
-
-// -------------------------------------------------------------------
-
-#ifndef _MSC_VER
-const int Test_EchoMetrics::kResidualEchoReturnLossFieldNumber;
-const int Test_EchoMetrics::kEchoReturnLossFieldNumber;
-const int Test_EchoMetrics::kEchoReturnLossEnhancementFieldNumber;
-const int Test_EchoMetrics::kANlpFieldNumber;
-#endif  // !_MSC_VER
-
-Test_EchoMetrics::Test_EchoMetrics()
-  : ::google::protobuf::MessageLite() {
-  SharedCtor();
-}
-
-void Test_EchoMetrics::InitAsDefaultInstance() {
-  residualechoreturnloss_ = const_cast< ::audio_processing_unittest::Test_Statistic*>(&::audio_processing_unittest::Test_Statistic::default_instance());
-  echoreturnloss_ = const_cast< ::audio_processing_unittest::Test_Statistic*>(&::audio_processing_unittest::Test_Statistic::default_instance());
-  echoreturnlossenhancement_ = const_cast< ::audio_processing_unittest::Test_Statistic*>(&::audio_processing_unittest::Test_Statistic::default_instance());
-  anlp_ = const_cast< ::audio_processing_unittest::Test_Statistic*>(&::audio_processing_unittest::Test_Statistic::default_instance());
-}
-
-Test_EchoMetrics::Test_EchoMetrics(const Test_EchoMetrics& from)
-  : ::google::protobuf::MessageLite() {
-  SharedCtor();
-  MergeFrom(from);
-}
-
-void Test_EchoMetrics::SharedCtor() {
-  _cached_size_ = 0;
-  residualechoreturnloss_ = NULL;
-  echoreturnloss_ = NULL;
-  echoreturnlossenhancement_ = NULL;
-  anlp_ = NULL;
-  ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-Test_EchoMetrics::~Test_EchoMetrics() {
-  SharedDtor();
-}
-
-void Test_EchoMetrics::SharedDtor() {
-  if (this != default_instance_) {
-    delete residualechoreturnloss_;
-    delete echoreturnloss_;
-    delete echoreturnlossenhancement_;
-    delete anlp_;
-  }
-}
-
-void Test_EchoMetrics::SetCachedSize(int size) const {
-  GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
-  _cached_size_ = size;
-  GOOGLE_SAFE_CONCURRENT_WRITES_END();
-}
-const Test_EchoMetrics& Test_EchoMetrics::default_instance() {
-  if (default_instance_ == NULL) protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();  return *default_instance_;
-}
-
-Test_EchoMetrics* Test_EchoMetrics::default_instance_ = NULL;
-
-Test_EchoMetrics* Test_EchoMetrics::New() const {
-  return new Test_EchoMetrics;
-}
-
-void Test_EchoMetrics::Clear() {
-  if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
-    if (has_residualechoreturnloss()) {
-      if (residualechoreturnloss_ != NULL) residualechoreturnloss_->::audio_processing_unittest::Test_Statistic::Clear();
-    }
-    if (has_echoreturnloss()) {
-      if (echoreturnloss_ != NULL) echoreturnloss_->::audio_processing_unittest::Test_Statistic::Clear();
-    }
-    if (has_echoreturnlossenhancement()) {
-      if (echoreturnlossenhancement_ != NULL) echoreturnlossenhancement_->::audio_processing_unittest::Test_Statistic::Clear();
-    }
-    if (has_anlp()) {
-      if (anlp_ != NULL) anlp_->::audio_processing_unittest::Test_Statistic::Clear();
-    }
-  }
-  ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-bool Test_EchoMetrics::MergePartialFromCodedStream(
-    ::google::protobuf::io::CodedInputStream* input) {
-#define DO_(EXPRESSION) if (!(EXPRESSION)) return false
-  ::google::protobuf::uint32 tag;
-  while ((tag = input->ReadTag()) != 0) {
-    switch (::google::protobuf::internal::WireFormatLite::GetTagFieldNumber(tag)) {
-      // optional .audio_processing_unittest.Test.Statistic residualEchoReturnLoss = 1;
-      case 1: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
-          DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
-               input, mutable_residualechoreturnloss()));
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(18)) goto parse_echoReturnLoss;
-        break;
-      }
-      
-      // optional .audio_processing_unittest.Test.Statistic echoReturnLoss = 2;
-      case 2: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
-         parse_echoReturnLoss:
-          DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
-               input, mutable_echoreturnloss()));
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(26)) goto parse_echoReturnLossEnhancement;
-        break;
-      }
-      
-      // optional .audio_processing_unittest.Test.Statistic echoReturnLossEnhancement = 3;
-      case 3: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
-         parse_echoReturnLossEnhancement:
-          DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
-               input, mutable_echoreturnlossenhancement()));
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(34)) goto parse_aNlp;
-        break;
-      }
-      
-      // optional .audio_processing_unittest.Test.Statistic aNlp = 4;
-      case 4: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
-         parse_aNlp:
-          DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
-               input, mutable_anlp()));
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectAtEnd()) return true;
-        break;
-      }
-      
-      default: {
-      handle_uninterpreted:
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_END_GROUP) {
-          return true;
-        }
-        DO_(::google::protobuf::internal::WireFormatLite::SkipField(input, tag, NULL));
-        break;
-      }
-    }
-  }
-  return true;
-#undef DO_
-}
-
-void Test_EchoMetrics::SerializeWithCachedSizes(
-    ::google::protobuf::io::CodedOutputStream* output) const {
-  // optional .audio_processing_unittest.Test.Statistic residualEchoReturnLoss = 1;
-  if (has_residualechoreturnloss()) {
-    ::google::protobuf::internal::WireFormatLite::WriteMessage(
-      1, this->residualechoreturnloss(), output);
-  }
-  
-  // optional .audio_processing_unittest.Test.Statistic echoReturnLoss = 2;
-  if (has_echoreturnloss()) {
-    ::google::protobuf::internal::WireFormatLite::WriteMessage(
-      2, this->echoreturnloss(), output);
-  }
-  
-  // optional .audio_processing_unittest.Test.Statistic echoReturnLossEnhancement = 3;
-  if (has_echoreturnlossenhancement()) {
-    ::google::protobuf::internal::WireFormatLite::WriteMessage(
-      3, this->echoreturnlossenhancement(), output);
-  }
-  
-  // optional .audio_processing_unittest.Test.Statistic aNlp = 4;
-  if (has_anlp()) {
-    ::google::protobuf::internal::WireFormatLite::WriteMessage(
-      4, this->anlp(), output);
-  }
-  
-}
-
-int Test_EchoMetrics::ByteSize() const {
-  int total_size = 0;
-  
-  if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
-    // optional .audio_processing_unittest.Test.Statistic residualEchoReturnLoss = 1;
-    if (has_residualechoreturnloss()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
-          this->residualechoreturnloss());
-    }
-    
-    // optional .audio_processing_unittest.Test.Statistic echoReturnLoss = 2;
-    if (has_echoreturnloss()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
-          this->echoreturnloss());
-    }
-    
-    // optional .audio_processing_unittest.Test.Statistic echoReturnLossEnhancement = 3;
-    if (has_echoreturnlossenhancement()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
-          this->echoreturnlossenhancement());
-    }
-    
-    // optional .audio_processing_unittest.Test.Statistic aNlp = 4;
-    if (has_anlp()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
-          this->anlp());
-    }
-    
-  }
-  GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
-  _cached_size_ = total_size;
-  GOOGLE_SAFE_CONCURRENT_WRITES_END();
-  return total_size;
-}
-
-void Test_EchoMetrics::CheckTypeAndMergeFrom(
-    const ::google::protobuf::MessageLite& from) {
-  MergeFrom(*::google::protobuf::down_cast<const Test_EchoMetrics*>(&from));
-}
-
-void Test_EchoMetrics::MergeFrom(const Test_EchoMetrics& from) {
-  GOOGLE_CHECK_NE(&from, this);
-  if (from._has_bits_[0 / 32] & (0xffu << (0 % 32))) {
-    if (from.has_residualechoreturnloss()) {
-      mutable_residualechoreturnloss()->::audio_processing_unittest::Test_Statistic::MergeFrom(from.residualechoreturnloss());
-    }
-    if (from.has_echoreturnloss()) {
-      mutable_echoreturnloss()->::audio_processing_unittest::Test_Statistic::MergeFrom(from.echoreturnloss());
-    }
-    if (from.has_echoreturnlossenhancement()) {
-      mutable_echoreturnlossenhancement()->::audio_processing_unittest::Test_Statistic::MergeFrom(from.echoreturnlossenhancement());
-    }
-    if (from.has_anlp()) {
-      mutable_anlp()->::audio_processing_unittest::Test_Statistic::MergeFrom(from.anlp());
-    }
-  }
-}
-
-void Test_EchoMetrics::CopyFrom(const Test_EchoMetrics& from) {
-  if (&from == this) return;
-  Clear();
-  MergeFrom(from);
-}
-
-bool Test_EchoMetrics::IsInitialized() const {
-  
-  return true;
-}
-
-void Test_EchoMetrics::Swap(Test_EchoMetrics* other) {
-  if (other != this) {
-    std::swap(residualechoreturnloss_, other->residualechoreturnloss_);
-    std::swap(echoreturnloss_, other->echoreturnloss_);
-    std::swap(echoreturnlossenhancement_, other->echoreturnlossenhancement_);
-    std::swap(anlp_, other->anlp_);
-    std::swap(_has_bits_[0], other->_has_bits_[0]);
-    std::swap(_cached_size_, other->_cached_size_);
-  }
-}
-
-::std::string Test_EchoMetrics::GetTypeName() const {
-  return "audio_processing_unittest.Test.EchoMetrics";
-}
-
-
-// -------------------------------------------------------------------
-
-#ifndef _MSC_VER
-const int Test::kNumReverseChannelsFieldNumber;
-const int Test::kNumChannelsFieldNumber;
-const int Test::kSampleRateFieldNumber;
-const int Test::kHasEchoCountFieldNumber;
-const int Test::kHasVoiceCountFieldNumber;
-const int Test::kIsSaturatedCountFieldNumber;
-const int Test::kEchoMetricsFieldNumber;
-#endif  // !_MSC_VER
-
-Test::Test()
-  : ::google::protobuf::MessageLite() {
-  SharedCtor();
-}
-
-void Test::InitAsDefaultInstance() {
-  echometrics_ = const_cast< ::audio_processing_unittest::Test_EchoMetrics*>(&::audio_processing_unittest::Test_EchoMetrics::default_instance());
-}
-
-Test::Test(const Test& from)
-  : ::google::protobuf::MessageLite() {
-  SharedCtor();
-  MergeFrom(from);
-}
-
-void Test::SharedCtor() {
-  _cached_size_ = 0;
-  numreversechannels_ = 0;
-  numchannels_ = 0;
-  samplerate_ = 0;
-  hasechocount_ = 0;
-  hasvoicecount_ = 0;
-  issaturatedcount_ = 0;
-  echometrics_ = NULL;
-  ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-Test::~Test() {
-  SharedDtor();
-}
-
-void Test::SharedDtor() {
-  if (this != default_instance_) {
-    delete echometrics_;
-  }
-}
-
-void Test::SetCachedSize(int size) const {
-  GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
-  _cached_size_ = size;
-  GOOGLE_SAFE_CONCURRENT_WRITES_END();
-}
-const Test& Test::default_instance() {
-  if (default_instance_ == NULL) protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();  return *default_instance_;
-}
-
-Test* Test::default_instance_ = NULL;
-
-Test* Test::New() const {
-  return new Test;
-}
-
-void Test::Clear() {
-  if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
-    numreversechannels_ = 0;
-    numchannels_ = 0;
-    samplerate_ = 0;
-    hasechocount_ = 0;
-    hasvoicecount_ = 0;
-    issaturatedcount_ = 0;
-    if (has_echometrics()) {
-      if (echometrics_ != NULL) echometrics_->::audio_processing_unittest::Test_EchoMetrics::Clear();
-    }
-  }
-  ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-bool Test::MergePartialFromCodedStream(
-    ::google::protobuf::io::CodedInputStream* input) {
-#define DO_(EXPRESSION) if (!(EXPRESSION)) return false
-  ::google::protobuf::uint32 tag;
-  while ((tag = input->ReadTag()) != 0) {
-    switch (::google::protobuf::internal::WireFormatLite::GetTagFieldNumber(tag)) {
-      // optional int32 numReverseChannels = 1;
-      case 1: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
-          DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
-                   ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
-                 input, &numreversechannels_)));
-          set_has_numreversechannels();
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(16)) goto parse_numChannels;
-        break;
-      }
-      
-      // optional int32 numChannels = 2;
-      case 2: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
-         parse_numChannels:
-          DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
-                   ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
-                 input, &numchannels_)));
-          set_has_numchannels();
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(24)) goto parse_sampleRate;
-        break;
-      }
-      
-      // optional int32 sampleRate = 3;
-      case 3: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
-         parse_sampleRate:
-          DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
-                   ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
-                 input, &samplerate_)));
-          set_has_samplerate();
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(32)) goto parse_hasEchoCount;
-        break;
-      }
-      
-      // optional int32 hasEchoCount = 4;
-      case 4: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
-         parse_hasEchoCount:
-          DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
-                   ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
-                 input, &hasechocount_)));
-          set_has_hasechocount();
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(40)) goto parse_hasVoiceCount;
-        break;
-      }
-      
-      // optional int32 hasVoiceCount = 5;
-      case 5: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
-         parse_hasVoiceCount:
-          DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
-                   ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
-                 input, &hasvoicecount_)));
-          set_has_hasvoicecount();
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(48)) goto parse_isSaturatedCount;
-        break;
-      }
-      
-      // optional int32 isSaturatedCount = 6;
-      case 6: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_VARINT) {
-         parse_isSaturatedCount:
-          DO_((::google::protobuf::internal::WireFormatLite::ReadPrimitive<
-                   ::google::protobuf::int32, ::google::protobuf::internal::WireFormatLite::TYPE_INT32>(
-                 input, &issaturatedcount_)));
-          set_has_issaturatedcount();
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(58)) goto parse_echoMetrics;
-        break;
-      }
-      
-      // optional .audio_processing_unittest.Test.EchoMetrics echoMetrics = 7;
-      case 7: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
-         parse_echoMetrics:
-          DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
-               input, mutable_echometrics()));
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectAtEnd()) return true;
-        break;
-      }
-      
-      default: {
-      handle_uninterpreted:
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_END_GROUP) {
-          return true;
-        }
-        DO_(::google::protobuf::internal::WireFormatLite::SkipField(input, tag, NULL));
-        break;
-      }
-    }
-  }
-  return true;
-#undef DO_
-}
-
-void Test::SerializeWithCachedSizes(
-    ::google::protobuf::io::CodedOutputStream* output) const {
-  // optional int32 numReverseChannels = 1;
-  if (has_numreversechannels()) {
-    ::google::protobuf::internal::WireFormatLite::WriteInt32(1, this->numreversechannels(), output);
-  }
-  
-  // optional int32 numChannels = 2;
-  if (has_numchannels()) {
-    ::google::protobuf::internal::WireFormatLite::WriteInt32(2, this->numchannels(), output);
-  }
-  
-  // optional int32 sampleRate = 3;
-  if (has_samplerate()) {
-    ::google::protobuf::internal::WireFormatLite::WriteInt32(3, this->samplerate(), output);
-  }
-  
-  // optional int32 hasEchoCount = 4;
-  if (has_hasechocount()) {
-    ::google::protobuf::internal::WireFormatLite::WriteInt32(4, this->hasechocount(), output);
-  }
-  
-  // optional int32 hasVoiceCount = 5;
-  if (has_hasvoicecount()) {
-    ::google::protobuf::internal::WireFormatLite::WriteInt32(5, this->hasvoicecount(), output);
-  }
-  
-  // optional int32 isSaturatedCount = 6;
-  if (has_issaturatedcount()) {
-    ::google::protobuf::internal::WireFormatLite::WriteInt32(6, this->issaturatedcount(), output);
-  }
-  
-  // optional .audio_processing_unittest.Test.EchoMetrics echoMetrics = 7;
-  if (has_echometrics()) {
-    ::google::protobuf::internal::WireFormatLite::WriteMessage(
-      7, this->echometrics(), output);
-  }
-  
-}
-
-int Test::ByteSize() const {
-  int total_size = 0;
-  
-  if (_has_bits_[0 / 32] & (0xffu << (0 % 32))) {
-    // optional int32 numReverseChannels = 1;
-    if (has_numreversechannels()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::Int32Size(
-          this->numreversechannels());
-    }
-    
-    // optional int32 numChannels = 2;
-    if (has_numchannels()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::Int32Size(
-          this->numchannels());
-    }
-    
-    // optional int32 sampleRate = 3;
-    if (has_samplerate()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::Int32Size(
-          this->samplerate());
-    }
-    
-    // optional int32 hasEchoCount = 4;
-    if (has_hasechocount()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::Int32Size(
-          this->hasechocount());
-    }
-    
-    // optional int32 hasVoiceCount = 5;
-    if (has_hasvoicecount()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::Int32Size(
-          this->hasvoicecount());
-    }
-    
-    // optional int32 isSaturatedCount = 6;
-    if (has_issaturatedcount()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::Int32Size(
-          this->issaturatedcount());
-    }
-    
-    // optional .audio_processing_unittest.Test.EchoMetrics echoMetrics = 7;
-    if (has_echometrics()) {
-      total_size += 1 +
-        ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
-          this->echometrics());
-    }
-    
-  }
-  GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
-  _cached_size_ = total_size;
-  GOOGLE_SAFE_CONCURRENT_WRITES_END();
-  return total_size;
-}
-
-void Test::CheckTypeAndMergeFrom(
-    const ::google::protobuf::MessageLite& from) {
-  MergeFrom(*::google::protobuf::down_cast<const Test*>(&from));
-}
-
-void Test::MergeFrom(const Test& from) {
-  GOOGLE_CHECK_NE(&from, this);
-  if (from._has_bits_[0 / 32] & (0xffu << (0 % 32))) {
-    if (from.has_numreversechannels()) {
-      set_numreversechannels(from.numreversechannels());
-    }
-    if (from.has_numchannels()) {
-      set_numchannels(from.numchannels());
-    }
-    if (from.has_samplerate()) {
-      set_samplerate(from.samplerate());
-    }
-    if (from.has_hasechocount()) {
-      set_hasechocount(from.hasechocount());
-    }
-    if (from.has_hasvoicecount()) {
-      set_hasvoicecount(from.hasvoicecount());
-    }
-    if (from.has_issaturatedcount()) {
-      set_issaturatedcount(from.issaturatedcount());
-    }
-    if (from.has_echometrics()) {
-      mutable_echometrics()->::audio_processing_unittest::Test_EchoMetrics::MergeFrom(from.echometrics());
-    }
-  }
-}
-
-void Test::CopyFrom(const Test& from) {
-  if (&from == this) return;
-  Clear();
-  MergeFrom(from);
-}
-
-bool Test::IsInitialized() const {
-  
-  return true;
-}
-
-void Test::Swap(Test* other) {
-  if (other != this) {
-    std::swap(numreversechannels_, other->numreversechannels_);
-    std::swap(numchannels_, other->numchannels_);
-    std::swap(samplerate_, other->samplerate_);
-    std::swap(hasechocount_, other->hasechocount_);
-    std::swap(hasvoicecount_, other->hasvoicecount_);
-    std::swap(issaturatedcount_, other->issaturatedcount_);
-    std::swap(echometrics_, other->echometrics_);
-    std::swap(_has_bits_[0], other->_has_bits_[0]);
-    std::swap(_cached_size_, other->_cached_size_);
-  }
-}
-
-::std::string Test::GetTypeName() const {
-  return "audio_processing_unittest.Test";
-}
-
-
-// ===================================================================
-
-#ifndef _MSC_VER
-const int OutputData::kTestFieldNumber;
-#endif  // !_MSC_VER
-
-OutputData::OutputData()
-  : ::google::protobuf::MessageLite() {
-  SharedCtor();
-}
-
-void OutputData::InitAsDefaultInstance() {
-}
-
-OutputData::OutputData(const OutputData& from)
-  : ::google::protobuf::MessageLite() {
-  SharedCtor();
-  MergeFrom(from);
-}
-
-void OutputData::SharedCtor() {
-  _cached_size_ = 0;
-  ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-OutputData::~OutputData() {
-  SharedDtor();
-}
-
-void OutputData::SharedDtor() {
-  if (this != default_instance_) {
-  }
-}
-
-void OutputData::SetCachedSize(int size) const {
-  GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
-  _cached_size_ = size;
-  GOOGLE_SAFE_CONCURRENT_WRITES_END();
-}
-const OutputData& OutputData::default_instance() {
-  if (default_instance_ == NULL) protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();  return *default_instance_;
-}
-
-OutputData* OutputData::default_instance_ = NULL;
-
-OutputData* OutputData::New() const {
-  return new OutputData;
-}
-
-void OutputData::Clear() {
-  test_.Clear();
-  ::memset(_has_bits_, 0, sizeof(_has_bits_));
-}
-
-bool OutputData::MergePartialFromCodedStream(
-    ::google::protobuf::io::CodedInputStream* input) {
-#define DO_(EXPRESSION) if (!(EXPRESSION)) return false
-  ::google::protobuf::uint32 tag;
-  while ((tag = input->ReadTag()) != 0) {
-    switch (::google::protobuf::internal::WireFormatLite::GetTagFieldNumber(tag)) {
-      // repeated .audio_processing_unittest.Test test = 1;
-      case 1: {
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_LENGTH_DELIMITED) {
-         parse_test:
-          DO_(::google::protobuf::internal::WireFormatLite::ReadMessageNoVirtual(
-                input, add_test()));
-        } else {
-          goto handle_uninterpreted;
-        }
-        if (input->ExpectTag(10)) goto parse_test;
-        if (input->ExpectAtEnd()) return true;
-        break;
-      }
-      
-      default: {
-      handle_uninterpreted:
-        if (::google::protobuf::internal::WireFormatLite::GetTagWireType(tag) ==
-            ::google::protobuf::internal::WireFormatLite::WIRETYPE_END_GROUP) {
-          return true;
-        }
-        DO_(::google::protobuf::internal::WireFormatLite::SkipField(input, tag, NULL));
-        break;
-      }
-    }
-  }
-  return true;
-#undef DO_
-}
-
-void OutputData::SerializeWithCachedSizes(
-    ::google::protobuf::io::CodedOutputStream* output) const {
-  // repeated .audio_processing_unittest.Test test = 1;
-  for (int i = 0; i < this->test_size(); i++) {
-    ::google::protobuf::internal::WireFormatLite::WriteMessage(
-      1, this->test(i), output);
-  }
-  
-}
-
-int OutputData::ByteSize() const {
-  int total_size = 0;
-  
-  // repeated .audio_processing_unittest.Test test = 1;
-  total_size += 1 * this->test_size();
-  for (int i = 0; i < this->test_size(); i++) {
-    total_size +=
-      ::google::protobuf::internal::WireFormatLite::MessageSizeNoVirtual(
-        this->test(i));
-  }
-  
-  GOOGLE_SAFE_CONCURRENT_WRITES_BEGIN();
-  _cached_size_ = total_size;
-  GOOGLE_SAFE_CONCURRENT_WRITES_END();
-  return total_size;
-}
-
-void OutputData::CheckTypeAndMergeFrom(
-    const ::google::protobuf::MessageLite& from) {
-  MergeFrom(*::google::protobuf::down_cast<const OutputData*>(&from));
-}
-
-void OutputData::MergeFrom(const OutputData& from) {
-  GOOGLE_CHECK_NE(&from, this);
-  test_.MergeFrom(from.test_);
-}
-
-void OutputData::CopyFrom(const OutputData& from) {
-  if (&from == this) return;
-  Clear();
-  MergeFrom(from);
-}
-
-bool OutputData::IsInitialized() const {
-  
-  return true;
-}
-
-void OutputData::Swap(OutputData* other) {
-  if (other != this) {
-    test_.Swap(&other->test_);
-    std::swap(_has_bits_[0], other->_has_bits_[0]);
-    std::swap(_cached_size_, other->_cached_size_);
-  }
-}
-
-::std::string OutputData::GetTypeName() const {
-  return "audio_processing_unittest.OutputData";
-}
-
-
-// @@protoc_insertion_point(namespace_scope)
-
-}  // namespace audio_processing_unittest
-
-// @@protoc_insertion_point(global_scope)
diff --git a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.h b/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.h
deleted file mode 100644
index 34c21b2..0000000
--- a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.pb.h
+++ /dev/null
@@ -1,862 +0,0 @@
-// Generated by the protocol buffer compiler.  DO NOT EDIT!
-// source: audio_processing_unittest.proto
-
-#ifndef PROTOBUF_audio_5fprocessing_5funittest_2eproto__INCLUDED
-#define PROTOBUF_audio_5fprocessing_5funittest_2eproto__INCLUDED
-
-#include <string>
-
-#include <google/protobuf/stubs/common.h>
-
-#if GOOGLE_PROTOBUF_VERSION < 2004000
-#error This file was generated by a newer version of protoc which is
-#error incompatible with your Protocol Buffer headers.  Please update
-#error your headers.
-#endif
-#if 2004000 < GOOGLE_PROTOBUF_MIN_PROTOC_VERSION
-#error This file was generated by an older version of protoc which is
-#error incompatible with your Protocol Buffer headers.  Please
-#error regenerate this file with a newer version of protoc.
-#endif
-
-#include <google/protobuf/generated_message_util.h>
-#include <google/protobuf/repeated_field.h>
-#include <google/protobuf/extension_set.h>
-// @@protoc_insertion_point(includes)
-
-namespace audio_processing_unittest {
-
-// Internal implementation detail -- do not call these.
-void  protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
-void protobuf_AssignDesc_audio_5fprocessing_5funittest_2eproto();
-void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto();
-
-class Test;
-class Test_Statistic;
-class Test_EchoMetrics;
-class OutputData;
-
-// ===================================================================
-
-class Test_Statistic : public ::google::protobuf::MessageLite {
- public:
-  Test_Statistic();
-  virtual ~Test_Statistic();
-  
-  Test_Statistic(const Test_Statistic& from);
-  
-  inline Test_Statistic& operator=(const Test_Statistic& from) {
-    CopyFrom(from);
-    return *this;
-  }
-  
-  static const Test_Statistic& default_instance();
-  
-  void Swap(Test_Statistic* other);
-  
-  // implements Message ----------------------------------------------
-  
-  Test_Statistic* New() const;
-  void CheckTypeAndMergeFrom(const ::google::protobuf::MessageLite& from);
-  void CopyFrom(const Test_Statistic& from);
-  void MergeFrom(const Test_Statistic& from);
-  void Clear();
-  bool IsInitialized() const;
-  
-  int ByteSize() const;
-  bool MergePartialFromCodedStream(
-      ::google::protobuf::io::CodedInputStream* input);
-  void SerializeWithCachedSizes(
-      ::google::protobuf::io::CodedOutputStream* output) const;
-  int GetCachedSize() const { return _cached_size_; }
-  private:
-  void SharedCtor();
-  void SharedDtor();
-  void SetCachedSize(int size) const;
-  public:
-  
-  ::std::string GetTypeName() const;
-  
-  // nested types ----------------------------------------------------
-  
-  // accessors -------------------------------------------------------
-  
-  // optional int32 instant = 1;
-  inline bool has_instant() const;
-  inline void clear_instant();
-  static const int kInstantFieldNumber = 1;
-  inline ::google::protobuf::int32 instant() const;
-  inline void set_instant(::google::protobuf::int32 value);
-  
-  // optional int32 average = 2;
-  inline bool has_average() const;
-  inline void clear_average();
-  static const int kAverageFieldNumber = 2;
-  inline ::google::protobuf::int32 average() const;
-  inline void set_average(::google::protobuf::int32 value);
-  
-  // optional int32 maximum = 3;
-  inline bool has_maximum() const;
-  inline void clear_maximum();
-  static const int kMaximumFieldNumber = 3;
-  inline ::google::protobuf::int32 maximum() const;
-  inline void set_maximum(::google::protobuf::int32 value);
-  
-  // optional int32 minimum = 4;
-  inline bool has_minimum() const;
-  inline void clear_minimum();
-  static const int kMinimumFieldNumber = 4;
-  inline ::google::protobuf::int32 minimum() const;
-  inline void set_minimum(::google::protobuf::int32 value);
-  
-  // @@protoc_insertion_point(class_scope:audio_processing_unittest.Test.Statistic)
- private:
-  inline void set_has_instant();
-  inline void clear_has_instant();
-  inline void set_has_average();
-  inline void clear_has_average();
-  inline void set_has_maximum();
-  inline void clear_has_maximum();
-  inline void set_has_minimum();
-  inline void clear_has_minimum();
-  
-  ::google::protobuf::int32 instant_;
-  ::google::protobuf::int32 average_;
-  ::google::protobuf::int32 maximum_;
-  ::google::protobuf::int32 minimum_;
-  
-  mutable int _cached_size_;
-  ::google::protobuf::uint32 _has_bits_[(4 + 31) / 32];
-  
-  friend void  protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
-  friend void protobuf_AssignDesc_audio_5fprocessing_5funittest_2eproto();
-  friend void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto();
-  
-  void InitAsDefaultInstance();
-  static Test_Statistic* default_instance_;
-};
-// -------------------------------------------------------------------
-
-class Test_EchoMetrics : public ::google::protobuf::MessageLite {
- public:
-  Test_EchoMetrics();
-  virtual ~Test_EchoMetrics();
-  
-  Test_EchoMetrics(const Test_EchoMetrics& from);
-  
-  inline Test_EchoMetrics& operator=(const Test_EchoMetrics& from) {
-    CopyFrom(from);
-    return *this;
-  }
-  
-  static const Test_EchoMetrics& default_instance();
-  
-  void Swap(Test_EchoMetrics* other);
-  
-  // implements Message ----------------------------------------------
-  
-  Test_EchoMetrics* New() const;
-  void CheckTypeAndMergeFrom(const ::google::protobuf::MessageLite& from);
-  void CopyFrom(const Test_EchoMetrics& from);
-  void MergeFrom(const Test_EchoMetrics& from);
-  void Clear();
-  bool IsInitialized() const;
-  
-  int ByteSize() const;
-  bool MergePartialFromCodedStream(
-      ::google::protobuf::io::CodedInputStream* input);
-  void SerializeWithCachedSizes(
-      ::google::protobuf::io::CodedOutputStream* output) const;
-  int GetCachedSize() const { return _cached_size_; }
-  private:
-  void SharedCtor();
-  void SharedDtor();
-  void SetCachedSize(int size) const;
-  public:
-  
-  ::std::string GetTypeName() const;
-  
-  // nested types ----------------------------------------------------
-  
-  // accessors -------------------------------------------------------
-  
-  // optional .audio_processing_unittest.Test.Statistic residualEchoReturnLoss = 1;
-  inline bool has_residualechoreturnloss() const;
-  inline void clear_residualechoreturnloss();
-  static const int kResidualEchoReturnLossFieldNumber = 1;
-  inline const ::audio_processing_unittest::Test_Statistic& residualechoreturnloss() const;
-  inline ::audio_processing_unittest::Test_Statistic* mutable_residualechoreturnloss();
-  inline ::audio_processing_unittest::Test_Statistic* release_residualechoreturnloss();
-  
-  // optional .audio_processing_unittest.Test.Statistic echoReturnLoss = 2;
-  inline bool has_echoreturnloss() const;
-  inline void clear_echoreturnloss();
-  static const int kEchoReturnLossFieldNumber = 2;
-  inline const ::audio_processing_unittest::Test_Statistic& echoreturnloss() const;
-  inline ::audio_processing_unittest::Test_Statistic* mutable_echoreturnloss();
-  inline ::audio_processing_unittest::Test_Statistic* release_echoreturnloss();
-  
-  // optional .audio_processing_unittest.Test.Statistic echoReturnLossEnhancement = 3;
-  inline bool has_echoreturnlossenhancement() const;
-  inline void clear_echoreturnlossenhancement();
-  static const int kEchoReturnLossEnhancementFieldNumber = 3;
-  inline const ::audio_processing_unittest::Test_Statistic& echoreturnlossenhancement() const;
-  inline ::audio_processing_unittest::Test_Statistic* mutable_echoreturnlossenhancement();
-  inline ::audio_processing_unittest::Test_Statistic* release_echoreturnlossenhancement();
-  
-  // optional .audio_processing_unittest.Test.Statistic aNlp = 4;
-  inline bool has_anlp() const;
-  inline void clear_anlp();
-  static const int kANlpFieldNumber = 4;
-  inline const ::audio_processing_unittest::Test_Statistic& anlp() const;
-  inline ::audio_processing_unittest::Test_Statistic* mutable_anlp();
-  inline ::audio_processing_unittest::Test_Statistic* release_anlp();
-  
-  // @@protoc_insertion_point(class_scope:audio_processing_unittest.Test.EchoMetrics)
- private:
-  inline void set_has_residualechoreturnloss();
-  inline void clear_has_residualechoreturnloss();
-  inline void set_has_echoreturnloss();
-  inline void clear_has_echoreturnloss();
-  inline void set_has_echoreturnlossenhancement();
-  inline void clear_has_echoreturnlossenhancement();
-  inline void set_has_anlp();
-  inline void clear_has_anlp();
-  
-  ::audio_processing_unittest::Test_Statistic* residualechoreturnloss_;
-  ::audio_processing_unittest::Test_Statistic* echoreturnloss_;
-  ::audio_processing_unittest::Test_Statistic* echoreturnlossenhancement_;
-  ::audio_processing_unittest::Test_Statistic* anlp_;
-  
-  mutable int _cached_size_;
-  ::google::protobuf::uint32 _has_bits_[(4 + 31) / 32];
-  
-  friend void  protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
-  friend void protobuf_AssignDesc_audio_5fprocessing_5funittest_2eproto();
-  friend void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto();
-  
-  void InitAsDefaultInstance();
-  static Test_EchoMetrics* default_instance_;
-};
-// -------------------------------------------------------------------
-
-class Test : public ::google::protobuf::MessageLite {
- public:
-  Test();
-  virtual ~Test();
-  
-  Test(const Test& from);
-  
-  inline Test& operator=(const Test& from) {
-    CopyFrom(from);
-    return *this;
-  }
-  
-  static const Test& default_instance();
-  
-  void Swap(Test* other);
-  
-  // implements Message ----------------------------------------------
-  
-  Test* New() const;
-  void CheckTypeAndMergeFrom(const ::google::protobuf::MessageLite& from);
-  void CopyFrom(const Test& from);
-  void MergeFrom(const Test& from);
-  void Clear();
-  bool IsInitialized() const;
-  
-  int ByteSize() const;
-  bool MergePartialFromCodedStream(
-      ::google::protobuf::io::CodedInputStream* input);
-  void SerializeWithCachedSizes(
-      ::google::protobuf::io::CodedOutputStream* output) const;
-  int GetCachedSize() const { return _cached_size_; }
-  private:
-  void SharedCtor();
-  void SharedDtor();
-  void SetCachedSize(int size) const;
-  public:
-  
-  ::std::string GetTypeName() const;
-  
-  // nested types ----------------------------------------------------
-  
-  typedef Test_Statistic Statistic;
-  typedef Test_EchoMetrics EchoMetrics;
-  
-  // accessors -------------------------------------------------------
-  
-  // optional int32 numReverseChannels = 1;
-  inline bool has_numreversechannels() const;
-  inline void clear_numreversechannels();
-  static const int kNumReverseChannelsFieldNumber = 1;
-  inline ::google::protobuf::int32 numreversechannels() const;
-  inline void set_numreversechannels(::google::protobuf::int32 value);
-  
-  // optional int32 numChannels = 2;
-  inline bool has_numchannels() const;
-  inline void clear_numchannels();
-  static const int kNumChannelsFieldNumber = 2;
-  inline ::google::protobuf::int32 numchannels() const;
-  inline void set_numchannels(::google::protobuf::int32 value);
-  
-  // optional int32 sampleRate = 3;
-  inline bool has_samplerate() const;
-  inline void clear_samplerate();
-  static const int kSampleRateFieldNumber = 3;
-  inline ::google::protobuf::int32 samplerate() const;
-  inline void set_samplerate(::google::protobuf::int32 value);
-  
-  // optional int32 hasEchoCount = 4;
-  inline bool has_hasechocount() const;
-  inline void clear_hasechocount();
-  static const int kHasEchoCountFieldNumber = 4;
-  inline ::google::protobuf::int32 hasechocount() const;
-  inline void set_hasechocount(::google::protobuf::int32 value);
-  
-  // optional int32 hasVoiceCount = 5;
-  inline bool has_hasvoicecount() const;
-  inline void clear_hasvoicecount();
-  static const int kHasVoiceCountFieldNumber = 5;
-  inline ::google::protobuf::int32 hasvoicecount() const;
-  inline void set_hasvoicecount(::google::protobuf::int32 value);
-  
-  // optional int32 isSaturatedCount = 6;
-  inline bool has_issaturatedcount() const;
-  inline void clear_issaturatedcount();
-  static const int kIsSaturatedCountFieldNumber = 6;
-  inline ::google::protobuf::int32 issaturatedcount() const;
-  inline void set_issaturatedcount(::google::protobuf::int32 value);
-  
-  // optional .audio_processing_unittest.Test.EchoMetrics echoMetrics = 7;
-  inline bool has_echometrics() const;
-  inline void clear_echometrics();
-  static const int kEchoMetricsFieldNumber = 7;
-  inline const ::audio_processing_unittest::Test_EchoMetrics& echometrics() const;
-  inline ::audio_processing_unittest::Test_EchoMetrics* mutable_echometrics();
-  inline ::audio_processing_unittest::Test_EchoMetrics* release_echometrics();
-  
-  // @@protoc_insertion_point(class_scope:audio_processing_unittest.Test)
- private:
-  inline void set_has_numreversechannels();
-  inline void clear_has_numreversechannels();
-  inline void set_has_numchannels();
-  inline void clear_has_numchannels();
-  inline void set_has_samplerate();
-  inline void clear_has_samplerate();
-  inline void set_has_hasechocount();
-  inline void clear_has_hasechocount();
-  inline void set_has_hasvoicecount();
-  inline void clear_has_hasvoicecount();
-  inline void set_has_issaturatedcount();
-  inline void clear_has_issaturatedcount();
-  inline void set_has_echometrics();
-  inline void clear_has_echometrics();
-  
-  ::google::protobuf::int32 numreversechannels_;
-  ::google::protobuf::int32 numchannels_;
-  ::google::protobuf::int32 samplerate_;
-  ::google::protobuf::int32 hasechocount_;
-  ::google::protobuf::int32 hasvoicecount_;
-  ::google::protobuf::int32 issaturatedcount_;
-  ::audio_processing_unittest::Test_EchoMetrics* echometrics_;
-  
-  mutable int _cached_size_;
-  ::google::protobuf::uint32 _has_bits_[(7 + 31) / 32];
-  
-  friend void  protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
-  friend void protobuf_AssignDesc_audio_5fprocessing_5funittest_2eproto();
-  friend void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto();
-  
-  void InitAsDefaultInstance();
-  static Test* default_instance_;
-};
-// -------------------------------------------------------------------
-
-class OutputData : public ::google::protobuf::MessageLite {
- public:
-  OutputData();
-  virtual ~OutputData();
-  
-  OutputData(const OutputData& from);
-  
-  inline OutputData& operator=(const OutputData& from) {
-    CopyFrom(from);
-    return *this;
-  }
-  
-  static const OutputData& default_instance();
-  
-  void Swap(OutputData* other);
-  
-  // implements Message ----------------------------------------------
-  
-  OutputData* New() const;
-  void CheckTypeAndMergeFrom(const ::google::protobuf::MessageLite& from);
-  void CopyFrom(const OutputData& from);
-  void MergeFrom(const OutputData& from);
-  void Clear();
-  bool IsInitialized() const;
-  
-  int ByteSize() const;
-  bool MergePartialFromCodedStream(
-      ::google::protobuf::io::CodedInputStream* input);
-  void SerializeWithCachedSizes(
-      ::google::protobuf::io::CodedOutputStream* output) const;
-  int GetCachedSize() const { return _cached_size_; }
-  private:
-  void SharedCtor();
-  void SharedDtor();
-  void SetCachedSize(int size) const;
-  public:
-  
-  ::std::string GetTypeName() const;
-  
-  // nested types ----------------------------------------------------
-  
-  // accessors -------------------------------------------------------
-  
-  // repeated .audio_processing_unittest.Test test = 1;
-  inline int test_size() const;
-  inline void clear_test();
-  static const int kTestFieldNumber = 1;
-  inline const ::audio_processing_unittest::Test& test(int index) const;
-  inline ::audio_processing_unittest::Test* mutable_test(int index);
-  inline ::audio_processing_unittest::Test* add_test();
-  inline const ::google::protobuf::RepeatedPtrField< ::audio_processing_unittest::Test >&
-      test() const;
-  inline ::google::protobuf::RepeatedPtrField< ::audio_processing_unittest::Test >*
-      mutable_test();
-  
-  // @@protoc_insertion_point(class_scope:audio_processing_unittest.OutputData)
- private:
-  
-  ::google::protobuf::RepeatedPtrField< ::audio_processing_unittest::Test > test_;
-  
-  mutable int _cached_size_;
-  ::google::protobuf::uint32 _has_bits_[(1 + 31) / 32];
-  
-  friend void  protobuf_AddDesc_audio_5fprocessing_5funittest_2eproto();
-  friend void protobuf_AssignDesc_audio_5fprocessing_5funittest_2eproto();
-  friend void protobuf_ShutdownFile_audio_5fprocessing_5funittest_2eproto();
-  
-  void InitAsDefaultInstance();
-  static OutputData* default_instance_;
-};
-// ===================================================================
-
-
-// ===================================================================
-
-// Test_Statistic
-
-// optional int32 instant = 1;
-inline bool Test_Statistic::has_instant() const {
-  return (_has_bits_[0] & 0x00000001u) != 0;
-}
-inline void Test_Statistic::set_has_instant() {
-  _has_bits_[0] |= 0x00000001u;
-}
-inline void Test_Statistic::clear_has_instant() {
-  _has_bits_[0] &= ~0x00000001u;
-}
-inline void Test_Statistic::clear_instant() {
-  instant_ = 0;
-  clear_has_instant();
-}
-inline ::google::protobuf::int32 Test_Statistic::instant() const {
-  return instant_;
-}
-inline void Test_Statistic::set_instant(::google::protobuf::int32 value) {
-  set_has_instant();
-  instant_ = value;
-}
-
-// optional int32 average = 2;
-inline bool Test_Statistic::has_average() const {
-  return (_has_bits_[0] & 0x00000002u) != 0;
-}
-inline void Test_Statistic::set_has_average() {
-  _has_bits_[0] |= 0x00000002u;
-}
-inline void Test_Statistic::clear_has_average() {
-  _has_bits_[0] &= ~0x00000002u;
-}
-inline void Test_Statistic::clear_average() {
-  average_ = 0;
-  clear_has_average();
-}
-inline ::google::protobuf::int32 Test_Statistic::average() const {
-  return average_;
-}
-inline void Test_Statistic::set_average(::google::protobuf::int32 value) {
-  set_has_average();
-  average_ = value;
-}
-
-// optional int32 maximum = 3;
-inline bool Test_Statistic::has_maximum() const {
-  return (_has_bits_[0] & 0x00000004u) != 0;
-}
-inline void Test_Statistic::set_has_maximum() {
-  _has_bits_[0] |= 0x00000004u;
-}
-inline void Test_Statistic::clear_has_maximum() {
-  _has_bits_[0] &= ~0x00000004u;
-}
-inline void Test_Statistic::clear_maximum() {
-  maximum_ = 0;
-  clear_has_maximum();
-}
-inline ::google::protobuf::int32 Test_Statistic::maximum() const {
-  return maximum_;
-}
-inline void Test_Statistic::set_maximum(::google::protobuf::int32 value) {
-  set_has_maximum();
-  maximum_ = value;
-}
-
-// optional int32 minimum = 4;
-inline bool Test_Statistic::has_minimum() const {
-  return (_has_bits_[0] & 0x00000008u) != 0;
-}
-inline void Test_Statistic::set_has_minimum() {
-  _has_bits_[0] |= 0x00000008u;
-}
-inline void Test_Statistic::clear_has_minimum() {
-  _has_bits_[0] &= ~0x00000008u;
-}
-inline void Test_Statistic::clear_minimum() {
-  minimum_ = 0;
-  clear_has_minimum();
-}
-inline ::google::protobuf::int32 Test_Statistic::minimum() const {
-  return minimum_;
-}
-inline void Test_Statistic::set_minimum(::google::protobuf::int32 value) {
-  set_has_minimum();
-  minimum_ = value;
-}
-
-// -------------------------------------------------------------------
-
-// Test_EchoMetrics
-
-// optional .audio_processing_unittest.Test.Statistic residualEchoReturnLoss = 1;
-inline bool Test_EchoMetrics::has_residualechoreturnloss() const {
-  return (_has_bits_[0] & 0x00000001u) != 0;
-}
-inline void Test_EchoMetrics::set_has_residualechoreturnloss() {
-  _has_bits_[0] |= 0x00000001u;
-}
-inline void Test_EchoMetrics::clear_has_residualechoreturnloss() {
-  _has_bits_[0] &= ~0x00000001u;
-}
-inline void Test_EchoMetrics::clear_residualechoreturnloss() {
-  if (residualechoreturnloss_ != NULL) residualechoreturnloss_->::audio_processing_unittest::Test_Statistic::Clear();
-  clear_has_residualechoreturnloss();
-}
-inline const ::audio_processing_unittest::Test_Statistic& Test_EchoMetrics::residualechoreturnloss() const {
-  return residualechoreturnloss_ != NULL ? *residualechoreturnloss_ : *default_instance_->residualechoreturnloss_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::mutable_residualechoreturnloss() {
-  set_has_residualechoreturnloss();
-  if (residualechoreturnloss_ == NULL) residualechoreturnloss_ = new ::audio_processing_unittest::Test_Statistic;
-  return residualechoreturnloss_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::release_residualechoreturnloss() {
-  clear_has_residualechoreturnloss();
-  ::audio_processing_unittest::Test_Statistic* temp = residualechoreturnloss_;
-  residualechoreturnloss_ = NULL;
-  return temp;
-}
-
-// optional .audio_processing_unittest.Test.Statistic echoReturnLoss = 2;
-inline bool Test_EchoMetrics::has_echoreturnloss() const {
-  return (_has_bits_[0] & 0x00000002u) != 0;
-}
-inline void Test_EchoMetrics::set_has_echoreturnloss() {
-  _has_bits_[0] |= 0x00000002u;
-}
-inline void Test_EchoMetrics::clear_has_echoreturnloss() {
-  _has_bits_[0] &= ~0x00000002u;
-}
-inline void Test_EchoMetrics::clear_echoreturnloss() {
-  if (echoreturnloss_ != NULL) echoreturnloss_->::audio_processing_unittest::Test_Statistic::Clear();
-  clear_has_echoreturnloss();
-}
-inline const ::audio_processing_unittest::Test_Statistic& Test_EchoMetrics::echoreturnloss() const {
-  return echoreturnloss_ != NULL ? *echoreturnloss_ : *default_instance_->echoreturnloss_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::mutable_echoreturnloss() {
-  set_has_echoreturnloss();
-  if (echoreturnloss_ == NULL) echoreturnloss_ = new ::audio_processing_unittest::Test_Statistic;
-  return echoreturnloss_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::release_echoreturnloss() {
-  clear_has_echoreturnloss();
-  ::audio_processing_unittest::Test_Statistic* temp = echoreturnloss_;
-  echoreturnloss_ = NULL;
-  return temp;
-}
-
-// optional .audio_processing_unittest.Test.Statistic echoReturnLossEnhancement = 3;
-inline bool Test_EchoMetrics::has_echoreturnlossenhancement() const {
-  return (_has_bits_[0] & 0x00000004u) != 0;
-}
-inline void Test_EchoMetrics::set_has_echoreturnlossenhancement() {
-  _has_bits_[0] |= 0x00000004u;
-}
-inline void Test_EchoMetrics::clear_has_echoreturnlossenhancement() {
-  _has_bits_[0] &= ~0x00000004u;
-}
-inline void Test_EchoMetrics::clear_echoreturnlossenhancement() {
-  if (echoreturnlossenhancement_ != NULL) echoreturnlossenhancement_->::audio_processing_unittest::Test_Statistic::Clear();
-  clear_has_echoreturnlossenhancement();
-}
-inline const ::audio_processing_unittest::Test_Statistic& Test_EchoMetrics::echoreturnlossenhancement() const {
-  return echoreturnlossenhancement_ != NULL ? *echoreturnlossenhancement_ : *default_instance_->echoreturnlossenhancement_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::mutable_echoreturnlossenhancement() {
-  set_has_echoreturnlossenhancement();
-  if (echoreturnlossenhancement_ == NULL) echoreturnlossenhancement_ = new ::audio_processing_unittest::Test_Statistic;
-  return echoreturnlossenhancement_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::release_echoreturnlossenhancement() {
-  clear_has_echoreturnlossenhancement();
-  ::audio_processing_unittest::Test_Statistic* temp = echoreturnlossenhancement_;
-  echoreturnlossenhancement_ = NULL;
-  return temp;
-}
-
-// optional .audio_processing_unittest.Test.Statistic aNlp = 4;
-inline bool Test_EchoMetrics::has_anlp() const {
-  return (_has_bits_[0] & 0x00000008u) != 0;
-}
-inline void Test_EchoMetrics::set_has_anlp() {
-  _has_bits_[0] |= 0x00000008u;
-}
-inline void Test_EchoMetrics::clear_has_anlp() {
-  _has_bits_[0] &= ~0x00000008u;
-}
-inline void Test_EchoMetrics::clear_anlp() {
-  if (anlp_ != NULL) anlp_->::audio_processing_unittest::Test_Statistic::Clear();
-  clear_has_anlp();
-}
-inline const ::audio_processing_unittest::Test_Statistic& Test_EchoMetrics::anlp() const {
-  return anlp_ != NULL ? *anlp_ : *default_instance_->anlp_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::mutable_anlp() {
-  set_has_anlp();
-  if (anlp_ == NULL) anlp_ = new ::audio_processing_unittest::Test_Statistic;
-  return anlp_;
-}
-inline ::audio_processing_unittest::Test_Statistic* Test_EchoMetrics::release_anlp() {
-  clear_has_anlp();
-  ::audio_processing_unittest::Test_Statistic* temp = anlp_;
-  anlp_ = NULL;
-  return temp;
-}
-
-// -------------------------------------------------------------------
-
-// Test
-
-// optional int32 numReverseChannels = 1;
-inline bool Test::has_numreversechannels() const {
-  return (_has_bits_[0] & 0x00000001u) != 0;
-}
-inline void Test::set_has_numreversechannels() {
-  _has_bits_[0] |= 0x00000001u;
-}
-inline void Test::clear_has_numreversechannels() {
-  _has_bits_[0] &= ~0x00000001u;
-}
-inline void Test::clear_numreversechannels() {
-  numreversechannels_ = 0;
-  clear_has_numreversechannels();
-}
-inline ::google::protobuf::int32 Test::numreversechannels() const {
-  return numreversechannels_;
-}
-inline void Test::set_numreversechannels(::google::protobuf::int32 value) {
-  set_has_numreversechannels();
-  numreversechannels_ = value;
-}
-
-// optional int32 numChannels = 2;
-inline bool Test::has_numchannels() const {
-  return (_has_bits_[0] & 0x00000002u) != 0;
-}
-inline void Test::set_has_numchannels() {
-  _has_bits_[0] |= 0x00000002u;
-}
-inline void Test::clear_has_numchannels() {
-  _has_bits_[0] &= ~0x00000002u;
-}
-inline void Test::clear_numchannels() {
-  numchannels_ = 0;
-  clear_has_numchannels();
-}
-inline ::google::protobuf::int32 Test::numchannels() const {
-  return numchannels_;
-}
-inline void Test::set_numchannels(::google::protobuf::int32 value) {
-  set_has_numchannels();
-  numchannels_ = value;
-}
-
-// optional int32 sampleRate = 3;
-inline bool Test::has_samplerate() const {
-  return (_has_bits_[0] & 0x00000004u) != 0;
-}
-inline void Test::set_has_samplerate() {
-  _has_bits_[0] |= 0x00000004u;
-}
-inline void Test::clear_has_samplerate() {
-  _has_bits_[0] &= ~0x00000004u;
-}
-inline void Test::clear_samplerate() {
-  samplerate_ = 0;
-  clear_has_samplerate();
-}
-inline ::google::protobuf::int32 Test::samplerate() const {
-  return samplerate_;
-}
-inline void Test::set_samplerate(::google::protobuf::int32 value) {
-  set_has_samplerate();
-  samplerate_ = value;
-}
-
-// optional int32 hasEchoCount = 4;
-inline bool Test::has_hasechocount() const {
-  return (_has_bits_[0] & 0x00000008u) != 0;
-}
-inline void Test::set_has_hasechocount() {
-  _has_bits_[0] |= 0x00000008u;
-}
-inline void Test::clear_has_hasechocount() {
-  _has_bits_[0] &= ~0x00000008u;
-}
-inline void Test::clear_hasechocount() {
-  hasechocount_ = 0;
-  clear_has_hasechocount();
-}
-inline ::google::protobuf::int32 Test::hasechocount() const {
-  return hasechocount_;
-}
-inline void Test::set_hasechocount(::google::protobuf::int32 value) {
-  set_has_hasechocount();
-  hasechocount_ = value;
-}
-
-// optional int32 hasVoiceCount = 5;
-inline bool Test::has_hasvoicecount() const {
-  return (_has_bits_[0] & 0x00000010u) != 0;
-}
-inline void Test::set_has_hasvoicecount() {
-  _has_bits_[0] |= 0x00000010u;
-}
-inline void Test::clear_has_hasvoicecount() {
-  _has_bits_[0] &= ~0x00000010u;
-}
-inline void Test::clear_hasvoicecount() {
-  hasvoicecount_ = 0;
-  clear_has_hasvoicecount();
-}
-inline ::google::protobuf::int32 Test::hasvoicecount() const {
-  return hasvoicecount_;
-}
-inline void Test::set_hasvoicecount(::google::protobuf::int32 value) {
-  set_has_hasvoicecount();
-  hasvoicecount_ = value;
-}
-
-// optional int32 isSaturatedCount = 6;
-inline bool Test::has_issaturatedcount() const {
-  return (_has_bits_[0] & 0x00000020u) != 0;
-}
-inline void Test::set_has_issaturatedcount() {
-  _has_bits_[0] |= 0x00000020u;
-}
-inline void Test::clear_has_issaturatedcount() {
-  _has_bits_[0] &= ~0x00000020u;
-}
-inline void Test::clear_issaturatedcount() {
-  issaturatedcount_ = 0;
-  clear_has_issaturatedcount();
-}
-inline ::google::protobuf::int32 Test::issaturatedcount() const {
-  return issaturatedcount_;
-}
-inline void Test::set_issaturatedcount(::google::protobuf::int32 value) {
-  set_has_issaturatedcount();
-  issaturatedcount_ = value;
-}
-
-// optional .audio_processing_unittest.Test.EchoMetrics echoMetrics = 7;
-inline bool Test::has_echometrics() const {
-  return (_has_bits_[0] & 0x00000040u) != 0;
-}
-inline void Test::set_has_echometrics() {
-  _has_bits_[0] |= 0x00000040u;
-}
-inline void Test::clear_has_echometrics() {
-  _has_bits_[0] &= ~0x00000040u;
-}
-inline void Test::clear_echometrics() {
-  if (echometrics_ != NULL) echometrics_->::audio_processing_unittest::Test_EchoMetrics::Clear();
-  clear_has_echometrics();
-}
-inline const ::audio_processing_unittest::Test_EchoMetrics& Test::echometrics() const {
-  return echometrics_ != NULL ? *echometrics_ : *default_instance_->echometrics_;
-}
-inline ::audio_processing_unittest::Test_EchoMetrics* Test::mutable_echometrics() {
-  set_has_echometrics();
-  if (echometrics_ == NULL) echometrics_ = new ::audio_processing_unittest::Test_EchoMetrics;
-  return echometrics_;
-}
-inline ::audio_processing_unittest::Test_EchoMetrics* Test::release_echometrics() {
-  clear_has_echometrics();
-  ::audio_processing_unittest::Test_EchoMetrics* temp = echometrics_;
-  echometrics_ = NULL;
-  return temp;
-}
-
-// -------------------------------------------------------------------
-
-// OutputData
-
-// repeated .audio_processing_unittest.Test test = 1;
-inline int OutputData::test_size() const {
-  return test_.size();
-}
-inline void OutputData::clear_test() {
-  test_.Clear();
-}
-inline const ::audio_processing_unittest::Test& OutputData::test(int index) const {
-  return test_.Get(index);
-}
-inline ::audio_processing_unittest::Test* OutputData::mutable_test(int index) {
-  return test_.Mutable(index);
-}
-inline ::audio_processing_unittest::Test* OutputData::add_test() {
-  return test_.Add();
-}
-inline const ::google::protobuf::RepeatedPtrField< ::audio_processing_unittest::Test >&
-OutputData::test() const {
-  return test_;
-}
-inline ::google::protobuf::RepeatedPtrField< ::audio_processing_unittest::Test >*
-OutputData::mutable_test() {
-  return &test_;
-}
-
-
-// @@protoc_insertion_point(namespace_scope)
-
-}  // namespace audio_processing_unittest
-
-// @@protoc_insertion_point(global_scope)
-
-#endif  // PROTOBUF_audio_5fprocessing_5funittest_2eproto__INCLUDED
diff --git a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.proto b/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.proto
deleted file mode 100644
index 8520e64..0000000
--- a/src/modules/audio_processing/main/test/unit_test/audio_processing_unittest.proto
+++ /dev/null
@@ -1,33 +0,0 @@
-package audio_processing_unittest;
-option optimize_for = LITE_RUNTIME;
-
-message Test {
-  optional int32 numReverseChannels = 1;
-  optional int32 numChannels = 2;
-  optional int32 sampleRate = 3;
-
-  optional int32 hasEchoCount = 4;
-  optional int32 hasVoiceCount = 5;
-  optional int32 isSaturatedCount = 6;
-
-  message Statistic {
-    optional int32 instant = 1;
-    optional int32 average = 2;
-    optional int32 maximum = 3;
-    optional int32 minimum = 4;
-  }
-
-  message EchoMetrics {
-    optional Statistic residualEchoReturnLoss = 1;
-    optional Statistic echoReturnLoss = 2;
-    optional Statistic echoReturnLossEnhancement = 3;
-    optional Statistic aNlp = 4;
-  }
-
-  optional EchoMetrics echoMetrics = 7;
-}
-
-message OutputData {
-  repeated Test test = 1;
-}
-
diff --git a/src/modules/audio_processing/main/test/unit_test/unit_test.cc b/src/modules/audio_processing/main/test/unit_test/unit_test.cc
deleted file mode 100644
index 3a6fce5..0000000
--- a/src/modules/audio_processing/main/test/unit_test/unit_test.cc
+++ /dev/null
@@ -1,881 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstdio>
-
-#include <gtest/gtest.h>
-
-#include "audio_processing.h"
-#include "audio_processing_unittest.pb.h"
-#include "event_wrapper.h"
-#include "module_common_types.h"
-#include "thread_wrapper.h"
-#include "trace.h"
-#include "signal_processing_library.h"
-
-using webrtc::AudioProcessing;
-using webrtc::AudioFrame;
-using webrtc::GainControl;
-using webrtc::NoiseSuppression;
-using webrtc::EchoCancellation;
-using webrtc::EventWrapper;
-using webrtc::Trace;
-using webrtc::LevelEstimator;
-using webrtc::EchoCancellation;
-using webrtc::EchoControlMobile;
-using webrtc::VoiceDetection;
-
-namespace {
-// When true, this will compare the output data with the results stored to
-// file. This is the typical case. When the file should be updated, it can
-// be set to false with the command-line switch --write_output_data.
-bool global_read_output_data = true;
-
-class ApmEnvironment : public ::testing::Environment {
- public:
-  virtual void SetUp() {
-    Trace::CreateTrace();
-    ASSERT_EQ(0, Trace::SetTraceFile("apm_trace.txt"));
-  }
-
-  virtual void TearDown() {
-    Trace::ReturnTrace();
-  }
-};
-
-class ApmTest : public ::testing::Test {
- protected:
-  ApmTest();
-  virtual void SetUp();
-  virtual void TearDown();
-
-  webrtc::AudioProcessing* apm_;
-  webrtc::AudioFrame* frame_;
-  webrtc::AudioFrame* revframe_;
-  FILE* far_file_;
-  FILE* near_file_;
-  bool update_output_data_;
-};
-
-ApmTest::ApmTest()
-    : apm_(NULL),
-      far_file_(NULL),
-      near_file_(NULL),
-      frame_(NULL),
-      revframe_(NULL) {}
-
-void ApmTest::SetUp() {
-  apm_ = AudioProcessing::Create(0);
-  ASSERT_TRUE(apm_ != NULL);
-
-  frame_ = new AudioFrame();
-  revframe_ = new AudioFrame();
-
-  ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
-  ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(2, 2));
-  ASSERT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(2));
-
-  frame_->_payloadDataLengthInSamples = 320;
-  frame_->_audioChannel = 2;
-  frame_->_frequencyInHz = 32000;
-  revframe_->_payloadDataLengthInSamples = 320;
-  revframe_->_audioChannel = 2;
-  revframe_->_frequencyInHz = 32000;
-
-  far_file_ = fopen("aec_far.pcm", "rb");
-  ASSERT_TRUE(far_file_ != NULL) << "Could not open input file aec_far.pcm\n";
-  near_file_ = fopen("aec_near.pcm", "rb");
-  ASSERT_TRUE(near_file_ != NULL) << "Could not open input file aec_near.pcm\n";
-}
-
-void ApmTest::TearDown() {
-  if (frame_) {
-    delete frame_;
-  }
-  frame_ = NULL;
-
-  if (revframe_) {
-    delete revframe_;
-  }
-  revframe_ = NULL;
-
-  if (far_file_) {
-    ASSERT_EQ(0, fclose(far_file_));
-  }
-  far_file_ = NULL;
-
-  if (near_file_) {
-    ASSERT_EQ(0, fclose(near_file_));
-  }
-  near_file_ = NULL;
-
-  if (apm_ != NULL) {
-    AudioProcessing::Destroy(apm_);
-  }
-  apm_ = NULL;
-}
-
-void MixStereoToMono(WebRtc_Word16* stereo,
-                     WebRtc_Word16* mono,
-                     int numSamples) {
-  for (int i = 0; i < numSamples; i++) {
-    int int32 = (static_cast<int>(stereo[i * 2]) +
-                 static_cast<int>(stereo[i * 2 + 1])) >> 1;
-    mono[i] = static_cast<WebRtc_Word16>(int32);
-  }
-}
-
-void WriteMessageLiteToFile(const char* filename,
-                            const ::google::protobuf::MessageLite& message) {
-  assert(filename != NULL);
-
-  FILE* file = fopen(filename, "wb");
-  ASSERT_TRUE(file != NULL) << "Could not open " << filename;
-  int size = message.ByteSize();
-  ASSERT_GT(size, 0);
-  unsigned char* array = new unsigned char[size];
-  ASSERT_TRUE(message.SerializeToArray(array, size));
-
-  ASSERT_EQ(1, fwrite(&size, sizeof(int), 1, file));
-  ASSERT_EQ(size, fwrite(array, sizeof(unsigned char), size, file));
-
-  delete [] array;
-  fclose(file);
-}
-
-void ReadMessageLiteFromFile(const char* filename,
-                             ::google::protobuf::MessageLite* message) {
-  assert(filename != NULL);
-  assert(message != NULL);
-
-  FILE* file = fopen(filename, "rb");
-  ASSERT_TRUE(file != NULL) << "Could not open " << filename;
-  int size = 0;
-  ASSERT_EQ(1, fread(&size, sizeof(int), 1, file));
-  ASSERT_GT(size, 0);
-  unsigned char* array = new unsigned char[size];
-  ASSERT_EQ(size, fread(array, sizeof(unsigned char), size, file));
-
-  ASSERT_TRUE(message->ParseFromArray(array, size));
-
-  delete [] array;
-  fclose(file);
-}
-
-struct ThreadData {
-  ThreadData(int thread_num_, AudioProcessing* ap_)
-      : thread_num(thread_num_),
-        error(false),
-        ap(ap_) {}
-  int thread_num;
-  bool error;
-  AudioProcessing* ap;
-};
-
-// Don't use GTest here; non-thread-safe on Windows (as of 1.5.0).
-bool DeadlockProc(void* thread_object) {
-  ThreadData* thread_data = static_cast<ThreadData*>(thread_object);
-  AudioProcessing* ap = thread_data->ap;
-  int err = ap->kNoError;
-
-  AudioFrame primary_frame;
-  AudioFrame reverse_frame;
-  primary_frame._payloadDataLengthInSamples = 320;
-  primary_frame._audioChannel = 2;
-  primary_frame._frequencyInHz = 32000;
-  reverse_frame._payloadDataLengthInSamples = 320;
-  reverse_frame._audioChannel = 2;
-  reverse_frame._frequencyInHz = 32000;
-
-  ap->echo_cancellation()->Enable(true);
-  ap->gain_control()->Enable(true);
-  ap->high_pass_filter()->Enable(true);
-  ap->level_estimator()->Enable(true);
-  ap->noise_suppression()->Enable(true);
-  ap->voice_detection()->Enable(true);
-
-  if (thread_data->thread_num % 2 == 0) {
-    err = ap->AnalyzeReverseStream(&reverse_frame);
-    if (err != ap->kNoError) {
-      printf("Error in AnalyzeReverseStream(): %d\n", err);
-      thread_data->error = true;
-      return false;
-    }
-  }
-
-  if (thread_data->thread_num % 2 == 1) {
-    ap->set_stream_delay_ms(0);
-    ap->echo_cancellation()->set_stream_drift_samples(0);
-    ap->gain_control()->set_stream_analog_level(0);
-    err = ap->ProcessStream(&primary_frame);
-    if (err == ap->kStreamParameterNotSetError) {
-      printf("Expected kStreamParameterNotSetError in ProcessStream(): %d\n",
-          err);
-    } else if (err != ap->kNoError) {
-      printf("Error in ProcessStream(): %d\n", err);
-      thread_data->error = true;
-      return false;
-    }
-    ap->gain_control()->stream_analog_level();
-  }
-
-  EventWrapper* event = EventWrapper::Create();
-  event->Wait(1);
-  delete event;
-  event = NULL;
-
-  return true;
-}
-
-/*TEST_F(ApmTest, Deadlock) {
-  const int num_threads = 16;
-  std::vector<ThreadWrapper*> threads(num_threads);
-  std::vector<ThreadData*> thread_data(num_threads);
-
-  ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
-  ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(2, 2));
-  ASSERT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(2));
-
-  for (int i = 0; i < num_threads; i++) {
-    thread_data[i] = new ThreadData(i, apm_);
-    threads[i] = ThreadWrapper::CreateThread(DeadlockProc,
-                                             thread_data[i],
-                                             kNormalPriority,
-                                             0);
-    ASSERT_TRUE(threads[i] != NULL);
-    unsigned int thread_id = 0;
-    threads[i]->Start(thread_id);
-  }
-
-  EventWrapper* event = EventWrapper::Create();
-  ASSERT_EQ(kEventTimeout, event->Wait(5000));
-  delete event;
-  event = NULL;
-
-  for (int i = 0; i < num_threads; i++) {
-    // This will return false if the thread has deadlocked.
-    ASSERT_TRUE(threads[i]->Stop());
-    ASSERT_FALSE(thread_data[i]->error);
-    delete threads[i];
-    threads[i] = NULL;
-    delete thread_data[i];
-    thread_data[i] = NULL;
-  }
-}*/
-
-TEST_F(ApmTest, StreamParameters) {
-  // No errors when the components are disabled.
-  EXPECT_EQ(apm_->kNoError,
-            apm_->ProcessStream(frame_));
-
-  // Missing agc level
-  EXPECT_EQ(apm_->kNoError, apm_->Initialize());
-  EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
-  EXPECT_EQ(apm_->kStreamParameterNotSetError,
-            apm_->ProcessStream(frame_));
-  EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
-  EXPECT_EQ(apm_->kNoError,
-            apm_->echo_cancellation()->set_stream_drift_samples(0));
-  EXPECT_EQ(apm_->kStreamParameterNotSetError,
-            apm_->ProcessStream(frame_));
-  EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
-
-  // Missing delay
-  EXPECT_EQ(apm_->kNoError, apm_->Initialize());
-  EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
-  EXPECT_EQ(apm_->kStreamParameterNotSetError,
-            apm_->ProcessStream(frame_));
-  EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
-  EXPECT_EQ(apm_->kNoError,
-            apm_->echo_cancellation()->set_stream_drift_samples(0));
-  EXPECT_EQ(apm_->kNoError,
-            apm_->gain_control()->set_stream_analog_level(127));
-  EXPECT_EQ(apm_->kStreamParameterNotSetError,
-            apm_->ProcessStream(frame_));
-  EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
-
-  // Missing drift
-  EXPECT_EQ(apm_->kNoError, apm_->Initialize());
-  EXPECT_EQ(apm_->kNoError,
-            apm_->echo_cancellation()->enable_drift_compensation(true));
-  EXPECT_EQ(apm_->kStreamParameterNotSetError,
-            apm_->ProcessStream(frame_));
-  EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
-  EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
-  EXPECT_EQ(apm_->kNoError,
-            apm_->gain_control()->set_stream_analog_level(127));
-  EXPECT_EQ(apm_->kStreamParameterNotSetError,
-            apm_->ProcessStream(frame_));
-
-  // No stream parameters
-  EXPECT_EQ(apm_->kNoError, apm_->Initialize());
-  EXPECT_EQ(apm_->kNoError,
-            apm_->AnalyzeReverseStream(revframe_));
-  EXPECT_EQ(apm_->kStreamParameterNotSetError,
-            apm_->ProcessStream(frame_));
-
-  // All there
-  EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
-  EXPECT_EQ(apm_->kNoError, apm_->Initialize());
-  EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
-  EXPECT_EQ(apm_->kNoError,
-            apm_->echo_cancellation()->set_stream_drift_samples(0));
-  EXPECT_EQ(apm_->kNoError,
-            apm_->gain_control()->set_stream_analog_level(127));
-  EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
-}
-
-TEST_F(ApmTest, Channels) {
-  // Testing number of invalid channels
-  EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(0, 1));
-  EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 0));
-  EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(3, 1));
-  EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 3));
-  EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(0));
-  EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(3));
-  // Testing number of valid channels
-  for (int i = 1; i < 3; i++) {
-    for (int j = 1; j < 3; j++) {
-      if (j > i) {
-        EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(i, j));
-      } else {
-        EXPECT_EQ(apm_->kNoError, apm_->set_num_channels(i, j));
-        EXPECT_EQ(j, apm_->num_output_channels());
-      }
-    }
-    EXPECT_EQ(i, apm_->num_input_channels());
-    EXPECT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(i));
-    EXPECT_EQ(i, apm_->num_reverse_channels());
-  }
-}
-
-TEST_F(ApmTest, SampleRates) {
-  // Testing invalid sample rates
-  EXPECT_EQ(apm_->kBadParameterError, apm_->set_sample_rate_hz(10000));
-  // Testing valid sample rates
-  int fs[] = {8000, 16000, 32000};
-  for (size_t i = 0; i < sizeof(fs) / sizeof(*fs); i++) {
-    EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(fs[i]));
-    EXPECT_EQ(fs[i], apm_->sample_rate_hz());
-  }
-}
-
-TEST_F(ApmTest, Process) {
-  GOOGLE_PROTOBUF_VERIFY_VERSION;
-  audio_processing_unittest::OutputData output_data;
-
-  if (global_read_output_data) {
-    ReadMessageLiteFromFile("output_data.pb", &output_data);
-
-  } else {
-    // We don't have a file; add the required tests to the protobuf.
-    int rev_ch[] = {1, 2};
-    int ch[] = {1, 2};
-    int fs[] = {8000, 16000, 32000};
-    for (size_t i = 0; i < sizeof(rev_ch) / sizeof(*rev_ch); i++) {
-      for (size_t j = 0; j < sizeof(ch) / sizeof(*ch); j++) {
-        for (size_t k = 0; k < sizeof(fs) / sizeof(*fs); k++) {
-          audio_processing_unittest::Test* test = output_data.add_test();
-          test->set_numreversechannels(rev_ch[i]);
-          test->set_numchannels(ch[j]);
-          test->set_samplerate(fs[k]);
-        }
-      }
-    }
-  }
-
-  EXPECT_EQ(apm_->kNoError,
-            apm_->echo_cancellation()->enable_drift_compensation(true));
-  EXPECT_EQ(apm_->kNoError,
-            apm_->echo_cancellation()->enable_metrics(true));
-  EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
-
-  EXPECT_EQ(apm_->kNoError,
-            apm_->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
-  EXPECT_EQ(apm_->kNoError,
-            apm_->gain_control()->set_analog_level_limits(0, 255));
-  EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
-
-  EXPECT_EQ(apm_->kNoError,
-            apm_->high_pass_filter()->Enable(true));
-
-  //EXPECT_EQ(apm_->kNoError,
-  //          apm_->level_estimator()->Enable(true));
-
-  EXPECT_EQ(apm_->kNoError,
-            apm_->noise_suppression()->Enable(true));
-
-  EXPECT_EQ(apm_->kNoError,
-            apm_->voice_detection()->Enable(true));
-
-  for (int i = 0; i < output_data.test_size(); i++) {
-    printf("Running test %d of %d...\n", i + 1, output_data.test_size());
-
-    audio_processing_unittest::Test* test = output_data.mutable_test(i);
-    const int num_samples = test->samplerate() / 100;
-    revframe_->_payloadDataLengthInSamples = num_samples;
-    revframe_->_audioChannel = test->numreversechannels();
-    revframe_->_frequencyInHz = test->samplerate();
-    frame_->_payloadDataLengthInSamples = num_samples;
-    frame_->_audioChannel = test->numchannels();
-    frame_->_frequencyInHz = test->samplerate();
-
-    EXPECT_EQ(apm_->kNoError, apm_->Initialize());
-    ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(test->samplerate()));
-    ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(frame_->_audioChannel,
-                                                     frame_->_audioChannel));
-    ASSERT_EQ(apm_->kNoError,
-        apm_->set_num_reverse_channels(revframe_->_audioChannel));
-
-
-    int has_echo_count = 0;
-    int has_voice_count = 0;
-    int is_saturated_count = 0;
-
-    while (1) {
-      WebRtc_Word16 temp_data[640];
-      int analog_level = 127;
-
-      // Read far-end frame
-      size_t read_count = fread(temp_data,
-                                sizeof(WebRtc_Word16),
-                                num_samples * 2,
-                                far_file_);
-      if (read_count != static_cast<size_t>(num_samples * 2)) {
-        // Check that the file really ended.
-        ASSERT_NE(0, feof(far_file_));
-        break; // This is expected.
-      }
-
-      if (revframe_->_audioChannel == 1) {
-        MixStereoToMono(temp_data, revframe_->_payloadData,
-            revframe_->_payloadDataLengthInSamples);
-      } else {
-        memcpy(revframe_->_payloadData,
-               &temp_data[0],
-               sizeof(WebRtc_Word16) * read_count);
-      }
-
-      EXPECT_EQ(apm_->kNoError,
-          apm_->AnalyzeReverseStream(revframe_));
-
-      EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
-      EXPECT_EQ(apm_->kNoError,
-          apm_->echo_cancellation()->set_stream_drift_samples(0));
-      EXPECT_EQ(apm_->kNoError,
-          apm_->gain_control()->set_stream_analog_level(analog_level));
-
-      // Read near-end frame
-      read_count = fread(temp_data,
-                         sizeof(WebRtc_Word16),
-                         num_samples * 2,
-                         near_file_);
-      if (read_count != static_cast<size_t>(num_samples * 2)) {
-        // Check that the file really ended.
-        ASSERT_NE(0, feof(near_file_));
-        break; // This is expected.
-      }
-
-      if (frame_->_audioChannel == 1) {
-        MixStereoToMono(temp_data, frame_->_payloadData, num_samples);
-      } else {
-        memcpy(frame_->_payloadData,
-               &temp_data[0],
-               sizeof(WebRtc_Word16) * read_count);
-      }
-
-      EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
-
-      if (apm_->echo_cancellation()->stream_has_echo()) {
-        has_echo_count++;
-      }
-
-      analog_level = apm_->gain_control()->stream_analog_level();
-      if (apm_->gain_control()->stream_is_saturated()) {
-        is_saturated_count++;
-      }
-      if (apm_->voice_detection()->stream_has_voice()) {
-        has_voice_count++;
-      }
-    }
-
-    //<-- Statistics -->
-    //LevelEstimator::Metrics far_metrics;
-    //LevelEstimator::Metrics near_metrics;
-    //EchoCancellation::Metrics echo_metrics;
-    //LevelEstimator::Metrics far_metrics_ref_;
-    //LevelEstimator::Metrics near_metrics_ref_;
-    //EchoCancellation::Metrics echo_metrics_ref_;
-    //EXPECT_EQ(apm_->kNoError,
-    //          apm_->echo_cancellation()->GetMetrics(&echo_metrics));
-    //EXPECT_EQ(apm_->kNoError,
-    //          apm_->level_estimator()->GetMetrics(&near_metrics,
-
-    // TODO(ajm): check echo metrics and output audio.
-    if (global_read_output_data) {
-      EXPECT_EQ(has_echo_count,
-                test->hasechocount());
-      EXPECT_EQ(has_voice_count,
-                test->hasvoicecount());
-      EXPECT_EQ(is_saturated_count,
-                test->issaturatedcount());
-    } else {
-      test->set_hasechocount(has_echo_count);
-      test->set_hasvoicecount(has_voice_count);
-      test->set_issaturatedcount(is_saturated_count);
-    }
-
-    rewind(far_file_);
-    rewind(near_file_);
-  }
-
-  if (!global_read_output_data) {
-    WriteMessageLiteToFile("output_data.pb", output_data);
-  }
-
-  google::protobuf::ShutdownProtobufLibrary();
-}
-
-TEST_F(ApmTest, EchoCancellation) {
-  EXPECT_EQ(apm_->kNoError,
-            apm_->echo_cancellation()->enable_drift_compensation(true));
-  EXPECT_TRUE(apm_->echo_cancellation()->is_drift_compensation_enabled());
-  EXPECT_EQ(apm_->kNoError,
-            apm_->echo_cancellation()->enable_drift_compensation(false));
-  EXPECT_FALSE(apm_->echo_cancellation()->is_drift_compensation_enabled());
-
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->echo_cancellation()->set_device_sample_rate_hz(4000));
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->echo_cancellation()->set_device_sample_rate_hz(100000));
-
-  int rate[] = {16000, 44100, 48000};
-  for (size_t i = 0; i < sizeof(rate)/sizeof(*rate); i++) {
-    EXPECT_EQ(apm_->kNoError,
-        apm_->echo_cancellation()->set_device_sample_rate_hz(rate[i]));
-    EXPECT_EQ(rate[i],
-        apm_->echo_cancellation()->device_sample_rate_hz());
-  }
-
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->echo_cancellation()->set_suppression_level(
-          static_cast<EchoCancellation::SuppressionLevel>(-1)));
-
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->echo_cancellation()->set_suppression_level(
-          static_cast<EchoCancellation::SuppressionLevel>(4)));
-
-  EchoCancellation::SuppressionLevel level[] = {
-    EchoCancellation::kLowSuppression,
-    EchoCancellation::kModerateSuppression,
-    EchoCancellation::kHighSuppression,
-  };
-  for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
-    EXPECT_EQ(apm_->kNoError,
-        apm_->echo_cancellation()->set_suppression_level(level[i]));
-    EXPECT_EQ(level[i],
-        apm_->echo_cancellation()->suppression_level());
-  }
-
-  EchoCancellation::Metrics metrics;
-  EXPECT_EQ(apm_->kNotEnabledError,
-            apm_->echo_cancellation()->GetMetrics(&metrics));
-
-  EXPECT_EQ(apm_->kNoError,
-            apm_->echo_cancellation()->enable_metrics(true));
-  EXPECT_TRUE(apm_->echo_cancellation()->are_metrics_enabled());
-  EXPECT_EQ(apm_->kNoError,
-            apm_->echo_cancellation()->enable_metrics(false));
-  EXPECT_FALSE(apm_->echo_cancellation()->are_metrics_enabled());
-
-  EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
-  EXPECT_TRUE(apm_->echo_cancellation()->is_enabled());
-  EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(false));
-  EXPECT_FALSE(apm_->echo_cancellation()->is_enabled());
-}
-
-TEST_F(ApmTest, EchoControlMobile) {
-  // AECM won't use super-wideband.
-  EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
-  EXPECT_EQ(apm_->kBadSampleRateError, apm_->echo_control_mobile()->Enable(true));
-  EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
-  // Turn AECM on (and AEC off)
-  EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
-  EXPECT_TRUE(apm_->echo_control_mobile()->is_enabled());
-
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->echo_control_mobile()->set_routing_mode(
-      static_cast<EchoControlMobile::RoutingMode>(-1)));
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->echo_control_mobile()->set_routing_mode(
-      static_cast<EchoControlMobile::RoutingMode>(5)));
-
-  // Toggle routing modes
-  EchoControlMobile::RoutingMode mode[] = {
-      EchoControlMobile::kQuietEarpieceOrHeadset,
-      EchoControlMobile::kEarpiece,
-      EchoControlMobile::kLoudEarpiece,
-      EchoControlMobile::kSpeakerphone,
-      EchoControlMobile::kLoudSpeakerphone,
-  };
-  for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
-    EXPECT_EQ(apm_->kNoError,
-        apm_->echo_control_mobile()->set_routing_mode(mode[i]));
-    EXPECT_EQ(mode[i],
-        apm_->echo_control_mobile()->routing_mode());
-  }
-  // Turn comfort noise off/on
-  EXPECT_EQ(apm_->kNoError,
-      apm_->echo_control_mobile()->enable_comfort_noise(false));
-  EXPECT_FALSE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
-  EXPECT_EQ(apm_->kNoError,
-      apm_->echo_control_mobile()->enable_comfort_noise(true));
-  EXPECT_TRUE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
-  // Turn AECM off
-  EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(false));
-  EXPECT_FALSE(apm_->echo_control_mobile()->is_enabled());
-}
-
-TEST_F(ApmTest, GainControl) {
-  // Testing gain modes
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->gain_control()->set_mode(static_cast<GainControl::Mode>(-1)));
-
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->gain_control()->set_mode(static_cast<GainControl::Mode>(3)));
-
-  EXPECT_EQ(apm_->kNoError,
-      apm_->gain_control()->set_mode(
-      apm_->gain_control()->mode()));
-
-  GainControl::Mode mode[] = {
-    GainControl::kAdaptiveAnalog,
-    GainControl::kAdaptiveDigital,
-    GainControl::kFixedDigital
-  };
-  for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
-    EXPECT_EQ(apm_->kNoError,
-        apm_->gain_control()->set_mode(mode[i]));
-    EXPECT_EQ(mode[i], apm_->gain_control()->mode());
-  }
-  // Testing invalid target levels
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->gain_control()->set_target_level_dbfs(-3));
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->gain_control()->set_target_level_dbfs(-40));
-  // Testing valid target levels
-  EXPECT_EQ(apm_->kNoError,
-      apm_->gain_control()->set_target_level_dbfs(
-      apm_->gain_control()->target_level_dbfs()));
-
-  int level_dbfs[] = {0, 6, 31};
-  for (size_t i = 0; i < sizeof(level_dbfs)/sizeof(*level_dbfs); i++) {
-    EXPECT_EQ(apm_->kNoError,
-        apm_->gain_control()->set_target_level_dbfs(level_dbfs[i]));
-    EXPECT_EQ(level_dbfs[i], apm_->gain_control()->target_level_dbfs());
-  }
-
-  // Testing invalid compression gains
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->gain_control()->set_compression_gain_db(-1));
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->gain_control()->set_compression_gain_db(100));
-
-  // Testing valid compression gains
-  EXPECT_EQ(apm_->kNoError,
-      apm_->gain_control()->set_compression_gain_db(
-      apm_->gain_control()->compression_gain_db()));
-
-  int gain_db[] = {0, 10, 90};
-  for (size_t i = 0; i < sizeof(gain_db)/sizeof(*gain_db); i++) {
-    EXPECT_EQ(apm_->kNoError,
-        apm_->gain_control()->set_compression_gain_db(gain_db[i]));
-    EXPECT_EQ(gain_db[i], apm_->gain_control()->compression_gain_db());
-  }
-
-  // Testing limiter off/on
-  EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(false));
-  EXPECT_FALSE(apm_->gain_control()->is_limiter_enabled());
-  EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(true));
-  EXPECT_TRUE(apm_->gain_control()->is_limiter_enabled());
-
-  // Testing invalid level limits
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->gain_control()->set_analog_level_limits(-1, 512));
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->gain_control()->set_analog_level_limits(100000, 512));
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->gain_control()->set_analog_level_limits(512, -1));
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->gain_control()->set_analog_level_limits(512, 100000));
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->gain_control()->set_analog_level_limits(512, 255));
-
-  // Testing valid level limits
-  EXPECT_EQ(apm_->kNoError,
-      apm_->gain_control()->set_analog_level_limits(
-      apm_->gain_control()->analog_level_minimum(),
-      apm_->gain_control()->analog_level_maximum()));
-
-  int min_level[] = {0, 255, 1024};
-  for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
-    EXPECT_EQ(apm_->kNoError,
-        apm_->gain_control()->set_analog_level_limits(min_level[i], 1024));
-    EXPECT_EQ(min_level[i], apm_->gain_control()->analog_level_minimum());
-  }
-
-  int max_level[] = {0, 1024, 65535};
-  for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
-    EXPECT_EQ(apm_->kNoError,
-        apm_->gain_control()->set_analog_level_limits(0, max_level[i]));
-    EXPECT_EQ(max_level[i], apm_->gain_control()->analog_level_maximum());
-  }
-
-  // TODO(ajm): stream_is_saturated() and stream_analog_level()
-
-  // Turn AGC off
-  EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
-  EXPECT_FALSE(apm_->gain_control()->is_enabled());
-}
-
-TEST_F(ApmTest, NoiseSuppression) {
-  // Tesing invalid suppression levels
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->noise_suppression()->set_level(
-          static_cast<NoiseSuppression::Level>(-1)));
-
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->noise_suppression()->set_level(
-          static_cast<NoiseSuppression::Level>(5)));
-
-  // Tesing valid suppression levels
-  NoiseSuppression::Level level[] = {
-    NoiseSuppression::kLow,
-    NoiseSuppression::kModerate,
-    NoiseSuppression::kHigh,
-    NoiseSuppression::kVeryHigh
-  };
-  for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
-    EXPECT_EQ(apm_->kNoError,
-        apm_->noise_suppression()->set_level(level[i]));
-    EXPECT_EQ(level[i], apm_->noise_suppression()->level());
-  }
-
-  // Turing NS on/off
-  EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(true));
-  EXPECT_TRUE(apm_->noise_suppression()->is_enabled());
-  EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(false));
-  EXPECT_FALSE(apm_->noise_suppression()->is_enabled());
-}
-
-TEST_F(ApmTest, HighPassFilter) {
-  // Turing HP filter on/off
-  EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(true));
-  EXPECT_TRUE(apm_->high_pass_filter()->is_enabled());
-  EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(false));
-  EXPECT_FALSE(apm_->high_pass_filter()->is_enabled());
-}
-
-TEST_F(ApmTest, LevelEstimator) {
-  // Turing Level estimator on/off
-  EXPECT_EQ(apm_->kUnsupportedComponentError,
-            apm_->level_estimator()->Enable(true));
-  EXPECT_FALSE(apm_->level_estimator()->is_enabled());
-  EXPECT_EQ(apm_->kUnsupportedComponentError,
-            apm_->level_estimator()->Enable(false));
-  EXPECT_FALSE(apm_->level_estimator()->is_enabled());
-}
-
-TEST_F(ApmTest, VoiceDetection) {
-  // Test external VAD
-  EXPECT_EQ(apm_->kNoError,
-            apm_->voice_detection()->set_stream_has_voice(true));
-  EXPECT_TRUE(apm_->voice_detection()->stream_has_voice());
-  EXPECT_EQ(apm_->kNoError,
-            apm_->voice_detection()->set_stream_has_voice(false));
-  EXPECT_FALSE(apm_->voice_detection()->stream_has_voice());
-
-  // Tesing invalid likelihoods
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->voice_detection()->set_likelihood(
-          static_cast<VoiceDetection::Likelihood>(-1)));
-
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->voice_detection()->set_likelihood(
-          static_cast<VoiceDetection::Likelihood>(5)));
-
-  // Tesing valid likelihoods
-  VoiceDetection::Likelihood likelihood[] = {
-      VoiceDetection::kVeryLowLikelihood,
-      VoiceDetection::kLowLikelihood,
-      VoiceDetection::kModerateLikelihood,
-      VoiceDetection::kHighLikelihood
-  };
-  for (size_t i = 0; i < sizeof(likelihood)/sizeof(*likelihood); i++) {
-    EXPECT_EQ(apm_->kNoError,
-              apm_->voice_detection()->set_likelihood(likelihood[i]));
-    EXPECT_EQ(likelihood[i], apm_->voice_detection()->likelihood());
-  }
-
-  /* TODO(bjornv): Enable once VAD supports other frame lengths than 10 ms
-  // Tesing invalid frame sizes
-  EXPECT_EQ(apm_->kBadParameterError,
-      apm_->voice_detection()->set_frame_size_ms(12));
-
-  // Tesing valid frame sizes
-  for (int i = 10; i <= 30; i += 10) {
-    EXPECT_EQ(apm_->kNoError,
-        apm_->voice_detection()->set_frame_size_ms(i));
-    EXPECT_EQ(i, apm_->voice_detection()->frame_size_ms());
-  }
-  */
-
-  // Turing VAD on/off
-  EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
-  EXPECT_TRUE(apm_->voice_detection()->is_enabled());
-  EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
-  EXPECT_FALSE(apm_->voice_detection()->is_enabled());
-
-  // TODO(bjornv): Add tests for streamed voice; stream_has_voice()
-}
-
-// Below are some ideas for tests from VPM.
-
-/*TEST_F(VideoProcessingModuleTest, GetVersionTest)
-{
-}
-
-TEST_F(VideoProcessingModuleTest, HandleNullBuffer)
-{
-}
-
-TEST_F(VideoProcessingModuleTest, HandleBadSize)
-{
-}
-
-TEST_F(VideoProcessingModuleTest, IdenticalResultsAfterReset)
-{
-}
-*/
-}  // namespace
-
-int main(int argc, char** argv) {
-  ::testing::InitGoogleTest(&argc, argv);
-  ApmEnvironment* env = new ApmEnvironment; // GTest takes ownership.
-  ::testing::AddGlobalTestEnvironment(env);
-
-  for (int i = 1; i < argc; i++) {
-    if (strcmp(argv[i], "--write_output_data") == 0) {
-      global_read_output_data = false;
-    }
-  }
-
-  return RUN_ALL_TESTS();
-}
diff --git a/src/modules/audio_processing/ns/main/interface/noise_suppression.h b/src/modules/audio_processing/ns/main/interface/noise_suppression.h
deleted file mode 100644
index b8983b0..0000000
--- a/src/modules/audio_processing/ns/main/interface/noise_suppression.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_
-
-#include "typedefs.h"
-
-typedef struct NsHandleT NsHandle;
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * This function returns the version number of the code.
- *
- * Input:
- *      - version       : Pointer to a character array where the version
- *                        info is stored.
- *      - length        : Length of version.
- *
- * Return value         :  0 - Ok
- *                        -1 - Error (probably length is not sufficient)
- */
-int WebRtcNs_get_version(char *version, short length);
-
-
-/*
- * This function creates an instance to the noise reduction structure
- *
- * Input:
- *      - NS_inst       : Pointer to noise reduction instance that should be
- *                        created
- *
- * Output:
- *      - NS_inst       : Pointer to created noise reduction instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int WebRtcNs_Create(NsHandle **NS_inst);
-
-
-/*
- * This function frees the dynamic memory of a specified Noise Reduction
- * instance.
- *
- * Input:
- *      - NS_inst       : Pointer to NS instance that should be freed
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int WebRtcNs_Free(NsHandle *NS_inst);
-
-
-/*
- * This function initializes a NS instance
- *
- * Input:
- *      - NS_inst       : Instance that should be initialized
- *      - fs            : sampling frequency
- *
- * Output:
- *      - NS_inst       : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int WebRtcNs_Init(NsHandle *NS_inst, WebRtc_UWord32 fs);
-
-/*
- * This changes the aggressiveness of the noise suppression method.
- *
- * Input:
- *      - NS_inst       : Instance that should be initialized
- *      - mode          : 0: Mild, 1: Medium , 2: Aggressive
- *
- * Output:
- *      - NS_inst       : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int WebRtcNs_set_policy(NsHandle *NS_inst, int mode);
-
-
-/*
- * This functions does Noise Suppression for the inserted speech frame. The
- * input and output signals should always be 10ms (80 or 160 samples).
- *
- * Input
- *      - NS_inst       : VAD Instance. Needs to be initiated before call.
- *      - spframe       : Pointer to speech frame buffer for L band
- *      - spframe_H     : Pointer to speech frame buffer for H band
- *      - fs            : sampling frequency
- *
- * Output:
- *      - NS_inst       : Updated NS instance
- *      - outframe      : Pointer to output frame for L band
- *      - outframe_H    : Pointer to output frame for H band
- *
- * Return value         :  0 - OK
- *                        -1 - Error
- */
-int WebRtcNs_Process(NsHandle *NS_inst,
-                     short *spframe,
-                     short *spframe_H,
-                     short *outframe,
-                     short *outframe_H);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_H_
diff --git a/src/modules/audio_processing/ns/main/interface/noise_suppression_x.h b/src/modules/audio_processing/ns/main/interface/noise_suppression_x.h
deleted file mode 100644
index 35fea2f..0000000
--- a/src/modules/audio_processing/ns/main/interface/noise_suppression_x.h
+++ /dev/null
@@ -1,123 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
-
-#include "signal_processing_library.h"
-
-typedef struct NsxHandleT NsxHandle;
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * This function returns the version number of the code.
- *
- * Input:
- *      - version       : Pointer to a character array where the version
- *                        info is stored.
- *      - length        : Length of version.
- *
- * Return value         :  0 - Ok
- *                        -1 - Error (probably length is not sufficient)
- */
-int WebRtcNsx_get_version(char *version, short length);
-
-
-/*
- * This function creates an instance to the noise reduction structure
- *
- * Input:
- *      - nsxInst       : Pointer to noise reduction instance that should be
- *                       created
- *
- * Output:
- *      - nsxInst       : Pointer to created noise reduction instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int WebRtcNsx_Create(NsxHandle **nsxInst);
-
-
-/*
- * This function frees the dynamic memory of a specified Noise Suppression
- * instance.
- *
- * Input:
- *      - nsxInst       : Pointer to NS instance that should be freed
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int WebRtcNsx_Free(NsxHandle *nsxInst);
-
-
-/*
- * This function initializes a NS instance
- *
- * Input:
- *      - nsxInst       : Instance that should be initialized
- *      - fs            : sampling frequency
- *
- * Output:
- *      - nsxInst       : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int WebRtcNsx_Init(NsxHandle *nsxInst, WebRtc_UWord32 fs);
-
-/*
- * This changes the aggressiveness of the noise suppression method.
- *
- * Input:
- *      - nsxInst       : Instance that should be initialized
- *      - mode          : 0: Mild, 1: Medium , 2: Aggressive
- *
- * Output:
- *      - nsxInst       : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int WebRtcNsx_set_policy(NsxHandle *nsxInst, int mode);
-
-/*
- * This functions does noise suppression for the inserted speech frame. The
- * input and output signals should always be 10ms (80 or 160 samples).
- *
- * Input
- *      - nsxInst       : NSx instance. Needs to be initiated before call.
- *      - speechFrame   : Pointer to speech frame buffer for L band
- *      - speechFrameHB : Pointer to speech frame buffer for H band
- *      - fs            : sampling frequency
- *
- * Output:
- *      - nsxInst       : Updated NSx instance
- *      - outFrame      : Pointer to output frame for L band
- *      - outFrameHB    : Pointer to output frame for H band
- *
- * Return value         :  0 - OK
- *                        -1 - Error
- */
-int WebRtcNsx_Process(NsxHandle *nsxInst,
-                      short *speechFrame,
-                      short *speechFrameHB,
-                      short *outFrame,
-                      short *outFrameHB);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_INTERFACE_NOISE_SUPPRESSION_X_H_
diff --git a/src/modules/audio_processing/ns/main/source/Android.mk b/src/modules/audio_processing/ns/main/source/Android.mk
deleted file mode 100644
index f7444d5..0000000
--- a/src/modules/audio_processing/ns/main/source/Android.mk
+++ /dev/null
@@ -1,50 +0,0 @@
-# This file is generated by gyp; do not edit. This means you!
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_ns
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := \
-    noise_suppression_x.c \
-    nsx_core.c 
-
-# floating point
-# noise_suppression.c ns_core.c 
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=  
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
-    '-DNO_HEAPCHECKER' \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_THREAD_RR' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' 
-
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../../../../.. \
-    $(LOCAL_PATH)/../interface \
-    $(LOCAL_PATH)/../../../utility \
-    $(LOCAL_PATH)/../../../../../common_audio/signal_processing_library/main/interface 
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS := 
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
-    libdl \
-    libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/ns/main/source/defines.h b/src/modules/audio_processing/ns/main/source/defines.h
deleted file mode 100644
index d253967..0000000
--- a/src/modules/audio_processing/ns/main/source/defines.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
-
-//#define PROCESS_FLOW_0    // Use the traditional method.
-//#define PROCESS_FLOW_1    // Use traditional with DD estimate of prior SNR.
-#define PROCESS_FLOW_2    // Use the new method of speech/noise classification.
-
-#define BLOCKL_MAX          160 // max processing block length: 160
-#define ANAL_BLOCKL_MAX     256 // max analysis block length: 256
-#define HALF_ANAL_BLOCKL    129 // half max analysis block length + 1
-
-#define QUANTILE            (float)0.25
-
-#define SIMULT              3
-#define END_STARTUP_LONG    200
-#define END_STARTUP_SHORT   50
-#define FACTOR              (float)40.0
-#define WIDTH               (float)0.01
-
-#define SMOOTH              (float)0.75 // filter smoothing
-// Length of fft work arrays.
-#define IP_LENGTH (ANAL_BLOCKL_MAX >> 1) // must be at least ceil(2 + sqrt(ANAL_BLOCKL_MAX/2))
-#define W_LENGTH (ANAL_BLOCKL_MAX >> 1)
-
-//PARAMETERS FOR NEW METHOD
-#define DD_PR_SNR           (float)0.98 // DD update of prior SNR
-#define LRT_TAVG            (float)0.50 // tavg parameter for LRT (previously 0.90)
-#define SPECT_FL_TAVG       (float)0.30 // tavg parameter for spectral flatness measure
-#define SPECT_DIFF_TAVG     (float)0.30 // tavg parameter for spectral difference measure
-#define PRIOR_UPDATE        (float)0.10 // update parameter of prior model
-#define NOISE_UPDATE        (float)0.90 // update parameter for noise
-#define SPEECH_UPDATE       (float)0.99 // update parameter when likely speech
-#define WIDTH_PR_MAP        (float)4.0  // width parameter in sigmoid map for prior model
-#define LRT_FEATURE_THR     (float)0.5  // default threshold for LRT feature
-#define SF_FEATURE_THR      (float)0.5  // default threshold for Spectral Flatness feature
-#define SD_FEATURE_THR      (float)0.5  // default threshold for Spectral Difference feature
-#define PROB_RANGE          (float)0.20 // probability threshold for noise state in
-                                        // speech/noise likelihood
-#define HIST_PAR_EST         1000       // histogram size for estimation of parameters
-#define GAMMA_PAUSE         (float)0.05 // update for conservative noise estimate
-//
-#define B_LIM               (float)0.5  // threshold in final energy gain factor calculation
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
diff --git a/src/modules/audio_processing/ns/main/source/noise_suppression.c b/src/modules/audio_processing/ns/main/source/noise_suppression.c
deleted file mode 100644
index aed10b1..0000000
--- a/src/modules/audio_processing/ns/main/source/noise_suppression.c
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "noise_suppression.h"
-#include "ns_core.h"
-#include "defines.h"
-
-int WebRtcNs_get_version(char *versionStr, short length)
-{
-    const char version[] = "NS 2.2.0";
-    const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
-
-    if (versionStr == NULL) {
-        return -1;
-    }
-
-    if (versionLen > length) {
-        return -1;
-    }
-
-    strncpy(versionStr, version, versionLen);
-
-    return 0;
-}
-
-int WebRtcNs_Create(NsHandle **NS_inst)
-{
-    *NS_inst = (NsHandle*) malloc(sizeof(NSinst_t));
-    if (*NS_inst!=NULL) {
-        (*(NSinst_t**)NS_inst)->initFlag=0;
-        return 0;
-    } else {
-        return -1;
-    }
-
-}
-
-int WebRtcNs_Free(NsHandle *NS_inst)
-{
-    free(NS_inst);
-    return 0;
-}
-
-
-int WebRtcNs_Init(NsHandle *NS_inst, WebRtc_UWord32 fs)
-{
-    return WebRtcNs_InitCore((NSinst_t*) NS_inst, fs);
-}
-
-int WebRtcNs_set_policy(NsHandle *NS_inst, int mode)
-{
-    return WebRtcNs_set_policy_core((NSinst_t*) NS_inst, mode);
-}
-
-
-int WebRtcNs_Process(NsHandle *NS_inst, short *spframe, short *spframe_H, short *outframe, short *outframe_H)
-{
-    return WebRtcNs_ProcessCore((NSinst_t*) NS_inst, spframe, spframe_H, outframe, outframe_H);
-}
diff --git a/src/modules/audio_processing/ns/main/source/noise_suppression_x.c b/src/modules/audio_processing/ns/main/source/noise_suppression_x.c
deleted file mode 100644
index f1ad730..0000000
--- a/src/modules/audio_processing/ns/main/source/noise_suppression_x.c
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "noise_suppression_x.h"
-#include "nsx_core.h"
-#include "nsx_defines.h"
-
-int WebRtcNsx_get_version(char *versionStr, short length)
-{
-    const char version[] = "NS\t3.1.0";
-    const short versionLen = (short)strlen(version) + 1; // +1 for null-termination
-
-    if (versionStr == NULL)
-    {
-        return -1;
-    }
-
-    if (versionLen > length)
-    {
-        return -1;
-    }
-
-    strncpy(versionStr, version, versionLen);
-
-    return 0;
-}
-
-int WebRtcNsx_Create(NsxHandle **nsxInst)
-{
-    *nsxInst = (NsxHandle*)malloc(sizeof(NsxInst_t));
-    if (*nsxInst != NULL)
-    {
-        (*(NsxInst_t**)nsxInst)->initFlag = 0;
-        return 0;
-    } else
-    {
-        return -1;
-    }
-
-}
-
-int WebRtcNsx_Free(NsxHandle *nsxInst)
-{
-    free(nsxInst);
-    return 0;
-}
-
-int WebRtcNsx_Init(NsxHandle *nsxInst, WebRtc_UWord32 fs)
-{
-    return WebRtcNsx_InitCore((NsxInst_t*)nsxInst, fs);
-}
-
-int WebRtcNsx_set_policy(NsxHandle *nsxInst, int mode)
-{
-    return WebRtcNsx_set_policy_core((NsxInst_t*)nsxInst, mode);
-}
-
-int WebRtcNsx_Process(NsxHandle *nsxInst, short *speechFrame, short *speechFrameHB,
-                      short *outFrame, short *outFrameHB)
-{
-    return WebRtcNsx_ProcessCore((NsxInst_t*)nsxInst, speechFrame, speechFrameHB, outFrame,
-                              outFrameHB);
-}
-
diff --git a/src/modules/audio_processing/ns/main/source/ns.gyp b/src/modules/audio_processing/ns/main/source/ns.gyp
deleted file mode 100644
index c8488b2..0000000
--- a/src/modules/audio_processing/ns/main/source/ns.gyp
+++ /dev/null
@@ -1,67 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../../../../common_settings.gypi',
-  ],
-  'targets': [
-    {
-      'target_name': 'ns',
-      'type': '<(library)',
-      'dependencies': [
-        '../../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
-        '../../../utility/util.gyp:apm_util'
-      ],
-      'include_dirs': [
-        '../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-        ],
-      },
-      'sources': [
-        '../interface/noise_suppression.h',
-        'noise_suppression.c',
-        'windows_private.h',
-        'defines.h',
-        'ns_core.c',
-        'ns_core.h',
-      ],
-    },
-    {
-      'target_name': 'ns_fix',
-      'type': '<(library)',
-      'dependencies': [
-        '../../../../../common_audio/signal_processing_library/main/source/spl.gyp:spl',
-      ],
-      'include_dirs': [
-        '../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-        ],
-      },
-      'sources': [
-        '../interface/noise_suppression_x.h',
-        'noise_suppression_x.c',
-        'nsx_defines.h',
-        'nsx_core.c',
-        'nsx_core.h',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/audio_processing/ns/main/source/ns_core.c b/src/modules/audio_processing/ns/main/source/ns_core.c
deleted file mode 100644
index 10a1b83..0000000
--- a/src/modules/audio_processing/ns/main/source/ns_core.c
+++ /dev/null
@@ -1,1500 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-#include <math.h>
-//#include <stdio.h>
-#include <stdlib.h>
-#include "noise_suppression.h"
-#include "ns_core.h"
-#include "windows_private.h"
-#include "fft4g.h"
-#include "signal_processing_library.h"
-
-// Set Feature Extraction Parameters
-void WebRtcNs_set_feature_extraction_parameters(NSinst_t *inst)
-{
-    //bin size of histogram
-    inst->featureExtractionParams.binSizeLrt      = (float)0.1;
-    inst->featureExtractionParams.binSizeSpecFlat = (float)0.05;
-    inst->featureExtractionParams.binSizeSpecDiff = (float)0.1;
-
-    //range of histogram over which lrt threshold is computed
-    inst->featureExtractionParams.rangeAvgHistLrt = (float)1.0;
-
-    //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
-    // thresholds for prior model
-    inst->featureExtractionParams.factor1ModelPars = (float)1.20; //for lrt and spectral diff
-    inst->featureExtractionParams.factor2ModelPars = (float)0.9;  //for spectral_flatness:
-                                                     // used when noise is flatter than speech
-
-    //peak limit for spectral flatness (varies between 0 and 1)
-    inst->featureExtractionParams.thresPosSpecFlat = (float)0.6;
-
-    //limit on spacing of two highest peaks in histogram: spacing determined by bin size
-    inst->featureExtractionParams.limitPeakSpacingSpecFlat = 2
-            * inst->featureExtractionParams.binSizeSpecFlat;
-    inst->featureExtractionParams.limitPeakSpacingSpecDiff = 2
-            * inst->featureExtractionParams.binSizeSpecDiff;
-
-    //limit on relevance of second peak:
-    inst->featureExtractionParams.limitPeakWeightsSpecFlat = (float)0.5;
-    inst->featureExtractionParams.limitPeakWeightsSpecDiff = (float)0.5;
-
-    // fluctuation limit of lrt feature
-    inst->featureExtractionParams.thresFluctLrt = (float)0.05;
-
-    //limit on the max and min values for the feature thresholds
-    inst->featureExtractionParams.maxLrt = (float)1.0;
-    inst->featureExtractionParams.minLrt = (float)0.20;
-
-    inst->featureExtractionParams.maxSpecFlat = (float)0.95;
-    inst->featureExtractionParams.minSpecFlat = (float)0.10;
-
-    inst->featureExtractionParams.maxSpecDiff = (float)1.0;
-    inst->featureExtractionParams.minSpecDiff = (float)0.16;
-
-    //criteria of weight of histogram peak  to accept/reject feature
-    inst->featureExtractionParams.thresWeightSpecFlat = (int)(0.3
-            * (inst->modelUpdatePars[1])); //for spectral flatness
-    inst->featureExtractionParams.thresWeightSpecDiff = (int)(0.3
-            * (inst->modelUpdatePars[1])); //for spectral difference
-}
-
-// Initialize state
-int WebRtcNs_InitCore(NSinst_t *inst, WebRtc_UWord32 fs)
-{
-    int i;
-    //We only support 10ms frames
-
-    //check for valid pointer
-    if (inst == NULL)
-    {
-        return -1;
-    }
-
-    // Initialization of struct
-    if (fs == 8000 || fs == 16000 || fs == 32000)
-    {
-        inst->fs = fs;
-    }
-    else
-    {
-        return -1;
-    }
-    inst->windShift = 0;
-    if (fs == 8000)
-    {
-        // We only support 10ms frames
-        inst->blockLen = 80;
-        inst->blockLen10ms = 80;
-        inst->anaLen = 128;
-        inst->window = kBlocks80w128;
-        inst->outLen = 0;
-    }
-    else if (fs == 16000)
-    {
-        // We only support 10ms frames
-        inst->blockLen = 160;
-        inst->blockLen10ms = 160;
-        inst->anaLen = 256;
-        inst->window = kBlocks160w256;
-        inst->outLen = 0;
-    }
-    else if (fs==32000)
-    {
-        // We only support 10ms frames
-        inst->blockLen = 160;
-        inst->blockLen10ms = 160;
-        inst->anaLen = 256;
-        inst->window = kBlocks160w256;
-        inst->outLen = 0;
-    }
-    inst->magnLen = inst->anaLen / 2 + 1; // Number of frequency bins
-
-    // Initialize fft work arrays.
-    inst->ip[0] = 0; // Setting this triggers initialization.
-    memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
-    rdft(inst->anaLen, 1, inst->dataBuf, inst->ip, inst->wfft);
-
-    memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
-    memset(inst->syntBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
-
-    //for HB processing
-    memset(inst->dataBufHB, 0, sizeof(float) * ANAL_BLOCKL_MAX);
-
-    //for quantile noise estimation
-    memset(inst->quantile, 0, sizeof(float) * HALF_ANAL_BLOCKL);
-    for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++)
-    {
-        inst->lquantile[i] = (float)8.0;
-        inst->density[i] = (float)0.3;
-    }
-
-    for (i = 0; i < SIMULT; i++)
-    {
-        inst->counter[i] = (int)floor((float)(END_STARTUP_LONG * (i + 1)) / (float)SIMULT);
-    }
-
-    inst->updates = 0;
-
-    // Wiener filter initialization
-    for (i = 0; i < HALF_ANAL_BLOCKL; i++)
-    {
-        inst->smooth[i] = (float)1.0;
-    }
-
-    // Set the aggressiveness: default
-    inst->aggrMode = 0;
-
-    //initialize variables for new method
-    inst->priorSpeechProb = (float)0.5; //prior prob for speech/noise
-    for (i = 0; i < HALF_ANAL_BLOCKL; i++)
-    {
-        inst->magnPrev[i]      = (float)0.0; //previous mag spectrum
-        inst->noisePrev[i]     = (float)0.0; //previous noise-spectrum
-        inst->logLrtTimeAvg[i] = LRT_FEATURE_THR; //smooth LR ratio (same as threshold)
-        inst->magnAvgPause[i]  = (float)0.0; //conservative noise spectrum estimate
-        inst->speechProbHB[i]  = (float)0.0; //for estimation of HB in second pass
-        inst->initMagnEst[i]   = (float)0.0; //initial average mag spectrum
-    }
-
-    //feature quantities
-    inst->featureData[0] = SF_FEATURE_THR;  //spectral flatness (start on threshold)
-    inst->featureData[1] = (float)0.0;      //spectral entropy: not used in this version
-    inst->featureData[2] = (float)0.0;      //spectral variance: not used in this version
-    inst->featureData[3] = LRT_FEATURE_THR; //average lrt factor (start on threshold)
-    inst->featureData[4] = SF_FEATURE_THR;  //spectral template diff (start on threshold)
-    inst->featureData[5] = (float)0.0;      //normalization for spectral-diff
-    inst->featureData[6] = (float)0.0;      //window time-average of input magnitude spectrum
-
-    //histogram quantities: used to estimate/update thresholds for features
-    for (i = 0; i < HIST_PAR_EST; i++)
-    {
-        inst->histLrt[i] = 0;
-        inst->histSpecFlat[i] = 0;
-        inst->histSpecDiff[i] = 0;
-    }
-
-    inst->blockInd = -1; //frame counter
-    inst->priorModelPars[0] = LRT_FEATURE_THR; //default threshold for lrt feature
-    inst->priorModelPars[1] = (float)0.5;      //threshold for spectral flatness:
-                                               // determined on-line
-    inst->priorModelPars[2] = (float)1.0;      //sgn_map par for spectral measure:
-                                               // 1 for flatness measure
-    inst->priorModelPars[3] = (float)0.5;      //threshold for template-difference feature:
-                                               // determined on-line
-    inst->priorModelPars[4] = (float)1.0;      //default weighting parameter for lrt feature
-    inst->priorModelPars[5] = (float)0.0;      //default weighting parameter for
-                                               // spectral flatness feature
-    inst->priorModelPars[6] = (float)0.0;      //default weighting parameter for
-                                               // spectral difference feature
-
-    inst->modelUpdatePars[0] = 2;   //update flag for parameters:
-                                    // 0 no update, 1=update once, 2=update every window
-    inst->modelUpdatePars[1] = 500; //window for update
-    inst->modelUpdatePars[2] = 0;   //counter for update of conservative noise spectrum
-    //counter if the feature thresholds are updated during the sequence
-    inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
-
-    inst->signalEnergy = 0.0;
-    inst->sumMagn = 0.0;
-    inst->whiteNoiseLevel = 0.0;
-    inst->pinkNoiseNumerator = 0.0;
-    inst->pinkNoiseExp = 0.0;
-
-    WebRtcNs_set_feature_extraction_parameters(inst); // Set feature configuration
-
-    //default mode
-    WebRtcNs_set_policy_core(inst, 0);
-
-
-    memset(inst->outBuf, 0, sizeof(float) * 3 * BLOCKL_MAX);
-
-    inst->initFlag = 1;
-    return 0;
-}
-
-int WebRtcNs_set_policy_core(NSinst_t *inst, int mode)
-{
-    // allow for modes:0,1,2,3
-    if (mode < 0 || mode > 3)
-    {
-        return (-1);
-    }
-
-    inst->aggrMode = mode;
-    if (mode == 0)
-    {
-        inst->overdrive = (float)1.0;
-        inst->denoiseBound = (float)0.5;
-        inst->gainmap = 0;
-    }
-    else if (mode == 1)
-    {
-        //inst->overdrive = (float)1.25;
-        inst->overdrive = (float)1.0;
-        inst->denoiseBound = (float)0.25;
-        inst->gainmap = 1;
-    }
-    else if (mode == 2)
-    {
-        //inst->overdrive = (float)1.25;
-        inst->overdrive = (float)1.1;
-        inst->denoiseBound = (float)0.125;
-        inst->gainmap = 1;
-    }
-    else if (mode == 3)
-    {
-        //inst->overdrive = (float)1.30;
-        inst->overdrive = (float)1.25;
-        inst->denoiseBound = (float)0.09;
-        inst->gainmap = 1;
-    }
-    return 0;
-}
-
-// Estimate noise
-void WebRtcNs_NoiseEstimation(NSinst_t *inst, float *magn, float *noise)
-{
-    int i, s, offset;
-    float lmagn[HALF_ANAL_BLOCKL], delta;
-
-    if (inst->updates < END_STARTUP_LONG)
-    {
-        inst->updates++;
-    }
-
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        lmagn[i] = (float)log(magn[i]);
-    }
-
-    // loop over simultaneous estimates
-    for (s = 0; s < SIMULT; s++)
-    {
-        offset = s * inst->magnLen;
-
-        // newquantest(...)
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            // compute delta
-            if (inst->density[offset + i] > 1.0)
-            {
-                delta = FACTOR * (float)1.0 / inst->density[offset + i];
-            }
-            else
-            {
-                delta = FACTOR;
-            }
-
-            // update log quantile estimate
-            if (lmagn[i] > inst->lquantile[offset + i])
-            {
-                inst->lquantile[offset + i] += QUANTILE * delta
-                        / (float)(inst->counter[s] + 1);
-            }
-            else
-            {
-                inst->lquantile[offset + i] -= ((float)1.0 - QUANTILE) * delta
-                        / (float)(inst->counter[s] + 1);
-            }
-
-            // update density estimate
-            if (fabs(lmagn[i] - inst->lquantile[offset + i]) < WIDTH)
-            {
-                inst->density[offset + i] = ((float)inst->counter[s] * inst->density[offset
-                        + i] + (float)1.0 / ((float)2.0 * WIDTH)) / (float)(inst->counter[s]
-                        + 1);
-            }
-        } // end loop over magnitude spectrum
-
-        if (inst->counter[s] >= END_STARTUP_LONG)
-        {
-            inst->counter[s] = 0;
-            if (inst->updates >= END_STARTUP_LONG)
-            {
-                for (i = 0; i < inst->magnLen; i++)
-                {
-                    inst->quantile[i] = (float)exp(inst->lquantile[offset + i]);
-                }
-            }
-        }
-
-        inst->counter[s]++;
-    } // end loop over simultaneous estimates
-
-    // Sequentially update the noise during startup
-    if (inst->updates < END_STARTUP_LONG)
-    {
-        // Use the last "s" to get noise during startup that differ from zero.
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            inst->quantile[i] = (float)exp(inst->lquantile[offset + i]);
-        }
-    }
-
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        noise[i] = inst->quantile[i];
-    }
-}
-
-// Extract thresholds for feature parameters
-// histograms are computed over some window_size (given by inst->modelUpdatePars[1])
-// thresholds and weights are extracted every window
-// flag 0 means update histogram only, flag 1 means compute the thresholds/weights
-// threshold and weights are returned in: inst->priorModelPars
-void WebRtcNs_FeatureParameterExtraction(NSinst_t *inst, int flag)
-{
-    int i, useFeatureSpecFlat, useFeatureSpecDiff, numHistLrt;
-    int maxPeak1, maxPeak2;
-    int weightPeak1SpecFlat, weightPeak2SpecFlat, weightPeak1SpecDiff, weightPeak2SpecDiff;
-
-    float binMid, featureSum;
-    float posPeak1SpecFlat, posPeak2SpecFlat, posPeak1SpecDiff, posPeak2SpecDiff;
-    float fluctLrt, avgHistLrt, avgSquareHistLrt, avgHistLrtCompl;
-
-    //3 features: lrt, flatness, difference
-    //lrt_feature = inst->featureData[3];
-    //flat_feature = inst->featureData[0];
-    //diff_feature = inst->featureData[4];
-
-    //update histograms
-    if (flag == 0)
-    {
-        // LRT
-        if ((inst->featureData[3] < HIST_PAR_EST * inst->featureExtractionParams.binSizeLrt)
-                && (inst->featureData[3] >= 0.0))
-        {
-            i = (int)(inst->featureData[3] / inst->featureExtractionParams.binSizeLrt);
-            inst->histLrt[i]++;
-        }
-        // Spectral flatness
-        if ((inst->featureData[0] < HIST_PAR_EST
-                    * inst->featureExtractionParams.binSizeSpecFlat)
-                && (inst->featureData[0] >= 0.0))
-        {
-            i = (int)(inst->featureData[0] / inst->featureExtractionParams.binSizeSpecFlat);
-            inst->histSpecFlat[i]++;
-        }
-        // Spectral difference
-        if ((inst->featureData[4] < HIST_PAR_EST
-                    * inst->featureExtractionParams.binSizeSpecDiff)
-                && (inst->featureData[4] >= 0.0))
-        {
-            i = (int)(inst->featureData[4] / inst->featureExtractionParams.binSizeSpecDiff);
-            inst->histSpecDiff[i]++;
-        }
-    }
-
-    // extract parameters for speech/noise probability
-    if (flag == 1)
-    {
-        //lrt feature: compute the average over inst->featureExtractionParams.rangeAvgHistLrt
-        avgHistLrt = 0.0;
-        avgHistLrtCompl = 0.0;
-        avgSquareHistLrt = 0.0;
-        numHistLrt = 0;
-        for (i = 0; i < HIST_PAR_EST; i++)
-        {
-            binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeLrt;
-            if (binMid <= inst->featureExtractionParams.rangeAvgHistLrt)
-            {
-                avgHistLrt += inst->histLrt[i] * binMid;
-                numHistLrt += inst->histLrt[i];
-            }
-            avgSquareHistLrt += inst->histLrt[i] * binMid * binMid;
-            avgHistLrtCompl += inst->histLrt[i] * binMid;
-        }
-        if (numHistLrt > 0)
-        {
-            avgHistLrt = avgHistLrt / ((float)numHistLrt);
-        }
-        avgHistLrtCompl = avgHistLrtCompl / ((float)inst->modelUpdatePars[1]);
-        avgSquareHistLrt = avgSquareHistLrt / ((float)inst->modelUpdatePars[1]);
-        fluctLrt = avgSquareHistLrt - avgHistLrt * avgHistLrtCompl;
-        // get threshold for lrt feature:
-        if (fluctLrt < inst->featureExtractionParams.thresFluctLrt)
-        {
-            //very low fluct, so likely noise
-            inst->priorModelPars[0] = inst->featureExtractionParams.maxLrt;
-        }
-        else
-        {
-            inst->priorModelPars[0] = inst->featureExtractionParams.factor1ModelPars
-                    * avgHistLrt;
-            // check if value is within min/max range
-            if (inst->priorModelPars[0] < inst->featureExtractionParams.minLrt)
-            {
-                inst->priorModelPars[0] = inst->featureExtractionParams.minLrt;
-            }
-            if (inst->priorModelPars[0] > inst->featureExtractionParams.maxLrt)
-            {
-                inst->priorModelPars[0] = inst->featureExtractionParams.maxLrt;
-            }
-        }
-        // done with lrt feature
-
-        //
-        // for spectral flatness and spectral difference: compute the main peaks of histogram
-        maxPeak1 = 0;
-        maxPeak2 = 0;
-        posPeak1SpecFlat = 0.0;
-        posPeak2SpecFlat = 0.0;
-        weightPeak1SpecFlat = 0;
-        weightPeak2SpecFlat = 0;
-
-        // peaks for flatness
-        for (i = 0; i < HIST_PAR_EST; i++)
-        {
-            binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecFlat;
-            if (inst->histSpecFlat[i] > maxPeak1)
-            {
-                // Found new "first" peak
-                maxPeak2 = maxPeak1;
-                weightPeak2SpecFlat = weightPeak1SpecFlat;
-                posPeak2SpecFlat = posPeak1SpecFlat;
-
-                maxPeak1 = inst->histSpecFlat[i];
-                weightPeak1SpecFlat = inst->histSpecFlat[i];
-                posPeak1SpecFlat = binMid;
-            }
-            else if (inst->histSpecFlat[i] > maxPeak2)
-            {
-                // Found new "second" peak
-                maxPeak2 = inst->histSpecFlat[i];
-                weightPeak2SpecFlat = inst->histSpecFlat[i];
-                posPeak2SpecFlat = binMid;
-            }
-        }
-
-        //compute two peaks for spectral difference
-        maxPeak1 = 0;
-        maxPeak2 = 0;
-        posPeak1SpecDiff = 0.0;
-        posPeak2SpecDiff = 0.0;
-        weightPeak1SpecDiff = 0;
-        weightPeak2SpecDiff = 0;
-        // peaks for spectral difference
-        for (i = 0; i < HIST_PAR_EST; i++)
-        {
-            binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecDiff;
-            if (inst->histSpecDiff[i] > maxPeak1)
-            {
-                // Found new "first" peak
-                maxPeak2 = maxPeak1;
-                weightPeak2SpecDiff = weightPeak1SpecDiff;
-                posPeak2SpecDiff = posPeak1SpecDiff;
-
-                maxPeak1 = inst->histSpecDiff[i];
-                weightPeak1SpecDiff = inst->histSpecDiff[i];
-                posPeak1SpecDiff = binMid;
-            }
-            else if (inst->histSpecDiff[i] > maxPeak2)
-            {
-                // Found new "second" peak
-                maxPeak2 = inst->histSpecDiff[i];
-                weightPeak2SpecDiff = inst->histSpecDiff[i];
-                posPeak2SpecDiff = binMid;
-            }
-        }
-
-        // for spectrum flatness feature
-        useFeatureSpecFlat = 1;
-        // merge the two peaks if they are close
-        if ((fabs(posPeak2SpecFlat - posPeak1SpecFlat)
-                < inst->featureExtractionParams.limitPeakSpacingSpecFlat)
-                && (weightPeak2SpecFlat
-                        > inst->featureExtractionParams.limitPeakWeightsSpecFlat
-                                * weightPeak1SpecFlat))
-        {
-            weightPeak1SpecFlat += weightPeak2SpecFlat;
-            posPeak1SpecFlat = (float)0.5 * (posPeak1SpecFlat + posPeak2SpecFlat);
-        }
-        //reject if weight of peaks is not large enough, or peak value too small
-        if (weightPeak1SpecFlat < inst->featureExtractionParams.thresWeightSpecFlat
-                || posPeak1SpecFlat < inst->featureExtractionParams.thresPosSpecFlat)
-        {
-            useFeatureSpecFlat = 0;
-        }
-        // if selected, get the threshold
-        if (useFeatureSpecFlat == 1)
-        {
-            // compute the threshold
-            inst->priorModelPars[1] = inst->featureExtractionParams.factor2ModelPars
-                    * posPeak1SpecFlat;
-            //check if value is within min/max range
-            if (inst->priorModelPars[1] < inst->featureExtractionParams.minSpecFlat)
-            {
-                inst->priorModelPars[1] = inst->featureExtractionParams.minSpecFlat;
-            }
-            if (inst->priorModelPars[1] > inst->featureExtractionParams.maxSpecFlat)
-            {
-                inst->priorModelPars[1] = inst->featureExtractionParams.maxSpecFlat;
-            }
-        }
-        // done with flatness feature
-
-        // for template feature
-        useFeatureSpecDiff = 1;
-        // merge the two peaks if they are close
-        if ((fabs(posPeak2SpecDiff - posPeak1SpecDiff)
-                < inst->featureExtractionParams.limitPeakSpacingSpecDiff)
-                && (weightPeak2SpecDiff
-                        > inst->featureExtractionParams.limitPeakWeightsSpecDiff
-                                * weightPeak1SpecDiff))
-        {
-            weightPeak1SpecDiff += weightPeak2SpecDiff;
-            posPeak1SpecDiff = (float)0.5 * (posPeak1SpecDiff + posPeak2SpecDiff);
-        }
-        // get the threshold value
-        inst->priorModelPars[3] = inst->featureExtractionParams.factor1ModelPars
-                * posPeak1SpecDiff;
-        //reject if weight of peaks is not large enough
-        if (weightPeak1SpecDiff < inst->featureExtractionParams.thresWeightSpecDiff)
-        {
-            useFeatureSpecDiff = 0;
-        }
-        //check if value is within min/max range
-        if (inst->priorModelPars[3] < inst->featureExtractionParams.minSpecDiff)
-        {
-            inst->priorModelPars[3] = inst->featureExtractionParams.minSpecDiff;
-        }
-        if (inst->priorModelPars[3] > inst->featureExtractionParams.maxSpecDiff)
-        {
-            inst->priorModelPars[3] = inst->featureExtractionParams.maxSpecDiff;
-        }
-        // done with spectral difference feature
-
-        // don't use template feature if fluctuation of lrt feature is very low:
-        //  most likely just noise state
-        if (fluctLrt < inst->featureExtractionParams.thresFluctLrt)
-        {
-            useFeatureSpecDiff = 0;
-        }
-
-        // select the weights between the features
-        // inst->priorModelPars[4] is weight for lrt: always selected
-        // inst->priorModelPars[5] is weight for spectral flatness
-        // inst->priorModelPars[6] is weight for spectral difference
-        featureSum = (float)(1 + useFeatureSpecFlat + useFeatureSpecDiff);
-        inst->priorModelPars[4] = (float)1.0 / featureSum;
-        inst->priorModelPars[5] = ((float)useFeatureSpecFlat) / featureSum;
-        inst->priorModelPars[6] = ((float)useFeatureSpecDiff) / featureSum;
-
-        // set hists to zero for next update
-        if (inst->modelUpdatePars[0] >= 1)
-        {
-            for (i = 0; i < HIST_PAR_EST; i++)
-            {
-                inst->histLrt[i] = 0;
-                inst->histSpecFlat[i] = 0;
-                inst->histSpecDiff[i] = 0;
-            }
-        }
-    } // end of flag == 1
-}
-
-// Compute spectral flatness on input spectrum
-// magnIn is the magnitude spectrum
-// spectral flatness is returned in inst->featureData[0]
-void WebRtcNs_ComputeSpectralFlatness(NSinst_t *inst, float *magnIn)
-{
-    int i;
-    int shiftLP = 1; //option to remove first bin(s) from spectral measures
-    float avgSpectralFlatnessNum, avgSpectralFlatnessDen, spectralTmp;
-
-    // comute spectral measures
-    // for flatness
-    avgSpectralFlatnessNum = 0.0;
-    avgSpectralFlatnessDen = inst->sumMagn;
-    for (i = 0; i < shiftLP; i++)
-    {
-        avgSpectralFlatnessDen -= magnIn[i];
-    }
-    // compute log of ratio of the geometric to arithmetic mean: check for log(0) case
-    for (i = shiftLP; i < inst->magnLen; i++)
-    {
-        if (magnIn[i] > 0.0)
-        {
-            avgSpectralFlatnessNum += (float)log(magnIn[i]);
-        }
-        else
-        {
-            inst->featureData[0] -= SPECT_FL_TAVG * inst->featureData[0];
-            return;
-        }
-    }
-    //normalize
-    avgSpectralFlatnessDen = avgSpectralFlatnessDen / inst->magnLen;
-    avgSpectralFlatnessNum = avgSpectralFlatnessNum / inst->magnLen;
-
-    //ratio and inverse log: check for case of log(0)
-    spectralTmp = (float)exp(avgSpectralFlatnessNum) / avgSpectralFlatnessDen;
-
-    //time-avg update of spectral flatness feature
-    inst->featureData[0] += SPECT_FL_TAVG * (spectralTmp - inst->featureData[0]);
-    // done with flatness feature
-}
-
-// Compute the difference measure between input spectrum and a template/learned noise spectrum
-// magnIn is the input spectrum
-// the reference/template spectrum is inst->magnAvgPause[i]
-// returns (normalized) spectral difference in inst->featureData[4]
-void WebRtcNs_ComputeSpectralDifference(NSinst_t *inst, float *magnIn)
-{
-    // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / var(magnAvgPause)
-    int i;
-    float avgPause, avgMagn, covMagnPause, varPause, varMagn, avgDiffNormMagn;
-
-    avgPause = 0.0;
-    avgMagn = inst->sumMagn;
-    // compute average quantities
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        //conservative smooth noise spectrum from pause frames
-        avgPause += inst->magnAvgPause[i];
-    }
-    avgPause = avgPause / ((float)inst->magnLen);
-    avgMagn = avgMagn / ((float)inst->magnLen);
-
-    covMagnPause = 0.0;
-    varPause = 0.0;
-    varMagn = 0.0;
-    // compute variance and covariance quantities
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        covMagnPause += (magnIn[i] - avgMagn) * (inst->magnAvgPause[i] - avgPause);
-        varPause += (inst->magnAvgPause[i] - avgPause) * (inst->magnAvgPause[i] - avgPause);
-        varMagn += (magnIn[i] - avgMagn) * (magnIn[i] - avgMagn);
-    }
-    covMagnPause = covMagnPause / ((float)inst->magnLen);
-    varPause = varPause / ((float)inst->magnLen);
-    varMagn = varMagn / ((float)inst->magnLen);
-    // update of average magnitude spectrum
-    inst->featureData[6] += inst->signalEnergy;
-
-    avgDiffNormMagn = varMagn - (covMagnPause * covMagnPause) / (varPause + (float)0.0001);
-    // normalize and compute time-avg update of difference feature
-    avgDiffNormMagn = (float)(avgDiffNormMagn / (inst->featureData[5] + (float)0.0001));
-    inst->featureData[4] += SPECT_DIFF_TAVG * (avgDiffNormMagn - inst->featureData[4]);
-}
-
-// Compute speech/noise probability
-// speech/noise probability is returned in: probSpeechFinal
-//magn is the input magnitude spectrum
-//noise is the noise spectrum
-//snrLocPrior is the prior snr for each freq.
-//snr loc_post is the post snr for each freq.
-void WebRtcNs_SpeechNoiseProb(NSinst_t *inst, float *probSpeechFinal, float *snrLocPrior,
-                              float *snrLocPost)
-{
-    int i, sgnMap;
-    float invLrt, gainPrior, indPrior;
-    float logLrtTimeAvgKsum, besselTmp;
-    float indicator0, indicator1, indicator2;
-    float tmpFloat1, tmpFloat2;
-    float weightIndPrior0, weightIndPrior1, weightIndPrior2;
-    float threshPrior0, threshPrior1, threshPrior2;
-    float widthPrior, widthPrior0, widthPrior1, widthPrior2;
-
-    widthPrior0 = WIDTH_PR_MAP;
-    widthPrior1 = (float)2.0 * WIDTH_PR_MAP; //width for pause region:
-                                             // lower range, so increase width in tanh map
-    widthPrior2 = (float)2.0 * WIDTH_PR_MAP; //for spectral-difference measure
-
-    //threshold parameters for features
-    threshPrior0 = inst->priorModelPars[0];
-    threshPrior1 = inst->priorModelPars[1];
-    threshPrior2 = inst->priorModelPars[3];
-
-    //sign for flatness feature
-    sgnMap = (int)(inst->priorModelPars[2]);
-
-    //weight parameters for features
-    weightIndPrior0 = inst->priorModelPars[4];
-    weightIndPrior1 = inst->priorModelPars[5];
-    weightIndPrior2 = inst->priorModelPars[6];
-
-    // compute feature based on average LR factor
-    // this is the average over all frequencies of the smooth log lrt
-    logLrtTimeAvgKsum = 0.0;
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        tmpFloat1 = (float)1.0 + (float)2.0 * snrLocPrior[i];
-        tmpFloat2 = (float)2.0 * snrLocPrior[i] / (tmpFloat1 + (float)0.0001);
-        besselTmp = (snrLocPost[i] + (float)1.0) * tmpFloat2;
-        inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - (float)log(tmpFloat1)
-                - inst->logLrtTimeAvg[i]);
-        logLrtTimeAvgKsum += inst->logLrtTimeAvg[i];
-    }
-    logLrtTimeAvgKsum = (float)logLrtTimeAvgKsum / (inst->magnLen);
-    inst->featureData[3] = logLrtTimeAvgKsum;
-    // done with computation of LR factor
-
-    //
-    //compute the indicator functions
-    //
-
-    // average lrt feature
-    widthPrior = widthPrior0;
-    //use larger width in tanh map for pause regions
-    if (logLrtTimeAvgKsum < threshPrior0)
-    {
-        widthPrior = widthPrior1;
-    }
-    // compute indicator function: sigmoid map
-    indicator0 = (float)0.5 * ((float)tanh(widthPrior * (logLrtTimeAvgKsum - threshPrior0))
-            + (float)1.0);
-
-    //spectral flatness feature
-    tmpFloat1 = inst->featureData[0];
-    widthPrior = widthPrior0;
-    //use larger width in tanh map for pause regions
-    if (sgnMap == 1 && (tmpFloat1 > threshPrior1))
-    {
-        widthPrior = widthPrior1;
-    }
-    if (sgnMap == -1 && (tmpFloat1 < threshPrior1))
-    {
-        widthPrior = widthPrior1;
-    }
-    // compute indicator function: sigmoid map
-    indicator1 = (float)0.5 * ((float)tanh(
-                                           (float)sgnMap * widthPrior * (threshPrior1
-                                                   - tmpFloat1)) + (float)1.0);
-
-    //for template spectrum-difference
-    tmpFloat1 = inst->featureData[4];
-    widthPrior = widthPrior0;
-    //use larger width in tanh map for pause regions
-    if (tmpFloat1 < threshPrior2)
-    {
-        widthPrior = widthPrior2;
-    }
-    // compute indicator function: sigmoid map
-    indicator2 = (float)0.5 * ((float)tanh(widthPrior * (tmpFloat1 - threshPrior2))
-            + (float)1.0);
-
-    //combine the indicator function with the feature weights
-    indPrior = weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2
-            * indicator2;
-    // done with computing indicator function
-
-    //compute the prior probability
-    inst->priorSpeechProb += PRIOR_UPDATE * (indPrior - inst->priorSpeechProb);
-    // make sure probabilities are within range: keep floor to 0.01
-    if (inst->priorSpeechProb > 1.0)
-    {
-        inst->priorSpeechProb = (float)1.0;
-    }
-    if (inst->priorSpeechProb < 0.01)
-    {
-        inst->priorSpeechProb = (float)0.01;
-    }
-
-    //final speech probability: combine prior model with LR factor:
-    gainPrior = ((float)1.0 - inst->priorSpeechProb) / (inst->priorSpeechProb + (float)0.0001);
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        invLrt = (float)exp(-inst->logLrtTimeAvg[i]);
-        invLrt = (float)gainPrior * invLrt;
-        probSpeechFinal[i] = (float)1.0 / ((float)1.0 + invLrt);
-    }
-}
-
-int WebRtcNs_ProcessCore(NSinst_t *inst,
-                         short *speechFrame,
-                         short *speechFrameHB,
-                         short *outFrame,
-                         short *outFrameHB)
-{
-    // main routine for noise reduction
-
-    int     flagHB = 0;
-    int     i;
-    const int kStartBand = 5; // Skip first frequency bins during estimation.
-    int     updateParsFlag;
-
-    float   energy1, energy2, gain, factor, factor1, factor2;
-    float   signalEnergy, sumMagn;
-    float   snrPrior, currentEstimateStsa;
-    float   tmpFloat1, tmpFloat2, tmpFloat3, probSpeech, probNonSpeech;
-    float   gammaNoiseTmp, gammaNoiseOld;
-    float   noiseUpdateTmp, fTmp, dTmp;
-    float   fin[BLOCKL_MAX], fout[BLOCKL_MAX];
-    float   winData[ANAL_BLOCKL_MAX];
-    float   magn[HALF_ANAL_BLOCKL], noise[HALF_ANAL_BLOCKL];
-    float   theFilter[HALF_ANAL_BLOCKL], theFilterTmp[HALF_ANAL_BLOCKL];
-    float   snrLocPost[HALF_ANAL_BLOCKL], snrLocPrior[HALF_ANAL_BLOCKL];
-    float   probSpeechFinal[HALF_ANAL_BLOCKL], previousEstimateStsa[HALF_ANAL_BLOCKL];
-    float   real[ANAL_BLOCKL_MAX], imag[HALF_ANAL_BLOCKL];
-    // Variables during startup
-    float   sum_log_i = 0.0;
-    float   sum_log_i_square = 0.0;
-    float   sum_log_magn = 0.0;
-    float   sum_log_i_log_magn = 0.0;
-    float   parametric_noise = 0.0;
-    float   parametric_exp = 0.0;
-    float   parametric_num = 0.0;
-
-    // SWB variables
-    int     deltaBweHB = 1;
-    int     deltaGainHB = 1;
-    float   decayBweHB = 1.0;
-    float   gainMapParHB = 1.0;
-    float   gainTimeDomainHB = 1.0;
-    float   avgProbSpeechHB, avgProbSpeechHBTmp, avgFilterGainHB, gainModHB;
-
-    // Check that initiation has been done
-    if (inst->initFlag != 1)
-    {
-        return (-1);
-    }
-    // Check for valid pointers based on sampling rate
-    if (inst->fs == 32000)
-    {
-        if (speechFrameHB == NULL)
-        {
-            return -1;
-        }
-        flagHB = 1;
-        // range for averaging low band quantities for H band gain
-        deltaBweHB = (int)inst->magnLen / 4;
-        deltaGainHB = deltaBweHB;
-    }
-    //
-    updateParsFlag = inst->modelUpdatePars[0];
-    //
-
-    //for LB do all processing
-    // convert to float
-    for (i = 0; i < inst->blockLen10ms; i++)
-    {
-        fin[i] = (float)speechFrame[i];
-    }
-    // update analysis buffer for L band
-    memcpy(inst->dataBuf, inst->dataBuf + inst->blockLen10ms,
-           sizeof(float) * (inst->anaLen - inst->blockLen10ms));
-    memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, fin,
-           sizeof(float) * inst->blockLen10ms);
-
-    if (flagHB == 1)
-    {
-        // convert to float
-        for (i = 0; i < inst->blockLen10ms; i++)
-        {
-            fin[i] = (float)speechFrameHB[i];
-        }
-        // update analysis buffer for H band
-        memcpy(inst->dataBufHB, inst->dataBufHB + inst->blockLen10ms,
-               sizeof(float) * (inst->anaLen - inst->blockLen10ms));
-        memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, fin,
-               sizeof(float) * inst->blockLen10ms);
-    }
-
-    // check if processing needed
-    if (inst->outLen == 0)
-    {
-        // windowing
-        energy1 = 0.0;
-        for (i = 0; i < inst->anaLen; i++)
-        {
-            winData[i] = inst->window[i] * inst->dataBuf[i];
-            energy1 += winData[i] * winData[i];
-        }
-        if (energy1 == 0.0)
-        {
-            // synthesize the special case of zero input
-            // we want to avoid updating statistics in this case:
-            // Updating feature statistics when we have zeros only will cause thresholds to
-            // move towards zero signal situations. This in turn has the effect that once the
-            // signal is "turned on" (non-zero values) everything will be treated as speech
-            // and there is no noise suppression effect. Depending on the duration of the
-            // inactive signal it takes a considerable amount of time for the system to learn
-            // what is noise and what is speech.
-
-            // read out fully processed segment
-            for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++)
-            {
-                fout[i - inst->windShift] = inst->syntBuf[i];
-            }
-            // update synthesis buffer
-            memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen,
-                   sizeof(float) * (inst->anaLen - inst->blockLen));
-            memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0,
-                   sizeof(float) * inst->blockLen);
-
-            // out buffer
-            inst->outLen = inst->blockLen - inst->blockLen10ms;
-            if (inst->blockLen > inst->blockLen10ms)
-            {
-                for (i = 0; i < inst->outLen; i++)
-                {
-                    inst->outBuf[i] = fout[i + inst->blockLen10ms];
-                }
-            }
-            // convert to short
-            for (i = 0; i < inst->blockLen10ms; i++)
-            {
-                dTmp = fout[i];
-                if (dTmp < WEBRTC_SPL_WORD16_MIN)
-                {
-                    dTmp = WEBRTC_SPL_WORD16_MIN;
-                }
-                else if (dTmp > WEBRTC_SPL_WORD16_MAX)
-                {
-                    dTmp = WEBRTC_SPL_WORD16_MAX;
-                }
-                outFrame[i] = (short)dTmp;
-            }
-
-            // for time-domain gain of HB
-            if (flagHB == 1)
-            {
-                for (i = 0; i < inst->blockLen10ms; i++)
-                {
-                    dTmp = inst->dataBufHB[i];
-                    if (dTmp < WEBRTC_SPL_WORD16_MIN)
-                    {
-                        dTmp = WEBRTC_SPL_WORD16_MIN;
-                    }
-                    else if (dTmp > WEBRTC_SPL_WORD16_MAX)
-                    {
-                        dTmp = WEBRTC_SPL_WORD16_MAX;
-                    }
-                    outFrameHB[i] = (short)dTmp;
-                }
-            } // end of H band gain computation
-            //
-            return 0;
-        }
-
-        //
-        inst->blockInd++; // Update the block index only when we process a block.
-        // FFT
-        rdft(inst->anaLen, 1, winData, inst->ip, inst->wfft);
-
-        imag[0] = 0;
-        real[0] = winData[0];
-        magn[0] = (float)(fabs(real[0]) + 1.0f);
-        imag[inst->magnLen - 1] = 0;
-        real[inst->magnLen - 1] = winData[1];
-        magn[inst->magnLen - 1] = (float)(fabs(real[inst->magnLen - 1]) + 1.0f);
-        signalEnergy = (float)(real[0] * real[0]) + (float)(real[inst->magnLen - 1]
-                * real[inst->magnLen - 1]);
-        sumMagn = magn[0] + magn[inst->magnLen - 1];
-        if (inst->blockInd < END_STARTUP_SHORT)
-        {
-            inst->initMagnEst[0] += magn[0];
-            inst->initMagnEst[inst->magnLen - 1] += magn[inst->magnLen - 1];
-            tmpFloat2 = log((float)(inst->magnLen - 1));
-            sum_log_i = tmpFloat2;
-            sum_log_i_square = tmpFloat2 * tmpFloat2;
-            tmpFloat1 = log(magn[inst->magnLen - 1]);
-            sum_log_magn = tmpFloat1;
-            sum_log_i_log_magn = tmpFloat2 * tmpFloat1;
-        }
-        for (i = 1; i < inst->magnLen - 1; i++)
-        {
-            real[i] = winData[2 * i];
-            imag[i] = winData[2 * i + 1];
-            // magnitude spectrum
-            fTmp = real[i] * real[i];
-            fTmp += imag[i] * imag[i];
-            signalEnergy += fTmp;
-            magn[i] = ((float)sqrt(fTmp)) + 1.0f;
-            sumMagn += magn[i];
-            if (inst->blockInd < END_STARTUP_SHORT)
-            {
-                inst->initMagnEst[i] += magn[i];
-                if (i >= kStartBand)
-                {
-                    tmpFloat2 = log((float)i);
-                    sum_log_i += tmpFloat2;
-                    sum_log_i_square += tmpFloat2 * tmpFloat2;
-                    tmpFloat1 = log(magn[i]);
-                    sum_log_magn += tmpFloat1;
-                    sum_log_i_log_magn += tmpFloat2 * tmpFloat1;
-                }
-            }
-        }
-        signalEnergy = signalEnergy / ((float)inst->magnLen);
-        inst->signalEnergy = signalEnergy;
-        inst->sumMagn = sumMagn;
-
-        //compute spectral flatness on input spectrum
-        WebRtcNs_ComputeSpectralFlatness(inst, magn);
-        // quantile noise estimate
-        WebRtcNs_NoiseEstimation(inst, magn, noise);
-        //compute simplified noise model during startup
-        if (inst->blockInd < END_STARTUP_SHORT)
-        {
-            // Estimate White noise
-            inst->whiteNoiseLevel += sumMagn / ((float)inst->magnLen) * inst->overdrive;
-            // Estimate Pink noise parameters
-            tmpFloat1 = sum_log_i_square * ((float)(inst->magnLen - kStartBand));
-            tmpFloat1 -= (sum_log_i * sum_log_i);
-            tmpFloat2 = (sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn);
-            tmpFloat3 = tmpFloat2 / tmpFloat1;
-            // Constrain the estimated spectrum to be positive
-            if (tmpFloat3 < 0.0f)
-            {
-                tmpFloat3 = 0.0f;
-            }
-            inst->pinkNoiseNumerator += tmpFloat3;
-            tmpFloat2 = (sum_log_i * sum_log_magn);
-            tmpFloat2 -= ((float)(inst->magnLen - kStartBand)) * sum_log_i_log_magn;
-            tmpFloat3 = tmpFloat2 / tmpFloat1;
-            // Constrain the pink noise power to be in the interval [0, 1];
-            if (tmpFloat3 < 0.0f)
-            {
-                tmpFloat3 = 0.0f;
-            }
-            if (tmpFloat3 > 1.0f)
-            {
-                tmpFloat3 = 1.0f;
-            }
-            inst->pinkNoiseExp += tmpFloat3;
-
-            // Calculate frequency independent parts of parametric noise estimate.
-            if (inst->pinkNoiseExp == 0.0f)
-            {
-                // Use white noise estimate
-                parametric_noise = inst->whiteNoiseLevel;
-            }
-            else
-            {
-                // Use pink noise estimate
-                parametric_num = exp(inst->pinkNoiseNumerator / (float)(inst->blockInd + 1));
-                parametric_num *= (float)(inst->blockInd + 1);
-                parametric_exp = inst->pinkNoiseExp / (float)(inst->blockInd + 1);
-                parametric_noise = parametric_num / pow((float)kStartBand, parametric_exp);
-            }
-            for (i = 0; i < inst->magnLen; i++)
-            {
-                // Estimate the background noise using the white and pink noise parameters
-                if ((inst->pinkNoiseExp > 0.0f) && (i >= kStartBand))
-                {
-                    // Use pink noise estimate
-                    parametric_noise = parametric_num / pow((float)i, parametric_exp);
-                }
-                theFilterTmp[i] = (inst->initMagnEst[i] - inst->overdrive * parametric_noise);
-                theFilterTmp[i] /= (inst->initMagnEst[i] + (float)0.0001);
-                // Weight quantile noise with modeled noise
-                noise[i] *= (inst->blockInd);
-                tmpFloat2 = parametric_noise * (END_STARTUP_SHORT - inst->blockInd);
-                noise[i] += (tmpFloat2 / (float)(inst->blockInd + 1));
-                noise[i] /= END_STARTUP_SHORT;
-            }
-        }
-        //compute average signal during END_STARTUP_LONG time:
-        // used to normalize spectral difference measure
-        if (inst->blockInd < END_STARTUP_LONG)
-        {
-            inst->featureData[5] *= inst->blockInd;
-            inst->featureData[5] += signalEnergy;
-            inst->featureData[5] /= (inst->blockInd + 1);
-        }
-
-#ifdef PROCESS_FLOW_0
-        if (inst->blockInd > END_STARTUP_LONG)
-        {
-            //option: average the quantile noise: for check with AEC2
-            for (i = 0; i < inst->magnLen; i++)
-            {
-                noise[i] = (float)0.6 * inst->noisePrev[i] + (float)0.4 * noise[i];
-            }
-            for (i = 0; i < inst->magnLen; i++)
-            {
-                // Wiener with over sub-substraction:
-                theFilter[i] = (magn[i] - inst->overdrive * noise[i]) / (magn[i] + (float)0.0001);
-            }
-        }
-#else
-        //start processing at frames == converged+1
-            //
-        // STEP 1: compute  prior and post snr based on quantile noise est
-        //
-
-        // compute DD estimate of prior SNR: needed for new method
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            // post snr
-            snrLocPost[i] = (float)0.0;
-            if (magn[i] > noise[i])
-            {
-                snrLocPost[i] = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
-            }
-            // previous post snr
-            // previous estimate: based on previous frame with gain filter
-            previousEstimateStsa[i] = inst->magnPrev[i] / (inst->noisePrev[i] + (float)0.0001)
-                    * (inst->smooth[i]);
-            // DD estimate is sum of two terms: current estimate and previous estimate
-            // directed decision update of snrPrior
-            snrLocPrior[i] = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR)
-                    * snrLocPost[i];
-            // post and prior snr needed for step 2
-        } // end of loop over freqs
-#ifdef PROCESS_FLOW_1
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            // gain filter
-            tmpFloat1 = inst->overdrive + snrLocPrior[i];
-            tmpFloat2 = (float)snrLocPrior[i] / tmpFloat1;
-            theFilter[i] = (float)tmpFloat2;
-        } // end of loop over freqs
-#endif
-        // done with step 1: dd computation of prior and post snr
-
-        //
-        //STEP 2: compute speech/noise likelihood
-        //
-#ifdef PROCESS_FLOW_2
-        // compute difference of input spectrum with learned/estimated noise spectrum
-        WebRtcNs_ComputeSpectralDifference(inst, magn);
-        // compute histograms for parameter decisions (thresholds and weights for features)
-        // parameters are extracted once every window time (=inst->modelUpdatePars[1])
-        if (updateParsFlag >= 1)
-        {
-            // counter update
-            inst->modelUpdatePars[3]--;
-            // update histogram
-            if (inst->modelUpdatePars[3] > 0)
-            {
-                WebRtcNs_FeatureParameterExtraction(inst, 0);
-            }
-            // compute model parameters
-            if (inst->modelUpdatePars[3] == 0)
-            {
-                WebRtcNs_FeatureParameterExtraction(inst, 1);
-                inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
-                // if wish to update only once, set flag to zero
-                if (updateParsFlag == 1)
-                {
-                    inst->modelUpdatePars[0] = 0;
-                }
-                else
-                {
-                    // update every window:
-                    // get normalization for spectral difference for next window estimate
-                    inst->featureData[6] = inst->featureData[6]
-                            / ((float)inst->modelUpdatePars[1]);
-                    inst->featureData[5] = (float)0.5 * (inst->featureData[6]
-                            + inst->featureData[5]);
-                    inst->featureData[6] = (float)0.0;
-                }
-            }
-        }
-        // compute speech/noise probability
-        WebRtcNs_SpeechNoiseProb(inst, probSpeechFinal, snrLocPrior, snrLocPost);
-        // time-avg parameter for noise update
-        gammaNoiseTmp = NOISE_UPDATE;
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            probSpeech = probSpeechFinal[i];
-            probNonSpeech = (float)1.0 - probSpeech;
-            // temporary noise update:
-            // use it for speech frames if update value is less than previous
-            noiseUpdateTmp = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp)
-                    * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
-            //
-            // time-constant based on speech/noise state
-            gammaNoiseOld = gammaNoiseTmp;
-            gammaNoiseTmp = NOISE_UPDATE;
-            // increase gamma (i.e., less noise update) for frame likely to be speech
-            if (probSpeech > PROB_RANGE)
-            {
-                gammaNoiseTmp = SPEECH_UPDATE;
-            }
-            // conservative noise update
-            if (probSpeech < PROB_RANGE)
-            {
-                inst->magnAvgPause[i] += GAMMA_PAUSE * (magn[i] - inst->magnAvgPause[i]);
-            }
-            // noise update
-            if (gammaNoiseTmp == gammaNoiseOld)
-            {
-                noise[i] = noiseUpdateTmp;
-            }
-            else
-            {
-                noise[i] = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp)
-                        * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
-                // allow for noise update downwards:
-                //  if noise update decreases the noise, it is safe, so allow it to happen
-                if (noiseUpdateTmp < noise[i])
-                {
-                    noise[i] = noiseUpdateTmp;
-                }
-            }
-        } // end of freq loop
-        // done with step 2: noise update
-
-        //
-        // STEP 3: compute dd update of prior snr and post snr based on new noise estimate
-        //
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            // post and prior snr
-            currentEstimateStsa = (float)0.0;
-            if (magn[i] > noise[i])
-            {
-                currentEstimateStsa = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
-            }
-            // DD estimate is sume of two terms: current estimate and previous estimate
-            // directed decision update of snrPrior
-            snrPrior = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR)
-                    * currentEstimateStsa;
-            // gain filter
-            tmpFloat1 = inst->overdrive + snrPrior;
-            tmpFloat2 = (float)snrPrior / tmpFloat1;
-            theFilter[i] = (float)tmpFloat2;
-        } // end of loop over freqs
-        // done with step3
-#endif
-#endif
-
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            // flooring bottom
-            if (theFilter[i] < inst->denoiseBound)
-            {
-                theFilter[i] = inst->denoiseBound;
-            }
-            // flooring top
-            if (theFilter[i] > (float)1.0)
-            {
-                theFilter[i] = 1.0;
-            }
-            if (inst->blockInd < END_STARTUP_SHORT)
-            {
-                // flooring bottom
-                if (theFilterTmp[i] < inst->denoiseBound)
-                {
-                    theFilterTmp[i] = inst->denoiseBound;
-                }
-                // flooring top
-                if (theFilterTmp[i] > (float)1.0)
-                {
-                    theFilterTmp[i] = 1.0;
-                }
-                // Weight the two suppression filters
-                theFilter[i] *= (inst->blockInd);
-                theFilterTmp[i] *= (END_STARTUP_SHORT - inst->blockInd);
-                theFilter[i] += theFilterTmp[i];
-                theFilter[i] /= (END_STARTUP_SHORT);
-            }
-            // smoothing
-#ifdef PROCESS_FLOW_0
-            inst->smooth[i] *= SMOOTH; // value set to 0.7 in define.h file
-            inst->smooth[i] += ((float)1.0 - SMOOTH) * theFilter[i];
-#else
-            inst->smooth[i] = theFilter[i];
-#endif
-            real[i] *= inst->smooth[i];
-            imag[i] *= inst->smooth[i];
-        }
-        // keep track of noise and magn spectrum for next frame
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            inst->noisePrev[i] = noise[i];
-            inst->magnPrev[i] = magn[i];
-        }
-        // back to time domain
-        winData[0] = real[0];
-        winData[1] = real[inst->magnLen - 1];
-        for (i = 1; i < inst->magnLen - 1; i++)
-        {
-            winData[2 * i] = real[i];
-            winData[2 * i + 1] = imag[i];
-        }
-        rdft(inst->anaLen, -1, winData, inst->ip, inst->wfft);
-
-        for (i = 0; i < inst->anaLen; i++)
-        {
-            real[i] = 2.0f * winData[i] / inst->anaLen; // fft scaling
-        }
-
-        //scale factor: only do it after END_STARTUP_LONG time
-        factor = (float)1.0;
-        if (inst->gainmap == 1 && inst->blockInd > END_STARTUP_LONG)
-        {
-            factor1 = (float)1.0;
-            factor2 = (float)1.0;
-
-            energy2 = 0.0;
-            for (i = 0; i < inst->anaLen;i++)
-            {
-                energy2 += (float)real[i] * (float)real[i];
-            }
-            gain = (float)sqrt(energy2 / (energy1 + (float)1.0));
-
-#ifdef PROCESS_FLOW_2
-            // scaling for new version
-            if (gain > B_LIM)
-            {
-                factor1 = (float)1.0 + (float)1.3 * (gain - B_LIM);
-                if (gain * factor1 > (float)1.0)
-                {
-                    factor1 = (float)1.0 / gain;
-                }
-            }
-            if (gain < B_LIM)
-            {
-                //don't reduce scale too much for pause regions:
-                // attenuation here should be controlled by flooring
-                if (gain <= inst->denoiseBound)
-                {
-                    gain = inst->denoiseBound;
-                }
-                factor2 = (float)1.0 - (float)0.3 * (B_LIM - gain);
-            }
-            //combine both scales with speech/noise prob:
-            // note prior (priorSpeechProb) is not frequency dependent
-            factor = inst->priorSpeechProb * factor1 + ((float)1.0 - inst->priorSpeechProb)
-                    * factor2;
-#else
-            if (gain > B_LIM)
-            {
-                factor = (float)1.0 + (float)1.3 * (gain - B_LIM);
-            }
-            else
-            {
-                factor = (float)1.0 + (float)2.0 * (gain - B_LIM);
-            }
-            if (gain * factor > (float)1.0)
-            {
-                factor = (float)1.0 / gain;
-            }
-#endif
-        } // out of inst->gainmap==1
-
-        // synthesis
-        for (i = 0; i < inst->anaLen; i++)
-        {
-            inst->syntBuf[i] += factor * inst->window[i] * (float)real[i];
-        }
-        // read out fully processed segment
-        for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++)
-        {
-            fout[i - inst->windShift] = inst->syntBuf[i];
-        }
-        // update synthesis buffer
-        memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen,
-               sizeof(float) * (inst->anaLen - inst->blockLen));
-        memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0,
-               sizeof(float) * inst->blockLen);
-
-        // out buffer
-        inst->outLen = inst->blockLen - inst->blockLen10ms;
-        if (inst->blockLen > inst->blockLen10ms)
-        {
-            for (i = 0; i < inst->outLen; i++)
-            {
-                inst->outBuf[i] = fout[i + inst->blockLen10ms];
-            }
-        }
-    } // end of if out.len==0
-    else
-    {
-        for (i = 0; i < inst->blockLen10ms; i++)
-        {
-            fout[i] = inst->outBuf[i];
-        }
-        memcpy(inst->outBuf, inst->outBuf + inst->blockLen10ms,
-               sizeof(float) * (inst->outLen - inst->blockLen10ms));
-        memset(inst->outBuf + inst->outLen - inst->blockLen10ms, 0,
-               sizeof(float) * inst->blockLen10ms);
-        inst->outLen -= inst->blockLen10ms;
-    }
-
-    // convert to short
-    for (i = 0; i < inst->blockLen10ms; i++)
-    {
-        dTmp = fout[i];
-        if (dTmp < WEBRTC_SPL_WORD16_MIN)
-        {
-            dTmp = WEBRTC_SPL_WORD16_MIN;
-        }
-        else if (dTmp > WEBRTC_SPL_WORD16_MAX)
-        {
-            dTmp = WEBRTC_SPL_WORD16_MAX;
-        }
-        outFrame[i] = (short)dTmp;
-    }
-
-    // for time-domain gain of HB
-    if (flagHB == 1)
-    {
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            inst->speechProbHB[i] = probSpeechFinal[i];
-        }
-        if (inst->blockInd > END_STARTUP_LONG)
-        {
-            // average speech prob from low band
-            // avg over second half (i.e., 4->8kHz) of freq. spectrum
-            avgProbSpeechHB = 0.0;
-            for (i = inst->magnLen - deltaBweHB - 1; i < inst->magnLen - 1; i++)
-            {
-                avgProbSpeechHB += inst->speechProbHB[i];
-            }
-            avgProbSpeechHB = avgProbSpeechHB / ((float)deltaBweHB);
-            // average filter gain from low band
-            // average over second half (i.e., 4->8kHz) of freq. spectrum
-            avgFilterGainHB = 0.0;
-            for (i = inst->magnLen - deltaGainHB - 1; i < inst->magnLen - 1; i++)
-            {
-                avgFilterGainHB += inst->smooth[i];
-            }
-            avgFilterGainHB = avgFilterGainHB / ((float)(deltaGainHB));
-            avgProbSpeechHBTmp = (float)2.0 * avgProbSpeechHB - (float)1.0;
-            // gain based on speech prob:
-            gainModHB = (float)0.5 * ((float)1.0 + (float)tanh(gainMapParHB * avgProbSpeechHBTmp));
-            //combine gain with low band gain
-            gainTimeDomainHB = (float)0.5 * gainModHB + (float)0.5 * avgFilterGainHB;
-            if (avgProbSpeechHB >= (float)0.5)
-            {
-                gainTimeDomainHB = (float)0.25 * gainModHB + (float)0.75 * avgFilterGainHB;
-            }
-            gainTimeDomainHB = gainTimeDomainHB * decayBweHB;
-        } // end of converged
-        //make sure gain is within flooring range
-        // flooring bottom
-        if (gainTimeDomainHB < inst->denoiseBound)
-        {
-            gainTimeDomainHB = inst->denoiseBound;
-        }
-        // flooring top
-        if (gainTimeDomainHB > (float)1.0)
-        {
-            gainTimeDomainHB = 1.0;
-        }
-        //apply gain
-        for (i = 0; i < inst->blockLen10ms; i++)
-        {
-            dTmp = gainTimeDomainHB * inst->dataBufHB[i];
-            if (dTmp < WEBRTC_SPL_WORD16_MIN)
-            {
-                dTmp = WEBRTC_SPL_WORD16_MIN;
-            }
-            else if (dTmp > WEBRTC_SPL_WORD16_MAX)
-            {
-                dTmp = WEBRTC_SPL_WORD16_MAX;
-            }
-            outFrameHB[i] = (short)dTmp;
-        }
-    } // end of H band gain computation
-    //
-
-    return 0;
-}
diff --git a/src/modules/audio_processing/ns/main/source/ns_core.h b/src/modules/audio_processing/ns/main/source/ns_core.h
deleted file mode 100644
index f72e22b..0000000
--- a/src/modules/audio_processing/ns/main/source/ns_core.h
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
-
-#include "defines.h"
-
-typedef struct NSParaExtract_t_ {
-
-    //bin size of histogram
-    float binSizeLrt;
-    float binSizeSpecFlat;
-    float binSizeSpecDiff;
-    //range of histogram over which lrt threshold is computed
-    float rangeAvgHistLrt;
-    //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
-    //thresholds for prior model
-    float factor1ModelPars; //for lrt and spectral difference
-    float factor2ModelPars; //for spectral_flatness: used when noise is flatter than speech
-    //peak limit for spectral flatness (varies between 0 and 1)
-    float thresPosSpecFlat;
-    //limit on spacing of two highest peaks in histogram: spacing determined by bin size
-    float limitPeakSpacingSpecFlat;
-    float limitPeakSpacingSpecDiff;
-    //limit on relevance of second peak:
-    float limitPeakWeightsSpecFlat;
-    float limitPeakWeightsSpecDiff;
-    //limit on fluctuation of lrt feature
-    float thresFluctLrt;
-    //limit on the max and min values for the feature thresholds
-    float maxLrt;
-    float minLrt;
-    float maxSpecFlat;
-    float minSpecFlat;
-    float maxSpecDiff;
-    float minSpecDiff;
-    //criteria of weight of histogram peak  to accept/reject feature
-    int thresWeightSpecFlat;
-    int thresWeightSpecDiff;
-
-} NSParaExtract_t;
-
-typedef struct NSinst_t_ {
-
-    WebRtc_UWord32  fs;
-    int             blockLen;
-    int             blockLen10ms;
-    int             windShift;
-    int             outLen;
-    int             anaLen;
-    int             magnLen;
-    int             aggrMode;
-    const float*    window;
-    float           dataBuf[ANAL_BLOCKL_MAX];
-    float           syntBuf[ANAL_BLOCKL_MAX];
-    float           outBuf[3 * BLOCKL_MAX];
-
-    int             initFlag;
-    // parameters for quantile noise estimation
-    float           density[SIMULT * HALF_ANAL_BLOCKL];
-    float           lquantile[SIMULT * HALF_ANAL_BLOCKL];
-    float           quantile[HALF_ANAL_BLOCKL];
-    int             counter[SIMULT];
-    int             updates;
-    // parameters for Wiener filter
-    float           smooth[HALF_ANAL_BLOCKL];
-    float           overdrive;
-    float           denoiseBound;
-    int             gainmap;
-    // fft work arrays.
-    int             ip[IP_LENGTH];
-    float           wfft[W_LENGTH];
-
-    // parameters for new method: some not needed, will reduce/cleanup later
-    WebRtc_Word32   blockInd;                           //frame index counter
-    int             modelUpdatePars[4];                 //parameters for updating or estimating
-                                                        // thresholds/weights for prior model
-    float           priorModelPars[7];                  //parameters for prior model
-    float           noisePrev[HALF_ANAL_BLOCKL];        //noise spectrum from previous frame
-    float           magnPrev[HALF_ANAL_BLOCKL];         //magnitude spectrum of previous frame
-    float           logLrtTimeAvg[HALF_ANAL_BLOCKL];    //log lrt factor with time-smoothing
-    float           priorSpeechProb;                    //prior speech/noise probability
-    float           featureData[7];                     //data for features
-    float           magnAvgPause[HALF_ANAL_BLOCKL];     //conservative noise spectrum estimate
-    float           signalEnergy;                       //energy of magn
-    float           sumMagn;                            //sum of magn
-    float           whiteNoiseLevel;                    //initial noise estimate
-    float           initMagnEst[HALF_ANAL_BLOCKL];      //initial magnitude spectrum estimate
-    float           pinkNoiseNumerator;                 //pink noise parameter: numerator
-    float           pinkNoiseExp;                       //pink noise parameter: power of freq
-    NSParaExtract_t featureExtractionParams;            //parameters for feature extraction
-    //histograms for parameter estimation
-    int             histLrt[HIST_PAR_EST];
-    int             histSpecFlat[HIST_PAR_EST];
-    int             histSpecDiff[HIST_PAR_EST];
-    //quantities for high band estimate
-    float           speechProbHB[HALF_ANAL_BLOCKL];     //final speech/noise prob: prior + LRT
-    float           dataBufHB[ANAL_BLOCKL_MAX];         //buffering data for HB
-
-} NSinst_t;
-
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/****************************************************************************
- * WebRtcNs_InitCore(...)
- *
- * This function initializes a noise suppression instance
- *
- * Input:
- *      - inst          : Instance that should be initialized
- *      - fs            : Sampling frequency
- *
- * Output:
- *      - inst          : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int WebRtcNs_InitCore(NSinst_t *inst, WebRtc_UWord32 fs);
-
-/****************************************************************************
- * WebRtcNs_set_policy_core(...)
- *
- * This changes the aggressiveness of the noise suppression method.
- *
- * Input:
- *      - inst          : Instance that should be initialized
- *      - mode          : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB)
- *
- * Output:
- *      - NS_inst      : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int WebRtcNs_set_policy_core(NSinst_t *inst, int mode);
-
-/****************************************************************************
- * WebRtcNs_ProcessCore
- *
- * Do noise suppression.
- *
- * Input:
- *      - inst          : Instance that should be initialized
- *      - inFrameLow    : Input speech frame for lower band
- *      - inFrameHigh   : Input speech frame for higher band
- *
- * Output:
- *      - inst          : Updated instance
- *      - outFrameLow   : Output speech frame for lower band
- *      - outFrameHigh  : Output speech frame for higher band
- *
- * Return value         :  0 - OK
- *                        -1 - Error
- */
-
-
-int WebRtcNs_ProcessCore(NSinst_t *inst,
-                         short *inFrameLow,
-                         short *inFrameHigh,
-                         short *outFrameLow,
-                         short *outFrameHigh);
-
-
-#ifdef __cplusplus
-}
-#endif
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
diff --git a/src/modules/audio_processing/ns/main/source/nsx_core.c b/src/modules/audio_processing/ns/main/source/nsx_core.c
deleted file mode 100644
index 01d3e54..0000000
--- a/src/modules/audio_processing/ns/main/source/nsx_core.c
+++ /dev/null
@@ -1,2493 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "noise_suppression_x.h"
-
-#include <assert.h>
-#include <math.h>
-#include <string.h>
-#include <stdlib.h>
-
-#include "nsx_core.h"
-
-// Skip first frequency bins during estimation. (0 <= value < 64)
-static const int kStartBand = 5;
-
-// Rounding
-static const WebRtc_Word16 kRoundTable[16] = {0, 1, 2, 4, 8, 16, 32, 64, 128, 256, 512, 1024,
-        2048, 4096, 8192, 16384};
-
-// Constants to compensate for shifting signal log(2^shifts).
-static const WebRtc_Word16 kLogTable[9] = {0, 177, 355, 532, 710, 887, 1065, 1242, 1420};
-
-static const WebRtc_Word16 kCounterDiv[201] = {32767, 16384, 10923, 8192, 6554, 5461, 4681,
-        4096, 3641, 3277, 2979, 2731, 2521, 2341, 2185, 2048, 1928, 1820, 1725, 1638, 1560,
-        1489, 1425, 1365, 1311, 1260, 1214, 1170, 1130, 1092, 1057, 1024, 993, 964, 936, 910,
-        886, 862, 840, 819, 799, 780, 762, 745, 728, 712, 697, 683, 669, 655, 643, 630, 618,
-        607, 596, 585, 575, 565, 555, 546, 537, 529, 520, 512, 504, 496, 489, 482, 475, 468,
-        462, 455, 449, 443, 437, 431, 426, 420, 415, 410, 405, 400, 395, 390, 386, 381, 377,
-        372, 368, 364, 360, 356, 352, 349, 345, 341, 338, 334, 331, 328, 324, 321, 318, 315,
-        312, 309, 306, 303, 301, 298, 295, 293, 290, 287, 285, 282, 280, 278, 275, 273, 271,
-        269, 266, 264, 262, 260, 258, 256, 254, 252, 250, 248, 246, 245, 243, 241, 239, 237,
-        236, 234, 232, 231, 229, 228, 226, 224, 223, 221, 220, 218, 217, 216, 214, 213, 211,
-        210, 209, 207, 206, 205, 204, 202, 201, 200, 199, 197, 196, 195, 194, 193, 192, 191,
-        189, 188, 187, 186, 185, 184, 183, 182, 181, 180, 179, 178, 177, 176, 175, 174, 173,
-        172, 172, 171, 170, 169, 168, 167, 166, 165, 165, 164, 163};
-
-static const WebRtc_Word16 kLogTableFrac[256] = {
-      0,   1,   3,   4,   6,   7,   9,  10,  11,  13,  14,  16,  17,  18,  20,  21,
-     22,  24,  25,  26,  28,  29,  30,  32,  33,  34,  36,  37,  38,  40,  41,  42,
-     44,  45,  46,  47,  49,  50,  51,  52,  54,  55,  56,  57,  59,  60,  61,  62,
-     63,  65,  66,  67,  68,  69,  71,  72,  73,  74,  75,  77,  78,  79,  80,  81,
-     82,  84,  85,  86,  87,  88,  89,  90,  92,  93,  94,  95,  96,  97,  98,  99,
-    100, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 116, 117,
-    118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133,
-    134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149,
-    150, 151, 152, 153, 154, 155, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164,
-    165, 166, 167, 168, 169, 169, 170, 171, 172, 173, 174, 175, 176, 177, 178, 178,
-    179, 180, 181, 182, 183, 184, 185, 185, 186, 187, 188, 189, 190, 191, 192, 192,
-    193, 194, 195, 196, 197, 198, 198, 199, 200, 201, 202, 203, 203, 204, 205, 206,
-    207, 208, 208, 209, 210, 211, 212, 212, 213, 214, 215, 216, 216, 217, 218, 219,
-    220, 220, 221, 222, 223, 224, 224, 225, 226, 227, 228, 228, 229, 230, 231, 231,
-    232, 233, 234, 234, 235, 236, 237, 238, 238, 239, 240, 241, 241, 242, 243, 244,
-    244, 245, 246, 247, 247, 248, 249, 249, 250, 251, 252, 252, 253, 254, 255, 255
-};
-
-static const WebRtc_Word16 kPowTableFrac[1024] = {
-    0,    1,    1,    2,    3,    3,    4,    5,
-    6,    6,    7,    8,    8,    9,   10,   10,
-   11,   12,   13,   13,   14,   15,   15,   16,
-   17,   17,   18,   19,   20,   20,   21,   22,
-   22,   23,   24,   25,   25,   26,   27,   27,
-   28,   29,   30,   30,   31,   32,   32,   33,
-   34,   35,   35,   36,   37,   37,   38,   39,
-   40,   40,   41,   42,   42,   43,   44,   45,
-   45,   46,   47,   48,   48,   49,   50,   50,
-   51,   52,   53,   53,   54,   55,   56,   56,
-   57,   58,   58,   59,   60,   61,   61,   62,
-   63,   64,   64,   65,   66,   67,   67,   68,
-   69,   69,   70,   71,   72,   72,   73,   74,
-   75,   75,   76,   77,   78,   78,   79,   80,
-   81,   81,   82,   83,   84,   84,   85,   86,
-   87,   87,   88,   89,   90,   90,   91,   92,
-   93,   93,   94,   95,   96,   96,   97,   98,
-   99,  100,  100,  101,  102,  103,  103,  104,
-  105,  106,  106,  107,  108,  109,  109,  110,
-  111,  112,  113,  113,  114,  115,  116,  116,
-  117,  118,  119,  119,  120,  121,  122,  123,
-  123,  124,  125,  126,  126,  127,  128,  129,
-  130,  130,  131,  132,  133,  133,  134,  135,
-  136,  137,  137,  138,  139,  140,  141,  141,
-  142,  143,  144,  144,  145,  146,  147,  148,
-  148,  149,  150,  151,  152,  152,  153,  154,
-  155,  156,  156,  157,  158,  159,  160,  160,
-  161,  162,  163,  164,  164,  165,  166,  167,
-  168,  168,  169,  170,  171,  172,  173,  173,
-  174,  175,  176,  177,  177,  178,  179,  180,
-  181,  181,  182,  183,  184,  185,  186,  186,
-  187,  188,  189,  190,  190,  191,  192,  193,
-  194,  195,  195,  196,  197,  198,  199,  200,
-  200,  201,  202,  203,  204,  205,  205,  206,
-  207,  208,  209,  210,  210,  211,  212,  213,
-  214,  215,  215,  216,  217,  218,  219,  220,
-  220,  221,  222,  223,  224,  225,  225,  226,
-  227,  228,  229,  230,  231,  231,  232,  233,
-  234,  235,  236,  237,  237,  238,  239,  240,
-  241,  242,  243,  243,  244,  245,  246,  247,
-  248,  249,  249,  250,  251,  252,  253,  254,
-  255,  255,  256,  257,  258,  259,  260,  261,
-  262,  262,  263,  264,  265,  266,  267,  268,
-  268,  269,  270,  271,  272,  273,  274,  275,
-  276,  276,  277,  278,  279,  280,  281,  282,
-  283,  283,  284,  285,  286,  287,  288,  289,
-  290,  291,  291,  292,  293,  294,  295,  296,
-  297,  298,  299,  299,  300,  301,  302,  303,
-  304,  305,  306,  307,  308,  308,  309,  310,
-  311,  312,  313,  314,  315,  316,  317,  318,
-  318,  319,  320,  321,  322,  323,  324,  325,
-  326,  327,  328,  328,  329,  330,  331,  332,
-  333,  334,  335,  336,  337,  338,  339,  339,
-  340,  341,  342,  343,  344,  345,  346,  347,
-  348,  349,  350,  351,  352,  352,  353,  354,
-  355,  356,  357,  358,  359,  360,  361,  362,
-  363,  364,  365,  366,  367,  367,  368,  369,
-  370,  371,  372,  373,  374,  375,  376,  377,
-  378,  379,  380,  381,  382,  383,  384,  385,
-  385,  386,  387,  388,  389,  390,  391,  392,
-  393,  394,  395,  396,  397,  398,  399,  400,
-  401,  402,  403,  404,  405,  406,  407,  408,
-  409,  410,  410,  411,  412,  413,  414,  415,
-  416,  417,  418,  419,  420,  421,  422,  423,
-  424,  425,  426,  427,  428,  429,  430,  431,
-  432,  433,  434,  435,  436,  437,  438,  439,
-  440,  441,  442,  443,  444,  445,  446,  447,
-  448,  449,  450,  451,  452,  453,  454,  455,
-  456,  457,  458,  459,  460,  461,  462,  463,
-  464,  465,  466,  467,  468,  469,  470,  471,
-  472,  473,  474,  475,  476,  477,  478,  479,
-  480,  481,  482,  483,  484,  485,  486,  487,
-  488,  489,  490,  491,  492,  493,  494,  495,
-  496,  498,  499,  500,  501,  502,  503,  504,
-  505,  506,  507,  508,  509,  510,  511,  512,
-  513,  514,  515,  516,  517,  518,  519,  520,
-  521,  522,  523,  525,  526,  527,  528,  529,
-  530,  531,  532,  533,  534,  535,  536,  537,
-  538,  539,  540,  541,  542,  544,  545,  546,
-  547,  548,  549,  550,  551,  552,  553,  554,
-  555,  556,  557,  558,  560,  561,  562,  563,
-  564,  565,  566,  567,  568,  569,  570,  571,
-  572,  574,  575,  576,  577,  578,  579,  580,
-  581,  582,  583,  584,  585,  587,  588,  589,
-  590,  591,  592,  593,  594,  595,  596,  597,
-  599,  600,  601,  602,  603,  604,  605,  606,
-  607,  608,  610,  611,  612,  613,  614,  615,
-  616,  617,  618,  620,  621,  622,  623,  624,
-  625,  626,  627,  628,  630,  631,  632,  633,
-  634,  635,  636,  637,  639,  640,  641,  642,
-  643,  644,  645,  646,  648,  649,  650,  651,
-  652,  653,  654,  656,  657,  658,  659,  660,
-  661,  662,  664,  665,  666,  667,  668,  669,
-  670,  672,  673,  674,  675,  676,  677,  678,
-  680,  681,  682,  683,  684,  685,  687,  688,
-  689,  690,  691,  692,  693,  695,  696,  697,
-  698,  699,  700,  702,  703,  704,  705,  706,
-  708,  709,  710,  711,  712,  713,  715,  716,
-  717,  718,  719,  720,  722,  723,  724,  725,
-  726,  728,  729,  730,  731,  732,  733,  735,
-  736,  737,  738,  739,  741,  742,  743,  744,
-  745,  747,  748,  749,  750,  751,  753,  754,
-  755,  756,  757,  759,  760,  761,  762,  763,
-  765,  766,  767,  768,  770,  771,  772,  773,
-  774,  776,  777,  778,  779,  780,  782,  783,
-  784,  785,  787,  788,  789,  790,  792,  793,
-  794,  795,  796,  798,  799,  800,  801,  803,
-  804,  805,  806,  808,  809,  810,  811,  813,
-  814,  815,  816,  818,  819,  820,  821,  823,
-  824,  825,  826,  828,  829,  830,  831,  833,
-  834,  835,  836,  838,  839,  840,  841,  843,
-  844,  845,  846,  848,  849,  850,  851,  853,
-  854,  855,  857,  858,  859,  860,  862,  863,
-  864,  866,  867,  868,  869,  871,  872,  873,
-  874,  876,  877,  878,  880,  881,  882,  883,
-  885,  886,  887,  889,  890,  891,  893,  894,
-  895,  896,  898,  899,  900,  902,  903,  904,
-  906,  907,  908,  909,  911,  912,  913,  915,
-  916,  917,  919,  920,  921,  923,  924,  925,
-  927,  928,  929,  931,  932,  933,  935,  936,
-  937,  938,  940,  941,  942,  944,  945,  946,
-  948,  949,  950,  952,  953,  955,  956,  957,
-  959,  960,  961,  963,  964,  965,  967,  968,
-  969,  971,  972,  973,  975,  976,  977,  979,
-  980,  981,  983,  984,  986,  987,  988,  990,
-  991,  992,  994,  995,  996,  998,  999, 1001,
- 1002, 1003, 1005, 1006, 1007, 1009, 1010, 1012,
- 1013, 1014, 1016, 1017, 1018, 1020, 1021, 1023
-};
-
-static const WebRtc_Word16 kIndicatorTable[17] = {0, 2017, 3809, 5227, 6258, 6963, 7424, 7718,
-        7901, 8014, 8084, 8126, 8152, 8168, 8177, 8183, 8187};
-
-// hybrib Hanning & flat window
-static const WebRtc_Word16 kBlocks80w128x[128] = {
-        0,    536,   1072,   1606,   2139,   2669,   3196,   3720,   4240,   4756,   5266,
-     5771,   6270,   6762,   7246,   7723,   8192,   8652,   9102,   9543,   9974,  10394,
-    10803,  11200,  11585,  11958,  12318,  12665,  12998,  13318,  13623,  13913,  14189,
-    14449,  14694,  14924,  15137,  15334,  15515,  15679,  15826,  15956,  16069,  16165,
-    16244,  16305,  16349,  16375,  16384,  16384,  16384,  16384,  16384,  16384,  16384,
-    16384,  16384,  16384,  16384,  16384,  16384,  16384,  16384,  16384,  16384,  16384,
-    16384,  16384,  16384,  16384,  16384,  16384,  16384,  16384,  16384,  16384,  16384,
-    16384,  16384,  16384,  16384,  16375,  16349,  16305,  16244,  16165,  16069,  15956,
-    15826,  15679,  15515,  15334,  15137,  14924,  14694,  14449,  14189,  13913,  13623,
-    13318,  12998,  12665,  12318,  11958,  11585,  11200,  10803,  10394,   9974,   9543,
-     9102,   8652,   8192,   7723,   7246,   6762,   6270,   5771,   5266,   4756,   4240,
-     3720,   3196,   2669,   2139,   1606,   1072,    536
-};
-
-// hybrib Hanning & flat window
-static const WebRtc_Word16 kBlocks160w256x[256] = {
-    0,   268,   536,   804,  1072,  1339,  1606,  1872,
- 2139,  2404,  2669,  2933,  3196,  3459,  3720,  3981,
- 4240,  4499,  4756,  5012,  5266,  5520,  5771,  6021,
- 6270,  6517,  6762,  7005,  7246,  7486,  7723,  7959,
- 8192,  8423,  8652,  8878,  9102,  9324,  9543,  9760,
- 9974, 10185, 10394, 10600, 10803, 11003, 11200, 11394,
-11585, 11773, 11958, 12140, 12318, 12493, 12665, 12833,
-12998, 13160, 13318, 13472, 13623, 13770, 13913, 14053,
-14189, 14321, 14449, 14574, 14694, 14811, 14924, 15032,
-15137, 15237, 15334, 15426, 15515, 15599, 15679, 15754,
-15826, 15893, 15956, 16015, 16069, 16119, 16165, 16207,
-16244, 16277, 16305, 16329, 16349, 16364, 16375, 16382,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
-16384, 16382, 16375, 16364, 16349, 16329, 16305, 16277,
-16244, 16207, 16165, 16119, 16069, 16015, 15956, 15893,
-15826, 15754, 15679, 15599, 15515, 15426, 15334, 15237,
-15137, 15032, 14924, 14811, 14694, 14574, 14449, 14321,
-14189, 14053, 13913, 13770, 13623, 13472, 13318, 13160,
-12998, 12833, 12665, 12493, 12318, 12140, 11958, 11773,
-11585, 11394, 11200, 11003, 10803, 10600, 10394, 10185,
- 9974,  9760,  9543,  9324,  9102,  8878,  8652,  8423,
- 8192,  7959,  7723,  7486,  7246,  7005,  6762,  6517,
- 6270,  6021,  5771,  5520,  5266,  5012,  4756,  4499,
- 4240,  3981,  3720,  3459,  3196,  2933,  2669,  2404,
- 2139,  1872,  1606,  1339,  1072,   804,   536,   268
-};
-
-// Gain factor table: Input value in Q8 and output value in Q13
-static const WebRtc_Word16 kFactor1Table[257] = {
-        8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8233, 8274, 8315, 8355, 8396, 8436, 8475, 8515, 8554, 8592, 8631, 8669,
-        8707, 8745, 8783, 8820, 8857, 8894, 8931, 8967, 9003, 9039, 9075, 9111, 9146, 9181,
-        9216, 9251, 9286, 9320, 9354, 9388, 9422, 9456, 9489, 9523, 9556, 9589, 9622, 9655,
-        9687, 9719, 9752, 9784, 9816, 9848, 9879, 9911, 9942, 9973, 10004, 10035, 10066,
-        10097, 10128, 10158, 10188, 10218, 10249, 10279, 10308, 10338, 10368, 10397, 10426,
-        10456, 10485, 10514, 10543, 10572, 10600, 10629, 10657, 10686, 10714, 10742, 10770,
-        10798, 10826, 10854, 10882, 10847, 10810, 10774, 10737, 10701, 10666, 10631, 10596,
-        10562, 10527, 10494, 10460, 10427, 10394, 10362, 10329, 10297, 10266, 10235, 10203,
-        10173, 10142, 10112, 10082, 10052, 10023, 9994, 9965, 9936, 9908, 9879, 9851, 9824,
-        9796, 9769, 9742, 9715, 9689, 9662, 9636, 9610, 9584, 9559, 9534, 9508, 9484, 9459,
-        9434, 9410, 9386, 9362, 9338, 9314, 9291, 9268, 9245, 9222, 9199, 9176, 9154, 9132,
-        9110, 9088, 9066, 9044, 9023, 9002, 8980, 8959, 8939, 8918, 8897, 8877, 8857, 8836,
-        8816, 8796, 8777, 8757, 8738, 8718, 8699, 8680, 8661, 8642, 8623, 8605, 8586, 8568,
-        8550, 8532, 8514, 8496, 8478, 8460, 8443, 8425, 8408, 8391, 8373, 8356, 8339, 8323,
-        8306, 8289, 8273, 8256, 8240, 8224, 8208, 8192
-};
-
-// Gain factor table: Input value in Q8 and output value in Q13
-static const WebRtc_Word16 kFactor2Aggressiveness1[257] = {
-        7577, 7577, 7577, 7577, 7577, 7577,
-        7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7596, 7614, 7632,
-        7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
-        7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
-        8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
-        8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
-};
-
-// Gain factor table: Input value in Q8 and output value in Q13
-static const WebRtc_Word16 kFactor2Aggressiveness2[257] = {
-        7270, 7270, 7270, 7270, 7270, 7306,
-        7339, 7369, 7397, 7424, 7448, 7472, 7495, 7517, 7537, 7558, 7577, 7596, 7614, 7632,
-        7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
-        7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
-        8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
-        8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
-};
-
-// Gain factor table: Input value in Q8 and output value in Q13
-static const WebRtc_Word16 kFactor2Aggressiveness3[257] = {
-        7184, 7184, 7184, 7229, 7270, 7306,
-        7339, 7369, 7397, 7424, 7448, 7472, 7495, 7517, 7537, 7558, 7577, 7596, 7614, 7632,
-        7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
-        7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
-        8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
-        8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
-        8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
-};
-
-// sum of log2(i) from table index to inst->anaLen2 in Q5
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kSumLogIndex[66] = {
-        0,  22917,  22917,  22885,  22834,  22770,  22696,  22613,
-    22524,  22428,  22326,  22220,  22109,  21994,  21876,  21754,
-    21629,  21501,  21370,  21237,  21101,  20963,  20822,  20679,
-    20535,  20388,  20239,  20089,  19937,  19783,  19628,  19470,
-    19312,  19152,  18991,  18828,  18664,  18498,  18331,  18164,
-    17994,  17824,  17653,  17480,  17306,  17132,  16956,  16779,
-    16602,  16423,  16243,  16063,  15881,  15699,  15515,  15331,
-    15146,  14960,  14774,  14586,  14398,  14209,  14019,  13829,
-    13637,  13445
-};
-
-// sum of log2(i)^2 from table index to inst->anaLen2 in Q2
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kSumSquareLogIndex[66] = {
-        0,  16959,  16959,  16955,  16945,  16929,  16908,  16881,
-    16850,  16814,  16773,  16729,  16681,  16630,  16575,  16517,
-    16456,  16392,  16325,  16256,  16184,  16109,  16032,  15952,
-    15870,  15786,  15700,  15612,  15521,  15429,  15334,  15238,
-    15140,  15040,  14938,  14834,  14729,  14622,  14514,  14404,
-    14292,  14179,  14064,  13947,  13830,  13710,  13590,  13468,
-    13344,  13220,  13094,  12966,  12837,  12707,  12576,  12444,
-    12310,  12175,  12039,  11902,  11763,  11624,  11483,  11341,
-    11198,  11054
-};
-
-// log2(table index) in Q12
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kLogIndex[129] = {
-        0,      0,   4096,   6492,   8192,   9511,  10588,  11499,
-    12288,  12984,  13607,  14170,  14684,  15157,  15595,  16003,
-    16384,  16742,  17080,  17400,  17703,  17991,  18266,  18529,
-    18780,  19021,  19253,  19476,  19691,  19898,  20099,  20292,
-    20480,  20662,  20838,  21010,  21176,  21338,  21496,  21649,
-    21799,  21945,  22087,  22226,  22362,  22495,  22625,  22752,
-    22876,  22998,  23117,  23234,  23349,  23462,  23572,  23680,
-    23787,  23892,  23994,  24095,  24195,  24292,  24388,  24483,
-    24576,  24668,  24758,  24847,  24934,  25021,  25106,  25189,
-    25272,  25354,  25434,  25513,  25592,  25669,  25745,  25820,
-    25895,  25968,  26041,  26112,  26183,  26253,  26322,  26390,
-    26458,  26525,  26591,  26656,  26721,  26784,  26848,  26910,
-    26972,  27033,  27094,  27154,  27213,  27272,  27330,  27388,
-    27445,  27502,  27558,  27613,  27668,  27722,  27776,  27830,
-    27883,  27935,  27988,  28039,  28090,  28141,  28191,  28241,
-    28291,  28340,  28388,  28437,  28484,  28532,  28579,  28626,
-    28672
-};
-
-// determinant of estimation matrix in Q0 corresponding to the log2 tables above
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kDeterminantEstMatrix[66] = {
-        0,  29814,  25574,  22640,  20351,  18469,  16873,  15491,
-    14277,  13199,  12233,  11362,  10571,   9851,   9192,   8587,
-     8030,   7515,   7038,   6596,   6186,   5804,   5448,   5115,
-     4805,   4514,   4242,   3988,   3749,   3524,   3314,   3116,
-     2930,   2755,   2590,   2435,   2289,   2152,   2022,   1900,
-     1785,   1677,   1575,   1478,   1388,   1302,   1221,   1145,
-     1073,   1005,    942,    881,    825,    771,    721,    674,
-      629,    587,    547,    510,    475,    442,    411,    382,
-      355,    330
-};
-
-void WebRtcNsx_UpdateNoiseEstimate(NsxInst_t *inst, int offset)
-{
-    WebRtc_Word32 tmp32no1 = 0;
-    WebRtc_Word32 tmp32no2 = 0;
-
-    WebRtc_Word16 tmp16no1 = 0;
-    WebRtc_Word16 tmp16no2 = 0;
-    WebRtc_Word16 exp2Const = 11819; // Q13
-
-    int i = 0;
-
-    tmp16no2 = WebRtcSpl_MaxValueW16(inst->noiseEstLogQuantile + offset, inst->magnLen);
-    inst->qNoise = 14
-            - (int)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(exp2Const, tmp16no2, 21);
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        // inst->quantile[i]=exp(inst->lquantile[offset+i]);
-        // in Q21
-        tmp32no2 = WEBRTC_SPL_MUL_16_16(exp2Const, inst->noiseEstLogQuantile[offset + i]);
-        tmp32no1 = (0x00200000 | (tmp32no2 & 0x001FFFFF));
-        tmp16no1 = -(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21);
-        tmp16no1 += 21;// shift 21 to get result in Q0
-        tmp16no1 -= (WebRtc_Word16)inst->qNoise; //shift to get result in Q(qNoise)
-        if (tmp16no1 > 0)
-        {
-            inst->noiseEstQuantile[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1 +
-                                                       kRoundTable[tmp16no1], tmp16no1);
-        }
-        else
-        {
-            inst->noiseEstQuantile[i] = (WebRtc_Word16)WEBRTC_SPL_LSHIFT_W32(tmp32no1,
-                                                                             -tmp16no1);
-        }
-    }
-}
-
-void WebRtcNsx_CalcParametricNoiseEstimate(NsxInst_t *inst,
-                                           WebRtc_Word16 pink_noise_exp_avg,
-                                           WebRtc_Word32 pink_noise_num_avg,
-                                           int freq_index,
-                                           WebRtc_UWord32 *noise_estimate,
-                                           WebRtc_UWord32 *noise_estimate_avg)
-{
-    WebRtc_Word32 tmp32no1 = 0;
-    WebRtc_Word32 tmp32no2 = 0;
-
-    WebRtc_Word16 int_part = 0;
-    WebRtc_Word16 frac_part = 0;
-
-    // Use pink noise estimate
-    // noise_estimate = 2^(pinkNoiseNumerator + pinkNoiseExp * log2(j))
-    assert(freq_index > 0);
-    tmp32no2 = WEBRTC_SPL_MUL_16_16(pink_noise_exp_avg, kLogIndex[freq_index]); // Q26
-    tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 15); // Q11
-    tmp32no1 = pink_noise_num_avg - tmp32no2; // Q11
-
-    // Calculate output: 2^tmp32no1
-    // Output in Q(minNorm-stages)
-    tmp32no1 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)(inst->minNorm - inst->stages), 11);
-    if (tmp32no1 > 0)
-    {
-        int_part = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 11);
-        frac_part = (WebRtc_Word16)(tmp32no1 & 0x000007ff); // Q11
-        // Piecewise linear approximation of 'b' in
-        // 2^(int_part+frac_part) = 2^int_part * (1 + b)
-        // 'b' is given in Q11 and below stored in frac_part.
-        if (WEBRTC_SPL_RSHIFT_W32(frac_part, 10))
-        {
-            // Upper fractional part
-            tmp32no2 = WEBRTC_SPL_MUL_32_16(2048 - frac_part, 1244); // Q21
-            tmp32no2 = 2048 - WEBRTC_SPL_RSHIFT_W32(tmp32no2, 10);
-        }
-        else
-        {
-            // Lower fractional part
-            tmp32no2 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(frac_part, 804), 10);
-        }
-        // Shift fractional part to Q(minNorm-stages)
-        tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, int_part - 11);
-        *noise_estimate_avg = WEBRTC_SPL_LSHIFT_U32(1, int_part) + (WebRtc_UWord32)tmp32no2;
-        // Scale up to initMagnEst, which is not block averaged
-        *noise_estimate = (*noise_estimate_avg) * (WebRtc_UWord32)(inst->blockIndex + 1);
-    }
-}
-
-// Initialize state
-WebRtc_Word32 WebRtcNsx_InitCore(NsxInst_t *inst, WebRtc_UWord32 fs)
-{
-    int i;
-
-    //check for valid pointer
-    if (inst == NULL)
-    {
-        return -1;
-    }
-    //
-
-    // Initialization of struct
-    if (fs == 8000 || fs == 16000 || fs == 32000)
-    {
-        inst->fs = fs;
-    } else
-    {
-        return -1;
-    }
-
-    if (fs == 8000)
-    {
-        inst->blockLen10ms = 80;
-        inst->anaLen = 128;
-        inst->stages = 7;
-        inst->window = kBlocks80w128x;
-        inst->thresholdLogLrt = 131072; //default threshold for LRT feature
-        inst->maxLrt = 0x0040000;
-        inst->minLrt = 52429;
-    } else if (fs == 16000)
-    {
-        inst->blockLen10ms = 160;
-        inst->anaLen = 256;
-        inst->stages = 8;
-        inst->window = kBlocks160w256x;
-        inst->thresholdLogLrt = 212644; //default threshold for LRT feature
-        inst->maxLrt = 0x0080000;
-        inst->minLrt = 104858;
-    } else if (fs == 32000)
-    {
-        inst->blockLen10ms = 160;
-        inst->anaLen = 256;
-        inst->stages = 8;
-        inst->window = kBlocks160w256x;
-        inst->thresholdLogLrt = 212644; //default threshold for LRT feature
-        inst->maxLrt = 0x0080000;
-        inst->minLrt = 104858;
-    }
-    inst->anaLen2 = WEBRTC_SPL_RSHIFT_W16(inst->anaLen, 1);
-    inst->magnLen = inst->anaLen2 + 1;
-
-    WebRtcSpl_ZerosArrayW16(inst->analysisBuffer, ANAL_BLOCKL_MAX);
-    WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer, ANAL_BLOCKL_MAX);
-
-    // for HB processing
-    WebRtcSpl_ZerosArrayW16(inst->dataBufHBFX, ANAL_BLOCKL_MAX);
-    // for quantile noise estimation
-    WebRtcSpl_ZerosArrayW16(inst->noiseEstQuantile, HALF_ANAL_BLOCKL);
-    for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++)
-    {
-        inst->noiseEstLogQuantile[i] = 2048; // Q8
-        inst->noiseEstDensity[i] = 153; // Q9
-    }
-    for (i = 0; i < SIMULT; i++)
-    {
-        inst->noiseEstCounter[i] = (WebRtc_Word16)(END_STARTUP_LONG * (i + 1)) / SIMULT;
-    }
-
-    // Initialize suppression filter with ones
-    WebRtcSpl_MemSetW16((WebRtc_Word16*)inst->noiseSupFilter, 16384, HALF_ANAL_BLOCKL);
-
-    // Set the aggressiveness: default
-    inst->aggrMode = 0;
-
-    //initialize variables for new method
-    inst->priorNonSpeechProb = 8192; // Q14(0.5) prior probability for speech/noise
-    for (i = 0; i < HALF_ANAL_BLOCKL; i++)
-    {
-        inst->prevMagnU16[i] = 0;
-        inst->prevNoiseU32[i] = 0; //previous noise-spectrum
-        inst->logLrtTimeAvgW32[i] = 0; //smooth LR ratio
-        inst->avgMagnPause[i] = 0; //conservative noise spectrum estimate
-        inst->initMagnEst[i] = 0; //initial average magnitude spectrum
-    }
-
-    //feature quantities
-    inst->thresholdSpecDiff = 50; //threshold for difference feature: determined on-line
-    inst->thresholdSpecFlat = 20480; //threshold for flatness: determined on-line
-    inst->featureLogLrt = inst->thresholdLogLrt; //average LRT factor (= threshold)
-    inst->featureSpecFlat = inst->thresholdSpecFlat; //spectral flatness (= threshold)
-    inst->featureSpecDiff = inst->thresholdSpecDiff; //spectral difference (= threshold)
-    inst->weightLogLrt = 6; //default weighting par for LRT feature
-    inst->weightSpecFlat = 0; //default weighting par for spectral flatness feature
-    inst->weightSpecDiff = 0; //default weighting par for spectral difference feature
-
-    inst->curAvgMagnEnergy = 0; //window time-average of input magnitude spectrum
-    inst->timeAvgMagnEnergy = 0; //normalization for spectral difference
-    inst->timeAvgMagnEnergyTmp = 0; //normalization for spectral difference
-
-    //histogram quantities: used to estimate/update thresholds for features
-    WebRtcSpl_ZerosArrayW16(inst->histLrt, HIST_PAR_EST);
-    WebRtcSpl_ZerosArrayW16(inst->histSpecDiff, HIST_PAR_EST);
-    WebRtcSpl_ZerosArrayW16(inst->histSpecFlat, HIST_PAR_EST);
-
-    inst->blockIndex = -1; //frame counter
-
-    //inst->modelUpdate    = 500;   //window for update
-    inst->modelUpdate = (1 << STAT_UPDATES); //window for update
-    inst->cntThresUpdate = 0; //counter feature thresholds updates
-
-    inst->sumMagn = 0;
-    inst->magnEnergy = 0;
-    inst->prevQMagn = 0;
-    inst->qNoise = 0;
-    inst->prevQNoise = 0;
-
-    inst->energyIn = 0;
-    inst->scaleEnergyIn = 0;
-
-    inst->whiteNoiseLevel = 0;
-    inst->pinkNoiseNumerator = 0;
-    inst->pinkNoiseExp = 0;
-    inst->minNorm = 15; // Start with full scale
-    inst->zeroInputSignal = 0;
-
-    //default mode
-    WebRtcNsx_set_policy_core(inst, 0);
-
-#ifdef NS_FILEDEBUG
-    inst->infile=fopen("indebug.pcm","wb");
-    inst->outfile=fopen("outdebug.pcm","wb");
-    inst->file1=fopen("file1.pcm","wb");
-    inst->file2=fopen("file2.pcm","wb");
-    inst->file3=fopen("file3.pcm","wb");
-    inst->file4=fopen("file4.pcm","wb");
-    inst->file5=fopen("file5.pcm","wb");
-#endif
-
-    inst->initFlag = 1;
-
-    return 0;
-}
-
-int WebRtcNsx_set_policy_core(NsxInst_t *inst, int mode)
-{
-    // allow for modes:0,1,2,3
-    if (mode < 0 || mode > 3)
-    {
-        return -1;
-    }
-
-    inst->aggrMode = mode;
-    if (mode == 0)
-    {
-        inst->overdrive = 256; // Q8(1.0)
-        inst->denoiseBound = 8192; // Q14(0.5)
-        inst->gainMap = 0; // No gain compensation
-    } else if (mode == 1)
-    {
-        inst->overdrive = 256; // Q8(1.0)
-        inst->denoiseBound = 4096; // Q14(0.25)
-        inst->factor2Table = kFactor2Aggressiveness1;
-        inst->gainMap = 1;
-    } else if (mode == 2)
-    {
-        inst->overdrive = 282; // ~= Q8(1.1)
-        inst->denoiseBound = 2048; // Q14(0.125)
-        inst->factor2Table = kFactor2Aggressiveness2;
-        inst->gainMap = 1;
-    } else if (mode == 3)
-    {
-        inst->overdrive = 320; // Q8(1.25)
-        inst->denoiseBound = 1475; // ~= Q14(0.09)
-        inst->factor2Table = kFactor2Aggressiveness3;
-        inst->gainMap = 1;
-    }
-    return 0;
-}
-
-void WebRtcNsx_NoiseEstimation(NsxInst_t *inst, WebRtc_UWord16 *magn, WebRtc_UWord32 *noise,
-                               WebRtc_Word16 *qNoise)
-{
-    WebRtc_Word32 numerator;
-
-    WebRtc_Word16 lmagn[HALF_ANAL_BLOCKL], counter, countDiv, countProd, delta, zeros, frac;
-    WebRtc_Word16 log2, tabind, logval, tmp16, tmp16no1, tmp16no2;
-    WebRtc_Word16 log2Const = 22713; // Q15
-    WebRtc_Word16 widthFactor = 21845;
-
-    int i, s, offset;
-
-    numerator = FACTOR_Q16;
-
-    tabind = inst->stages - inst->normData;
-    if (tabind < 0)
-    {
-        logval = -kLogTable[-tabind];
-    } else
-    {
-        logval = kLogTable[tabind];
-    }
-
-    // lmagn(i)=log(magn(i))=log(2)*log2(magn(i))
-    // magn is in Q(-stages), and the real lmagn values are:
-    // real_lmagn(i)=log(magn(i)*2^stages)=log(magn(i))+log(2^stages)
-    // lmagn in Q8
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        if (magn[i])
-        {
-            zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magn[i]);
-            frac = (WebRtc_Word16)((((WebRtc_UWord32)magn[i] << zeros) & 0x7FFFFFFF) >> 23);
-            // log2(magn(i))
-            log2 = (WebRtc_Word16)(((31 - zeros) << 8) + kLogTableFrac[frac]);
-            // log2(magn(i))*log(2)
-            lmagn[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(log2, log2Const, 15);
-            // + log(2^stages)
-            lmagn[i] += logval;
-        } else
-        {
-            lmagn[i] = logval;//0;
-        }
-    }
-
-    // loop over simultaneous estimates
-    for (s = 0; s < SIMULT; s++)
-    {
-        offset = s * inst->magnLen;
-
-        // Get counter values from state
-        counter = inst->noiseEstCounter[s];
-        countDiv = kCounterDiv[counter];
-        countProd = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(counter, countDiv);
-
-        // quant_est(...)
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            // compute delta
-            if (inst->noiseEstDensity[offset + i] > 512)
-            {
-                delta = WebRtcSpl_DivW32W16ResW16(numerator,
-                                                  inst->noiseEstDensity[offset + i]);
-            } else
-            {
-                delta = FACTOR_Q7;
-            }
-
-            // update log quantile estimate
-            tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delta, countDiv, 14);
-            if (lmagn[i] > inst->noiseEstLogQuantile[offset + i])
-            {
-                // +=QUANTILE*delta/(inst->counter[s]+1) QUANTILE=0.25, =1 in Q2
-                // CounterDiv=1/inst->counter[s] in Q15
-                tmp16 += 2;
-                tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 2);
-                inst->noiseEstLogQuantile[offset + i] += tmp16no1;
-            } else
-            {
-                tmp16 += 1;
-                tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 1);
-                // *(1-QUANTILE), in Q2 QUANTILE=0.25, 1-0.25=0.75=3 in Q2
-                tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, 3, 1);
-                inst->noiseEstLogQuantile[offset + i] -= tmp16no2;
-            }
-
-            // update density estimate
-            if (WEBRTC_SPL_ABS_W16(lmagn[i] - inst->noiseEstLogQuantile[offset + i])
-                    < WIDTH_Q8)
-            {
-                tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
-                        inst->noiseEstDensity[offset + i], countProd, 15);
-                tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(widthFactor,
-                                                                               countDiv, 15);
-                inst->noiseEstDensity[offset + i] = tmp16no1 + tmp16no2;
-            }
-        } // end loop over magnitude spectrum
-
-        if (counter >= END_STARTUP_LONG)
-        {
-            inst->noiseEstCounter[s] = 0;
-            if (inst->blockIndex >= END_STARTUP_LONG)
-            {
-                WebRtcNsx_UpdateNoiseEstimate(inst, offset);
-            }
-        }
-        inst->noiseEstCounter[s]++;
-
-    } // end loop over simultaneous estimates
-
-    // Sequentially update the noise during startup
-    if (inst->blockIndex < END_STARTUP_LONG)
-    {
-        WebRtcNsx_UpdateNoiseEstimate(inst, offset);
-    }
-
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        noise[i] = (WebRtc_UWord32)(inst->noiseEstQuantile[i]); // Q(qNoise)
-    }
-    (*qNoise) = (WebRtc_Word16)inst->qNoise;
-}
-
-// Extract thresholds for feature parameters
-// histograms are computed over some window_size (given by window_pars)
-// thresholds and weights are extracted every window
-// flag 0 means update histogram only, flag 1 means compute the thresholds/weights
-// threshold and weights are returned in: inst->priorModelPars
-void WebRtcNsx_FeatureParameterExtraction(NsxInst_t *inst, int flag)
-{
-    WebRtc_UWord32 tmpU32;
-    WebRtc_UWord32 histIndex;
-    WebRtc_UWord32 posPeak1SpecFlatFX, posPeak2SpecFlatFX;
-    WebRtc_UWord32 posPeak1SpecDiffFX, posPeak2SpecDiffFX;
-
-    WebRtc_Word32 tmp32;
-    WebRtc_Word32 fluctLrtFX, thresFluctLrtFX;
-    WebRtc_Word32 avgHistLrtFX, avgSquareHistLrtFX, avgHistLrtComplFX;
-
-    WebRtc_Word16 j;
-    WebRtc_Word16 numHistLrt;
-
-    int i;
-    int useFeatureSpecFlat, useFeatureSpecDiff, featureSum;
-    int maxPeak1, maxPeak2;
-    int weightPeak1SpecFlat, weightPeak2SpecFlat;
-    int weightPeak1SpecDiff, weightPeak2SpecDiff;
-
-    //update histograms
-    if (!flag)
-    {
-        // LRT
-        // Type casting to UWord32 is safe since negative values will not be wrapped to larger
-        // values than HIST_PAR_EST
-        histIndex = (WebRtc_UWord32)(inst->featureLogLrt);
-        if (histIndex < HIST_PAR_EST)
-        {
-            inst->histLrt[histIndex]++;
-        }
-        // Spectral flatness
-        // (inst->featureSpecFlat*20)>>10 = (inst->featureSpecFlat*5)>>8
-        histIndex = WEBRTC_SPL_RSHIFT_U32(inst->featureSpecFlat * 5, 8);
-        if (histIndex < HIST_PAR_EST)
-        {
-            inst->histSpecFlat[histIndex]++;
-        }
-        // Spectral difference
-        histIndex = HIST_PAR_EST;
-        if (inst->timeAvgMagnEnergy)
-        {
-            // Guard against division by zero
-            // If timeAvgMagnEnergy == 0 we have no normalizing statistics and therefore can't
-            // update the histogram
-            histIndex = WEBRTC_SPL_UDIV((inst->featureSpecDiff * 5) >> inst->stages,
-                                        inst->timeAvgMagnEnergy);
-        }
-        if (histIndex < HIST_PAR_EST)
-        {
-            inst->histSpecDiff[histIndex]++;
-        }
-    }
-
-    // extract parameters for speech/noise probability
-    if (flag)
-    {
-        useFeatureSpecDiff = 1;
-        //for LRT feature:
-        // compute the average over inst->featureExtractionParams.rangeAvgHistLrt
-        avgHistLrtFX = 0;
-        avgSquareHistLrtFX = 0;
-        numHistLrt = 0;
-        for (i = 0; i < BIN_SIZE_LRT; i++)
-        {
-            j = (2 * i + 1);
-            tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
-            avgHistLrtFX += tmp32;
-            numHistLrt += inst->histLrt[i];
-            avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
-        }
-        avgHistLrtComplFX = avgHistLrtFX;
-        for (; i < HIST_PAR_EST; i++)
-        {
-            j = (2 * i + 1);
-            tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
-            avgHistLrtComplFX += tmp32;
-            avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
-        }
-        fluctLrtFX = WEBRTC_SPL_MUL(avgSquareHistLrtFX, numHistLrt);
-        fluctLrtFX -= WEBRTC_SPL_MUL(avgHistLrtFX, avgHistLrtComplFX);
-        thresFluctLrtFX = THRES_FLUCT_LRT * numHistLrt;
-        // get threshold for LRT feature:
-        tmpU32 = (FACTOR_1_LRT_DIFF * (WebRtc_UWord32)avgHistLrtFX);
-        if ((fluctLrtFX < thresFluctLrtFX) || (numHistLrt == 0) || (tmpU32
-                > (WebRtc_UWord32)(100 * numHistLrt)))
-        {
-            inst->thresholdLogLrt = inst->maxLrt; //very low fluctuation, so likely noise
-        } else
-        {
-            tmp32 = (WebRtc_Word32)((tmpU32 << (9 + inst->stages)) / numHistLrt / 25);
-            // check if value is within min/max range
-            inst->thresholdLogLrt = WEBRTC_SPL_SAT(inst->maxLrt, tmp32, inst->minLrt);
-        }
-        if (fluctLrtFX < thresFluctLrtFX)
-        {
-            // Do not use difference feature if fluctuation of LRT feature is very low:
-            // most likely just noise state
-            useFeatureSpecDiff = 0;
-        }
-
-        // for spectral flatness and spectral difference: compute the main peaks of histogram
-        maxPeak1 = 0;
-        maxPeak2 = 0;
-        posPeak1SpecFlatFX = 0;
-        posPeak2SpecFlatFX = 0;
-        weightPeak1SpecFlat = 0;
-        weightPeak2SpecFlat = 0;
-
-        // peaks for flatness
-        for (i = 0; i < HIST_PAR_EST; i++)
-        {
-            if (inst->histSpecFlat[i] > maxPeak1)
-            {
-                // Found new "first" peak
-                maxPeak2 = maxPeak1;
-                weightPeak2SpecFlat = weightPeak1SpecFlat;
-                posPeak2SpecFlatFX = posPeak1SpecFlatFX;
-
-                maxPeak1 = inst->histSpecFlat[i];
-                weightPeak1SpecFlat = inst->histSpecFlat[i];
-                posPeak1SpecFlatFX = (WebRtc_UWord32)(2 * i + 1);
-            } else if (inst->histSpecFlat[i] > maxPeak2)
-            {
-                // Found new "second" peak
-                maxPeak2 = inst->histSpecFlat[i];
-                weightPeak2SpecFlat = inst->histSpecFlat[i];
-                posPeak2SpecFlatFX = (WebRtc_UWord32)(2 * i + 1);
-            }
-        }
-
-        // for spectral flatness feature
-        useFeatureSpecFlat = 1;
-        // merge the two peaks if they are close
-        if ((posPeak1SpecFlatFX - posPeak2SpecFlatFX < LIM_PEAK_SPACE_FLAT_DIFF)
-                && (weightPeak2SpecFlat * LIM_PEAK_WEIGHT_FLAT_DIFF > weightPeak1SpecFlat))
-        {
-            weightPeak1SpecFlat += weightPeak2SpecFlat;
-            posPeak1SpecFlatFX = (posPeak1SpecFlatFX + posPeak2SpecFlatFX) >> 1;
-        }
-        //reject if weight of peaks is not large enough, or peak value too small
-        if (weightPeak1SpecFlat < THRES_WEIGHT_FLAT_DIFF || posPeak1SpecFlatFX
-                < THRES_PEAK_FLAT)
-        {
-            useFeatureSpecFlat = 0;
-        } else // if selected, get the threshold
-        {
-            // compute the threshold and check if value is within min/max range
-            inst->thresholdSpecFlat = WEBRTC_SPL_SAT(MAX_FLAT_Q10, FACTOR_2_FLAT_Q10
-                                                     * posPeak1SpecFlatFX, MIN_FLAT_Q10); //Q10
-        }
-        // done with flatness feature
-
-        if (useFeatureSpecDiff)
-        {
-            //compute two peaks for spectral difference
-            maxPeak1 = 0;
-            maxPeak2 = 0;
-            posPeak1SpecDiffFX = 0;
-            posPeak2SpecDiffFX = 0;
-            weightPeak1SpecDiff = 0;
-            weightPeak2SpecDiff = 0;
-            // peaks for spectral difference
-            for (i = 0; i < HIST_PAR_EST; i++)
-            {
-                if (inst->histSpecDiff[i] > maxPeak1)
-                {
-                    // Found new "first" peak
-                    maxPeak2 = maxPeak1;
-                    weightPeak2SpecDiff = weightPeak1SpecDiff;
-                    posPeak2SpecDiffFX = posPeak1SpecDiffFX;
-
-                    maxPeak1 = inst->histSpecDiff[i];
-                    weightPeak1SpecDiff = inst->histSpecDiff[i];
-                    posPeak1SpecDiffFX = (WebRtc_UWord32)(2 * i + 1);
-                } else if (inst->histSpecDiff[i] > maxPeak2)
-                {
-                    // Found new "second" peak
-                    maxPeak2 = inst->histSpecDiff[i];
-                    weightPeak2SpecDiff = inst->histSpecDiff[i];
-                    posPeak2SpecDiffFX = (WebRtc_UWord32)(2 * i + 1);
-                }
-            }
-
-            // merge the two peaks if they are close
-            if ((posPeak1SpecDiffFX - posPeak2SpecDiffFX < LIM_PEAK_SPACE_FLAT_DIFF)
-                    && (weightPeak2SpecDiff * LIM_PEAK_WEIGHT_FLAT_DIFF > weightPeak1SpecDiff))
-            {
-                weightPeak1SpecDiff += weightPeak2SpecDiff;
-                posPeak1SpecDiffFX = (posPeak1SpecDiffFX + posPeak2SpecDiffFX) >> 1;
-            }
-            // get the threshold value and check if value is within min/max range
-            inst->thresholdSpecDiff = WEBRTC_SPL_SAT(MAX_DIFF, FACTOR_1_LRT_DIFF
-                                                     * posPeak1SpecDiffFX, MIN_DIFF); //5x bigger
-            //reject if weight of peaks is not large enough
-            if (weightPeak1SpecDiff < THRES_WEIGHT_FLAT_DIFF)
-            {
-                useFeatureSpecDiff = 0;
-            }
-            // done with spectral difference feature
-        }
-
-        // select the weights between the features
-        // inst->priorModelPars[4] is weight for LRT: always selected
-        featureSum = 6 / (1 + useFeatureSpecFlat + useFeatureSpecDiff);
-        inst->weightLogLrt = featureSum;
-        inst->weightSpecFlat = useFeatureSpecFlat * featureSum;
-        inst->weightSpecDiff = useFeatureSpecDiff * featureSum;
-
-        // set histograms to zero for next update
-        WebRtcSpl_ZerosArrayW16(inst->histLrt, HIST_PAR_EST);
-        WebRtcSpl_ZerosArrayW16(inst->histSpecDiff, HIST_PAR_EST);
-        WebRtcSpl_ZerosArrayW16(inst->histSpecFlat, HIST_PAR_EST);
-    } // end of flag == 1
-}
-
-
-// Compute spectral flatness on input spectrum
-// magn is the magnitude spectrum
-// spectral flatness is returned in inst->featureSpecFlat
-void WebRtcNsx_ComputeSpectralFlatness(NsxInst_t *inst, WebRtc_UWord16 *magn)
-{
-    WebRtc_UWord32 tmpU32;
-    WebRtc_UWord32 avgSpectralFlatnessNum, avgSpectralFlatnessDen;
-
-    WebRtc_Word32 tmp32;
-    WebRtc_Word32 currentSpectralFlatness, logCurSpectralFlatness;
-
-    WebRtc_Word16 zeros, frac, intPart;
-
-    int i;
-
-    // for flatness
-    avgSpectralFlatnessNum = 0;
-    avgSpectralFlatnessDen = inst->sumMagn - (WebRtc_UWord32)magn[0]; // Q(normData-stages)
-
-    // compute log of ratio of the geometric to arithmetic mean: check for log(0) case
-    // flatness = exp( sum(log(magn[i]))/N - log(sum(magn[i])/N) )
-    //          = exp( sum(log(magn[i]))/N ) * N / sum(magn[i])
-    //          = 2^( sum(log2(magn[i]))/N - (log2(sum(magn[i])) - log2(N)) ) [This is used]
-    for (i = 1; i < inst->magnLen; i++)
-    {
-        // First bin is excluded from spectrum measures. Number of bins is now a power of 2
-        if (magn[i])
-        {
-            zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magn[i]);
-            frac = (WebRtc_Word16)(((WebRtc_UWord32)((WebRtc_UWord32)(magn[i]) << zeros)
-                    & 0x7FFFFFFF) >> 23);
-            // log2(magn(i))
-            tmpU32 = (WebRtc_UWord32)(((31 - zeros) << 8) + kLogTableFrac[frac]); // Q8
-            avgSpectralFlatnessNum += tmpU32; // Q8
-        } else
-        {
-            //if at least one frequency component is zero, treat separately
-            tmpU32 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecFlat, SPECT_FLAT_TAVG_Q14); // Q24
-            inst->featureSpecFlat -= WEBRTC_SPL_RSHIFT_U32(tmpU32, 14); // Q10
-            return;
-        }
-    }
-    //ratio and inverse log: check for case of log(0)
-    zeros = WebRtcSpl_NormU32(avgSpectralFlatnessDen);
-    frac = (WebRtc_Word16)(((avgSpectralFlatnessDen << zeros) & 0x7FFFFFFF) >> 23);
-    // log2(avgSpectralFlatnessDen)
-    tmp32 = (WebRtc_Word32)(((31 - zeros) << 8) + kLogTableFrac[frac]); // Q8
-    logCurSpectralFlatness = (WebRtc_Word32)avgSpectralFlatnessNum;
-    logCurSpectralFlatness += ((WebRtc_Word32)(inst->stages - 1) << (inst->stages + 7)); // Q(8+stages-1)
-    logCurSpectralFlatness -= (tmp32 << (inst->stages - 1));
-    logCurSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(logCurSpectralFlatness, 10 - inst->stages); // Q17
-    tmp32 = (WebRtc_Word32)(0x00020000 | (WEBRTC_SPL_ABS_W32(logCurSpectralFlatness)
-            & 0x0001FFFF)); //Q17
-    intPart = -(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(logCurSpectralFlatness, 17);
-    intPart += 7; // Shift 7 to get the output in Q10 (from Q17 = -17+10)
-    if (intPart > 0)
-    {
-        currentSpectralFlatness = WEBRTC_SPL_RSHIFT_W32(tmp32, intPart);
-    } else
-    {
-        currentSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(tmp32, -intPart);
-    }
-
-    //time average update of spectral flatness feature
-    tmp32 = currentSpectralFlatness - (WebRtc_Word32)inst->featureSpecFlat; // Q10
-    tmp32 = WEBRTC_SPL_MUL_32_16(SPECT_FLAT_TAVG_Q14, tmp32); // Q24
-    inst->featureSpecFlat = (WebRtc_UWord32)((WebRtc_Word32)inst->featureSpecFlat
-            + WEBRTC_SPL_RSHIFT_W32(tmp32, 14)); // Q10
-    // done with flatness feature
-}
-
-
-// Compute the difference measure between input spectrum and a template/learned noise spectrum
-// magn_tmp is the input spectrum
-// the reference/template spectrum is  inst->magn_avg_pause[i]
-// returns (normalized) spectral difference in inst->featureSpecDiff
-void WebRtcNsx_ComputeSpectralDifference(NsxInst_t *inst, WebRtc_UWord16 *magnIn)
-{
-    // This is to be calculated:
-    // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / var(magnAvgPause)
-
-    WebRtc_UWord32 tmpU32no1, tmpU32no2;
-    WebRtc_UWord32 varMagnUFX, varPauseUFX, avgDiffNormMagnUFX;
-
-    WebRtc_Word32 tmp32no1, tmp32no2;
-    WebRtc_Word32 avgPauseFX, avgMagnFX, covMagnPauseFX;
-    WebRtc_Word32 maxPause, minPause;
-
-    WebRtc_Word16 tmp16no1;
-
-    int i, norm32, nShifts;
-
-    avgPauseFX = 0;
-    maxPause = 0;
-    minPause = inst->avgMagnPause[0]; // Q(prevQMagn)
-    // compute average quantities
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        // Compute mean of magn_pause
-        avgPauseFX += inst->avgMagnPause[i]; // in Q(prevQMagn)
-        maxPause = WEBRTC_SPL_MAX(maxPause, inst->avgMagnPause[i]);
-        minPause = WEBRTC_SPL_MIN(minPause, inst->avgMagnPause[i]);
-    }
-    // normalize by replacing div of "inst->magnLen" with "inst->stages-1" shifts
-    avgPauseFX = WEBRTC_SPL_RSHIFT_W32(avgPauseFX, inst->stages - 1);
-    avgMagnFX = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(inst->sumMagn, inst->stages - 1);
-    // Largest possible deviation in magnPause for (co)var calculations
-    tmp32no1 = WEBRTC_SPL_MAX(maxPause - avgPauseFX, avgPauseFX - minPause);
-    // Get number of shifts to make sure we don't get wrap around in varPause
-    nShifts = WEBRTC_SPL_MAX(0, 10 + inst->stages - WebRtcSpl_NormW32(tmp32no1));
-
-    varMagnUFX = 0;
-    varPauseUFX = 0;
-    covMagnPauseFX = 0;
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        // Compute var and cov of magn and magn_pause
-        tmp16no1 = (WebRtc_Word16)((WebRtc_Word32)magnIn[i] - avgMagnFX);
-        tmp32no2 = inst->avgMagnPause[i] - avgPauseFX;
-        varMagnUFX += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1); // Q(2*qMagn)
-        tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16no1); // Q(prevQMagn+qMagn)
-        covMagnPauseFX += tmp32no1; // Q(prevQMagn+qMagn)
-        tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, nShifts); // Q(prevQMagn-minPause)
-        varPauseUFX += (WebRtc_UWord32)WEBRTC_SPL_MUL(tmp32no1, tmp32no1); // Q(2*(prevQMagn-minPause))
-    }
-    //update of average magnitude spectrum: Q(-2*stages) and averaging replaced by shifts
-    inst->curAvgMagnEnergy += WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy, 2 * inst->normData
-                                                    + inst->stages - 1);
-
-    avgDiffNormMagnUFX = varMagnUFX; // Q(2*qMagn)
-    if ((varPauseUFX) && (covMagnPauseFX))
-    {
-        tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_ABS_W32(covMagnPauseFX); // Q(prevQMagn+qMagn)
-        norm32 = WebRtcSpl_NormU32(tmpU32no1) - 16;
-        if (norm32 > 0)
-        {
-            tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32); // Q(prevQMagn+qMagn+norm32)
-        } else
-        {
-            tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, -norm32); // Q(prevQMagn+qMagn+norm32)
-        }
-        tmpU32no2 = WEBRTC_SPL_UMUL(tmpU32no1, tmpU32no1); // Q(2*(prevQMagn+qMagn-norm32))
-
-        nShifts += norm32;
-        nShifts <<= 1;
-        if (nShifts < 0)
-        {
-            varPauseUFX >>= (-nShifts); // Q(2*(qMagn+norm32+minPause))
-            nShifts = 0;
-        }
-        tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no2, varPauseUFX); // Q(2*(qMagn+norm32-16+minPause))
-        tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, nShifts);
-
-        avgDiffNormMagnUFX -= WEBRTC_SPL_MIN(avgDiffNormMagnUFX, tmpU32no1); // Q(2*qMagn)
-    }
-    //normalize and compute time average update of difference feature
-    tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(avgDiffNormMagnUFX, 2 * inst->normData);
-    if (inst->featureSpecDiff > tmpU32no1)
-    {
-        tmpU32no2 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecDiff - tmpU32no1,
-                                          SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
-        inst->featureSpecDiff -= WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
-    } else
-    {
-        tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no1 - inst->featureSpecDiff,
-                                          SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
-        inst->featureSpecDiff += WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
-    }
-}
-
-// Compute speech/noise probability
-// speech/noise probability is returned in: probSpeechFinal
-//snrLocPrior is the prior SNR for each frequency (in Q11)
-//snrLocPost is the post SNR for each frequency (in Q11)
-void WebRtcNsx_SpeechNoiseProb(NsxInst_t *inst, WebRtc_UWord16 *nonSpeechProbFinal,
-                               WebRtc_UWord32 *priorLocSnr, WebRtc_UWord32 *postLocSnr)
-{
-    WebRtc_UWord32 zeros, num, den, tmpU32no1, tmpU32no2, tmpU32no3;
-
-    WebRtc_Word32 invLrtFX, indPriorFX, tmp32, tmp32no1, tmp32no2, besselTmpFX32;
-    WebRtc_Word32 frac32, logTmp;
-    WebRtc_Word32 logLrtTimeAvgKsumFX;
-
-    WebRtc_Word16 indPriorFX16;
-    WebRtc_Word16 tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac, intPart;
-
-    int i, normTmp, normTmp2, nShifts;
-
-    // compute feature based on average LR factor
-    // this is the average over all frequencies of the smooth log LRT
-    logLrtTimeAvgKsumFX = 0;
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        besselTmpFX32 = (WebRtc_Word32)postLocSnr[i]; // Q11
-        normTmp = WebRtcSpl_NormU32(postLocSnr[i]);
-        num = WEBRTC_SPL_LSHIFT_U32(postLocSnr[i], normTmp); // Q(11+normTmp)
-        if (normTmp > 10)
-        {
-            den = WEBRTC_SPL_LSHIFT_U32(priorLocSnr[i], normTmp - 11); // Q(normTmp)
-        } else
-        {
-            den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp)
-        }
-        besselTmpFX32 -= WEBRTC_SPL_UDIV(num, den); // Q11
-
-        // inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - log(snrLocPrior) - inst->logLrtTimeAvg[i]);
-        // Here, LRT_TAVG = 0.5
-        zeros = WebRtcSpl_NormU32(priorLocSnr[i]);
-        frac32 = (WebRtc_Word32)(((priorLocSnr[i] << zeros) & 0x7FFFFFFF) >> 19);
-        tmp32 = WEBRTC_SPL_MUL(frac32, frac32);
-        tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(tmp32, -43), 19);
-        tmp32 += WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)frac32, 5412, 12);
-        frac32 = tmp32 + 37;
-        // tmp32 = log2(priorLocSnr[i])
-        tmp32 = (WebRtc_Word32)(((31 - zeros) << 12) + frac32) - (11 << 12); // Q12
-        logTmp = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32, 178), 8); // log2(priorLocSnr[i])*log(2)
-        tmp32no1 = WEBRTC_SPL_RSHIFT_W32(logTmp + inst->logLrtTimeAvgW32[i], 1); // Q12
-        inst->logLrtTimeAvgW32[i] += (besselTmpFX32 - tmp32no1); // Q12
-
-        logLrtTimeAvgKsumFX += inst->logLrtTimeAvgW32[i]; // Q12
-    }
-    inst->featureLogLrt = WEBRTC_SPL_RSHIFT_W32(logLrtTimeAvgKsumFX * 5, inst->stages + 10); // 5 = BIN_SIZE_LRT / 2
-    // done with computation of LR factor
-
-    //
-    //compute the indicator functions
-    //
-
-    // average LRT feature
-    // FLOAT code
-    // indicator0 = 0.5 * (tanh(widthPrior * (logLrtTimeAvgKsum - threshPrior0)) + 1.0);
-    tmpIndFX = 16384; // Q14(1.0)
-    tmp32no1 = logLrtTimeAvgKsumFX - inst->thresholdLogLrt; // Q12
-    nShifts = 7 - inst->stages; // WIDTH_PR_MAP_SHIFT - inst->stages + 5;
-    //use larger width in tanh map for pause regions
-    if (tmp32no1 < 0)
-    {
-        tmpIndFX = 0;
-        tmp32no1 = -tmp32no1;
-        //widthPrior = widthPrior * 2.0;
-        nShifts++;
-    }
-    tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, nShifts); // Q14
-    // compute indicator function: sigmoid map
-    tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 14);
-    if ((tableIndex < 16) && (tableIndex >= 0))
-    {
-        tmp16no2 = kIndicatorTable[tableIndex];
-        tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
-        frac = (WebRtc_Word16)(tmp32no1 & 0x00003fff); // Q14
-        tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
-        if (tmpIndFX == 0)
-        {
-            tmpIndFX = 8192 - tmp16no2; // Q14
-        } else
-        {
-            tmpIndFX = 8192 + tmp16no2; // Q14
-        }
-    }
-    indPriorFX = WEBRTC_SPL_MUL_16_16(inst->weightLogLrt, tmpIndFX); // 6*Q14
-
-    //spectral flatness feature
-    if (inst->weightSpecFlat)
-    {
-        tmpU32no1 = WEBRTC_SPL_UMUL(inst->featureSpecFlat, 400); // Q10
-        tmpIndFX = 16384; // Q14(1.0)
-        //use larger width in tanh map for pause regions
-        tmpU32no2 = inst->thresholdSpecFlat - tmpU32no1; //Q10
-        nShifts = 4;
-        if (inst->thresholdSpecFlat < tmpU32no1)
-        {
-            tmpIndFX = 0;
-            tmpU32no2 = tmpU32no1 - inst->thresholdSpecFlat;
-            //widthPrior = widthPrior * 2.0;
-            nShifts++;
-        }
-        tmp32no1 = (WebRtc_Word32)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2,
-                                                                            nShifts), 25); //Q14
-        tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts), 25); //Q14
-        // compute indicator function: sigmoid map
-        // FLOAT code
-        // indicator1 = 0.5 * (tanh(sgnMap * widthPrior * (threshPrior1 - tmpFloat1)) + 1.0);
-        tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
-        if (tableIndex < 16)
-        {
-            tmp16no2 = kIndicatorTable[tableIndex];
-            tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
-            frac = (WebRtc_Word16)(tmpU32no1 & 0x00003fff); // Q14
-            tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
-            if (tmpIndFX)
-            {
-                tmpIndFX = 8192 + tmp16no2; // Q14
-            } else
-            {
-                tmpIndFX = 8192 - tmp16no2; // Q14
-            }
-        }
-        indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecFlat, tmpIndFX); // 6*Q14
-    }
-
-    //for template spectral-difference
-    if (inst->weightSpecDiff)
-    {
-        tmpU32no1 = 0;
-        if (inst->featureSpecDiff)
-        {
-            normTmp = WEBRTC_SPL_MIN(20 - inst->stages,
-                                     WebRtcSpl_NormU32(inst->featureSpecDiff));
-            tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp); // Q(normTmp-2*stages)
-            tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy, 20 - inst->stages
-                                              - normTmp);
-            if (tmpU32no2)
-            {
-                tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q14?? Q(20 - inst->stages)
-            } else
-            {
-                tmpU32no1 = (WebRtc_UWord32)(0x7fffffff);
-            }
-        }
-        tmpU32no3 = WEBRTC_SPL_UDIV(WEBRTC_SPL_LSHIFT_U32(inst->thresholdSpecDiff, 17), 25);
-        tmpU32no2 = tmpU32no1 - tmpU32no3;
-        nShifts = 1;
-        tmpIndFX = 16384; // Q14(1.0)
-        //use larger width in tanh map for pause regions
-        if (tmpU32no2 & 0x80000000)
-        {
-            tmpIndFX = 0;
-            tmpU32no2 = tmpU32no3 - tmpU32no1;
-            //widthPrior = widthPrior * 2.0;
-            nShifts--;
-        }
-        tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, nShifts);
-        // compute indicator function: sigmoid map
-        /* FLOAT code
-         indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0);
-         */
-        tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
-        if (tableIndex < 16)
-        {
-            tmp16no2 = kIndicatorTable[tableIndex];
-            tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
-            frac = (WebRtc_Word16)(tmpU32no1 & 0x00003fff); // Q14
-            tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16no1, frac,
-                                                                            14);
-            if (tmpIndFX)
-            {
-                tmpIndFX = 8192 + tmp16no2;
-            } else
-            {
-                tmpIndFX = 8192 - tmp16no2;
-            }
-        }
-        indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecDiff, tmpIndFX); // 6*Q14
-    }
-
-    //combine the indicator function with the feature weights
-    // FLOAT code
-    // indPrior = 1 - (weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2 * indicator2);
-    indPriorFX16 = WebRtcSpl_DivW32W16ResW16(98307 - indPriorFX, 6); // Q14
-    // done with computing indicator function
-
-    //compute the prior probability
-    // FLOAT code
-    // inst->priorNonSpeechProb += PRIOR_UPDATE * (indPriorNonSpeech - inst->priorNonSpeechProb);
-    tmp16 = indPriorFX16 - inst->priorNonSpeechProb; // Q14
-    inst->priorNonSpeechProb += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(PRIOR_UPDATE_Q14,
-                                                                         tmp16, 14); // Q14
-
-    //final speech probability: combine prior model with LR factor:
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        // FLOAT code
-        // invLrt = exp(inst->logLrtTimeAvg[i]);
-        // invLrt = inst->priorSpeechProb * invLrt;
-        // nonSpeechProbFinal[i] = (1.0 - inst->priorSpeechProb) / (1.0 - inst->priorSpeechProb + invLrt);
-        // invLrt = (1.0 - inst->priorNonSpeechProb) * invLrt;
-        // nonSpeechProbFinal[i] = inst->priorNonSpeechProb / (inst->priorNonSpeechProb + invLrt);
-        nonSpeechProbFinal[i] = 0; // Q8
-        if ((inst->logLrtTimeAvgW32[i] < 65300) && (inst->priorNonSpeechProb > 0))
-        {
-            tmp32no1 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(inst->logLrtTimeAvgW32[i], 23637),
-                                             14); // Q12
-            intPart = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 12);
-            if (intPart < -8)
-            {
-                intPart = -8;
-            }
-            frac = (WebRtc_Word16)(tmp32no1 & 0x00000fff); // Q12
-            // Quadratic approximation of 2^frac
-            tmp32no2 = WEBRTC_SPL_RSHIFT_W32(frac * frac * 44, 19); // Q12
-            tmp32no2 += WEBRTC_SPL_MUL_16_16_RSFT(frac, 84, 7); // Q12
-            invLrtFX = WEBRTC_SPL_LSHIFT_W32(1, 8 + intPart)
-                    + WEBRTC_SPL_SHIFT_W32(tmp32no2, intPart - 4); // Q8
-
-            normTmp = WebRtcSpl_NormW32(invLrtFX);
-            normTmp2 = WebRtcSpl_NormW16((16384 - inst->priorNonSpeechProb));
-            if (normTmp + normTmp2 < 15)
-            {
-                invLrtFX = WEBRTC_SPL_RSHIFT_W32(invLrtFX, 15 - normTmp2 - normTmp); // Q(normTmp+normTmp2-7)
-                tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb)); // Q(normTmp+normTmp2+7)
-                invLrtFX = WEBRTC_SPL_SHIFT_W32(tmp32no1, 7 - normTmp - normTmp2); // Q14
-            } else
-            {
-                tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb)); // Q22
-                invLrtFX = WEBRTC_SPL_RSHIFT_W32(tmp32no1, 8); // Q14
-            }
-
-            tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inst->priorNonSpeechProb, 8); // Q22
-            nonSpeechProbFinal[i] = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32no1,
-                    (WebRtc_Word32)inst->priorNonSpeechProb
-                    + invLrtFX); // Q8
-            if (7 - normTmp - normTmp2 > 0)
-            {
-                nonSpeechProbFinal[i] = 0; // Q8
-            }
-        }
-    }
-}
-
-// Transform input (speechFrame) to frequency domain magnitude (magnU16)
-void WebRtcNsx_DataAnalysis(NsxInst_t *inst, short *speechFrame, WebRtc_UWord16 *magnU16)
-{
-
-    WebRtc_UWord32 tmpU32no1, tmpU32no2;
-
-    WebRtc_Word32   tmp_1_w32 = 0;
-    WebRtc_Word32   tmp_2_w32 = 0;
-    WebRtc_Word32   sum_log_magn = 0;
-    WebRtc_Word32   sum_log_i_log_magn = 0;
-
-    WebRtc_UWord16  sum_log_magn_u16 = 0;
-    WebRtc_UWord16  tmp_u16 = 0;
-
-    WebRtc_Word16   sum_log_i = 0;
-    WebRtc_Word16   sum_log_i_square = 0;
-    WebRtc_Word16   frac = 0;
-    WebRtc_Word16   log2 = 0;
-    WebRtc_Word16   matrix_determinant = 0;
-    WebRtc_Word16   winData[ANAL_BLOCKL_MAX], maxWinData;
-    WebRtc_Word16   realImag[ANAL_BLOCKL_MAX << 1];
-
-    int i, j;
-    int outCFFT;
-    int zeros;
-    int net_norm = 0;
-    int right_shifts_in_magnU16 = 0;
-    int right_shifts_in_initMagnEst = 0;
-
-    // For lower band do all processing
-    // update analysis buffer for L band
-    WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer, inst->analysisBuffer + inst->blockLen10ms,
-                          inst->anaLen - inst->blockLen10ms);
-    WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer + inst->anaLen - inst->blockLen10ms,
-                          speechFrame, inst->blockLen10ms);
-
-    // Window data before FFT
-    for (i = 0; i < inst->anaLen; i++)
-    {
-        winData[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(inst->window[i],
-                inst->analysisBuffer[i],
-                14); // Q0
-    }
-    // Get input energy
-    inst->energyIn = WebRtcSpl_Energy(winData, (int)inst->anaLen, &(inst->scaleEnergyIn));
-
-    // Reset zero input flag
-    inst->zeroInputSignal = 0;
-    // Acquire norm for winData
-    maxWinData = WebRtcSpl_MaxAbsValueW16(winData, inst->anaLen);
-    inst->normData = WebRtcSpl_NormW16(maxWinData);
-    if (maxWinData == 0)
-    {
-        // Treat zero input separately.
-        inst->zeroInputSignal = 1;
-        return;
-    }
-
-    // Determine the net normalization in the frequency domain
-    net_norm = inst->stages - inst->normData;
-    // Track lowest normalization factor and use it to prevent wrap around in shifting
-    right_shifts_in_magnU16 = inst->normData - inst->minNorm;
-    right_shifts_in_initMagnEst = WEBRTC_SPL_MAX(-right_shifts_in_magnU16, 0);
-    inst->minNorm -= right_shifts_in_initMagnEst;
-    right_shifts_in_magnU16 = WEBRTC_SPL_MAX(right_shifts_in_magnU16, 0);
-
-    // create realImag as winData interleaved with zeros (= imag. part), normalize it
-    for (i = 0; i < inst->anaLen; i++)
-    {
-        j = WEBRTC_SPL_LSHIFT_W16(i, 1);
-        realImag[j] = WEBRTC_SPL_LSHIFT_W16(winData[i], inst->normData); // Q(normData)
-        realImag[j + 1] = 0; // Insert zeros in imaginary part
-    }
-
-    // bit-reverse position of elements in array and FFT the array
-    WebRtcSpl_ComplexBitReverse(realImag, inst->stages); // Q(normData-stages)
-    outCFFT = WebRtcSpl_ComplexFFT(realImag, inst->stages, 1);
-
-    inst->imag[0] = 0; // Q(normData-stages)
-    inst->imag[inst->anaLen2] = 0;
-    inst->real[0] = realImag[0]; // Q(normData-stages)
-    inst->real[inst->anaLen2] = realImag[inst->anaLen];
-    // Q(2*(normData-stages))
-    inst->magnEnergy = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(inst->real[0], inst->real[0]);
-    inst->magnEnergy += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(inst->real[inst->anaLen2],
-                                                             inst->real[inst->anaLen2]);
-    magnU16[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(inst->real[0]); // Q(normData-stages)
-    magnU16[inst->anaLen2] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(inst->real[inst->anaLen2]);
-    inst->sumMagn = (WebRtc_UWord32)magnU16[0]; // Q(normData-stages)
-    inst->sumMagn += (WebRtc_UWord32)magnU16[inst->anaLen2];
-
-    // Gather information during startup for noise parameter estimation
-    if (inst->blockIndex < END_STARTUP_SHORT)
-    {
-        // Switch initMagnEst to Q(minNorm-stages)
-        inst->initMagnEst[0] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[0],
-                                                     right_shifts_in_initMagnEst);
-        inst->initMagnEst[inst->anaLen2] =
-                WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[inst->anaLen2],
-                                      right_shifts_in_initMagnEst); // Q(minNorm-stages)
-
-        // Shift magnU16 to same domain as initMagnEst
-        tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[0],
-                                          right_shifts_in_magnU16); // Q(minNorm-stages)
-        tmpU32no2 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[inst->anaLen2],
-                                          right_shifts_in_magnU16); // Q(minNorm-stages)
-
-        // Update initMagnEst
-        inst->initMagnEst[0] += tmpU32no1; // Q(minNorm-stages)
-        inst->initMagnEst[inst->anaLen2] += tmpU32no2; // Q(minNorm-stages)
-
-        log2 = 0;
-        if (magnU16[inst->anaLen2])
-        {
-            // Calculate log2(magnU16[inst->anaLen2])
-            zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magnU16[inst->anaLen2]);
-            frac = (WebRtc_Word16)((((WebRtc_UWord32)magnU16[inst->anaLen2] << zeros) &
-                    0x7FFFFFFF) >> 23); // Q8
-            // log2(magnU16(i)) in Q8
-            log2 = (WebRtc_Word16)(((31 - zeros) << 8) + kLogTableFrac[frac]);
-        }
-
-        sum_log_magn = (WebRtc_Word32)log2; // Q8
-        // sum_log_i_log_magn in Q17
-        sum_log_i_log_magn = (WEBRTC_SPL_MUL_16_16(kLogIndex[inst->anaLen2], log2) >> 3);
-    }
-
-    for (i = 1; i < inst->anaLen2; i++)
-    {
-        j = WEBRTC_SPL_LSHIFT_W16(i, 1);
-        inst->real[i] = realImag[j];
-        inst->imag[i] = -realImag[j + 1];
-        // magnitude spectrum
-        // energy in Q(2*(normData-stages))
-        tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j], realImag[j]);
-        tmpU32no1 += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j + 1], realImag[j + 1]);
-        inst->magnEnergy += tmpU32no1; // Q(2*(normData-stages))
-
-        magnU16[i] = (WebRtc_UWord16)WebRtcSpl_Sqrt(tmpU32no1); // Q(normData-stages)
-        inst->sumMagn += (WebRtc_UWord32)magnU16[i]; // Q(normData-stages)
-        if (inst->blockIndex < END_STARTUP_SHORT)
-        {
-            // Switch initMagnEst to Q(minNorm-stages)
-            inst->initMagnEst[i] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i],
-                                                         right_shifts_in_initMagnEst);
-
-            // Shift magnU16 to same domain as initMagnEst, i.e., Q(minNorm-stages)
-            tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[i],
-                                              right_shifts_in_magnU16);
-            // Update initMagnEst
-            inst->initMagnEst[i] += tmpU32no1; // Q(minNorm-stages)
-
-            if (i >= kStartBand)
-            {
-                // For pink noise estimation. Collect data neglecting lower frequency band
-                log2 = 0;
-                if (magnU16[i])
-                {
-                    zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magnU16[i]);
-                    frac = (WebRtc_Word16)((((WebRtc_UWord32)magnU16[i] << zeros) &
-                            0x7FFFFFFF) >> 23);
-                    // log2(magnU16(i)) in Q8
-                    log2 = (WebRtc_Word16)(((31 - zeros) << 8) + kLogTableFrac[frac]);
-                }
-                sum_log_magn += (WebRtc_Word32)log2; // Q8
-                // sum_log_i_log_magn in Q17
-                sum_log_i_log_magn += (WEBRTC_SPL_MUL_16_16(kLogIndex[i], log2) >> 3);
-            }
-        }
-    }
-
-    //compute simplified noise model during startup
-    if (inst->blockIndex < END_STARTUP_SHORT)
-    {
-        // Estimate White noise
-        // Switch whiteNoiseLevel to Q(minNorm-stages)
-        inst->whiteNoiseLevel = WEBRTC_SPL_RSHIFT_U32(inst->whiteNoiseLevel,
-                                                      right_shifts_in_initMagnEst);
-
-        // Update the average magnitude spectrum, used as noise estimate.
-        tmpU32no1 = WEBRTC_SPL_UMUL_32_16(inst->sumMagn, inst->overdrive);
-        tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, inst->stages + 8);
-
-        // Replacing division above with 'stages' shifts
-        // Shift to same Q-domain as whiteNoiseLevel
-        tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, right_shifts_in_magnU16);
-        // This operation is safe from wrap around as long as END_STARTUP_SHORT < 128
-        assert(END_STARTUP_SHORT < 128);
-        inst->whiteNoiseLevel += tmpU32no1; // Q(minNorm-stages)
-
-        // Estimate Pink noise parameters
-        // Denominator used in both parameter estimates.
-        // The value is only dependent on the size of the frequency band (kStartBand)
-        // and to reduce computational complexity stored in a table (kDeterminantEstMatrix[])
-        matrix_determinant = kDeterminantEstMatrix[kStartBand]; // Q0
-        sum_log_i = kSumLogIndex[kStartBand]; // Q5
-        sum_log_i_square = kSumSquareLogIndex[kStartBand]; // Q2
-        if (inst->fs == 8000)
-        {
-            // Adjust values to shorter blocks in narrow band.
-            tmp_1_w32 = (WebRtc_Word32)matrix_determinant;
-            tmp_1_w32 += WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], sum_log_i, 9);
-            tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], kSumLogIndex[65], 10);
-            tmp_1_w32 -= WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)sum_log_i_square, 4);
-            tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)(inst->magnLen
-                    - kStartBand), kSumSquareLogIndex[65], 2);
-            matrix_determinant = (WebRtc_Word16)tmp_1_w32;
-            sum_log_i -= kSumLogIndex[65]; // Q5
-            sum_log_i_square -= kSumSquareLogIndex[65]; // Q2
-        }
-
-        // Necessary number of shifts to fit sum_log_magn in a word16
-        zeros = 16 - WebRtcSpl_NormW32(sum_log_magn);
-        if (zeros < 0)
-        {
-            zeros = 0;
-        }
-        tmp_1_w32 = WEBRTC_SPL_LSHIFT_W32(sum_log_magn, 1); // Q9
-        sum_log_magn_u16 = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W32(tmp_1_w32, zeros);//Q(9-zeros)
-
-        // Calculate and update pinkNoiseNumerator. Result in Q11.
-        tmp_2_w32 = WEBRTC_SPL_MUL_16_U16(sum_log_i_square, sum_log_magn_u16); // Q(11-zeros)
-        tmpU32no1 = WEBRTC_SPL_RSHIFT_U32((WebRtc_UWord32)sum_log_i_log_magn, 12); // Q5
-
-        // Shift the largest value of sum_log_i and tmp32no3 before multiplication
-        tmp_u16 = WEBRTC_SPL_LSHIFT_U16((WebRtc_UWord16)sum_log_i, 1); // Q6
-        if ((WebRtc_UWord32)sum_log_i > tmpU32no1)
-        {
-            tmp_u16 = WEBRTC_SPL_RSHIFT_U16(tmp_u16, zeros);
-        }
-        else
-        {
-            tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, zeros);
-        }
-        tmp_2_w32 -= (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(tmpU32no1, tmp_u16); // Q(11-zeros)
-        matrix_determinant = WEBRTC_SPL_RSHIFT_W16(matrix_determinant, zeros); // Q(-zeros)
-        tmp_2_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q11
-        tmp_2_w32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)net_norm, 11); // Q11
-        if (tmp_2_w32 < 0)
-        {
-            tmp_2_w32 = 0;
-        }
-        inst->pinkNoiseNumerator += tmp_2_w32; // Q11
-
-        // Calculate and update pinkNoiseExp. Result in Q14.
-        tmp_2_w32 = WEBRTC_SPL_MUL_16_U16(sum_log_i, sum_log_magn_u16); // Q(14-zeros)
-        tmp_1_w32 = WEBRTC_SPL_RSHIFT_W32(sum_log_i_log_magn, 3 + zeros);
-        tmp_1_w32 = WEBRTC_SPL_MUL((WebRtc_Word32)(inst->magnLen - kStartBand),
-                                   tmp_1_w32);
-        tmp_2_w32 -= tmp_1_w32; // Q(14-zeros)
-        if (tmp_2_w32 > 0)
-        {
-            // If the exponential parameter is negative force it to zero, which means a
-            // flat spectrum.
-            tmp_1_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q14
-            inst->pinkNoiseExp += WEBRTC_SPL_SAT(16384, tmp_1_w32, 0); // Q14
-        }
-    }
-}
-
-void WebRtcNsx_DataSynthesis(NsxInst_t *inst, short *outFrame)
-{
-    WebRtc_Word32 tmp32no1;
-    WebRtc_Word32 energyOut;
-
-    WebRtc_Word16 realImag[ANAL_BLOCKL_MAX << 1];
-    WebRtc_Word16 tmp16no1, tmp16no2;
-    WebRtc_Word16 energyRatio;
-    WebRtc_Word16 gainFactor, gainFactor1, gainFactor2;
-
-    int i, j;
-    int outCIFFT;
-    int scaleEnergyOut = 0;
-
-    if (inst->zeroInputSignal)
-    {
-        // synthesize the special case of zero input
-        // read out fully processed segment
-        for (i = 0; i < inst->blockLen10ms; i++)
-        {
-            outFrame[i] = inst->synthesisBuffer[i]; // Q0
-        }
-        // update synthesis buffer
-        WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
-                              inst->synthesisBuffer + inst->blockLen10ms,
-                              inst->anaLen - inst->blockLen10ms);
-        WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer + inst->anaLen - inst->blockLen10ms,
-                                inst->blockLen10ms);
-        return;
-    }
-    // Filter the data in the frequency domain
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        inst->real[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->real[i],
-                (WebRtc_Word16)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
-        inst->imag[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->imag[i],
-                (WebRtc_Word16)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
-    }
-    // back to time domain
-    // Create spectrum
-    realImag[0] = inst->real[0];
-    realImag[1] = -inst->imag[0];
-    for (i = 1; i < inst->anaLen2; i++)
-    {
-        j = WEBRTC_SPL_LSHIFT_W16(i, 1);
-        tmp16no1 = (inst->anaLen << 1) - j;
-        realImag[j] = inst->real[i];
-        realImag[j + 1] = -inst->imag[i];
-        realImag[tmp16no1] = inst->real[i];
-        realImag[tmp16no1 + 1] = inst->imag[i];
-    }
-    realImag[inst->anaLen] = inst->real[inst->anaLen2];
-    realImag[inst->anaLen + 1] = -inst->imag[inst->anaLen2];
-
-    // bit-reverse position of elements in array and IFFT it
-    WebRtcSpl_ComplexBitReverse(realImag, inst->stages);
-    outCIFFT = WebRtcSpl_ComplexIFFT(realImag, inst->stages, 1);
-
-    for (i = 0; i < inst->anaLen; i++)
-    {
-        j = WEBRTC_SPL_LSHIFT_W16(i, 1);
-        tmp32no1 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)realImag[j], outCIFFT - inst->normData);
-        inst->real[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, tmp32no1,
-                                                      WEBRTC_SPL_WORD16_MIN);
-    }
-
-    //scale factor: only do it after END_STARTUP_LONG time
-    gainFactor = 8192; // 8192 = Q13(1.0)
-    if (inst->gainMap == 1 &&
-        inst->blockIndex > END_STARTUP_LONG &&
-        inst->energyIn > 0)
-    {
-        energyOut = WebRtcSpl_Energy(inst->real, (int)inst->anaLen, &scaleEnergyOut); // Q(-scaleEnergyOut)
-        if (scaleEnergyOut == 0 && !(energyOut & 0x7f800000))
-        {
-            energyOut = WEBRTC_SPL_SHIFT_W32(energyOut, 8 + scaleEnergyOut
-                                             - inst->scaleEnergyIn);
-        } else
-        {
-            inst->energyIn = WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 8 + scaleEnergyOut
-                                                   - inst->scaleEnergyIn); // Q(-8-scaleEnergyOut)
-        }
-
-        assert(inst->energyIn > 0);
-        energyRatio = (WebRtc_Word16)WEBRTC_SPL_DIV(energyOut
-                + WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 1), inst->energyIn); // Q8
-
-        //         // original FLOAT code
-        //         if (gain > blim) {
-        //         factor1=1.0+1.3*(gain-blim);
-        //         if (gain*factor1 > 1.0) { // FLOAT
-        //         factor1 = 1.0/gain; // FLOAT
-        //         }
-        //         }
-        //         else {
-        //         factor1=1.0; // FLOAT
-        //         }
-        //
-        //         if (gain > blim) {
-        //         factor2=1.0; //FLOAT
-        //         }
-        //         else {
-        //         //don't reduce scale too much for pause regions: attenuation here should be controlled by flooring
-        //         factor2=1.0-0.3*(blim-gain); // FLOAT
-        //         if (gain <= inst->denoiseBound) {
-        //         factor2=1.0-0.3*(blim-inst->denoiseBound); // FLOAT
-        //         }
-        //         }
-
-        // all done in lookup tables now
-        gainFactor1 = kFactor1Table[energyRatio]; // Q8
-        gainFactor2 = inst->factor2Table[energyRatio]; // Q8
-
-        //combine both scales with speech/noise prob: note prior (priorSpeechProb) is not frequency dependent
-
-        // factor = inst->priorSpeechProb*factor1 + (1.0-inst->priorSpeechProb)*factor2; // original code
-        tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(16384 - inst->priorNonSpeechProb,
-                                                            gainFactor1, 14); // Q13 16384 = Q14(1.0)
-        tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->priorNonSpeechProb,
-                                                            gainFactor2, 14); // Q13;
-        gainFactor = tmp16no1 + tmp16no2; // Q13
-    } // out of flag_gain_map==1
-
-    // synthesis
-    for (i = 0; i < inst->anaLen; i++)
-    {
-        tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(inst->window[i],
-                                                                       inst->real[i], 14); // Q0, window in Q14
-        tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16no1, gainFactor, 13); // Q0
-        // Down shift with rounding
-        tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, tmp32no1,
-                                                 WEBRTC_SPL_WORD16_MIN); // Q0
-        inst->synthesisBuffer[i] = WEBRTC_SPL_ADD_SAT_W16(inst->synthesisBuffer[i], tmp16no2); // Q0
-    }
-
-    // read out fully processed segment
-    for (i = 0; i < inst->blockLen10ms; i++)
-    {
-        outFrame[i] = inst->synthesisBuffer[i]; // Q0
-    }
-    // update synthesis buffer
-    WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer, inst->synthesisBuffer + inst->blockLen10ms,
-                          inst->anaLen - inst->blockLen10ms);
-    WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer + inst->anaLen - inst->blockLen10ms,
-                            inst->blockLen10ms);
-}
-
-int WebRtcNsx_ProcessCore(NsxInst_t *inst, short *speechFrame, short *speechFrameHB,
-                          short *outFrame, short *outFrameHB)
-{
-    // main routine for noise suppression
-
-    WebRtc_UWord32 tmpU32no1, tmpU32no2, tmpU32no3;
-    WebRtc_UWord32 satMax, maxNoiseU32;
-    WebRtc_UWord32 tmpMagnU32, tmpNoiseU32;
-    WebRtc_UWord32 nearMagnEst;
-    WebRtc_UWord32 noiseUpdateU32;
-    WebRtc_UWord32 noiseU32[HALF_ANAL_BLOCKL];
-    WebRtc_UWord32 postLocSnr[HALF_ANAL_BLOCKL];
-    WebRtc_UWord32 priorLocSnr[HALF_ANAL_BLOCKL];
-    WebRtc_UWord32 prevNearSnr[HALF_ANAL_BLOCKL];
-    WebRtc_UWord32 curNearSnr;
-    WebRtc_UWord32 priorSnr;
-    WebRtc_UWord32 noise_estimate = 0;
-    WebRtc_UWord32 noise_estimate_avg = 0;
-    WebRtc_UWord32 numerator = 0;
-
-    WebRtc_Word32 tmp32no1, tmp32no2;
-    WebRtc_Word32 pink_noise_num_avg = 0;
-
-    WebRtc_UWord16 tmpU16no1;
-    WebRtc_UWord16 magnU16[HALF_ANAL_BLOCKL];
-    WebRtc_UWord16 prevNoiseU16[HALF_ANAL_BLOCKL];
-    WebRtc_UWord16 nonSpeechProbFinal[HALF_ANAL_BLOCKL];
-    WebRtc_UWord16 gammaNoise, prevGammaNoise;
-    WebRtc_UWord16 noiseSupFilterTmp[HALF_ANAL_BLOCKL];
-
-    WebRtc_Word16 qMagn, qNoise;
-    WebRtc_Word16 avgProbSpeechHB, gainModHB, avgFilterGainHB, gainTimeDomainHB;
-    WebRtc_Word16 tmp16no1;
-    WebRtc_Word16 int_part = 0;
-    WebRtc_Word16 frac_part = 0;
-    WebRtc_Word16 pink_noise_exp_avg = 0;
-
-    int i;
-    int nShifts, postShifts;
-    int norm32no1, norm32no2;
-    int flag, sign;
-    int q_domain_to_use = 0;
-
-#ifdef NS_FILEDEBUG
-    fwrite(spframe, sizeof(short), inst->blockLen10ms, inst->infile);
-#endif
-
-    // Check that initialization has been done
-    if (inst->initFlag != 1)
-    {
-        return -1;
-    }
-    // Check for valid pointers based on sampling rate
-    if ((inst->fs == 32000) && (speechFrameHB == NULL))
-    {
-        return -1;
-    }
-
-    // Store speechFrame and transform to frequency domain
-    WebRtcNsx_DataAnalysis(inst, speechFrame, magnU16);
-
-    if (inst->zeroInputSignal)
-    {
-        WebRtcNsx_DataSynthesis(inst, outFrame);
-
-        if (inst->fs == 32000)
-        {
-            // update analysis buffer for H band
-            // append new data to buffer FX
-            WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX, inst->dataBufHBFX + inst->blockLen10ms,
-                                  inst->anaLen - inst->blockLen10ms);
-            WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX + inst->anaLen - inst->blockLen10ms,
-                                  speechFrameHB, inst->blockLen10ms);
-            for (i = 0; i < inst->blockLen10ms; i++)
-            {
-                outFrameHB[i] = inst->dataBufHBFX[i]; // Q0
-            }
-        } // end of H band gain computation
-        return 0;
-    }
-
-    // Update block index when we have something to process
-    inst->blockIndex++;
-    //
-
-    // Norm of magn
-    qMagn = inst->normData - inst->stages;
-
-    // Compute spectral flatness on input spectrum
-    WebRtcNsx_ComputeSpectralFlatness(inst, magnU16);
-
-    // quantile noise estimate
-    WebRtcNsx_NoiseEstimation(inst, magnU16, noiseU32, &qNoise);
-
-    //noise estimate from previous frame
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        prevNoiseU16[i] = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], 11); // Q(prevQNoise)
-    }
-
-    if (inst->blockIndex < END_STARTUP_SHORT)
-    {
-        // Noise Q-domain to be used later; see description at end of section.
-        q_domain_to_use = WEBRTC_SPL_MIN((int)qNoise, inst->minNorm - inst->stages);
-
-        // Calculate frequency independent parts in parametric noise estimate and calculate
-        // the estimate for the lower frequency band (same values for all frequency bins)
-        if (inst->pinkNoiseExp)
-        {
-            pink_noise_exp_avg = (WebRtc_Word16)WebRtcSpl_DivW32W16(inst->pinkNoiseExp,
-                                                (WebRtc_Word16)(inst->blockIndex + 1)); // Q14
-            pink_noise_num_avg = WebRtcSpl_DivW32W16(inst->pinkNoiseNumerator,
-                                 (WebRtc_Word16)(inst->blockIndex + 1)); // Q11
-            WebRtcNsx_CalcParametricNoiseEstimate(inst,
-                                                  pink_noise_exp_avg,
-                                                  pink_noise_num_avg,
-                                                  kStartBand,
-                                                  &noise_estimate,
-                                                  &noise_estimate_avg);
-        }
-        else
-        {
-            // Use white noise estimate if we have poor pink noise parameter estimates
-            noise_estimate = inst->whiteNoiseLevel; // Q(minNorm-stages)
-            noise_estimate_avg = noise_estimate / (inst->blockIndex + 1); // Q(minNorm-stages)
-        }
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            // Estimate the background noise using the pink noise parameters if permitted
-            if ((inst->pinkNoiseExp) && (i >= kStartBand))
-            {
-                // Reset noise_estimate
-                noise_estimate = 0;
-                noise_estimate_avg = 0;
-                // Calculate the parametric noise estimate for current frequency bin
-                WebRtcNsx_CalcParametricNoiseEstimate(inst,
-                                                      pink_noise_exp_avg,
-                                                      pink_noise_num_avg,
-                                                      i,
-                                                      &noise_estimate,
-                                                      &noise_estimate_avg);
-            }
-            // Calculate parametric Wiener filter
-            noiseSupFilterTmp[i] = inst->denoiseBound;
-            if (inst->initMagnEst[i])
-            {
-                // numerator = (initMagnEst - noise_estimate * overdrive)
-                // Result in Q(8+minNorm-stages)
-                tmpU32no1 = WEBRTC_SPL_UMUL_32_16(noise_estimate, inst->overdrive);
-                numerator = WEBRTC_SPL_LSHIFT_U32(inst->initMagnEst[i], 8);
-                if (numerator > tmpU32no1)
-                {
-                    // Suppression filter coefficient larger than zero, so calculate.
-                    numerator -= tmpU32no1;
-
-                    // Determine number of left shifts in numerator for best accuracy after
-                    // division
-                    nShifts = WebRtcSpl_NormU32(numerator);
-                    nShifts = WEBRTC_SPL_SAT(6, nShifts, 0);
-
-                    // Shift numerator to Q(nShifts+8+minNorm-stages)
-                    numerator = WEBRTC_SPL_LSHIFT_U32(numerator, nShifts);
-
-                    // Shift denominator to Q(nShifts-6+minNorm-stages)
-                    tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i], 6 - nShifts);
-                    tmpU32no2 = WEBRTC_SPL_UDIV(numerator, tmpU32no1); // Q14
-                    noiseSupFilterTmp[i] = (WebRtc_UWord16)WEBRTC_SPL_SAT(16384, tmpU32no2,
-                                           (WebRtc_UWord32)(inst->denoiseBound)); // Q14
-                }
-            }
-            // Weight quantile noise 'noiseU32' with modeled noise 'noise_estimate_avg'
-            // 'noiseU32 is in Q(qNoise) and 'noise_estimate' in Q(minNorm-stages)
-            // To guarantee that we do not get wrap around when shifting to the same domain
-            // we use the lowest one. Furthermore, we need to save 6 bits for the weighting.
-            // 'noise_estimate_avg' can handle this operation by construction, but 'noiseU32'
-            // may not.
-
-            // Shift 'noiseU32' to 'q_domain_to_use'
-            tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], (int)qNoise - q_domain_to_use);
-            // Shift 'noise_estimate_avg' to 'q_domain_to_use'
-            tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noise_estimate_avg, inst->minNorm - inst->stages
-                                              - q_domain_to_use);
-            // Make a simple check to see if we have enough room for weighting 'tmpU32no1'
-            // without wrap around
-            nShifts = 0;
-            if (tmpU32no1 & 0xfc000000) {
-                tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 6);
-                tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6);
-                nShifts = 6;
-            }
-            // Add them together and divide by startup length
-            noiseU32[i] = WebRtcSpl_DivU32U16(tmpU32no1 + tmpU32no2, END_STARTUP_SHORT);
-            // Shift back if necessary
-            noiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], nShifts);
-        }
-        // Update new Q-domain for 'noiseU32'
-        qNoise = q_domain_to_use;
-    }
-    // compute average signal during END_STARTUP_LONG time:
-    // used to normalize spectral difference measure
-    if (inst->blockIndex < END_STARTUP_LONG)
-    {
-        // substituting division with shift ending up in Q(-2*stages)
-        inst->timeAvgMagnEnergyTmp
-                += WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy,
-                                         2 * inst->normData + inst->stages - 1);
-        inst->timeAvgMagnEnergy = WebRtcSpl_DivU32U16(inst->timeAvgMagnEnergyTmp,
-                                                      inst->blockIndex + 1);
-    }
-
-    //start processing at frames == converged+1
-    // STEP 1: compute prior and post SNR based on quantile noise estimates
-
-    // compute direct decision (DD) estimate of prior SNR: needed for new method
-    satMax = (WebRtc_UWord32)1048575;// Largest possible value without getting overflow despite shifting 12 steps
-    postShifts = 6 + qMagn - qNoise;
-    nShifts = 5 - inst->prevQMagn + inst->prevQNoise;
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        // FLOAT:
-        // post SNR
-        // postLocSnr[i] = 0.0;
-        // if (magn[i] > noise[i])
-        // {
-        //   postLocSnr[i] = magn[i] / (noise[i] + 0.0001);
-        // }
-        // // previous post SNR
-        // // previous estimate: based on previous frame with gain filter (smooth is previous filter)
-        //
-        // prevNearSnr[i] = inst->prevMagnU16[i] / (inst->noisePrev[i] + 0.0001) * (inst->smooth[i]);
-        //
-        // // DD estimate is sum of two terms: current estimate and previous estimate
-        // // directed decision update of priorSnr (or we actually store [2*priorSnr+1])
-        //
-        // priorLocSnr[i] = DD_PR_SNR * prevNearSnr[i] + (1.0 - DD_PR_SNR) * (postLocSnr[i] - 1.0);
-
-        // calculate post SNR: output in Q11
-        postLocSnr[i] = 2048; // 1.0 in Q11
-        tmpU32no1 = WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)magnU16[i], 6); // Q(6+qMagn)
-        if (postShifts < 0)
-        {
-            tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], -postShifts); // Q(6+qMagn)
-        } else
-        {
-            tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], postShifts); // Q(6+qMagn)
-        }
-        if (tmpU32no1 > tmpU32no2)
-        {
-            // Current magnitude larger than noise
-            tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, 11); // Q(17+qMagn)
-            if (tmpU32no2)
-            {
-                tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
-                postLocSnr[i] = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
-            } else
-            {
-                postLocSnr[i] = satMax;
-            }
-        }
-
-        // calculate prevNearSnr[i] and save for later instead of recalculating it later
-        nearMagnEst = WEBRTC_SPL_UMUL_16_16(inst->prevMagnU16[i], inst->noiseSupFilter[i]); // Q(prevQMagn+14)
-        tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(nearMagnEst, 3); // Q(prevQMagn+17)
-        tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], nShifts); // Q(prevQMagn+6)
-
-        if (tmpU32no2)
-        {
-            tmpU32no1 = WEBRTC_SPL_DIV(tmpU32no1, tmpU32no2); // Q11
-            tmpU32no1 = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
-        } else
-        {
-            tmpU32no1 = satMax; // Q11
-        }
-        prevNearSnr[i] = tmpU32no1; // Q11
-
-        //directed decision update of priorSnr
-        tmpU32no1 = WEBRTC_SPL_UMUL_32_16(prevNearSnr[i], DD_PR_SNR_Q11); // Q22
-        tmpU32no2 = WEBRTC_SPL_UMUL_32_16(postLocSnr[i] - 2048, ONE_MINUS_DD_PR_SNR_Q11); // Q22
-        priorSnr = tmpU32no1 + tmpU32no2 + 512; // Q22 (added 512 for rounding)
-        // priorLocSnr = 1 + 2*priorSnr
-        priorLocSnr[i] = 2048 + WEBRTC_SPL_RSHIFT_U32(priorSnr, 10); // Q11
-    } // end of loop over frequencies
-    // done with step 1: DD computation of prior and post SNR
-
-    // STEP 2: compute speech/noise likelihood
-
-    //compute difference of input spectrum with learned/estimated noise spectrum
-    WebRtcNsx_ComputeSpectralDifference(inst, magnU16);
-    //compute histograms for determination of parameters (thresholds and weights for features)
-    //parameters are extracted once every window time (=inst->modelUpdate)
-    //counter update
-    inst->cntThresUpdate++;
-    flag = (int)(inst->cntThresUpdate == inst->modelUpdate);
-    //update histogram
-    WebRtcNsx_FeatureParameterExtraction(inst, flag);
-    //compute model parameters
-    if (flag)
-    {
-        inst->cntThresUpdate = 0; // Reset counter
-        //update every window:
-        // get normalization for spectral difference for next window estimate
-
-        // Shift to Q(-2*stages)
-        inst->curAvgMagnEnergy = WEBRTC_SPL_RSHIFT_U32(inst->curAvgMagnEnergy, STAT_UPDATES);
-
-        tmpU32no1 = (inst->curAvgMagnEnergy + inst->timeAvgMagnEnergy + 1) >> 1; //Q(-2*stages)
-        // Update featureSpecDiff
-        if ((tmpU32no1 != inst->timeAvgMagnEnergy) && (inst->featureSpecDiff))
-        {
-            norm32no1 = 0;
-            tmpU32no3 = tmpU32no1;
-            while (0xFFFF0000 & tmpU32no3)
-            {
-                tmpU32no3 >>= 1;
-                norm32no1++;
-            }
-            tmpU32no2 = inst->featureSpecDiff;
-            while (0xFFFF0000 & tmpU32no2)
-            {
-                tmpU32no2 >>= 1;
-                norm32no1++;
-            }
-            tmpU32no3 = WEBRTC_SPL_UMUL(tmpU32no3, tmpU32no2);
-            tmpU32no3 = WEBRTC_SPL_UDIV(tmpU32no3, inst->timeAvgMagnEnergy);
-            if (WebRtcSpl_NormU32(tmpU32no3) < norm32no1)
-            {
-                inst->featureSpecDiff = 0x007FFFFF;
-            } else
-            {
-                inst->featureSpecDiff = WEBRTC_SPL_MIN(0x007FFFFF,
-                                        WEBRTC_SPL_LSHIFT_U32(tmpU32no3, norm32no1));
-            }
-        }
-
-        inst->timeAvgMagnEnergy = tmpU32no1; // Q(-2*stages)
-        inst->curAvgMagnEnergy = 0;
-    }
-
-    //compute speech/noise probability
-    WebRtcNsx_SpeechNoiseProb(inst, nonSpeechProbFinal, priorLocSnr, postLocSnr);
-
-    //time-avg parameter for noise update
-    gammaNoise = NOISE_UPDATE_Q8; // Q8
-
-    maxNoiseU32 = 0;
-    postShifts = inst->prevQNoise - qMagn;
-    nShifts = inst->prevQMagn - qMagn;
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        // temporary noise update: use it for speech frames if update value is less than previous
-        // the formula has been rewritten into:
-        // noiseUpdate = noisePrev[i] + (1 - gammaNoise) * nonSpeechProb * (magn[i] - noisePrev[i])
-
-        if (postShifts < 0)
-        {
-            tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(magnU16[i], -postShifts); // Q(prevQNoise)
-        } else
-        {
-            tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], postShifts); // Q(prevQNoise)
-        }
-        if (prevNoiseU16[i] > tmpU32no2)
-        {
-            sign = -1;
-            tmpU32no1 = prevNoiseU16[i] - tmpU32no2;
-        } else
-        {
-            sign = 1;
-            tmpU32no1 = tmpU32no2 - prevNoiseU16[i];
-        }
-        noiseUpdateU32 = inst->prevNoiseU32[i]; // Q(prevQNoise+11)
-        tmpU32no3 = 0;
-        if ((tmpU32no1) && (nonSpeechProbFinal[i]))
-        {
-            // This value will be used later, if gammaNoise changes
-            tmpU32no3 = WEBRTC_SPL_UMUL_32_16(tmpU32no1, nonSpeechProbFinal[i]); // Q(prevQNoise+8)
-            if (0x7c000000 & tmpU32no3)
-            {
-                // Shifting required before multiplication
-                tmpU32no2
-                        = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
-            } else
-            {
-                // We can do shifting after multiplication
-                tmpU32no2
-                        = WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
-            }
-            if (sign > 0)
-            {
-                noiseUpdateU32 += tmpU32no2; // Q(prevQNoise+11)
-            } else
-            {
-                // This operation is safe. We can never get wrap around, since worst
-                // case scenario means magnU16 = 0
-                noiseUpdateU32 -= tmpU32no2; // Q(prevQNoise+11)
-            }
-        }
-
-        //increase gamma (i.e., less noise update) for frame likely to be speech
-        prevGammaNoise = gammaNoise;
-        gammaNoise = NOISE_UPDATE_Q8;
-        //time-constant based on speech/noise state
-        //increase gamma (i.e., less noise update) for frames likely to be speech
-        if (nonSpeechProbFinal[i] < ONE_MINUS_PROB_RANGE_Q8)
-        {
-            gammaNoise = GAMMA_NOISE_TRANS_AND_SPEECH_Q8;
-        }
-
-        if (prevGammaNoise != gammaNoise)
-        {
-            // new noise update
-            // this line is the same as above, only that the result is stored in a different variable and the gammaNoise
-            // has changed
-            //
-            // noiseUpdate = noisePrev[i] + (1 - gammaNoise) * nonSpeechProb * (magn[i] - noisePrev[i])
-
-            if (0x7c000000 & tmpU32no3)
-            {
-                // Shifting required before multiplication
-                tmpU32no2
-                        = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
-            } else
-            {
-                // We can do shifting after multiplication
-                tmpU32no2
-                        = WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
-            }
-            if (sign > 0)
-            {
-                tmpU32no1 = inst->prevNoiseU32[i] + tmpU32no2; // Q(prevQNoise+11)
-            } else
-            {
-                tmpU32no1 = inst->prevNoiseU32[i] - tmpU32no2; // Q(prevQNoise+11)
-            }
-            if (noiseUpdateU32 > tmpU32no1)
-            {
-                noiseUpdateU32 = tmpU32no1; // Q(prevQNoise+11)
-            }
-        }
-        noiseU32[i] = noiseUpdateU32; // Q(prevQNoise+11)
-        if (noiseUpdateU32 > maxNoiseU32)
-        {
-            maxNoiseU32 = noiseUpdateU32;
-        }
-
-        // conservative noise update
-        // // original FLOAT code
-        // if (prob_speech < PROB_RANGE) {
-        // inst->avgMagnPause[i] = inst->avgMagnPause[i] + (1.0 - gamma_pause)*(magn[i] - inst->avgMagnPause[i]);
-        // }
-
-        tmp32no2 = WEBRTC_SPL_SHIFT_W32(inst->avgMagnPause[i], -nShifts);
-        if (nonSpeechProbFinal[i] > ONE_MINUS_PROB_RANGE_Q8)
-        {
-            if (nShifts < 0)
-            {
-                tmp32no1 = (WebRtc_Word32)magnU16[i] - tmp32no2; // Q(qMagn)
-                tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
-                tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + 128, 8); // Q(qMagn)
-            } else
-            {
-                tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)magnU16[i], nShifts)
-                        - inst->avgMagnPause[i]; // Q(qMagn+nShifts)
-                tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
-                tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + (128 << nShifts), 8 + nShifts); // Q(qMagn)
-            }
-            tmp32no2 += tmp32no1; // Q(qMagn)
-        }
-        inst->avgMagnPause[i] = tmp32no2;
-    } // end of frequency loop
-
-    norm32no1 = WebRtcSpl_NormU32(maxNoiseU32);
-    qNoise = inst->prevQNoise + norm32no1 - 5;
-    // done with step 2: noise update
-
-    // STEP 3: compute dd update of prior snr and post snr based on new noise estimate
-    nShifts = inst->prevQNoise + 11 - qMagn;
-    for (i = 0; i < inst->magnLen; i++)
-    {
-        // FLOAT code
-        // // post and prior SNR
-        // curNearSnr = 0.0;
-        // if (magn[i] > noise[i])
-        // {
-        // curNearSnr = magn[i] / (noise[i] + 0.0001) - 1.0;
-        // }
-        // // DD estimate is sum of two terms: current estimate and previous estimate
-        // // directed decision update of snrPrior
-        // snrPrior = DD_PR_SNR * prevNearSnr[i] + (1.0 - DD_PR_SNR) * curNearSnr;
-        // // gain filter
-        // tmpFloat1 = inst->overdrive + snrPrior;
-        // tmpFloat2 = snrPrior / tmpFloat1;
-        // theFilter[i] = tmpFloat2;
-
-        // calculate curNearSnr again, this is necessary because a new noise estimate has been made since then. for the original
-        curNearSnr = 0; // Q11
-        if (nShifts < 0)
-        {
-            // This case is equivalent with magn < noise which implies curNearSnr = 0;
-            tmpMagnU32 = (WebRtc_UWord32)magnU16[i]; // Q(qMagn)
-            tmpNoiseU32 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], -nShifts); // Q(qMagn)
-        } else if (nShifts > 17)
-        {
-            tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], 17); // Q(qMagn+17)
-            tmpNoiseU32 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], nShifts - 17); // Q(qMagn+17)
-        } else
-        {
-            tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)magnU16[i], nShifts); // Q(qNoise_prev+11)
-            tmpNoiseU32 = noiseU32[i]; // Q(qNoise_prev+11)
-        }
-        if (tmpMagnU32 > tmpNoiseU32)
-        {
-            tmpU32no1 = tmpMagnU32 - tmpNoiseU32; // Q(qCur)
-            norm32no2 = WEBRTC_SPL_MIN(11, WebRtcSpl_NormU32(tmpU32no1));
-            tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32no2); // Q(qCur+norm32no2)
-            tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpNoiseU32, 11 - norm32no2); // Q(qCur+norm32no2-11)
-            if (tmpU32no2)
-            {
-                tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
-            }
-            curNearSnr = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
-        }
-
-        //directed decision update of priorSnr
-        // FLOAT
-        // priorSnr = DD_PR_SNR * prevNearSnr + (1.0-DD_PR_SNR) * curNearSnr;
-
-        tmpU32no1 = WEBRTC_SPL_UMUL_32_16(prevNearSnr[i], DD_PR_SNR_Q11); // Q22
-        tmpU32no2 = WEBRTC_SPL_UMUL_32_16(curNearSnr, ONE_MINUS_DD_PR_SNR_Q11); // Q22
-        priorSnr = tmpU32no1 + tmpU32no2; // Q22
-
-        //gain filter
-        tmpU32no1 = (WebRtc_UWord32)(inst->overdrive)
-                + WEBRTC_SPL_RSHIFT_U32(priorSnr + 8192, 14); // Q8
-        tmpU16no1 = (WebRtc_UWord16)WEBRTC_SPL_UDIV(priorSnr + (tmpU32no1 >> 1), tmpU32no1); // Q14
-        inst->noiseSupFilter[i] = WEBRTC_SPL_SAT(16384, tmpU16no1, inst->denoiseBound); // 16384 = Q14(1.0) // Q14
-
-        // Weight in the parametric Wiener filter during startup
-        if (inst->blockIndex < END_STARTUP_SHORT)
-        {
-            // Weight the two suppression filters
-            tmpU32no1 = WEBRTC_SPL_UMUL_16_16(inst->noiseSupFilter[i],
-                                              (WebRtc_UWord16)inst->blockIndex);
-            tmpU32no2 = WEBRTC_SPL_UMUL_16_16(noiseSupFilterTmp[i],
-                                              (WebRtc_UWord16)(END_STARTUP_SHORT
-                                                      - inst->blockIndex));
-            tmpU32no1 += tmpU32no2;
-            inst->noiseSupFilter[i] = (WebRtc_UWord16)WebRtcSpl_DivU32U16(tmpU32no1,
-                                                                          END_STARTUP_SHORT);
-        }
-    } // end of loop over frequencies
-    //done with step3
-
-    // save noise and magnitude spectrum for next frame
-    inst->prevQNoise = qNoise;
-    inst->prevQMagn = qMagn;
-    if (norm32no1 > 5)
-    {
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            inst->prevNoiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], norm32no1 - 5); // Q(qNoise+11)
-            inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
-        }
-    } else
-    {
-        for (i = 0; i < inst->magnLen; i++)
-        {
-            inst->prevNoiseU32[i] = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], 5 - norm32no1); // Q(qNoise+11)
-            inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
-        }
-    }
-
-    WebRtcNsx_DataSynthesis(inst, outFrame);
-#ifdef NS_FILEDEBUG
-    fwrite(outframe, sizeof(short), inst->blockLen10ms, inst->outfile);
-#endif
-
-    //for H band:
-    // only update data buffer, then apply time-domain gain is applied derived from L band
-    if (inst->fs == 32000)
-    {
-        // update analysis buffer for H band
-        // append new data to buffer FX
-        WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX, inst->dataBufHBFX + inst->blockLen10ms, inst->anaLen - inst->blockLen10ms);
-        WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX + inst->anaLen - inst->blockLen10ms, speechFrameHB, inst->blockLen10ms);
-        // range for averaging low band quantities for H band gain
-
-        gainTimeDomainHB = 16384; // 16384 = Q14(1.0)
-        //average speech prob from low band
-        //average filter gain from low band
-        //avg over second half (i.e., 4->8kHz) of freq. spectrum
-        tmpU32no1 = 0; // Q12
-        tmpU16no1 = 0; // Q8
-        for (i = inst->anaLen2 - (inst->anaLen2 >> 2); i < inst->anaLen2; i++)
-        {
-            tmpU16no1 += nonSpeechProbFinal[i]; // Q8
-            tmpU32no1 += (WebRtc_UWord32)(inst->noiseSupFilter[i]); // Q14
-        }
-        avgProbSpeechHB = (WebRtc_Word16)(4096
-                - WEBRTC_SPL_RSHIFT_U16(tmpU16no1, inst->stages - 7)); // Q12
-        avgFilterGainHB = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, inst->stages - 3); // Q14
-
-        // // original FLOAT code
-        // // gain based on speech probability:
-        // avg_prob_speech_tt=(float)2.0*avg_prob_speech-(float)1.0;
-        // gain_mod=(float)0.5*((float)1.0+(float)tanh(avg_prob_speech_tt)); // between 0 and 1
-
-        // gain based on speech probability:
-        // original expression: "0.5 * (1 + tanh(2x-1))"
-        // avgProbSpeechHB has been anyway saturated to a value between 0 and 1 so the other cases don't have to be dealt with
-        // avgProbSpeechHB and gainModHB are in Q12, 3607 = Q12(0.880615234375) which is a zero point of
-        // |0.5 * (1 + tanh(2x-1)) - x| - |0.5 * (1 + tanh(2x-1)) - 0.880615234375| meaning that from that point the error of approximating
-        // the expression with f(x) = x would be greater than the error of approximating the expression with f(x) = 0.880615234375
-        // error: "|0.5 * (1 + tanh(2x-1)) - x| from x=0 to 0.880615234375" -> http://www.wolframalpha.com/input/?i=|0.5+*+(1+%2B+tanh(2x-1))+-+x|+from+x%3D0+to+0.880615234375
-        // and:  "|0.5 * (1 + tanh(2x-1)) - 0.880615234375| from x=0.880615234375 to 1" -> http://www.wolframalpha.com/input/?i=+|0.5+*+(1+%2B+tanh(2x-1))+-+0.880615234375|+from+x%3D0.880615234375+to+1
-        gainModHB = WEBRTC_SPL_MIN(avgProbSpeechHB, 3607);
-
-        // // original FLOAT code
-        // //combine gain with low band gain
-        // if (avg_prob_speech < (float)0.5) {
-        // gain_time_domain_HB=(float)0.5*gain_mod+(float)0.5*avg_filter_gain;
-        // }
-        // else {
-        // gain_time_domain_HB=(float)0.25*gain_mod+(float)0.75*avg_filter_gain;
-        // }
-
-
-        //combine gain with low band gain
-        if (avgProbSpeechHB < 2048)
-        { // 2048 = Q12(0.5)
-            // the next two lines in float are  "gain_time_domain = 0.5 * gain_mod + 0.5 * avg_filter_gain"; Q2(0.5) = 2 equals one left shift
-            gainTimeDomainHB = (gainModHB << 1) + (avgFilterGainHB >> 1); // Q14
-        } else
-        {
-            // "gain_time_domain = 0.25 * gain_mod + 0.75 * agv_filter_gain;"
-            gainTimeDomainHB = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(3, avgFilterGainHB, 2); // 3 = Q2(0.75); Q14
-            gainTimeDomainHB += gainModHB; // Q14
-        }
-        //make sure gain is within flooring range
-        gainTimeDomainHB
-                = WEBRTC_SPL_SAT(16384, gainTimeDomainHB, (WebRtc_Word16)(inst->denoiseBound)); // 16384 = Q14(1.0)
-
-
-        //apply gain
-        for (i = 0; i < inst->blockLen10ms; i++)
-        {
-            outFrameHB[i]
-                    = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gainTimeDomainHB, inst->dataBufHBFX[i], 14); // Q0
-        }
-    } // end of H band gain computation
-
-    return 0;
-}
diff --git a/src/modules/audio_processing/ns/main/source/nsx_core.h b/src/modules/audio_processing/ns/main/source/nsx_core.h
deleted file mode 100644
index 2e74303..0000000
--- a/src/modules/audio_processing/ns/main/source/nsx_core.h
+++ /dev/null
@@ -1,169 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
-
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-#include "nsx_defines.h"
-
-#ifdef NS_FILEDEBUG
-#include <stdio.h>
-#endif
-
-typedef struct NsxInst_t_
-{
-    WebRtc_UWord32          fs;
-
-    const WebRtc_Word16*    window;
-    WebRtc_Word16           analysisBuffer[ANAL_BLOCKL_MAX];
-    WebRtc_Word16           synthesisBuffer[ANAL_BLOCKL_MAX];
-    WebRtc_UWord16          noiseSupFilter[HALF_ANAL_BLOCKL];
-    WebRtc_UWord16          overdrive; /* Q8 */
-    WebRtc_UWord16          denoiseBound; /* Q14 */
-    const WebRtc_Word16*    factor2Table;
-    WebRtc_Word16           noiseEstLogQuantile[SIMULT * HALF_ANAL_BLOCKL];
-    WebRtc_Word16           noiseEstDensity[SIMULT * HALF_ANAL_BLOCKL];
-    WebRtc_Word16           noiseEstCounter[SIMULT];
-    WebRtc_Word16           noiseEstQuantile[HALF_ANAL_BLOCKL];
-
-    WebRtc_Word16           anaLen;
-    int                     anaLen2;
-    int                     magnLen;
-    int                     aggrMode;
-    int                     stages;
-    int                     initFlag;
-    int                     gainMap;
-
-    WebRtc_Word32           maxLrt;
-    WebRtc_Word32           minLrt;
-    WebRtc_Word32           logLrtTimeAvgW32[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing in Q8
-    WebRtc_Word32           featureLogLrt;
-    WebRtc_Word32           thresholdLogLrt;
-    WebRtc_Word16           weightLogLrt;
-
-    WebRtc_UWord32          featureSpecDiff;
-    WebRtc_UWord32          thresholdSpecDiff;
-    WebRtc_Word16           weightSpecDiff;
-
-    WebRtc_UWord32          featureSpecFlat;
-    WebRtc_UWord32          thresholdSpecFlat;
-    WebRtc_Word16           weightSpecFlat;
-
-    WebRtc_Word32           avgMagnPause[HALF_ANAL_BLOCKL]; //conservative estimate of noise spectrum
-    WebRtc_UWord32          magnEnergy;
-    WebRtc_UWord32          sumMagn;
-    WebRtc_UWord32          curAvgMagnEnergy;
-    WebRtc_UWord32          timeAvgMagnEnergy;
-    WebRtc_UWord32          timeAvgMagnEnergyTmp;
-
-    WebRtc_UWord32          whiteNoiseLevel;              //initial noise estimate
-    WebRtc_UWord32          initMagnEst[HALF_ANAL_BLOCKL];//initial magnitude spectrum estimate
-    WebRtc_Word32           pinkNoiseNumerator;           //pink noise parameter: numerator
-    WebRtc_Word32           pinkNoiseExp;                 //pink noise parameter: power of freq
-    int                     minNorm;                      //smallest normalization factor
-    int                     zeroInputSignal;              //zero input signal flag
-
-    WebRtc_UWord32          prevNoiseU32[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame
-    WebRtc_UWord16          prevMagnU16[HALF_ANAL_BLOCKL]; //magnitude spectrum from previous frame
-    WebRtc_Word16           priorNonSpeechProb; //prior speech/noise probability // Q14
-
-    int                     blockIndex; //frame index counter
-    int                     modelUpdate; //parameter for updating or estimating thresholds/weights for prior model
-    int                     cntThresUpdate;
-
-    //histograms for parameter estimation
-    WebRtc_Word16           histLrt[HIST_PAR_EST];
-    WebRtc_Word16           histSpecFlat[HIST_PAR_EST];
-    WebRtc_Word16           histSpecDiff[HIST_PAR_EST];
-
-    //quantities for high band estimate
-    WebRtc_Word16           dataBufHBFX[ANAL_BLOCKL_MAX]; /* Q0 */
-
-    int                     qNoise;
-    int                     prevQNoise;
-    int                     prevQMagn;
-    int                     blockLen10ms;
-
-    WebRtc_Word16           real[ANAL_BLOCKL_MAX];
-    WebRtc_Word16           imag[ANAL_BLOCKL_MAX];
-    WebRtc_Word32           energyIn;
-    int                     scaleEnergyIn;
-    int                     normData;
-
-} NsxInst_t;
-
-#ifdef __cplusplus
-extern "C"
-{
-#endif
-
-/****************************************************************************
- * WebRtcNsx_InitCore(...)
- *
- * This function initializes a noise suppression instance
- *
- * Input:
- *      - inst          : Instance that should be initialized
- *      - fs            : Sampling frequency
- *
- * Output:
- *      - inst          : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-WebRtc_Word32 WebRtcNsx_InitCore(NsxInst_t *inst, WebRtc_UWord32 fs);
-
-/****************************************************************************
- * WebRtcNsx_set_policy_core(...)
- *
- * This changes the aggressiveness of the noise suppression method.
- *
- * Input:
- *      - inst          : Instance that should be initialized
- *      - mode          : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB)
- *
- * Output:
- *      - NS_inst      : Initialized instance
- *
- * Return value         :  0 - Ok
- *                        -1 - Error
- */
-int WebRtcNsx_set_policy_core(NsxInst_t *inst, int mode);
-
-/****************************************************************************
- * WebRtcNsx_ProcessCore
- *
- * Do noise suppression.
- *
- * Input:
- *      - inst          : Instance that should be initialized
- *      - inFrameLow    : Input speech frame for lower band
- *      - inFrameHigh   : Input speech frame for higher band
- *
- * Output:
- *      - inst          : Updated instance
- *      - outFrameLow   : Output speech frame for lower band
- *      - outFrameHigh  : Output speech frame for higher band
- *
- * Return value         :  0 - OK
- *                        -1 - Error
- */
-int WebRtcNsx_ProcessCore(NsxInst_t *inst, short *inFrameLow, short *inFrameHigh,
-                          short *outFrameLow, short *outFrameHigh);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
diff --git a/src/modules/audio_processing/ns/main/source/nsx_defines.h b/src/modules/audio_processing/ns/main/source/nsx_defines.h
deleted file mode 100644
index 58796b9..0000000
--- a/src/modules/audio_processing/ns/main/source/nsx_defines.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
-
-#define ANAL_BLOCKL_MAX         256 // max analysis block length
-#define HALF_ANAL_BLOCKL        129 // half max analysis block length + 1
-#define SIMULT                  3
-#define END_STARTUP_LONG        200
-#define END_STARTUP_SHORT       50
-#define FACTOR_Q16              (WebRtc_Word32)2621440 // 40 in Q16
-#define FACTOR_Q7               (WebRtc_Word16)5120 // 40 in Q7
-#define WIDTH_Q8                3 // 0.01 in Q8 (or 25 )
-//PARAMETERS FOR NEW METHOD
-#define DD_PR_SNR_Q11           2007 // ~= Q11(0.98) DD update of prior SNR
-#define ONE_MINUS_DD_PR_SNR_Q11 41 // DD update of prior SNR
-#define SPECT_FLAT_TAVG_Q14     4915 // (0.30) tavg parameter for spectral flatness measure
-#define SPECT_DIFF_TAVG_Q8      77 // (0.30) tavg parameter for spectral flatness measure
-#define PRIOR_UPDATE_Q14        1638 // Q14(0.1) update parameter of prior model
-#define NOISE_UPDATE_Q8         26 // 26 ~= Q8(0.1) update parameter for noise
-// probability threshold for noise state in speech/noise likelihood
-#define ONE_MINUS_PROB_RANGE_Q8 205 // 205 ~= Q8(0.8)
-#define HIST_PAR_EST            1000 // histogram size for estimation of parameters
-//FEATURE EXTRACTION CONFIG
-//bin size of histogram
-#define BIN_SIZE_LRT            10
-//scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
-// thresholds for prior model
-#define FACTOR_1_LRT_DIFF       6 //for LRT and spectral difference (5 times bigger)
-//for spectral_flatness: used when noise is flatter than speech (10 times bigger)
-#define FACTOR_2_FLAT_Q10       922
-//peak limit for spectral flatness (varies between 0 and 1)
-#define THRES_PEAK_FLAT         24 // * 2 * BIN_SIZE_FLAT_FX
-//limit on spacing of two highest peaks in histogram: spacing determined by bin size
-#define LIM_PEAK_SPACE_FLAT_DIFF    4 // * 2 * BIN_SIZE_DIFF_FX
-//limit on relevance of second peak:
-#define LIM_PEAK_WEIGHT_FLAT_DIFF   2
-#define THRES_FLUCT_LRT         10240 //=20 * inst->modelUpdate; fluctuation limit of LRT feat.
-//limit on the max and min values for the feature thresholds
-#define MAX_FLAT_Q10            38912 //  * 2 * BIN_SIZE_FLAT_FX
-#define MIN_FLAT_Q10            4096 //  * 2 * BIN_SIZE_FLAT_FX
-#define MAX_DIFF                100 // * 2 * BIN_SIZE_DIFF_FX
-#define MIN_DIFF                16 // * 2 * BIN_SIZE_DIFF_FX
-//criteria of weight of histogram peak  to accept/reject feature
-#define THRES_WEIGHT_FLAT_DIFF  154//(int)(0.3*(inst->modelUpdate)) for flatness and difference
-//
-#define STAT_UPDATES            9 // Update every 512 = 1 << 9 block
-#define ONE_MINUS_GAMMA_PAUSE_Q8    13 // ~= Q8(0.05) update for conservative noise estimate
-#define GAMMA_NOISE_TRANS_AND_SPEECH_Q8 3 // ~= Q8(0.01) update for transition and noise region
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
diff --git a/src/modules/audio_processing/ns/main/source/windows_private.h b/src/modules/audio_processing/ns/main/source/windows_private.h
deleted file mode 100644
index 8f9006e..0000000
--- a/src/modules/audio_processing/ns/main/source/windows_private.h
+++ /dev/null
@@ -1,573 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
-
-// Hanning window for 4ms 16kHz
-static const float kHanning64w128[128] = {
-0.00000000000000f, 0.02454122852291f, 0.04906767432742f,
-0.07356456359967f, 0.09801714032956f, 0.12241067519922f,
-0.14673047445536f, 0.17096188876030f, 0.19509032201613f,
-0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
-0.29028467725446f, 0.31368174039889f, 0.33688985339222f,
-0.35989503653499f, 0.38268343236509f, 0.40524131400499f,
-0.42755509343028f, 0.44961132965461f, 0.47139673682600f,
-0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
-0.55557023301960f, 0.57580819141785f, 0.59569930449243f,
-0.61523159058063f, 0.63439328416365f, 0.65317284295378f,
-0.67155895484702f, 0.68954054473707f, 0.70710678118655f,
-0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
-0.77301045336274f, 0.78834642762661f, 0.80320753148064f,
-0.81758481315158f, 0.83146961230255f, 0.84485356524971f,
-0.85772861000027f, 0.87008699110871f, 0.88192126434835f,
-0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
-0.92387953251129f, 0.93299279883474f, 0.94154406518302f,
-0.94952818059304f, 0.95694033573221f, 0.96377606579544f,
-0.97003125319454f, 0.97570213003853f, 0.98078528040323f,
-0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
-0.99518472667220f, 0.99729045667869f, 0.99879545620517f,
-0.99969881869620f, 1.00000000000000f,
-0.99969881869620f, 0.99879545620517f, 0.99729045667869f,
-0.99518472667220f, 0.99247953459871f, 0.98917650996478f,
-0.98527764238894f, 0.98078528040323f, 0.97570213003853f,
-0.97003125319454f, 0.96377606579544f, 0.95694033573221f,
-0.94952818059304f, 0.94154406518302f, 0.93299279883474f,
-0.92387953251129f, 0.91420975570353f, 0.90398929312344f,
-0.89322430119552f, 0.88192126434835f, 0.87008699110871f,
-0.85772861000027f, 0.84485356524971f, 0.83146961230255f,
-0.81758481315158f, 0.80320753148064f, 0.78834642762661f,
-0.77301045336274f, 0.75720884650648f, 0.74095112535496f,
-0.72424708295147f, 0.70710678118655f, 0.68954054473707f,
-0.67155895484702f, 0.65317284295378f, 0.63439328416365f,
-0.61523159058063f, 0.59569930449243f, 0.57580819141785f,
-0.55557023301960f, 0.53499761988710f, 0.51410274419322f,
-0.49289819222978f, 0.47139673682600f, 0.44961132965461f,
-0.42755509343028f, 0.40524131400499f, 0.38268343236509f,
-0.35989503653499f, 0.33688985339222f, 0.31368174039889f,
-0.29028467725446f, 0.26671275747490f, 0.24298017990326f,
-0.21910124015687f, 0.19509032201613f, 0.17096188876030f,
-0.14673047445536f, 0.12241067519922f, 0.09801714032956f,
-0.07356456359967f, 0.04906767432742f, 0.02454122852291f
-};
-
-
-
-// hybrib Hanning & flat window
-static const float kBlocks80w128[128] = {
-(float)0.00000000, (float)0.03271908, (float)0.06540313, (float)0.09801714, (float)0.13052619,
-(float)0.16289547, (float)0.19509032, (float)0.22707626, (float)0.25881905, (float)0.29028468,
-(float)0.32143947, (float)0.35225005, (float)0.38268343, (float)0.41270703, (float)0.44228869,
-(float)0.47139674, (float)0.50000000, (float)0.52806785, (float)0.55557023, (float)0.58247770,
-(float)0.60876143, (float)0.63439328, (float)0.65934582, (float)0.68359230, (float)0.70710678,
-(float)0.72986407, (float)0.75183981, (float)0.77301045, (float)0.79335334, (float)0.81284668,
-(float)0.83146961, (float)0.84920218, (float)0.86602540, (float)0.88192126, (float)0.89687274,
-(float)0.91086382, (float)0.92387953, (float)0.93590593, (float)0.94693013, (float)0.95694034,
-(float)0.96592583, (float)0.97387698, (float)0.98078528, (float)0.98664333, (float)0.99144486,
-(float)0.99518473, (float)0.99785892, (float)0.99946459, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)0.99946459, (float)0.99785892, (float)0.99518473, (float)0.99144486,
-(float)0.98664333, (float)0.98078528, (float)0.97387698, (float)0.96592583, (float)0.95694034,
-(float)0.94693013, (float)0.93590593, (float)0.92387953, (float)0.91086382, (float)0.89687274,
-(float)0.88192126, (float)0.86602540, (float)0.84920218, (float)0.83146961, (float)0.81284668,
-(float)0.79335334, (float)0.77301045, (float)0.75183981, (float)0.72986407, (float)0.70710678,
-(float)0.68359230, (float)0.65934582, (float)0.63439328, (float)0.60876143, (float)0.58247770,
-(float)0.55557023, (float)0.52806785, (float)0.50000000, (float)0.47139674, (float)0.44228869,
-(float)0.41270703, (float)0.38268343, (float)0.35225005, (float)0.32143947, (float)0.29028468,
-(float)0.25881905, (float)0.22707626, (float)0.19509032, (float)0.16289547, (float)0.13052619,
-(float)0.09801714, (float)0.06540313, (float)0.03271908
-};
-
-// hybrib Hanning & flat window
-static const float kBlocks160w256[256] = {
-(float)0.00000000, (float)0.01636173, (float)0.03271908, (float)0.04906767, (float)0.06540313,
-(float)0.08172107, (float)0.09801714, (float)0.11428696, (float)0.13052619, (float)0.14673047,
-(float)0.16289547, (float)0.17901686, (float)0.19509032, (float)0.21111155, (float)0.22707626,
-(float)0.24298018, (float)0.25881905, (float)0.27458862, (float)0.29028468, (float)0.30590302,
-(float)0.32143947, (float)0.33688985, (float)0.35225005, (float)0.36751594, (float)0.38268343,
-(float)0.39774847, (float)0.41270703, (float)0.42755509, (float)0.44228869, (float)0.45690388,
-(float)0.47139674, (float)0.48576339, (float)0.50000000, (float)0.51410274, (float)0.52806785,
-(float)0.54189158, (float)0.55557023, (float)0.56910015, (float)0.58247770, (float)0.59569930,
-(float)0.60876143, (float)0.62166057, (float)0.63439328, (float)0.64695615, (float)0.65934582,
-(float)0.67155895, (float)0.68359230, (float)0.69544264, (float)0.70710678, (float)0.71858162,
-(float)0.72986407, (float)0.74095113, (float)0.75183981, (float)0.76252720, (float)0.77301045,
-(float)0.78328675, (float)0.79335334, (float)0.80320753, (float)0.81284668, (float)0.82226822,
-(float)0.83146961, (float)0.84044840, (float)0.84920218, (float)0.85772861, (float)0.86602540,
-(float)0.87409034, (float)0.88192126, (float)0.88951608, (float)0.89687274, (float)0.90398929,
-(float)0.91086382, (float)0.91749450, (float)0.92387953, (float)0.93001722, (float)0.93590593,
-(float)0.94154407, (float)0.94693013, (float)0.95206268, (float)0.95694034, (float)0.96156180,
-(float)0.96592583, (float)0.97003125, (float)0.97387698, (float)0.97746197, (float)0.98078528,
-(float)0.98384601, (float)0.98664333, (float)0.98917651, (float)0.99144486, (float)0.99344778,
-(float)0.99518473, (float)0.99665524, (float)0.99785892, (float)0.99879546, (float)0.99946459,
-(float)0.99986614, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)0.99986614, (float)0.99946459, (float)0.99879546, (float)0.99785892,
-(float)0.99665524, (float)0.99518473, (float)0.99344778, (float)0.99144486, (float)0.98917651,
-(float)0.98664333, (float)0.98384601, (float)0.98078528, (float)0.97746197, (float)0.97387698,
-(float)0.97003125, (float)0.96592583, (float)0.96156180, (float)0.95694034, (float)0.95206268,
-(float)0.94693013, (float)0.94154407, (float)0.93590593, (float)0.93001722, (float)0.92387953,
-(float)0.91749450, (float)0.91086382, (float)0.90398929, (float)0.89687274, (float)0.88951608,
-(float)0.88192126, (float)0.87409034, (float)0.86602540, (float)0.85772861, (float)0.84920218,
-(float)0.84044840, (float)0.83146961, (float)0.82226822, (float)0.81284668, (float)0.80320753,
-(float)0.79335334, (float)0.78328675, (float)0.77301045, (float)0.76252720, (float)0.75183981,
-(float)0.74095113, (float)0.72986407, (float)0.71858162, (float)0.70710678, (float)0.69544264,
-(float)0.68359230, (float)0.67155895, (float)0.65934582, (float)0.64695615, (float)0.63439328,
-(float)0.62166057, (float)0.60876143, (float)0.59569930, (float)0.58247770, (float)0.56910015,
-(float)0.55557023, (float)0.54189158, (float)0.52806785, (float)0.51410274, (float)0.50000000,
-(float)0.48576339, (float)0.47139674, (float)0.45690388, (float)0.44228869, (float)0.42755509,
-(float)0.41270703, (float)0.39774847, (float)0.38268343, (float)0.36751594, (float)0.35225005,
-(float)0.33688985, (float)0.32143947, (float)0.30590302, (float)0.29028468, (float)0.27458862,
-(float)0.25881905, (float)0.24298018, (float)0.22707626, (float)0.21111155, (float)0.19509032,
-(float)0.17901686, (float)0.16289547, (float)0.14673047, (float)0.13052619, (float)0.11428696,
-(float)0.09801714, (float)0.08172107, (float)0.06540313, (float)0.04906767, (float)0.03271908,
-(float)0.01636173
-};
-
-// hybrib Hanning & flat window: for 20ms
-static const float kBlocks320w512[512] = {
-(float)0.00000000, (float)0.00818114, (float)0.01636173, (float)0.02454123, (float)0.03271908,
-(float)0.04089475, (float)0.04906767, (float)0.05723732, (float)0.06540313, (float)0.07356456,
-(float)0.08172107, (float)0.08987211, (float)0.09801714, (float)0.10615561, (float)0.11428696,
-(float)0.12241068, (float)0.13052619, (float)0.13863297, (float)0.14673047, (float)0.15481816,
-(float)0.16289547, (float)0.17096189, (float)0.17901686, (float)0.18705985, (float)0.19509032,
-(float)0.20310773, (float)0.21111155, (float)0.21910124, (float)0.22707626, (float)0.23503609,
-(float)0.24298018, (float)0.25090801, (float)0.25881905, (float)0.26671276, (float)0.27458862,
-(float)0.28244610, (float)0.29028468, (float)0.29810383, (float)0.30590302, (float)0.31368174,
-(float)0.32143947, (float)0.32917568, (float)0.33688985, (float)0.34458148, (float)0.35225005,
-(float)0.35989504, (float)0.36751594, (float)0.37511224, (float)0.38268343, (float)0.39022901,
-(float)0.39774847, (float)0.40524131, (float)0.41270703, (float)0.42014512, (float)0.42755509,
-(float)0.43493645, (float)0.44228869, (float)0.44961133, (float)0.45690388, (float)0.46416584,
-(float)0.47139674, (float)0.47859608, (float)0.48576339, (float)0.49289819, (float)0.50000000,
-(float)0.50706834, (float)0.51410274, (float)0.52110274, (float)0.52806785, (float)0.53499762,
-(float)0.54189158, (float)0.54874927, (float)0.55557023, (float)0.56235401, (float)0.56910015,
-(float)0.57580819, (float)0.58247770, (float)0.58910822, (float)0.59569930, (float)0.60225052,
-(float)0.60876143, (float)0.61523159, (float)0.62166057, (float)0.62804795, (float)0.63439328,
-(float)0.64069616, (float)0.64695615, (float)0.65317284, (float)0.65934582, (float)0.66547466,
-(float)0.67155895, (float)0.67759830, (float)0.68359230, (float)0.68954054, (float)0.69544264,
-(float)0.70129818, (float)0.70710678, (float)0.71286806, (float)0.71858162, (float)0.72424708,
-(float)0.72986407, (float)0.73543221, (float)0.74095113, (float)0.74642045, (float)0.75183981,
-(float)0.75720885, (float)0.76252720, (float)0.76779452, (float)0.77301045, (float)0.77817464,
-(float)0.78328675, (float)0.78834643, (float)0.79335334, (float)0.79830715, (float)0.80320753,
-(float)0.80805415, (float)0.81284668, (float)0.81758481, (float)0.82226822, (float)0.82689659,
-(float)0.83146961, (float)0.83598698, (float)0.84044840, (float)0.84485357, (float)0.84920218,
-(float)0.85349396, (float)0.85772861, (float)0.86190585, (float)0.86602540, (float)0.87008699,
-(float)0.87409034, (float)0.87803519, (float)0.88192126, (float)0.88574831, (float)0.88951608,
-(float)0.89322430, (float)0.89687274, (float)0.90046115, (float)0.90398929, (float)0.90745693,
-(float)0.91086382, (float)0.91420976, (float)0.91749450, (float)0.92071783, (float)0.92387953,
-(float)0.92697940, (float)0.93001722, (float)0.93299280, (float)0.93590593, (float)0.93875641,
-(float)0.94154407, (float)0.94426870, (float)0.94693013, (float)0.94952818, (float)0.95206268,
-(float)0.95453345, (float)0.95694034, (float)0.95928317, (float)0.96156180, (float)0.96377607,
-(float)0.96592583, (float)0.96801094, (float)0.97003125, (float)0.97198664, (float)0.97387698,
-(float)0.97570213, (float)0.97746197, (float)0.97915640, (float)0.98078528, (float)0.98234852,
-(float)0.98384601, (float)0.98527764, (float)0.98664333, (float)0.98794298, (float)0.98917651,
-(float)0.99034383, (float)0.99144486, (float)0.99247953, (float)0.99344778, (float)0.99434953,
-(float)0.99518473, (float)0.99595331, (float)0.99665524, (float)0.99729046, (float)0.99785892,
-(float)0.99836060, (float)0.99879546, (float)0.99916346, (float)0.99946459, (float)0.99969882,
-(float)0.99986614, (float)0.99996653, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
-(float)1.00000000, (float)0.99996653, (float)0.99986614, (float)0.99969882, (float)0.99946459,
-(float)0.99916346, (float)0.99879546, (float)0.99836060, (float)0.99785892, (float)0.99729046,
-(float)0.99665524, (float)0.99595331, (float)0.99518473, (float)0.99434953, (float)0.99344778,
-(float)0.99247953, (float)0.99144486, (float)0.99034383, (float)0.98917651, (float)0.98794298,
-(float)0.98664333, (float)0.98527764, (float)0.98384601, (float)0.98234852, (float)0.98078528,
-(float)0.97915640, (float)0.97746197, (float)0.97570213, (float)0.97387698, (float)0.97198664,
-(float)0.97003125, (float)0.96801094, (float)0.96592583, (float)0.96377607, (float)0.96156180,
-(float)0.95928317, (float)0.95694034, (float)0.95453345, (float)0.95206268, (float)0.94952818,
-(float)0.94693013, (float)0.94426870, (float)0.94154407, (float)0.93875641, (float)0.93590593,
-(float)0.93299280, (float)0.93001722, (float)0.92697940, (float)0.92387953, (float)0.92071783,
-(float)0.91749450, (float)0.91420976, (float)0.91086382, (float)0.90745693, (float)0.90398929,
-(float)0.90046115, (float)0.89687274, (float)0.89322430, (float)0.88951608, (float)0.88574831,
-(float)0.88192126, (float)0.87803519, (float)0.87409034, (float)0.87008699, (float)0.86602540,
-(float)0.86190585, (float)0.85772861, (float)0.85349396, (float)0.84920218, (float)0.84485357,
-(float)0.84044840, (float)0.83598698, (float)0.83146961, (float)0.82689659, (float)0.82226822,
-(float)0.81758481, (float)0.81284668, (float)0.80805415, (float)0.80320753, (float)0.79830715,
-(float)0.79335334, (float)0.78834643, (float)0.78328675, (float)0.77817464, (float)0.77301045,
-(float)0.76779452, (float)0.76252720, (float)0.75720885, (float)0.75183981, (float)0.74642045,
-(float)0.74095113, (float)0.73543221, (float)0.72986407, (float)0.72424708, (float)0.71858162,
-(float)0.71286806, (float)0.70710678, (float)0.70129818, (float)0.69544264, (float)0.68954054,
-(float)0.68359230, (float)0.67759830, (float)0.67155895, (float)0.66547466, (float)0.65934582,
-(float)0.65317284, (float)0.64695615, (float)0.64069616, (float)0.63439328, (float)0.62804795,
-(float)0.62166057, (float)0.61523159, (float)0.60876143, (float)0.60225052, (float)0.59569930,
-(float)0.58910822, (float)0.58247770, (float)0.57580819, (float)0.56910015, (float)0.56235401,
-(float)0.55557023, (float)0.54874927, (float)0.54189158, (float)0.53499762, (float)0.52806785,
-(float)0.52110274, (float)0.51410274, (float)0.50706834, (float)0.50000000, (float)0.49289819,
-(float)0.48576339, (float)0.47859608, (float)0.47139674, (float)0.46416584, (float)0.45690388,
-(float)0.44961133, (float)0.44228869, (float)0.43493645, (float)0.42755509, (float)0.42014512,
-(float)0.41270703, (float)0.40524131, (float)0.39774847, (float)0.39022901, (float)0.38268343,
-(float)0.37511224, (float)0.36751594, (float)0.35989504, (float)0.35225005, (float)0.34458148,
-(float)0.33688985, (float)0.32917568, (float)0.32143947, (float)0.31368174, (float)0.30590302,
-(float)0.29810383, (float)0.29028468, (float)0.28244610, (float)0.27458862, (float)0.26671276,
-(float)0.25881905, (float)0.25090801, (float)0.24298018, (float)0.23503609, (float)0.22707626,
-(float)0.21910124, (float)0.21111155, (float)0.20310773, (float)0.19509032, (float)0.18705985,
-(float)0.17901686, (float)0.17096189, (float)0.16289547, (float)0.15481816, (float)0.14673047,
-(float)0.13863297, (float)0.13052619, (float)0.12241068, (float)0.11428696, (float)0.10615561,
-(float)0.09801714, (float)0.08987211, (float)0.08172107, (float)0.07356456, (float)0.06540313,
-(float)0.05723732, (float)0.04906767, (float)0.04089475, (float)0.03271908, (float)0.02454123,
-(float)0.01636173, (float)0.00818114
-};
-
-
-// Hanning window: for 15ms at 16kHz with symmetric zeros
-static const float kBlocks240w512[512] = {
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00654494, (float)0.01308960, (float)0.01963369,
-(float)0.02617695, (float)0.03271908, (float)0.03925982, (float)0.04579887, (float)0.05233596,
-(float)0.05887080, (float)0.06540313, (float)0.07193266, (float)0.07845910, (float)0.08498218,
-(float)0.09150162, (float)0.09801714, (float)0.10452846, (float)0.11103531, (float)0.11753740,
-(float)0.12403446, (float)0.13052620, (float)0.13701233, (float)0.14349262, (float)0.14996676,
-(float)0.15643448, (float)0.16289547, (float)0.16934951, (float)0.17579629, (float)0.18223552,
-(float)0.18866697, (float)0.19509032, (float)0.20150533, (float)0.20791170, (float)0.21430916,
-(float)0.22069745, (float)0.22707628, (float)0.23344538, (float)0.23980446, (float)0.24615330,
-(float)0.25249159, (float)0.25881904, (float)0.26513544, (float)0.27144045, (float)0.27773386,
-(float)0.28401536, (float)0.29028466, (float)0.29654160, (float)0.30278578, (float)0.30901700,
-(float)0.31523499, (float)0.32143945, (float)0.32763019, (float)0.33380687, (float)0.33996925,
-(float)0.34611708, (float)0.35225007, (float)0.35836795, (float)0.36447051, (float)0.37055743,
-(float)0.37662852, (float)0.38268346, (float)0.38872197, (float)0.39474389, (float)0.40074885,
-(float)0.40673664, (float)0.41270703, (float)0.41865975, (float)0.42459452, (float)0.43051112,
-(float)0.43640924, (float)0.44228873, (float)0.44814920, (float)0.45399052, (float)0.45981237,
-(float)0.46561453, (float)0.47139674, (float)0.47715878, (float)0.48290035, (float)0.48862126,
-(float)0.49432120, (float)0.50000000, (float)0.50565743, (float)0.51129311, (float)0.51690692,
-(float)0.52249855, (float)0.52806789, (float)0.53361452, (float)0.53913832, (float)0.54463905,
-(float)0.55011642, (float)0.55557024, (float)0.56100029, (float)0.56640625, (float)0.57178795,
-(float)0.57714522, (float)0.58247769, (float)0.58778524, (float)0.59306765, (float)0.59832460,
-(float)0.60355598, (float)0.60876143, (float)0.61394083, (float)0.61909395, (float)0.62422055,
-(float)0.62932038, (float)0.63439333, (float)0.63943899, (float)0.64445734, (float)0.64944810,
-(float)0.65441096, (float)0.65934587, (float)0.66425246, (float)0.66913062, (float)0.67398012,
-(float)0.67880076, (float)0.68359232, (float)0.68835455, (float)0.69308740, (float)0.69779050,
-(float)0.70246369, (float)0.70710677, (float)0.71171963, (float)0.71630198, (float)0.72085363,
-(float)0.72537440, (float)0.72986406, (float)0.73432255, (float)0.73874950, (float)0.74314487,
-(float)0.74750835, (float)0.75183982, (float)0.75613910, (float)0.76040596, (float)0.76464027,
-(float)0.76884186, (float)0.77301043, (float)0.77714598, (float)0.78124821, (float)0.78531694,
-(float)0.78935206, (float)0.79335338, (float)0.79732066, (float)0.80125386, (float)0.80515265,
-(float)0.80901700, (float)0.81284672, (float)0.81664157, (float)0.82040149, (float)0.82412618,
-(float)0.82781565, (float)0.83146966, (float)0.83508795, (float)0.83867061, (float)0.84221727,
-(float)0.84572780, (float)0.84920216, (float)0.85264021, (float)0.85604161, (float)0.85940641,
-(float)0.86273444, (float)0.86602545, (float)0.86927933, (float)0.87249607, (float)0.87567532,
-(float)0.87881714, (float)0.88192129, (float)0.88498765, (float)0.88801610, (float)0.89100653,
-(float)0.89395881, (float)0.89687276, (float)0.89974827, (float)0.90258533, (float)0.90538365,
-(float)0.90814316, (float)0.91086388, (float)0.91354549, (float)0.91618794, (float)0.91879123,
-(float)0.92135513, (float)0.92387950, (float)0.92636442, (float)0.92880958, (float)0.93121493,
-(float)0.93358046, (float)0.93590593, (float)0.93819135, (float)0.94043654, (float)0.94264150,
-(float)0.94480604, (float)0.94693011, (float)0.94901365, (float)0.95105654, (float)0.95305866,
-(float)0.95501995, (float)0.95694035, (float)0.95881975, (float)0.96065807, (float)0.96245527,
-(float)0.96421117, (float)0.96592581, (float)0.96759909, (float)0.96923089, (float)0.97082120,
-(float)0.97236991, (float)0.97387701, (float)0.97534233, (float)0.97676587, (float)0.97814763,
-(float)0.97948742, (float)0.98078531, (float)0.98204112, (float)0.98325491, (float)0.98442656,
-(float)0.98555607, (float)0.98664331, (float)0.98768836, (float)0.98869103, (float)0.98965138,
-(float)0.99056935, (float)0.99144489, (float)0.99227792, (float)0.99306846, (float)0.99381649,
-(float)0.99452192, (float)0.99518472, (float)0.99580491, (float)0.99638247, (float)0.99691731,
-(float)0.99740952, (float)0.99785894, (float)0.99826562, (float)0.99862951, (float)0.99895066,
-(float)0.99922901, (float)0.99946457, (float)0.99965733, (float)0.99980724, (float)0.99991435,
-(float)0.99997860, (float)1.00000000, (float)0.99997860, (float)0.99991435, (float)0.99980724,
-(float)0.99965733, (float)0.99946457, (float)0.99922901, (float)0.99895066, (float)0.99862951,
-(float)0.99826562, (float)0.99785894, (float)0.99740946, (float)0.99691731, (float)0.99638247,
-(float)0.99580491, (float)0.99518472, (float)0.99452192, (float)0.99381644, (float)0.99306846,
-(float)0.99227792, (float)0.99144489, (float)0.99056935, (float)0.98965138, (float)0.98869103,
-(float)0.98768836, (float)0.98664331, (float)0.98555607, (float)0.98442656, (float)0.98325491,
-(float)0.98204112, (float)0.98078525, (float)0.97948742, (float)0.97814757, (float)0.97676587,
-(float)0.97534227, (float)0.97387695, (float)0.97236991, (float)0.97082120, (float)0.96923089,
-(float)0.96759909, (float)0.96592581, (float)0.96421117, (float)0.96245521, (float)0.96065807,
-(float)0.95881969, (float)0.95694029, (float)0.95501995, (float)0.95305860, (float)0.95105648,
-(float)0.94901365, (float)0.94693011, (float)0.94480604, (float)0.94264150, (float)0.94043654,
-(float)0.93819129, (float)0.93590593, (float)0.93358046, (float)0.93121493, (float)0.92880952,
-(float)0.92636436, (float)0.92387950, (float)0.92135507, (float)0.91879123, (float)0.91618794,
-(float)0.91354543, (float)0.91086382, (float)0.90814310, (float)0.90538365, (float)0.90258527,
-(float)0.89974827, (float)0.89687276, (float)0.89395875, (float)0.89100647, (float)0.88801610,
-(float)0.88498759, (float)0.88192123, (float)0.87881714, (float)0.87567532, (float)0.87249595,
-(float)0.86927933, (float)0.86602539, (float)0.86273432, (float)0.85940641, (float)0.85604161,
-(float)0.85264009, (float)0.84920216, (float)0.84572780, (float)0.84221715, (float)0.83867055,
-(float)0.83508795, (float)0.83146954, (float)0.82781565, (float)0.82412612, (float)0.82040137,
-(float)0.81664157, (float)0.81284660, (float)0.80901700, (float)0.80515265, (float)0.80125374,
-(float)0.79732066, (float)0.79335332, (float)0.78935200, (float)0.78531694, (float)0.78124815,
-(float)0.77714586, (float)0.77301049, (float)0.76884180, (float)0.76464021, (float)0.76040596,
-(float)0.75613904, (float)0.75183970, (float)0.74750835, (float)0.74314481, (float)0.73874938,
-(float)0.73432249, (float)0.72986400, (float)0.72537428, (float)0.72085363, (float)0.71630186,
-(float)0.71171951, (float)0.70710677, (float)0.70246363, (float)0.69779032, (float)0.69308734,
-(float)0.68835449, (float)0.68359220, (float)0.67880070, (float)0.67398006, (float)0.66913044,
-(float)0.66425240, (float)0.65934575, (float)0.65441096, (float)0.64944804, (float)0.64445722,
-(float)0.63943905, (float)0.63439327, (float)0.62932026, (float)0.62422055, (float)0.61909389,
-(float)0.61394072, (float)0.60876143, (float)0.60355592, (float)0.59832448, (float)0.59306765,
-(float)0.58778518, (float)0.58247757, (float)0.57714522, (float)0.57178789, (float)0.56640613,
-(float)0.56100023, (float)0.55557019, (float)0.55011630, (float)0.54463905, (float)0.53913826,
-(float)0.53361434, (float)0.52806783, (float)0.52249849, (float)0.51690674, (float)0.51129305,
-(float)0.50565726, (float)0.50000006, (float)0.49432117, (float)0.48862115, (float)0.48290038,
-(float)0.47715873, (float)0.47139663, (float)0.46561456, (float)0.45981231, (float)0.45399037,
-(float)0.44814920, (float)0.44228864, (float)0.43640912, (float)0.43051112, (float)0.42459446,
-(float)0.41865960, (float)0.41270703, (float)0.40673658, (float)0.40074870, (float)0.39474386,
-(float)0.38872188, (float)0.38268328, (float)0.37662849, (float)0.37055734, (float)0.36447033,
-(float)0.35836792, (float)0.35224995, (float)0.34611690, (float)0.33996922, (float)0.33380675,
-(float)0.32763001, (float)0.32143945, (float)0.31523487, (float)0.30901679, (float)0.30278572,
-(float)0.29654145, (float)0.29028472, (float)0.28401530, (float)0.27773371, (float)0.27144048,
-(float)0.26513538, (float)0.25881892, (float)0.25249159, (float)0.24615324, (float)0.23980433,
-(float)0.23344538, (float)0.22707619, (float)0.22069728, (float)0.21430916, (float)0.20791161,
-(float)0.20150517, (float)0.19509031, (float)0.18866688, (float)0.18223536, (float)0.17579627,
-(float)0.16934940, (float)0.16289529, (float)0.15643445, (float)0.14996666, (float)0.14349243,
-(float)0.13701232, (float)0.13052608, (float)0.12403426, (float)0.11753736, (float)0.11103519,
-(float)0.10452849, (float)0.09801710, (float)0.09150149, (float)0.08498220, (float)0.07845904,
-(float)0.07193252, (float)0.06540315, (float)0.05887074, (float)0.05233581, (float)0.04579888,
-(float)0.03925974, (float)0.03271893, (float)0.02617695, (float)0.01963361, (float)0.01308943,
-(float)0.00654493, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000
-};
-
-
-// Hanning window: for 30ms with 1024 fft with symmetric zeros at 16kHz
-static const float kBlocks480w1024[1024] = {
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00327249, (float)0.00654494,
-(float)0.00981732, (float)0.01308960, (float)0.01636173, (float)0.01963369, (float)0.02290544,
-(float)0.02617695, (float)0.02944817, (float)0.03271908, (float)0.03598964, (float)0.03925982,
-(float)0.04252957, (float)0.04579887, (float)0.04906768, (float)0.05233596, (float)0.05560368,
-(float)0.05887080, (float)0.06213730, (float)0.06540313, (float)0.06866825, (float)0.07193266,
-(float)0.07519628, (float)0.07845910, (float)0.08172107, (float)0.08498218, (float)0.08824237,
-(float)0.09150162, (float)0.09475989, (float)0.09801714, (float)0.10127335, (float)0.10452846,
-(float)0.10778246, (float)0.11103531, (float)0.11428697, (float)0.11753740, (float)0.12078657,
-(float)0.12403446, (float)0.12728101, (float)0.13052620, (float)0.13376999, (float)0.13701233,
-(float)0.14025325, (float)0.14349262, (float)0.14673047, (float)0.14996676, (float)0.15320145,
-(float)0.15643448, (float)0.15966582, (float)0.16289547, (float)0.16612339, (float)0.16934951,
-(float)0.17257382, (float)0.17579629, (float)0.17901687, (float)0.18223552, (float)0.18545224,
-(float)0.18866697, (float)0.19187967, (float)0.19509032, (float)0.19829889, (float)0.20150533,
-(float)0.20470962, (float)0.20791170, (float)0.21111156, (float)0.21430916, (float)0.21750447,
-(float)0.22069745, (float)0.22388805, (float)0.22707628, (float)0.23026206, (float)0.23344538,
-(float)0.23662618, (float)0.23980446, (float)0.24298020, (float)0.24615330, (float)0.24932377,
-(float)0.25249159, (float)0.25565669, (float)0.25881904, (float)0.26197866, (float)0.26513544,
-(float)0.26828939, (float)0.27144045, (float)0.27458861, (float)0.27773386, (float)0.28087610,
-(float)0.28401536, (float)0.28715158, (float)0.29028466, (float)0.29341471, (float)0.29654160,
-(float)0.29966527, (float)0.30278578, (float)0.30590302, (float)0.30901700, (float)0.31212768,
-(float)0.31523499, (float)0.31833893, (float)0.32143945, (float)0.32453656, (float)0.32763019,
-(float)0.33072028, (float)0.33380687, (float)0.33688986, (float)0.33996925, (float)0.34304500,
-(float)0.34611708, (float)0.34918544, (float)0.35225007, (float)0.35531089, (float)0.35836795,
-(float)0.36142117, (float)0.36447051, (float)0.36751595, (float)0.37055743, (float)0.37359497,
-(float)0.37662852, (float)0.37965801, (float)0.38268346, (float)0.38570479, (float)0.38872197,
-(float)0.39173502, (float)0.39474389, (float)0.39774847, (float)0.40074885, (float)0.40374491,
-(float)0.40673664, (float)0.40972406, (float)0.41270703, (float)0.41568562, (float)0.41865975,
-(float)0.42162940, (float)0.42459452, (float)0.42755508, (float)0.43051112, (float)0.43346250,
-(float)0.43640924, (float)0.43935132, (float)0.44228873, (float)0.44522133, (float)0.44814920,
-(float)0.45107228, (float)0.45399052, (float)0.45690390, (float)0.45981237, (float)0.46271592,
-(float)0.46561453, (float)0.46850815, (float)0.47139674, (float)0.47428030, (float)0.47715878,
-(float)0.48003215, (float)0.48290035, (float)0.48576337, (float)0.48862126, (float)0.49147385,
-(float)0.49432120, (float)0.49716330, (float)0.50000000, (float)0.50283140, (float)0.50565743,
-(float)0.50847799, (float)0.51129311, (float)0.51410276, (float)0.51690692, (float)0.51970553,
-(float)0.52249855, (float)0.52528602, (float)0.52806789, (float)0.53084403, (float)0.53361452,
-(float)0.53637928, (float)0.53913832, (float)0.54189163, (float)0.54463905, (float)0.54738063,
-(float)0.55011642, (float)0.55284631, (float)0.55557024, (float)0.55828828, (float)0.56100029,
-(float)0.56370628, (float)0.56640625, (float)0.56910014, (float)0.57178795, (float)0.57446963,
-(float)0.57714522, (float)0.57981455, (float)0.58247769, (float)0.58513463, (float)0.58778524,
-(float)0.59042960, (float)0.59306765, (float)0.59569931, (float)0.59832460, (float)0.60094351,
-(float)0.60355598, (float)0.60616195, (float)0.60876143, (float)0.61135441, (float)0.61394083,
-(float)0.61652070, (float)0.61909395, (float)0.62166059, (float)0.62422055, (float)0.62677383,
-(float)0.62932038, (float)0.63186020, (float)0.63439333, (float)0.63691956, (float)0.63943899,
-(float)0.64195162, (float)0.64445734, (float)0.64695615, (float)0.64944810, (float)0.65193301,
-(float)0.65441096, (float)0.65688187, (float)0.65934587, (float)0.66180271, (float)0.66425246,
-(float)0.66669512, (float)0.66913062, (float)0.67155898, (float)0.67398012, (float)0.67639405,
-(float)0.67880076, (float)0.68120021, (float)0.68359232, (float)0.68597710, (float)0.68835455,
-(float)0.69072467, (float)0.69308740, (float)0.69544262, (float)0.69779050, (float)0.70013082,
-(float)0.70246369, (float)0.70478904, (float)0.70710677, (float)0.70941699, (float)0.71171963,
-(float)0.71401459, (float)0.71630198, (float)0.71858168, (float)0.72085363, (float)0.72311789,
-(float)0.72537440, (float)0.72762316, (float)0.72986406, (float)0.73209721, (float)0.73432255,
-(float)0.73653996, (float)0.73874950, (float)0.74095118, (float)0.74314487, (float)0.74533057,
-(float)0.74750835, (float)0.74967808, (float)0.75183982, (float)0.75399351, (float)0.75613910,
-(float)0.75827658, (float)0.76040596, (float)0.76252723, (float)0.76464027, (float)0.76674515,
-(float)0.76884186, (float)0.77093029, (float)0.77301043, (float)0.77508241, (float)0.77714598,
-(float)0.77920127, (float)0.78124821, (float)0.78328675, (float)0.78531694, (float)0.78733873,
-(float)0.78935206, (float)0.79135692, (float)0.79335338, (float)0.79534125, (float)0.79732066,
-(float)0.79929149, (float)0.80125386, (float)0.80320752, (float)0.80515265, (float)0.80708915,
-(float)0.80901700, (float)0.81093621, (float)0.81284672, (float)0.81474853, (float)0.81664157,
-(float)0.81852591, (float)0.82040149, (float)0.82226825, (float)0.82412618, (float)0.82597536,
-(float)0.82781565, (float)0.82964706, (float)0.83146966, (float)0.83328325, (float)0.83508795,
-(float)0.83688378, (float)0.83867061, (float)0.84044838, (float)0.84221727, (float)0.84397703,
-(float)0.84572780, (float)0.84746957, (float)0.84920216, (float)0.85092574, (float)0.85264021,
-(float)0.85434544, (float)0.85604161, (float)0.85772866, (float)0.85940641, (float)0.86107504,
-(float)0.86273444, (float)0.86438453, (float)0.86602545, (float)0.86765707, (float)0.86927933,
-(float)0.87089235, (float)0.87249607, (float)0.87409031, (float)0.87567532, (float)0.87725097,
-(float)0.87881714, (float)0.88037390, (float)0.88192129, (float)0.88345921, (float)0.88498765,
-(float)0.88650668, (float)0.88801610, (float)0.88951612, (float)0.89100653, (float)0.89248741,
-(float)0.89395881, (float)0.89542055, (float)0.89687276, (float)0.89831537, (float)0.89974827,
-(float)0.90117162, (float)0.90258533, (float)0.90398932, (float)0.90538365, (float)0.90676826,
-(float)0.90814316, (float)0.90950841, (float)0.91086388, (float)0.91220951, (float)0.91354549,
-(float)0.91487163, (float)0.91618794, (float)0.91749454, (float)0.91879123, (float)0.92007810,
-(float)0.92135513, (float)0.92262226, (float)0.92387950, (float)0.92512691, (float)0.92636442,
-(float)0.92759192, (float)0.92880958, (float)0.93001723, (float)0.93121493, (float)0.93240267,
-(float)0.93358046, (float)0.93474817, (float)0.93590593, (float)0.93705362, (float)0.93819135,
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-(float)0.03925974, (float)0.03598953, (float)0.03271893, (float)0.02944798, (float)0.02617695,
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-(float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000};
-
-#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
diff --git a/src/modules/audio_processing/utility/Android.mk b/src/modules/audio_processing/utility/Android.mk
deleted file mode 100644
index 7e758ce..0000000
--- a/src/modules/audio_processing/utility/Android.mk
+++ /dev/null
@@ -1,49 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_apm_utility
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := fft4g.c \
-    ring_buffer.c
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=  
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
-    '-DNO_HEAPCHECKER' \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_THREAD_RR' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' 
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := \
-    $(LOCAL_PATH)
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS := 
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
-    libdl \
-    libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/modules/audio_processing/utility/fft4g.c b/src/modules/audio_processing/utility/fft4g.c
deleted file mode 100644
index 9a84368..0000000
--- a/src/modules/audio_processing/utility/fft4g.c
+++ /dev/null
@@ -1,1356 +0,0 @@
-/*
- * http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
- * Copyright Takuya OOURA, 1996-2001
- *
- * You may use, copy, modify and distribute this code for any purpose (include
- * commercial use) and without fee. Please refer to this package when you modify
- * this code.
- *
- * Changes:
- * Trivial type modifications by the WebRTC authors.
- */
-
-/*
-Fast Fourier/Cosine/Sine Transform
-    dimension   :one
-    data length :power of 2
-    decimation  :frequency
-    radix       :4, 2
-    data        :inplace
-    table       :use
-functions
-    cdft: Complex Discrete Fourier Transform
-    rdft: Real Discrete Fourier Transform
-    ddct: Discrete Cosine Transform
-    ddst: Discrete Sine Transform
-    dfct: Cosine Transform of RDFT (Real Symmetric DFT)
-    dfst: Sine Transform of RDFT (Real Anti-symmetric DFT)
-function prototypes
-    void cdft(int, int, float *, int *, float *);
-    void rdft(int, int, float *, int *, float *);
-    void ddct(int, int, float *, int *, float *);
-    void ddst(int, int, float *, int *, float *);
-    void dfct(int, float *, float *, int *, float *);
-    void dfst(int, float *, float *, int *, float *);
-
-
--------- Complex DFT (Discrete Fourier Transform) --------
-    [definition]
-        <case1>
-            X[k] = sum_j=0^n-1 x[j]*exp(2*pi*i*j*k/n), 0<=k<n
-        <case2>
-            X[k] = sum_j=0^n-1 x[j]*exp(-2*pi*i*j*k/n), 0<=k<n
-        (notes: sum_j=0^n-1 is a summation from j=0 to n-1)
-    [usage]
-        <case1>
-            ip[0] = 0; // first time only
-            cdft(2*n, 1, a, ip, w);
-        <case2>
-            ip[0] = 0; // first time only
-            cdft(2*n, -1, a, ip, w);
-    [parameters]
-        2*n            :data length (int)
-                        n >= 1, n = power of 2
-        a[0...2*n-1]   :input/output data (float *)
-                        input data
-                            a[2*j] = Re(x[j]),
-                            a[2*j+1] = Im(x[j]), 0<=j<n
-                        output data
-                            a[2*k] = Re(X[k]),
-                            a[2*k+1] = Im(X[k]), 0<=k<n
-        ip[0...*]      :work area for bit reversal (int *)
-                        length of ip >= 2+sqrt(n)
-                        strictly,
-                        length of ip >=
-                            2+(1<<(int)(log(n+0.5)/log(2))/2).
-                        ip[0],ip[1] are pointers of the cos/sin table.
-        w[0...n/2-1]   :cos/sin table (float *)
-                        w[],ip[] are initialized if ip[0] == 0.
-    [remark]
-        Inverse of
-            cdft(2*n, -1, a, ip, w);
-        is
-            cdft(2*n, 1, a, ip, w);
-            for (j = 0; j <= 2 * n - 1; j++) {
-                a[j] *= 1.0 / n;
-            }
-        .
-
-
--------- Real DFT / Inverse of Real DFT --------
-    [definition]
-        <case1> RDFT
-            R[k] = sum_j=0^n-1 a[j]*cos(2*pi*j*k/n), 0<=k<=n/2
-            I[k] = sum_j=0^n-1 a[j]*sin(2*pi*j*k/n), 0<k<n/2
-        <case2> IRDFT (excluding scale)
-            a[k] = (R[0] + R[n/2]*cos(pi*k))/2 +
-                   sum_j=1^n/2-1 R[j]*cos(2*pi*j*k/n) +
-                   sum_j=1^n/2-1 I[j]*sin(2*pi*j*k/n), 0<=k<n
-    [usage]
-        <case1>
-            ip[0] = 0; // first time only
-            rdft(n, 1, a, ip, w);
-        <case2>
-            ip[0] = 0; // first time only
-            rdft(n, -1, a, ip, w);
-    [parameters]
-        n              :data length (int)
-                        n >= 2, n = power of 2
-        a[0...n-1]     :input/output data (float *)
-                        <case1>
-                            output data
-                                a[2*k] = R[k], 0<=k<n/2
-                                a[2*k+1] = I[k], 0<k<n/2
-                                a[1] = R[n/2]
-                        <case2>
-                            input data
-                                a[2*j] = R[j], 0<=j<n/2
-                                a[2*j+1] = I[j], 0<j<n/2
-                                a[1] = R[n/2]
-        ip[0...*]      :work area for bit reversal (int *)
-                        length of ip >= 2+sqrt(n/2)
-                        strictly,
-                        length of ip >=
-                            2+(1<<(int)(log(n/2+0.5)/log(2))/2).
-                        ip[0],ip[1] are pointers of the cos/sin table.
-        w[0...n/2-1]   :cos/sin table (float *)
-                        w[],ip[] are initialized if ip[0] == 0.
-    [remark]
-        Inverse of
-            rdft(n, 1, a, ip, w);
-        is
-            rdft(n, -1, a, ip, w);
-            for (j = 0; j <= n - 1; j++) {
-                a[j] *= 2.0 / n;
-            }
-        .
-
-
--------- DCT (Discrete Cosine Transform) / Inverse of DCT --------
-    [definition]
-        <case1> IDCT (excluding scale)
-            C[k] = sum_j=0^n-1 a[j]*cos(pi*j*(k+1/2)/n), 0<=k<n
-        <case2> DCT
-            C[k] = sum_j=0^n-1 a[j]*cos(pi*(j+1/2)*k/n), 0<=k<n
-    [usage]
-        <case1>
-            ip[0] = 0; // first time only
-            ddct(n, 1, a, ip, w);
-        <case2>
-            ip[0] = 0; // first time only
-            ddct(n, -1, a, ip, w);
-    [parameters]
-        n              :data length (int)
-                        n >= 2, n = power of 2
-        a[0...n-1]     :input/output data (float *)
-                        output data
-                            a[k] = C[k], 0<=k<n
-        ip[0...*]      :work area for bit reversal (int *)
-                        length of ip >= 2+sqrt(n/2)
-                        strictly,
-                        length of ip >=
-                            2+(1<<(int)(log(n/2+0.5)/log(2))/2).
-                        ip[0],ip[1] are pointers of the cos/sin table.
-        w[0...n*5/4-1] :cos/sin table (float *)
-                        w[],ip[] are initialized if ip[0] == 0.
-    [remark]
-        Inverse of
-            ddct(n, -1, a, ip, w);
-        is
-            a[0] *= 0.5;
-            ddct(n, 1, a, ip, w);
-            for (j = 0; j <= n - 1; j++) {
-                a[j] *= 2.0 / n;
-            }
-        .
-
-
--------- DST (Discrete Sine Transform) / Inverse of DST --------
-    [definition]
-        <case1> IDST (excluding scale)
-            S[k] = sum_j=1^n A[j]*sin(pi*j*(k+1/2)/n), 0<=k<n
-        <case2> DST
-            S[k] = sum_j=0^n-1 a[j]*sin(pi*(j+1/2)*k/n), 0<k<=n
-    [usage]
-        <case1>
-            ip[0] = 0; // first time only
-            ddst(n, 1, a, ip, w);
-        <case2>
-            ip[0] = 0; // first time only
-            ddst(n, -1, a, ip, w);
-    [parameters]
-        n              :data length (int)
-                        n >= 2, n = power of 2
-        a[0...n-1]     :input/output data (float *)
-                        <case1>
-                            input data
-                                a[j] = A[j], 0<j<n
-                                a[0] = A[n]
-                            output data
-                                a[k] = S[k], 0<=k<n
-                        <case2>
-                            output data
-                                a[k] = S[k], 0<k<n
-                                a[0] = S[n]
-        ip[0...*]      :work area for bit reversal (int *)
-                        length of ip >= 2+sqrt(n/2)
-                        strictly,
-                        length of ip >=
-                            2+(1<<(int)(log(n/2+0.5)/log(2))/2).
-                        ip[0],ip[1] are pointers of the cos/sin table.
-        w[0...n*5/4-1] :cos/sin table (float *)
-                        w[],ip[] are initialized if ip[0] == 0.
-    [remark]
-        Inverse of
-            ddst(n, -1, a, ip, w);
-        is
-            a[0] *= 0.5;
-            ddst(n, 1, a, ip, w);
-            for (j = 0; j <= n - 1; j++) {
-                a[j] *= 2.0 / n;
-            }
-        .
-
-
--------- Cosine Transform of RDFT (Real Symmetric DFT) --------
-    [definition]
-        C[k] = sum_j=0^n a[j]*cos(pi*j*k/n), 0<=k<=n
-    [usage]
-        ip[0] = 0; // first time only
-        dfct(n, a, t, ip, w);
-    [parameters]
-        n              :data length - 1 (int)
-                        n >= 2, n = power of 2
-        a[0...n]       :input/output data (float *)
-                        output data
-                            a[k] = C[k], 0<=k<=n
-        t[0...n/2]     :work area (float *)
-        ip[0...*]      :work area for bit reversal (int *)
-                        length of ip >= 2+sqrt(n/4)
-                        strictly,
-                        length of ip >=
-                            2+(1<<(int)(log(n/4+0.5)/log(2))/2).
-                        ip[0],ip[1] are pointers of the cos/sin table.
-        w[0...n*5/8-1] :cos/sin table (float *)
-                        w[],ip[] are initialized if ip[0] == 0.
-    [remark]
-        Inverse of
-            a[0] *= 0.5;
-            a[n] *= 0.5;
-            dfct(n, a, t, ip, w);
-        is
-            a[0] *= 0.5;
-            a[n] *= 0.5;
-            dfct(n, a, t, ip, w);
-            for (j = 0; j <= n; j++) {
-                a[j] *= 2.0 / n;
-            }
-        .
-
-
--------- Sine Transform of RDFT (Real Anti-symmetric DFT) --------
-    [definition]
-        S[k] = sum_j=1^n-1 a[j]*sin(pi*j*k/n), 0<k<n
-    [usage]
-        ip[0] = 0; // first time only
-        dfst(n, a, t, ip, w);
-    [parameters]
-        n              :data length + 1 (int)
-                        n >= 2, n = power of 2
-        a[0...n-1]     :input/output data (float *)
-                        output data
-                            a[k] = S[k], 0<k<n
-                        (a[0] is used for work area)
-        t[0...n/2-1]   :work area (float *)
-        ip[0...*]      :work area for bit reversal (int *)
-                        length of ip >= 2+sqrt(n/4)
-                        strictly,
-                        length of ip >=
-                            2+(1<<(int)(log(n/4+0.5)/log(2))/2).
-                        ip[0],ip[1] are pointers of the cos/sin table.
-        w[0...n*5/8-1] :cos/sin table (float *)
-                        w[],ip[] are initialized if ip[0] == 0.
-    [remark]
-        Inverse of
-            dfst(n, a, t, ip, w);
-        is
-            dfst(n, a, t, ip, w);
-            for (j = 1; j <= n - 1; j++) {
-                a[j] *= 2.0 / n;
-            }
-        .
-
-
-Appendix :
-    The cos/sin table is recalculated when the larger table required.
-    w[] and ip[] are compatible with all routines.
-*/
-
-void cdft(int n, int isgn, float *a, int *ip, float *w)
-{
-    void makewt(int nw, int *ip, float *w);
-    void bitrv2(int n, int *ip, float *a);
-    void bitrv2conj(int n, int *ip, float *a);
-    void cftfsub(int n, float *a, float *w);
-    void cftbsub(int n, float *a, float *w);
-
-    if (n > (ip[0] << 2)) {
-        makewt(n >> 2, ip, w);
-    }
-    if (n > 4) {
-        if (isgn >= 0) {
-            bitrv2(n, ip + 2, a);
-            cftfsub(n, a, w);
-        } else {
-            bitrv2conj(n, ip + 2, a);
-            cftbsub(n, a, w);
-        }
-    } else if (n == 4) {
-        cftfsub(n, a, w);
-    }
-}
-
-
-void rdft(int n, int isgn, float *a, int *ip, float *w)
-{
-    void makewt(int nw, int *ip, float *w);
-    void makect(int nc, int *ip, float *c);
-    void bitrv2(int n, int *ip, float *a);
-    void cftfsub(int n, float *a, float *w);
-    void cftbsub(int n, float *a, float *w);
-    void rftfsub(int n, float *a, int nc, float *c);
-    void rftbsub(int n, float *a, int nc, float *c);
-    int nw, nc;
-    float xi;
-
-    nw = ip[0];
-    if (n > (nw << 2)) {
-        nw = n >> 2;
-        makewt(nw, ip, w);
-    }
-    nc = ip[1];
-    if (n > (nc << 2)) {
-        nc = n >> 2;
-        makect(nc, ip, w + nw);
-    }
-    if (isgn >= 0) {
-        if (n > 4) {
-            bitrv2(n, ip + 2, a);
-            cftfsub(n, a, w);
-            rftfsub(n, a, nc, w + nw);
-        } else if (n == 4) {
-            cftfsub(n, a, w);
-        }
-        xi = a[0] - a[1];
-        a[0] += a[1];
-        a[1] = xi;
-    } else {
-        a[1] = 0.5f * (a[0] - a[1]);
-        a[0] -= a[1];
-        if (n > 4) {
-            rftbsub(n, a, nc, w + nw);
-            bitrv2(n, ip + 2, a);
-            cftbsub(n, a, w);
-        } else if (n == 4) {
-            cftfsub(n, a, w);
-        }
-    }
-}
-
-
-void ddct(int n, int isgn, float *a, int *ip, float *w)
-{
-    void makewt(int nw, int *ip, float *w);
-    void makect(int nc, int *ip, float *c);
-    void bitrv2(int n, int *ip, float *a);
-    void cftfsub(int n, float *a, float *w);
-    void cftbsub(int n, float *a, float *w);
-    void rftfsub(int n, float *a, int nc, float *c);
-    void rftbsub(int n, float *a, int nc, float *c);
-    void dctsub(int n, float *a, int nc, float *c);
-    int j, nw, nc;
-    float xr;
-
-    nw = ip[0];
-    if (n > (nw << 2)) {
-        nw = n >> 2;
-        makewt(nw, ip, w);
-    }
-    nc = ip[1];
-    if (n > nc) {
-        nc = n;
-        makect(nc, ip, w + nw);
-    }
-    if (isgn < 0) {
-        xr = a[n - 1];
-        for (j = n - 2; j >= 2; j -= 2) {
-            a[j + 1] = a[j] - a[j - 1];
-            a[j] += a[j - 1];
-        }
-        a[1] = a[0] - xr;
-        a[0] += xr;
-        if (n > 4) {
-            rftbsub(n, a, nc, w + nw);
-            bitrv2(n, ip + 2, a);
-            cftbsub(n, a, w);
-        } else if (n == 4) {
-            cftfsub(n, a, w);
-        }
-    }
-    dctsub(n, a, nc, w + nw);
-    if (isgn >= 0) {
-        if (n > 4) {
-            bitrv2(n, ip + 2, a);
-            cftfsub(n, a, w);
-            rftfsub(n, a, nc, w + nw);
-        } else if (n == 4) {
-            cftfsub(n, a, w);
-        }
-        xr = a[0] - a[1];
-        a[0] += a[1];
-        for (j = 2; j < n; j += 2) {
-            a[j - 1] = a[j] - a[j + 1];
-            a[j] += a[j + 1];
-        }
-        a[n - 1] = xr;
-    }
-}
-
-
-void ddst(int n, int isgn, float *a, int *ip, float *w)
-{
-    void makewt(int nw, int *ip, float *w);
-    void makect(int nc, int *ip, float *c);
-    void bitrv2(int n, int *ip, float *a);
-    void cftfsub(int n, float *a, float *w);
-    void cftbsub(int n, float *a, float *w);
-    void rftfsub(int n, float *a, int nc, float *c);
-    void rftbsub(int n, float *a, int nc, float *c);
-    void dstsub(int n, float *a, int nc, float *c);
-    int j, nw, nc;
-    float xr;
-
-    nw = ip[0];
-    if (n > (nw << 2)) {
-        nw = n >> 2;
-        makewt(nw, ip, w);
-    }
-    nc = ip[1];
-    if (n > nc) {
-        nc = n;
-        makect(nc, ip, w + nw);
-    }
-    if (isgn < 0) {
-        xr = a[n - 1];
-        for (j = n - 2; j >= 2; j -= 2) {
-            a[j + 1] = -a[j] - a[j - 1];
-            a[j] -= a[j - 1];
-        }
-        a[1] = a[0] + xr;
-        a[0] -= xr;
-        if (n > 4) {
-            rftbsub(n, a, nc, w + nw);
-            bitrv2(n, ip + 2, a);
-            cftbsub(n, a, w);
-        } else if (n == 4) {
-            cftfsub(n, a, w);
-        }
-    }
-    dstsub(n, a, nc, w + nw);
-    if (isgn >= 0) {
-        if (n > 4) {
-            bitrv2(n, ip + 2, a);
-            cftfsub(n, a, w);
-            rftfsub(n, a, nc, w + nw);
-        } else if (n == 4) {
-            cftfsub(n, a, w);
-        }
-        xr = a[0] - a[1];
-        a[0] += a[1];
-        for (j = 2; j < n; j += 2) {
-            a[j - 1] = -a[j] - a[j + 1];
-            a[j] -= a[j + 1];
-        }
-        a[n - 1] = -xr;
-    }
-}
-
-
-void dfct(int n, float *a, float *t, int *ip, float *w)
-{
-    void makewt(int nw, int *ip, float *w);
-    void makect(int nc, int *ip, float *c);
-    void bitrv2(int n, int *ip, float *a);
-    void cftfsub(int n, float *a, float *w);
-    void rftfsub(int n, float *a, int nc, float *c);
-    void dctsub(int n, float *a, int nc, float *c);
-    int j, k, l, m, mh, nw, nc;
-    float xr, xi, yr, yi;
-
-    nw = ip[0];
-    if (n > (nw << 3)) {
-        nw = n >> 3;
-        makewt(nw, ip, w);
-    }
-    nc = ip[1];
-    if (n > (nc << 1)) {
-        nc = n >> 1;
-        makect(nc, ip, w + nw);
-    }
-    m = n >> 1;
-    yi = a[m];
-    xi = a[0] + a[n];
-    a[0] -= a[n];
-    t[0] = xi - yi;
-    t[m] = xi + yi;
-    if (n > 2) {
-        mh = m >> 1;
-        for (j = 1; j < mh; j++) {
-            k = m - j;
-            xr = a[j] - a[n - j];
-            xi = a[j] + a[n - j];
-            yr = a[k] - a[n - k];
-            yi = a[k] + a[n - k];
-            a[j] = xr;
-            a[k] = yr;
-            t[j] = xi - yi;
-            t[k] = xi + yi;
-        }
-        t[mh] = a[mh] + a[n - mh];
-        a[mh] -= a[n - mh];
-        dctsub(m, a, nc, w + nw);
-        if (m > 4) {
-            bitrv2(m, ip + 2, a);
-            cftfsub(m, a, w);
-            rftfsub(m, a, nc, w + nw);
-        } else if (m == 4) {
-            cftfsub(m, a, w);
-        }
-        a[n - 1] = a[0] - a[1];
-        a[1] = a[0] + a[1];
-        for (j = m - 2; j >= 2; j -= 2) {
-            a[2 * j + 1] = a[j] + a[j + 1];
-            a[2 * j - 1] = a[j] - a[j + 1];
-        }
-        l = 2;
-        m = mh;
-        while (m >= 2) {
-            dctsub(m, t, nc, w + nw);
-            if (m > 4) {
-                bitrv2(m, ip + 2, t);
-                cftfsub(m, t, w);
-                rftfsub(m, t, nc, w + nw);
-            } else if (m == 4) {
-                cftfsub(m, t, w);
-            }
-            a[n - l] = t[0] - t[1];
-            a[l] = t[0] + t[1];
-            k = 0;
-            for (j = 2; j < m; j += 2) {
-                k += l << 2;
-                a[k - l] = t[j] - t[j + 1];
-                a[k + l] = t[j] + t[j + 1];
-            }
-            l <<= 1;
-            mh = m >> 1;
-            for (j = 0; j < mh; j++) {
-                k = m - j;
-                t[j] = t[m + k] - t[m + j];
-                t[k] = t[m + k] + t[m + j];
-            }
-            t[mh] = t[m + mh];
-            m = mh;
-        }
-        a[l] = t[0];
-        a[n] = t[2] - t[1];
-        a[0] = t[2] + t[1];
-    } else {
-        a[1] = a[0];
-        a[2] = t[0];
-        a[0] = t[1];
-    }
-}
-
-
-void dfst(int n, float *a, float *t, int *ip, float *w)
-{
-    void makewt(int nw, int *ip, float *w);
-    void makect(int nc, int *ip, float *c);
-    void bitrv2(int n, int *ip, float *a);
-    void cftfsub(int n, float *a, float *w);
-    void rftfsub(int n, float *a, int nc, float *c);
-    void dstsub(int n, float *a, int nc, float *c);
-    int j, k, l, m, mh, nw, nc;
-    float xr, xi, yr, yi;
-
-    nw = ip[0];
-    if (n > (nw << 3)) {
-        nw = n >> 3;
-        makewt(nw, ip, w);
-    }
-    nc = ip[1];
-    if (n > (nc << 1)) {
-        nc = n >> 1;
-        makect(nc, ip, w + nw);
-    }
-    if (n > 2) {
-        m = n >> 1;
-        mh = m >> 1;
-        for (j = 1; j < mh; j++) {
-            k = m - j;
-            xr = a[j] + a[n - j];
-            xi = a[j] - a[n - j];
-            yr = a[k] + a[n - k];
-            yi = a[k] - a[n - k];
-            a[j] = xr;
-            a[k] = yr;
-            t[j] = xi + yi;
-            t[k] = xi - yi;
-        }
-        t[0] = a[mh] - a[n - mh];
-        a[mh] += a[n - mh];
-        a[0] = a[m];
-        dstsub(m, a, nc, w + nw);
-        if (m > 4) {
-            bitrv2(m, ip + 2, a);
-            cftfsub(m, a, w);
-            rftfsub(m, a, nc, w + nw);
-        } else if (m == 4) {
-            cftfsub(m, a, w);
-        }
-        a[n - 1] = a[1] - a[0];
-        a[1] = a[0] + a[1];
-        for (j = m - 2; j >= 2; j -= 2) {
-            a[2 * j + 1] = a[j] - a[j + 1];
-            a[2 * j - 1] = -a[j] - a[j + 1];
-        }
-        l = 2;
-        m = mh;
-        while (m >= 2) {
-            dstsub(m, t, nc, w + nw);
-            if (m > 4) {
-                bitrv2(m, ip + 2, t);
-                cftfsub(m, t, w);
-                rftfsub(m, t, nc, w + nw);
-            } else if (m == 4) {
-                cftfsub(m, t, w);
-            }
-            a[n - l] = t[1] - t[0];
-            a[l] = t[0] + t[1];
-            k = 0;
-            for (j = 2; j < m; j += 2) {
-                k += l << 2;
-                a[k - l] = -t[j] - t[j + 1];
-                a[k + l] = t[j] - t[j + 1];
-            }
-            l <<= 1;
-            mh = m >> 1;
-            for (j = 1; j < mh; j++) {
-                k = m - j;
-                t[j] = t[m + k] + t[m + j];
-                t[k] = t[m + k] - t[m + j];
-            }
-            t[0] = t[m + mh];
-            m = mh;
-        }
-        a[l] = t[0];
-    }
-    a[0] = 0;
-}
-
-
-/* -------- initializing routines -------- */
-
-
-#include <math.h>
-
-void makewt(int nw, int *ip, float *w)
-{
-    void bitrv2(int n, int *ip, float *a);
-    int j, nwh;
-    float delta, x, y;
-
-    ip[0] = nw;
-    ip[1] = 1;
-    if (nw > 2) {
-        nwh = nw >> 1;
-        delta = (float)atan(1.0f) / nwh;
-        w[0] = 1;
-        w[1] = 0;
-        w[nwh] = (float)cos(delta * nwh);
-        w[nwh + 1] = w[nwh];
-        if (nwh > 2) {
-            for (j = 2; j < nwh; j += 2) {
-                x = (float)cos(delta * j);
-                y = (float)sin(delta * j);
-                w[j] = x;
-                w[j + 1] = y;
-                w[nw - j] = y;
-                w[nw - j + 1] = x;
-            }
-            bitrv2(nw, ip + 2, w);
-        }
-    }
-}
-
-
-void makect(int nc, int *ip, float *c)
-{
-    int j, nch;
-    float delta;
-
-    ip[1] = nc;
-    if (nc > 1) {
-        nch = nc >> 1;
-        delta = (float)atan(1.0f) / nch;
-        c[0] = (float)cos(delta * nch);
-        c[nch] = 0.5f * c[0];
-        for (j = 1; j < nch; j++) {
-            c[j] = 0.5f * (float)cos(delta * j);
-            c[nc - j] = 0.5f * (float)sin(delta * j);
-        }
-    }
-}
-
-
-/* -------- child routines -------- */
-
-
-void bitrv2(int n, int *ip, float *a)
-{
-    int j, j1, k, k1, l, m, m2;
-    float xr, xi, yr, yi;
-
-    ip[0] = 0;
-    l = n;
-    m = 1;
-    while ((m << 3) < l) {
-        l >>= 1;
-        for (j = 0; j < m; j++) {
-            ip[m + j] = ip[j] + l;
-        }
-        m <<= 1;
-    }
-    m2 = 2 * m;
-    if ((m << 3) == l) {
-        for (k = 0; k < m; k++) {
-            for (j = 0; j < k; j++) {
-                j1 = 2 * j + ip[k];
-                k1 = 2 * k + ip[j];
-                xr = a[j1];
-                xi = a[j1 + 1];
-                yr = a[k1];
-                yi = a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-                j1 += m2;
-                k1 += 2 * m2;
-                xr = a[j1];
-                xi = a[j1 + 1];
-                yr = a[k1];
-                yi = a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-                j1 += m2;
-                k1 -= m2;
-                xr = a[j1];
-                xi = a[j1 + 1];
-                yr = a[k1];
-                yi = a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-                j1 += m2;
-                k1 += 2 * m2;
-                xr = a[j1];
-                xi = a[j1 + 1];
-                yr = a[k1];
-                yi = a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-            }
-            j1 = 2 * k + m2 + ip[k];
-            k1 = j1 + m2;
-            xr = a[j1];
-            xi = a[j1 + 1];
-            yr = a[k1];
-            yi = a[k1 + 1];
-            a[j1] = yr;
-            a[j1 + 1] = yi;
-            a[k1] = xr;
-            a[k1 + 1] = xi;
-        }
-    } else {
-        for (k = 1; k < m; k++) {
-            for (j = 0; j < k; j++) {
-                j1 = 2 * j + ip[k];
-                k1 = 2 * k + ip[j];
-                xr = a[j1];
-                xi = a[j1 + 1];
-                yr = a[k1];
-                yi = a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-                j1 += m2;
-                k1 += m2;
-                xr = a[j1];
-                xi = a[j1 + 1];
-                yr = a[k1];
-                yi = a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-            }
-        }
-    }
-}
-
-
-void bitrv2conj(int n, int *ip, float *a)
-{
-    int j, j1, k, k1, l, m, m2;
-    float xr, xi, yr, yi;
-
-    ip[0] = 0;
-    l = n;
-    m = 1;
-    while ((m << 3) < l) {
-        l >>= 1;
-        for (j = 0; j < m; j++) {
-            ip[m + j] = ip[j] + l;
-        }
-        m <<= 1;
-    }
-    m2 = 2 * m;
-    if ((m << 3) == l) {
-        for (k = 0; k < m; k++) {
-            for (j = 0; j < k; j++) {
-                j1 = 2 * j + ip[k];
-                k1 = 2 * k + ip[j];
-                xr = a[j1];
-                xi = -a[j1 + 1];
-                yr = a[k1];
-                yi = -a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-                j1 += m2;
-                k1 += 2 * m2;
-                xr = a[j1];
-                xi = -a[j1 + 1];
-                yr = a[k1];
-                yi = -a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-                j1 += m2;
-                k1 -= m2;
-                xr = a[j1];
-                xi = -a[j1 + 1];
-                yr = a[k1];
-                yi = -a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-                j1 += m2;
-                k1 += 2 * m2;
-                xr = a[j1];
-                xi = -a[j1 + 1];
-                yr = a[k1];
-                yi = -a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-            }
-            k1 = 2 * k + ip[k];
-            a[k1 + 1] = -a[k1 + 1];
-            j1 = k1 + m2;
-            k1 = j1 + m2;
-            xr = a[j1];
-            xi = -a[j1 + 1];
-            yr = a[k1];
-            yi = -a[k1 + 1];
-            a[j1] = yr;
-            a[j1 + 1] = yi;
-            a[k1] = xr;
-            a[k1 + 1] = xi;
-            k1 += m2;
-            a[k1 + 1] = -a[k1 + 1];
-        }
-    } else {
-        a[1] = -a[1];
-        a[m2 + 1] = -a[m2 + 1];
-        for (k = 1; k < m; k++) {
-            for (j = 0; j < k; j++) {
-                j1 = 2 * j + ip[k];
-                k1 = 2 * k + ip[j];
-                xr = a[j1];
-                xi = -a[j1 + 1];
-                yr = a[k1];
-                yi = -a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-                j1 += m2;
-                k1 += m2;
-                xr = a[j1];
-                xi = -a[j1 + 1];
-                yr = a[k1];
-                yi = -a[k1 + 1];
-                a[j1] = yr;
-                a[j1 + 1] = yi;
-                a[k1] = xr;
-                a[k1 + 1] = xi;
-            }
-            k1 = 2 * k + ip[k];
-            a[k1 + 1] = -a[k1 + 1];
-            a[k1 + m2 + 1] = -a[k1 + m2 + 1];
-        }
-    }
-}
-
-
-void cftfsub(int n, float *a, float *w)
-{
-    void cft1st(int n, float *a, float *w);
-    void cftmdl(int n, int l, float *a, float *w);
-    int j, j1, j2, j3, l;
-    float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
-    l = 2;
-    if (n > 8) {
-        cft1st(n, a, w);
-        l = 8;
-        while ((l << 2) < n) {
-            cftmdl(n, l, a, w);
-            l <<= 2;
-        }
-    }
-    if ((l << 2) == n) {
-        for (j = 0; j < l; j += 2) {
-            j1 = j + l;
-            j2 = j1 + l;
-            j3 = j2 + l;
-            x0r = a[j] + a[j1];
-            x0i = a[j + 1] + a[j1 + 1];
-            x1r = a[j] - a[j1];
-            x1i = a[j + 1] - a[j1 + 1];
-            x2r = a[j2] + a[j3];
-            x2i = a[j2 + 1] + a[j3 + 1];
-            x3r = a[j2] - a[j3];
-            x3i = a[j2 + 1] - a[j3 + 1];
-            a[j] = x0r + x2r;
-            a[j + 1] = x0i + x2i;
-            a[j2] = x0r - x2r;
-            a[j2 + 1] = x0i - x2i;
-            a[j1] = x1r - x3i;
-            a[j1 + 1] = x1i + x3r;
-            a[j3] = x1r + x3i;
-            a[j3 + 1] = x1i - x3r;
-        }
-    } else {
-        for (j = 0; j < l; j += 2) {
-            j1 = j + l;
-            x0r = a[j] - a[j1];
-            x0i = a[j + 1] - a[j1 + 1];
-            a[j] += a[j1];
-            a[j + 1] += a[j1 + 1];
-            a[j1] = x0r;
-            a[j1 + 1] = x0i;
-        }
-    }
-}
-
-
-void cftbsub(int n, float *a, float *w)
-{
-    void cft1st(int n, float *a, float *w);
-    void cftmdl(int n, int l, float *a, float *w);
-    int j, j1, j2, j3, l;
-    float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
-    l = 2;
-    if (n > 8) {
-        cft1st(n, a, w);
-        l = 8;
-        while ((l << 2) < n) {
-            cftmdl(n, l, a, w);
-            l <<= 2;
-        }
-    }
-    if ((l << 2) == n) {
-        for (j = 0; j < l; j += 2) {
-            j1 = j + l;
-            j2 = j1 + l;
-            j3 = j2 + l;
-            x0r = a[j] + a[j1];
-            x0i = -a[j + 1] - a[j1 + 1];
-            x1r = a[j] - a[j1];
-            x1i = -a[j + 1] + a[j1 + 1];
-            x2r = a[j2] + a[j3];
-            x2i = a[j2 + 1] + a[j3 + 1];
-            x3r = a[j2] - a[j3];
-            x3i = a[j2 + 1] - a[j3 + 1];
-            a[j] = x0r + x2r;
-            a[j + 1] = x0i - x2i;
-            a[j2] = x0r - x2r;
-            a[j2 + 1] = x0i + x2i;
-            a[j1] = x1r - x3i;
-            a[j1 + 1] = x1i - x3r;
-            a[j3] = x1r + x3i;
-            a[j3 + 1] = x1i + x3r;
-        }
-    } else {
-        for (j = 0; j < l; j += 2) {
-            j1 = j + l;
-            x0r = a[j] - a[j1];
-            x0i = -a[j + 1] + a[j1 + 1];
-            a[j] += a[j1];
-            a[j + 1] = -a[j + 1] - a[j1 + 1];
-            a[j1] = x0r;
-            a[j1 + 1] = x0i;
-        }
-    }
-}
-
-
-void cft1st(int n, float *a, float *w)
-{
-    int j, k1, k2;
-    float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
-    float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
-    x0r = a[0] + a[2];
-    x0i = a[1] + a[3];
-    x1r = a[0] - a[2];
-    x1i = a[1] - a[3];
-    x2r = a[4] + a[6];
-    x2i = a[5] + a[7];
-    x3r = a[4] - a[6];
-    x3i = a[5] - a[7];
-    a[0] = x0r + x2r;
-    a[1] = x0i + x2i;
-    a[4] = x0r - x2r;
-    a[5] = x0i - x2i;
-    a[2] = x1r - x3i;
-    a[3] = x1i + x3r;
-    a[6] = x1r + x3i;
-    a[7] = x1i - x3r;
-    wk1r = w[2];
-    x0r = a[8] + a[10];
-    x0i = a[9] + a[11];
-    x1r = a[8] - a[10];
-    x1i = a[9] - a[11];
-    x2r = a[12] + a[14];
-    x2i = a[13] + a[15];
-    x3r = a[12] - a[14];
-    x3i = a[13] - a[15];
-    a[8] = x0r + x2r;
-    a[9] = x0i + x2i;
-    a[12] = x2i - x0i;
-    a[13] = x0r - x2r;
-    x0r = x1r - x3i;
-    x0i = x1i + x3r;
-    a[10] = wk1r * (x0r - x0i);
-    a[11] = wk1r * (x0r + x0i);
-    x0r = x3i + x1r;
-    x0i = x3r - x1i;
-    a[14] = wk1r * (x0i - x0r);
-    a[15] = wk1r * (x0i + x0r);
-    k1 = 0;
-    for (j = 16; j < n; j += 16) {
-        k1 += 2;
-        k2 = 2 * k1;
-        wk2r = w[k1];
-        wk2i = w[k1 + 1];
-        wk1r = w[k2];
-        wk1i = w[k2 + 1];
-        wk3r = wk1r - 2 * wk2i * wk1i;
-        wk3i = 2 * wk2i * wk1r - wk1i;
-        x0r = a[j] + a[j + 2];
-        x0i = a[j + 1] + a[j + 3];
-        x1r = a[j] - a[j + 2];
-        x1i = a[j + 1] - a[j + 3];
-        x2r = a[j + 4] + a[j + 6];
-        x2i = a[j + 5] + a[j + 7];
-        x3r = a[j + 4] - a[j + 6];
-        x3i = a[j + 5] - a[j + 7];
-        a[j] = x0r + x2r;
-        a[j + 1] = x0i + x2i;
-        x0r -= x2r;
-        x0i -= x2i;
-        a[j + 4] = wk2r * x0r - wk2i * x0i;
-        a[j + 5] = wk2r * x0i + wk2i * x0r;
-        x0r = x1r - x3i;
-        x0i = x1i + x3r;
-        a[j + 2] = wk1r * x0r - wk1i * x0i;
-        a[j + 3] = wk1r * x0i + wk1i * x0r;
-        x0r = x1r + x3i;
-        x0i = x1i - x3r;
-        a[j + 6] = wk3r * x0r - wk3i * x0i;
-        a[j + 7] = wk3r * x0i + wk3i * x0r;
-        wk1r = w[k2 + 2];
-        wk1i = w[k2 + 3];
-        wk3r = wk1r - 2 * wk2r * wk1i;
-        wk3i = 2 * wk2r * wk1r - wk1i;
-        x0r = a[j + 8] + a[j + 10];
-        x0i = a[j + 9] + a[j + 11];
-        x1r = a[j + 8] - a[j + 10];
-        x1i = a[j + 9] - a[j + 11];
-        x2r = a[j + 12] + a[j + 14];
-        x2i = a[j + 13] + a[j + 15];
-        x3r = a[j + 12] - a[j + 14];
-        x3i = a[j + 13] - a[j + 15];
-        a[j + 8] = x0r + x2r;
-        a[j + 9] = x0i + x2i;
-        x0r -= x2r;
-        x0i -= x2i;
-        a[j + 12] = -wk2i * x0r - wk2r * x0i;
-        a[j + 13] = -wk2i * x0i + wk2r * x0r;
-        x0r = x1r - x3i;
-        x0i = x1i + x3r;
-        a[j + 10] = wk1r * x0r - wk1i * x0i;
-        a[j + 11] = wk1r * x0i + wk1i * x0r;
-        x0r = x1r + x3i;
-        x0i = x1i - x3r;
-        a[j + 14] = wk3r * x0r - wk3i * x0i;
-        a[j + 15] = wk3r * x0i + wk3i * x0r;
-    }
-}
-
-
-void cftmdl(int n, int l, float *a, float *w)
-{
-    int j, j1, j2, j3, k, k1, k2, m, m2;
-    float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
-    float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
-    m = l << 2;
-    for (j = 0; j < l; j += 2) {
-        j1 = j + l;
-        j2 = j1 + l;
-        j3 = j2 + l;
-        x0r = a[j] + a[j1];
-        x0i = a[j + 1] + a[j1 + 1];
-        x1r = a[j] - a[j1];
-        x1i = a[j + 1] - a[j1 + 1];
-        x2r = a[j2] + a[j3];
-        x2i = a[j2 + 1] + a[j3 + 1];
-        x3r = a[j2] - a[j3];
-        x3i = a[j2 + 1] - a[j3 + 1];
-        a[j] = x0r + x2r;
-        a[j + 1] = x0i + x2i;
-        a[j2] = x0r - x2r;
-        a[j2 + 1] = x0i - x2i;
-        a[j1] = x1r - x3i;
-        a[j1 + 1] = x1i + x3r;
-        a[j3] = x1r + x3i;
-        a[j3 + 1] = x1i - x3r;
-    }
-    wk1r = w[2];
-    for (j = m; j < l + m; j += 2) {
-        j1 = j + l;
-        j2 = j1 + l;
-        j3 = j2 + l;
-        x0r = a[j] + a[j1];
-        x0i = a[j + 1] + a[j1 + 1];
-        x1r = a[j] - a[j1];
-        x1i = a[j + 1] - a[j1 + 1];
-        x2r = a[j2] + a[j3];
-        x2i = a[j2 + 1] + a[j3 + 1];
-        x3r = a[j2] - a[j3];
-        x3i = a[j2 + 1] - a[j3 + 1];
-        a[j] = x0r + x2r;
-        a[j + 1] = x0i + x2i;
-        a[j2] = x2i - x0i;
-        a[j2 + 1] = x0r - x2r;
-        x0r = x1r - x3i;
-        x0i = x1i + x3r;
-        a[j1] = wk1r * (x0r - x0i);
-        a[j1 + 1] = wk1r * (x0r + x0i);
-        x0r = x3i + x1r;
-        x0i = x3r - x1i;
-        a[j3] = wk1r * (x0i - x0r);
-        a[j3 + 1] = wk1r * (x0i + x0r);
-    }
-    k1 = 0;
-    m2 = 2 * m;
-    for (k = m2; k < n; k += m2) {
-        k1 += 2;
-        k2 = 2 * k1;
-        wk2r = w[k1];
-        wk2i = w[k1 + 1];
-        wk1r = w[k2];
-        wk1i = w[k2 + 1];
-        wk3r = wk1r - 2 * wk2i * wk1i;
-        wk3i = 2 * wk2i * wk1r - wk1i;
-        for (j = k; j < l + k; j += 2) {
-            j1 = j + l;
-            j2 = j1 + l;
-            j3 = j2 + l;
-            x0r = a[j] + a[j1];
-            x0i = a[j + 1] + a[j1 + 1];
-            x1r = a[j] - a[j1];
-            x1i = a[j + 1] - a[j1 + 1];
-            x2r = a[j2] + a[j3];
-            x2i = a[j2 + 1] + a[j3 + 1];
-            x3r = a[j2] - a[j3];
-            x3i = a[j2 + 1] - a[j3 + 1];
-            a[j] = x0r + x2r;
-            a[j + 1] = x0i + x2i;
-            x0r -= x2r;
-            x0i -= x2i;
-            a[j2] = wk2r * x0r - wk2i * x0i;
-            a[j2 + 1] = wk2r * x0i + wk2i * x0r;
-            x0r = x1r - x3i;
-            x0i = x1i + x3r;
-            a[j1] = wk1r * x0r - wk1i * x0i;
-            a[j1 + 1] = wk1r * x0i + wk1i * x0r;
-            x0r = x1r + x3i;
-            x0i = x1i - x3r;
-            a[j3] = wk3r * x0r - wk3i * x0i;
-            a[j3 + 1] = wk3r * x0i + wk3i * x0r;
-        }
-        wk1r = w[k2 + 2];
-        wk1i = w[k2 + 3];
-        wk3r = wk1r - 2 * wk2r * wk1i;
-        wk3i = 2 * wk2r * wk1r - wk1i;
-        for (j = k + m; j < l + (k + m); j += 2) {
-            j1 = j + l;
-            j2 = j1 + l;
-            j3 = j2 + l;
-            x0r = a[j] + a[j1];
-            x0i = a[j + 1] + a[j1 + 1];
-            x1r = a[j] - a[j1];
-            x1i = a[j + 1] - a[j1 + 1];
-            x2r = a[j2] + a[j3];
-            x2i = a[j2 + 1] + a[j3 + 1];
-            x3r = a[j2] - a[j3];
-            x3i = a[j2 + 1] - a[j3 + 1];
-            a[j] = x0r + x2r;
-            a[j + 1] = x0i + x2i;
-            x0r -= x2r;
-            x0i -= x2i;
-            a[j2] = -wk2i * x0r - wk2r * x0i;
-            a[j2 + 1] = -wk2i * x0i + wk2r * x0r;
-            x0r = x1r - x3i;
-            x0i = x1i + x3r;
-            a[j1] = wk1r * x0r - wk1i * x0i;
-            a[j1 + 1] = wk1r * x0i + wk1i * x0r;
-            x0r = x1r + x3i;
-            x0i = x1i - x3r;
-            a[j3] = wk3r * x0r - wk3i * x0i;
-            a[j3 + 1] = wk3r * x0i + wk3i * x0r;
-        }
-    }
-}
-
-
-void rftfsub(int n, float *a, int nc, float *c)
-{
-    int j, k, kk, ks, m;
-    float wkr, wki, xr, xi, yr, yi;
-
-    m = n >> 1;
-    ks = 2 * nc / m;
-    kk = 0;
-    for (j = 2; j < m; j += 2) {
-        k = n - j;
-        kk += ks;
-        wkr = 0.5f - c[nc - kk];
-        wki = c[kk];
-        xr = a[j] - a[k];
-        xi = a[j + 1] + a[k + 1];
-        yr = wkr * xr - wki * xi;
-        yi = wkr * xi + wki * xr;
-        a[j] -= yr;
-        a[j + 1] -= yi;
-        a[k] += yr;
-        a[k + 1] -= yi;
-    }
-}
-
-
-void rftbsub(int n, float *a, int nc, float *c)
-{
-    int j, k, kk, ks, m;
-    float wkr, wki, xr, xi, yr, yi;
-
-    a[1] = -a[1];
-    m = n >> 1;
-    ks = 2 * nc / m;
-    kk = 0;
-    for (j = 2; j < m; j += 2) {
-        k = n - j;
-        kk += ks;
-        wkr = 0.5f - c[nc - kk];
-        wki = c[kk];
-        xr = a[j] - a[k];
-        xi = a[j + 1] + a[k + 1];
-        yr = wkr * xr + wki * xi;
-        yi = wkr * xi - wki * xr;
-        a[j] -= yr;
-        a[j + 1] = yi - a[j + 1];
-        a[k] += yr;
-        a[k + 1] = yi - a[k + 1];
-    }
-    a[m + 1] = -a[m + 1];
-}
-
-
-void dctsub(int n, float *a, int nc, float *c)
-{
-    int j, k, kk, ks, m;
-    float wkr, wki, xr;
-
-    m = n >> 1;
-    ks = nc / n;
-    kk = 0;
-    for (j = 1; j < m; j++) {
-        k = n - j;
-        kk += ks;
-        wkr = c[kk] - c[nc - kk];
-        wki = c[kk] + c[nc - kk];
-        xr = wki * a[j] - wkr * a[k];
-        a[j] = wkr * a[j] + wki * a[k];
-        a[k] = xr;
-    }
-    a[m] *= c[0];
-}
-
-
-void dstsub(int n, float *a, int nc, float *c)
-{
-    int j, k, kk, ks, m;
-    float wkr, wki, xr;
-
-    m = n >> 1;
-    ks = nc / n;
-    kk = 0;
-    for (j = 1; j < m; j++) {
-        k = n - j;
-        kk += ks;
-        wkr = c[kk] - c[nc - kk];
-        wki = c[kk] + c[nc - kk];
-        xr = wki * a[k] - wkr * a[j];
-        a[k] = wkr * a[k] + wki * a[j];
-        a[j] = xr;
-    }
-    a[m] *= c[0];
-}
-
diff --git a/src/modules/audio_processing/utility/fft4g.h b/src/modules/audio_processing/utility/fft4g.h
deleted file mode 100644
index 373ff14..0000000
--- a/src/modules/audio_processing/utility/fft4g.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_FFT4G_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_FFT4G_H_
-
-void rdft(int, int, float *, int *, float *);
-void cdft(int, int, float *, int *, float *);
-
-#endif
-
diff --git a/src/modules/audio_processing/utility/ring_buffer.c b/src/modules/audio_processing/utility/ring_buffer.c
deleted file mode 100644
index ea2e354..0000000
--- a/src/modules/audio_processing/utility/ring_buffer.c
+++ /dev/null
@@ -1,239 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Provides a generic ring buffer that can be written to and read from with
- * arbitrarily sized blocks. The AEC uses this for several different tasks.
- */
-
-#include <stdlib.h>
-#include <string.h>
-#include "ring_buffer.h"
-
-typedef struct {
-    int readPos;
-    int writePos;
-    int size;
-    char rwWrap;
-    bufdata_t *data;
-} buf_t;
-
-enum {SAME_WRAP, DIFF_WRAP};
-
-int WebRtcApm_CreateBuffer(void **bufInst, int size)
-{
-    buf_t *buf = NULL;
-
-    if (size < 0) {
-        return -1;
-    }
-
-    buf = malloc(sizeof(buf_t));
-    *bufInst = buf;
-    if (buf == NULL) {
-        return -1;
-    }
-
-    buf->data = malloc(size*sizeof(bufdata_t));
-    if (buf->data == NULL) {
-        free(buf);
-        buf = NULL;
-        return -1;
-    }
-
-    buf->size = size;
-    return 0;
-}
-
-int WebRtcApm_InitBuffer(void *bufInst)
-{
-    buf_t *buf = (buf_t*)bufInst;
-
-    buf->readPos = 0;
-    buf->writePos = 0;
-    buf->rwWrap = SAME_WRAP;
-
-    // Initialize buffer to zeros
-    memset(buf->data, 0, sizeof(bufdata_t)*buf->size);
-
-    return 0;
-}
-
-int WebRtcApm_FreeBuffer(void *bufInst)
-{
-    buf_t *buf = (buf_t*)bufInst;
-
-    if (buf == NULL) {
-        return -1;
-    }
-
-    free(buf->data);
-    free(buf);
-
-    return 0;
-}
-
-int WebRtcApm_ReadBuffer(void *bufInst, bufdata_t *data, int size)
-{
-    buf_t *buf = (buf_t*)bufInst;
-    int n = 0, margin = 0;
-
-    if (size <= 0 || size > buf->size) {
-        return -1;
-    }
-
-    n = size;
-    if (buf->rwWrap == DIFF_WRAP) {
-        margin = buf->size - buf->readPos;
-        if (n > margin) {
-            buf->rwWrap = SAME_WRAP;
-            memcpy(data, buf->data + buf->readPos,
-                sizeof(bufdata_t)*margin);
-            buf->readPos = 0;
-            n = size - margin;
-        }
-        else {
-            memcpy(data, buf->data + buf->readPos,
-                sizeof(bufdata_t)*n);
-            buf->readPos += n;
-            return n;
-        }
-    }
-
-    if (buf->rwWrap == SAME_WRAP) {
-        margin = buf->writePos - buf->readPos;
-        if (margin > n)
-            margin = n;
-        memcpy(data + size - n, buf->data + buf->readPos,
-            sizeof(bufdata_t)*margin);
-        buf->readPos += margin;
-        n -= margin;
-    }
-
-    return size - n;
-}
-
-int WebRtcApm_WriteBuffer(void *bufInst, const bufdata_t *data, int size)
-{
-    buf_t *buf = (buf_t*)bufInst;
-    int n = 0, margin = 0;
-
-    if (size < 0 || size > buf->size) {
-        return -1;
-    }
-
-    n = size;
-    if (buf->rwWrap == SAME_WRAP) {
-        margin = buf->size - buf->writePos;
-        if (n > margin) {
-            buf->rwWrap = DIFF_WRAP;
-            memcpy(buf->data + buf->writePos, data,
-                sizeof(bufdata_t)*margin);
-            buf->writePos = 0;
-            n = size - margin;
-        }
-        else {
-            memcpy(buf->data + buf->writePos, data,
-                sizeof(bufdata_t)*n);
-            buf->writePos += n;
-            return n;
-        }
-    }
-
-    if (buf->rwWrap == DIFF_WRAP) {
-        margin = buf->readPos - buf->writePos;
-        if (margin > n)
-            margin = n;
-        memcpy(buf->data + buf->writePos, data + size - n,
-            sizeof(bufdata_t)*margin);
-        buf->writePos += margin;
-        n -= margin;
-    }
-
-    return size - n;
-}
-
-int WebRtcApm_FlushBuffer(void *bufInst, int size)
-{
-    buf_t *buf = (buf_t*)bufInst;
-    int n = 0, margin = 0;
-
-    if (size <= 0 || size > buf->size) {
-        return -1;
-    }
-
-    n = size;
-    if (buf->rwWrap == DIFF_WRAP) {
-        margin = buf->size - buf->readPos;
-        if (n > margin) {
-            buf->rwWrap = SAME_WRAP;
-            buf->readPos = 0;
-            n = size - margin;
-        }
-        else {
-            buf->readPos += n;
-            return n;
-        }
-    }
-
-    if (buf->rwWrap == SAME_WRAP) {
-        margin = buf->writePos - buf->readPos;
-        if (margin > n)
-            margin = n;
-        buf->readPos += margin;
-        n -= margin;
-    }
-
-    return size - n;
-}
-
-int WebRtcApm_StuffBuffer(void *bufInst, int size)
-{
-    buf_t *buf = (buf_t*)bufInst;
-    int n = 0, margin = 0;
-
-    if (size <= 0 || size > buf->size) {
-        return -1;
-    }
-
-    n = size;
-    if (buf->rwWrap == SAME_WRAP) {
-        margin = buf->readPos;
-        if (n > margin) {
-            buf->rwWrap = DIFF_WRAP;
-            buf->readPos = buf->size - 1;
-            n -= margin + 1;
-        }
-        else {
-            buf->readPos -= n;
-            return n;
-        }
-    }
-
-    if (buf->rwWrap == DIFF_WRAP) {
-        margin = buf->readPos - buf->writePos;
-        if (margin > n)
-            margin = n;
-        buf->readPos -= margin;
-        n -= margin;
-    }
-
-    return size - n;
-}
-
-int WebRtcApm_get_buffer_size(const void *bufInst)
-{
-    const buf_t *buf = (buf_t*)bufInst;
-
-    if (buf->rwWrap == SAME_WRAP)
-        return buf->writePos - buf->readPos;
-    else
-        return buf->size - buf->readPos + buf->writePos;
-}
diff --git a/src/modules/audio_processing/utility/ring_buffer.h b/src/modules/audio_processing/utility/ring_buffer.h
deleted file mode 100644
index 0fd261d..0000000
--- a/src/modules/audio_processing/utility/ring_buffer.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Specifies the interface for the AEC generic buffer.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
-
-// Determines buffer datatype
-typedef short bufdata_t;
-
-// Unless otherwise specified, functions return 0 on success and -1 on error
-int WebRtcApm_CreateBuffer(void **bufInst, int size);
-int WebRtcApm_InitBuffer(void *bufInst);
-int WebRtcApm_FreeBuffer(void *bufInst);
-
-// Returns number of samples read
-int WebRtcApm_ReadBuffer(void *bufInst, bufdata_t *data, int size);
-
-// Returns number of samples written
-int WebRtcApm_WriteBuffer(void *bufInst, const bufdata_t *data, int size);
-
-// Returns number of samples flushed
-int WebRtcApm_FlushBuffer(void *bufInst, int size);
-
-// Returns number of samples stuffed
-int WebRtcApm_StuffBuffer(void *bufInst, int size);
-
-// Returns number of samples in buffer
-int WebRtcApm_get_buffer_size(const void *bufInst);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
diff --git a/src/modules/audio_processing/utility/util.gyp b/src/modules/audio_processing/utility/util.gyp
deleted file mode 100644
index 3348da8..0000000
--- a/src/modules/audio_processing/utility/util.gyp
+++ /dev/null
@@ -1,36 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../../common_settings.gypi',
-  ],
-  'targets': [
-    {
-      'target_name': 'apm_util',
-      'type': '<(library)',
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '.',
-        ],
-      },
-      'sources': [
-        'ring_buffer.c',
-        'ring_buffer.h',
-        'fft4g.c',
-        'fft4g.h',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/modules/interface/module.h b/src/modules/interface/module.h
deleted file mode 100644
index f270978..0000000
--- a/src/modules/interface/module.h
+++ /dev/null
@@ -1,33 +0,0 @@
-#ifndef MODULE_H
-#define MODULE_H
-
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-class Module
-{
-public:
-    // Returns version of the module and its components.
-    virtual int32_t Version(char* version,
-                            uint32_t& remainingBufferInBytes,
-                            uint32_t& position) const = 0;
-
-    // Change the unique identifier of this object.
-    virtual int32_t ChangeUniqueId(const int32_t id) = 0;
-
-    // Returns the number of milliseconds until the module want a worker
-    // thread to call Process.
-    virtual int32_t TimeUntilNextProcess() = 0 ;
-
-    // Process any pending tasks such as timeouts.
-    virtual int32_t Process() = 0 ;
-
-protected:
-    virtual ~Module() {}
-};
-
-} // namespace webrtc
-
-#endif // MODULE_H
diff --git a/src/modules/interface/module_common_types.h b/src/modules/interface/module_common_types.h
deleted file mode 100644
index 1865b0f..0000000
--- a/src/modules/interface/module_common_types.h
+++ /dev/null
@@ -1,1012 +0,0 @@
-#ifndef MODULE_COMMON_TYPES_H
-#define MODULE_COMMON_TYPES_H
-
-#include <cstring> // memcpy
-#include <assert.h>
-
-#include "typedefs.h"
-#include "common_types.h"
-
-#ifdef _WIN32
-    #pragma warning(disable:4351)       // remove warning "new behavior: elements of array
-                                        // 'array' will be default initialized"
-#endif
-
-namespace webrtc
-{
-struct RTPHeader
-{
-    bool           markerBit;
-    WebRtc_UWord8  payloadType;
-    WebRtc_UWord16 sequenceNumber;
-    WebRtc_UWord32 timestamp;
-    WebRtc_UWord32 ssrc;
-    WebRtc_UWord8  numCSRCs;
-    WebRtc_UWord32 arrOfCSRCs[kRtpCsrcSize];
-    WebRtc_UWord8  paddingLength;
-    WebRtc_UWord16 headerLength;
-};
-
-struct RTPAudioHeader
-{
-    WebRtc_UWord8  numEnergy;                         // number of valid entries in arrOfEnergy
-    WebRtc_UWord8  arrOfEnergy[kRtpCsrcSize];   // one energy byte (0-9) per channel
-    bool           isCNG;                             // is this CNG
-    WebRtc_UWord8  channel;                           // number of channels 2 = stereo
-};
-
-struct RTPVideoHeaderH263
-{
-    bool independentlyDecodable;  // H.263-1998 if no P bit it's not independently decodable
-    bool bits;                    // H.263 mode B, Xor the lasy byte of previus packet with the
-                                  // first byte of this packet
-};
-enum {kNoPictureId = -1};
-struct RTPVideoHeaderVP8
-{
-    bool           startBit;        // Start of partition.
-    bool           stopBit;         // Stop of partition.
-    WebRtc_Word16  pictureId;       // Picture ID index, 15 bits;
-                                    // kNoPictureId if PictureID does not exist.
-    bool           nonReference;    // Frame is discardable.
-};
-union RTPVideoTypeHeader
-{
-    RTPVideoHeaderH263      H263;
-    RTPVideoHeaderVP8       VP8;
-};
-
-enum RTPVideoCodecTypes
-{
-    kRTPVideoGeneric  = 0,
-    kRTPVideoH263     = 1,
-    kRTPVideoMPEG4    = 5,
-    kRTPVideoVP8      = 8,
-    kRTPVideoNoVideo  = 10,
-    kRTPVideoFEC      = 11,
-    kRTPVideoI420     = 12
-};
-struct RTPVideoHeader
-{
-    WebRtc_UWord16          width;                  // size
-    WebRtc_UWord16          height;
-
-    bool                    isFirstPacket;   // first packet in frame
-    RTPVideoCodecTypes      codec;
-    RTPVideoTypeHeader      codecHeader;
-};
-union RTPTypeHeader
-{
-    RTPAudioHeader  Audio;
-    RTPVideoHeader  Video;
-};
-
-struct WebRtcRTPHeader
-{
-    RTPHeader       header;
-    FrameType       frameType;
-    RTPTypeHeader   type;
-};
-
-class RTPFragmentationHeader
-{
-public:
-    RTPFragmentationHeader() :
-        fragmentationVectorSize(0),
-        fragmentationOffset(NULL),
-        fragmentationLength(NULL),
-        fragmentationTimeDiff(NULL),
-        fragmentationPlType(NULL)
-    {};
-
-    ~RTPFragmentationHeader()
-    {
-        delete [] fragmentationOffset;
-        delete [] fragmentationLength;
-        delete [] fragmentationTimeDiff;
-        delete [] fragmentationPlType;
-    }
-
-    RTPFragmentationHeader& operator=(const RTPFragmentationHeader& header)
-    {
-        if(this == &header)
-        {
-            return *this;
-        }
-
-        if(header.fragmentationVectorSize != fragmentationVectorSize)
-        {
-            // new size of vectors
-
-            // delete old
-            delete [] fragmentationOffset;
-            fragmentationOffset = NULL;
-            delete [] fragmentationLength;
-            fragmentationLength = NULL;
-            delete [] fragmentationTimeDiff;
-            fragmentationTimeDiff = NULL;
-            delete [] fragmentationPlType;
-            fragmentationPlType = NULL;
-
-            if(header.fragmentationVectorSize > 0)
-            {
-                // allocate new
-                if(header.fragmentationOffset)
-                {
-                    fragmentationOffset = new WebRtc_UWord32[header.fragmentationVectorSize];
-                }
-                if(header.fragmentationLength)
-                {
-                    fragmentationLength = new WebRtc_UWord32[header.fragmentationVectorSize];
-                }
-                if(header.fragmentationTimeDiff)
-                {
-                    fragmentationTimeDiff = new WebRtc_UWord16[header.fragmentationVectorSize];
-                }
-                if(header.fragmentationPlType)
-                {
-                    fragmentationPlType = new WebRtc_UWord8[header.fragmentationVectorSize];
-                }
-            }
-            // set new size
-            fragmentationVectorSize =   header.fragmentationVectorSize;
-        }
-
-        if(header.fragmentationVectorSize > 0)
-        {
-            // copy values
-            if(header.fragmentationOffset)
-            {
-                memcpy(fragmentationOffset, header.fragmentationOffset,
-                        header.fragmentationVectorSize * sizeof(WebRtc_UWord32));
-            }
-            if(header.fragmentationLength)
-            {
-                memcpy(fragmentationLength, header.fragmentationLength,
-                        header.fragmentationVectorSize * sizeof(WebRtc_UWord32));
-            }
-            if(header.fragmentationTimeDiff)
-            {
-                memcpy(fragmentationTimeDiff, header.fragmentationTimeDiff,
-                        header.fragmentationVectorSize * sizeof(WebRtc_UWord16));
-            }
-            if(header.fragmentationPlType)
-            {
-                memcpy(fragmentationPlType, header.fragmentationPlType,
-                        header.fragmentationVectorSize * sizeof(WebRtc_UWord8));
-            }
-        }
-        return *this;
-    }
-    void VerifyAndAllocateFragmentationHeader( const WebRtc_UWord16 size)
-    {
-        if( fragmentationVectorSize < size)
-        {
-            WebRtc_UWord16 oldVectorSize = fragmentationVectorSize;
-            {
-                // offset
-                WebRtc_UWord32* oldOffsets = fragmentationOffset;
-                fragmentationOffset = new WebRtc_UWord32[size];
-                memset(fragmentationOffset+oldVectorSize, 0,
-                       sizeof(WebRtc_UWord32)*(size-oldVectorSize));
-                // copy old values
-                memcpy(fragmentationOffset,oldOffsets, sizeof(WebRtc_UWord32) * oldVectorSize);
-                delete[] oldOffsets;
-            }
-            // length
-            {
-                WebRtc_UWord32* oldLengths = fragmentationLength;
-                fragmentationLength = new WebRtc_UWord32[size];
-                memset(fragmentationLength+oldVectorSize, 0,
-                       sizeof(WebRtc_UWord32) * (size- oldVectorSize));
-                memcpy(fragmentationLength, oldLengths,
-                       sizeof(WebRtc_UWord32) * oldVectorSize);
-                delete[] oldLengths;
-            }
-            // time diff
-            {
-                WebRtc_UWord16* oldTimeDiffs = fragmentationTimeDiff;
-                fragmentationTimeDiff = new WebRtc_UWord16[size];
-                memset(fragmentationTimeDiff+oldVectorSize, 0,
-                       sizeof(WebRtc_UWord16) * (size- oldVectorSize));
-                memcpy(fragmentationTimeDiff, oldTimeDiffs,
-                       sizeof(WebRtc_UWord16) * oldVectorSize);
-                delete[] oldTimeDiffs;
-            }
-            // payload type
-            {
-                WebRtc_UWord8* oldTimePlTypes = fragmentationPlType;
-                fragmentationPlType = new WebRtc_UWord8[size];
-                memset(fragmentationPlType+oldVectorSize, 0,
-                       sizeof(WebRtc_UWord8) * (size- oldVectorSize));
-                memcpy(fragmentationPlType, oldTimePlTypes,
-                       sizeof(WebRtc_UWord8) * oldVectorSize);
-                delete[] oldTimePlTypes;
-            }
-            fragmentationVectorSize = size;
-        }
-    }
-
-    WebRtc_UWord16    fragmentationVectorSize;    // Number of fragmentations
-    WebRtc_UWord32*   fragmentationOffset;        // Offset of pointer to data for each fragm.
-    WebRtc_UWord32*   fragmentationLength;        // Data size for each fragmentation
-    WebRtc_UWord16*   fragmentationTimeDiff;      // Timestamp difference relative "now" for
-                                                  // each fragmentation
-    WebRtc_UWord8*    fragmentationPlType;        // Payload type of each fragmentation
-};
-
-struct RTCPVoIPMetric
-{
-    // RFC 3611 4.7
-    WebRtc_UWord8     lossRate;
-    WebRtc_UWord8     discardRate;
-    WebRtc_UWord8     burstDensity;
-    WebRtc_UWord8     gapDensity;
-    WebRtc_UWord16    burstDuration;
-    WebRtc_UWord16    gapDuration;
-    WebRtc_UWord16    roundTripDelay;
-    WebRtc_UWord16    endSystemDelay;
-    WebRtc_UWord8     signalLevel;
-    WebRtc_UWord8     noiseLevel;
-    WebRtc_UWord8     RERL;
-    WebRtc_UWord8     Gmin;
-    WebRtc_UWord8     Rfactor;
-    WebRtc_UWord8     extRfactor;
-    WebRtc_UWord8     MOSLQ;
-    WebRtc_UWord8     MOSCQ;
-    WebRtc_UWord8     RXconfig;
-    WebRtc_UWord16    JBnominal;
-    WebRtc_UWord16    JBmax;
-    WebRtc_UWord16    JBabsMax;
-};
-
-// class describing a complete, or parts of an encoded frame.
-class EncodedVideoData
-{
-public:
-    EncodedVideoData() :
-        completeFrame(false),
-        missingFrame(false),
-        payloadData(NULL),
-        payloadSize(0),
-        bufferSize(0)
-    {};
-
-    EncodedVideoData(const EncodedVideoData& data)
-    {
-        payloadType         = data.payloadType;
-        timeStamp           = data.timeStamp;
-        renderTimeMs        = data.renderTimeMs;
-        encodedWidth        = data.encodedWidth;
-        encodedHeight       = data.encodedHeight;
-        completeFrame       = data.completeFrame;
-        missingFrame        = data.missingFrame;
-        payloadSize         = data.payloadSize;
-        fragmentationHeader = data.fragmentationHeader;
-        frameType           = data.frameType;
-        codec               = data.codec;
-        if (data.payloadSize > 0)
-        {
-            payloadData = new WebRtc_UWord8[data.payloadSize];
-            memcpy(payloadData, data.payloadData, data.payloadSize);
-        }
-        else
-        {
-            payloadData = NULL;
-        }
-    }
-
-
-    ~EncodedVideoData()
-    {
-        delete [] payloadData;
-    };
-
-    EncodedVideoData& operator=(const EncodedVideoData& data)
-    {
-        if (this == &data)
-        {
-            return *this;
-        }
-        payloadType         = data.payloadType;
-        timeStamp           = data.timeStamp;
-        renderTimeMs        = data.renderTimeMs;
-        encodedWidth        = data.encodedWidth;
-        encodedHeight       = data.encodedHeight;
-        completeFrame       = data.completeFrame;
-        missingFrame        = data.missingFrame;
-        payloadSize         = data.payloadSize;
-        fragmentationHeader = data.fragmentationHeader;
-        frameType           = data.frameType;
-        codec               = data.codec;
-        if (data.payloadSize > 0)
-        {
-            delete [] payloadData;
-            payloadData = new WebRtc_UWord8[data.payloadSize];
-            memcpy(payloadData, data.payloadData, data.payloadSize);
-            bufferSize = data.payloadSize;
-        }
-        return *this;
-    };
-    void VerifyAndAllocate( const WebRtc_UWord32 size)
-    {
-        if (bufferSize < size)
-        {
-            WebRtc_UWord8* oldPayload = payloadData;
-            payloadData = new WebRtc_UWord8[size];
-            memcpy(payloadData, oldPayload, sizeof(WebRtc_UWord8) * payloadSize);
-
-            bufferSize = size;
-            delete[] oldPayload;
-        }
-    }
-
-    WebRtc_UWord8               payloadType;
-    WebRtc_UWord32              timeStamp;
-    WebRtc_Word64               renderTimeMs;
-    WebRtc_UWord32              encodedWidth;
-    WebRtc_UWord32              encodedHeight;
-    bool                        completeFrame;
-    bool                        missingFrame;
-    WebRtc_UWord8*              payloadData;
-    WebRtc_UWord32              payloadSize;
-    WebRtc_UWord32              bufferSize;
-    RTPFragmentationHeader      fragmentationHeader;
-    FrameType                   frameType;
-    VideoCodecType              codec;
-};
-
-// Video Content Metrics
-struct VideoContentMetrics
-{
-    VideoContentMetrics(): motionMagnitudeNZ(0), sizeZeroMotion(0), spatialPredErr(0),
-            spatialPredErrH(0), spatialPredErrV(0), motionPredErr(0), 
-            motionHorizontalness(0), motionClusterDistortion(0),
-            nativeWidth(0), nativeHeight(0), contentChange(false) {   }
-    void Reset(){ motionMagnitudeNZ = 0; sizeZeroMotion = 0; spatialPredErr = 0;
-            spatialPredErrH = 0; spatialPredErrV = 0; motionPredErr = 0; 
-            motionHorizontalness = 0; motionClusterDistortion = 0;
-            nativeWidth = 0; nativeHeight = 0; contentChange = false; }
-
-    float            motionMagnitudeNZ;
-    float            sizeZeroMotion;
-    float            spatialPredErr;
-    float            spatialPredErrH;
-    float            spatialPredErrV;
-    float            motionPredErr;
-    float            motionHorizontalness;
-    float            motionClusterDistortion;
-    WebRtc_UWord32   nativeWidth;
-    WebRtc_UWord32   nativeHeight;
-    WebRtc_UWord32   nativeFrameRate;
-    bool             contentChange;
-};
-
-/*************************************************
- *
- * VideoFrame class
- *
- * The VideoFrame class allows storing and
- * handling of video frames.
- *
- *
- *************************************************/
-class VideoFrame
-{
-public:
-    VideoFrame();
-    ~VideoFrame();
-    /**
-    * Verifies that current allocated buffer size is larger than or equal to the input size.
-    * If the current buffer size is smaller, a new allocation is made and the old buffer data
-    * is copied to the new buffer.
-    * Buffer size is updated to minimumSize.
-    */
-    WebRtc_Word32 VerifyAndAllocate(const WebRtc_UWord32 minimumSize);
-    /**
-    *    Update length of data buffer in frame. Function verifies that new length is less or
-    *    equal to allocated size.
-    */
-    WebRtc_Word32 SetLength(const WebRtc_UWord32 newLength);
-    /*
-    *    Swap buffer and size data
-    */
-    WebRtc_Word32 Swap(WebRtc_UWord8*& newMemory,
-                       WebRtc_UWord32& newLength,
-                       WebRtc_UWord32& newSize);
-    /*
-    *    Swap buffer and size data
-    */
-    WebRtc_Word32 SwapFrame(VideoFrame& videoFrame);
-    /**
-    *    Copy buffer: If newLength is bigger than allocated size, a new buffer of size length
-    *    is allocated.
-    */
-    WebRtc_Word32 CopyFrame(const VideoFrame& videoFrame);
-    /**
-    *    Copy buffer: If newLength is bigger than allocated size, a new buffer of size length
-    *    is allocated.
-    */
-    WebRtc_Word32 CopyFrame(WebRtc_UWord32 length, const WebRtc_UWord8* sourceBuffer);
-    /**
-    *    Delete VideoFrame and resets members to zero
-    */
-    void Free();
-    /**
-    *   Set frame timestamp (90kHz)
-    */
-    void SetTimeStamp(const WebRtc_UWord32 timeStamp) {_timeStamp = timeStamp;}
-    /**
-    *   Get pointer to frame buffer
-    */
-    WebRtc_UWord8*    Buffer() const {return _buffer;}
-
-    WebRtc_UWord8*&   Buffer() {return _buffer;}
-
-    /**
-    *   Get allocated buffer size
-    */
-    WebRtc_UWord32    Size() const {return _bufferSize;}
-    /**
-    *   Get frame length
-    */
-    WebRtc_UWord32    Length() const {return _bufferLength;}
-    /**
-    *   Get frame timestamp (90kHz)
-    */
-    WebRtc_UWord32    TimeStamp() const {return _timeStamp;}
-    /**
-    *   Get frame width
-    */
-    WebRtc_UWord32    Width() const {return _width;}
-    /**
-    *   Get frame height
-    */
-    WebRtc_UWord32    Height() const {return _height;}
-    /**
-    *   Set frame width
-    */
-    void   SetWidth(const WebRtc_UWord32 width)  {_width = width;}
-    /**
-    *   Set frame height
-    */
-    void  SetHeight(const WebRtc_UWord32 height) {_height = height;}
-    /**
-    *   Set render time in miliseconds
-    */
-    void SetRenderTime(const WebRtc_Word64 renderTimeMs) {_renderTimeMs = renderTimeMs;}
-    /**
-    *  Get render time in miliseconds
-    */
-    WebRtc_Word64    RenderTimeMs() const {return _renderTimeMs;}
-
-private:
-    void Set(WebRtc_UWord8* buffer,
-             WebRtc_UWord32 size,
-             WebRtc_UWord32 length,
-             WebRtc_UWord32 timeStamp);
-
-    WebRtc_UWord8*          _buffer;          // Pointer to frame buffer
-    WebRtc_UWord32          _bufferSize;      // Allocated buffer size
-    WebRtc_UWord32          _bufferLength;    // Length (in bytes) of buffer
-    WebRtc_UWord32          _timeStamp;       // Timestamp of frame (90kHz)
-    WebRtc_UWord32          _width;
-    WebRtc_UWord32          _height;
-    WebRtc_Word64           _renderTimeMs;
-}; // end of VideoFrame class declaration
-
-// inline implementation of VideoFrame class:
-inline
-VideoFrame::VideoFrame():
-    _buffer(0),
-    _bufferSize(0),
-    _bufferLength(0),
-    _timeStamp(0),
-    _width(0),
-    _height(0),
-    _renderTimeMs(0)
-{
-    //
-}
-inline
-VideoFrame::~VideoFrame()
-{
-    if(_buffer)
-    {
-        delete [] _buffer;
-        _buffer = NULL;
-    }
-}
-
-
-inline
-WebRtc_Word32
-VideoFrame::VerifyAndAllocate(const WebRtc_UWord32 minimumSize)
-{
-    if (minimumSize < 1)
-    {
-        return -1;
-    }
-    if(minimumSize > _bufferSize)
-    {
-        // create buffer of sufficient size
-        WebRtc_UWord8* newBufferBuffer = new WebRtc_UWord8[minimumSize];
-        if(_buffer)
-        {
-            // copy old data
-            memcpy(newBufferBuffer, _buffer, _bufferSize);
-            delete [] _buffer;
-        }
-        _buffer = newBufferBuffer;
-        _bufferSize = minimumSize;
-    }
-     return 0;
-}
-
-inline
-WebRtc_Word32
-VideoFrame::SetLength(const WebRtc_UWord32 newLength)
-{
-    if (newLength >_bufferSize )
-    { // can't accomodate new value
-        return -1;
-    }
-     _bufferLength = newLength;
-     return 0;
-}
-
-inline
-WebRtc_Word32
-VideoFrame::SwapFrame(VideoFrame& videoFrame)
-{
-    WebRtc_UWord32 tmpTimeStamp  = _timeStamp;
-    WebRtc_UWord32 tmpWidth      = _width;
-    WebRtc_UWord32 tmpHeight     = _height;
-    WebRtc_Word64  tmpRenderTime = _renderTimeMs;
-
-    _timeStamp = videoFrame._timeStamp;
-    _width = videoFrame._width;
-    _height = videoFrame._height;
-    _renderTimeMs = videoFrame._renderTimeMs;
-
-    videoFrame._timeStamp = tmpTimeStamp;
-    videoFrame._width = tmpWidth;
-    videoFrame._height = tmpHeight;
-    videoFrame._renderTimeMs = tmpRenderTime;
-
-    return Swap(videoFrame._buffer, videoFrame._bufferLength, videoFrame._bufferSize);
-}
-
-inline
-WebRtc_Word32
-VideoFrame::Swap(WebRtc_UWord8*& newMemory, WebRtc_UWord32& newLength, WebRtc_UWord32& newSize)
-{
-    WebRtc_UWord8* tmpBuffer = _buffer;
-    WebRtc_UWord32 tmpLength = _bufferLength;
-    WebRtc_UWord32 tmpSize = _bufferSize;
-    _buffer = newMemory;
-    _bufferLength = newLength;
-    _bufferSize = newSize;
-    newMemory = tmpBuffer;
-    newLength = tmpLength;
-    newSize = tmpSize;
-    return 0;
-}
-
-inline
-WebRtc_Word32
-VideoFrame::CopyFrame(WebRtc_UWord32 length, const WebRtc_UWord8* sourceBuffer)
-{
-    if (length > _bufferSize)
-    {
-        WebRtc_Word32 ret = VerifyAndAllocate(length);
-        if (ret < 0)
-        {
-            return ret;
-        }
-    }
-     memcpy(_buffer, sourceBuffer, length);
-    _bufferLength = length;
-    return 0;
-}
-
-inline
-WebRtc_Word32
-VideoFrame::CopyFrame(const VideoFrame& videoFrame)
-{
-    if(CopyFrame(videoFrame.Length(), videoFrame.Buffer()) != 0)
-    {
-        return -1;
-    }
-    _timeStamp = videoFrame._timeStamp;
-    _width = videoFrame._width;
-    _height = videoFrame._height;
-    _renderTimeMs = videoFrame._renderTimeMs;
-    return 0;
-}
-
-inline
-void
-VideoFrame::Free()
-{
-    _timeStamp = 0;
-    _bufferLength = 0;
-    _bufferSize = 0;
-    _height = 0;
-    _width = 0;
-    _renderTimeMs = 0;
-
-    if(_buffer)
-    {
-        delete [] _buffer;
-        _buffer = NULL;
-    }
-}
-
-
-/*************************************************
- *
- * AudioFrame class
- *
- * The AudioFrame class holds up to 60 ms wideband
- * audio. It allows for adding and subtracting frames
- * while keeping track of the resulting states.
- *
- * Note
- * - The +operator assume that you would never add
- *   exact opposite frames when deciding the resulting
- *   state. To do this use the -operator.
- *
- * - _audioChannel of 1 indicated mono, and 2
- *   indicates stereo.
- *
- * - _payloadDataLengthInSamples is the number of
- *   samples per channel. Therefore, the total
- *   number of samples in _payloadData is
- *   (_payloadDataLengthInSamples * _audioChannel).
- *
- * - Stereo data is stored in interleaved fashion
- *   starting with the left channel.
- *
- *************************************************/
-class AudioFrame
-{
-public:
-    enum{kMaxAudioFrameSizeSamples = 3840}; // stereo 32KHz 60ms 2*32*60
-
-    enum VADActivity
-    {
-        kVadActive  = 0,
-        kVadPassive = 1,
-        kVadUnknown = 2
-    };
-    enum SpeechType
-    {
-        kNormalSpeech = 0,
-        kPLC          = 1,
-        kCNG          = 2,
-        kPLCCNG       = 3,
-        kUndefined    = 4
-    };
-
-    AudioFrame();
-    virtual ~AudioFrame();
-
-    WebRtc_Word32 UpdateFrame(
-        const WebRtc_Word32  id,
-        const WebRtc_UWord32 timeStamp,
-        const WebRtc_Word16* payloadData,
-        const WebRtc_UWord16 payloadDataLengthInSamples,
-        const WebRtc_UWord32 frequencyInHz,
-        const SpeechType     speechType,
-        const VADActivity    vadActivity,
-        const WebRtc_UWord8  audioChannel = 1,
-        const WebRtc_Word32  volume = -1,
-        const WebRtc_Word32  energy = -1);
-
-    AudioFrame& Append(const AudioFrame& rhs);
-
-    void Mute() const;
-
-    AudioFrame& operator=(const AudioFrame& rhs);
-    AudioFrame& operator>>=(const WebRtc_Word32 rhs);
-    AudioFrame& operator+=(const AudioFrame& rhs);
-    AudioFrame& operator-=(const AudioFrame& rhs);
-
-    WebRtc_Word32  _id;
-    WebRtc_UWord32 _timeStamp;
-
-    // Supporting Stereo, stereo samples are interleaved
-    mutable WebRtc_Word16 _payloadData[kMaxAudioFrameSizeSamples];
-    WebRtc_UWord16 _payloadDataLengthInSamples;
-    WebRtc_UWord32 _frequencyInHz;
-    WebRtc_UWord8  _audioChannel;
-    SpeechType   _speechType;
-    VADActivity  _vadActivity;
-
-    WebRtc_UWord32 _energy;
-    WebRtc_Word32  _volume;
-};
-
-inline
-AudioFrame::AudioFrame()
-    :
-    _id(-1),
-    _timeStamp(0),
-    _payloadData(),
-    _payloadDataLengthInSamples(0),
-    _frequencyInHz(0),
-    _audioChannel(1),
-    _speechType(kUndefined),
-    _vadActivity(kVadUnknown),
-    _energy(0xffffffff),
-    _volume(0xffffffff)
-{
-}
-
-inline
-AudioFrame::~AudioFrame()
-{
-}
-
-inline
-WebRtc_Word32
-AudioFrame::UpdateFrame(
-    const WebRtc_Word32  id,
-    const WebRtc_UWord32 timeStamp,
-    const WebRtc_Word16* payloadData,
-    const WebRtc_UWord16 payloadDataLengthInSamples,
-    const WebRtc_UWord32 frequencyInHz,
-    const SpeechType     speechType,
-    const VADActivity    vadActivity,
-    const WebRtc_UWord8  audioChannel,
-    const WebRtc_Word32  volume,
-    const WebRtc_Word32  energy)
-{
-    _id            = id;
-    _timeStamp     = timeStamp;
-    _frequencyInHz = frequencyInHz;
-    _speechType    = speechType;
-    _vadActivity   = vadActivity;
-    _volume        = volume;
-    _audioChannel  = audioChannel;
-    _energy        = energy;
-
-    if((payloadDataLengthInSamples > kMaxAudioFrameSizeSamples) ||
-        (audioChannel > 2) || (audioChannel < 1))
-    {
-        _payloadDataLengthInSamples = 0;
-        return -1;
-    }
-    _payloadDataLengthInSamples = payloadDataLengthInSamples;
-    if(payloadData != NULL)
-    {
-        memcpy(_payloadData, payloadData, sizeof(WebRtc_Word16) *
-            payloadDataLengthInSamples * _audioChannel);
-    }
-    else
-    {
-        memset(_payloadData,0,sizeof(WebRtc_Word16) *
-            payloadDataLengthInSamples * _audioChannel);
-    }
-    return 0;
-}
-
-inline
-void
-AudioFrame::Mute() const
-{
-  memset(_payloadData, 0, _payloadDataLengthInSamples * sizeof(WebRtc_Word16));
-}
-
-inline
-AudioFrame&
-AudioFrame::operator=(const AudioFrame& rhs)
-{
-    // Sanity Check
-    if((rhs._payloadDataLengthInSamples > kMaxAudioFrameSizeSamples) ||
-        (rhs._audioChannel > 2) ||
-        (rhs._audioChannel < 1))
-    {
-        return *this;
-    }
-    if(this == &rhs)
-    {
-        return *this;
-    }
-    _id               = rhs._id;
-    _timeStamp        = rhs._timeStamp;
-    _frequencyInHz    = rhs._frequencyInHz;
-    _speechType       = rhs._speechType;
-    _vadActivity      = rhs._vadActivity;
-    _volume           = rhs._volume;
-    _audioChannel     = rhs._audioChannel;
-    _energy           = rhs._energy;
-
-    _payloadDataLengthInSamples = rhs._payloadDataLengthInSamples;
-    memcpy(_payloadData, rhs._payloadData,
-        sizeof(WebRtc_Word16) * rhs._payloadDataLengthInSamples * _audioChannel);
-
-    return *this;
-}
-
-inline
-AudioFrame&
-AudioFrame::operator>>=(const WebRtc_Word32 rhs)
-{
-    assert((_audioChannel > 0) && (_audioChannel < 3));
-    if((_audioChannel > 2) ||
-        (_audioChannel < 1))
-    {
-        return *this;
-    }
-    for(WebRtc_UWord16 i = 0; i < _payloadDataLengthInSamples * _audioChannel; i++)
-    {
-        _payloadData[i] = WebRtc_Word16(_payloadData[i] >> rhs);
-    }
-    return *this;
-}
-
-inline
-AudioFrame&
-AudioFrame::Append(const AudioFrame& rhs)
-{
-    // Sanity check
-    assert((_audioChannel > 0) && (_audioChannel < 3));
-    if((_audioChannel > 2) ||
-        (_audioChannel < 1))
-    {
-        return *this;
-    }
-    if(_audioChannel != rhs._audioChannel)
-    {
-        return *this;
-    }
-    if((_vadActivity == kVadActive) ||
-        rhs._vadActivity == kVadActive)
-    {
-        _vadActivity = kVadActive;
-    }
-    else if((_vadActivity == kVadUnknown) ||
-        rhs._vadActivity == kVadUnknown)
-    {
-        _vadActivity = kVadUnknown;
-    }
-    if(_speechType != rhs._speechType)
-    {
-        _speechType = kUndefined;
-    }
-
-    WebRtc_UWord16 offset = _payloadDataLengthInSamples * _audioChannel;
-    for(WebRtc_UWord16 i = 0;
-        i < rhs._payloadDataLengthInSamples * rhs._audioChannel;
-        i++)
-    {
-        _payloadData[offset+i] = rhs._payloadData[i];
-    }
-    _payloadDataLengthInSamples += rhs._payloadDataLengthInSamples;
-    return *this;
-}
-
-// merge vectors
-inline
-AudioFrame&
-AudioFrame::operator+=(const AudioFrame& rhs)
-{
-    // Sanity check
-    assert((_audioChannel > 0) && (_audioChannel < 3));
-    if((_audioChannel > 2) ||
-        (_audioChannel < 1))
-    {
-        return *this;
-    }
-    if(_audioChannel != rhs._audioChannel)
-    {
-        return *this;
-    }
-    bool noPrevData = false;
-    if(_payloadDataLengthInSamples != rhs._payloadDataLengthInSamples)
-    {
-        if(_payloadDataLengthInSamples == 0)
-        {
-            // special case we have no data to start with
-            _payloadDataLengthInSamples = rhs._payloadDataLengthInSamples;
-            noPrevData = true;
-        } else
-        {
-          return *this;
-        }
-    }
-
-    if((_vadActivity == kVadActive) ||
-        rhs._vadActivity == kVadActive)
-    {
-        _vadActivity = kVadActive;
-    }
-    else if((_vadActivity == kVadUnknown) ||
-        rhs._vadActivity == kVadUnknown)
-    {
-        _vadActivity = kVadUnknown;
-    }
-
-    if(_speechType != rhs._speechType)
-    {
-        _speechType = kUndefined;
-    }
-
-    if(noPrevData)
-    {
-        memcpy(_payloadData, rhs._payloadData,
-          sizeof(WebRtc_Word16) * rhs._payloadDataLengthInSamples * _audioChannel);
-    } else
-    {
-      // IMPROVEMENT this can be done very fast in assembly
-      for(WebRtc_UWord16 i = 0; i < _payloadDataLengthInSamples * _audioChannel; i++)
-      {
-          WebRtc_Word32 wrapGuard = (WebRtc_Word32)_payloadData[i] +
-                  (WebRtc_Word32)rhs._payloadData[i];
-          if(wrapGuard < -32768)
-          {
-              _payloadData[i] = -32768;
-          }else if(wrapGuard > 32767)
-          {
-              _payloadData[i] = 32767;
-          }else
-          {
-              _payloadData[i] = (WebRtc_Word16)wrapGuard;
-          }
-      }
-    }
-    _energy = 0xffffffff;
-    _volume = 0xffffffff;
-    return *this;
-}
-
-inline
-AudioFrame&
-AudioFrame::operator-=(const AudioFrame& rhs)
-{
-    // Sanity check
-    assert((_audioChannel > 0) && (_audioChannel < 3));
-    if((_audioChannel > 2)||
-        (_audioChannel < 1))
-    {
-        return *this;
-    }
-    if((_payloadDataLengthInSamples != rhs._payloadDataLengthInSamples) ||
-        (_audioChannel != rhs._audioChannel))
-    {
-        return *this;
-    }
-    if((_vadActivity != kVadPassive) ||
-        rhs._vadActivity != kVadPassive)
-    {
-        _vadActivity = kVadUnknown;
-    }
-    _speechType = kUndefined;
-
-    for(WebRtc_UWord16 i = 0; i < _payloadDataLengthInSamples * _audioChannel; i++)
-    {
-        WebRtc_Word32 wrapGuard = (WebRtc_Word32)_payloadData[i] -
-                (WebRtc_Word32)rhs._payloadData[i];
-        if(wrapGuard < -32768)
-        {
-            _payloadData[i] = -32768;
-        }
-        else if(wrapGuard > 32767)
-        {
-            _payloadData[i] = 32767;
-        }
-        else
-        {
-            _payloadData[i] = (WebRtc_Word16)wrapGuard;
-        }
-    }
-    _energy = 0xffffffff;
-    _volume = 0xffffffff;
-    return *this;
-}
-
-} // namespace webrtc
-
-#endif // MODULE_COMMON_TYPES_H
diff --git a/src/system_wrappers/OWNERS b/src/system_wrappers/OWNERS
deleted file mode 100644
index 32dcbbc..0000000
--- a/src/system_wrappers/OWNERS
+++ /dev/null
@@ -1,7 +0,0 @@
-hellner@google.com
-pwestin@google.com
-perkj@google.com
-henrika@google.com
-grunell@google.com
-mflodman@google.com
-niklase@google.com
\ No newline at end of file
diff --git a/src/system_wrappers/interface/aligned_malloc.h b/src/system_wrappers/interface/aligned_malloc.h
deleted file mode 100644
index c229435..0000000
--- a/src/system_wrappers/interface/aligned_malloc.h
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
-
-#include <stddef.h>
-
-namespace webrtc
-{
-    void* AlignedMalloc(
-        size_t size,
-        size_t alignment);
-    void AlignedFree(
-        void* memBlock);
-}
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
diff --git a/src/system_wrappers/interface/atomic32_wrapper.h b/src/system_wrappers/interface/atomic32_wrapper.h
deleted file mode 100644
index 40862fb..0000000
--- a/src/system_wrappers/interface/atomic32_wrapper.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Atomic system independant 32-bit integer.
-// Note: uses full memory barriers.
-// Note: assumes 32-bit (or higher) system
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_WRAPPER_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-class Atomic32Impl;
-class Atomic32Wrapper
-{
-public:
-    Atomic32Wrapper(WebRtc_Word32 initialValue = 0);
-    ~Atomic32Wrapper();
-
-    // Prefix operator!
-    WebRtc_Word32 operator++();
-    WebRtc_Word32 operator--();
-
-    Atomic32Wrapper& operator=(const Atomic32Wrapper& rhs);
-    Atomic32Wrapper& operator=(WebRtc_Word32 rhs);
-
-    WebRtc_Word32 operator+=(WebRtc_Word32 rhs);
-    WebRtc_Word32 operator-=(WebRtc_Word32 rhs);
-
-    // Sets the value atomically to newValue if the value equals compare value.
-    // The function returns true if the exchange happened.
-    bool CompareExchange(WebRtc_Word32 newValue, WebRtc_Word32 compareValue);
-    WebRtc_Word32 Value() const;
-private:
-    // Disable the + and - operator since it's unclear what these operations
-    // should do.
-    Atomic32Wrapper operator+(const Atomic32Wrapper& rhs);
-    Atomic32Wrapper operator-(const Atomic32Wrapper& rhs);
-
-    WebRtc_Word32& operator++(int);
-    WebRtc_Word32& operator--(int);
-
-    // Cheshire cat to hide the implementation (faster than
-    // using virtual functions)
-    Atomic32Impl& _impl;
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_WRAPPER_H_
diff --git a/src/system_wrappers/interface/condition_variable_wrapper.h b/src/system_wrappers/interface/condition_variable_wrapper.h
deleted file mode 100644
index c040fbf..0000000
--- a/src/system_wrappers/interface/condition_variable_wrapper.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class ConditionVariableWrapper
-{
-public:
-    // Factory method, constructor disabled.
-    static ConditionVariableWrapper* CreateConditionVariable();
-
-    virtual ~ConditionVariableWrapper() {}
-
-    // Calling thread will atomically release critSect and wait until next
-    // some other thread calls Wake() or WakeAll().
-    virtual void SleepCS(CriticalSectionWrapper& critSect) = 0;
-
-    // Same as above but with a timeout.
-    virtual bool SleepCS(CriticalSectionWrapper& critSect,
-                         unsigned long maxTimeInMS) = 0;
-
-    // Wakes one thread calling SleepCS().
-    virtual void Wake() = 0;
-
-    // Wakes all threads calling SleepCS().
-    virtual void WakeAll() = 0;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
diff --git a/src/system_wrappers/interface/constructor_magic.h b/src/system_wrappers/interface/constructor_magic.h
deleted file mode 100644
index b2aabc5..0000000
--- a/src/system_wrappers/interface/constructor_magic.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * WebRtc
- * Copy from third_party/libjingle/source/talk/base/constructormagic.h
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONSTRUCTOR_MAGIC_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONSTRUCTOR_MAGIC_H_
-
-#ifndef DISALLOW_ASSIGN
-#define DISALLOW_ASSIGN(TypeName) \
-  void operator=(const TypeName&)
-#endif
-
-#ifndef DISALLOW_COPY_AND_ASSIGN
-// A macro to disallow the evil copy constructor and operator= functions
-// This should be used in the private: declarations for a class
-#define DISALLOW_COPY_AND_ASSIGN(TypeName)    \
-  TypeName(const TypeName&);                    \
-  DISALLOW_ASSIGN(TypeName)
-#endif
-
-#ifndef DISALLOW_EVIL_CONSTRUCTORS
-// Alternative, less-accurate legacy name.
-#define DISALLOW_EVIL_CONSTRUCTORS(TypeName) \
-  DISALLOW_COPY_AND_ASSIGN(TypeName)
-#endif
-
-#ifndef DISALLOW_IMPLICIT_CONSTRUCTORS
-// A macro to disallow all the implicit constructors, namely the
-// default constructor, copy constructor and operator= functions.
-//
-// This should be used in the private: declarations for a class
-// that wants to prevent anyone from instantiating it. This is
-// especially useful for classes containing only static methods.
-#define DISALLOW_IMPLICIT_CONSTRUCTORS(TypeName) \
-  TypeName();                                    \
-  DISALLOW_EVIL_CONSTRUCTORS(TypeName)
-#endif
-
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONSTRUCTOR_MAGIC_H_
diff --git a/src/system_wrappers/interface/cpu_features_wrapper.h b/src/system_wrappers/interface/cpu_features_wrapper.h
deleted file mode 100644
index 5d8a828..0000000
--- a/src/system_wrappers/interface/cpu_features_wrapper.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
-
-#if defined(__cplusplus) || defined(c_plusplus)
-extern "C" {
-#endif
-
-// list of features.
-typedef enum {
-  kSSE2,
-  kSSE3
-} CPUFeature;
-
-typedef int (*WebRtc_CPUInfo)(CPUFeature feature);
-// returns true if the CPU supports the feature.
-extern WebRtc_CPUInfo WebRtc_GetCPUInfo;
-// No CPU feature is available => straight C path.
-extern WebRtc_CPUInfo WebRtc_GetCPUInfoNoASM;
-
-#if defined(__cplusplus) || defined(c_plusplus)
-}    // extern "C"
-#endif
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
diff --git a/src/system_wrappers/interface/cpu_wrapper.h b/src/system_wrappers/interface/cpu_wrapper.h
deleted file mode 100644
index b72c20c..0000000
--- a/src/system_wrappers/interface/cpu_wrapper.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_WRAPPER_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-class CpuWrapper
-{
-public:
-    static WebRtc_UWord32 DetectNumberOfCores();
-
-    static CpuWrapper* CreateCpu();
-    virtual ~CpuWrapper() {}
-
-    // Returns the average CPU usage for all processors. The CPU usage can be
-    // between and including 0 to 100 (%)
-    virtual WebRtc_Word32 CpuUsage() = 0;
-    virtual WebRtc_Word32 CpuUsage(WebRtc_Word8* processName,
-                                   WebRtc_UWord32 length) = 0;
-    virtual WebRtc_Word32 CpuUsage(WebRtc_UWord32  dwProcessID) = 0;
-
-    // The CPU usage per core is returned in cpu_usage. The CPU can be between
-    // and including 0 to 100 (%)
-    // Note that the pointer passed as cpu_usage is redirected to a local member
-    // of the CPU wrapper.
-    // numCores is the number of cores in the cpu_usage array.
-    virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& numCores,
-                                            WebRtc_UWord32*& cpu_usage) = 0;
-
-    virtual void Reset() = 0;
-    virtual void Stop() = 0;
-
-protected:
-    CpuWrapper() {}
-
-private:
-    static WebRtc_UWord32 _numberOfCores;
-
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_WRAPPER_H_
diff --git a/src/system_wrappers/interface/critical_section_wrapper.h b/src/system_wrappers/interface/critical_section_wrapper.h
deleted file mode 100644
index ad31497..0000000
--- a/src/system_wrappers/interface/critical_section_wrapper.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
-
-// If the critical section is heavily contended it may be beneficial to use
-// read/write locks instead.
-
-#include "common_types.h"
-
-namespace webrtc {
-class CriticalSectionWrapper
-{
-public:
-    // Factory method, constructor disabled
-    static CriticalSectionWrapper* CreateCriticalSection();
-
-    virtual ~CriticalSectionWrapper() {}
-
-    // Tries to grab lock, beginning of a critical section. Will wait for the
-    // lock to become available if the grab failed.
-    virtual void Enter() = 0;
-
-    // Returns a grabbed lock, end of critical section.
-    virtual void Leave() = 0;
-};
-
-// RAII extension of the critical section. Prevents Enter/Leave missmatches and
-// provides more compact critical section syntax.
-class CriticalSectionScoped
-{
-public:
-    CriticalSectionScoped(CriticalSectionWrapper& critsec)
-        :
-        _ptrCritSec(&critsec)
-    {
-        _ptrCritSec->Enter();
-    }
-
-    ~CriticalSectionScoped()
-    {
-        if (_ptrCritSec)
-        {
-            Leave();
-        }
-    }
-
-private:
-    void Leave()
-    {
-        _ptrCritSec->Leave();
-        _ptrCritSec = 0;
-    }
-
-    CriticalSectionWrapper* _ptrCritSec;
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
diff --git a/src/system_wrappers/interface/event_wrapper.h b/src/system_wrappers/interface/event_wrapper.h
deleted file mode 100644
index 0c9a908..0000000
--- a/src/system_wrappers/interface/event_wrapper.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
-
-namespace webrtc {
-enum EventTypeWrapper
-{
-    kEventSignaled = 1,
-    kEventError = 2,
-    kEventTimeout = 3
-};
-
-#define WEBRTC_EVENT_10_SEC   10000
-#define WEBRTC_EVENT_INFINITE 0xffffffff
-
-class EventWrapper
-{
-public:
-    // Factory method. Constructor disabled.
-    static EventWrapper* Create();
-    virtual ~EventWrapper() {}
-
-    // Releases threads who are calling Wait() and has started waiting. Please
-    // note that a thread calling Wait() will not start waiting immediately.
-    // assumptions to the contrary is a very common source of issues in
-    // multithreaded programming.
-    // Set is sticky in the sense that it will release at least one thread
-    // either immediately or some time in the future.
-    virtual bool Set() = 0;
-
-    // Prevents future Wait() calls from finishing without a new Set() call.
-    virtual bool Reset() = 0;
-
-    // Puts the calling thread into a wait state. The thread may be released
-    // by a Set() call depending on if other threads are waiting and if so on
-    // timing. The thread that was released will call Reset() before leaving
-    // preventing more threads from being released. If multiple threads
-    // are waiting for the same Set(), only one (random) thread is guaranteed to
-    // be released. It is possible that multiple (random) threads are released
-    // Depending on timing.
-    virtual EventTypeWrapper Wait(unsigned long maxTime) = 0;
-
-    // Starts a timer that will call a non-sticky version of Set() either once
-    // or periodically. If the timer is periodic it ensures that there is no
-    // drift over time relative to the system clock.
-    virtual bool StartTimer(bool periodic, unsigned long time) = 0;
-
-    virtual bool StopTimer() = 0;
-
-    // Only implemented on Windows
-    // Returns 1 if a key has been pressed since last call to this function.
-    // -1 indicates failure
-    // 0 indicates no key has been pressed since last call
-    // TODO(hellner) this function does not seem to belong here
-    static int KeyPressed();
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
diff --git a/src/system_wrappers/interface/file_wrapper.h b/src/system_wrappers/interface/file_wrapper.h
deleted file mode 100644
index 8f0cd8c..0000000
--- a/src/system_wrappers/interface/file_wrapper.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
-
-#include "common_types.h"
-#include "typedefs.h"
-
-// Implementation of an InStream and OutStream that can read (exclusive) or
-// write from/to a file.
-
-namespace webrtc {
-class FileWrapper : public InStream, public OutStream
-{
-public:
-    enum { kMaxFileNameSize = 1024};
-    enum { kFileMaxTextMessageSize = 1024};
-
-    // Factory method. Constructor disabled.
-    static FileWrapper* Create();
-
-    // Returns true if a file has been opened.
-    virtual bool Open() const = 0;
-
-    // Opens a file in read or write mode, decided by the readOnly parameter.
-    virtual WebRtc_Word32 OpenFile(const WebRtc_Word8* fileNameUTF8,
-                                   const bool readOnly,
-                                   const bool loop = false,
-                                   const bool text = false) = 0;
-
-    virtual WebRtc_Word32 CloseFile() = 0;
-
-    // Limits the file size.
-    virtual WebRtc_Word32 SetMaxFileSize(WebRtc_Word32 bytes)  = 0;
-
-    // Flush any pending writes.
-    virtual WebRtc_Word32 Flush() = 0;
-
-    // Returns the opened file's name in fileNameUTF8. size is the allocated
-    // size of fileNameUTF8. The name will be truncated if the size of
-    // fileNameUTF8 is to small.
-    virtual WebRtc_Word32 FileName(WebRtc_Word8* fileNameUTF8,
-                                   WebRtc_UWord32 size) const = 0;
-
-    // Write text to the opened file. The written text can contain plain text
-    // and text with type specifiers in the same way as sprintf works.
-    virtual WebRtc_Word32 WriteText(const WebRtc_Word8* text, ...) = 0;
-
-    // Reads len number of bytes from buf to file.
-    virtual int Read(void* buf, int len) = 0;
-
-    // Writes len number of bytes to buf from file. Please note that the actual
-    // writing to file may happen some time later. Call flush to force a write
-    // to take affect
-    virtual bool Write(const void *buf,int len) = 0;
-
-    // Rewinds the file to the start. Only available when OpenFile() has been
-    // called with loop argument set to true. Or readOnly argument has been set
-    // to false.
-    virtual int Rewind() = 0;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
diff --git a/src/system_wrappers/interface/list_wrapper.h b/src/system_wrappers/interface/list_wrapper.h
deleted file mode 100644
index bc10ad4..0000000
--- a/src/system_wrappers/interface/list_wrapper.h
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LIST_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LIST_WRAPPER_H_
-
-#include "constructor_magic.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class ListItem
-{
-friend class ListWrapper;
-
-public:
-    ListItem(const void* ptr);
-    ListItem(const unsigned int item);
-    virtual ~ListItem();
-    void* GetItem() const;
-    unsigned int GetUnsignedItem() const;
-
-protected:
-    ListItem* next_;
-    ListItem* prev_;
-
-private:
-    const void*         item_ptr_;
-    const unsigned int  item_;
-    DISALLOW_COPY_AND_ASSIGN(ListItem);
-};
-
-class ListWrapper
-{
-public:
-    ListWrapper();
-    virtual ~ListWrapper();
-
-    // Returns the number of elements stored in the list.
-    unsigned int GetSize() const;
-
-    // Puts a pointer to anything last in the list.
-    int PushBack(const void* ptr);
-    // Puts a pointer to anything first in the list.
-    int PushFront(const void* ptr);
-
-    // Puts a copy of the specified integer last in the list.
-    int PushBack(const unsigned int item_id);
-    // Puts a copy of the specified integer first in the list.
-    int PushFront(const unsigned int item_id);
-
-    // Pops the first ListItem from the list
-    int PopFront();
-
-    // Pops the last ListItem from the list
-    int PopBack();
-
-    // Returns true if the list is empty
-    bool Empty() const;
-
-    // Returns a pointer to the first ListItem in the list.
-    ListItem* First() const;
-
-    // Returns a pointer to the last ListItem in the list.
-    ListItem* Last() const;
-
-    // Returns a pointer to the ListItem stored after item in the list.
-    ListItem* Next(ListItem* item) const;
-
-    // Returns a pointer to the ListItem stored before item in the list.
-    ListItem* Previous(ListItem* item) const;
-
-    // Removes item from the list.
-    int Erase(ListItem* item);
-
-    // Insert list item after existing_previous_item. Please note that new_item
-    // must be created using new ListItem(). The map will take ownership of
-    // new_item following a successfull insert. If insert fails new_item will
-    // not be released by the List
-    int Insert(ListItem* existing_previous_item,
-               ListItem* new_item);
-
-    // Insert list item before existing_next_item. Please note that new_item
-    // must be created using new ListItem(). The map will take ownership of
-    // new_item following a successfull insert. If insert fails new_item will
-    // not be released by the List
-    int InsertBefore(ListItem* existing_next_item,
-                     ListItem* new_item);
-
-private:
-    void PushBackImpl(ListItem* item);
-    void PushFrontImpl(ListItem* item);
-
-    CriticalSectionWrapper* critical_section_;
-    ListItem* first_;
-    ListItem* last_;
-    unsigned int size_;
-    DISALLOW_COPY_AND_ASSIGN(ListWrapper);
-};
-} //namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LIST_WRAPPER_H_
diff --git a/src/system_wrappers/interface/map_wrapper.h b/src/system_wrappers/interface/map_wrapper.h
deleted file mode 100644
index 9297382..0000000
--- a/src/system_wrappers/interface/map_wrapper.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_MAP_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_MAP_WRAPPER_H_
-
-#include <map>
-
-#include "constructor_magic.h"
-
-namespace webrtc {
-class MapItem
-{
-friend class MapWrapper;
-
-public:
-    MapItem(int id, void* ptr);
-    virtual ~MapItem();
-    void* GetItem();
-    int GetId();
-    unsigned int GetUnsignedId();
-    void SetItem(void* ptr);
-
-private:
-    int   item_id_;
-    void* item_pointer_;
-    DISALLOW_COPY_AND_ASSIGN(MapItem);
-};
-
-class MapWrapper
-{
-public:
-    MapWrapper();
-    ~MapWrapper();
-
-    // Puts a pointer to anything in the map and associates it with id. Note, id
-    // needs to be unique for all items in the map.
-    int Insert(int id, void* ptr);
-
-    // Removes item from map.
-    int Erase(MapItem* item);
-
-    // Finds item with associated with id and removes it from the map.
-    int Erase(int id);
-
-    // Returns the number of elements stored in the map.
-    int Size() const;
-
-    // Returns a pointer to the first MapItem in the map.
-    MapItem* First() const;
-
-    // Returns a pointer to the last MapItem in the map.
-    MapItem* Last() const;
-
-    // Returns a pointer to the MapItem stored after item in the map.
-    MapItem* Next(MapItem* item) const;
-
-    // Returns a pointer to the MapItem stored before item in the map.
-    MapItem* Previous(MapItem* item) const;
-
-    // Returns a pointer to the MapItem associated with id from the map.
-    MapItem* Find(int id) const;
-
-private:
-    std::map<int, MapItem*>    map_;
-    DISALLOW_COPY_AND_ASSIGN(MapWrapper);
-};
-} // namespace webrtc
-
-#endif  // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_MAP_WRAPPER_H_
diff --git a/src/system_wrappers/interface/rw_lock_wrapper.h b/src/system_wrappers/interface/rw_lock_wrapper.h
deleted file mode 100644
index f0842ac..0000000
--- a/src/system_wrappers/interface/rw_lock_wrapper.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
-
-// Note, Windows pre-Vista version of RW locks are not supported nativly. For
-// these OSs regular critical sections have been used to approximate RW lock
-// functionality and will therefore have worse performance.
-
-namespace webrtc {
-class RWLockWrapper
-{
-public:
-    static RWLockWrapper* CreateRWLock();
-    virtual ~RWLockWrapper();
-
-    virtual void AcquireLockExclusive() = 0;
-    virtual void ReleaseLockExclusive() = 0;
-
-    virtual void AcquireLockShared() = 0;
-    virtual void ReleaseLockShared() = 0;
-
-protected:
-    virtual int Init() = 0;
-};
-
-// RAII extensions of the RW lock. Prevents Acquire/Release missmatches and
-// provides more compact locking syntax.
-class ReadLockScoped
-{
-public:
-    ReadLockScoped(RWLockWrapper& rwLock)
-        :
-        _rwLock(rwLock)
-    {
-        _rwLock.AcquireLockShared();
-    }
-
-    ~ReadLockScoped()
-    {
-        _rwLock.ReleaseLockShared();
-    }
-
-private:
-    RWLockWrapper& _rwLock;
-};
-
-class WriteLockScoped
-{
-public:
-    WriteLockScoped(RWLockWrapper& rwLock)
-        :
-        _rwLock(rwLock)
-    {
-        _rwLock.AcquireLockExclusive();
-    }
-
-    ~WriteLockScoped()
-    {
-        _rwLock.ReleaseLockExclusive();
-    }
-
-private:
-    RWLockWrapper& _rwLock;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
diff --git a/src/system_wrappers/interface/sort.h b/src/system_wrappers/interface/sort.h
deleted file mode 100644
index fb25ecf..0000000
--- a/src/system_wrappers/interface/sort.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Generic unstable sorting routines.
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
-
-#include "typedefs.h"
-#include "common_types.h"
-
-namespace webrtc
-{
-    enum Type
-    {
-        TYPE_Word8,
-        TYPE_UWord8,
-        TYPE_Word16,
-        TYPE_UWord16,
-        TYPE_Word32,
-        TYPE_UWord32,
-        TYPE_Word64,
-        TYPE_UWord64,
-        TYPE_Float32,
-        TYPE_Float64
-    };
-    // Sorts intrinsic data types.
-    //
-    // data          [in/out] A pointer to an array of intrinsic type.
-    //               Upon return it will be sorted in ascending order.
-    // numOfElements The number of elements in the array.
-    // dataType      Enum corresponding to the type of the array.
-    //
-    // returns 0 on success, -1 on failure.
-    WebRtc_Word32 Sort(void* data, WebRtc_UWord32 numOfElements, Type dataType);
-
-    // Sorts arbitrary data types. This requires an array of intrinsically typed
-    // key values which will be used to sort the data array. There must be a
-    // one-to-one correspondence between data elements and key elements, with
-    // corresponding elements sharing the same position in their respective
-    // arrays.
-    //
-    // data          [in/out] A pointer to an array of arbitrary type.
-    //               Upon return it will be sorted in ascending order.
-    // key           [in] A pointer to an array of keys used to sort the
-    //               data array.
-    // numOfElements The number of elements in the arrays.
-    // sizeOfElement The size, in bytes, of the data array.
-    // keyType       Enum corresponding to the type of the key array.
-    //
-    // returns 0 on success, -1 on failure.
-    //
-    WebRtc_Word32 KeySort(void* data, void* key, WebRtc_UWord32 numOfElements,
-                          WebRtc_UWord32 sizeOfElement, Type keyType);
-}
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
diff --git a/src/system_wrappers/interface/thread_wrapper.h b/src/system_wrappers/interface/thread_wrapper.h
deleted file mode 100644
index eccf3c2..0000000
--- a/src/system_wrappers/interface/thread_wrapper.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// System independant wrapper for spawning threads
-// Note: the spawned thread will loop over the callback function until stopped.
-// Note: The callback function is expected to return every 2 seconds or more
-// often.
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
-
-namespace webrtc {
-// Object that will be passed by the spawned thread when it enters the callback
-// function.
-#define ThreadObj void*
-
-// Callback function that the spawned thread will enter once spawned
-typedef  bool(*ThreadRunFunction)(ThreadObj);
-
-enum ThreadPriority
-{
-    kLowPriority = 1,
-    kNormalPriority = 2,
-    kHighPriority = 3,
-    kHighestPriority = 4,
-    kRealtimePriority = 5
-};
-
-class ThreadWrapper
-{
-public:
-    enum {kThreadMaxNameLength = 64};
-
-    virtual ~ThreadWrapper() {};
-
-    // Factory method. Constructor disabled.
-    //
-    // func        Pointer to a, by user, specified callback function.
-    // obj         Object associated with the thread. Passed in the callback
-    //             function.
-    // prio        Thread priority. May require root/admin rights.
-    // threadName  NULL terminated thread name, will be visable in the Windows
-    //             debugger.
-    static ThreadWrapper* CreateThread(ThreadRunFunction func = 0,
-                                    ThreadObj obj= 0,
-                                    ThreadPriority prio = kNormalPriority,
-                                    const char* threadName = 0);
-
-    // Non blocking termination of the spawned thread. Note that it is not safe
-    // to delete this class until the spawned thread has been reclaimed.
-    virtual void SetNotAlive() = 0;
-
-    // Spawns the thread. This will start the triggering of the callback
-    // function.
-    virtual bool Start(unsigned int& id) = 0;
-
-    // Sets the threads CPU affinity. CPUs are listed 0 - (number of CPUs - 1).
-    // The numbers in processorNumbers specify which CPUs are allowed to run the
-    // thread. processorNumbers should not contain any duplicates and elements
-    // should be lower than (number of CPUs - 1). amountOfProcessors should be
-    // equal to the number of processors listed in processorNumbers
-    virtual bool SetAffinity(const int* /*processorNumbers*/,
-                             const unsigned int /*amountOfProcessors*/)
-                            {return false;}
-
-    // Stops the spawned thread and waits for it to be reclaimed with a timeout
-    // of two seconds. Will return false if the thread was not reclaimed.
-    // Multiple tries to Stop are allowed (e.g. to wait longer than 2 seconds).
-    // It's ok to call Stop() even if the spawned thread has been reclaimed.
-    virtual bool Stop() = 0;
-
-    // Stops the spawned thread dead in its tracks. Will likely result in a
-    // corrupt state. There should be an extremely good reason for even looking
-    // at this function. Can cause many problems deadlock being one of them.
-    virtual bool Shutdown() {return false;}
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
diff --git a/src/system_wrappers/interface/tick_util.h b/src/system_wrappers/interface/tick_util.h
deleted file mode 100644
index 4c28067..0000000
--- a/src/system_wrappers/interface/tick_util.h
+++ /dev/null
@@ -1,304 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// System independant wrapper for polling elapsed time in ms and us.
-// The implementation works in the tick domain which can be mapped over to the
-// time domain.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
-
-#if _WIN32
-#include <windows.h>
-#include <mmsystem.h>
-#elif WEBRTC_LINUX
-#include <ctime>
-#else
-#include <sys/time.h>
-#include <time.h>
-#endif
-
-#include "typedefs.h"
-
-namespace webrtc {
-class TickInterval;
-
-class TickTime
-{
-public:
-    // Current time in the tick domain.
-    static TickTime Now();
-
-    // Now in the time domain in ms.
-    static WebRtc_Word64 MillisecondTimestamp();
-
-    // Now in the time domain in us.
-    static WebRtc_Word64 MicrosecondTimestamp();
-
-    WebRtc_Word64 Ticks() const;
-
-    static WebRtc_Word64 MillisecondsToTicks(const WebRtc_Word64 ms);
-
-    static WebRtc_Word64 TicksToMilliseconds(const WebRtc_Word64 ticks);
-
-    // Returns a TickTime that is ticks later than the passed TickTime
-    friend TickTime operator+(const TickTime lhs, const WebRtc_Word64 ticks);
-    TickTime& operator+=(const WebRtc_Word64& rhs);
-
-
-    // Returns a TickInterval that is the difference in ticks beween rhs and lhs
-    friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);
-private:
-    WebRtc_Word64 _ticks;
-};
-
-class TickInterval
-{
-public:
-    TickInterval();
-
-    WebRtc_Word64 Milliseconds() const;
-    WebRtc_Word64 Microseconds() const;
-
-    // Returns the sum of two TickIntervals as a TickInterval
-    friend TickInterval operator+(const TickInterval& lhs,
-                                  const TickInterval& rhs);
-    TickInterval& operator-=(const TickInterval& rhs);
-
-    // Returns a TickInterval corresponding to rhs - lhs
-    friend TickInterval operator-(const TickInterval& lhs,
-                                  const TickInterval& rhs);
-    TickInterval& operator+=(const TickInterval& rhs);
-
-private:
-    TickInterval(WebRtc_Word64 interval);
-
-    friend class TickTime;
-    friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);
-
-private:
-    WebRtc_Word64 _interval;
-};
-
-inline TickInterval operator+(const TickInterval& lhs, const TickInterval& rhs)
-{
-    return TickInterval(lhs._interval + rhs._interval);
-}
-
-inline TickInterval operator-(const TickInterval& lhs, const TickInterval& rhs)
-{
-    return TickInterval(lhs._interval - rhs._interval);
-}
-
-inline TickInterval operator-(const TickTime& lhs,const TickTime& rhs)
-{
-    return TickInterval(lhs._ticks - rhs._ticks);
-}
-
-inline TickTime operator+(const TickTime lhs, const WebRtc_Word64 ticks)
-{
-    TickTime time = lhs;
-    time._ticks += ticks;
-    return time;
-}
-
-inline TickTime TickTime::Now()
-{
-    TickTime result;
-#if _WIN32
-    #ifdef USE_QUERY_PERFORMANCE_COUNTER
-        // QueryPerformanceCounter returns the value from the TSC which is
-        // incremented at the CPU frequency. The algorithm used requires
-        // the CPU frequency to be constant. Technology like speed stepping
-        // which has variable CPU frequency will therefore yield unpredictable,
-        // incorrect time estimations.
-        LARGE_INTEGER qpcnt;
-        QueryPerformanceCounter(&qpcnt);
-        result._ticks = qpcnt.QuadPart;
-    #else
-        static volatile LONG lastTimeGetTime = 0;
-        static volatile WebRtc_Word64 numWrapTimeGetTime = 0;
-        volatile LONG* lastTimeGetTimePtr = &lastTimeGetTime;
-        DWORD now = timeGetTime();
-        // Atomically update the last gotten time
-        DWORD old = InterlockedExchange(lastTimeGetTimePtr, now);
-        if(now < old)
-        {
-            // If now is earlier than old, there may have been a race between
-            // threads.
-            // 0x0fffffff ~3.1 days, the code will not take that long to execute
-            // so it must have been a wrap around.
-            if(old > 0xf0000000 && now < 0x0fffffff) 
-            {
-                numWrapTimeGetTime++;
-            }
-        }
-        result._ticks = now + (numWrapTimeGetTime<<32);
-    #endif
-#elif defined(WEBRTC_LINUX)
-    struct timespec ts;
-    #ifdef WEBRTC_CLOCK_TYPE_REALTIME
-        clock_gettime(CLOCK_REALTIME, &ts);
-    #else
-        clock_gettime(CLOCK_MONOTONIC, &ts);
-    #endif
-    result._ticks = 1000000000LL * static_cast<WebRtc_Word64>(ts.tv_sec) + static_cast<WebRtc_Word64>(ts.tv_nsec);
-#else
-    struct timeval tv;
-    gettimeofday(&tv, NULL);
-    result._ticks = 1000000LL * static_cast<WebRtc_Word64>(tv.tv_sec) + static_cast<WebRtc_Word64>(tv.tv_usec);
-#endif
-    return result;
-}
-
-inline WebRtc_Word64 TickTime::MillisecondTimestamp()
-{
-    TickTime now = TickTime::Now();
-#if _WIN32
-    #ifdef USE_QUERY_PERFORMANCE_COUNTER
-        LARGE_INTEGER qpfreq;
-        QueryPerformanceFrequency(&qpfreq);
-        return (now._ticks * 1000) / qpfreq.QuadPart;
-    #else
-        return now._ticks;
-    #endif
-#elif WEBRTC_LINUX
-    return now._ticks / 1000000LL;
-#else
-    return now._ticks / 1000LL;
-#endif
-}
-
-inline WebRtc_Word64 TickTime::MicrosecondTimestamp()
-{
-    TickTime now = TickTime::Now();
-
-#if _WIN32
-    #ifdef USE_QUERY_PERFORMANCE_COUNTER
-        LARGE_INTEGER qpfreq;
-        QueryPerformanceFrequency(&qpfreq);
-        return (now._ticks * 1000) / (qpfreq.QuadPart/1000);
-    #else
-        return now._ticks *1000LL;
-    #endif
-#elif WEBRTC_LINUX
-    return now._ticks / 1000LL;
-#else
-    return now._ticks;
-#endif
-}
-
-inline WebRtc_Word64 TickTime::Ticks() const
-{
-    return _ticks;
-}
-
-inline WebRtc_Word64 TickTime::MillisecondsToTicks(const WebRtc_Word64 ms)
-{
-#if _WIN32
-    #ifdef USE_QUERY_PERFORMANCE_COUNTER
-        LARGE_INTEGER qpfreq;
-        QueryPerformanceFrequency(&qpfreq);
-        return (qpfreq.QuadPart * ms) / 1000;
-    #else
-        return ms;
-    #endif
-#elif WEBRTC_LINUX
-    return ms * 1000000LL;
-#else
-    return ms * 1000LL;
-#endif
-}
-
-inline WebRtc_Word64 TickTime::TicksToMilliseconds(const WebRtc_Word64 ticks)
-{
-#if _WIN32
-    #ifdef USE_QUERY_PERFORMANCE_COUNTER
-        LARGE_INTEGER qpfreq;
-        QueryPerformanceFrequency(&qpfreq);
-        return (ticks * 1000) / qpfreq.QuadPart;
-    #else
-        return ticks;
-    #endif
-#elif WEBRTC_LINUX
-    return ticks / 1000000LL;
-#else
-    return ticks / 1000LL;
-#endif
-}
-
-inline TickTime& TickTime::operator+=(const WebRtc_Word64& ticks)
-{
-    _ticks += ticks;
-    return *this;
-}
-
-inline TickInterval::TickInterval() : _interval(0)
-{
-}
-
-inline TickInterval::TickInterval(const WebRtc_Word64 interval)
-    : _interval(interval)
-{
-}
-
-inline WebRtc_Word64 TickInterval::Milliseconds() const
-{
-#if _WIN32
-    #ifdef USE_QUERY_PERFORMANCE_COUNTER
-        LARGE_INTEGER qpfreq;
-        QueryPerformanceFrequency(&qpfreq);
-        return (_interval * 1000) / qpfreq.QuadPart;
-    #else
-	// _interval is in ms
-        return _interval;
-    #endif
-#elif WEBRTC_LINUX
-    // _interval is in ns
-    return _interval / 1000000;
-#else
-    // _interval is usecs
-    return _interval / 1000;
-#endif
-}
-
-inline WebRtc_Word64 TickInterval::Microseconds() const
-{
-#if _WIN32
-    #ifdef USE_QUERY_PERFORMANCE_COUNTER
-        LARGE_INTEGER qpfreq;
-        QueryPerformanceFrequency(&qpfreq);
-        return (_interval * 1000000) / qpfreq.QuadPart;
-    #else
-	// _interval is in ms
-        return _interval *1000LL;
-    #endif
-#elif WEBRTC_LINUX
-    // _interval is in ns
-    return _interval / 1000;
-#else
-    // _interval is usecs
-    return _interval;
-#endif
-}
-
-inline TickInterval& TickInterval::operator+=(const TickInterval& rhs)
-{
-    _interval += rhs._interval;
-    return *this;
-}
-
-inline TickInterval& TickInterval::operator-=(const TickInterval& rhs)
-{
-    _interval -= rhs._interval;
-    return *this;
-}
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
diff --git a/src/system_wrappers/interface/trace.h b/src/system_wrappers/interface/trace.h
deleted file mode 100644
index 0f7df4d..0000000
--- a/src/system_wrappers/interface/trace.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// System independant wrapper for logging runtime information to file.
-// Note: All log messages will be written to the same trace file.
-// Note: If to many messages are written to file there will be a build up of
-//       messages. Apply filtering to avoid that.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
-
-#include "common_types.h"
-#include "typedefs.h"
-
-#ifdef WEBRTC_NO_TRACE
-    #define WEBRTC_TRACE
-#else
-    // Ideally we would use __VA_ARGS__ but it's not supported by all compilers
-    // such as VS2003 (it's supported in VS2005). TODO (hellner) why
-    // would this be better than current implementation (not convinced)?
-    #define WEBRTC_TRACE Trace::Add
-#endif
-
-namespace webrtc {
-class Trace
-{
-public:
-
-    // Increments the reference count to the trace.
-    static void CreateTrace();
-    // Decrements the reference count to the trace.
-    static void ReturnTrace();
-    // Note: any instance that writes to the trace file should increment and
-    // decrement the reference count on construction and destruction
-    // respectively
-
-    // Specifies what type of messages should be written to the trace file. The
-    // filter parameter is a bitmask where each message type is enumerated by
-    // the TraceLevel enumerator. TODO (hellner) why is the
-    // TraceLevel enumerator not defined in this file?
-    static WebRtc_Word32 SetLevelFilter(const WebRtc_UWord32 filter);
-
-    // Returns what type of messages are written to the trace file.
-    static WebRtc_Word32 LevelFilter(WebRtc_UWord32& filter);
-
-    // Sets the file name. If addFileCounter is false the same file will be
-    // reused when it fills up. If it's true a new file with incremented name
-    // will be used.
-    static WebRtc_Word32 SetTraceFile(const WebRtc_Word8* fileName,
-                                      const bool addFileCounter = false);
-
-    // Returns the name of the file that the trace is currently writing to.
-    static WebRtc_Word32 TraceFile(WebRtc_Word8 fileName[1024]);
-
-    // Registers callback to receive trace messages. TODO (hellner)
-    // why not use OutStream instead? Why is TraceCallback not defined in this
-    // file
-    static WebRtc_Word32 SetTraceCallback(TraceCallback* callback);
-
-    // Adds a trace message for writing to file. The message is put in a queue
-    // for writing to file whenever possible for performance reasons. I.e. there
-    // is a crash it is possible that the last, vital logs are not logged yet.
-    // level is the the type of message to log. If that type of messages is
-    // filtered it will not be written to file. module is an identifier for what
-    // part of the code the message is comming.
-    // id is an identifier that should be unique for that set of classes that
-    // are associated (e.g. all instances owned by an engine).
-    // msg and the elipsis are the same as e.g. sprintf.
-    // TODO (hellner) Why is TraceModule not defined in this file?
-    static void Add(const TraceLevel level,
-                    const TraceModule module,
-                    const WebRtc_Word32 id,
-                    const char* msg, ...);
-
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
diff --git a/src/system_wrappers/source/Android.mk b/src/system_wrappers/source/Android.mk
deleted file mode 100644
index f8e406f..0000000
--- a/src/system_wrappers/source/Android.mk
+++ /dev/null
@@ -1,67 +0,0 @@
-# This file is generated by gyp; do not edit. This means you!
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc_system_wrappers
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := \
-    map.cc \
-    rw_lock_generic.cc \
-    sort.cc \
-    aligned_malloc.cc \
-    atomic32.cc \
-    condition_variable.cc \
-    cpu.cc \
-    cpu_features.cc \
-    critical_section.cc \
-    event.cc \
-    file_impl.cc \
-    list_no_stl.cc \
-    rw_lock.cc \
-    thread.cc \
-    trace_impl.cc \
-    condition_variable_linux.cc \
-    cpu_linux.cc \
-    critical_section_linux.cc \
-    event_linux.cc \
-    thread_linux.cc \
-    trace_linux.cc \
-    rw_lock_linux.cc 
-
-# Flags passed to both C and C++ files.
-MY_CFLAGS :=  
-MY_CFLAGS_C :=
-MY_DEFS := '-DNO_TCMALLOC' \
-    '-DNO_HEAPCHECKER' \
-    '-DWEBRTC_TARGET_PC' \
-    '-DWEBRTC_LINUX' \
-    '-DWEBRTC_CLOCK_TYPE_REALTIME' \
-    '-DWEBRTC_THREAD_RR' \
-    '-DWEBRTC_ANDROID' \
-    '-DANDROID' 
-LOCAL_CFLAGS := $(MY_CFLAGS_C) $(MY_CFLAGS) $(MY_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := $(LOCAL_PATH)/../.. \
-    $(LOCAL_PATH)/spreadsortlib \
-    $(LOCAL_PATH)/../interface 
-
-# Flags passed to only C++ (and not C) files.
-LOCAL_CPPFLAGS := 
-
-LOCAL_LDFLAGS :=
-
-LOCAL_STATIC_LIBRARIES :=
-
-LOCAL_SHARED_LIBRARIES := libcutils \
-    libdl \
-    libstlport
-LOCAL_ADDITIONAL_DEPENDENCIES :=
-
-include external/stlport/libstlport.mk
-include $(BUILD_STATIC_LIBRARY)
diff --git a/src/system_wrappers/source/aligned_malloc.cc b/src/system_wrappers/source/aligned_malloc.cc
deleted file mode 100644
index 6225753..0000000
--- a/src/system_wrappers/source/aligned_malloc.cc
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "aligned_malloc.h"
-
-#include <assert.h>
-#include <memory.h>
-
-#ifdef ANDROID
-#include <stdlib.h>
-#endif
-
-#if WEBRTC_MAC
-  #include <malloc/malloc.h>
-#else
-  #include <malloc.h>
-#endif
-
-#if _WIN32
-    #include <windows.h>
-#else
-    #include <stdint.h>
-#endif
-
-#include "typedefs.h"
-
-// Ok reference on memory alignment:
-// http://stackoverflow.com/questions/227897/solve-the-memory-alignment-in-c-interview-question-that-stumped-me
-
-namespace webrtc
-{
-// TODO (hellner) better to create just one memory block and
-//                           interpret the first sizeof(AlignedMemory) bytes as
-//                           an AlignedMemory struct.
-struct AlignedMemory
-{
-  void* alignedBuffer;
-  void* memoryPointer;
-};
-
-void* AlignedMalloc(size_t size, size_t alignment)
-{
-    if(alignment == 0)
-    {
-        // Don't allow alignment 0 since it's undefined.
-        return NULL;
-    }
-    // Make sure that the alignment is an integer power of two or fail.
-    if(alignment & (alignment - 1))
-    {
-        return NULL;
-    }
-
-    AlignedMemory* returnValue = new AlignedMemory();
-    if(returnValue == NULL)
-    {
-        return NULL;
-    }
-
-    // The memory is aligned towards the lowest address that so only
-    // alignment - 1 bytes needs to be allocated.
-    // A pointer to AlignedMemory must be stored so that it can be retreived for
-    // deletion, ergo the sizeof(uintptr_t).
-    returnValue->memoryPointer = malloc(size + sizeof(uintptr_t) +
-                                        alignment - 1);
-    if(returnValue->memoryPointer == NULL)
-    {
-        delete returnValue;
-        return NULL;
-    }
-
-    // Alligning after the sizeof(header) bytes will leave room for the header
-    // in the same memory block.
-    uintptr_t alignStartPos = (uintptr_t)returnValue->memoryPointer;
-    alignStartPos += sizeof(uintptr_t);
-
-    // The buffer should be aligned with 'alignment' bytes. The - 1 guarantees
-    // that we align towards the lowest address.
-    uintptr_t alignedPos = (alignStartPos + alignment - 1) & ~(alignment - 1);
-
-    // alignedPos is the address sought for.
-    returnValue->alignedBuffer = (void*)alignedPos;
-
-    // Store the address to the AlignedMemory struct in the header so that a
-    // it's possible to reclaim all memory.
-    uintptr_t headerPos = alignedPos;
-    headerPos -= sizeof(uintptr_t);
-    void* headerPtr = (void*) headerPos;
-    uintptr_t headerValue = (uintptr_t)returnValue;
-    memcpy(headerPtr,&headerValue,sizeof(uintptr_t));
-
-    return returnValue->alignedBuffer;
-}
-
-void AlignedFree(void* memBlock)
-{
-    if(memBlock == NULL)
-    {
-        return;
-    }
-    uintptr_t alignedPos = (uintptr_t)memBlock;
-    uintptr_t headerPos = alignedPos - sizeof(uintptr_t);
-
-    // Read out the address of the AlignedMemory struct from the header.
-    uintptr_t* headerPtr = (uintptr_t*)headerPos;
-    AlignedMemory* deleteMemory = (AlignedMemory*) *headerPtr;
-
-    if(deleteMemory->memoryPointer != NULL)
-    {
-        free(deleteMemory->memoryPointer);
-    }
-    delete deleteMemory;
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/atomic32.cc b/src/system_wrappers/source/atomic32.cc
deleted file mode 100644
index 3d6849e..0000000
--- a/src/system_wrappers/source/atomic32.cc
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "atomic32_wrapper.h"
-
-#if defined(_WIN32)
-    #include "atomic32_windows.h"
-#elif defined(WEBRTC_LINUX)
-    #include "atomic32_linux.h"
-#elif defined(WEBRTC_MAC)
-    #include "atomic32_mac.h"
-#else
-    #error unsupported os!
-#endif
-
-namespace webrtc {
-Atomic32Wrapper::Atomic32Wrapper(WebRtc_Word32 initialValue)
-    : _impl(*new Atomic32Impl(initialValue))
-{
-}
-
-Atomic32Wrapper::~Atomic32Wrapper()
-{
-    delete &_impl;
-}
-
-WebRtc_Word32 Atomic32Wrapper::operator++()
-{
-    return ++_impl;
-}
-
-WebRtc_Word32 Atomic32Wrapper::operator--()
-{
-    return --_impl;
-}
-
-// Read and write to properly aligned variables are atomic operations.
-// Ex reference (for Windows): http://msdn.microsoft.com/en-us/library/ms684122(v=VS.85).aspx
-// TODO (hellner) operator= and Atomic32Wrapper::Value() can be fully
-// implemented here.
-Atomic32Wrapper& Atomic32Wrapper::operator=(const Atomic32Wrapper& rhs)
-{
-    if(this == &rhs)
-    {
-        return *this;
-    }
-    _impl = rhs._impl;
-    return *this;
-}
-
-Atomic32Wrapper& Atomic32Wrapper::operator=(WebRtc_Word32 rhs)
-{
-    _impl = rhs;
-    return *this;
-}
-
-WebRtc_Word32 Atomic32Wrapper::operator+=(WebRtc_Word32 rhs)
-{
-    return _impl += rhs;
-}
-
-WebRtc_Word32 Atomic32Wrapper::operator-=(WebRtc_Word32 rhs)
-{
-    return _impl -= rhs;
-}
-
-bool Atomic32Wrapper::CompareExchange(WebRtc_Word32 newValue,
-                                      WebRtc_Word32 compareValue)
-{
-    return _impl.CompareExchange(newValue,compareValue);
-}
-
-WebRtc_Word32 Atomic32Wrapper::Value() const
-{
-    return _impl.Value();
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/atomic32_linux.h b/src/system_wrappers/source/atomic32_linux.h
deleted file mode 100644
index f9f5650..0000000
--- a/src/system_wrappers/source/atomic32_linux.h
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Atomic system independant 32-bit signed integer.
-// Linux implementation.
-// Note: Requires gcc 4.1.2 or later.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_LINUX_H_
-
-#include <inttypes.h>
-#include <malloc.h>
-
-#include "common_types.h"
-
-namespace webrtc {
-class Atomic32Impl
-{
-public:
-    inline Atomic32Impl(WebRtc_Word32 initialValue);
-    inline ~Atomic32Impl();
-
-    inline WebRtc_Word32 operator++();
-    inline WebRtc_Word32 operator--();
-
-    inline Atomic32Impl& operator=(const Atomic32Impl& rhs);
-    inline Atomic32Impl& operator=(WebRtc_Word32 rhs);
-    inline WebRtc_Word32 operator+=(WebRtc_Word32 rhs);
-    inline WebRtc_Word32 operator-=(WebRtc_Word32 rhs);
-
-    inline bool CompareExchange(WebRtc_Word32 newValue,
-                                WebRtc_Word32 compareValue);
-
-    inline WebRtc_Word32 Value() const;
-private:
-    void*        _ptrMemory;
-    // Volatile ensures full memory barriers.
-    volatile WebRtc_Word32* _value;
-};
-
-// TODO (hellner) use aligned_malloc instead of doing it manually.
-inline Atomic32Impl::Atomic32Impl(WebRtc_Word32 initialValue)
-    : _ptrMemory(NULL),
-      _value(NULL)
-{   // Align the memory associated with _value on a 32-bit boundary. This is a
-    // requirement for the used Linux APIs to be atomic.
-    // Keep _ptrMemory to be able to reclaim memory.
-    _ptrMemory = malloc(sizeof(WebRtc_Word32)*2);
-    _value = (WebRtc_Word32*) (((uintptr_t)_ptrMemory+3)&(~0x3));
-    *_value = initialValue;
-}
-
-inline Atomic32Impl::~Atomic32Impl()
-{
-    if(_ptrMemory != NULL)
-    {
-        free(_ptrMemory);
-    }
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator++()
-{
-    WebRtc_Word32 returnValue = __sync_fetch_and_add(_value,1);
-    returnValue++;
-    return returnValue;
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator--()
-{
-    WebRtc_Word32 returnValue = __sync_fetch_and_sub(_value,1);
-    returnValue--;
-    return returnValue;
-}
-
-inline Atomic32Impl& Atomic32Impl::operator=(const Atomic32Impl& rhs)
-{
-    *_value = *rhs._value;
-    return *this;
-}
-
-inline Atomic32Impl& Atomic32Impl::operator=(WebRtc_Word32 rhs)
-{
-    *_value = rhs;
-    return *this;
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator+=(WebRtc_Word32 rhs)
-{
-    WebRtc_Word32 returnValue = __sync_fetch_and_add(_value,rhs);
-    returnValue += rhs;
-    return returnValue;
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator-=(WebRtc_Word32 rhs)
-{
-    WebRtc_Word32 returnValue = __sync_fetch_and_sub(_value,rhs);
-    returnValue -= rhs;
-    return returnValue;
-}
-
-inline bool Atomic32Impl::CompareExchange(WebRtc_Word32 newValue,
-                                          WebRtc_Word32 compareValue)
-{
-    return __sync_bool_compare_and_swap(_value,compareValue,newValue);
-}
-
-inline WebRtc_Word32 Atomic32Impl::Value() const
-{
-    return *_value;
-}
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_LINUX_H_
diff --git a/src/system_wrappers/source/atomic32_mac.h b/src/system_wrappers/source/atomic32_mac.h
deleted file mode 100644
index bf8febc..0000000
--- a/src/system_wrappers/source/atomic32_mac.h
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// Atomic system independant 32-bit signed integer.
-// Mac implementation.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_MAC_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_MAC_H_
-
-#include <stdlib.h>
-#include <libkern/OSAtomic.h>
-
-#include "common_types.h"
-
-namespace webrtc {
-class Atomic32Impl
-{
-public:
-    inline Atomic32Impl(WebRtc_Word32 initialValue);
-    inline ~Atomic32Impl();
-
-    inline WebRtc_Word32 operator++();
-    inline WebRtc_Word32 operator--();
-
-    inline Atomic32Impl& operator=(const Atomic32Impl& rhs);
-    inline Atomic32Impl& operator=(WebRtc_Word32 rhs);
-    inline WebRtc_Word32 operator+=(WebRtc_Word32 rhs);
-    inline WebRtc_Word32 operator-=(WebRtc_Word32 rhs);
-
-    inline bool CompareExchange(WebRtc_Word32 newValue,
-                                WebRtc_Word32 compareValue);
-
-    inline WebRtc_Word32 Value() const;
-private:
-    void*        _ptrMemory;
-    // Volatile ensures full memory barriers.
-    volatile WebRtc_Word32* _value;
-};
-
-// TODO (hellner) use aligned_malloc instead of doing it manually.
-inline Atomic32Impl::Atomic32Impl(WebRtc_Word32 initialValue)
-    :
-    _ptrMemory(NULL),
-    _value(NULL)
-{   // Align the memory associated with _value on a 32-bit boundary. This is a
-    // requirement for the used Mac APIs to be atomic.
-    // Keep _ptrMemory to be able to reclaim memory.
-    _ptrMemory = malloc(sizeof(WebRtc_Word32)*2);
-    _value = (WebRtc_Word32*) (((uintptr_t)_ptrMemory+3)&(~0x3));
-    *_value = initialValue;
-}
-
-inline Atomic32Impl::~Atomic32Impl()
-{
-    if(_ptrMemory != NULL)
-    {
-        free(_ptrMemory);
-    }
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator++()
-{
-    return OSAtomicIncrement32Barrier(
-               reinterpret_cast<volatile int32_t*>(_value));
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator--()
-{
-    return OSAtomicDecrement32Barrier(
-               reinterpret_cast<volatile int32_t*>(_value));
-}
-
-inline Atomic32Impl& Atomic32Impl::operator=(const Atomic32Impl& rhs)
-{
-    *_value = *rhs._value;
-    return *this;
-}
-
-inline Atomic32Impl& Atomic32Impl::operator=(WebRtc_Word32 rhs)
-{
-    *_value = rhs;
-    return *this;
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator+=(WebRtc_Word32 rhs)
-{
-    return OSAtomicAdd32Barrier(rhs,
-                                reinterpret_cast<volatile int32_t*>(_value));
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator-=(WebRtc_Word32 rhs)
-{
-    return OSAtomicAdd32Barrier(-rhs,
-                                reinterpret_cast<volatile int32_t*>(_value));
-}
-
-inline bool Atomic32Impl::CompareExchange(WebRtc_Word32 newValue,
-                                          WebRtc_Word32 compareValue)
-{
-    return OSAtomicCompareAndSwap32Barrier(
-               compareValue,
-               newValue,
-               reinterpret_cast<volatile int32_t*>(_value));
-}
-
-inline WebRtc_Word32 Atomic32Impl::Value() const
-{
-    return *_value;
-}
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_MAC_H_
diff --git a/src/system_wrappers/source/condition_variable.cc b/src/system_wrappers/source/condition_variable.cc
deleted file mode 100644
index 7ca1b56..0000000
--- a/src/system_wrappers/source/condition_variable.cc
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#if defined(_WIN32)
-   #include <windows.h>
-   #include "condition_variable_wrapper.h"
-   #include "condition_variable_windows.h"
-#elif defined(WEBRTC_LINUX)
-   #include <pthread.h>
-   #include "condition_variable_wrapper.h"
-   #include "condition_variable_linux.h"
-#elif defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
-   #include <pthread.h>
-   #include "condition_variable_wrapper.h"
-   #include "condition_variable_linux.h"
- #endif
-
-namespace webrtc {
-ConditionVariableWrapper*
-ConditionVariableWrapper::CreateConditionVariable()
-{
-#if defined(_WIN32)
-    return new ConditionVariableWindows;
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
-    return ConditionVariableLinux::Create();
-#else
-    return NULL;
-#endif
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/condition_variable_linux.cc b/src/system_wrappers/source/condition_variable_linux.cc
deleted file mode 100644
index 778c2cf..0000000
--- a/src/system_wrappers/source/condition_variable_linux.cc
+++ /dev/null
@@ -1,151 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "condition_variable_linux.h"
-
-#if defined(WEBRTC_LINUX)
-#include <ctime>
-#else
-#include <sys/time.h>
-#endif
-
-#include <errno.h>
-
-#include "critical_section_linux.h"
-
-namespace webrtc {
-ConditionVariableWrapper* ConditionVariableLinux::Create()
-{
-    ConditionVariableLinux* ptr = new ConditionVariableLinux;
-    if (!ptr)
-    {
-        return NULL;
-    }
-
-    const int error = ptr->Construct();
-    if (error)
-    {
-        delete ptr;
-        return NULL;
-    }
-
-    return ptr;
-}
-
-ConditionVariableLinux::ConditionVariableLinux()
-{
-}
-
-int ConditionVariableLinux::Construct()
-{
-    int result = 0;
-#ifdef WEBRTC_CLOCK_TYPE_REALTIME
-    result = pthread_cond_init(&_cond, NULL);
-#else
-    pthread_condattr_t condAttr;
-    result = pthread_condattr_init(&condAttr);
-    if (result != 0)
-    {
-        return -1;
-    }
-    result = pthread_condattr_setclock(&condAttr, CLOCK_MONOTONIC);
-    if (result != 0)
-    {
-        return -1;
-    }
-    result = pthread_cond_init(&_cond, &condAttr);
-    if (result != 0)
-    {
-        return -1;
-    }
-    result = pthread_condattr_destroy(&condAttr);
-    if (result != 0)
-    {
-        return -1;
-    }
-#endif
-    return 0;
-}
-
-ConditionVariableLinux::~ConditionVariableLinux()
-{
-    pthread_cond_destroy(&_cond);
-}
-
-void ConditionVariableLinux::SleepCS(CriticalSectionWrapper& critSect)
-{
-    CriticalSectionLinux* cs = reinterpret_cast<CriticalSectionLinux*>(
-                                   &critSect);
-    pthread_cond_wait(&_cond, &cs->_mutex);
-}
-
-
-bool
-ConditionVariableLinux::SleepCS(
-    CriticalSectionWrapper& critSect,
-    unsigned long maxTimeInMS)
-{
-    const unsigned long INFINITE =  0xFFFFFFFF;
-
-    const int MILLISECONDS_PER_SECOND      = 1000;
-#ifndef WEBRTC_LINUX
-    const int MICROSECONDS_PER_MILLISECOND = 1000;
-#endif
-    const int NANOSECONDS_PER_SECOND       = 1000000000;
-    const int NANOSECONDS_PER_MILLISECOND  = 1000000;
-
-    CriticalSectionLinux* cs = reinterpret_cast<CriticalSectionLinux*>(
-                                   &critSect);
-
-    if (maxTimeInMS != INFINITE)
-    {
-        timespec ts;
-#ifndef WEBRTC_MAC
-#ifdef WEBRTC_CLOCK_TYPE_REALTIME
-        clock_gettime(CLOCK_REALTIME, &ts);
-#else
-        clock_gettime(CLOCK_MONOTONIC, &ts);
-#endif
-#else
-        struct timeval tv;
-        gettimeofday(&tv, 0);
-        ts.tv_sec  = tv.tv_sec;
-        ts.tv_nsec = tv.tv_usec * MICROSECONDS_PER_MILLISECOND;
-#endif
-
-        ts.tv_sec += maxTimeInMS / MILLISECONDS_PER_SECOND;
-        ts.tv_nsec += (maxTimeInMS - ((maxTimeInMS / MILLISECONDS_PER_SECOND)*
-                      MILLISECONDS_PER_SECOND)) * NANOSECONDS_PER_MILLISECOND;
-
-        if (ts.tv_nsec >= NANOSECONDS_PER_SECOND)
-        {
-            ts.tv_sec += ts.tv_nsec / NANOSECONDS_PER_SECOND;
-            ts.tv_nsec %= NANOSECONDS_PER_SECOND;
-        }
-        const int res = pthread_cond_timedwait(&_cond, &cs->_mutex, &ts);
-        return (res == ETIMEDOUT) ? false : true;
-    }
-    else
-    {
-        pthread_cond_wait(&_cond, &cs->_mutex);
-        return true;
-    }
-}
-
-void ConditionVariableLinux::Wake()
-{
-    pthread_cond_signal(&_cond);
-}
-
-void ConditionVariableLinux::WakeAll()
-{
-    pthread_cond_broadcast(&_cond);
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/condition_variable_linux.h b/src/system_wrappers/source/condition_variable_linux.h
deleted file mode 100644
index 0300c5b..0000000
--- a/src/system_wrappers/source/condition_variable_linux.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_LINUX_H_
-
-#include "condition_variable_wrapper.h"
-
-#include <pthread.h>
-
-namespace webrtc {
-class ConditionVariableLinux : public ConditionVariableWrapper
-{
-public:
-    static ConditionVariableWrapper* Create();
-    ~ConditionVariableLinux();
-
-    void SleepCS(CriticalSectionWrapper& critSect);
-    bool SleepCS(CriticalSectionWrapper& critSect, unsigned long maxTimeInMS);
-    void Wake();
-    void WakeAll();
-
-private:
-    ConditionVariableLinux();
-    int Construct();
-
-private:
-    pthread_cond_t _cond;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_LINUX_H_
diff --git a/src/system_wrappers/source/cpu.cc b/src/system_wrappers/source/cpu.cc
deleted file mode 100644
index 2285872..0000000
--- a/src/system_wrappers/source/cpu.cc
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "cpu_wrapper.h"
-
-#if defined(_WIN32)
-    #include <Windows.h>
-    #include "engine_configurations.h"
-    #include "cpu_windows.h"
-#elif defined(WEBRTC_MAC)
-    #include <sys/types.h>
-    #include <sys/sysctl.h>
-    #include "cpu_mac.h"
-#elif defined(WEBRTC_MAC_INTEL)
-    #include "cpu_mac.h"
-#elif defined(ANDROID)
-    // Not implemented yet, might be possible to use Linux implementation
-#else // defined(WEBRTC_LINUX)
-    #include <sys/sysinfo.h>
-    #include "cpu_linux.h"
-#endif
-
-#include "trace.h"
-
-namespace webrtc {
-WebRtc_UWord32 CpuWrapper::_numberOfCores = 0;
-
-WebRtc_UWord32 CpuWrapper::DetectNumberOfCores()
-{
-    if (!_numberOfCores)
-    {
-#if defined(_WIN32)
-        SYSTEM_INFO si;
-        GetSystemInfo(&si);
-        _numberOfCores = static_cast<WebRtc_UWord32>(si.dwNumberOfProcessors);
-        WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
-                     "Available number of cores:%d", _numberOfCores);
-
-#elif defined(WEBRTC_LINUX) && !defined(ANDROID)
-        _numberOfCores = get_nprocs();
-        WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
-                     "Available number of cores:%d", _numberOfCores);
-
-#elif (defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL))
-        int name[] = {CTL_HW, HW_AVAILCPU};
-        int ncpu;
-        size_t size = sizeof(ncpu);
-        if(0 == sysctl(name, 2, &ncpu, &size, NULL, 0))
-        {
-            _numberOfCores = static_cast<WebRtc_UWord32>(ncpu);
-            WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
-                         "Available number of cores:%d", _numberOfCores);
-    } else
-    {
-            WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
-                         "Failed to get number of cores");
-            _numberOfCores = 1;
-    }
-#else
-        WEBRTC_TRACE(kTraceWarning, kTraceUtility, -1,
-                     "No function to get number of cores");
-        _numberOfCores = 1;
-#endif
-    }
-    return _numberOfCores;
-}
-
-CpuWrapper* CpuWrapper::CreateCpu()
-{
-#if defined(_WIN32)
-   return new CpuWindows();
-#elif (defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL))
-    return new CpuWrapperMac();
-#elif defined(ANDROID)
-    return 0;
-#else
-    return new CpuLinux();
-#endif
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/cpu_features.cc b/src/system_wrappers/source/cpu_features.cc
deleted file mode 100644
index 850dc9b..0000000
--- a/src/system_wrappers/source/cpu_features.cc
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "cpu_features_wrapper.h"
-
-// No CPU feature is available => straight C path.
-int GetCPUInfoNoASM(CPUFeature feature) {
-  (void)feature;
-  return 0;
-}
-
-// Intrinsic for "cpuid".
-#if defined(__pic__) && defined(__i386__)
-static inline void cpuid(int cpu_info[4], int info_type) {
-  __asm__ volatile (
-    "mov %%ebx, %%edi\n"
-    "cpuid\n"
-    "xchg %%edi, %%ebx\n"
-    : "=a"(cpu_info[0]), "=D"(cpu_info[1]), "=c"(cpu_info[2]), "=d"(cpu_info[3])
-    : "a"(info_type));
-}
-#elif defined(__i386__) || defined(__x86_64__)
-static inline void cpuid(int cpu_info[4], int info_type) {
-  __asm__ volatile (
-    "cpuid\n"
-    : "=a"(cpu_info[0]), "=b"(cpu_info[1]), "=c"(cpu_info[2]), "=d"(cpu_info[3])
-    : "a"(info_type));
-}
-#endif
-
-#if defined(__i386__) || defined(__x86_64__)
-// Actual feature detection for x86.
-static int GetCPUInfo(CPUFeature feature) {
-  int cpu_info[4];
-  cpuid(cpu_info, 1);
-  if (feature == kSSE2) {
-    return 0 != (cpu_info[3] & 0x04000000);
-  }
-  if (feature == kSSE3) {
-    return 0 != (cpu_info[2] & 0x00000001);
-  }
-  return 0;
-}
-#else
-// Default to straight C for other platforms.
-static int GetCPUInfo(CPUFeature feature) {
-  (void)feature;
-  return 0;
-}
-#endif
-
-WebRtc_CPUInfo WebRtc_GetCPUInfo = GetCPUInfo;
-WebRtc_CPUInfo WebRtc_GetCPUInfoNoASM = GetCPUInfoNoASM;
diff --git a/src/system_wrappers/source/cpu_linux.cc b/src/system_wrappers/source/cpu_linux.cc
deleted file mode 100644
index eff9704..0000000
--- a/src/system_wrappers/source/cpu_linux.cc
+++ /dev/null
@@ -1,172 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "cpu_linux.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <unistd.h>
-
-namespace webrtc {
-CpuLinux::CpuLinux()
-{
-    m_oldBusyTime = 0;
-    m_oldIdleTime = 0;
-    m_numCores = 0;
-    m_numCores = GetNumCores();
-    m_oldBusyTimeMulti = new long long[m_numCores];
-    memset(m_oldBusyTimeMulti, 0, sizeof(long long) * m_numCores);
-    m_oldIdleTimeMulti = new long long[m_numCores];
-    memset(m_oldIdleTimeMulti, 0, sizeof(long long) * m_numCores);
-    m_idleArray = new long long[m_numCores];
-    memset(m_idleArray, 0, sizeof(long long) * m_numCores);
-    m_busyArray = new long long[m_numCores];
-    memset(m_busyArray, 0, sizeof(long long) * m_numCores);
-    m_resultArray = new WebRtc_UWord32[m_numCores];
-
-    GetData(m_oldBusyTime, m_oldIdleTime, m_busyArray, m_idleArray);
-}
-
-CpuLinux::~CpuLinux()
-{
-    delete [] m_oldBusyTimeMulti;
-    delete [] m_oldIdleTimeMulti;
-    delete [] m_idleArray;
-    delete [] m_busyArray;
-    delete [] m_resultArray;
-}
-
-WebRtc_Word32 CpuLinux::CpuUsage()
-{
-    WebRtc_UWord32 dummy = 0;
-    WebRtc_UWord32* dummyArray = NULL;
-    return CpuUsageMultiCore(dummy, dummyArray);
-}
-
-WebRtc_Word32 CpuLinux::CpuUsageMultiCore(WebRtc_UWord32& numCores,
-                                          WebRtc_UWord32*& coreArray)
-{
-    coreArray = m_resultArray;
-    numCores = m_numCores;
-    long long busy = 0;
-    long long idle = 0;
-    GetData(busy, idle, m_busyArray, m_idleArray);
-
-    long long deltaBusy = busy - m_oldBusyTime;
-    long long deltaIdle = idle - m_oldIdleTime;
-    m_oldBusyTime = busy;
-    m_oldIdleTime = idle;
-
-    int retVal = -1;
-    if (deltaBusy + deltaIdle == 0)
-    {
-        retVal = 0;
-    }
-    else
-    {
-        retVal = (int)(100 * (deltaBusy) / (deltaBusy + deltaIdle));
-    }
-
-    if (coreArray == NULL)
-    {
-      return retVal;
-    }
-
-    for (WebRtc_UWord32 i = 0; i < m_numCores; i++)
-    {
-        deltaBusy = m_busyArray[i] - m_oldBusyTimeMulti[i];
-        deltaIdle = m_idleArray[i] - m_oldIdleTimeMulti[i];
-        m_oldBusyTimeMulti[i] = m_busyArray[i];
-        m_oldIdleTimeMulti[i] = m_idleArray[i];
-        if(deltaBusy + deltaIdle == 0)
-        {
-            coreArray[i] = 0;
-        }
-        else
-        {
-            coreArray[i] = (int)(100 * (deltaBusy) / (deltaBusy+deltaIdle));
-        }
-    }
-    return retVal;
-}
-
-
-int CpuLinux::GetData(long long& busy, long long& idle, long long*& busyArray,
-                      long long*& idleArray)
-{
-    FILE* fp = fopen("/proc/stat", "r");
-    if (!fp)
-    {
-        return -1;
-    }
-
-    char line[100];
-    char* dummy = fgets(line, 100, fp);
-    char firstWord[100];
-    sscanf(line, "%s ", firstWord);
-    if(strncmp(firstWord, "cpu", 3)!=0)
-    {
-        return -1;
-    }
-    char sUser[100];
-    char sNice[100];
-    char sSystem[100];
-    char sIdle[100];
-    sscanf(line, "%s %s %s %s %s ", firstWord, sUser, sNice, sSystem, sIdle);
-    long long luser = atoll(sUser);
-    long long lnice = atoll(sNice);
-    long long lsystem = atoll(sSystem);
-    long long lidle = atoll (sIdle);
-
-    busy = luser + lnice + lsystem;
-    idle = lidle;
-    for (WebRtc_UWord32 i = 0; i < m_numCores; i++)
-    {
-        dummy = fgets(line, 100, fp);
-        sscanf(line, "%s %s %s %s %s ", firstWord, sUser, sNice, sSystem,
-               sIdle);
-        luser = atoll(sUser);
-        lnice = atoll(sNice);
-        lsystem = atoll(sSystem);
-        lidle = atoll (sIdle);
-        busyArray[i] = luser + lnice + lsystem;
-        idleArray[i] = lidle;
-    }
-    fclose(fp);
-    return 0;
-}
-
-int CpuLinux::GetNumCores()
-{
-    FILE* fp = fopen("/proc/stat", "r");
-    if (!fp)
-    {
-        return -1;
-    }
-    // Skip first line
-    char line[100];
-    char* dummy = fgets(line, 100, fp);
-    int numCores = -1;
-    char firstWord[100];
-    do
-    {
-        numCores++;
-        if (fgets(line, 100, fp))
-        {
-            sscanf(line, "%s ", firstWord);
-        } else {
-            break;
-        }
-    } while (strncmp(firstWord, "cpu", 3) == 0);
-    fclose(fp);
-    return numCores;
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/cpu_linux.h b/src/system_wrappers/source/cpu_linux.h
deleted file mode 100644
index 9b22e83..0000000
--- a/src/system_wrappers/source/cpu_linux.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_LINUX_H_
-
-#include "cpu_wrapper.h"
-
-namespace webrtc {
-class CpuLinux : public CpuWrapper
-{
-public:
-    CpuLinux();
-    virtual ~CpuLinux();
-
-    virtual WebRtc_Word32 CpuUsage();
-    virtual WebRtc_Word32 CpuUsage(WebRtc_Word8* /*pProcessName*/,
-                                   WebRtc_UWord32 /*length*/) {return 0;}
-    virtual WebRtc_Word32 CpuUsage(WebRtc_UWord32 /*dwProcessID*/) {return 0;}
-
-    virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& numCores,
-                                            WebRtc_UWord32*& array);
-
-    virtual void Reset() {return;}
-    virtual void Stop() {return;}
-private:
-    int GetData(long long& busy, long long& idle, long long*& busyArray,
-                long long*& idleArray);
-    int GetNumCores();
-
-    long long m_oldBusyTime;
-    long long m_oldIdleTime;
-
-    long long* m_oldBusyTimeMulti;
-    long long* m_oldIdleTimeMulti;
-
-    long long* m_idleArray;
-    long long* m_busyArray;
-    WebRtc_UWord32* m_resultArray;
-    WebRtc_UWord32  m_numCores;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_LINUX_H_
diff --git a/src/system_wrappers/source/cpu_mac.cc b/src/system_wrappers/source/cpu_mac.cc
deleted file mode 100644
index c2a11e1..0000000
--- a/src/system_wrappers/source/cpu_mac.cc
+++ /dev/null
@@ -1,132 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "cpu_mac.h"
-
-#include <iostream>
-#include <mach/mach.h>
-#include <mach/mach_error.h>
-
-#include "tick_util.h"
-
-namespace webrtc {
-CpuWrapperMac::CpuWrapperMac() : _cpuUsage(NULL)
-{
-    natural_t cpuCount;
-    processor_info_array_t infoArray;
-    mach_msg_type_number_t infoCount;
-
-    kern_return_t error = host_processor_info(mach_host_self(),
-                                              PROCESSOR_CPU_LOAD_INFO,
-                                              &cpuCount,
-                                              &infoArray,
-                                              &infoCount);
-    if (error)
-    {
-        return;
-    }
-
-    _cpuUsage = new WebRtc_UWord32[cpuCount];
-    _lastTickCount = new WebRtc_Word64[cpuCount];
-    _lastTime = TickTime::MillisecondTimestamp();
-
-    processor_cpu_load_info_data_t* cpuLoadInfo =
-        (processor_cpu_load_info_data_t*) infoArray;
-    for (unsigned int cpu= 0; cpu < cpuCount; cpu++)
-    {
-        WebRtc_Word64 ticks = 0;
-        for (int state = 0; state < 2; state++)
-        {
-            ticks += cpuLoadInfo[cpu].cpu_ticks[state];
-        }
-        _lastTickCount[cpu] = ticks;
-    }
-    vm_deallocate(mach_task_self(), (vm_address_t)infoArray, infoCount);
-}
-
-CpuWrapperMac::~CpuWrapperMac()
-{
-    delete _cpuUsage;
-    delete _lastTickCount;
-}
-
-WebRtc_Word32 CpuWrapperMac::CpuUsage()
-{
-    WebRtc_UWord32 numCores;
-    WebRtc_UWord32* array = NULL;
-    return CpuUsageMultiCore(numCores, array);
-}
-
-WebRtc_Word32
-CpuWrapperMac::CpuUsageMultiCore(WebRtc_UWord32& numCores,
-                                 WebRtc_UWord32*& array)
-{
-    natural_t cpuCount;
-    processor_info_array_t infoArray;
-    mach_msg_type_number_t infoCount;
-
-    // sanity check
-    if(_cpuUsage == NULL)
-    {
-        return -1;
-    }
-    WebRtc_Word64 now = TickTime::MillisecondTimestamp();
-    WebRtc_Word64 timeDiffMS = now - _lastTime;
-    // TODO(hellner) why block here? Why not just return the old
-    //                          value? Is this behavior consistent across all
-    //                          platforms?
-    // Make sure that at least 500 ms pass between calls.
-    if(timeDiffMS < 500)
-    {
-        usleep((500-timeDiffMS)*1000);
-        return CpuUsageMultiCore(numCores, array);
-    }
-    _lastTime = now;
-
-     kern_return_t error = host_processor_info(mach_host_self(),
-                                              PROCESSOR_CPU_LOAD_INFO,
-                                              &cpuCount,
-                                              &infoArray,
-                                              &infoCount);
-    if (error)
-    {
-        return -1;
-    }
-
-    processor_cpu_load_info_data_t* cpuLoadInfo =
-        (processor_cpu_load_info_data_t*) infoArray;
-
-    WebRtc_Word32 totalCpuUsage = 0;
-    for (unsigned int cpu = 0; cpu < cpuCount; cpu++)
-    {
-        WebRtc_Word64 ticks = 0;
-        for (int state = 0; state < 2; state++)
-        {
-            ticks += cpuLoadInfo[cpu].cpu_ticks[state];
-        }
-        if(timeDiffMS <= 0)
-        {
-            _cpuUsage[cpu] = 0;
-        }else {
-            _cpuUsage[cpu] = (WebRtc_UWord32)((1000 *
-                                              (ticks - _lastTickCount[cpu])) /
-                                              timeDiffMS);
-        }
-        _lastTickCount[cpu] = ticks;
-        totalCpuUsage += _cpuUsage[cpu];
-    }
-
-    vm_deallocate(mach_task_self(), (vm_address_t)infoArray, infoCount);
-
-    numCores = cpuCount;
-    array = _cpuUsage;
-    return totalCpuUsage/cpuCount;
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/cpu_mac.h b/src/system_wrappers/source/cpu_mac.h
deleted file mode 100644
index 04cd097..0000000
--- a/src/system_wrappers/source/cpu_mac.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_MAC_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_MAC_H_
-
-#include "cpu_wrapper.h"
-
-namespace webrtc {
-class CpuWrapperMac : public CpuWrapper
-{
-public:
-    CpuWrapperMac();
-    virtual ~CpuWrapperMac();
-
-    virtual WebRtc_Word32 CpuUsage();
-    virtual WebRtc_Word32 CpuUsage(WebRtc_Word8* /*pProcessName*/,
-                                   WebRtc_UWord32 /*length*/) {return -1;}
-    virtual WebRtc_Word32 CpuUsage(WebRtc_UWord32  /*dwProcessID*/) {return -1;}
-
-    // Note: this class will block the call and sleep if called too fast
-    // This function blocks the calling thread if the thread is calling it more
-    // often than every 500 ms.
-    virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& numCores,
-                                            WebRtc_UWord32*& array);
-
-    virtual void Reset() {}
-    virtual void Stop() {}
-
-private:
-    WebRtc_UWord32* _cpuUsage;
-    WebRtc_Word64*  _lastTickCount;
-    WebRtc_Word64   _lastTime;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_MAC_H_
diff --git a/src/system_wrappers/source/critical_section.cc b/src/system_wrappers/source/critical_section.cc
deleted file mode 100644
index 213c352..0000000
--- a/src/system_wrappers/source/critical_section.cc
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#if defined(_WIN32)
-    #include <windows.h>
-    #include "critical_section_windows.h"
-#else
-    #include "critical_section_linux.h"
-#endif
-
-namespace webrtc {
-CriticalSectionWrapper* CriticalSectionWrapper::CreateCriticalSection()
-{
-#ifdef _WIN32
-    return new CriticalSectionWindows();
-#else
-    return new CriticalSectionLinux();
-#endif
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/critical_section_linux.cc b/src/system_wrappers/source/critical_section_linux.cc
deleted file mode 100644
index 35e81ae..0000000
--- a/src/system_wrappers/source/critical_section_linux.cc
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "critical_section_linux.h"
-
-namespace webrtc {
-CriticalSectionLinux::CriticalSectionLinux()
-{
-    pthread_mutexattr_t attr;
-    pthread_mutexattr_init(&attr);
-    pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE);
-    pthread_mutex_init(&_mutex, &attr);
-}
-
-CriticalSectionLinux::~CriticalSectionLinux()
-{
-    pthread_mutex_destroy(&_mutex);
-}
-
-void
-CriticalSectionLinux::Enter()
-{
-    pthread_mutex_lock(&_mutex);
-}
-
-void
-CriticalSectionLinux::Leave()
-{
-    pthread_mutex_unlock(&_mutex);
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/critical_section_linux.h b/src/system_wrappers/source/critical_section_linux.h
deleted file mode 100644
index 5ada1cb..0000000
--- a/src/system_wrappers/source/critical_section_linux.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_LINUX_H_
-
-#include "critical_section_wrapper.h"
-
-#include <pthread.h>
-
-namespace webrtc {
-class CriticalSectionLinux : public CriticalSectionWrapper
-{
-public:
-    CriticalSectionLinux();
-
-    virtual ~CriticalSectionLinux();
-
-    virtual void Enter();
-    virtual void Leave();
-
-private:
-    pthread_mutex_t _mutex;
-    friend class ConditionVariableLinux;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_LINUX_H_
diff --git a/src/system_wrappers/source/event.cc b/src/system_wrappers/source/event.cc
deleted file mode 100644
index 384b961..0000000
--- a/src/system_wrappers/source/event.cc
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "event_wrapper.h"
-
-#if defined(_WIN32)
-    #include <windows.h>
-    #include "event_windows.h"
-#else
-    #include <pthread.h>
-    #include "event_linux.h"
-#endif
-
-namespace webrtc {
-EventWrapper* EventWrapper::Create()
-{
-#if defined(_WIN32)
-    return new EventWindows();
-#else
-    return EventLinux::Create();
-#endif
-}
-
-int EventWrapper::KeyPressed()
-{
-#if defined(_WIN32)
-    int keyDown = 0;
-    for(int key = 0x20; key < 0x90; key++)
-    {
-        short res = GetAsyncKeyState(key);
-        keyDown |= res%2;  // Get the LSB
-    }
-    if(keyDown)
-    {
-        return 1;
-    }
-    else
-    {
-        return 0;
-    }
-#else
-    return -1;
-#endif
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/event_linux.cc b/src/system_wrappers/source/event_linux.cc
deleted file mode 100644
index dddd31c..0000000
--- a/src/system_wrappers/source/event_linux.cc
+++ /dev/null
@@ -1,324 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "event_linux.h"
-
-#include <errno.h>
-#include <pthread.h>
-#include <signal.h>
-#include <stdio.h>
-#include <string.h>
-#include <sys/time.h>
-#include <unistd.h>
-
-namespace webrtc {
-const long int E6 = 1000000;
-const long int E9 = 1000 * E6;
-
-EventWrapper* EventLinux::Create()
-{
-    EventLinux* ptr = new EventLinux;
-    if (!ptr)
-    {
-        return NULL;
-    }
-
-    const int error = ptr->Construct();
-    if (error)
-    {
-        delete ptr;
-        return NULL;
-    }
-    return ptr;
-}
-
-
-EventLinux::EventLinux()
-    : _timerThread(0),
-      _timerEvent(0),
-      _periodic(false),
-      _time(0),
-      _count(0),
-      _state(kDown)
-{
-}
-
-int EventLinux::Construct()
-{
-    // Set start time to zero
-    memset(&_tCreate, 0, sizeof(_tCreate));
-
-    int result = pthread_mutex_init(&mutex, 0);
-    if (result != 0)
-    {
-        return -1;
-    }
-#ifdef WEBRTC_CLOCK_TYPE_REALTIME
-    result = pthread_cond_init(&cond, 0);
-    if (result != 0)
-    {
-        return -1;
-    }
-#else
-    pthread_condattr_t condAttr;
-    result = pthread_condattr_init(&condAttr);
-    if (result != 0)
-    {
-        return -1;
-    }
-    result = pthread_condattr_setclock(&condAttr, CLOCK_MONOTONIC);
-    if (result != 0)
-    {
-        return -1;
-    }
-    result = pthread_cond_init(&cond, &condAttr);
-    if (result != 0)
-    {
-        return -1;
-    }
-    result = pthread_condattr_destroy(&condAttr);
-    if (result != 0)
-    {
-        return -1;
-    }
-#endif
-    return 0;
-}
-
-EventLinux::~EventLinux()
-{
-    StopTimer();
-    pthread_cond_destroy(&cond);
-    pthread_mutex_destroy(&mutex);
-}
-
-bool EventLinux::Reset()
-{
-    if (0 != pthread_mutex_lock(&mutex))
-    {
-        return false;
-    }
-    _state = kDown;
-    pthread_mutex_unlock(&mutex);
-    return true;
-}
-
-bool EventLinux::Set()
-{
-    if (0 != pthread_mutex_lock(&mutex))
-    {
-        return false;
-    }
-    _state = kUp;
-     // Release all waiting threads
-    pthread_cond_broadcast(&cond);
-    pthread_mutex_unlock(&mutex);
-    return true;
-}
-
-EventTypeWrapper EventLinux::Wait(unsigned long timeout)
-{
-    int retVal = 0;
-    if (0 != pthread_mutex_lock(&mutex))
-    {
-        return kEventError;
-    }
-
-    if (kDown == _state)
-    {
-        if (WEBRTC_EVENT_INFINITE != timeout)
-        {
-            timespec tEnd;
-#ifndef WEBRTC_MAC
-#ifdef WEBRTC_CLOCK_TYPE_REALTIME
-            clock_gettime(CLOCK_REALTIME, &tEnd);
-#else
-            clock_gettime(CLOCK_MONOTONIC, &tEnd);
-#endif
-#else
-            timeval tVal;
-            struct timezone tZone;
-            tZone.tz_minuteswest = 0;
-            tZone.tz_dsttime = 0;
-            gettimeofday(&tVal,&tZone);
-            TIMEVAL_TO_TIMESPEC(&tVal,&tEnd);
-#endif
-            tEnd.tv_sec  += timeout / 1000;
-            tEnd.tv_nsec += (timeout - (timeout / 1000) * 1000) * E6;
-
-            if (tEnd.tv_nsec >= E9)
-            {
-                tEnd.tv_sec++;
-                tEnd.tv_nsec -= E9;
-            }
-            retVal = pthread_cond_timedwait(&cond, &mutex, &tEnd);
-        } else {
-            retVal = pthread_cond_wait(&cond, &mutex);
-        }
-    }
-
-    _state = kDown;
-    pthread_mutex_unlock(&mutex);
-
-    switch(retVal)
-    {
-    case 0:
-        return kEventSignaled;
-    case ETIMEDOUT:
-        return kEventTimeout;
-    default:
-        return kEventError;
-    }
-}
-
-EventTypeWrapper EventLinux::Wait(timespec& tPulse)
-{
-    int retVal = 0;
-    if (0 != pthread_mutex_lock(&mutex))
-    {
-        return kEventError;
-    }
-
-    if (kUp != _state)
-    {
-        retVal = pthread_cond_timedwait(&cond, &mutex, &tPulse);
-    }
-    _state = kDown;
-
-    pthread_mutex_unlock(&mutex);
-
-    switch(retVal)
-    {
-    case 0:
-        return kEventSignaled;
-    case ETIMEDOUT:
-        return kEventTimeout;
-    default:
-        return kEventError;
-    }
-}
-
-bool EventLinux::StartTimer(bool periodic, unsigned long time)
-{
-    if (_timerThread)
-    {
-        if(_periodic)
-        {
-            // Timer already started.
-            return false;
-        } else  {
-            // New one shot timer
-            _time = time;
-            _tCreate.tv_sec = 0;
-            _timerEvent->Set();
-            return true;
-        }
-    }
-
-    // Start the timer thread
-    _timerEvent = static_cast<EventLinux*>(EventWrapper::Create());
-    const char* threadName = "WebRtc_event_timer_thread";
-    _timerThread = ThreadWrapper::CreateThread(Run, this, kRealtimePriority,
-                                               threadName);
-    _periodic = periodic;
-    _time = time;
-    unsigned int id = 0;
-    if (_timerThread->Start(id))
-    {
-        return true;
-    }
-    return false;
-}
-
-bool EventLinux::Run(ThreadObj obj)
-{
-    return static_cast<EventLinux*>(obj)->Process();
-}
-
-bool EventLinux::Process()
-{
-    if (_tCreate.tv_sec == 0)
-    {
-#ifndef WEBRTC_MAC
-#ifdef WEBRTC_CLOCK_TYPE_REALTIME
-        clock_gettime(CLOCK_REALTIME, &_tCreate);
-#else
-        clock_gettime(CLOCK_MONOTONIC, &_tCreate);
-#endif
-#else
-        timeval tVal;
-        struct timezone tZone;
-        tZone.tz_minuteswest = 0;
-        tZone.tz_dsttime = 0;
-        gettimeofday(&tVal,&tZone);
-        TIMEVAL_TO_TIMESPEC(&tVal,&_tCreate);
-#endif
-        _count=0;
-    }
-
-    timespec tEnd;
-    unsigned long long time = _time * ++_count;
-    tEnd.tv_sec  = _tCreate.tv_sec + time/1000;
-    tEnd.tv_nsec = _tCreate.tv_nsec + (time - (time/1000)*1000)*E6;
-
-    if ( tEnd.tv_nsec >= E9 )
-    {
-        tEnd.tv_sec++;
-        tEnd.tv_nsec -= E9;
-    }
-
-    switch(_timerEvent->Wait(tEnd))
-    {
-    case kEventSignaled:
-        return true;
-    case kEventError:
-        return false;
-    case kEventTimeout:
-        break;
-    }
-    if(_periodic || _count==1)
-    {
-        Set();
-    }
-    return true;
-}
-
-bool EventLinux::StopTimer()
-{
-    if(_timerThread)
-    {
-        _timerThread->SetNotAlive();
-    }
-    if (_timerEvent)
-    {
-        _timerEvent->Set();
-    }
-    if (_timerThread)
-    {
-        if(!_timerThread->Stop())
-        {
-            return false;
-        }
-
-        delete _timerThread;
-        _timerThread = 0;
-    }
-    if (_timerEvent)
-    {
-        delete _timerEvent;
-        _timerEvent = 0;
-    }
-
-    // Set time to zero to force new reference time for the timer.
-    memset(&_tCreate, 0, sizeof(_tCreate));
-    _count=0;
-    return true;
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/event_linux.h b/src/system_wrappers/source/event_linux.h
deleted file mode 100644
index 17d193f..0000000
--- a/src/system_wrappers/source/event_linux.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_LINUX_H_
-
-#include "event_wrapper.h"
-
-#include <pthread.h>
-#include <time.h>
-
-#include "thread_wrapper.h"
-
-namespace webrtc {
-enum State
-{
-    kUp = 1,
-    kDown = 2
-};
-
-class EventLinux : public EventWrapper
-{
-public:
-    static EventWrapper* Create();
-
-    virtual ~EventLinux();
-
-    virtual EventTypeWrapper Wait(unsigned long maxTime);
-    virtual bool Set();
-    virtual bool Reset();
-
-    virtual bool StartTimer(bool periodic, unsigned long time);
-    virtual bool StopTimer();
-
-private:
-    EventLinux();
-    int Construct();
-
-    static bool Run(ThreadObj obj);
-    bool Process();
-    EventTypeWrapper Wait(timespec& tPulse);
-
-
-private:
-    pthread_cond_t  cond;
-    pthread_mutex_t mutex;
-
-    ThreadWrapper* _timerThread;
-    EventLinux*    _timerEvent;
-    timespec       _tCreate;
-
-    bool          _periodic;
-    unsigned long _time;  // In ms
-    unsigned long _count;
-    State         _state;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_LINUX_H_
diff --git a/src/system_wrappers/source/file_impl.cc b/src/system_wrappers/source/file_impl.cc
deleted file mode 100644
index 6046c2c..0000000
--- a/src/system_wrappers/source/file_impl.cc
+++ /dev/null
@@ -1,267 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "file_impl.h"
-
-#include <cassert>
-
-#ifdef _WIN32
-    #include <Windows.h>
-#else
-    #include <stdarg.h>
-    #include <string.h>
-#endif
-
-namespace webrtc {
-FileWrapper* FileWrapper::Create()
-{
-    return new FileWrapperImpl();
-}
-
-FileWrapperImpl::FileWrapperImpl()
-    : _id(NULL),
-      _open(false),
-      _looping(false),
-      _readOnly(false),
-      _text(false),
-      _maxSizeInBytes(-1),
-      _sizeInBytes(0)
-{
-    memset(_fileNameUTF8, 0, kMaxFileNameSize);
-}
-
-FileWrapperImpl::~FileWrapperImpl()
-{
-    if (_id != NULL)
-    {
-        fclose(_id);
-    }
-}
-
-WebRtc_Word32 FileWrapperImpl::CloseFile()
-{
-    if (_id != NULL)
-    {
-        fclose(_id);
-        _id = NULL;
-    }
-    memset(_fileNameUTF8, 0, kMaxFileNameSize);
-    _open = false;
-    return 0;
-}
-
-int FileWrapperImpl::Rewind()
-{
-    if(_looping || !_readOnly)
-    {
-        if (_id != NULL)
-        {
-            _sizeInBytes = 0;
-            return fseek(_id, 0, SEEK_SET);
-        }
-    }
-    return -1;
-}
-
-WebRtc_Word32 FileWrapperImpl::SetMaxFileSize(WebRtc_Word32 bytes)
-{
-    _maxSizeInBytes = bytes;
-    return 0;
-}
-
-WebRtc_Word32 FileWrapperImpl::Flush()
-{
-    if (_id != NULL)
-    {
-        return fflush(_id);
-    }
-    return -1;
-}
-
-WebRtc_Word32 FileWrapperImpl::FileName(WebRtc_Word8* fileNameUTF8,
-                                        WebRtc_UWord32 size) const
-{
-    WebRtc_Word32 len = static_cast<WebRtc_Word32>(strlen(_fileNameUTF8));
-    if(len > kMaxFileNameSize)
-    {
-        assert(false);
-        return -1;
-    }
-    if(len < 1)
-    {
-        return -1;
-    }
-    // Make sure to NULL terminate
-    if(size < (WebRtc_UWord32)len)
-    {
-        len = size - 1;
-    }
-    memcpy(fileNameUTF8, _fileNameUTF8, len);
-    fileNameUTF8[len] = 0;
-    return 0;
-}
-
-bool
-FileWrapperImpl::Open() const
-{
-    return _open;
-}
-
-WebRtc_Word32 FileWrapperImpl::OpenFile(const WebRtc_Word8 *fileNameUTF8,
-                                        const bool readOnly, const bool loop,
-                                        const bool text)
-{
-    WebRtc_Word32 length = (WebRtc_Word32)strlen(fileNameUTF8);
-    if (length > kMaxFileNameSize)
-    {
-        return -1;
-    }
-
-    _readOnly = readOnly;
-
-    FILE *tmpId = NULL;
-#if defined _WIN32
-    wchar_t wideFileName[kMaxFileNameSize];
-    wideFileName[0] = 0;
-
-    MultiByteToWideChar(CP_UTF8,
-                        0 /*UTF8 flag*/,
-                        fileNameUTF8,
-                        -1 /*Null terminated string*/,
-                        wideFileName,
-                        kMaxFileNameSize);
-    if(text)
-    {
-        if(readOnly)
-        {
-            tmpId = _wfopen(wideFileName, L"rt");
-        } else {
-            tmpId = _wfopen(wideFileName, L"wt");
-        }
-    } else {
-        if(readOnly)
-        {
-            tmpId = _wfopen(wideFileName, L"rb");
-        } else {
-            tmpId = _wfopen(wideFileName, L"wb");
-        }
-    }
-#else
-    if(text)
-    {
-        if(readOnly)
-        {
-            tmpId = fopen(fileNameUTF8, "rt");
-        } else {
-            tmpId = fopen(fileNameUTF8, "wt");
-        }
-    } else {
-        if(readOnly)
-        {
-            tmpId = fopen(fileNameUTF8, "rb");
-        } else {
-            tmpId = fopen(fileNameUTF8, "wb");
-        }
-    }
-#endif
-
-    if (tmpId != NULL)
-    {
-        // + 1 comes fro copying the NULL termination charachter too
-        memcpy(_fileNameUTF8, fileNameUTF8, length + 1);
-        if (_id != NULL)
-        {
-            fclose(_id);
-        }
-        _id = tmpId;
-        _looping = loop;
-        _open = true;
-        return 0;
-    }
-    return -1;
-}
-
-int FileWrapperImpl::Read(void *buf, int len)
-{
-    if(len < 0)
-    {
-        return 0;
-    }
-    if (_id != NULL)
-    {
-        WebRtc_Word32 res = static_cast<WebRtc_Word32>(fread(buf, 1, len, _id));
-        if (res != len)
-        {
-            if(!_looping)
-            {
-                CloseFile();
-            }
-        }
-        return res;
-    }
-    return -1;
-}
-
-WebRtc_Word32 FileWrapperImpl::WriteText(const WebRtc_Word8* text, ...)
-{
-    assert(!_readOnly);
-    assert(!_text);
-
-    if (_id == NULL)
-    {
-        return -1;
-    }
-
-    char tempBuff[kFileMaxTextMessageSize];
-    if (text)
-    {
-        va_list args;
-        va_start(args, text);
-#ifdef _WIN32
-        _vsnprintf(tempBuff, kFileMaxTextMessageSize-1, text, args);
-#else
-        vsnprintf(tempBuff, kFileMaxTextMessageSize-1, text, args);
-#endif
-        va_end(args);
-        WebRtc_Word32 nBytes;
-        nBytes = fprintf(_id, "%s", tempBuff);
-        if (nBytes > 0)
-        {
-            return 0;
-        }
-        CloseFile();
-    }
-    return -1;
-}
-
-bool FileWrapperImpl::Write(const void* buf, int len)
-{
-    assert(!_readOnly);
-    if (_id != NULL)
-    {
-        // Check if it's time to stop writing.
-        if ((_maxSizeInBytes != -1) &&
-             _sizeInBytes + len > (WebRtc_UWord32)_maxSizeInBytes)
-        {
-            Flush();
-            return false;
-        }
-
-        size_t nBytes = fwrite((WebRtc_UWord8*)buf, 1, len, _id);
-        if (nBytes > 0)
-        {
-            _sizeInBytes += static_cast<WebRtc_Word32>(nBytes);
-            return true;
-        }
-        CloseFile();
-    }
-    return false;
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/file_impl.h b/src/system_wrappers/source/file_impl.h
deleted file mode 100644
index cf6b734..0000000
--- a/src/system_wrappers/source/file_impl.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_FILE_IMPL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_FILE_IMPL_H_
-
-#include "file_wrapper.h"
-
-#include <stdio.h>
-
-namespace webrtc {
-class FileWrapperImpl : public FileWrapper
-{
-public:
-    FileWrapperImpl();
-    virtual ~FileWrapperImpl();
-
-    virtual WebRtc_Word32 FileName(WebRtc_Word8* fileNameUTF8,
-                                   WebRtc_UWord32 size) const;
-
-    virtual bool Open() const;
-
-    virtual WebRtc_Word32 OpenFile(const WebRtc_Word8* fileNameUTF8,
-                                 const bool readOnly,
-                                 const bool loop = false,
-                                 const bool text = false);
-
-    virtual WebRtc_Word32 CloseFile();
-    virtual WebRtc_Word32 SetMaxFileSize(WebRtc_Word32 bytes);
-    virtual WebRtc_Word32 Flush();
-
-    virtual int Read(void* buf, int len);
-    virtual bool Write(const void *buf, int len);
-    virtual int Rewind();
-
-    virtual WebRtc_Word32 WriteText(const WebRtc_Word8* text, ...);
-
-private:
-    FILE*          _id;
-    bool           _open;
-    bool           _looping;
-    bool           _readOnly;
-    bool           _text;
-    WebRtc_Word32  _maxSizeInBytes; // -1 indicates file size limitation is off
-    WebRtc_UWord32 _sizeInBytes;
-    WebRtc_Word8   _fileNameUTF8[kMaxFileNameSize];
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_FILE_IMPL_H_
diff --git a/src/system_wrappers/source/list_no_stl.cc b/src/system_wrappers/source/list_no_stl.cc
deleted file mode 100644
index d45f27b..0000000
--- a/src/system_wrappers/source/list_no_stl.cc
+++ /dev/null
@@ -1,289 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "list_wrapper.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-namespace webrtc {
-ListItem::ListItem(const void* item)
-    : next_(0),
-      prev_(0),
-      item_ptr_(item),
-      item_(0)
-{
-}
-
-ListItem::ListItem(const unsigned int item)
-    : next_(0),
-      prev_(0),
-      item_ptr_(0),
-      item_(item)
-{
-}
-
-ListItem::~ListItem()
-{
-}
-
-void* ListItem::GetItem() const
-{
-    return const_cast<void*>(item_ptr_);
-}
-
-unsigned int ListItem::GetUnsignedItem() const
-{
-    return item_;
-}
-
-ListWrapper::ListWrapper()
-    : critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
-      first_(0),
-      last_(0),
-      size_(0)
-{
-}
-
-ListWrapper::~ListWrapper()
-{
-    if (!Empty())
-    {
-        // TODO (hellner) I'm not sure this loggin is useful.
-        WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
-                   "Potential memory leak in ListWrapper");
-        // Remove all remaining list items.
-        while (Erase(First()) == 0)
-        {}
-    }
-    delete critical_section_;
-}
-
-bool ListWrapper::Empty() const
-{
-    return !first_ && !last_;
-}
-
-unsigned int ListWrapper::GetSize() const
-{
-    return size_;
-}
-
-int ListWrapper::PushBack(const void* ptr)
-{
-    ListItem* item = new ListItem(ptr);
-    CriticalSectionScoped lock(*critical_section_);
-    PushBackImpl(item);
-    return 0;
-}
-
-int ListWrapper::PushBack(const unsigned int item_id)
-{
-    ListItem* item = new ListItem(item_id);
-    CriticalSectionScoped lock(*critical_section_);
-    PushBackImpl(item);
-    return 0;
-}
-
-int ListWrapper::PushFront(const unsigned int item_id)
-{
-    ListItem* item = new ListItem(item_id);
-    CriticalSectionScoped lock(*critical_section_);
-    PushFrontImpl(item);
-    return 0;
-}
-
-int ListWrapper::PushFront(const void* ptr)
-{
-    ListItem* item = new ListItem(ptr);
-    CriticalSectionScoped lock(*critical_section_);
-    PushFrontImpl(item);
-    return 0;
-}
-
-int ListWrapper::PopFront()
-{
-    return Erase(first_);
-}
-
-int ListWrapper::PopBack()
-{
-    return Erase(last_);
-}
-
-ListItem* ListWrapper::First() const
-{
-    return first_;
-}
-
-ListItem* ListWrapper::Last() const
-{
-    return last_;
-}
-
-ListItem* ListWrapper::Next(ListItem* item) const
-{
-    if(!item)
-    {
-        return 0;
-    }
-    return item->next_;
-}
-
-ListItem* ListWrapper::Previous(ListItem* item) const
-{
-    if (!item)
-    {
-        return 0;
-    }
-    return item->prev_;
-}
-
-int ListWrapper::Insert(ListItem* existing_previous_item, ListItem* new_item)
-{
-    if (!new_item)
-    {
-        return -1;
-    }
-    // Allow existing_previous_item to be NULL if the list is empty.
-    // TODO (hellner) why allow this? Keep it as is for now to avoid
-    // breaking API contract.
-    if (!existing_previous_item && !Empty())
-    {
-        return -1;
-    }
-    CriticalSectionScoped lock(*critical_section_);
-    if (!existing_previous_item)
-    {
-        PushBackImpl(new_item);
-        return 0;
-    }
-    ListItem* next_item = existing_previous_item->next_;
-    new_item->next_ = existing_previous_item->next_;
-    new_item->prev_ = existing_previous_item;
-    existing_previous_item->next_ = new_item;
-    if (next_item)
-    {
-        next_item->prev_ = new_item;
-    }
-    else
-    {
-        last_ = new_item;
-    }
-    size_++;
-    return 0;
-}
-
-int ListWrapper::InsertBefore(ListItem* existing_next_item,
-                              ListItem* new_item)
-{
-    if (!new_item)
-    {
-        return -1;
-    }
-    // Allow existing_next_item to be NULL if the list is empty.
-    // Todo: why allow this? Keep it as is for now to avoid breaking API
-    // contract.
-    if (!existing_next_item && !Empty())
-    {
-        return -1;
-    }
-    CriticalSectionScoped lock(*critical_section_);
-    if (!existing_next_item)
-    {
-        PushBackImpl(new_item);
-        return 0;
-    }
-
-    ListItem* previous_item = existing_next_item->prev_;
-    new_item->next_ = existing_next_item;
-    new_item->prev_ = previous_item;
-    existing_next_item->prev_ = new_item;
-    if (previous_item)
-    {
-        previous_item->next_ = new_item;
-    }
-    else
-    {
-        first_ = new_item;
-    }
-    size_++;
-    return 0;
-}
-
-int ListWrapper::Erase(ListItem* item)
-{
-    if (!item)
-    {
-        return -1;
-    }
-    size_--;
-    ListItem* previous_item = item->prev_;
-    ListItem* next_item = item->next_;
-    if (!previous_item)
-    {
-        if(next_item)
-        {
-            next_item->prev_ = 0;
-        }
-        first_ = next_item;
-    }
-    else
-    {
-        previous_item->next_ = next_item;
-    }
-    if (!next_item)
-    {
-        if(previous_item)
-        {
-            previous_item->next_ = 0;
-        }
-        last_ = previous_item;
-    }
-    else
-    {
-        next_item->prev_ = previous_item;
-    }
-    delete item;
-    return 0;
-}
-
-void ListWrapper::PushBackImpl(ListItem* item)
-{
-    if (Empty())
-    {
-        first_ = item;
-        last_ = item;
-        size_++;
-        return;
-    }
-
-    item->prev_ = last_;
-    last_->next_ = item;
-    last_ = item;
-    size_++;
-}
-
-void ListWrapper::PushFrontImpl(ListItem* item)
-{
-    if (Empty())
-    {
-        first_ = item;
-        last_ = item;
-        size_++;
-        return;
-    }
-
-    item->next_ = first_;
-    first_->prev_ = item;
-    first_ = item;
-    size_++;
-}
-} //namespace webrtc
diff --git a/src/system_wrappers/source/list_no_stl.h b/src/system_wrappers/source/list_no_stl.h
deleted file mode 100644
index 26d844c..0000000
--- a/src/system_wrappers/source/list_no_stl.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_NO_STL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_NO_STL_H_
-
-#include "constructor_magic.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class ListNoStlItem
-{
-public:
-    ListNoStlItem(const void* ptr);
-    ListNoStlItem(const unsigned int item);
-    virtual ~ListNoStlItem();
-    void* GetItem() const;
-    unsigned int GetUnsignedItem() const;
-
-protected:
-    ListNoStlItem* next_;
-    ListNoStlItem* prev_;
-
-private:
-    friend class ListNoStl;
-
-    const void*         item_ptr_;
-    const unsigned int  item_;
-    DISALLOW_COPY_AND_ASSIGN(ListNoStlItem);
-};
-
-
-class ListNoStl
-{
-public:
-    ListNoStl();
-    virtual ~ListNoStl();
-
-    // ListWrapper functions
-    unsigned int GetSize() const;
-    int PushBack(const void* ptr);
-    int PushBack(const unsigned int item_id);
-    int PushFront(const void* ptr);
-    int PushFront(const unsigned int item_id);
-    int PopFront();
-    int PopBack();
-    bool Empty() const;
-    ListNoStlItem* First() const;
-    ListNoStlItem* Last() const;
-    ListNoStlItem* Next(ListNoStlItem* item) const;
-    ListNoStlItem* Previous(ListNoStlItem* item) const;
-    int Erase(ListNoStlItem* item);
-    int Insert(ListNoStlItem* existing_previous_item,
-               ListNoStlItem* new_item);
-
-    int InsertBefore(ListNoStlItem* existing_next_item,
-                     ListNoStlItem* new_item);
-
-private:
-    void PushBack(ListNoStlItem* item);
-    void PushFront(ListNoStlItem* item);
-
-    CriticalSectionWrapper* critical_section_;
-    ListNoStlItem* first_;
-    ListNoStlItem* last_;
-    unsigned int size_;
-    DISALLOW_COPY_AND_ASSIGN(ListNoStl);
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_NO_STL_H_
diff --git a/src/system_wrappers/source/list_stl.cc b/src/system_wrappers/source/list_stl.cc
deleted file mode 100644
index dcc63c3..0000000
--- a/src/system_wrappers/source/list_stl.cc
+++ /dev/null
@@ -1,244 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "list_wrapper.h"
-
-#include "trace.h"
-
-namespace webrtc {
-ListItem::ListItem(const void* item)
-    : this_iter_(),
-      item_ptr_(item),
-      item_(0)
-{
-}
-
-ListItem::ListItem(const unsigned int item)
-    : this_iter_(),
-      item_ptr_(0),
-      item_(item)
-{
-}
-
-ListItem::~ListItem()
-{
-}
-
-void* ListItem::GetItem() const
-{
-    return const_cast<void*>(item_ptr_);
-}
-
-unsigned int ListItem::GetUnsignedItem() const
-{
-    return item_;
-}
-
-ListWrapper::ListWrapper() : list_()
-{
-}
-
-ListWrapper::~ListWrapper()
-{
-    if (!Empty())
-    {
-        // TODO (hellner) I'm not sure this loggin is useful.
-        WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
-                   "Potential memory leak in ListWrapper");
-        // Remove all remaining list items.
-        while (Erase(First()) == 0)
-        {}
-    }
-}
-
-bool ListWrapper::Empty() const
-{
-    return list_.empty();
-}
-
-unsigned int ListWrapper::GetSize() const
-{
-    return list_.size();
-}
-
-int ListWrapper::PushBack(const void* ptr)
-{
-    ListItem* item = new ListItem(ptr);
-    list_.push_back(item);
-    return 0;
-}
-
-int ListWrapper::PushBack(const unsigned int item_id)
-{
-    ListItem* item = new ListItem(item_id);
-    list_.push_back(item);
-    return 0;
-}
-
-int ListWrapper::PushFront(const unsigned int item_id)
-{
-    ListItem* item = new ListItem(item_id);
-    list_.push_front(item);
-    return 0;
-}
-
-int ListWrapper::PushFront(const void* ptr)
-{
-    ListItem* item = new ListItem(ptr);
-    list_.push_front(item);
-    return 0;
-}
-
-int ListWrapper::PopFront()
-{
-    if(list_.empty())
-    {
-        return -1;
-    }
-    list_.pop_front();
-    return 0;
-}
-
-int ListWrapper::PopBack()
-{
-    if(list_.empty())
-    {
-        return -1;
-    }
-    list_.pop_back();
-    return 0;
-}
-
-ListItem* ListWrapper::First() const
-{
-    if(list_.empty())
-    {
-        return NULL;
-    }
-    std::list<ListItem*>::iterator item_iter = list_.begin();
-    ListItem* return_item = (*item_iter);
-    return_item->this_iter_ = item_iter;
-    return return_item;
-}
-
-ListItem* ListWrapper::Last() const
-{
-    if(list_.empty())
-    {
-        return NULL;
-    }
-    // std::list::end() addresses the last item + 1. Decrement so that the
-    // actual last is accessed.
-    std::list<ListItem*>::iterator item_iter = list_.end();
-    --item_iter;
-    ListItem* return_item = (*item_iter);
-    return_item->this_iter_ = item_iter;
-    return return_item;
-}
-
-ListItem* ListWrapper::Next(ListItem* item) const
-{
-    if(item == NULL)
-    {
-        return NULL;
-    }
-    std::list<ListItem*>::iterator item_iter = item->this_iter_;
-    ++item_iter;
-    if (item_iter == list_.end())
-    {
-        return NULL;
-    }
-    ListItem* return_item = (*item_iter);
-    return_item->this_iter_ = item_iter;
-    return return_item;
-}
-
-ListItem* ListWrapper::Previous(ListItem* item) const
-{
-    if(item == NULL)
-    {
-        return NULL;
-    }
-    std::list<ListItem*>::iterator item_iter = item->this_iter_;
-    if (item_iter == list_.begin())
-    {
-      return NULL;
-    }
-    --item_iter;
-    ListItem* return_item = (*item_iter);
-    return_item->this_iter_ = item_iter;
-    return return_item;
-}
-
-int ListWrapper::Insert(ListItem* existing_previous_item,
-                        ListItem* new_item)
-{
-    // Allow existingPreviousItem to be NULL if the list is empty.
-    // TODO (hellner) why allow this? Keep it as is for now to avoid
-    // breaking API contract.
-    if (!existing_previous_item && !Empty())
-    {
-        return -1;
-    }
-
-    if (!new_item)
-    {
-        return -1;
-    }
-
-    std::list<ListItem*>::iterator insert_location = list_.begin();
-    if (!Empty())
-    {
-        insert_location = existing_previous_item->this_iter_;
-        if(insert_location != list_.end())
-        {
-            ++insert_location;
-        }
-    }
-
-    list_.insert(insert_location,new_item);
-    return 0;
-}
-
-int ListWrapper::InsertBefore(ListItem* existing_next_item,
-                           ListItem* new_item)
-{
-    // Allow existing_next_item to be NULL if the list is empty.
-    // Todo: why allow this? Keep it as is for now to avoid breaking API
-    // contract.
-    if (!existing_next_item && !Empty())
-    {
-        return -1;
-    }
-    if (!new_item)
-    {
-        return -1;
-    }
-
-    std::list<ListItem*>::iterator insert_location = list_.begin();
-    if (!Empty())
-    {
-        insert_location = existing_next_item->this_iter_;
-    }
-
-    list_.insert(insert_location,new_item);
-    return 0;
-}
-
-int ListWrapper::Erase(ListItem* item)
-{
-    if(item == NULL)
-    {
-        return -1;
-    }
-    list_.erase(item->this_iter_);
-    return 0;
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/list_stl.h b/src/system_wrappers/source/list_stl.h
deleted file mode 100644
index b83a664..0000000
--- a/src/system_wrappers/source/list_stl.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_STL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_STL_H_
-
-#include <list>
-
-#include "constructor_magic.h"
-
-namespace webrtc {
-class ListItem
-{
-friend class ListWrapper;
-
-public:
-    ListItem(const void* ptr);
-    ListItem(const unsigned int item);
-    virtual ~ListItem();
-    void* GetItem() const;
-    unsigned int GetUnsignedItem() const;
-
-private:
-    mutable std::list<ListItem*>::iterator this_iter_;
-    const void*         item_ptr_;
-    const unsigned int  item_;
-    DISALLOW_COPY_AND_ASSIGN(ListItem);
-};
-
-class ListWrapper
-{
-public:
-    ListWrapper();
-    ~ListWrapper();
-
-    // ListWrapper functions
-    unsigned int GetSize() const;
-    int PushBack(const void* ptr);
-    int PushBack(const unsigned int item_id);
-    int PushFront(const void* ptr);
-    int PushFront(const unsigned int item_id);
-    int PopFront();
-    int PopBack();
-    bool Empty() const;
-    ListItem* First() const;
-    ListItem* Last() const;
-    ListItem* Next(ListItem* item) const;
-    ListItem* Previous(ListItem* item) const;
-    int Erase(ListItem* item);
-    int Insert(ListItem* existing_previous_item, ListItem* new_item);
-    int InsertBefore(ListItem* existing_next_item, ListItem* new_item);
-
-private:
-    mutable std::list<ListItem*> list_;
-    DISALLOW_COPY_AND_ASSIGN(ListWrapper);
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_STL_H_
diff --git a/src/system_wrappers/source/list_unittest.cc b/src/system_wrappers/source/list_unittest.cc
deleted file mode 100644
index 3f3c88f..0000000
--- a/src/system_wrappers/source/list_unittest.cc
+++ /dev/null
@@ -1,475 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gtest/gtest.h"
-
-#include "list_wrapper.h"
-
-using ::webrtc::ListWrapper;
-using ::webrtc::ListItem;
-
-// Note: kNumberOfElements needs to be even.
-const unsigned int kNumberOfElements = 10;
-
-// An opaque implementation of dynamic or statically allocated unsigned ints.
-// This class makes it possible to use the exact same code for testing of both
-// the dynamic and static implementation of ListWrapper.
-// Clarification: ListWrapper has two versions of PushBack(..). It takes an
-// unsigned integer or a void pointer. The integer implementation takes care
-// of memory management. The void pointer version expect the caller to manage
-// the memory associated with the void pointer.
-// This class works like the integer version but can be implemented on top of
-// either the integer version or void pointer version of ListWrapper.
-// Note: the non-virtual fuctions behave the same for both versions.
-class ListWrapperSimple {
-public:
-    static ListWrapperSimple* Create(bool static_allocation);
-    virtual ~ListWrapperSimple() {}
-
-    // These three functions should be used for manipulating ListItems so that
-    // they are the type corresponding to the underlying implementation.
-    virtual unsigned int GetUnsignedItem(
-        const ListItem* item) const = 0;
-    virtual ListItem* CreateListItem(unsigned int item_id) = 0;
-    virtual bool DestroyListItem(ListItem* item) = 0;
-
-    unsigned int GetSize() const {
-        return list_.GetSize();
-    }
-    virtual int PushBack(const unsigned int item_id) = 0;
-    virtual int PushFront(const unsigned int item_id) = 0;
-    virtual int PopFront() = 0;
-    virtual int PopBack() = 0;
-    bool Empty() const {
-        return list_.Empty();
-    }
-    ListItem* First() const {
-        return list_.First();
-    }
-    ListItem* Last() const {
-        return list_.Last();
-    }
-    ListItem* Next(ListItem* item) const {
-        return list_.Next(item);
-    }
-    ListItem* Previous(ListItem* item) const {
-        return list_.Previous(item);
-    }
-    virtual int Erase(ListItem* item) = 0;
-    int Insert(ListItem* existing_previous_item,
-                       ListItem* new_item) {
-        return list_.Insert(existing_previous_item, new_item);
-    }
-
-    int InsertBefore(ListItem* existing_next_item,
-                             ListItem* new_item) {
-        return list_.InsertBefore(existing_next_item, new_item);
-    }
-protected:
-    ListWrapperSimple() {}
-
-    ListWrapper list_;
-};
-
-class ListWrapperStatic : public ListWrapperSimple {
-public:
-    ListWrapperStatic() {}
-    virtual ~ListWrapperStatic() {}
-
-    virtual unsigned int GetUnsignedItem(const ListItem* item) const {
-        return item->GetUnsignedItem();
-    }
-    virtual ListItem* CreateListItem(unsigned int item_id) {
-        return new ListItem(item_id);
-    }
-    virtual bool DestroyListItem(ListItem* item) {
-        if (item == NULL) {
-            return false;
-        }
-        delete item;
-        return true;
-    }
-    virtual int PushBack(const unsigned int item_id) {
-        return list_.PushBack(item_id);
-    }
-    virtual int PushFront(const unsigned int item_id) {
-        return list_.PushFront(item_id);
-    }
-    virtual int PopFront() {
-        return list_.PopFront();
-    }
-    virtual int PopBack() {
-        return list_.PopBack();
-    }
-    virtual int Erase(ListItem* item) {
-        return list_.Erase(item);
-    }
-};
-
-class ListWrapperDynamic : public ListWrapperSimple {
-public:
-    ListWrapperDynamic() {}
-    virtual ~ListWrapperDynamic() {}
-
-    virtual unsigned int GetUnsignedItem(const ListItem* item) const {
-        const unsigned int* return_value_pointer =
-            reinterpret_cast<unsigned int*> (item->GetItem());
-        if (return_value_pointer == NULL) {
-            return -1;
-        }
-        return *return_value_pointer;
-    }
-    virtual ListItem* CreateListItem(unsigned int item_id) {
-        unsigned int* item_id_pointer = new unsigned int;
-        if (item_id_pointer == NULL) {
-            return NULL;
-        }
-        *item_id_pointer = item_id;
-        ListItem* return_value = new ListItem(
-            reinterpret_cast<void*>(item_id_pointer));
-        if (return_value == NULL) {
-            delete item_id_pointer;
-            return NULL;
-        }
-        return return_value;
-    }
-    virtual bool DestroyListItem(ListItem* item) {
-        if (item == NULL) {
-            return false;
-        }
-        bool return_value = false;
-        unsigned int* item_id_ptr = reinterpret_cast<unsigned int*>(
-            item->GetItem());
-        if (item_id_ptr != NULL) {
-            return_value = true;
-            delete item_id_ptr;
-        }
-        delete item;
-        return return_value;
-    }
-    virtual int PushBack(const unsigned int item_id) {
-        unsigned int* item_id_ptr = new unsigned int;
-        if (item_id_ptr == NULL) {
-            return -1;
-        }
-        *item_id_ptr = item_id;
-        const int return_value = list_.PushBack(
-            reinterpret_cast<void*>(item_id_ptr));
-        if (return_value != 0) {
-            delete item_id_ptr;
-        }
-        return return_value;
-    }
-    virtual int PushFront(const unsigned int item_id) {
-        unsigned int* item_id_ptr = new unsigned int;
-        if (item_id_ptr == NULL) {
-            return -1;
-        }
-        *item_id_ptr = item_id;
-        const int return_value = list_.PushFront(
-            reinterpret_cast<void*>(item_id_ptr));
-        if (return_value != 0) {
-            delete item_id_ptr;
-        }
-        return return_value;
-    }
-    virtual int PopFront() {
-        return Erase(list_.First());
-    }
-    virtual int PopBack() {
-        return Erase(list_.Last());
-    }
-    virtual int Erase(ListItem* item) {
-        if (item == NULL) {
-            return -1;
-        }
-        unsigned int* item_id_ptr = reinterpret_cast<unsigned int*> (
-            item->GetItem());
-        int return_value = -1;
-        if (item_id_ptr != NULL) {
-            delete item_id_ptr;
-            return_value = 0;
-        }
-        if (list_.Erase(item) != 0) {
-            return -1;
-        }
-        return return_value;
-    }
-};
-
-ListWrapperSimple* ListWrapperSimple::Create(bool static_allocation) {
-    if(static_allocation)
-    {
-        return new ListWrapperStatic();
-    }
-    return new ListWrapperDynamic();
-}
-
-void ClearList(ListWrapperSimple* list) {
-    if (list == NULL)
-    {
-        return;
-    }
-    while (list->Erase(list->First()) == 0) {
-    }
-}
-
-ListWrapperSimple* CreateAscendingList(bool static_allocation) {
-    ListWrapperSimple* return_value = ListWrapperSimple::Create(
-        static_allocation);
-    if (return_value == NULL) {
-        return NULL;
-    }
-    for (unsigned int i = 0; i < kNumberOfElements; ++i) {
-        if (return_value->PushBack(i) == -1) {
-            ClearList(return_value);
-            delete return_value;
-            return NULL;
-        }
-    }
-    return return_value;
-}
-
-ListWrapperSimple* CreateDecendingList(bool static_allocation) {
-    ListWrapperSimple* return_value = ListWrapperSimple::Create(
-        static_allocation);
-    if (return_value == NULL) {
-        return NULL;
-    }
-    for (unsigned int i = 0; i < kNumberOfElements; ++i) {
-        if (return_value->PushBack(kNumberOfElements - i - 1) == -1) {
-            ClearList(return_value);
-            delete return_value;
-            return NULL;
-        }
-    }
-    return return_value;
-}
-
-// [0,kNumberOfElements - 1,1,kNumberOfElements - 2,...] (this is why
-// kNumberOfElements need to be even)
-ListWrapperSimple* CreateInterleavedList(bool static_allocation) {
-    ListWrapperSimple* return_value = ListWrapperSimple::Create(
-        static_allocation);
-    if (return_value == NULL) {
-        return NULL;
-    }
-    unsigned int uneven_count = 0;
-    unsigned int even_count = 0;
-    for (unsigned int i = 0; i < kNumberOfElements; i++) {
-        unsigned int push_value = 0;
-        if ((i % 2) == 0) {
-            push_value = even_count;
-            even_count++;
-        } else {
-            push_value = kNumberOfElements - uneven_count - 1;
-            uneven_count++;
-        }
-        if (return_value->PushBack(push_value) == -1) {
-            ClearList(return_value);
-            delete return_value;
-            return NULL;
-        }
-    }
-    return return_value;
-}
-
-void PrintList(const ListWrapperSimple* list) {
-    ListItem* list_item = list->First();
-    printf("[");
-    while (list_item != NULL)
-    {
-        printf("%3u", list->GetUnsignedItem(list_item));
-        list_item = list->Next(list_item);
-    }
-    printf("]\n");
-}
-
-bool CompareLists(const ListWrapperSimple* lhs, const ListWrapperSimple* rhs) {
-    const unsigned int list_size = lhs->GetSize();
-    if (lhs->GetSize() != rhs->GetSize()) {
-        return false;
-    }
-    if (lhs->Empty()) {
-        return rhs->Empty();
-    }
-    unsigned int i = 0;
-    ListItem* lhs_item = lhs->First();
-    ListItem* rhs_item = rhs->First();
-    while (i < list_size) {
-        if (lhs_item == NULL) {
-            return false;
-        }
-        if (rhs_item == NULL) {
-            return false;
-        }
-        if (lhs->GetUnsignedItem(lhs_item) != rhs->GetUnsignedItem(rhs_item)) {
-            return false;
-        }
-        i++;
-        lhs_item = lhs->Next(lhs_item);
-        rhs_item = rhs->Next(rhs_item);
-    }
-    return true;
-}
-
-TEST(ListWrapperTest,ReverseNewIntList) {
-    // Create a new temporary list with elements reversed those of
-    // new_int_list_
-    const ListWrapperSimple* decending_list = CreateDecendingList(rand()%2);
-    ASSERT_FALSE(decending_list == NULL);
-    ASSERT_FALSE(decending_list->Empty());
-    ASSERT_EQ(kNumberOfElements,decending_list->GetSize());
-
-    const ListWrapperSimple* ascending_list = CreateAscendingList(rand()%2);
-    ASSERT_FALSE(ascending_list == NULL);
-    ASSERT_FALSE(ascending_list->Empty());
-    ASSERT_EQ(kNumberOfElements,ascending_list->GetSize());
-
-    ListWrapperSimple* list_to_reverse = ListWrapperSimple::Create(rand()%2);
-
-    // Reverse the list using PushBack and Previous.
-    for (ListItem* item = ascending_list->Last(); item != NULL;
-         item = ascending_list->Previous(item)) {
-         list_to_reverse->PushBack(ascending_list->GetUnsignedItem(item));
-    }
-
-    ASSERT_TRUE(CompareLists(decending_list,list_to_reverse));
-
-    ListWrapperSimple* list_to_un_reverse =
-        ListWrapperSimple::Create(rand()%2);
-    ASSERT_FALSE(list_to_un_reverse == NULL);
-    // Reverse the reversed list using PushFront and Next.
-    for (ListItem* item = list_to_reverse->First(); item != NULL;
-         item = list_to_reverse->Next(item)) {
-         list_to_un_reverse->PushFront(list_to_reverse->GetUnsignedItem(item));
-    }
-
-    ASSERT_TRUE(CompareLists(ascending_list,list_to_un_reverse));
-}
-
-TEST(ListWrapperTest,PopTest) {
-    ListWrapperSimple* ascending_list = CreateAscendingList(rand()%2);
-    ASSERT_FALSE(ascending_list == NULL);
-    ASSERT_FALSE(ascending_list->Empty());
-    EXPECT_EQ(0,ascending_list->PopFront());
-    EXPECT_EQ(1,ascending_list->GetUnsignedItem(ascending_list->First()));
-    
-    EXPECT_EQ(0,ascending_list->PopBack());
-    EXPECT_EQ(kNumberOfElements - 2,ascending_list->GetUnsignedItem(
-              ascending_list->Last()));
-    EXPECT_EQ(kNumberOfElements - 2, ascending_list->GetSize());
-}
-
-// Use Insert to interleave two lists.
-TEST(ListWrapperTest,InterLeaveTest) {
-    ListWrapperSimple* interleave_list = CreateAscendingList(rand()%2);
-    ASSERT_FALSE(interleave_list == NULL);
-    ASSERT_FALSE(interleave_list->Empty());
-
-    ListWrapperSimple* decending_list = CreateDecendingList(rand()%2);
-    ASSERT_FALSE(decending_list == NULL);
-
-    for (int i = 0; i < kNumberOfElements/2; ++i) {
-        ASSERT_EQ(0,interleave_list->PopBack());
-        ASSERT_EQ(0,decending_list->PopBack());
-    }
-    ASSERT_EQ(kNumberOfElements/2,interleave_list->GetSize());
-    ASSERT_EQ(kNumberOfElements/2,decending_list->GetSize());
-
-    int insert_position = kNumberOfElements/2;
-    ASSERT_EQ(insert_position * 2, kNumberOfElements);
-    while (!decending_list->Empty())
-    {
-        ListItem* item = decending_list->Last();
-        ASSERT_FALSE(item == NULL);
-
-        const unsigned int item_id = decending_list->GetUnsignedItem(item);
-        ASSERT_EQ(0,decending_list->Erase(item));
-
-        ListItem* insert_item = interleave_list->CreateListItem(item_id);
-        ASSERT_FALSE(insert_item == NULL);
-        item = interleave_list->First();
-        ASSERT_FALSE(item == NULL);
-        for (int j = 0; j < insert_position - 1; ++j) {
-            item = interleave_list->Next(item);
-            ASSERT_FALSE(item == NULL);
-        }
-        if (0 != interleave_list->Insert(item,insert_item)) {
-            interleave_list->DestroyListItem(insert_item);
-            FAIL();
-        }
-        --insert_position;
-    }
-    
-    ListWrapperSimple* interleaved_list = CreateInterleavedList(rand()%2);
-    ASSERT_FALSE(interleaved_list == NULL);
-    ASSERT_FALSE(interleaved_list->Empty());
-
-    ASSERT_TRUE(CompareLists(interleaved_list,interleave_list));
-}
-
-// Use InsertBefore to interleave two lists.
-TEST(ListWrapperTest,InterLeaveTestII) {
-    ListWrapperSimple* interleave_list = CreateDecendingList(rand()%2);
-    ASSERT_FALSE(interleave_list == NULL);
-    ASSERT_FALSE(interleave_list->Empty());
-
-    ListWrapperSimple* ascending_list = CreateAscendingList(rand()%2);
-    ASSERT_FALSE(ascending_list == NULL);
-
-    for (int i = 0; i < kNumberOfElements/2; ++i) {
-        ASSERT_EQ(0,interleave_list->PopBack());
-        ASSERT_EQ(0,ascending_list->PopBack());
-    }
-    ASSERT_EQ(kNumberOfElements/2,interleave_list->GetSize());
-    ASSERT_EQ(kNumberOfElements/2,ascending_list->GetSize());
-
-    int insert_position = kNumberOfElements/2;
-    ASSERT_EQ(insert_position * 2, kNumberOfElements);
-    while (!ascending_list->Empty())
-    {
-        ListItem* item = ascending_list->Last();
-        ASSERT_FALSE(item == NULL);
-
-        const unsigned int item_id = ascending_list->GetUnsignedItem(item);
-        ASSERT_EQ(0,ascending_list->Erase(item));
-
-        ListItem* insert_item = interleave_list->CreateListItem(item_id);
-        ASSERT_FALSE(insert_item == NULL);
-        item = interleave_list->First();
-        ASSERT_FALSE(item == NULL);
-        for (int j = 0; j < insert_position - 1; ++j) {
-            item = interleave_list->Next(item);
-            ASSERT_FALSE(item == NULL);
-        }
-        if (0 != interleave_list->InsertBefore(item,insert_item)) {
-            interleave_list->DestroyListItem(insert_item);
-            FAIL();
-        }
-        --insert_position;
-    }
-
-    ListWrapperSimple* interleaved_list = CreateInterleavedList(rand()%2);
-    ASSERT_FALSE(interleaved_list == NULL);
-    ASSERT_FALSE(interleaved_list->Empty());
-
-    ASSERT_TRUE(CompareLists(interleaved_list,interleave_list));
-}
-
-int main(int argc, char **argv)
-{
-    ::testing::InitGoogleTest(&argc, argv);
-    // Added return_value so that it's convenient to put a breakpoint before
-    // exiting please note that the return value from RUN_ALL_TESTS() must
-    // be returned by the main function.
-    const int return_value = RUN_ALL_TESTS();
-    return return_value;
-}
diff --git a/src/system_wrappers/source/map.cc b/src/system_wrappers/source/map.cc
deleted file mode 100644
index 0bff155..0000000
--- a/src/system_wrappers/source/map.cc
+++ /dev/null
@@ -1,166 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "map_wrapper.h"
-
-#include "trace.h"
-
-namespace webrtc {
-MapItem::MapItem(int id, void* item) : item_pointer_(item), item_id_(id)
-{
-}
-
-MapItem::~MapItem()
-{
-}
-
-void* MapItem::GetItem()
-{
-    return item_pointer_;
-}
-
-int MapItem::GetId()
-{
-    return item_id_;
-}
-
-unsigned int MapItem::GetUnsignedId()
-{
-    return static_cast<unsigned int>(item_id_);
-}
-
-void MapItem::SetItem(void* ptr)
-{
-    item_pointer_ = ptr;
-}
-
-MapWrapper::MapWrapper() : map_()
-{
-}
-
-MapWrapper::~MapWrapper()
-{
-    if (!map_.empty())
-    {
-        WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
-                   "Potential memory leak in MapWrapper");
-        // Remove all map items. Please note that std::map::clear() can't be
-        // used because each item has some dynamically allocated memory
-        // associated with it (i.e. using std::map::clear would introduce a
-        // memory leak).
-        while (Erase(First()) == 0)
-        {}
-    }
-}
-
-int MapWrapper::Size() const
-{
-    return (int)map_.size();
-}
-
-int MapWrapper::Insert(int id, void* ptr)
-{
-    map_[id] = new MapItem(id,ptr);
-    return 0;
-}
-
-MapItem* MapWrapper::First() const
-{
-    std::map<int, MapItem*>::const_iterator it = map_.begin();
-    if (it != map_.end())
-    {
-        return it->second;
-    }
-    return 0;
-}
-
-MapItem* MapWrapper::Last() const
-{
-    std::map<int, MapItem*>::const_reverse_iterator it = map_.rbegin();
-    if (it != map_.rend())
-    {
-        return it->second;
-    }
-    return 0;
-}
-
-MapItem* MapWrapper::Next(MapItem* item) const
-{
-    if (item == 0)
-    {
-        return 0;
-    }
-    std::map<int, MapItem*>::const_iterator it = map_.find(item->item_id_);
-    if (it != map_.end())
-    {
-        it++;
-        if (it != map_.end())
-        {
-            return it->second;
-        }
-    }
-    return 0;
-}
-
-MapItem* MapWrapper::Previous(MapItem* item) const
-{
-    if (item == 0)
-    {
-        return 0;
-    }
-
-    std::map<int, MapItem*>::const_iterator it = map_.find(item->item_id_);
-    if ((it != map_.end()) &&
-       (it != map_.begin()))
-    {
-        --it;
-        return it->second;
-    }
-    return 0;
-}
-
-MapItem* MapWrapper::Find(int id) const
-{
-    std::map<int, MapItem*>::const_iterator it = map_.find(id);
-    if (it != map_.end())
-    {
-        return it->second;
-    }
-    return 0;
-}
-
-int MapWrapper::Erase(MapItem* item)
-{
-    if (item == 0)
-    {
-        return -1;
-    }
-    std::map<int, MapItem*>::iterator it = map_.find(item->item_id_);
-    if (it != map_.end())
-    {
-        delete it->second;
-        map_.erase(it);
-        return 0;
-    }
-    return -1;
-}
-
-int MapWrapper::Erase(const int id)
-{
-    std::map<int, MapItem*>::iterator it = map_.find(id);
-    if (it != map_.end())
-    {
-        delete it->second;
-        map_.erase(it);
-        return 0;
-    }
-    return -1;
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/map_no_stl.cc b/src/system_wrappers/source/map_no_stl.cc
deleted file mode 100644
index cb0ac00..0000000
--- a/src/system_wrappers/source/map_no_stl.cc
+++ /dev/null
@@ -1,217 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "map_no_stl.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-namespace webrtc {
-MapNoStlItem::MapNoStlItem(int id, void* item)
-    : next_(0),
-      prev_(0),
-      item_id_(id),
-      item_ptr_(item)
-{
-}
-
-MapNoStlItem::~MapNoStlItem()
-{
-}
-
-void* MapNoStlItem::GetItem()
-{
-    return item_ptr_;
-}
-
-int MapNoStlItem::GetId()
-{
-    return item_id_;
-}
-
-unsigned int MapNoStlItem::GetUnsignedId()
-{
-    return static_cast<unsigned int>(item_id_);
-}
-
-void MapNoStlItem::SetItem(void* ptr)
-{
-    item_ptr_ = ptr;
-}
-
-MapNoStl::MapNoStl()
-    : critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
-      first_(0),
-      last_(0),
-      size_(0)
-{
-}
-
-MapNoStl::~MapNoStl()
-{
-    if (First())
-    {
-        WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
-                   "Potential memory leak in MapNoStl");
-        while (Erase(First()) == 0)
-        {}
-    }
-    delete critical_section_;
-}
-
-int MapNoStl::Size() const
-{
-    return size_;
-}
-
-int MapNoStl::Insert(int id, void* ptr)
-{
-    MapNoStlItem* new_item = new MapNoStlItem(id, ptr);
-
-    CriticalSectionScoped lock(*critical_section_);
-    MapNoStlItem* item = first_;
-    size_++;
-    if (!item)
-    {
-        first_ = new_item;
-        last_ = new_item;
-        return 0;
-    }
-    while(item->next_)
-    {
-        // Three scenarios
-        // 1. Item should be inserted first.
-        // 2. Item should be inserted between two items
-        // 3. Item should be inserted last
-        if (item->GetId() > id)
-        {
-            new_item->next_ = item;
-            item->prev_ = new_item;
-            if (item == first_)
-            {
-                first_ = new_item;
-            }
-            else
-            {
-                new_item->prev_ = item->prev_;
-                new_item->prev_->next_ = new_item;
-            }
-            return 0;
-        }
-        item = item->next_;
-    }
-    // 3
-    item->next_ = new_item;
-    new_item->prev_ = item;
-    last_ = new_item;
-    return 0;
-}
-
-MapNoStlItem* MapNoStl::First() const
-{
-    return first_;
-}
-
-MapNoStlItem* MapNoStl::Last() const
-{
-    return last_;
-}
-
-MapNoStlItem* MapNoStl::Next(MapNoStlItem* item) const
-{
-    if (!item)
-    {
-        return 0;
-    }
-    return item->next_;
-}
-
-MapNoStlItem* MapNoStl::Previous(MapNoStlItem* item) const
-{
-    if (!item)
-    {
-        return 0;
-    }
-    return item->prev_;
-}
-
-MapNoStlItem* MapNoStl::Find(int id) const
-{
-    CriticalSectionScoped lock(*critical_section_);
-    MapNoStlItem* item = Locate(id);
-    return item;
-}
-
-int MapNoStl::Erase(MapNoStlItem* item)
-{
-    if(!item)
-    {
-        return -1;
-    }
-    CriticalSectionScoped lock(*critical_section_);
-    return Remove(item);
-}
-
-int MapNoStl::Erase(const int id)
-{
-    CriticalSectionScoped lock(*critical_section_);
-    MapNoStlItem* item = Locate(id);
-    if(!item)
-    {
-        return -1;
-    }
-    return Remove(item);
-}
-
-MapNoStlItem* MapNoStl::Locate(int id) const
-{
-    MapNoStlItem* item = first_;
-    while(item)
-    {
-        if (item->GetId() == id)
-        {
-            return item;
-        }
-        item = item->next_;
-    }
-    return 0;
-}
-
-int MapNoStl::Remove(MapNoStlItem* item)
-{
-    if (!item)
-    {
-        return -1;
-    }
-    size_--;
-    MapNoStlItem* previous_item = item->prev_;
-    MapNoStlItem* next_item = item->next_;
-    if (!previous_item)
-    {
-        next_item->prev_ = 0;
-        first_ = next_item;
-    }
-    else
-    {
-        previous_item->next_ = next_item;
-    }
-    if (!next_item)
-    {
-        previous_item->next_ = 0;
-        last_ = previous_item;
-    }
-    else
-    {
-        next_item->prev_ = previous_item;
-    }
-    delete item;
-    return 0;
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/map_no_stl.h b/src/system_wrappers/source/map_no_stl.h
deleted file mode 100644
index 51bc011..0000000
--- a/src/system_wrappers/source/map_no_stl.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_MAP_NO_STL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_MAP_NO_STL_H_
-
-#include "constructor_magic.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class MapNoStlItem
-{
-friend class Map;
-
-public:
-    MapNoStlItem(int id, void* ptr);
-    virtual ~MapNoStlItem();
-    void* GetItem();
-    int GetId();
-    unsigned int GetUnsignedId();
-    void SetItem(void* ptr);
-
-protected:
-    MapNoStlItem* next_;
-    MapNoStlItem* prev_;
-
-private:
-    int item_id_;
-    void* item_ptr_;
-    DISALLOW_COPY_AND_ASSIGN(MapNoStlItem);
-};
-
-class MapNoStl
-{
-public:
-    MapNoStl();
-    virtual ~MapNoStl();
-
-    // MapWrapper functions.
-    int Insert(int id, void* ptr);
-    int Erase(MapNoStlItem* item);
-    int Erase(int id);
-    int Size() const;
-    MapNoStlItem* First() const;
-    MapNoStlItem* Last() const;
-    MapNoStlItem* Next(MapNoStlItem* item) const;
-    MapNoStlItem* Previous(MapNoStlItem* item) const;
-    MapNoStlItem* Find(int id) const;
-
-private:
-    MapNoStlItem* Locate(int id) const;
-    int Remove(MapNoStlItem* item);
-
-    CriticalSection* critical_section_;
-    MapNoStlItem* first_;
-    MapNoStlItem* last_;
-    int size_;
-    DISALLOW_COPY_AND_ASSIGN(MapNoStl);
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_MAP_NO_STL_H_
diff --git a/src/system_wrappers/source/map_unittest.cc b/src/system_wrappers/source/map_unittest.cc
deleted file mode 100644
index 8e8ea07..0000000
--- a/src/system_wrappers/source/map_unittest.cc
+++ /dev/null
@@ -1,231 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gtest/gtest.h"
-
-#include "map_wrapper.h"
-
-using ::webrtc::MapWrapper;
-using ::webrtc::MapItem;
-
-const unsigned int kNumberOfElements = 10;
-
-int* ItemPointer(MapItem* item) {
-    if (item == NULL) {
-        return NULL;
-    }
-    return reinterpret_cast<int*>(item->GetItem());
-}
-
-bool DeleteItemContent(MapItem* item) {
-    if(item == NULL) {
-        return false;
-    }
-    int* value_ptr = ItemPointer(item);
-    delete value_ptr;
-    return true;
-}
-
-int ItemValue(MapItem* item) {
-    if (item == NULL) {
-        return -1;
-    }
-    const int* value_ptr = ItemPointer(item);
-    if (value_ptr == 0) {
-        return -1;
-    }
-    return *value_ptr;
-}
-
-void PrintToConsole(const char* message, bool supress) {
-    if (supress) {
-        return;
-    }
-    printf(message);
-}
-
-bool CreateAscendingMap(MapWrapper* ascending_map) {
-    int* insert_value = NULL;
-    for (int i = 0; i < kNumberOfElements; ++i) {
-        insert_value = new int;
-        if (insert_value == NULL) {
-            return false;
-        }
-        *insert_value = i;
-        if (0 != ascending_map->Insert(
-                     i,
-                     reinterpret_cast<void*>(insert_value))) {
-            return false;
-        }
-    }
-    return true;
-}
-
-bool ClearMap(MapWrapper* clear_map) {
-    bool success = true;
-    while (clear_map->Size() != 0) {
-        MapItem* remove_item = clear_map->First();
-        if (remove_item == NULL) {
-            return false;
-        }
-        if (!DeleteItemContent(remove_item)) {
-            success = false;
-        }
-        if (clear_map->Erase(remove_item) != 0) {
-            return false;
-        }
-    }
-    return success;
-}
-
-void PrintMapItem(MapItem* item, bool supress) {
-    const int id = item->GetId();
-    const int value = ItemValue(item);
-    char print_buffer[16];
-    sprintf(print_buffer, "(%3i,%3i) ", id, value);
-    PrintToConsole(print_buffer, supress);
-}
-
-// Succeeds only if all the items were printed.
-bool PrintMap(const MapWrapper& print_map, bool supress) {
-    const int elements_to_print = print_map.Size();
-    int elements_printed = 0;
-    MapItem* item = print_map.First();
-    PrintToConsole("[", supress);
-    while (item != NULL) {
-        PrintMapItem(item, supress);
-        ++elements_printed;
-        item = print_map.Next(item);
-    }
-    PrintToConsole("]\n", supress);
-    return elements_printed == elements_to_print;
-}
-
-// Succeeds only if all the items were printed.
-bool ReversePrintMap(const MapWrapper& print_map, bool supress) {
-    const int elements_to_print = print_map.Size();
-    int elements_printed = 0;
-    MapItem* item = print_map.Last();
-    PrintToConsole("[", supress);
-    while (item != NULL) {
-        PrintMapItem(item, supress);
-        ++elements_printed;
-        item = print_map.Previous(item);
-    }
-    PrintToConsole("]\n", supress);
-    return elements_printed == elements_to_print;
-}
-
-// Returns true if the map items contain the same item.
-bool CompareItems(MapItem* lhs_item, MapItem* rhs_item) {
-    if ((lhs_item == NULL) || (rhs_item == NULL)) {
-        return false;
-    }
-    if (lhs_item->GetId() != rhs_item->GetId()) {
-        return false;
-    }
-    return lhs_item->GetItem() == rhs_item->GetItem();
-}
-
-// Returns true if the map contains the same items.
-bool CompareMaps(const MapWrapper& lhs, const MapWrapper& rhs) {
-    const int map_size = lhs.Size();
-    if (map_size != rhs.Size()) {
-        return false;
-    }
-    int item_count = 0;
-    MapItem* lhs_item = lhs.First();
-    while (lhs_item != NULL) {
-        MapItem* rhs_item = rhs.Find(lhs_item->GetId());
-        if (rhs_item == NULL) {
-            return false;
-        }
-        if (!CompareItems(lhs_item, rhs_item)) {
-            return false;
-        }
-        ++item_count;
-        lhs_item = lhs.Next(lhs_item);
-    }
-    return item_count == map_size;
-} 
-
-class MapWrapperTest : public ::testing::Test {
-protected:
-    virtual void SetUp() {
-        ASSERT_TRUE(CreateAscendingMap(&ascending_map_));
-    }
-
-    virtual void TearDown() {
-        EXPECT_TRUE(ClearMap(&ascending_map_));
-    }
-    MapWrapper ascending_map_;
-};
-
-TEST_F(MapWrapperTest,RemoveTest) {
-    // Erase using int id
-    { // Create local scope to avoid accidental re-use
-        MapItem* item_first = ascending_map_.First();
-        ASSERT_FALSE(item_first == NULL);
-        const int first_value_id = item_first->GetId();
-        const int first_value = ItemValue(item_first);
-        EXPECT_TRUE(first_value == 0);
-        EXPECT_EQ(first_value_id,first_value);
-        EXPECT_FALSE(NULL == ascending_map_.Find(first_value_id));
-        EXPECT_TRUE(DeleteItemContent(item_first));
-        ascending_map_.Erase(first_value_id);
-        EXPECT_TRUE(NULL == ascending_map_.Find(first_value_id));
-        EXPECT_EQ(kNumberOfElements-1,ascending_map_.Size());
-    }
-    
-    // Erase using MapItem* item
-    MapItem* item_last = ascending_map_.Last();
-    ASSERT_FALSE(item_last == NULL);
-    const int last_value_id = item_last->GetId();
-    const int last_value = ItemValue(item_last);
-    EXPECT_TRUE(last_value == kNumberOfElements - 1);    
-    EXPECT_EQ(last_value_id, last_value);
-    EXPECT_FALSE(NULL == ascending_map_.Find(last_value_id));
-    EXPECT_TRUE(DeleteItemContent(item_last));
-    ascending_map_.Erase(last_value_id);
-    EXPECT_TRUE(NULL == ascending_map_.Find(last_value_id));
-    EXPECT_EQ(kNumberOfElements-2,ascending_map_.Size());
-}
-
-TEST_F(MapWrapperTest, PrintTest) {
-    const bool supress = true; // Don't spam the console
-
-    EXPECT_TRUE(PrintMap(ascending_map_, supress));
-    EXPECT_TRUE(ReversePrintMap(ascending_map_, supress));
-}
-
-TEST_F(MapWrapperTest, CopyTest) {
-    MapWrapper compare_map;
-    ASSERT_TRUE(CreateAscendingMap(&compare_map));
-    const int map_size = compare_map.Size();
-    ASSERT_EQ(ascending_map_.Size(), map_size);
-    // CompareMaps compare the pointers not value of the pointers.
-    // (the values are the same since both are ascending maps).
-    EXPECT_FALSE(CompareMaps(compare_map,ascending_map_));
-
-    int copy_count = 0;
-    MapItem* ascend_item = ascending_map_.First();
-    while (ascend_item != NULL) {
-        MapItem* compare_item = compare_map.Find(ascend_item->GetId());
-        ASSERT_FALSE(compare_item == NULL);
-        DeleteItemContent(compare_item);
-        compare_item->SetItem(ascend_item->GetItem());
-        ascend_item = ascending_map_.Next(ascend_item);
-        ++copy_count;
-    }
-    EXPECT_TRUE(CompareMaps(compare_map,ascending_map_));
-    while (compare_map.Erase(compare_map.First()) == 0) {
-    }
-    EXPECT_EQ(map_size, copy_count);
-}
diff --git a/src/system_wrappers/source/rw_lock.cc b/src/system_wrappers/source/rw_lock.cc
deleted file mode 100644
index 47901d3..0000000
--- a/src/system_wrappers/source/rw_lock.cc
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rw_lock_wrapper.h"
-
-#include <assert.h>
-
-#if defined(_WIN32)
-    #include "rw_lock_windows.h"
-#elif defined(ANDROID)
-    #include <stdlib.h>
-    #include "rw_lock_generic.h"
-#else
-    #include "rw_lock_linux.h"
-#endif
-
-namespace webrtc {
-RWLockWrapper* RWLockWrapper::CreateRWLock()
-{
-#ifdef _WIN32
-    RWLockWrapper* lock =  new RWLockWindows();
-#elif defined(ANDROID)
-    RWLockWrapper* lock =  new RWLockWrapperGeneric();
-#else
-    RWLockWrapper* lock =  new RWLockLinux();
-#endif
-    if(lock->Init() != 0)
-    {
-        delete lock;
-        assert(false);
-        return NULL;
-    }
-    return lock;
-}
-
-RWLockWrapper::~RWLockWrapper()
-{
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/rw_lock_generic.cc b/src/system_wrappers/source/rw_lock_generic.cc
deleted file mode 100644
index a468ef3..0000000
--- a/src/system_wrappers/source/rw_lock_generic.cc
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rw_lock_generic.h"
-
-#include "condition_variable_wrapper.h"
-#include "critical_section_wrapper.h"
-
-namespace webrtc {
-RWLockWrapperGeneric::RWLockWrapperGeneric()
-    : _readersActive(0),
-      _writerActive(false),
-      _readersWaiting(0),
-      _writersWaiting(0)
-{
-    _critSectPtr  = CriticalSectionWrapper::CreateCriticalSection();
-    _readCondPtr  = ConditionVariableWrapper::CreateConditionVariable();
-    _writeCondPtr = ConditionVariableWrapper::CreateConditionVariable();
-}
-
-RWLockWrapperGeneric::~RWLockWrapperGeneric()
-{
-    delete _writeCondPtr;
-    delete _readCondPtr;
-    delete _critSectPtr;
-}
-
-int RWLockWrapperGeneric::Init()
-{
-    return 0;
-}
-
-void RWLockWrapperGeneric::AcquireLockExclusive()
-{
-    _critSectPtr->Enter();
-
-    if (_writerActive || _readersActive > 0)
-    {
-        ++_writersWaiting;
-
-        while (_writerActive || _readersActive > 0)
-        {
-            _writeCondPtr->SleepCS(*_critSectPtr);
-        }
-
-        --_writersWaiting;
-    }
-    _writerActive = true;
-    _critSectPtr->Leave();
-}
-
-void RWLockWrapperGeneric::ReleaseLockExclusive()
-{
-    _critSectPtr->Enter();
-
-    _writerActive = false;
-
-    if (_writersWaiting > 0)
-    {
-        _writeCondPtr->Wake();
-
-    }else if (_readersWaiting > 0)
-    {
-        _readCondPtr->WakeAll();
-    }
-    _critSectPtr->Leave();
-}
-
-void RWLockWrapperGeneric::AcquireLockShared()
-{
-    _critSectPtr->Enter();
-
-    if (_writerActive || _writersWaiting > 0)
-    {
-        ++_readersWaiting;
-
-        while (_writerActive || _writersWaiting > 0)
-        {
-            _readCondPtr->SleepCS(*_critSectPtr);
-        }
-        --_readersWaiting;
-    }
-    ++_readersActive;
-    _critSectPtr->Leave();
-}
-
-void RWLockWrapperGeneric::ReleaseLockShared()
-{
-    _critSectPtr->Enter();
-
-    --_readersActive;
-
-    if (_readersActive == 0 && _writersWaiting > 0)
-    {
-        _writeCondPtr->Wake();
-    }
-    _critSectPtr->Leave();
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/rw_lock_generic.h b/src/system_wrappers/source/rw_lock_generic.h
deleted file mode 100644
index fff5e5d..0000000
--- a/src/system_wrappers/source/rw_lock_generic.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_GENERIC_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_GENERIC_H_
-
-#include "rw_lock_wrapper.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class ConditionVariableWrapper;
-
-class RWLockWrapperGeneric : public RWLockWrapper
-{
-public:
-    RWLockWrapperGeneric();
-    virtual ~RWLockWrapperGeneric();
-
-    virtual void AcquireLockExclusive();
-    virtual void ReleaseLockExclusive();
-
-    virtual void AcquireLockShared();
-    virtual void ReleaseLockShared();
-
-protected:
-    virtual int Init();
-
-private:
-    CriticalSectionWrapper*   _critSectPtr;
-    ConditionVariableWrapper* _readCondPtr;
-    ConditionVariableWrapper* _writeCondPtr;
-
-    int  _readersActive;
-    bool _writerActive;
-    int  _readersWaiting;
-    int  _writersWaiting;
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_GENERIC_H_
diff --git a/src/system_wrappers/source/rw_lock_linux.cc b/src/system_wrappers/source/rw_lock_linux.cc
deleted file mode 100644
index 084dce8..0000000
--- a/src/system_wrappers/source/rw_lock_linux.cc
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rw_lock_linux.h"
-
-namespace webrtc {
-RWLockLinux::RWLockLinux() : _lock()
-{
-}
-
-RWLockLinux::~RWLockLinux()
-{
-    pthread_rwlock_destroy(&_lock);
-}
-
-int RWLockLinux::Init()
-{
-    return pthread_rwlock_init(&_lock, 0);
-}
-
-void RWLockLinux::AcquireLockExclusive()
-{
-    pthread_rwlock_wrlock(&_lock);
-}
-
-void RWLockLinux::ReleaseLockExclusive()
-{
-    pthread_rwlock_unlock(&_lock);
-}
-
-void RWLockLinux::AcquireLockShared()
-{
-    pthread_rwlock_rdlock(&_lock);
-}
-
-void RWLockLinux::ReleaseLockShared()
-{
-    pthread_rwlock_unlock(&_lock);
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/rw_lock_linux.h b/src/system_wrappers/source/rw_lock_linux.h
deleted file mode 100644
index 391ee8f..0000000
--- a/src/system_wrappers/source/rw_lock_linux.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_LINUX_H_
-
-#include "rw_lock_wrapper.h"
-
-#include <pthread.h>
-
-namespace webrtc {
-class RWLockLinux : public RWLockWrapper
-{
-public:
-    RWLockLinux();
-    virtual ~RWLockLinux();
-
-    virtual void AcquireLockExclusive();
-    virtual void ReleaseLockExclusive();
-
-    virtual void AcquireLockShared();
-    virtual void ReleaseLockShared();
-
-protected:
-    virtual int Init();
-
-private:
-    pthread_rwlock_t _lock;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_LINUX_H_
diff --git a/src/system_wrappers/source/sort.cc b/src/system_wrappers/source/sort.cc
deleted file mode 100644
index f44b644..0000000
--- a/src/system_wrappers/source/sort.cc
+++ /dev/null
@@ -1,551 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-// When the platform supports STL, the functions are implemented using a
-// templated spreadsort algorithm (http://sourceforge.net/projects/spreadsort/),
-// part of the Boost C++ library collection. Otherwise, the C standard library's
-// qsort() will be used.
-
-#include "sort.h"
-
-#include <cassert>
-#include <cstring>  // memcpy
-#include <new>      // nothrow new
-
-#ifdef NO_STL
-#include <cstdlib>      // qsort
-#else
-#include <algorithm>    // std::sort
-#include <vector>
-#include "spreadsort.hpp" // TODO (ajm) upgrade to spreadsortv2.
-#endif
-
-#ifdef NO_STL
-#define COMPARE_DEREFERENCED(XT, YT)        \
-    do                                      \
-    {                                       \
-        if ((XT) > (YT))                    \
-        {                                   \
-            return 1;                       \
-        }                                   \
-        else if ((XT) < (YT))               \
-        {                                   \
-            return -1;                      \
-        }                                   \
-                                            \
-        return 0;                           \
-    }                                       \
-    while(0)
-
-#define COMPARE_FOR_QSORT(X, Y, TYPE)                               \
-    do                                                              \
-    {                                                               \
-        TYPE xT = static_cast<TYPE>(*static_cast<const TYPE*>(X));  \
-        TYPE yT = static_cast<TYPE>(*static_cast<const TYPE*>(Y));  \
-        COMPARE_DEREFERENCED(xT, yT);                               \
-    }                                                               \
-    while(0)
-
-#define COMPARE_KEY_FOR_QSORT(SORT_KEY_X, SORT_KEY_Y, TYPE)         \
-    do                                                              \
-    {                                                               \
-        TYPE xT = static_cast<TYPE>(*static_cast<TYPE*>             \
-            (static_cast<const SortKey*>(SORT_KEY_X)->key));        \
-        TYPE yT = static_cast<TYPE>(*static_cast<TYPE*>             \
-            (static_cast<const SortKey*>(SORT_KEY_Y)->key));        \
-        COMPARE_DEREFERENCED(xT, yT);                               \
-    }                                                               \
-    while(0)
-
-#define KEY_QSORT(SORT_KEY, KEY, NUM_OF_ELEMENTS, KEY_TYPE, COMPARE_FUNC)     \
-    do                                                                        \
-    {                                                                         \
-        KEY_TYPE* keyT = (KEY_TYPE*)(key);                                    \
-        for (WebRtc_UWord32 i = 0; i < (NUM_OF_ELEMENTS); i++)                \
-        {                                                                     \
-            ptrSortKey[i].key = &keyT[i];                                     \
-            ptrSortKey[i].index = i;                                          \
-        }                                                                     \
-                                                                              \
-        qsort((SORT_KEY), (NUM_OF_ELEMENTS), sizeof(SortKey), (COMPARE_FUNC));\
-    }                                                                         \
-    while(0)
-#endif
-
-namespace webrtc
-{
-#ifdef NO_STL
-    struct SortKey
-    {
-        void* key;
-        WebRtc_UWord32 index;
-    };
-#else
-    template<typename KeyType>
-    struct SortKey
-    {
-        KeyType key;
-        WebRtc_UWord32 index;
-    };
-#endif
-
-    namespace // Unnamed namespace provides internal linkage.
-    {
-#ifdef NO_STL
-        int CompareWord8(const void* x, const void* y)
-        {
-            COMPARE_FOR_QSORT(x, y, WebRtc_Word8);
-        }
-
-        int CompareUWord8(const void* x, const void* y)
-        {
-            COMPARE_FOR_QSORT(x, y, WebRtc_UWord8);
-        }
-
-        int CompareWord16(const void* x, const void* y)
-        {
-            COMPARE_FOR_QSORT(x, y, WebRtc_Word16);
-        }
-
-        int CompareUWord16(const void* x, const void* y)
-        {
-            COMPARE_FOR_QSORT(x, y, WebRtc_UWord16);
-        }
-
-        int CompareWord32(const void* x, const void* y)
-        {
-            COMPARE_FOR_QSORT(x, y, WebRtc_Word32);
-        }
-
-        int CompareUWord32(const void* x, const void* y)
-        {
-            COMPARE_FOR_QSORT(x, y, WebRtc_UWord32);
-        }
-
-        int CompareWord64(const void* x, const void* y)
-        {
-            COMPARE_FOR_QSORT(x, y, WebRtc_Word64);
-        }
-
-        int CompareUWord64(const void* x, const void* y)
-        {
-            COMPARE_FOR_QSORT(x, y, WebRtc_UWord64);
-        }
-
-        int CompareFloat32(const void* x, const void* y)
-        {
-            COMPARE_FOR_QSORT(x, y, float);
-        }
-
-        int CompareFloat64(const void* x, const void* y)
-        {
-            COMPARE_FOR_QSORT(x, y, double);
-        }
-
-        int CompareKeyWord8(const void* sortKeyX, const void* sortKeyY)
-        {
-            COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word8);
-        }
-
-        int CompareKeyUWord8(const void* sortKeyX, const void* sortKeyY)
-        {
-            COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord8);
-        }
-
-        int CompareKeyWord16(const void* sortKeyX, const void* sortKeyY)
-        {
-            COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word16);
-        }
-
-        int CompareKeyUWord16(const void* sortKeyX, const void* sortKeyY)
-        {
-            COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord16);
-        }
-
-        int CompareKeyWord32(const void* sortKeyX, const void* sortKeyY)
-        {
-            COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word32);
-        }
-
-        int CompareKeyUWord32(const void* sortKeyX, const void* sortKeyY)
-        {
-            COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord32);
-        }
-
-        int CompareKeyWord64(const void* sortKeyX, const void* sortKeyY)
-        {
-            COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word64);
-        }
-
-        int CompareKeyUWord64(const void* sortKeyX, const void* sortKeyY)
-        {
-            COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord64);
-        }
-
-        int CompareKeyFloat32(const void* sortKeyX, const void* sortKeyY)
-        {
-            COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, float);
-        }
-
-        int CompareKeyFloat64(const void* sortKeyX, const void* sortKeyY)
-        {
-            COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, double);
-        }
-#else
-        template <typename KeyType>
-        struct KeyLessThan
-        {
-            bool operator()(const SortKey<KeyType>& sortKeyX,
-                const SortKey<KeyType>& sortKeyY) const
-            {
-                return sortKeyX.key < sortKeyY.key;
-            }
-        };
-
-        template <typename KeyType>
-        struct KeyRightShift
-        {
-            KeyType operator()(const SortKey<KeyType>& sortKey,
-                const unsigned offset) const
-            {
-                return sortKey.key >> offset;
-            }
-        };
-
-        template <typename DataType>
-        inline void IntegerSort(void* data, WebRtc_UWord32 numOfElements)
-        {
-            DataType* dataT = static_cast<DataType*>(data);
-            boost::integer_sort(dataT, dataT + numOfElements);
-        }
-
-        template <typename DataType, typename IntegerType>
-        inline void FloatSort(void* data, WebRtc_UWord32 numOfElements)
-        {
-            DataType* dataT = static_cast<DataType*>(data);
-            IntegerType cVal = 0;
-            boost::float_sort_cast(dataT, dataT + numOfElements, cVal);
-        }
-
-        template <typename DataType>
-        inline void StdSort(void* data, WebRtc_UWord32 numOfElements)
-        {
-            DataType* dataT = static_cast<DataType*>(data);
-            std::sort(dataT, dataT + numOfElements);
-        }
-
-        template<typename KeyType>
-        inline WebRtc_Word32 SetupKeySort(void* key,
-                                          SortKey<KeyType>*& ptrSortKey,
-                                          WebRtc_UWord32 numOfElements)
-        {
-            ptrSortKey = new(std::nothrow) SortKey<KeyType>[numOfElements];
-            if (ptrSortKey == NULL)
-            {
-                return -1;
-            }
-
-            KeyType* keyT = static_cast<KeyType*>(key);
-            for (WebRtc_UWord32 i = 0; i < numOfElements; i++)
-            {
-                ptrSortKey[i].key = keyT[i];
-                ptrSortKey[i].index = i;
-            }
-
-            return 0;
-        }
-
-        template<typename KeyType>
-        inline WebRtc_Word32 TeardownKeySort(void* data,
-                                             SortKey<KeyType>* ptrSortKey,
-            WebRtc_UWord32 numOfElements, WebRtc_UWord32 sizeOfElement)
-        {
-            WebRtc_UWord8* ptrData = static_cast<WebRtc_UWord8*>(data);
-            WebRtc_UWord8* ptrDataSorted = new(std::nothrow) WebRtc_UWord8
-                [numOfElements * sizeOfElement];
-            if (ptrDataSorted == NULL)
-            {
-                return -1;
-            }
-
-            for (WebRtc_UWord32 i = 0; i < numOfElements; i++)
-            {
-                memcpy(ptrDataSorted + i * sizeOfElement, ptrData +
-                       ptrSortKey[i].index * sizeOfElement, sizeOfElement);
-            }
-            memcpy(ptrData, ptrDataSorted, numOfElements * sizeOfElement);
-            delete[] ptrSortKey;
-            delete[] ptrDataSorted;
-            return 0;
-        }
-
-        template<typename KeyType>
-        inline WebRtc_Word32 IntegerKeySort(void* data, void* key,
-                                            WebRtc_UWord32 numOfElements,
-                                            WebRtc_UWord32 sizeOfElement)
-        {
-            SortKey<KeyType>* ptrSortKey;
-            if (SetupKeySort<KeyType>(key, ptrSortKey, numOfElements) != 0)
-            {
-                return -1;
-            }
-
-            boost::integer_sort(ptrSortKey, ptrSortKey + numOfElements,
-                KeyRightShift<KeyType>(), KeyLessThan<KeyType>());
-
-            if (TeardownKeySort<KeyType>(data, ptrSortKey, numOfElements,
-                    sizeOfElement) != 0)
-            {
-                return -1;
-            }
-
-            return 0;
-        }
-
-        template<typename KeyType>
-        inline WebRtc_Word32 StdKeySort(void* data, void* key,
-                                        WebRtc_UWord32 numOfElements,
-                                        WebRtc_UWord32 sizeOfElement)
-        {
-            SortKey<KeyType>* ptrSortKey;
-            if (SetupKeySort<KeyType>(key, ptrSortKey, numOfElements) != 0)
-            {
-                return -1;
-            }
-
-            std::sort(ptrSortKey, ptrSortKey + numOfElements,
-                KeyLessThan<KeyType>());
-
-            if (TeardownKeySort<KeyType>(data, ptrSortKey, numOfElements,
-                    sizeOfElement) != 0)
-            {
-                return -1;
-            }
-
-            return 0;
-        }
-#endif
-    }
-
-    WebRtc_Word32 Sort(void* data, WebRtc_UWord32 numOfElements, Type type)
-    {
-        if (data == NULL)
-        {
-            return -1;
-        }
-
-#ifdef NO_STL
-        switch (type)
-        {
-        case TYPE_Word8:
-            qsort(data, numOfElements, sizeof(WebRtc_Word8), CompareWord8);
-            break;
-        case TYPE_UWord8:
-            qsort(data, numOfElements, sizeof(WebRtc_UWord8), CompareUWord8);
-            break;
-        case TYPE_Word16:
-            qsort(data, numOfElements, sizeof(WebRtc_Word16), CompareWord16);
-            break;
-        case TYPE_UWord16:
-            qsort(data, numOfElements, sizeof(WebRtc_UWord16), CompareUWord16);
-            break;
-        case TYPE_Word32:
-            qsort(data, numOfElements, sizeof(WebRtc_Word32), CompareWord32);
-            break;
-        case TYPE_UWord32:
-            qsort(data, numOfElements, sizeof(WebRtc_UWord32), CompareUWord32);
-            break;
-        case TYPE_Word64:
-            qsort(data, numOfElements, sizeof(WebRtc_Word64), CompareWord64);
-            break;
-        case TYPE_UWord64:
-            qsort(data, numOfElements, sizeof(WebRtc_UWord64), CompareUWord64);
-            break;
-        case TYPE_Float32:
-            qsort(data, numOfElements, sizeof(float), CompareFloat32);
-            break;
-        case TYPE_Float64:
-            qsort(data, numOfElements, sizeof(double), CompareFloat64);
-            break;
-        default:
-            return -1;
-        }
-#else
-        // Fall back to std::sort for 64-bit types and floats due to compiler
-	// warnings and VS 2003 build crashes respectively with spreadsort.
-        switch (type)
-        {
-        case TYPE_Word8:
-            IntegerSort<WebRtc_Word8>(data, numOfElements);
-            break;
-        case TYPE_UWord8:
-            IntegerSort<WebRtc_UWord8>(data, numOfElements);
-            break;
-        case TYPE_Word16:
-            IntegerSort<WebRtc_Word16>(data, numOfElements);
-            break;
-        case TYPE_UWord16:
-            IntegerSort<WebRtc_UWord16>(data, numOfElements);
-            break;
-        case TYPE_Word32:
-            IntegerSort<WebRtc_Word32>(data, numOfElements);
-            break;
-        case TYPE_UWord32:
-            IntegerSort<WebRtc_UWord32>(data, numOfElements);
-            break;
-        case TYPE_Word64:
-            StdSort<WebRtc_Word64>(data, numOfElements);
-            break;
-        case TYPE_UWord64:
-            StdSort<WebRtc_UWord64>(data, numOfElements);
-            break;
-        case TYPE_Float32:
-            StdSort<float>(data, numOfElements);
-            break;
-        case TYPE_Float64:
-            StdSort<double>(data, numOfElements);
-            break;
-        default:
-            return -1;
-        }
-#endif
-        return 0;
-    }
-
-    WebRtc_Word32 KeySort(void* data, void* key, WebRtc_UWord32 numOfElements,
-                          WebRtc_UWord32 sizeOfElement, Type keyType)
-    {
-        if (data == NULL)
-        {
-            return -1;
-        }
-
-        if (key == NULL)
-        {
-            return -1;
-        }
-
-        if ((WebRtc_UWord64)numOfElements * sizeOfElement > 0xffffffff)
-        {
-            return -1;
-        }
-
-#ifdef NO_STL
-        SortKey* ptrSortKey = new(std::nothrow) SortKey[numOfElements];
-        if (ptrSortKey == NULL)
-        {
-            return -1;
-        }
-
-        switch (keyType)
-        {
-        case TYPE_Word8:
-            KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word8,
-                CompareKeyWord8);
-            break;
-        case TYPE_UWord8:
-            KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord8,
-                CompareKeyUWord8);
-            break;
-        case TYPE_Word16:
-            KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word16,
-                CompareKeyWord16);
-            break;
-        case TYPE_UWord16:
-            KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord16,
-                CompareKeyUWord16);
-            break;
-        case TYPE_Word32:
-            KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word32,
-                CompareKeyWord32);
-            break;
-        case TYPE_UWord32:
-            KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord32,
-                CompareKeyUWord32);
-            break;
-        case TYPE_Word64:
-            KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word64,
-                CompareKeyWord64);
-            break;
-        case TYPE_UWord64:
-            KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord64,
-                CompareKeyUWord64);
-            break;
-        case TYPE_Float32:
-            KEY_QSORT(ptrSortKey, key, numOfElements, float,
-                CompareKeyFloat32);
-            break;
-        case TYPE_Float64:
-            KEY_QSORT(ptrSortKey, key, numOfElements, double,
-                CompareKeyFloat64);
-            break;
-        default:
-            return -1;
-        }
-
-        // Shuffle into sorted position based on index map.
-        WebRtc_UWord8* ptrData = static_cast<WebRtc_UWord8*>(data);
-        WebRtc_UWord8* ptrDataSorted = new(std::nothrow) WebRtc_UWord8
-            [numOfElements * sizeOfElement];
-        if (ptrDataSorted == NULL)
-        {
-            return -1;
-        }
-
-        for (WebRtc_UWord32 i = 0; i < numOfElements; i++)
-        {
-            memcpy(ptrDataSorted + i * sizeOfElement, ptrData +
-                ptrSortKey[i].index * sizeOfElement, sizeOfElement);
-        }
-        memcpy(ptrData, ptrDataSorted, numOfElements * sizeOfElement);
-
-        delete[] ptrSortKey;
-        delete[] ptrDataSorted;
-
-        return 0;
-#else
-        // Fall back to std::sort for 64-bit types and floats due to compiler
-	// warnings and errors respectively with spreadsort.
-        switch (keyType)
-        {
-        case TYPE_Word8:
-            return IntegerKeySort<WebRtc_Word8>(data, key, numOfElements,
-                                                sizeOfElement);
-        case TYPE_UWord8:
-            return IntegerKeySort<WebRtc_UWord8>(data, key, numOfElements,
-                                                 sizeOfElement);
-        case TYPE_Word16:
-            return IntegerKeySort<WebRtc_Word16>(data, key, numOfElements,
-                                                 sizeOfElement);
-        case TYPE_UWord16:
-            return IntegerKeySort<WebRtc_UWord16>(data, key, numOfElements,
-                                                  sizeOfElement);
-        case TYPE_Word32:
-            return IntegerKeySort<WebRtc_Word32>(data, key, numOfElements,
-                                                 sizeOfElement);
-        case TYPE_UWord32:
-            return IntegerKeySort<WebRtc_UWord32>(data, key, numOfElements,
-                                                  sizeOfElement);
-        case TYPE_Word64:
-            return StdKeySort<WebRtc_Word64>(data, key, numOfElements,
-                                             sizeOfElement);
-        case TYPE_UWord64:
-            return StdKeySort<WebRtc_UWord64>(data, key, numOfElements,
-                                              sizeOfElement);
-        case TYPE_Float32:
-            return StdKeySort<float>(data, key, numOfElements, sizeOfElement);
-        case TYPE_Float64:
-            return StdKeySort<double>(data, key, numOfElements, sizeOfElement);
-        default:
-            return -1;
-        }
-#endif
-    }
-} // namespace webrtc
diff --git a/src/system_wrappers/source/spreadsortlib/constants.hpp b/src/system_wrappers/source/spreadsortlib/constants.hpp
deleted file mode 100644
index fa81ece..0000000
--- a/src/system_wrappers/source/spreadsortlib/constants.hpp
+++ /dev/null
@@ -1,42 +0,0 @@
-/*Boost Software License - Version 1.0 - August 17th, 2003
-
-Permission is hereby granted, free of charge, to any person or organization
-obtaining a copy of the software and accompanying documentation covered by
-this license (the "Software") to use, reproduce, display, distribute,
-execute, and transmit the Software, and to prepare derivative works of the
-Software, and to permit third-parties to whom the Software is furnished to
-do so, all subject to the following:
-
-The copyright notices in the Software and this entire statement, including
-the above license grant, this restriction and the following disclaimer,
-must be included in all copies of the Software, in whole or in part, and
-all derivative works of the Software, unless such copies or derivative
-works are solely in the form of machine-executable object code generated by
-a source language processor.
-
-THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
-IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
-FITNESS FOR A PARTICULAR PURPOSE, TITLE AND NON-INFRINGEMENT. IN NO EVENT
-SHALL THE COPYRIGHT HOLDERS OR ANYONE DISTRIBUTING THE SOFTWARE BE LIABLE
-FOR ANY DAMAGES OR OTHER LIABILITY, WHETHER IN CONTRACT, TORT OR OTHERWISE,
-ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
-DEALINGS IN THE SOFTWARE.*/
-#ifndef BOOST_SPREADSORT_CONSTANTS
-#define BOOST_SPREADSORT_CONSTANTS
-namespace boost {
-namespace detail {
-//Tuning constants
-//Sets the minimum number of items per bin.
-static const unsigned LOG_MEAN_BIN_SIZE = 2;
-//This should be tuned to your processor cache; if you go too large you get cache misses on bins
-//The smaller this number, the less worst-case memory usage.  If too small, too many recursions slow down spreadsort
-static const unsigned MAX_SPLITS = 10;
-//Used to force a comparison-based sorting for small bins, if it's faster.  Minimum value 0
-static const unsigned LOG_MIN_SPLIT_COUNT = 5;
-//There is a minimum size below which it is not worth using spreadsort
-static const long MIN_SORT_SIZE = 1000;
-//This is the constant on the log base n of m calculation; make this larger the faster std::sort is relative to spreadsort
-static const unsigned LOG_CONST = 2;
-}
-}
-#endif
diff --git a/src/system_wrappers/source/spreadsortlib/spreadsort.hpp b/src/system_wrappers/source/spreadsortlib/spreadsort.hpp
deleted file mode 100644
index 2d1529a..0000000
--- a/src/system_wrappers/source/spreadsortlib/spreadsort.hpp
+++ /dev/null
@@ -1,1688 +0,0 @@
-//Templated spread_sort library
-
-//          Copyright Steven J. Ross 2001 - 2009.
-// Distributed under the Boost Software License, Version 1.0.
-//    (See accompanying file LICENSE_1_0.txt or copy at
-//          http://www.boost.org/LICENSE_1_0.txt)
-
-//  See http://www.boost.org/ for updates, documentation, and revision history.
-		  
-/*
-Some improvements suggested by:
-Phil Endecott and Frank Gennari
-Cygwin fix provided by:
-Scott McMurray
-*/
-
-#ifndef BOOST_SPREAD_SORT_H
-#define BOOST_SPREAD_SORT_H
-#include <algorithm>
-#include <vector>
-#include "constants.hpp"
-#include <cstring>
-
-namespace boost {
-  namespace detail {
-  	//This only works on unsigned data types
-  	template <typename T>
-  	inline unsigned 
-  	rough_log_2_size(const T& input) 
-  	{
-  		unsigned result = 0;
-  		//The && is necessary on some compilers to avoid infinite loops; it doesn't significantly impair performance
-  		while((input >> result) && (result < (8*sizeof(T)))) ++result;
-  		return result;
-  	}
-
-  	//Gets the maximum size which we'll call spread_sort on to control worst-case performance
-  	//Maintains both a minimum size to recurse and a check of distribution size versus count
-  	//This is called for a set of bins, instead of bin-by-bin, to avoid performance overhead
-  	inline size_t
-  	get_max_count(unsigned log_range, size_t count)
-  	{
-  		unsigned divisor = rough_log_2_size(count);
-  		//Making sure the divisor is positive
-  		if(divisor > LOG_MEAN_BIN_SIZE)
-  			divisor -= LOG_MEAN_BIN_SIZE;
-  		else
-  			divisor = 1;
-  		unsigned relative_width = (LOG_CONST * log_range)/((divisor > MAX_SPLITS) ? MAX_SPLITS : divisor);
-  		//Don't try to bitshift more than the size of an element
-  		if((8*sizeof(size_t)) <= relative_width)
-  			relative_width = (8*sizeof(size_t)) - 1;
-  		return (size_t)1 << ((relative_width < (LOG_MEAN_BIN_SIZE + LOG_MIN_SPLIT_COUNT)) ? 
-  			(LOG_MEAN_BIN_SIZE + LOG_MIN_SPLIT_COUNT) :  relative_width);
-  	}
-
-  	//Find the minimum and maximum using <
-  	template <class RandomAccessIter>
-  	inline void 
-  	find_extremes(RandomAccessIter current, RandomAccessIter last, RandomAccessIter & max, RandomAccessIter & min)
-  	{
-  		min = max = current;
-  		//Start from the second item, as max and min are initialized to the first
-  		while(++current < last) {
-  			if(*max < *current)
-  				max = current;
-  			else if(*current < *min)
-  				min = current;
-  		}
-  	}
-
-  	//Uses a user-defined comparison operator to find minimum and maximum
-  	template <class RandomAccessIter, class compare>
-  	inline void 
-  	find_extremes(RandomAccessIter current, RandomAccessIter last, RandomAccessIter & max, RandomAccessIter & min, compare comp)
-  	{
-  		min = max = current;
-  		while(++current < last) {
-  			if(comp(*max, *current))
-  				max = current;
-  			else if(comp(*current, *min))
-  				min = current;
-  		}
-  	}
-
-  	//Gets a non-negative right bit shift to operate as a logarithmic divisor
-  	inline int
-  	get_log_divisor(size_t count, unsigned log_range)
-  	{
-  		int log_divisor;
-  		//If we can finish in one iteration without exceeding either (2 to the MAX_SPLITS) or n bins, do so
-  		if((log_divisor = log_range - rough_log_2_size(count)) <= 0 && log_range < MAX_SPLITS)
-  			log_divisor = 0;
-  		else {
-  			//otherwise divide the data into an optimized number of pieces
-  			log_divisor += LOG_MEAN_BIN_SIZE;
-  			if(log_divisor < 0)
-  				log_divisor = 0;
-  			//Cannot exceed MAX_SPLITS or cache misses slow down bin lookups dramatically
-  			if((log_range - log_divisor) > MAX_SPLITS)
-  				log_divisor = log_range - MAX_SPLITS;
-  		}
-  		return log_divisor;
-  	}
-
-  	template <class RandomAccessIter>
-  	inline RandomAccessIter * 
-  	size_bins(std::vector<size_t> &bin_sizes, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset, unsigned &cache_end, unsigned bin_count)
-  	{
-  		//Assure space for the size of each bin, followed by initializing sizes
-  		if(bin_count > bin_sizes.size())
-  			bin_sizes.resize(bin_count);
-  		for(size_t u = 0; u < bin_count; u++)
-  			bin_sizes[u] = 0;
-  		//Make sure there is space for the bins
-  		cache_end = cache_offset + bin_count;
-  		if(cache_end > bin_cache.size())
-  			bin_cache.resize(cache_end);
-  		return &(bin_cache[cache_offset]);
-  	}
-
-  	//Implementation for recursive integer sorting
-  	template <class RandomAccessIter, class div_type, class data_type>
-  	inline void 
-  	spread_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
-  				  , std::vector<size_t> &bin_sizes)
-  	{
-  		//This step is roughly 10% of runtime, but it helps avoid worst-case behavior and improve behavior with real data
-  		//If you know the maximum and minimum ahead of time, you can pass those values in and skip this step for the first iteration
-  		RandomAccessIter max, min;
-  		find_extremes(first, last, max, min);
-  		//max and min will be the same (the first item) iff all values are equivalent
-  		if(max == min)
-  			return;
-  		RandomAccessIter * target_bin;
-  		unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(*max >> 0) - (*min >> 0)));
-  		div_type div_min = *min >> log_divisor;
-  		div_type div_max = *max >> log_divisor;
-  		unsigned bin_count = div_max - div_min + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-  	
-  		//Calculating the size of each bin; this takes roughly 10% of runtime
-  		for (RandomAccessIter current = first; current != last;)
-  			bin_sizes[(*(current++) >> log_divisor) - div_min]++;
-  		//Assign the bin positions
-  		bins[0] = first;
-  		for(unsigned u = 0; u < bin_count - 1; u++)
-  			bins[u + 1] = bins[u] + bin_sizes[u];
-  
-  		//Swap into place
-  		//This dominates runtime, mostly in the swap and bin lookups
-  		RandomAccessIter nextbinstart = first;
-  		for(unsigned u = 0; u < bin_count - 1; ++u) {
-  			RandomAccessIter * local_bin = bins + u;
-  			nextbinstart += bin_sizes[u];
-  			//Iterating over each element in this bin
-  			for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  				//Swapping elements in current into place until the correct element has been swapped in
-  				for(target_bin = (bins + ((*current >> log_divisor) - div_min));  target_bin != local_bin; 
-  					target_bin = bins + ((*current >> log_divisor) - div_min)) {
-  					//3-way swap; this is about 1% faster than a 2-way swap with integers
-  					//The main advantage is less copies are involved per item put in the correct place
-  					data_type tmp;
-  					RandomAccessIter b = (*target_bin)++;
-  					RandomAccessIter * b_bin = bins + ((*b >> log_divisor) - div_min);
-  					if (b_bin != local_bin) {
-  						RandomAccessIter c = (*b_bin)++;
-  						tmp = *c;
-  						*c = *b;
-  					} 
-  					else
-  						tmp = *b;
-  					*b = *current;
-  					*current = tmp;
-  				}
-  			}
-  			*local_bin = nextbinstart;
-  		}
-  		bins[bin_count - 1] = last;
-  
-  		//If we've bucketsorted, the array is sorted and we should skip recursion
-  		if(!log_divisor)
-  			return;
-  
-  		//Recursing; log_divisor is the remaining range
-  		size_t max_count = get_max_count(log_divisor, last - first);
-  		RandomAccessIter lastPos = first;
-  		for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			//don't sort unless there are at least two items to compare
-  			if(count < 2)
-  				continue;
-  			//using std::sort if its worst-case is better
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[u]);
-  			else
-  				spread_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes);
-  		}
-  	}
-
-  	//Generic bitshift-based 3-way swapping code
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift>
-  	inline void inner_swap_loop(RandomAccessIter * bins, const RandomAccessIter & nextbinstart, unsigned ii, right_shift &shift
-  		, const unsigned log_divisor, const div_type div_min) 
-  	{
-  		RandomAccessIter * local_bin = bins + ii;
-  		for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  			for(RandomAccessIter * target_bin = (bins + (shift(*current, log_divisor) - div_min));  target_bin != local_bin; 
-  				target_bin = bins + (shift(*current, log_divisor) - div_min)) {
-  				data_type tmp;
-  				RandomAccessIter b = (*target_bin)++;
-  				RandomAccessIter * b_bin = bins + (shift(*b, log_divisor) - div_min);
-  				//Three-way swap; if the item to be swapped doesn't belong in the current bin, swap it to where it belongs
-  				if (b_bin != local_bin) {
-  					RandomAccessIter c = (*b_bin)++;
-  					tmp = *c;
-  					*c = *b;
-  				} 
-  				//Note: we could increment current once the swap is done in this case, but that seems to impair performance
-  				else
-  					tmp = *b;
-  				*b = *current;
-  				*current = tmp;
-  			}
-  		}
-  		*local_bin = nextbinstart;
-  	}
-
-  	//Standard swapping wrapper for ascending values
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift>
-  	inline void swap_loop(RandomAccessIter * bins, RandomAccessIter & nextbinstart, unsigned ii, right_shift &shift
-  		, const std::vector<size_t> &bin_sizes, const unsigned log_divisor, const div_type div_min) 
-  	{
-  		nextbinstart += bin_sizes[ii];
-  		inner_swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, log_divisor, div_min);
-  	}
-
-  	//Functor implementation for recursive sorting
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
-  	inline void 
-  	spread_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
-  					, std::vector<size_t> &bin_sizes, right_shift shift, compare comp)
-  	{
-  		RandomAccessIter max, min;
-  		find_extremes(first, last, max, min, comp);
-  		if(max == min)
-  			return;
-  		unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(shift(*max, 0)) - (shift(*min, 0))));
-  		div_type div_min = shift(*min, log_divisor);
-  		div_type div_max = shift(*max, log_divisor);
-  		unsigned bin_count = div_max - div_min + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-  			
-  		//Calculating the size of each bin
-  		for (RandomAccessIter current = first; current != last;)
-  			bin_sizes[shift(*(current++), log_divisor) - div_min]++;
-  		bins[0] = first;
-  		for(unsigned u = 0; u < bin_count - 1; u++)
-  			bins[u + 1] = bins[u] + bin_sizes[u];
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		for(unsigned u = 0; u < bin_count - 1; ++u)
-  			swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, u, shift, bin_sizes, log_divisor, div_min);
-  		bins[bin_count - 1] = last;
-  		
-  		//If we've bucketsorted, the array is sorted and we should skip recursion
-  		if(!log_divisor)
-  			return;
-  		
-  		//Recursing
-  		size_t max_count = get_max_count(log_divisor, last - first);
-  		RandomAccessIter lastPos = first;
-  		for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[u], comp);
-  			else
-  				spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift, compare>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift, comp);
-  		}
-  	}
-
-  	//Functor implementation for recursive sorting with only Shift overridden
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift>
-  	inline void 
-  	spread_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
-  					, std::vector<size_t> &bin_sizes, right_shift shift)
-  	{
-  		RandomAccessIter max, min;
-  		find_extremes(first, last, max, min);
-  		if(max == min)
-  			return;
-  		unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(shift(*max, 0)) - (shift(*min, 0))));
-  		div_type div_min = shift(*min, log_divisor);
-  		div_type div_max = shift(*max, log_divisor);
-  		unsigned bin_count = div_max - div_min + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-  			
-  		//Calculating the size of each bin
-  		for (RandomAccessIter current = first; current != last;)
-  			bin_sizes[shift(*(current++), log_divisor) - div_min]++;
-  		bins[0] = first;
-  		for(unsigned u = 0; u < bin_count - 1; u++)
-  			bins[u + 1] = bins[u] + bin_sizes[u];
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		for(unsigned ii = 0; ii < bin_count - 1; ++ii)
-  			swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, bin_sizes, log_divisor, div_min);
-  		bins[bin_count - 1] = last;
-  		
-  		//If we've bucketsorted, the array is sorted and we should skip recursion
-  		if(!log_divisor)
-  			return;
-  		
-  		//Recursing
-  		size_t max_count = get_max_count(log_divisor, last - first);
-  		RandomAccessIter lastPos = first;
-  		for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[u]);
-  			else
-  				spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift);
-  		}
-  	}
-
-  	//Holds the bin vector and makes the initial recursive call
-  	template <class RandomAccessIter, class div_type, class data_type>
-  	inline void 
-  	spread_sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type)
-  	{
-  		std::vector<size_t> bin_sizes;
-  		std::vector<RandomAccessIter> bin_cache;
-  		spread_sort_rec<RandomAccessIter, div_type, data_type>(first, last, bin_cache, 0, bin_sizes);
-  	}
-
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
-  	inline void 
-  	spread_sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift, compare comp)
-  	{
-  		std::vector<size_t> bin_sizes;
-  		std::vector<RandomAccessIter> bin_cache;
-  		spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift, compare>(first, last, bin_cache, 0, bin_sizes, shift, comp);
-  	}
-
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift>
-  	inline void 
-  	spread_sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift)
-  	{
-  		std::vector<size_t> bin_sizes;
-  		std::vector<RandomAccessIter> bin_cache;
-  		spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(first, last, bin_cache, 0, bin_sizes, shift);
-  	}
-  }
-
-  //Top-level sorting call for integers
-  template <class RandomAccessIter>
-  inline void integer_sort(RandomAccessIter first, RandomAccessIter last) 
-  {
-  	//Don't sort if it's too small to optimize
-  	if(last - first < detail::MIN_SORT_SIZE)
-  		std::sort(first, last);
-  	else
-  		detail::spread_sort(first, last, *first >> 0, *first);
-  }
-
-  //integer_sort with functors
-  template <class RandomAccessIter, class right_shift, class compare>
-  inline void integer_sort(RandomAccessIter first, RandomAccessIter last,
-  						right_shift shift, compare comp) {
-  	if(last - first < detail::MIN_SORT_SIZE)
-  		std::sort(first, last, comp);
-  	else
-  		detail::spread_sort(first, last, shift(*first, 0), *first, shift, comp);
-  }
-
-  //integer_sort with right_shift functor
-  template <class RandomAccessIter, class right_shift>
-  inline void integer_sort(RandomAccessIter first, RandomAccessIter last,
-  						right_shift shift) {
-  	if(last - first < detail::MIN_SORT_SIZE)
-  		std::sort(first, last);
-  	else
-  		detail::spread_sort(first, last, shift(*first, 0), *first, shift);
-  }
-
-  //------------------------------------------------------ float_sort source --------------------------------------
-  //Casts a RandomAccessIter to the specified data type
-  template<class cast_type, class RandomAccessIter>
-  inline cast_type
-  cast_float_iter(const RandomAccessIter & floatiter)
-  {
-  	cast_type result;
-  	std::memcpy(&result, &(*floatiter), sizeof(cast_type));
-  	return result;
-  }
-
-  //Casts a data element to the specified datinner_float_a type
-  template<class data_type, class cast_type>
-  inline cast_type
-  mem_cast(const data_type & data)
-  {
-  	cast_type result;
-  	std::memcpy(&result, &data, sizeof(cast_type));
-  	return result;
-  }
-
-  namespace detail {
-  	template <class RandomAccessIter, class div_type, class right_shift>
-  	inline void 
-  	find_extremes(RandomAccessIter current, RandomAccessIter last, div_type & max, div_type & min, right_shift shift)
-  	{
-  		min = max = shift(*current, 0);
-  		while(++current < last) {
-  			div_type value = shift(*current, 0);
-  			if(max < value)
-  				max = value;
-  			else if(value < min)
-  				min = value;
-  		}
-  	}
-
-  	//Specialized swap loops for floating-point casting
-  	template <class RandomAccessIter, class div_type, class data_type>
-  	inline void inner_float_swap_loop(RandomAccessIter * bins, const RandomAccessIter & nextbinstart, unsigned ii
-  		, const unsigned log_divisor, const div_type div_min) 
-  	{
-  		RandomAccessIter * local_bin = bins + ii;
-  		for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  			for(RandomAccessIter * target_bin = (bins + ((cast_float_iter<div_type, RandomAccessIter>(current) >> log_divisor) - div_min));  target_bin != local_bin; 
-  				target_bin = bins + ((cast_float_iter<div_type, RandomAccessIter>(current) >> log_divisor) - div_min)) {
-  				data_type tmp;
-  				RandomAccessIter b = (*target_bin)++;
-  				RandomAccessIter * b_bin = bins + ((cast_float_iter<div_type, RandomAccessIter>(b) >> log_divisor) - div_min);
-  				//Three-way swap; if the item to be swapped doesn't belong in the current bin, swap it to where it belongs
-  				if (b_bin != local_bin) {
-  					RandomAccessIter c = (*b_bin)++;
-  					tmp = *c;
-  					*c = *b;
-  				} 
-  				else
-  					tmp = *b;
-  				*b = *current;
-  				*current = tmp;
-  			}
-  		}
-  		*local_bin = nextbinstart;
-  	}
-
-  	template <class RandomAccessIter, class div_type, class data_type>
-  	inline void float_swap_loop(RandomAccessIter * bins, RandomAccessIter & nextbinstart, unsigned ii
-  		, const std::vector<size_t> &bin_sizes, const unsigned log_divisor, const div_type div_min) 
-  	{
-  		nextbinstart += bin_sizes[ii];
-  		inner_float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, ii, log_divisor, div_min);
-  	}
-
-  	template <class RandomAccessIter, class cast_type>
-  	inline void 
-  	find_extremes(RandomAccessIter current, RandomAccessIter last, cast_type & max, cast_type & min)
-  	{
-  		min = max = cast_float_iter<cast_type, RandomAccessIter>(current);
-  		while(++current < last) {
-  			cast_type value = cast_float_iter<cast_type, RandomAccessIter>(current);
-  			if(max < value)
-  				max = value;
-  			else if(value < min)
-  				min = value;
-  		}
-  	}
-
-  	//Special-case sorting of positive floats with casting instead of a right_shift
-  	template <class RandomAccessIter, class div_type, class data_type>
-  	inline void 
-  	positive_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
-  					, std::vector<size_t> &bin_sizes)
-  	{
-  		div_type max, min;
-  		find_extremes(first, last, max, min);
-  		if(max == min)
-  			return;
-  		unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
-  		div_type div_min = min >> log_divisor;
-  		div_type div_max = max >> log_divisor;
-  		unsigned bin_count = div_max - div_min + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-  			
-  		//Calculating the size of each bin
-  		for (RandomAccessIter current = first; current != last;)
-  			bin_sizes[(cast_float_iter<div_type, RandomAccessIter>(current++) >> log_divisor) - div_min]++;
-  		bins[0] = first;
-  		for(unsigned u = 0; u < bin_count - 1; u++)
-  			bins[u + 1] = bins[u] + bin_sizes[u];
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		for(unsigned u = 0; u < bin_count - 1; ++u)
-  			float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, u, bin_sizes, log_divisor, div_min);
-  		bins[bin_count - 1] = last;
-  		
-  		//Return if we've completed bucketsorting
-  		if(!log_divisor)
-  			return;
-  		
-  		//Recursing
-  		size_t max_count = get_max_count(log_divisor, last - first);
-  		RandomAccessIter lastPos = first;
-  		for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[u]);
-  			else
-  				positive_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes);
-  		}
-  	}
-
-  	//Sorting negative_ float_s
-  	//Note that bins are iterated in reverse order because max_neg_float = min_neg_int
-  	template <class RandomAccessIter, class div_type, class data_type>
-  	inline void 
-  	negative_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
-  					, std::vector<size_t> &bin_sizes)
-  	{
-  		div_type max, min;
-  		find_extremes(first, last, max, min);
-  		if(max == min)
-  			return;
-  		unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
-  		div_type div_min = min >> log_divisor;
-  		div_type div_max = max >> log_divisor;
-  		unsigned bin_count = div_max - div_min + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-  			
-  		//Calculating the size of each bin
-  		for (RandomAccessIter current = first; current != last;)
-  			bin_sizes[(cast_float_iter<div_type, RandomAccessIter>(current++) >> log_divisor) - div_min]++;
-  		bins[bin_count - 1] = first;
-  		for(int ii = bin_count - 2; ii >= 0; --ii)
-  			bins[ii] = bins[ii + 1] + bin_sizes[ii + 1];
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		//The last bin will always have the correct elements in it
-  		for(int ii = bin_count - 1; ii > 0; --ii)
-  			float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, ii, bin_sizes, log_divisor, div_min);
-  		//Since we don't process the last bin, we need to update its end position
-  		bin_cache[cache_offset] = last;
-  		
-  		//Return if we've completed bucketsorting
-  		if(!log_divisor)
-  			return;
-  		
-  		//Recursing
-  		size_t max_count = get_max_count(log_divisor, last - first);
-  		RandomAccessIter lastPos = first;
-  		for(int ii = cache_end - 1; ii >= (int)cache_offset; lastPos = bin_cache[ii], --ii) {
-  			size_t count = bin_cache[ii] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[ii]);
-  			else
-  				negative_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes);
-  		}
-  	}
-
-  	//Sorting negative_ float_s
-  	//Note that bins are iterated in reverse order because max_neg_float = min_neg_int
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift>
-  	inline void 
-  	negative_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
-  					, std::vector<size_t> &bin_sizes, right_shift shift)
-  	{
-  		div_type max, min;
-  		find_extremes(first, last, max, min, shift);
-  		if(max == min)
-  			return;
-  		unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
-  		div_type div_min = min >> log_divisor;
-  		div_type div_max = max >> log_divisor;
-  		unsigned bin_count = div_max - div_min + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-  			
-  		//Calculating the size of each bin
-  		for (RandomAccessIter current = first; current != last;)
-  			bin_sizes[shift(*(current++), log_divisor) - div_min]++;
-  		bins[bin_count - 1] = first;
-  		for(int ii = bin_count - 2; ii >= 0; --ii)
-  			bins[ii] = bins[ii + 1] + bin_sizes[ii + 1];
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		//The last bin will always have the correct elements in it
-  		for(int ii = bin_count - 1; ii > 0; --ii)
-  			swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, bin_sizes, log_divisor, div_min);
-  		//Since we don't process the last bin, we need to update its end position
-  		bin_cache[cache_offset] = last;
-  		
-  		//Return if we've completed bucketsorting
-  		if(!log_divisor)
-  			return;
-  		
-  		//Recursing
-  		size_t max_count = get_max_count(log_divisor, last - first);
-  		RandomAccessIter lastPos = first;
-  		for(int ii = cache_end - 1; ii >= (int)cache_offset; lastPos = bin_cache[ii], --ii) {
-  			size_t count = bin_cache[ii] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[ii]);
-  			else
-  				negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift);
-  		}
-  	}
-
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
-  	inline void 
-  	negative_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
-  					, std::vector<size_t> &bin_sizes, right_shift shift, compare comp)
-  	{
-  		div_type max, min;
-  		find_extremes(first, last, max, min, shift);
-  		if(max == min)
-  			return;
-  		unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
-  		div_type div_min = min >> log_divisor;
-  		div_type div_max = max >> log_divisor;
-  		unsigned bin_count = div_max - div_min + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-  			
-  		//Calculating the size of each bin
-  		for (RandomAccessIter current = first; current != last;)
-  			bin_sizes[shift(*(current++), log_divisor) - div_min]++;
-  		bins[bin_count - 1] = first;
-  		for(int ii = bin_count - 2; ii >= 0; --ii)
-  			bins[ii] = bins[ii + 1] + bin_sizes[ii + 1];
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		//The last bin will always have the correct elements in it
-  		for(int ii = bin_count - 1; ii > 0; --ii)
-  			swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, bin_sizes, log_divisor, div_min);
-  		//Since we don't process the last bin, we need to update its end position
-  		bin_cache[cache_offset] = last;
-  		
-  		//Return if we've completed bucketsorting
-  		if(!log_divisor)
-  			return;
-  		
-  		//Recursing
-  		size_t max_count = get_max_count(log_divisor, last - first);
-  		RandomAccessIter lastPos = first;
-  		for(int ii = cache_end - 1; ii >= (int)cache_offset; lastPos = bin_cache[ii], --ii) {
-  			size_t count = bin_cache[ii] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[ii], comp);
-  			else
-  				negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift, compare>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift, comp);
-  		}
-  	}
-
-  	//Casting special-case for floating-point sorting
-  	template <class RandomAccessIter, class div_type, class data_type>
-  	inline void 
-  	float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
-  					, std::vector<size_t> &bin_sizes)
-  	{
-  		div_type max, min;
-  		find_extremes(first, last, max, min);
-  		if(max == min)
-  			return;
-  		unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
-  		div_type div_min = min >> log_divisor;
-  		div_type div_max = max >> log_divisor;
-  		unsigned bin_count = div_max - div_min + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-  			
-  		//Calculating the size of each bin
-  		for (RandomAccessIter current = first; current != last;)
-  			bin_sizes[(cast_float_iter<div_type, RandomAccessIter>(current++) >> log_divisor) - div_min]++;
-  		//The index of the first positive bin
-  		div_type first_positive = (div_min < 0) ? -div_min : 0;
-  		//Resetting if all bins are negative
-  		if(cache_offset + first_positive > cache_end)
-  			first_positive = cache_end - cache_offset;
-  		//Reversing the order of the negative bins
-  		//Note that because of the negative/positive ordering direction flip
-  		//We can not depend upon bin order and positions matching up
-  		//so bin_sizes must be reused to contain the end of the bin
-  		if(first_positive > 0) {
-  			bins[first_positive - 1] = first;
-  			for(int ii = first_positive - 2; ii >= 0; --ii) {
-  				bins[ii] = first + bin_sizes[ii + 1];
-  				bin_sizes[ii] += bin_sizes[ii + 1];
-  			}
-  			//Handling positives following negatives
-  			if((unsigned)first_positive < bin_count) {
-  				bins[first_positive] = first + bin_sizes[0];
-  				bin_sizes[first_positive] += bin_sizes[0];
-  			}
-  		}
-  		else
-  			bins[0] = first;
-  		for(unsigned u = first_positive; u < bin_count - 1; u++) {
-  			bins[u + 1] = first + bin_sizes[u];
-  			bin_sizes[u + 1] += bin_sizes[u];
-  		}
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		for(unsigned u = 0; u < bin_count; ++u) {
-  			nextbinstart = first + bin_sizes[u];
-  			inner_float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, u, log_divisor, div_min);
-  		}
-  		
-  		if(!log_divisor)
-  			return;
-  		
-  		//Handling negative values first
-  		size_t max_count = get_max_count(log_divisor, last - first);
-  		RandomAccessIter lastPos = first;
-  		for(int ii = cache_offset + first_positive - 1; ii >= (int)cache_offset ; lastPos = bin_cache[ii--]) {
-  			size_t count = bin_cache[ii] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[ii]);
-  			//sort negative values using reversed-bin spread_sort
-  			else
-  				negative_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes);
-  		}
-  		
-  		for(unsigned u = cache_offset + first_positive; u < cache_end; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[u]);
-  			//sort positive values using normal spread_sort
-  			else
-  				positive_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes);
-  		}
-  	}
-
-  	//Functor implementation for recursive sorting
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift>
-  	inline void 
-  	float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
-  					, std::vector<size_t> &bin_sizes, right_shift shift)
-  	{
-  		div_type max, min;
-  		find_extremes(first, last, max, min, shift);
-  		if(max == min)
-  			return;
-  		unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
-  		div_type div_min = min >> log_divisor;
-  		div_type div_max = max >> log_divisor;
-  		unsigned bin_count = div_max - div_min + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-  			
-  		//Calculating the size of each bin
-  		for (RandomAccessIter current = first; current != last;)
-  			bin_sizes[shift(*(current++), log_divisor) - div_min]++;
-  		//The index of the first positive bin
-  		div_type first_positive = (div_min < 0) ? -div_min : 0;
-  		//Resetting if all bins are negative
-  		if(cache_offset + first_positive > cache_end)
-  			first_positive = cache_end - cache_offset;
-  		//Reversing the order of the negative bins
-  		//Note that because of the negative/positive ordering direction flip
-  		//We can not depend upon bin order and positions matching up
-  		//so bin_sizes must be reused to contain the end of the bin
-  		if(first_positive > 0) {
-  			bins[first_positive - 1] = first;
-  			for(int ii = first_positive - 2; ii >= 0; --ii) {
-  				bins[ii] = first + bin_sizes[ii + 1];
-  				bin_sizes[ii] += bin_sizes[ii + 1];
-  			}
-  			//Handling positives following negatives
-  			if((unsigned)first_positive < bin_count) {
-  				bins[first_positive] = first + bin_sizes[0];
-  				bin_sizes[first_positive] += bin_sizes[0];
-  			}
-  		}
-  		else
-  			bins[0] = first;
-  		for(unsigned u = first_positive; u < bin_count - 1; u++) {
-  			bins[u + 1] = first + bin_sizes[u];
-  			bin_sizes[u + 1] += bin_sizes[u];
-  		}
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		for(unsigned u = 0; u < bin_count; ++u) {
-  			nextbinstart = first + bin_sizes[u];
-  			inner_swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, u, shift, log_divisor, div_min);
-  		}
-  		
-  		//Return if we've completed bucketsorting
-  		if(!log_divisor)
-  			return;
-  		
-  		//Handling negative values first
-  		size_t max_count = get_max_count(log_divisor, last - first);
-  		RandomAccessIter lastPos = first;
-  		for(int ii = cache_offset + first_positive - 1; ii >= (int)cache_offset ; lastPos = bin_cache[ii--]) {
-  			size_t count = bin_cache[ii] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[ii]);
-  			//sort negative values using reversed-bin spread_sort
-  			else
-  				negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift);
-  		}
-  		
-  		for(unsigned u = cache_offset + first_positive; u < cache_end; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[u]);
-  			//sort positive values using normal spread_sort
-  			else
-  				spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift);
-  		}
-  	}
-
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
-  	inline void 
-  	float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
-  					, std::vector<size_t> &bin_sizes, right_shift shift, compare comp)
-  	{
-  		div_type max, min;
-  		find_extremes(first, last, max, min, shift);
-  		if(max == min)
-  			return;
-  		unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
-  		div_type div_min = min >> log_divisor;
-  		div_type div_max = max >> log_divisor;
-  		unsigned bin_count = div_max - div_min + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-  			
-  		//Calculating the size of each bin
-  		for (RandomAccessIter current = first; current != last;)
-  			bin_sizes[shift(*(current++), log_divisor) - div_min]++;
-  		//The index of the first positive bin
-  		div_type first_positive = (div_min < 0) ? -div_min : 0;
-  		//Resetting if all bins are negative
-  		if(cache_offset + first_positive > cache_end)
-  			first_positive = cache_end - cache_offset;
-  		//Reversing the order of the negative bins
-  		//Note that because of the negative/positive ordering direction flip
-  		//We can not depend upon bin order and positions matching up
-  		//so bin_sizes must be reused to contain the end of the bin
-  		if(first_positive > 0) {
-  			bins[first_positive - 1] = first;
-  			for(int ii = first_positive - 2; ii >= 0; --ii) {
-  				bins[ii] = first + bin_sizes[ii + 1];
-  				bin_sizes[ii] += bin_sizes[ii + 1];
-  			}
-  			//Handling positives following negatives
-  			if((unsigned)first_positive < bin_count) {
-  				bins[first_positive] = first + bin_sizes[0];
-  				bin_sizes[first_positive] += bin_sizes[0];
-  			}
-  		}
-  		else
-  			bins[0] = first;
-  		for(unsigned u = first_positive; u < bin_count - 1; u++) {
-  			bins[u + 1] = first + bin_sizes[u];
-  			bin_sizes[u + 1] += bin_sizes[u];
-  		}
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		for(unsigned u = 0; u < bin_count; ++u) {
-  			nextbinstart = first + bin_sizes[u];
-  			inner_swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, u, shift, log_divisor, div_min);
-  		}
-  		
-  		//Return if we've completed bucketsorting
-  		if(!log_divisor)
-  			return;
-  		
-  		//Handling negative values first
-  		size_t max_count = get_max_count(log_divisor, last - first);
-  		RandomAccessIter lastPos = first;
-  		for(int ii = cache_offset + first_positive - 1; ii >= (int)cache_offset ; lastPos = bin_cache[ii--]) {
-  			size_t count = bin_cache[ii] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[ii]);
-  			//sort negative values using reversed-bin spread_sort
-  			else
-  				negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift, comp);
-  		}
-  		
-  		for(unsigned u = cache_offset + first_positive; u < cache_end; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			if(count < 2)
-  				continue;
-  			if(count < max_count)
-  				std::sort(lastPos, bin_cache[u]);
-  			//sort positive values using normal spread_sort
-  			else
-  				spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift, comp);
-  		}
-  	}
-
-  	template <class RandomAccessIter, class cast_type, class data_type>
-  	inline void 
-  	float_Sort(RandomAccessIter first, RandomAccessIter last, cast_type, data_type)
-  	{
-  		std::vector<size_t> bin_sizes;
-  		std::vector<RandomAccessIter> bin_cache;
-  		float_sort_rec<RandomAccessIter, cast_type, data_type>(first, last, bin_cache, 0, bin_sizes);
-  	}
-
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift>
-  	inline void 
-  	float_Sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift)
-  	{
-  		std::vector<size_t> bin_sizes;
-  		std::vector<RandomAccessIter> bin_cache;
-  		float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(first, last, bin_cache, 0, bin_sizes, shift);
-  	}
-
-  	template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
-  	inline void 
-  	float_Sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift, compare comp)
-  	{
-  		std::vector<size_t> bin_sizes;
-  		std::vector<RandomAccessIter> bin_cache;
-  		float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(first, last, bin_cache, 0, bin_sizes, shift, comp);
-  	}
-  }
-
-  //float_sort with casting
-  //The cast_type must be equal in size to the data type, and must be a signed integer
-  template <class RandomAccessIter, class cast_type>
-  inline void float_sort_cast(RandomAccessIter first, RandomAccessIter last, cast_type cVal) 
-  {
-  	if(last - first < detail::MIN_SORT_SIZE)
-  		std::sort(first, last);
-  	else
-  		detail::float_Sort(first, last, cVal, *first);
-  }
-
-  //float_sort with casting to an int
-  //Only use this with IEEE floating-point numbers
-  template <class RandomAccessIter>
-  inline void float_sort_cast_to_int(RandomAccessIter first, RandomAccessIter last) 
-  {
-  	int cVal = 0;
-  	float_sort_cast(first, last, cVal);
-  }
-
-  //float_sort with functors
-  template <class RandomAccessIter, class right_shift>
-  inline void float_sort(RandomAccessIter first, RandomAccessIter last, right_shift shift) 
-  {
-  	if(last - first < detail::MIN_SORT_SIZE)
-  		std::sort(first, last);
-  	else
-  		detail::float_Sort(first, last, shift(*first, 0), *first, shift);
-  }
-
-  template <class RandomAccessIter, class right_shift, class compare>
-  inline void float_sort(RandomAccessIter first, RandomAccessIter last, right_shift shift, compare comp) 
-  {
-  	if(last - first < detail::MIN_SORT_SIZE)
-  		std::sort(first, last, comp);
-  	else
-  		detail::float_Sort(first, last, shift(*first, 0), *first, shift, comp);
-  }
-
-  //------------------------------------------------- string_sort source ---------------------------------------------
-  namespace detail {
-  	//Offsetting on identical characters.  This function works a character at a time for optimal worst-case performance.
-  	template<class RandomAccessIter>
-  	inline void
-  	update_offset(RandomAccessIter first, RandomAccessIter finish, unsigned &char_offset)
-  	{
-  		unsigned nextOffset = char_offset;
-  		bool done = false;
-  		while(!done) {
-  			RandomAccessIter curr = first;
-  			do {
-  				//ignore empties, but if the nextOffset would exceed the length or not match, exit; we've found the last matching character
-  				if((*curr).size() > char_offset && ((*curr).size() <= (nextOffset + 1) || (*curr)[nextOffset] != (*first)[nextOffset])) {
-  					done = true;
-  					break;
-  				}
-  			} while(++curr != finish);
-  			if(!done)
-  				++nextOffset;
-  		} 
-  		char_offset = nextOffset;
-  	}
-
-  	//Offsetting on identical characters.  This function works a character at a time for optimal worst-case performance.
-  	template<class RandomAccessIter, class get_char, class get_length>
-  	inline void
-  	update_offset(RandomAccessIter first, RandomAccessIter finish, unsigned &char_offset, get_char getchar, get_length length)
-  	{
-  		unsigned nextOffset = char_offset;
-  		bool done = false;
-  		while(!done) {
-  			RandomAccessIter curr = first;
-  			do {
-  				//ignore empties, but if the nextOffset would exceed the length or not match, exit; we've found the last matching character
-  				if(length(*curr) > char_offset && (length(*curr) <= (nextOffset + 1) || getchar((*curr), nextOffset) != getchar((*first), nextOffset))) {
-  					done = true;
-  					break;
-  				}
-  			} while(++curr != finish);
-  			if(!done)
-  				++nextOffset;
-  		} 
-  		char_offset = nextOffset;
-  	}
-
-  	//A comparison functor for strings that assumes they are identical up to char_offset
-  	template<class data_type, class unsignedchar_type>
-  	struct offset_lessthan {
-  		offset_lessthan(unsigned char_offset) : fchar_offset(char_offset){}
-  		inline bool operator()(const data_type &x, const data_type &y) const 
-  		{
-  			unsigned minSize = std::min(x.size(), y.size());
-  			for(unsigned u = fchar_offset; u < minSize; ++u) {
-  				if(static_cast<unsignedchar_type>(x[u]) < static_cast<unsignedchar_type>(y[u]))
-  					return true;
-  				else if(static_cast<unsignedchar_type>(y[u]) < static_cast<unsignedchar_type>(x[u]))
-  					return false;
-  			}
-  			return x.size() < y.size();
-  		}
-  		unsigned fchar_offset;
-  	};
-
-  	//A comparison functor for strings that assumes they are identical up to char_offset
-  	template<class data_type, class unsignedchar_type>
-  	struct offset_greaterthan {
-  		offset_greaterthan(unsigned char_offset) : fchar_offset(char_offset){}
-  		inline bool operator()(const data_type &x, const data_type &y) const 
-  		{
-  			unsigned minSize = std::min(x.size(), y.size());
-  			for(unsigned u = fchar_offset; u < minSize; ++u) {
-  				if(static_cast<unsignedchar_type>(x[u]) > static_cast<unsignedchar_type>(y[u]))
-  					return true;
-  				else if(static_cast<unsignedchar_type>(y[u]) > static_cast<unsignedchar_type>(x[u]))
-  					return false;
-  			}
-  			return x.size() > y.size();
-  		}
-  		unsigned fchar_offset;
-  	};
-
-  	//A comparison functor for strings that assumes they are identical up to char_offset
-  	template<class data_type, class get_char, class get_length>
-  	struct offset_char_lessthan {
-  		offset_char_lessthan(unsigned char_offset) : fchar_offset(char_offset){}
-  		inline bool operator()(const data_type &x, const data_type &y) const 
-  		{
-  			unsigned minSize = std::min(length(x), length(y));
-  			for(unsigned u = fchar_offset; u < minSize; ++u) {
-  				if(getchar(x, u) < getchar(y, u))
-  					return true;
-  				else if(getchar(y, u) < getchar(x, u))
-  					return false;
-  			}
-  			return length(x) < length(y);
-  		}
-  		unsigned fchar_offset;
-  		get_char getchar;
-  		get_length length;
-  	};
-
-  	//String sorting recursive implementation
-  	template <class RandomAccessIter, class data_type, class unsignedchar_type>
-  	inline void 
-  	string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
-  		, unsigned cache_offset, std::vector<size_t> &bin_sizes)
-  	{
-  		//This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
-  		//Iterate to the end of the empties.  If all empty, return
-  		while((*first).size() <= char_offset) {
-  			if(++first == last)
-  				return;
-  		}
-  		RandomAccessIter finish = last - 1;
-  		//Getting the last non-empty
-  		for(;(*finish).size() <= char_offset; --finish) { }
-  		++finish;
-  		//Offsetting on identical characters.  This section works a character at a time for optimal worst-case performance.
-  		update_offset(first, finish, char_offset);
-  		
-  		const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
-  		//Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
-  		const unsigned max_size = bin_count;
-  		const unsigned membin_count = bin_count + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count) + 1;
-  			
-  		//Calculating the size of each bin; this takes roughly 10% of runtime
-  		for (RandomAccessIter current = first; current != last; ++current) {
-  			if((*current).size() <= char_offset) {
-  				bin_sizes[0]++;
-  			}
-  			else
-  				bin_sizes[static_cast<unsignedchar_type>((*current)[char_offset]) + 1]++;
-  		}
-  		//Assign the bin positions
-  		bin_cache[cache_offset] = first;
-  		for(unsigned u = 0; u < membin_count - 1; u++)
-  			bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		//handling empty bins
-  		RandomAccessIter * local_bin = &(bin_cache[cache_offset]);
-  		nextbinstart +=	bin_sizes[0];
-  		RandomAccessIter * target_bin;
-  		//Iterating over each element in the bin of empties
-  		for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  			//empties belong in this bin
-  			while((*current).size() > char_offset) {
-  				target_bin = bins + static_cast<unsignedchar_type>((*current)[char_offset]);
-  				iter_swap(current, (*target_bin)++);
-  			}
-  		}
-  		*local_bin = nextbinstart;
-  		//iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
-  		unsigned last_bin = bin_count - 1;
-  		for(; last_bin && !bin_sizes[last_bin + 1]; --last_bin) { }
-  		//This dominates runtime, mostly in the swap and bin lookups
-  		for(unsigned u = 0; u < last_bin; ++u) {
-  			local_bin = bins + u;
-  			nextbinstart += bin_sizes[u + 1];
-  			//Iterating over each element in this bin
-  			for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  				//Swapping elements in current into place until the correct element has been swapped in
-  				for(target_bin = bins + static_cast<unsignedchar_type>((*current)[char_offset]);  target_bin != local_bin; 
-  					target_bin = bins + static_cast<unsignedchar_type>((*current)[char_offset]))
-  					iter_swap(current, (*target_bin)++);
-  			}
-  			*local_bin = nextbinstart;
-  		}
-  		bins[last_bin] = last;
-  		//Recursing
-  		RandomAccessIter lastPos = bin_cache[cache_offset];
-  		//Skip this loop for empties
-  		for(unsigned u = cache_offset + 1; u < cache_offset + last_bin + 2; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			//don't sort unless there are at least two items to compare
-  			if(count < 2)
-  				continue;
-  			//using std::sort if its worst-case is better
-  			if(count < max_size)
-  				std::sort(lastPos, bin_cache[u], offset_lessthan<data_type, unsignedchar_type>(char_offset + 1));
-  			else
-  				string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(lastPos, bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes);
-  		}
-  	}
-
-  	//Sorts strings in reverse order, with empties at the end
-  	template <class RandomAccessIter, class data_type, class unsignedchar_type>
-  	inline void 
-  	reverse_string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
-  		, unsigned cache_offset, std::vector<size_t> &bin_sizes)
-  	{
-  		//This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
-  		RandomAccessIter curr = first;
-  		//Iterate to the end of the empties.  If all empty, return
-  		while((*curr).size() <= char_offset) {
-  			if(++curr == last)
-  				return;
-  		}
-  		//Getting the last non-empty
-  		while((*(--last)).size() <= char_offset) { }
-  		++last;
-  		//Offsetting on identical characters.  This section works a character at a time for optimal worst-case performance.
-  		update_offset(curr, last, char_offset);
-  		RandomAccessIter * target_bin;
-  		
-  		const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
-  		//Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
-  		const unsigned max_size = bin_count;
-  		const unsigned membin_count = bin_count + 1;
-  		const unsigned max_bin = bin_count - 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count);
-  		RandomAccessIter * end_bin = &(bin_cache[cache_offset + max_bin]);
-  			
-  		//Calculating the size of each bin; this takes roughly 10% of runtime
-  		for (RandomAccessIter current = first; current != last; ++current) {
-  			if((*current).size() <= char_offset) {
-  				bin_sizes[bin_count]++;
-  			}
-  			else
-  				bin_sizes[max_bin - static_cast<unsignedchar_type>((*current)[char_offset])]++;
-  		}
-  		//Assign the bin positions
-  		bin_cache[cache_offset] = first;
-  		for(unsigned u = 0; u < membin_count - 1; u++)
-  			bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = last;
-  		//handling empty bins
-  		RandomAccessIter * local_bin = &(bin_cache[cache_offset + bin_count]);
-  		RandomAccessIter lastFull = *local_bin;
-  		//Iterating over each element in the bin of empties
-  		for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  			//empties belong in this bin
-  			while((*current).size() > char_offset) {
-  				target_bin = end_bin - static_cast<unsignedchar_type>((*current)[char_offset]);
-  				iter_swap(current, (*target_bin)++);
-  			}
-  		}
-  		*local_bin = nextbinstart;
-  		nextbinstart = first;
-  		//iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
-  		unsigned last_bin = max_bin;
-  		for(; last_bin && !bin_sizes[last_bin]; --last_bin) { }
-  		//This dominates runtime, mostly in the swap and bin lookups
-  		for(unsigned u = 0; u < last_bin; ++u) {
-  			local_bin = bins + u;
-  			nextbinstart += bin_sizes[u];
-  			//Iterating over each element in this bin
-  			for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  				//Swapping elements in current into place until the correct element has been swapped in
-  				for(target_bin = end_bin - static_cast<unsignedchar_type>((*current)[char_offset]);  target_bin != local_bin; 
-  					target_bin = end_bin - static_cast<unsignedchar_type>((*current)[char_offset]))
-  					iter_swap(current, (*target_bin)++);
-  			}
-  			*local_bin = nextbinstart;
-  		}
-  		bins[last_bin] = lastFull;
-  		//Recursing
-  		RandomAccessIter lastPos = first;
-  		//Skip this loop for empties
-  		for(unsigned u = cache_offset; u <= cache_offset + last_bin; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			//don't sort unless there are at least two items to compare
-  			if(count < 2)
-  				continue;
-  			//using std::sort if its worst-case is better
-  			if(count < max_size)
-  				std::sort(lastPos, bin_cache[u], offset_greaterthan<data_type, unsignedchar_type>(char_offset + 1));
-  			else
-  				reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(lastPos, bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes);
-  		}
-  	}
-
-  	//String sorting recursive implementation
-  	template <class RandomAccessIter, class data_type, class unsignedchar_type, class get_char, class get_length>
-  	inline void 
-  	string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
-  		, unsigned cache_offset, std::vector<size_t> &bin_sizes, get_char getchar, get_length length)
-  	{
-  		//This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
-  		//Iterate to the end of the empties.  If all empty, return
-  		while(length(*first) <= char_offset) {
-  			if(++first == last)
-  				return;
-  		}
-  		RandomAccessIter finish = last - 1;
-  		//Getting the last non-empty
-  		for(;length(*finish) <= char_offset; --finish) { }
-  		++finish;
-  		update_offset(first, finish, char_offset, getchar, length);
-  		
-  		const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
-  		//Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
-  		const unsigned max_size = bin_count;
-  		const unsigned membin_count = bin_count + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count) + 1;
-  			
-  		//Calculating the size of each bin; this takes roughly 10% of runtime
-  		for (RandomAccessIter current = first; current != last; ++current) {
-  			if(length(*current) <= char_offset) {
-  				bin_sizes[0]++;
-  			}
-  			else
-  				bin_sizes[getchar((*current), char_offset) + 1]++;
-  		}
-  		//Assign the bin positions
-  		bin_cache[cache_offset] = first;
-  		for(unsigned u = 0; u < membin_count - 1; u++)
-  			bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		//handling empty bins
-  		RandomAccessIter * local_bin = &(bin_cache[cache_offset]);
-  		nextbinstart +=	bin_sizes[0];
-  		RandomAccessIter * target_bin;
-  		//Iterating over each element in the bin of empties
-  		for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  			//empties belong in this bin
-  			while(length(*current) > char_offset) {
-  				target_bin = bins + getchar((*current), char_offset);
-  				iter_swap(current, (*target_bin)++);
-  			}
-  		}
-  		*local_bin = nextbinstart;
-  		//iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
-  		unsigned last_bin = bin_count - 1;
-  		for(; last_bin && !bin_sizes[last_bin + 1]; --last_bin) { }
-  		//This dominates runtime, mostly in the swap and bin lookups
-  		for(unsigned ii = 0; ii < last_bin; ++ii) {
-  			local_bin = bins + ii;
-  			nextbinstart += bin_sizes[ii + 1];
-  			//Iterating over each element in this bin
-  			for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  				//Swapping elements in current into place until the correct element has been swapped in
-  				for(target_bin = bins + getchar((*current), char_offset);  target_bin != local_bin; 
-  					target_bin = bins + getchar((*current), char_offset))
-  					iter_swap(current, (*target_bin)++);
-  			}
-  			*local_bin = nextbinstart;
-  		}
-  		bins[last_bin] = last;
-  		
-  		//Recursing
-  		RandomAccessIter lastPos = bin_cache[cache_offset];
-  		//Skip this loop for empties
-  		for(unsigned u = cache_offset + 1; u < cache_offset + last_bin + 2; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			//don't sort unless there are at least two items to compare
-  			if(count < 2)
-  				continue;
-  			//using std::sort if its worst-case is better
-  			if(count < max_size)
-  				std::sort(lastPos, bin_cache[u], offset_char_lessthan<data_type, get_char, get_length>(char_offset + 1));
-  			else
-  				string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length>(lastPos, bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes, getchar, length);
-  		}
-  	}
-
-  	//String sorting recursive implementation
-  	template <class RandomAccessIter, class data_type, class unsignedchar_type, class get_char, class get_length, class compare>
-  	inline void 
-  	string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
-  		, unsigned cache_offset, std::vector<size_t> &bin_sizes, get_char getchar, get_length length, compare comp)
-  	{
-  		//This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
-  		//Iterate to the end of the empties.  If all empty, return
-  		while(length(*first) <= char_offset) {
-  			if(++first == last)
-  				return;
-  		}
-  		RandomAccessIter finish = last - 1;
-  		//Getting the last non-empty
-  		for(;length(*finish) <= char_offset; --finish) { }
-  		++finish;
-  		update_offset(first, finish, char_offset, getchar, length);
-  		
-  		const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
-  		//Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
-  		const unsigned max_size = bin_count;
-  		const unsigned membin_count = bin_count + 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count) + 1;
-  			
-  		//Calculating the size of each bin; this takes roughly 10% of runtime
-  		for (RandomAccessIter current = first; current != last; ++current) {
-  			if(length(*current) <= char_offset) {
-  				bin_sizes[0]++;
-  			}
-  			else
-  				bin_sizes[getchar((*current), char_offset) + 1]++;
-  		}
-  		//Assign the bin positions
-  		bin_cache[cache_offset] = first;
-  		for(unsigned u = 0; u < membin_count - 1; u++)
-  			bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = first;
-  		//handling empty bins
-  		RandomAccessIter * local_bin = &(bin_cache[cache_offset]);
-  		nextbinstart +=	bin_sizes[0];
-  		RandomAccessIter * target_bin;
-  		//Iterating over each element in the bin of empties
-  		for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  			//empties belong in this bin
-  			while(length(*current) > char_offset) {
-  				target_bin = bins + getchar((*current), char_offset);
-  				iter_swap(current, (*target_bin)++);
-  			}
-  		}
-  		*local_bin = nextbinstart;
-  		//iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
-  		unsigned last_bin = bin_count - 1;
-  		for(; last_bin && !bin_sizes[last_bin + 1]; --last_bin) { }
-  		//This dominates runtime, mostly in the swap and bin lookups
-  		for(unsigned u = 0; u < last_bin; ++u) {
-  			local_bin = bins + u;
-  			nextbinstart += bin_sizes[u + 1];
-  			//Iterating over each element in this bin
-  			for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  				//Swapping elements in current into place until the correct element has been swapped in
-  				for(target_bin = bins + getchar((*current), char_offset);  target_bin != local_bin; 
-  					target_bin = bins + getchar((*current), char_offset))
-  					iter_swap(current, (*target_bin)++);
-  			}
-  			*local_bin = nextbinstart;
-  		}
-  		bins[last_bin] = last;
-  		
-  		//Recursing
-  		RandomAccessIter lastPos = bin_cache[cache_offset];
-  		//Skip this loop for empties
-  		for(unsigned u = cache_offset + 1; u < cache_offset + last_bin + 2; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			//don't sort unless there are at least two items to compare
-  			if(count < 2)
-  				continue;
-  			//using std::sort if its worst-case is better
-  			if(count < max_size)
-  				std::sort(lastPos, bin_cache[u], comp);
-  			else
-  				string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(lastPos
-  					, bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes, getchar, length, comp);
-  		}
-  	}
-
-  	//Sorts strings in reverse order, with empties at the end
-  	template <class RandomAccessIter, class data_type, class unsignedchar_type, class get_char, class get_length, class compare>
-  	inline void 
-  	reverse_string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
-  		, unsigned cache_offset, std::vector<size_t> &bin_sizes, get_char getchar, get_length length, compare comp)
-  	{
-  		//This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
-  		RandomAccessIter curr = first;
-  		//Iterate to the end of the empties.  If all empty, return
-  		while(length(*curr) <= char_offset) {
-  			if(++curr == last)
-  				return;
-  		}
-  		//Getting the last non-empty
-  		while(length(*(--last)) <= char_offset) { }
-  		++last;
-  		//Offsetting on identical characters.  This section works a character at a time for optimal worst-case performance.
-  		update_offset(first, last, char_offset, getchar, length);
-  		
-  		const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
-  		//Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
-  		const unsigned max_size = bin_count;
-  		const unsigned membin_count = bin_count + 1;
-  		const unsigned max_bin = bin_count - 1;
-  		unsigned cache_end;
-  		RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count);
-  		RandomAccessIter *end_bin = &(bin_cache[cache_offset + max_bin]);
-  			
-  		//Calculating the size of each bin; this takes roughly 10% of runtime
-  		for (RandomAccessIter current = first; current != last; ++current) {
-  			if(length(*current) <= char_offset) {
-  				bin_sizes[bin_count]++;
-  			}
-  			else
-  				bin_sizes[max_bin - getchar((*current), char_offset)]++;
-  		}
-  		//Assign the bin positions
-  		bin_cache[cache_offset] = first;
-  		for(unsigned u = 0; u < membin_count - 1; u++)
-  			bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
-  		
-  		//Swap into place
-  		RandomAccessIter nextbinstart = last;
-  		//handling empty bins
-  		RandomAccessIter * local_bin = &(bin_cache[cache_offset + bin_count]);
-  		RandomAccessIter lastFull = *local_bin;
-  		RandomAccessIter * target_bin;
-  		//Iterating over each element in the bin of empties
-  		for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  			//empties belong in this bin
-  			while(length(*current) > char_offset) {
-  				target_bin = end_bin - getchar((*current), char_offset);
-  				iter_swap(current, (*target_bin)++);
-  			}
-  		}
-  		*local_bin = nextbinstart;
-  		nextbinstart = first;
-  		//iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
-  		unsigned last_bin = max_bin;
-  		for(; last_bin && !bin_sizes[last_bin]; --last_bin) { }
-  		//This dominates runtime, mostly in the swap and bin lookups
-  		for(unsigned u = 0; u < last_bin; ++u) {
-  			local_bin = bins + u;
-  			nextbinstart += bin_sizes[u];
-  			//Iterating over each element in this bin
-  			for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
-  				//Swapping elements in current into place until the correct element has been swapped in
-  				for(target_bin = end_bin - getchar((*current), char_offset);  target_bin != local_bin; 
-  					target_bin = end_bin - getchar((*current), char_offset))
-  					iter_swap(current, (*target_bin)++);
-  			}
-  			*local_bin = nextbinstart;
-  		}
-  		bins[last_bin] = lastFull;
-  		//Recursing
-  		RandomAccessIter lastPos = first;
-  		//Skip this loop for empties
-  		for(unsigned u = cache_offset; u <= cache_offset + last_bin; lastPos = bin_cache[u], ++u) {
-  			size_t count = bin_cache[u] - lastPos;
-  			//don't sort unless there are at least two items to compare
-  			if(count < 2)
-  				continue;
-  			//using std::sort if its worst-case is better
-  			if(count < max_size)
-  				std::sort(lastPos, bin_cache[u], comp);
-  			else
-  				reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(lastPos
-  					, bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes, getchar, length, comp);
-  		}
-  	}
-
-  	//Holds the bin vector and makes the initial recursive call
-  	template <class RandomAccessIter, class data_type, class unsignedchar_type>
-  	inline void 
-  	string_sort(RandomAccessIter first, RandomAccessIter last, data_type, unsignedchar_type)
-  	{
-  		std::vector<size_t> bin_sizes;
-  		std::vector<RandomAccessIter> bin_cache;
-  		string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(first, last, 0, bin_cache, 0, bin_sizes);
-  	}
-
-  	//Holds the bin vector and makes the initial recursive call
-  	template <class RandomAccessIter, class data_type, class unsignedchar_type>
-  	inline void 
-  	reverse_string_sort(RandomAccessIter first, RandomAccessIter last, data_type, unsignedchar_type)
-  	{
-  		std::vector<size_t> bin_sizes;
-  		std::vector<RandomAccessIter> bin_cache;
-  		reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(first, last, 0, bin_cache, 0, bin_sizes);
-  	}
-
-  	//Holds the bin vector and makes the initial recursive call
-  	template <class RandomAccessIter, class get_char, class get_length, class data_type, class unsignedchar_type>
-  	inline void 
-  	string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, data_type, unsignedchar_type)
-  	{
-  		std::vector<size_t> bin_sizes;
-  		std::vector<RandomAccessIter> bin_cache;
-  		string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length>(first, last, 0, bin_cache, 0, bin_sizes, getchar, length);
-  	}
-
-  	//Holds the bin vector and makes the initial recursive call
-  	template <class RandomAccessIter, class get_char, class get_length, class compare, class data_type, class unsignedchar_type>
-  	inline void 
-  	string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp, data_type, unsignedchar_type)
-  	{
-  		std::vector<size_t> bin_sizes;
-  		std::vector<RandomAccessIter> bin_cache;
-  		string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(first, last, 0, bin_cache, 0, bin_sizes, getchar, length, comp);
-  	}
-
-  	//Holds the bin vector and makes the initial recursive call
-  	template <class RandomAccessIter, class get_char, class get_length, class compare, class data_type, class unsignedchar_type>
-  	inline void 
-  	reverse_string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp, data_type, unsignedchar_type)
-  	{
-  		std::vector<size_t> bin_sizes;
-  		std::vector<RandomAccessIter> bin_cache;
-  		reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(first, last, 0, bin_cache, 0, bin_sizes, getchar, length, comp);
-  	}
-  }
-
-  //Allows character-type overloads
-  template <class RandomAccessIter, class unsignedchar_type>
-  inline void string_sort(RandomAccessIter first, RandomAccessIter last, unsignedchar_type unused) 
-  {
-  	//Don't sort if it's too small to optimize
-  	if(last - first < detail::MIN_SORT_SIZE)
-  		std::sort(first, last);
-  	else
-  		detail::string_sort(first, last, *first, unused);
-  }
-
-  //Top-level sorting call; wraps using default of unsigned char
-  template <class RandomAccessIter>
-  inline void string_sort(RandomAccessIter first, RandomAccessIter last) 
-  {
-  	unsigned char unused = '\0';
-  	string_sort(first, last, unused);
-  }
-
-  //Allows character-type overloads
-  template <class RandomAccessIter, class compare, class unsignedchar_type>
-  inline void reverse_string_sort(RandomAccessIter first, RandomAccessIter last, compare comp, unsignedchar_type unused) 
-  {
-  	//Don't sort if it's too small to optimize
-  	if(last - first < detail::MIN_SORT_SIZE)
-  		std::sort(first, last, comp);
-  	else
-  		detail::reverse_string_sort(first, last, *first, unused);
-  }
-
-  //Top-level sorting call; wraps using default of unsigned char
-  template <class RandomAccessIter, class compare>
-  inline void reverse_string_sort(RandomAccessIter first, RandomAccessIter last, compare comp) 
-  {
-  	unsigned char unused = '\0';
-  	reverse_string_sort(first, last, comp, unused);
-  }
-
-  template <class RandomAccessIter, class get_char, class get_length>
-  inline void string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length) 
-  {
-  	//Don't sort if it's too small to optimize
-  	if(last - first < detail::MIN_SORT_SIZE)
-  		std::sort(first, last);
-  	else {
-  		//skipping past empties at the beginning, which allows us to get the character type 
-  		//.empty() is not used so as not to require a user declaration of it
-  		while(!length(*first)) {
-  			if(++first == last)
-  				return;
-  		}
-  		detail::string_sort(first, last, getchar, length, *first, getchar((*first), 0));
-  	}
-  }
-
-  template <class RandomAccessIter, class get_char, class get_length, class compare>
-  inline void string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp) 
-  {
-  	//Don't sort if it's too small to optimize
-  	if(last - first < detail::MIN_SORT_SIZE)
-  		std::sort(first, last, comp);
-  	else {
-  		//skipping past empties at the beginning, which allows us to get the character type 
-  		//.empty() is not used so as not to require a user declaration of it
-  		while(!length(*first)) {
-  			if(++first == last)
-  				return;
-  		}
-  		detail::string_sort(first, last, getchar, length, comp, *first, getchar((*first), 0));
-  	}
-  }
-
-  template <class RandomAccessIter, class get_char, class get_length, class compare>
-  inline void reverse_string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp) 
-  {
-  	//Don't sort if it's too small to optimize
-  	if(last - first < detail::MIN_SORT_SIZE)
-  		std::sort(first, last, comp);
-  	else {
-  		//skipping past empties at the beginning, which allows us to get the character type 
-  		//.empty() is not used so as not to require a user declaration of it
-  		while(!length(*(--last))) {
-  			//Note: if there is just one non-empty, and it's at the beginning, then it's already in sorted order
-  			if(first == last)
-  				return;
-  		}
-  		//making last just after the end of the non-empty part of the array
-  		++last;
-  		detail::reverse_string_sort(first, last, getchar, length, comp, *first, getchar((*first), 0));
-  	}
-  }
-}
-
-#endif
diff --git a/src/system_wrappers/source/system_wrappers.gyp b/src/system_wrappers/source/system_wrappers.gyp
deleted file mode 100644
index 0448941..0000000
--- a/src/system_wrappers/source/system_wrappers.gyp
+++ /dev/null
@@ -1,146 +0,0 @@
-# Copyright (c) 2009 The Chromium Authors. All rights reserved.
-# Use of this source code is governed by a BSD-style license that can be
-# found in the LICENSE file.
-
-# TODO: Rename files to use *_linux.cpp etc. names, to automatically include relevant files. Remove conditions section.
-
-{
-  'includes': [
-    '../../common_settings.gypi', # Common settings
-  ],
-  'targets': [
-    {
-      'target_name': 'system_wrappers',
-      'type': '<(library)',
-      'include_dirs': [
-        'spreadsortlib',
-        '../interface',
-      ],
-      'direct_dependent_settings': {
-        'include_dirs': [
-          '../interface',
-        ],
-      },
-      'sources': [
-        '../interface/aligned_malloc.h',
-        '../interface/atomic32_wrapper.h',
-        '../interface/condition_variable_wrapper.h',
-        '../interface/cpu_wrapper.h',
-        '../interface/cpu_features_wrapper.h',
-        '../interface/critical_section_wrapper.h',
-        '../interface/event_wrapper.h',
-        '../interface/file_wrapper.h',
-        '../interface/list_wrapper.h',
-        '../interface/map_wrapper.h',
-        '../interface/rw_lock_wrapper.h',
-        '../interface/sort.h',
-        '../interface/thread_wrapper.h',
-        '../interface/tick_util.h',
-        '../interface/trace.h',
-        'aligned_malloc.cc',
-        'atomic32.cc',
-        'atomic32_linux.h',
-        'atomic32_mac.h',
-        'atomic32_windows.h',
-        'condition_variable.cc',
-        'condition_variable_linux.h',
-        'condition_variable_windows.h',
-        'cpu.cc',
-        'cpu_linux.h',
-        'cpu_mac.h',
-        'cpu_windows.h',
-        'cpu_features.cc',
-        'critical_section.cc',
-        'critical_section_linux.h',
-        'critical_section_windows.h',
-        'event.cc',
-        'event_linux.h',
-        'event_windows.h',
-        'file_impl.cc',
-        'file_impl.h',
-        'list_no_stl.cc',
-        'map.cc',
-        'rw_lock.cc',
-        'rw_lock_linux.h',
-        'rw_lock_windows.h',
-        'sort.cc',
-        'thread.cc',
-        'thread_linux.h',
-        'thread_windows.h',
-        'trace_impl.cc',
-        'trace_impl.h',
-        'trace_linux.h',
-        'trace_windows.h',
-      ],
-      'conditions': [
-        ['OS=="linux"', {
-          'sources': [
-            'condition_variable_linux.cc',
-            'cpu_linux.cc',
-            'critical_section_linux.cc',
-            'event_linux.cc',
-            'thread_linux.cc',
-            'trace_linux.cc',
-            'rw_lock_linux.cc',
-          ],
-          'link_settings': {
-            'libraries': [
-              '-lrt',
-            ],
-          },
-        }],
-        ['OS=="mac"', {
-          'sources': [
-            'condition_variable_linux.cc',
-            'cpu_mac.cc',
-            'critical_section_linux.cc',
-            'event_linux.cc',
-            'rw_lock_linux.cc',
-            'thread_linux.cc',
-            'trace_linux.cc',
-          ],
-        }],
-        ['OS=="win"', {
-          'sources': [
-            'atomic32_windows.h',
-            'condition_variable_windows.cc',
-            'condition_variable_windows.h',
-            'cpu_windows.cc',
-            'cpu_windows.h',
-            'critical_section_windows.cc',
-            'critical_section_windows.h',
-            'event_windows.cc',
-            'event_windows.h',
-            'rw_lock_windows.cc',
-            'rw_lock_windows.h',
-            'thread_windows.cc',
-            'thread_windows.h',
-            'trace_windows.cc',
-            'trace_windows.h',
-          ],
-          'link_settings': {
-            'libraries': [
-              '-lwinmm.lib',
-            ],
-          },
-        }],
-	    ] # conditions
-    },
-    {
-      'target_name': 'system_wrappersTest',
-      'type': 'executable',
-      'dependencies': [
-        'system_wrappers'
-      ],
-      'sources': [
-        '../test/Test.cpp',
-      ],
-    },
-  ], # targets
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/system_wrappers/source/system_wrappers_tests.gyp b/src/system_wrappers/source/system_wrappers_tests.gyp
deleted file mode 100644
index 856f0c1..0000000
--- a/src/system_wrappers/source/system_wrappers_tests.gyp
+++ /dev/null
@@ -1,37 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS.  All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
-  'includes': [
-    '../../common_settings.gypi',
-  ],
-  'targets': [
-    {
-      'target_name': 'unittest',
-      'type': 'executable',
-      'dependencies': [
-        '../../system_wrappers/source/system_wrappers.gyp:system_wrappers',
-        '../../../testing/gtest.gyp:gtest',
-        '../../../testing/gtest.gyp:gtest_main',
-      ],
-      'include_dirs': [
-        '../../../testing/gtest/include',
-      ],
-      'sources': [
-        'list_unittest.cc',
-        'map_unittest.cc',
-      ],
-    },
-  ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/src/system_wrappers/source/thread.cc b/src/system_wrappers/source/thread.cc
deleted file mode 100644
index a136cba..0000000
--- a/src/system_wrappers/source/thread.cc
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "thread_wrapper.h"
-
-#if defined(_WIN32)
-    #include "thread_windows.h"
-#else
-    #include "thread_linux.h"
-#endif
-
-namespace webrtc {
-ThreadWrapper* ThreadWrapper::CreateThread(ThreadRunFunction func,
-                                           ThreadObj obj, ThreadPriority prio,
-                                           const char* threadName)
-{
-#if defined(_WIN32)
-    return new ThreadWindows(func, obj, prio, threadName);
-#else
-    return ThreadLinux::Create(func, obj, prio, threadName);
-#endif
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/thread_linux.cc b/src/system_wrappers/source/thread_linux.cc
deleted file mode 100644
index 1281c1b..0000000
--- a/src/system_wrappers/source/thread_linux.cc
+++ /dev/null
@@ -1,340 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "thread_linux.h"
-
-#include <errno.h>
-#include <string.h> // strncpy
-#include <time.h>   // nanosleep
-#include <unistd.h>
-#ifdef WEBRTC_LINUX
-#include <sys/types.h>
-#include <sched.h>
-#include <sys/syscall.h>
-#include <linux/unistd.h>
-#include <sys/prctl.h>
-#endif
-
-#include "event_wrapper.h"
-#include "trace.h"
-
-namespace webrtc {
-extern "C"
-{
-    static void* StartThread(void* lpParameter)
-    {
-        static_cast<ThreadLinux*>(lpParameter)->Run();
-        return 0;
-    }
-}
-
-#if (defined(WEBRTC_LINUX) && !defined(ANDROID))
-static pid_t gettid()
-{
-#if defined(__NR_gettid)
-    return  syscall(__NR_gettid);
-#else
-    return -1;
-#endif
-}
-#endif
-
-ThreadWrapper* ThreadLinux::Create(ThreadRunFunction func, ThreadObj obj,
-                                   ThreadPriority prio, const char* threadName)
-{
-    ThreadLinux* ptr = new ThreadLinux(func, obj, prio, threadName);
-    if (!ptr)
-    {
-        return NULL;
-    }
-    const int error = ptr->Construct();
-    if (error)
-    {
-        delete ptr;
-        return NULL;
-    }
-    return ptr;
-}
-
-ThreadLinux::ThreadLinux(ThreadRunFunction func, ThreadObj obj,
-                         ThreadPriority prio, const char* threadName)
-    : _runFunction(func),
-      _obj(obj),
-      _alive(false),
-      _dead(true),
-      _prio(prio),
-      _event(EventWrapper::Create()),
-      _setThreadName(false)
-{
-#ifdef WEBRTC_LINUX
-    _linuxPid = -1;
-#endif
-    if (threadName != NULL)
-    {
-        _setThreadName = true;
-        strncpy(_name, threadName, kThreadMaxNameLength);
-    }
-}
-
-int ThreadLinux::Construct()
-{
-    int result = 0;
-#if !defined(ANDROID)
-    // Enable immediate cancellation if requested, see Shutdown()
-    result = pthread_setcancelstate(PTHREAD_CANCEL_ENABLE, NULL);
-    if (result != 0)
-    {
-        return -1;
-    }
-    result = pthread_setcanceltype(PTHREAD_CANCEL_ASYNCHRONOUS, NULL);
-    if (result != 0)
-    {
-        return -1;
-    }
-#endif
-    result = pthread_attr_init(&_attr);
-    if (result != 0)
-    {
-        return -1;
-    }
-
-    return 0;
-}
-
-ThreadLinux::~ThreadLinux()
-{
-    pthread_attr_destroy(&_attr);
-    delete _event;
-}
-
-#define HAS_THREAD_ID !defined(MAC_IPHONE) && !defined(MAC_IPHONE_SIM)  &&  \
-                      !defined(WEBRTC_MAC) && !defined(WEBRTC_MAC_INTEL) && \
-                      !defined(MAC_DYLIB)  && !defined(MAC_INTEL_DYLIB)
-#if HAS_THREAD_ID
-bool ThreadLinux::Start(unsigned int& threadID)
-#else
-bool ThreadLinux::Start(unsigned int& /*threadID*/)
-#endif
-{
-    if (!_runFunction)
-    {
-        return false;
-    }
-    int result = pthread_attr_setdetachstate(&_attr, PTHREAD_CREATE_DETACHED);
-    // Set the stack stack size to 1M.
-    result |= pthread_attr_setstacksize(&_attr, 1024*1024);
-#ifdef WEBRTC_THREAD_RR
-    const int policy = SCHED_RR;
-#else
-    const int policy = SCHED_FIFO;
-#endif
-    _event->Reset();
-    result |= pthread_create(&_thread, &_attr, &StartThread, this);
-    if (result != 0)
-    {
-        return false;
-    }
-
-    // Wait up to 10 seconds for the OS to call the callback function. Prevents
-    // race condition if Stop() is called too quickly after start.
-    if (kEventSignaled != _event->Wait(WEBRTC_EVENT_10_SEC))
-    {
-        // Timed out. Something went wrong.
-        _runFunction = NULL;
-        return false;
-    }
-
-#if HAS_THREAD_ID
-    threadID = static_cast<unsigned int>(_thread);
-#endif
-    sched_param param;
-
-    const int minPrio = sched_get_priority_min(policy);
-    const int maxPrio = sched_get_priority_max(policy);
-    if ((minPrio == EINVAL) || (maxPrio == EINVAL))
-    {
-        return false;
-    }
-
-    switch (_prio)
-    {
-    case kLowPriority:
-        param.sched_priority = minPrio + 1;
-        break;
-    case kNormalPriority:
-        param.sched_priority = (minPrio + maxPrio) / 2;
-        break;
-    case kHighPriority:
-        param.sched_priority = maxPrio - 3;
-        break;
-    case kHighestPriority:
-        param.sched_priority = maxPrio - 2;
-        break;
-    case kRealtimePriority:
-        param.sched_priority = maxPrio - 1;
-        break;
-    default:
-        return false;
-    }
-    result = pthread_setschedparam(_thread, policy, &param);
-    if (result == EINVAL)
-    {
-        return false;
-    }
-    return true;
-}
-
-#if (defined(WEBRTC_LINUX) && !defined(ANDROID))
-bool ThreadLinux::SetAffinity(const int* processorNumbers,
-                              const unsigned int amountOfProcessors)
-{
-    if (!processorNumbers || (amountOfProcessors == 0))
-    {
-        return false;
-    }
-
-    cpu_set_t mask;
-    CPU_ZERO(&mask);
-
-    for(unsigned int processor = 0;
-        processor < amountOfProcessors;
-        processor++)
-    {
-        CPU_SET(processorNumbers[processor], &mask);
-    }
-    const int result = sched_setaffinity(_linuxPid, (unsigned int)sizeof(mask),
-                                         &mask);
-    if (result != 0)
-    {
-        return false;
-
-    }
-    return true;
-}
-#else
-// NOTE: On Mac OS X, use the Thread affinity API in
-// /usr/include/mach/thread_policy.h: thread_policy_set and mach_thread_self()
-// instead of Linux gettid() syscall.
-bool ThreadLinux::SetAffinity(const int* , const unsigned int)
-{
-    return false;
-}
-#endif
-
-void ThreadLinux::SetNotAlive()
-{
-    _alive = false;
-}
-
-bool ThreadLinux::Shutdown()
-{
-#if !defined(ANDROID)
-    if (_thread && (0 != pthread_cancel(_thread)))
-    {
-        return false;
-    }
-
-    return true;
-#else
-    return false;
-#endif
-}
-
-bool ThreadLinux::Stop()
-{
-    _alive = false;
-
-    // TODO (hellner) why not use an event here?
-    // Wait up to 10 seconds for the thread to terminate
-    for (int i = 0; i < 1000 && !_dead; i++)
-    {
-        timespec t;
-        t.tv_sec = 0;
-        t.tv_nsec = 10*1000*1000;
-        nanosleep(&t, NULL);
-    }
-    if (_dead)
-    {
-        return true;
-    }
-    else
-    {
-        return false;
-    }
-}
-
-void ThreadLinux::Run()
-{
-    _alive = true;
-    _dead  = false;
-#ifdef WEBRTC_LINUX
-    if(_linuxPid == -1)
-    {
-        _linuxPid = gettid();
-    }
-#endif
-    // The event the Start() is waiting for.
-    _event->Set();
-
-    if (_setThreadName)
-    {
-#ifdef WEBRTC_LINUX
-        WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,-1,
-                     "Thread with id:%d name:%s started ", _linuxPid, _name);
-        prctl(PR_SET_NAME, (unsigned long)_name, 0, 0, 0);
-#else
-        WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,-1,
-                     "Thread with name:%s started ", _name);
-#endif
-    }else
-    {
-#ifdef WEBRTC_LINUX
-        WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
-                     "Thread with id:%d without name started", _linuxPid);
-#else
-        WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
-                     "Thread without name started");
-#endif
-    }
-    do
-    {
-        if (_runFunction)
-        {
-            if (!_runFunction(_obj))
-            {
-                _alive = false;
-            }
-        }
-        else
-        {
-            _alive = false;
-        }
-    }
-    while (_alive);
-
-    if (_setThreadName)
-    {
-        // Don't set the name for the trace thread because it may cause a
-        // deadlock. TODO (hellner) there should be a better solution than
-        // coupling the thread and the trace class like this.
-        if (strcmp(_name, "Trace"))
-        {
-            WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,-1,
-                         "Thread with name:%s stopped", _name);
-        }
-    }
-    else
-    {
-        WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,-1,
-                     "Thread without name stopped");
-    }
-    _dead = true;
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/thread_linux.h b/src/system_wrappers/source/thread_linux.h
deleted file mode 100644
index 3e2b908..0000000
--- a/src/system_wrappers/source/thread_linux.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_LINUX_H_
-
-#include "thread_wrapper.h"
-#include <pthread.h>
-
-namespace webrtc {
-class EventWrapper;
-
-class ThreadLinux : public ThreadWrapper
-{
-public:
-    static ThreadWrapper* Create(ThreadRunFunction func, ThreadObj obj,
-                                 ThreadPriority prio, const char* threadName);
-
-    ThreadLinux(ThreadRunFunction func, ThreadObj obj, ThreadPriority prio,
-                const char* threadName);
-    ~ThreadLinux();
-
-    // From ThreadWrapper
-    virtual void SetNotAlive();
-    virtual bool Start(unsigned int& id);
-    // Not implemented on Mac
-    virtual bool SetAffinity(const int* processorNumbers,
-                             unsigned int amountOfProcessors);
-    virtual bool Stop();
-    virtual bool Shutdown();
-
-    void Run();
-
-private:
-    int Construct();
-
-private:
-    // processing function
-    ThreadRunFunction   _runFunction;
-    ThreadObj           _obj;
-
-    // internal state
-    bool                    _alive;
-    bool                    _dead;
-    ThreadPriority          _prio;
-    EventWrapper*           _event;
-
-    // zero-terminated thread name string
-    char                    _name[kThreadMaxNameLength];
-    bool                    _setThreadName;
-
-    // handle to thread
-    pthread_attr_t          _attr;
-    pthread_t               _thread;
-#ifdef WEBRTC_LINUX
-    pid_t                   _linuxPid;
-#endif
-
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_LINUX_H_
diff --git a/src/system_wrappers/source/trace_impl.cc b/src/system_wrappers/source/trace_impl.cc
deleted file mode 100644
index 0a5f9db..0000000
--- a/src/system_wrappers/source/trace_impl.cc
+++ /dev/null
@@ -1,949 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace_impl.h"
-
-#include <cassert>
-#include <string.h> // memset
-
-#ifdef _WIN32
-#include "trace_windows.h"
-#include "fix_interlocked_exchange_pointer_windows.h"
-#else
-#include <stdio.h>
-#include <time.h>
-#include <stdarg.h>
-#include "trace_linux.h"
-#endif // _WIN32
-
-#define KEY_LEN_CHARS 31
-
-#ifdef _WIN32
-    #pragma warning(disable:4355)
-// VS 2005: Disable warnings for default initialized arrays.
-    #pragma warning(disable:4351)
-#endif // _WIN32
-
-namespace webrtc {
-static WebRtc_UWord32 levelFilter = kTraceDefault;
-
-// Construct On First Use idiom. Avoids "static initialization order fiasco".
-Trace* TraceImpl::StaticInstance(TraceCount inc, const TraceLevel level)
-{
-    // TODO (hellner): use atomic wrapper instead.
-    static volatile long theTraceCount = 0;
-    static Trace* volatile theTrace = NULL;
-
-    TraceCreate state = WEBRTC_TRACE_EXIST;
-
-    // Sanitys to avoid taking lock unless absolutely necessary (for
-    // performance reasons). inc == WEBRTC_TRACE_INC_NO_CREATE) implies that
-    // a message will be written to file.
-    if(level != kTraceAll && inc == WEBRTC_TRACE_INC_NO_CREATE)
-    {
-        if(!(level & levelFilter))
-        {
-            return NULL;
-        }
-    }
-
-#ifndef _WIN32
-    // TODO (pwestin): crtiSect is never reclaimed. Fix memory leak.
-    static CriticalSectionWrapper* crtiSect(
-        CriticalSectionWrapper::CreateCriticalSection());
-    CriticalSectionScoped lock(*crtiSect);
-
-    if(inc == WEBRTC_TRACE_INC_NO_CREATE && theTraceCount == 0)
-    {
-        return NULL;
-    }
-
-    if(inc == WEBRTC_TRACE_INC || inc == WEBRTC_TRACE_INC_NO_CREATE)
-    {
-        theTraceCount++;
-        if(theTraceCount == 1)
-        {
-            state = WEBRTC_TRACE_CREATE;
-        }
-    } else {
-        theTraceCount--;
-        if(theTraceCount == 0)
-        {
-            state = WEBRTC_TRACE_DESTROY;
-        }
-    }
-    if(state == WEBRTC_TRACE_CREATE)
-    {
-        theTrace = TraceImpl::CreateTrace();
-
-    } else if(state == WEBRTC_TRACE_DESTROY) {
-        Trace* oldValue = theTrace;
-        theTrace = NULL;
-        // The lock is held by the scoped critical section. Release the lock
-        // temporarily so that the trace can be safely deleted. If the lock
-        // was kept during the delete, e.g. creating and destroying the trace
-        // too quickly may lead to a deadlock.
-        // This is due to the fact that starting and stopping a ThreadWrapper
-        // thread will trigger writing of trace messages.
-        // TODO (hellner): remove the tight coupling with the thread
-        //                 implementation.
-        crtiSect->Leave();
-        if(oldValue)
-        {
-            delete static_cast<TraceImpl*>(oldValue);
-        }
-        // Re-aqcuire the lock.
-        crtiSect->Enter();
-        return NULL;
-    }
-#else  // _WIN32
-    if(inc == WEBRTC_TRACE_INC_NO_CREATE && theTraceCount == 0)
-    {
-        return NULL;
-    }
-    if(inc == WEBRTC_TRACE_INC_NO_CREATE)
-    {
-        if(1 == InterlockedIncrement(&theTraceCount))
-        {
-            // The trace has been destroyed by some other thread. Rollback.
-            InterlockedDecrement(&theTraceCount);
-            assert(false);
-            return NULL;
-        }
-        // Sanity to catch corrupt state.
-        if(theTrace == NULL)
-        {
-            assert(false);
-            InterlockedDecrement(&theTraceCount);
-            return NULL;
-        }
-    } else if(inc == WEBRTC_TRACE_INC) {
-        if(theTraceCount == 0)
-        {
-            state = WEBRTC_TRACE_CREATE;
-        } else {
-            if(1 == InterlockedIncrement(&theTraceCount))
-            {
-                // InterlockedDecrement because reference count should not be
-                // updated just yet (that's done when the trace is created).
-                InterlockedDecrement(&theTraceCount);
-                state = WEBRTC_TRACE_CREATE;
-            }
-        }
-    } else {
-        int newValue = InterlockedDecrement(&theTraceCount);
-        if(newValue == 0)
-        {
-            state = WEBRTC_TRACE_DESTROY;
-        }
-    }
-
-    if(state == WEBRTC_TRACE_CREATE)
-    {
-        // Create trace and let whichever thread finishes first assign its local
-        // copy to the global instance. All other threads reclaim their local
-        // copy.
-        Trace* newTrace = TraceImpl::CreateTrace();
-        if(1 == InterlockedIncrement(&theTraceCount))
-        {
-            Trace* oldValue = (Trace*)InterlockedExchangePointer(
-                reinterpret_cast<void* volatile*>(&theTrace), newTrace);
-            assert(oldValue == NULL);
-            assert(theTrace);
-        } else {
-            InterlockedDecrement(&theTraceCount);
-            if(newTrace)
-            {
-                delete static_cast<TraceImpl*>(newTrace);
-            }
-        }
-        return NULL;
-    } else if(state == WEBRTC_TRACE_DESTROY)
-    {
-        Trace* oldValue = (Trace*)InterlockedExchangePointer(
-            reinterpret_cast<void* volatile*>(&theTrace), NULL);
-        if(oldValue)
-        {
-            delete static_cast<TraceImpl*>(oldValue);
-        }
-        return NULL;
-    }
-#endif // #ifndef _WIN32
-    return theTrace;
-}
-
-void Trace::CreateTrace()
-{
-    TraceImpl::StaticInstance(WEBRTC_TRACE_INC);
-}
-
-void Trace::ReturnTrace()
-{
-    TraceImpl::StaticInstance(WEBRTC_TRACE_DEC);
-}
-
-TraceImpl* TraceImpl::GetTrace(const TraceLevel level)
-{
-    return (TraceImpl*)StaticInstance(WEBRTC_TRACE_INC_NO_CREATE, level);
-}
-
-Trace* TraceImpl::CreateTrace()
-{
-#if defined(_WIN32)
-    return new TraceWindows();
-#else
-    return new TraceLinux();
-#endif
-}
-
-TraceImpl::TraceImpl()
-    : _critsectInterface(*CriticalSectionWrapper::CreateCriticalSection()),
-      _callback(NULL),
-      _rowCountText(0),
-      _fileCountText(0),
-      _traceFile(*FileWrapper::Create()),
-      _thread(*ThreadWrapper::CreateThread(TraceImpl::Run, this,
-                                           kHighestPriority, "Trace")),
-      _event(*EventWrapper::Create()),
-      _critsectArray(*CriticalSectionWrapper::CreateCriticalSection()),
-      _nextFreeIdx(),
-      _level(),
-      _length(),
-      _messageQueue(),
-      _activeQueue(0)
-{
-    _nextFreeIdx[0] = 0;
-    _nextFreeIdx[1] = 0;
-
-    unsigned int tid = 0;
-    _thread.Start(tid);
-
-    for(int m = 0; m < WEBRTC_TRACE_NUM_ARRAY; m++)
-    {
-        for(int n = 0; n < WEBRTC_TRACE_MAX_QUEUE; n++)
-        {
-            _messageQueue[m][n] = new
-                WebRtc_Word8[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
-        }
-    }
-}
-
-bool TraceImpl::StopThread()
-{
-    // Release the worker thread so that it can flush any lingering messages.
-    _event.Set();
-
-    // Allow 10 ms for pending messages to be flushed out.
-    // TODO (hellner): why not use condition variables to do this? Or let the
-    //                 worker thread die and let this thread flush remaining
-    //                 messages?
-#ifdef _WIN32
-    Sleep(10);
-#else
-    timespec t;
-    t.tv_sec = 0;
-    t.tv_nsec = 10*1000000;
-    nanosleep(&t,NULL);
-#endif
-
-    _thread.SetNotAlive();
-    // Make sure the thread finishes as quickly as possible (instead of having
-    // to wait for the timeout).
-    _event.Set();
-    bool stopped = _thread.Stop();
-
-    CriticalSectionScoped lock(_critsectInterface);
-    _traceFile.Flush();
-    _traceFile.CloseFile();
-    return stopped;
-}
-
-TraceImpl::~TraceImpl()
-{
-    StopThread();
-    delete &_event;
-    delete &_traceFile;
-    delete &_thread;
-    delete &_critsectInterface;
-    delete &_critsectArray;
-
-    for(int m = 0; m < WEBRTC_TRACE_NUM_ARRAY; m++)
-    {
-        for(int n = 0; n < WEBRTC_TRACE_MAX_QUEUE; n++)
-        {
-            delete [] _messageQueue[m][n];
-        }
-    }
-}
-
-WebRtc_Word32 TraceImpl::AddLevel(char* szMessage, const TraceLevel level) const
-{
-    switch (level)
-    {
-        case kTraceStateInfo:
-            sprintf (szMessage, "STATEINFO ; ");
-            break;
-        case kTraceWarning:
-            sprintf (szMessage, "WARNING   ; ");
-            break;
-        case kTraceError:
-            sprintf (szMessage, "ERROR     ; ");
-            break;
-        case kTraceCritical:
-            sprintf (szMessage, "CRITICAL  ; ");
-            break;
-        case kTraceInfo:
-            sprintf (szMessage, "DEBUGINFO ; ");
-            break;
-        case kTraceModuleCall:
-            sprintf (szMessage, "MODULECALL; ");
-            break;
-        case kTraceMemory:
-            sprintf (szMessage, "MEMORY    ; ");
-            break;
-        case kTraceTimer:
-            sprintf (szMessage, "TIMER     ; ");
-            break;
-        case kTraceStream:
-            sprintf (szMessage, "STREAM    ; ");
-            break;
-        case kTraceApiCall:
-            sprintf (szMessage, "APICALL   ; ");
-            break;
-        case kTraceDebug:
-            sprintf (szMessage, "DEBUG     ; ");
-            break;
-        default:
-            assert(false);
-            return 0;
-    }
-    // All messages are 12 characters.
-    return 12;
-}
-
-WebRtc_Word32 TraceImpl::AddModuleAndId(char* traceMessage,
-                                        const TraceModule module,
-                                        const WebRtc_Word32 id) const
-{
-    // Use long int to prevent problems with different definitions of
-    // WebRtc_Word32.
-    // TODO (hellner): is this actually a problem? If so, it should be better to
-    //                 clean up WebRtc_Word32
-    const long int idl = id;
-    if(idl != -1)
-    {
-        const unsigned long int idEngine = id>>16;
-        const unsigned long int idChannel = id & 0xffff;
-
-        switch (module)
-        {
-            case kTraceVoice:
-                sprintf(traceMessage, "       VOICE:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceVideo:
-                sprintf(traceMessage, "       VIDEO:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceUtility:
-                sprintf(traceMessage, "     UTILITY:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceRtpRtcp:
-                sprintf(traceMessage, "    RTP/RTCP:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceTransport:
-                sprintf(traceMessage, "   TRANSPORT:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceAudioCoding:
-                sprintf(traceMessage, "AUDIO CODING:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceSrtp:
-                sprintf(traceMessage, "        SRTP:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceAudioMixerServer:
-                sprintf(traceMessage, " AUDIO MIX/S:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceAudioMixerClient:
-                sprintf(traceMessage, " AUDIO MIX/C:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceVideoCoding:
-                sprintf(traceMessage, "VIDEO CODING:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceVideoMixer:
-                // Print sleep time and API call
-                sprintf(traceMessage, "   VIDEO MIX:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceFile:
-                sprintf(traceMessage, "        FILE:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceAudioProcessing:
-                sprintf(traceMessage, "  AUDIO PROC:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceAudioDevice:
-                sprintf(traceMessage, "AUDIO DEVICE:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceVideoRenderer:
-                sprintf(traceMessage, "VIDEO RENDER:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceVideoCapture:
-                sprintf(traceMessage, "VIDEO CAPTUR:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            case kTraceVideoPreocessing:
-                sprintf(traceMessage, "  VIDEO PROC:%5ld %5ld;", idEngine,
-                        idChannel);
-                break;
-            default:
-                assert(false);
-                return 0;
-        }
-    } else {
-        switch (module)
-        {
-            case kTraceVoice:
-                sprintf (traceMessage, "       VOICE:%11ld;", idl);
-                break;
-            case kTraceVideo:
-                sprintf (traceMessage, "       VIDEO:%11ld;", idl);
-                break;
-            case kTraceUtility:
-                sprintf (traceMessage, "     UTILITY:%11ld;", idl);
-                break;
-            case kTraceRtpRtcp:
-                sprintf (traceMessage, "    RTP/RTCP:%11ld;", idl);
-                break;
-            case kTraceTransport:
-                sprintf (traceMessage, "   TRANSPORT:%11ld;", idl);
-                break;
-            case kTraceAudioCoding:
-                sprintf (traceMessage, "AUDIO CODING:%11ld;", idl);
-                break;
-            case kTraceSrtp:
-                sprintf (traceMessage, "        SRTP:%11ld;", idl);
-                break;
-            case kTraceAudioMixerServer:
-                sprintf (traceMessage, " AUDIO MIX/S:%11ld;", idl);
-                break;
-            case kTraceAudioMixerClient:
-                sprintf (traceMessage, " AUDIO MIX/C:%11ld;", idl);
-                break;
-            case kTraceVideoCoding:
-                sprintf (traceMessage, "VIDEO CODING:%11ld;", idl);
-                break;
-            case kTraceVideoMixer:
-                sprintf (traceMessage, "   VIDEO MIX:%11ld;", idl);
-                break;
-            case kTraceFile:
-                sprintf (traceMessage, "        FILE:%11ld;", idl);
-                break;
-            case kTraceAudioProcessing:
-                sprintf (traceMessage, "  AUDIO PROC:%11ld;", idl);
-                break;
-            case kTraceAudioDevice:
-                sprintf (traceMessage, "AUDIO DEVICE:%11ld;", idl);
-                break;
-            case kTraceVideoRenderer:
-                sprintf (traceMessage, "VIDEO RENDER:%11ld;", idl);
-                break;
-            case kTraceVideoCapture:
-                sprintf (traceMessage, "VIDEO CAPTUR:%11ld;", idl);
-                break;
-            case kTraceVideoPreocessing:
-                sprintf (traceMessage, "  VIDEO PROC:%11ld;", idl);
-                break;
-            default:
-                assert(false);
-                return 0;
-        }
-    }
-    // All messages are 25 characters.
-    return 25;
-}
-
-WebRtc_Word32 TraceImpl::SetTraceFileImpl(const WebRtc_Word8* fileNameUTF8,
-                                          const bool addFileCounter)
-{
-    CriticalSectionScoped lock(_critsectInterface);
-
-    _traceFile.Flush();
-    _traceFile.CloseFile();
-
-    if(fileNameUTF8)
-    {
-        if(addFileCounter)
-        {
-            _fileCountText = 1;
-
-            WebRtc_Word8 fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize];
-            CreateFileName(fileNameUTF8, fileNameWithCounterUTF8,
-                           _fileCountText);
-            if(_traceFile.OpenFile(fileNameWithCounterUTF8, false, false,
-                                   true) == -1)
-            {
-                return -1;
-            }
-        }else {
-            _fileCountText = 0;
-            if(_traceFile.OpenFile(fileNameUTF8, false, false, true) == -1)
-            {
-                return -1;
-            }
-        }
-    }
-    _rowCountText = 0;
-    return 0;
-}
-
-WebRtc_Word32 TraceImpl::TraceFileImpl(
-    WebRtc_Word8 fileNameUTF8[FileWrapper::kMaxFileNameSize])
-{
-    CriticalSectionScoped lock(_critsectInterface);
-    return _traceFile.FileName(fileNameUTF8, FileWrapper::kMaxFileNameSize);
-}
-
-WebRtc_Word32 TraceImpl::SetTraceCallbackImpl(TraceCallback* callback)
-{
-    CriticalSectionScoped lock(_critsectInterface);
-    _callback = callback;
-    return 0;
-}
-
-WebRtc_Word32 TraceImpl::AddMessage(
-    char* traceMessage,
-    const char msg[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
-    const WebRtc_UWord16 writtenSoFar) const
-
-{
-    int length = 0;
-    if(writtenSoFar >= WEBRTC_TRACE_MAX_MESSAGE_SIZE)
-    {
-        return -1;
-    }
-    // - 2 to leave room for newline and NULL termination
-#ifdef _WIN32
-    length = _snprintf(traceMessage,
-                       WEBRTC_TRACE_MAX_MESSAGE_SIZE - writtenSoFar - 2,
-                       "%s",msg);
-    if(length < 0)
-    {
-        length = WEBRTC_TRACE_MAX_MESSAGE_SIZE - writtenSoFar - 2;
-        traceMessage[length] = 0;
-    }
-#else
-    length = snprintf(traceMessage,
-                      WEBRTC_TRACE_MAX_MESSAGE_SIZE-writtenSoFar-2, "%s",msg);
-    if(length < 0 || length > WEBRTC_TRACE_MAX_MESSAGE_SIZE-writtenSoFar - 2)
-    {
-        length = WEBRTC_TRACE_MAX_MESSAGE_SIZE - writtenSoFar - 2;
-        traceMessage[length] = 0;
-    }
-#endif
-    // Length with NULL termination.
-    return length+1;
-}
-
-void TraceImpl::AddMessageToList(
-    const char traceMessage[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
-    const WebRtc_UWord16 length,
-    const TraceLevel level)
-{
-    CriticalSectionScoped lock(_critsectArray);
-
-    if(_nextFreeIdx[_activeQueue] >= WEBRTC_TRACE_MAX_QUEUE)
-    {
-        if( ! _traceFile.Open() &&
-            !_callback)
-        {
-            // Keep at least the last 1/4 of old messages when not logging.
-            // TODO (hellner): isn't this redundant. The user will make it known
-            //                 when to start logging. Why keep messages before
-            //                 that?
-            for(int n = 0; n < WEBRTC_TRACE_MAX_QUEUE/4; n++)
-            {
-                const int lastQuarterOffset = (3*WEBRTC_TRACE_MAX_QUEUE/4);
-                memcpy(_messageQueue[_activeQueue][n],
-                       _messageQueue[_activeQueue][n + lastQuarterOffset],
-                       WEBRTC_TRACE_MAX_MESSAGE_SIZE);
-            }
-            _nextFreeIdx[_activeQueue] = WEBRTC_TRACE_MAX_QUEUE/4;
-        } else {
-            // More messages are being written than there is room for in the
-            // buffer. Drop any new messages.
-            // TODO (hellner): its probably better to drop old messages instead
-            //                 of new ones. One step further: if this happens
-            //                 it's due to writing faster than what can be
-            //                 processed. Maybe modify the filter at this point.
-            //                 E.g. turn of STREAM.
-            return;
-        }
-    }
-
-    WebRtc_UWord16 idx = _nextFreeIdx[_activeQueue];
-    _nextFreeIdx[_activeQueue]++;
-
-    _level[_activeQueue][idx] = level;
-    _length[_activeQueue][idx] = length;
-    memcpy(_messageQueue[_activeQueue][idx], traceMessage, length);
-
-    if(_nextFreeIdx[_activeQueue] == WEBRTC_TRACE_MAX_QUEUE-1)
-    {
-        // Loggin more messages than can be worked off. Log a warning.
-        memcpy(_messageQueue[_activeQueue][_nextFreeIdx[_activeQueue]],
-               "WARNING MISSING TRACE MESSAGES\n", 32);
-        _nextFreeIdx[_activeQueue]++;
-    }
-}
-
-bool TraceImpl::Run(void* obj)
-{
-    return static_cast<TraceImpl*>(obj)->Process();
-}
-
-bool TraceImpl::Process()
-{
-    if(_event.Wait(1000) == kEventSignaled)
-    {
-        if(_traceFile.Open() || _callback)
-        {
-            // File mode (not calback mode).
-            WriteToFile();
-        }
-    } else {
-        _traceFile.Flush();
-    }
-    return true;
-}
-
-void TraceImpl::WriteToFile()
-{
-    WebRtc_UWord8 localQueueActive = 0;
-    WebRtc_UWord16 localNextFreeIdx = 0;
-
-    // There are two buffer. One for reading (for writing to file) and one for
-    // writing (for storing new messages). Let new messages be posted to the
-    // unused buffer so that the current buffer can be flushed safely.
-    {
-        CriticalSectionScoped lock(_critsectArray);
-        localNextFreeIdx = _nextFreeIdx[_activeQueue];
-        _nextFreeIdx[_activeQueue] = 0;
-        localQueueActive = _activeQueue;
-        if(_activeQueue == 0)
-        {
-            _activeQueue = 1;
-        } else
-        {
-            _activeQueue = 0;
-        }
-    }
-    if(localNextFreeIdx == 0)
-    {
-        return;
-    }
-
-    CriticalSectionScoped lock(_critsectInterface);
-
-    for(WebRtc_UWord16 idx = 0; idx <localNextFreeIdx; idx++)
-    {
-        TraceLevel localLevel = _level[localQueueActive][idx];
-        if(_callback)
-        {
-            _callback->Print(localLevel, _messageQueue[localQueueActive][idx],
-                             _length[localQueueActive][idx]);
-        }
-        if(_traceFile.Open())
-        {
-            if(_rowCountText > WEBRTC_TRACE_MAX_FILE_SIZE)
-            {
-                // wrap file
-                _rowCountText = 0;
-                _traceFile.Flush();
-
-                if(_fileCountText == 0)
-                {
-                    _traceFile.Rewind();
-                } else
-                {
-                    WebRtc_Word8 oldFileName[FileWrapper::kMaxFileNameSize];
-                    WebRtc_Word8 newFileName[FileWrapper::kMaxFileNameSize];
-
-                    // get current name
-                    _traceFile.FileName(oldFileName,
-                                        FileWrapper::kMaxFileNameSize);
-                    _traceFile.CloseFile();
-
-                    _fileCountText++;
-
-                    UpdateFileName(oldFileName, newFileName, _fileCountText);
-
-                    if(_traceFile.OpenFile(newFileName, false, false,
-                                           true) == -1)
-                    {
-                        return;
-                    }
-                }
-            }
-            if(_rowCountText ==  0)
-            {
-                WebRtc_Word8 message[WEBRTC_TRACE_MAX_MESSAGE_SIZE + 1];
-                WebRtc_Word32 length = AddDateTimeInfo(message);
-                if(length != -1)
-                {
-                    message[length] = 0;
-                    message[length-1] = '\n';
-                    _traceFile.Write(message, length);
-                    _rowCountText++;
-                }
-                length = AddBuildInfo(message);
-                if(length != -1)
-                {
-                    message[length+1] = 0;
-                    message[length] = '\n';
-                    message[length-1] = '\n';
-                    _traceFile.Write(message, length+1);
-                    _rowCountText++;
-                    _rowCountText++;
-                }
-            }
-            WebRtc_UWord16 length = _length[localQueueActive][idx];
-            _messageQueue[localQueueActive][idx][length] = 0;
-            _messageQueue[localQueueActive][idx][length-1] = '\n';
-            _traceFile.Write(_messageQueue[localQueueActive][idx], length);
-            _rowCountText++;
-        }
-    }
-}
-
-void TraceImpl::AddImpl(const TraceLevel level, const TraceModule module,
-                        const WebRtc_Word32 id,
-                        const char msg[WEBRTC_TRACE_MAX_MESSAGE_SIZE])
-{
-    if (TraceCheck(level))
-    {
-        char traceMessage[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
-        char* meassagePtr = traceMessage;
-
-        WebRtc_Word32 len = 0;
-        WebRtc_Word32 ackLen = 0;
-
-        len = AddLevel(meassagePtr, level);
-        if(len == -1)
-        {
-            return;
-        }
-        meassagePtr += len;
-        ackLen += len;
-
-        len = AddTime(meassagePtr, level);
-        if(len == -1)
-        {
-            return;
-        }
-        meassagePtr += len;
-        ackLen += len;
-
-        len = AddModuleAndId(meassagePtr, module, id);
-        if(len == -1)
-        {
-            return;
-        }
-        meassagePtr += len;
-        ackLen += len;
-
-        len = AddThreadId(meassagePtr);
-        if(len == -1)
-        {
-            return;
-        }
-        meassagePtr += len;
-        ackLen += len;
-
-        len = AddMessage(meassagePtr, msg, (WebRtc_UWord16)ackLen);
-        if(len == -1)
-        {
-            return;
-        }
-        ackLen += len;
-        AddMessageToList(traceMessage,(WebRtc_UWord16)ackLen, level);
-
-        // Make sure that messages are written as soon as possible.
-        _event.Set();
-    }
-}
-
-bool TraceImpl::TraceCheck(const TraceLevel level) const
-{
-    return (level & levelFilter)? true:false;
-}
-
-bool TraceImpl::UpdateFileName(
-    const WebRtc_Word8 fileNameUTF8[FileWrapper::kMaxFileNameSize],
-    WebRtc_Word8 fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
-    const WebRtc_UWord32 newCount) const
-{
-    WebRtc_Word32 length = (WebRtc_Word32)strlen(fileNameUTF8);
-    if(length < 0)
-    {
-        return false;
-    }
-
-    WebRtc_Word32 lengthWithoutFileEnding = length-1;
-    while(lengthWithoutFileEnding > 0)
-    {
-        if(fileNameUTF8[lengthWithoutFileEnding] == '.')
-        {
-            break;
-        } else {
-            lengthWithoutFileEnding--;
-        }
-    }
-    if(lengthWithoutFileEnding == 0)
-    {
-        lengthWithoutFileEnding = length;
-    }
-    WebRtc_Word32 lengthTo_ = lengthWithoutFileEnding - 1;
-    while(lengthTo_ > 0)
-    {
-        if(fileNameUTF8[lengthTo_] == '_')
-        {
-            break;
-        } else {
-            lengthTo_--;
-        }
-    }
-
-    memcpy(fileNameWithCounterUTF8, fileNameUTF8, lengthTo_);
-    sprintf(fileNameWithCounterUTF8+lengthTo_, "_%lu%s", newCount,
-            fileNameUTF8+lengthWithoutFileEnding);
-    return true;
-}
-
-bool TraceImpl::CreateFileName(
-    const WebRtc_Word8 fileNameUTF8[FileWrapper::kMaxFileNameSize],
-    WebRtc_Word8 fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
-    const WebRtc_UWord32 newCount) const
-{
-    WebRtc_Word32 length = (WebRtc_Word32)strlen(fileNameUTF8);
-    if(length < 0)
-    {
-        return false;
-    }
-
-    WebRtc_Word32 lengthWithoutFileEnding = length-1;
-    while(lengthWithoutFileEnding > 0)
-    {
-        if(fileNameUTF8[lengthWithoutFileEnding] == '.')
-        {
-            break;
-        }else
-        {
-            lengthWithoutFileEnding--;
-        }
-    }
-    if(lengthWithoutFileEnding == 0)
-    {
-        lengthWithoutFileEnding = length;
-    }
-    memcpy(fileNameWithCounterUTF8, fileNameUTF8, lengthWithoutFileEnding);
-    sprintf(fileNameWithCounterUTF8+lengthWithoutFileEnding, "_%lu%s",
-            newCount, fileNameUTF8+lengthWithoutFileEnding);
-    return true;
-}
-
-WebRtc_Word32 Trace::SetLevelFilter(WebRtc_UWord32 filter)
-{
-    levelFilter = filter;
-    return 0;
-};
-
-WebRtc_Word32 Trace::LevelFilter(WebRtc_UWord32& filter)
-{
-    filter = levelFilter;
-    return 0;
-};
-
-WebRtc_Word32 Trace::TraceFile(WebRtc_Word8 fileName[FileWrapper::kMaxFileNameSize])
-{
-    TraceImpl* trace = TraceImpl::GetTrace();
-    if(trace)
-    {
-        int retVal = trace->TraceFileImpl(fileName);
-        ReturnTrace();
-        return retVal;
-    }
-    return -1;
-}
-
-WebRtc_Word32 Trace::SetTraceFile(const WebRtc_Word8* fileName,
-                                  const bool addFileCounter)
-{
-    TraceImpl* trace = TraceImpl::GetTrace();
-    if(trace)
-    {
-        int retVal = trace->SetTraceFileImpl(fileName, addFileCounter);
-        ReturnTrace();
-        return retVal;
-    }
-    return -1;
-}
-
-WebRtc_Word32 Trace::SetTraceCallback(TraceCallback* callback)
-{
-    TraceImpl* trace = TraceImpl::GetTrace();
-    if(trace)
-    {
-        int retVal = trace->SetTraceCallbackImpl(callback);
-        ReturnTrace();
-        return retVal;
-    }
-    return -1;
-}
-
-void Trace::Add(const TraceLevel level, const TraceModule module,
-                const WebRtc_Word32 id, const char* msg, ...)
-
-{
-    TraceImpl* trace = TraceImpl::GetTrace(level);
-    if(trace)
-    {
-        if(trace->TraceCheck(level))
-        {
-            char tempBuff[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
-            char* buff = 0;
-            if(msg)
-            {
-                va_list args;
-                va_start(args, msg);
-#ifdef _WIN32
-                _vsnprintf(tempBuff,WEBRTC_TRACE_MAX_MESSAGE_SIZE-1,msg,args);
-#else
-                vsnprintf(tempBuff,WEBRTC_TRACE_MAX_MESSAGE_SIZE-1,msg,args);
-#endif
-                va_end(args);
-                buff = tempBuff;
-            }
-            trace->AddImpl(level, module, id, buff);
-        }
-        ReturnTrace();
-    }
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/trace_impl.h b/src/system_wrappers/source/trace_impl.h
deleted file mode 100644
index 42e82fe..0000000
--- a/src/system_wrappers/source/trace_impl.h
+++ /dev/null
@@ -1,141 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_IMPL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_IMPL_H_
-
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/event_wrapper.h"
-#include "system_wrappers/interface/file_wrapper.h"
-#include "system_wrappers/interface/trace.h"
-#include "system_wrappers/interface/thread_wrapper.h"
-
-namespace webrtc {
-enum TraceCount
-{
-    WEBRTC_TRACE_DEC    = 0,
-    WEBRTC_TRACE_INC    = 1,
-    WEBRTC_TRACE_INC_NO_CREATE = 2
-};
-
-enum TraceCreate
-{
-    WEBRTC_TRACE_EXIST    = 0,
-    WEBRTC_TRACE_CREATE   = 1,
-    WEBRTC_TRACE_DESTROY  = 2
-};
-
-// TODO (pwestin) WEBRTC_TRACE_MAX_QUEUE needs to be tweaked
-// TODO (hellner) the buffer should be close to how much the system can write to
-//                file. Increasing the buffer will not solve anything. Sooner or
-//                later the buffer is going to fill up anyways.
-#if defined(MAC_IPHONE)
-    #define WEBRTC_TRACE_MAX_QUEUE  2000
-#else
-    #define WEBRTC_TRACE_MAX_QUEUE  8000
-#endif
-#define WEBRTC_TRACE_NUM_ARRAY 2
-#define WEBRTC_TRACE_MAX_MESSAGE_SIZE 256
-// Total buffer size is WEBRTC_TRACE_NUM_ARRAY (number of buffer partitions) *
-// WEBRTC_TRACE_MAX_QUEUE (number of lines per buffer partition) *
-// WEBRTC_TRACE_MAX_MESSAGE_SIZE (number of 1 byte charachters per line) =
-// 1 or 4 Mbyte
-
-#define WEBRTC_TRACE_MAX_FILE_SIZE 100*1000
-// Number of rows that may be written to file. On average 110 bytes per row (max
-// 256 bytes per row). So on average 110*100*1000 = 11 Mbyte, max 256*100*1000 =
-// 25.6 Mbyte
-
-class TraceImpl : public Trace
-{
-public:
-    virtual ~TraceImpl();
-
-    static Trace* CreateTrace();
-    static TraceImpl* GetTrace(const TraceLevel level = kTraceAll);
-
-    static Trace* StaticInstance(TraceCount inc,
-                                 const TraceLevel level = kTraceAll);
-
-    WebRtc_Word32 SetTraceFileImpl(const WebRtc_Word8* fileName,
-                                   const bool addFileCounter);
-    WebRtc_Word32 TraceFileImpl(
-        WebRtc_Word8 fileName[FileWrapper::kMaxFileNameSize]);
-
-    WebRtc_Word32 SetTraceCallbackImpl(TraceCallback* callback);
-
-    void AddImpl(const TraceLevel level, const TraceModule module,
-                 const WebRtc_Word32 id, const char* msg);
-
-    bool StopThread();
-
-    bool TraceCheck(const TraceLevel level) const;
-
-protected:
-    TraceImpl();
-
-    // OS specific implementations
-    virtual WebRtc_Word32 AddThreadId(char* traceMessage) const = 0;
-    virtual WebRtc_Word32 AddTime(char* traceMessage,
-                                  const TraceLevel level) const = 0;
-
-    virtual WebRtc_Word32 AddBuildInfo(char* traceMessage) const = 0;
-    virtual WebRtc_Word32 AddDateTimeInfo(char* traceMessage) const = 0;
-
-    static bool Run(void* obj);
-    bool Process();
-
-private:
-    WebRtc_Word32 AddLevel(char* szMessage, const TraceLevel level) const;
-
-    WebRtc_Word32 AddModuleAndId(char* traceMessage, const TraceModule module,
-                                 const WebRtc_Word32 id) const;
-
-    WebRtc_Word32 AddMessage(char* traceMessage,
-                             const char msg[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
-                             const WebRtc_UWord16 writtenSoFar) const;
-
-    void AddMessageToList(
-        const char traceMessage[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
-        const WebRtc_UWord16 length,
-        const TraceLevel level);
-
-    bool UpdateFileName(
-        const WebRtc_Word8 fileNameUTF8[FileWrapper::kMaxFileNameSize],
-        WebRtc_Word8 fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
-        const WebRtc_UWord32 newCount) const;
-
-    bool CreateFileName(
-        const WebRtc_Word8 fileNameUTF8[FileWrapper::kMaxFileNameSize],
-        WebRtc_Word8 fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
-        const WebRtc_UWord32 newCount) const;
-
-    void WriteToFile();
-
-    CriticalSectionWrapper& _critsectInterface;
-    TraceCallback* _callback;
-    WebRtc_UWord32 _rowCountText;
-    WebRtc_UWord32 _fileCountText;
-
-    FileWrapper& _traceFile;
-    ThreadWrapper& _thread;
-    EventWrapper& _event;
-
-    // _critsectArray protects _activeQueue
-    CriticalSectionWrapper& _critsectArray;
-    WebRtc_UWord16 _nextFreeIdx[WEBRTC_TRACE_NUM_ARRAY];
-    TraceLevel _level[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
-    WebRtc_UWord16 _length[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
-    WebRtc_Word8* _messageQueue[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
-    WebRtc_UWord8 _activeQueue;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_IMPL_H_
diff --git a/src/system_wrappers/source/trace_linux.cc b/src/system_wrappers/source/trace_linux.cc
deleted file mode 100644
index 8dba3be..0000000
--- a/src/system_wrappers/source/trace_linux.cc
+++ /dev/null
@@ -1,135 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace_linux.h"
-
-#include <cassert>
-#include <stdarg.h>
-#include <stdio.h>
-#include <string.h>
-#include <time.h>
-
-#ifdef ANDROID
-    #include <pthread.h>
-#else
-    #include <iostream>
-#endif
-
-#if defined(_DEBUG)
-    #define BUILDMODE "d"
-#elif defined(DEBUG)
-    #define BUILDMODE "d"
-#elif defined(NDEBUG)
-    #define BUILDMODE "r"
-#else
-    #define BUILDMODE "?"
-#endif
-#define BUILDTIME __TIME__
-#define BUILDDATE __DATE__
-// example: "Oct 10 2002 12:05:30 r"
-#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
-
-namespace webrtc {
-TraceLinux::TraceLinux()
-{
-    _prevAPITickCount = time(NULL);
-    _prevTickCount = _prevAPITickCount;
-}
-
-TraceLinux::~TraceLinux()
-{
-    StopThread();
-}
-
-WebRtc_Word32 TraceLinux::AddThreadId(char* traceMessage) const
-{
-    WebRtc_UWord64 threadId = (WebRtc_UWord64)pthread_self();
-    sprintf(traceMessage, "%10llu; ", threadId);
-    // 12 bytes are written.
-    return 12;
-}
-
-WebRtc_Word32 TraceLinux::AddTime(char* traceMessage,
-                                  const TraceLevel level) const
-{
-    time_t dwCurrentTimeInSeconds = time(NULL);
-    struct tm systemTime;
-    gmtime_r(&dwCurrentTimeInSeconds, &systemTime);
-
-    if(level == kTraceApiCall)
-    {
-        WebRtc_UWord32 dwDeltaTime = dwCurrentTimeInSeconds - _prevTickCount;
-        _prevTickCount = dwCurrentTimeInSeconds;
-
-        if(_prevTickCount == 0)
-        {
-            dwDeltaTime = 0;
-        }
-        if(dwDeltaTime > 0x0fffffff)
-        {
-            // Either wraparound or data race.
-            dwDeltaTime = 0;
-        }
-        if(dwDeltaTime > 99999)
-        {
-            dwDeltaTime = 99999;
-        }
-
-        sprintf(traceMessage, "(%2u:%2u:%2u:%3u |%5lu) ", systemTime.tm_hour,
-                systemTime.tm_min, systemTime.tm_sec, 0,
-                static_cast<unsigned long>(dwDeltaTime));
-    } else {
-        WebRtc_UWord32 dwDeltaTime = dwCurrentTimeInSeconds - _prevAPITickCount;
-        _prevAPITickCount = dwCurrentTimeInSeconds;
-        if(_prevAPITickCount == 0)
-        {
-            dwDeltaTime = 0;
-        }
-        if(dwDeltaTime > 0x0fffffff)
-        {
-            // Either wraparound or data race.
-            dwDeltaTime = 0;
-        }
-        if(dwDeltaTime > 99999)
-        {
-            dwDeltaTime = 99999;
-        }
-        sprintf(traceMessage, "(%2u:%2u:%2u:%3u |%5lu) ", systemTime.tm_hour,
-                systemTime.tm_min, systemTime.tm_sec, 0,
-                static_cast<unsigned long>(dwDeltaTime));
-    }
-    // Messages is 22 characters.
-    return 22;
-}
-
-WebRtc_Word32 TraceLinux::AddBuildInfo(char* traceMessage) const
-{
-    sprintf(traceMessage, "Build info: %s", BUILDINFO);
-    // Include NULL termination (hence + 1).
-    return strlen(traceMessage) + 1;
-}
-
-WebRtc_Word32 TraceLinux::AddDateTimeInfo(char* traceMessage) const
-{
-    time_t t;
-    time(&t);
-    sprintf(traceMessage, "Local Date: %s", ctime(&t));
-    WebRtc_Word32 len = static_cast<WebRtc_Word32>(strlen(traceMessage));
-
-    if ('\n' == traceMessage[len - 1])
-    {
-        traceMessage[len - 1] = '\0';
-        --len;
-    }
-
-    // Messages is 12 characters.
-    return len + 1;
-}
-} // namespace webrtc
diff --git a/src/system_wrappers/source/trace_linux.h b/src/system_wrappers/source/trace_linux.h
deleted file mode 100644
index 6e327a0..0000000
--- a/src/system_wrappers/source/trace_linux.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_LINUX_H_
-
-#include "critical_section_wrapper.h"
-#include "trace_impl.h"
-
-namespace webrtc {
-class TraceLinux : public TraceImpl
-{
-public:
-    TraceLinux();
-    virtual ~TraceLinux();
-
-    virtual WebRtc_Word32 AddThreadId(char *traceMessage) const;
-    virtual WebRtc_Word32 AddTime(char* traceMessage,
-                                  const TraceLevel level) const;
-
-    virtual WebRtc_Word32 AddBuildInfo(char* traceMessage) const;
-    virtual WebRtc_Word32 AddDateTimeInfo(char* traceMessage) const;
-
-private:
-    volatile mutable WebRtc_UWord32  _prevAPITickCount;
-    volatile mutable WebRtc_UWord32  _prevTickCount;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_LINUX_H_
diff --git a/src/system_wrappers/test/Test.cpp b/src/system_wrappers/test/Test.cpp
deleted file mode 100644
index 7a34166..0000000
--- a/src/system_wrappers/test/Test.cpp
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-#include <iostream>
-
-#ifdef _WIN32
-    #include <windows.h>
-    #include <tchar.h>
-#else
-    #include <stdio.h>
-    #define Sleep(x) usleep(x*1000)
-#endif
-
-#include "common_types.h"
-#include "trace.h"
-#include "cpu_wrapper.h"
-
-
-#ifdef _WIN32
-int _tmain(int argc, _TCHAR* argv[])
-#else
-int main(int argc, char* argv[])
-#endif
-{
-    Trace::CreateTrace();
-    Trace::SetTraceFile("testTrace.txt");
-    Trace::SetLevelFilter(webrtc::kTraceAll);
-
-    printf("Start system wrapper test\n");
-
-    printf("Number of cores detected:%u\n", (unsigned int)CpuWrapper::DetectNumberOfCores());
-
-    CpuWrapper* cpu = CpuWrapper::CreateCpu();
-
-    WebRtc_UWord32 numCores;
-    WebRtc_UWord32* cores;
-
-    for(int i = 0; i< 10;i++)
-    {
-        WebRtc_Word32 total = cpu->CpuUsageMultiCore(numCores, cores);
-
-        printf("\nNumCores:%d\n", (int)numCores);
-        printf("Total cpu:%d\n", (int)total);
-
-        for (WebRtc_UWord32 i = 0; i< numCores;i++)
-        {
-            printf("Core:%lu CPU:%lu \n", i, cores[i]);
-        }
-        Sleep(1000);
-    }
-
-    printf("Done system wrapper test\n");
-
-    delete cpu;
-
-    Trace::ReturnTrace();
-};
diff --git a/src/system_wrappers/test/TestSort/TestSort.cpp b/src/system_wrappers/test/TestSort/TestSort.cpp
deleted file mode 100644
index 6846a71..0000000
--- a/src/system_wrappers/test/TestSort/TestSort.cpp
+++ /dev/null
@@ -1,265 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstdio>
-#include <algorithm>
-#include <cstring>
-
-#include "sort.h"
-#include "tick_util.h"
-
-// Excellent work polluting the global namespace Visual Studio...
-#undef max
-#undef min
-#include <limits>
-
-template<typename KeyType>
-struct LotsOfData
-{
-    KeyType key;
-    char data[64];
-};
-
-template<typename DataType>
-int Compare(const void* dataX, const void* dataY)
-{
-    DataType dataX = (DataType)*(const DataType*)dataX;
-    DataType dataY = (DataType)*(const DataType*)dataY;
-    if (dataX > dataY)
-    {
-        return 1;
-    }
-    else if (dataX < dataY)
-    {
-        return -1;
-    }
-
-    return 0;
-};
-
-template<typename DataType, typename KeyType>
-int CompareKey(const void* dataX, const void* dataY)
-{
-    KeyType keyX = ((const DataType*)dataX)->key;
-    KeyType keyY = ((const DataType*)dataY)->key;
-    if (keyX > keyY)
-    {
-        return 1;
-    }
-    else if (keyX < keyY)
-    {
-        return -1;
-    }
-
-    return 0;
-}
-
-template<typename DataType>
-struct KeyLessThan
-{
-    bool operator()(const DataType &dataX, const DataType &dataY) const
-    {
-        return dataX.key < dataY.key;
-    }
-};
-
-const char* TypeEnumToString(webrtc::Type type)
-{
-    switch (type)
-    {
-        using namespace webrtc;
-        case TYPE_Word8:
-            return "Word8";
-        case TYPE_UWord8:
-            return "UWord8";
-        case TYPE_Word16:
-            return "Word16";
-        case TYPE_UWord16:
-            return "UWord16";
-        case TYPE_Word32:
-            return "Word32";
-        case TYPE_UWord32:
-            return "UWord32";
-        case TYPE_Word64:
-            return "Word64";
-        case TYPE_UWord64:
-            return "UWord64";
-        case TYPE_Float32:
-            return "Float32";
-        case TYPE_Float64:
-            return "Float64";
-        default:
-            return "Unrecognized";
-    }
-}
-
-template<typename Type>
-Type TypedRand()
-{
-    if (std::numeric_limits<Type>::is_integer)
-    {
-        double floatRand = static_cast<double>(rand()) / RAND_MAX;
-        if (std::numeric_limits<Type>::is_signed)
-        {
-            floatRand -= 0.5;
-        }
-
-        // Uniform [-max()/2, max()/2] for signed
-        //         [0, max()] for unsigned
-        return static_cast<Type>(floatRand * std::numeric_limits<Type>::max());
-    }
-    else // Floating point
-    {
-        // Uniform [-0.5, 0.5]
-        // The outer cast is to remove template warnings.
-        return static_cast<Type>((static_cast<Type>(rand()) / RAND_MAX) - 0.5);
-    }
-}
-
-template<typename KeyType>
-void RunSortTest(webrtc::Type sortType, bool keySort)
-{
-    enum { DataLength = 1000 };
-    enum { NumOfTests = 10000 };
-    KeyType key[DataLength];
-    KeyType keyRef[DataLength];
-    LotsOfData<KeyType> data[DataLength];
-    LotsOfData<KeyType> dataRef[DataLength];
-    WebRtc_Word32 retVal = 0;
-
-    if (keySort)
-    {
-        printf("Running %s KeySort() tests...\n", TypeEnumToString(sortType));
-    }
-    else
-    {
-        printf("Running %s Sort() tests...\n", TypeEnumToString(sortType));
-    }
-
-    TickInterval accTicks;
-    for (int i = 0; i < NumOfTests; i++)
-    {
-        for (int j = 0; j < DataLength; j++)
-        {
-            key[j] = TypedRand<KeyType>();
-            data[j].key = key[j];
-            // Write index to payload. We use this later for verification.
-            sprintf(data[j].data, "%d", j);
-        }
-
-        memcpy(dataRef, data, sizeof(data));
-        memcpy(keyRef, key, sizeof(key));
-
-        retVal = 0;
-        TickTime t0 = TickTime::Now();
-        if (keySort)
-        {
-            retVal = webrtc::KeySort(data, key, DataLength, sizeof(LotsOfData<KeyType>),
-                sortType);
-
-            //std::sort(data, data + DataLength, KeyLessThan<KeyType>());
-            //qsort(data, DataLength, sizeof(LotsOfData<KeyType>),
-            //    CompareKey<LotsOfData<KeyType>, KeyType>);
-        }
-        else
-        {
-            retVal = webrtc::Sort(key, DataLength, sortType);
-
-            //std::sort(key, key + DataLength);
-            //qsort(key, DataLength, sizeof(KeyType), Compare<KeyType>);
-        }
-        TickTime t1 = TickTime::Now();
-        accTicks += (t1 - t0);
-
-        if (retVal != 0)
-        {
-            printf("Test failed at iteration %d:\n", i);
-            printf("Sort returned an error. ");
-            printf("It likely does not support the requested type\nExiting...\n");
-            exit(0);
-        }
-
-        // Reference sort.
-        if (!keySort)
-        {
-            std::sort(keyRef, keyRef + DataLength);
-        }
-
-        if (keySort)
-        {
-            for (int j = 0; j < DataLength - 1; j++)
-            {
-                if (data[j].key > data[j + 1].key)
-                {
-                    printf("Test failed at iteration %d:\n", i);
-                    printf("Keys are not monotonically increasing\nExiting...\n");
-                    exit(0);
-                }
-
-                int index = atoi(data[j].data);
-                if (index < 0 || index >= DataLength || data[j].key != dataRef[index].key)
-                {
-                    printf("Test failed at iteration %d:\n", i);
-                    printf("Payload data is corrupt\nExiting...\n");
-                    exit(0);
-                }
-            }
-        }
-        else
-        {
-            for (int j = 0; j < DataLength - 1; j++)
-            {
-                if (key[j] > key[j + 1])
-                {
-                    printf("Test failed at iteration %d:\n", i);
-                    printf("Data is not monotonically increasing\nExiting...\n");
-                    exit(0);
-                }
-            }
-
-            if (memcmp(key, keyRef, sizeof(key)) != 0)
-            {
-                printf("Test failed at iteration %d:\n", i);
-                printf("Sort data differs from std::sort reference\nExiting...\n");
-                exit(0);
-            }
-        }
-    }
-
-    printf("Compliance test passed over %d iterations\n", NumOfTests);
-
-    WebRtc_Word64 executeTime = accTicks.Milliseconds();
-    printf("Execute time: %.2f s\n\n", (float)executeTime / 1000);
-}
-
-int main()
-{
-    // Seed rand().
-    srand(42);
-    bool keySort = false;
-    for (int i = 0; i < 2; i++) {
-        RunSortTest<WebRtc_Word8>(webrtc::TYPE_Word8, keySort);
-        RunSortTest<WebRtc_UWord8>(webrtc::TYPE_UWord8, keySort);
-        RunSortTest<WebRtc_Word16>(webrtc::TYPE_Word16, keySort);
-        RunSortTest<WebRtc_UWord16>(webrtc::TYPE_UWord16, keySort);
-        RunSortTest<WebRtc_Word32>(webrtc::TYPE_Word32, keySort);
-        RunSortTest<WebRtc_UWord32>(webrtc::TYPE_UWord32, keySort);
-        RunSortTest<WebRtc_Word64>(webrtc::TYPE_Word64, keySort);
-        RunSortTest<WebRtc_UWord64>(webrtc::TYPE_UWord64, keySort);
-        RunSortTest<float>(webrtc::TYPE_Float32, keySort);
-        RunSortTest<double>(webrtc::TYPE_Float64, keySort);
-
-        keySort = !keySort;
-    }
-
-    printf("All tests passed\n");
-
-    return 0;
-}
diff --git a/src/system_wrappers/test/list/list.cc b/src/system_wrappers/test/list/list.cc
deleted file mode 100644
index 5c4f0b9..0000000
--- a/src/system_wrappers/test/list/list.cc
+++ /dev/null
@@ -1,174 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "list_wrapper.h"
-
-const int kNumberOfElements = 10;
-
-void FailTest(bool failed)
-{
-    if (failed)
-    {
-        printf("Test failed!\n");
-        printf("Press enter to continue:");
-        getchar();
-        exit(0);
-    }
-}
-
-int GetStoredIntegerValue(ListItem* list_item)
-{
-    void* list_item_pointer = list_item->GetItem();
-    if (list_item_pointer != NULL)
-    {
-        return *(reinterpret_cast<int*>(list_item_pointer));
-    }
-    return static_cast<int>(list_item->GetUnsignedItem());
-}
-
-void PrintList(ListWrapper& list)
-{
-    ListItem* list_item = list.First();
-    printf("List: ");
-    while (list_item != NULL)
-    {
-        int item_value = GetStoredIntegerValue(list_item);
-        FailTest(item_value < 0);
-        printf(" %d",item_value);
-        list_item = list.Next(list_item);
-    }
-    printf("\n");
-}
-
-// The list should always be in ascending order
-void ListSanity(ListWrapper& list)
-{
-    if(list.Empty())
-    {
-      return;
-    }
-    ListItem* item_iter = list.First();
-    // Fake a previous value for the first iteration
-    int previous_value = GetStoredIntegerValue(item_iter) - 1;
-    while (item_iter != NULL)
-    {
-        const int value = GetStoredIntegerValue(item_iter);
-        FailTest(value != previous_value + 1);
-        previous_value = value;
-        item_iter = list.Next(item_iter);
-    }
-}
-
-int main(int /*argc*/, char* /*argv*/[])
-{
-    printf("List Test:\n");
-    int element_array[kNumberOfElements];
-    for (int i = 0; i < kNumberOfElements; i++)
-    {
-        element_array[i] = i;
-    }
-    // Test PushBack 1
-    ListWrapper test_list;
-    for (int i = 2; i < kNumberOfElements - 2; i++)
-    {
-        FailTest(test_list.PushBack((void*)&element_array[i]) != 0);
-    }
-    // Test PushBack 2
-    FailTest(test_list.PushBack(element_array[kNumberOfElements - 2]) != 0);
-    FailTest(test_list.PushBack(element_array[kNumberOfElements - 1]) != 0);
-    // Test PushFront 2
-    FailTest(test_list.PushFront(element_array[1]) != 0);
-    // Test PushFront 1
-    FailTest(test_list.PushFront((void*)&element_array[0]) != 0);
-    // Test GetSize
-    FailTest(test_list.GetSize() != kNumberOfElements);
-    PrintList(test_list);
-    //Test PopFront
-    FailTest(test_list.PopFront() != 0);
-    //Test PopBack
-    FailTest(test_list.PopBack() != 0);
-    // Test GetSize
-    FailTest(test_list.GetSize() != kNumberOfElements - 2);
-    // Test Empty
-    FailTest(test_list.Empty());
-    // Test First
-    ListItem* first_item = test_list.First();
-    FailTest(first_item == NULL);
-    // Test Last
-    ListItem* last_item = test_list.Last();
-    FailTest(last_item == NULL);
-    // Test Next
-    ListItem* second_item = test_list.Next(first_item);
-    FailTest(second_item == NULL);
-    FailTest(test_list.Next(last_item) != NULL);
-    FailTest(test_list.Next(NULL) != NULL);
-    // Test Previous
-    ListItem* second_to_last_item = test_list.Previous(last_item);
-    FailTest(second_to_last_item == NULL);
-    FailTest(test_list.Previous(first_item) != NULL);
-    FailTest(test_list.Previous(NULL) != NULL);
-    // Test GetUnsignedItem
-    FailTest(last_item->GetUnsignedItem() !=
-             kNumberOfElements - 2);
-    FailTest(last_item->GetItem() !=
-             NULL);
-    // Test GetItem
-    FailTest(GetStoredIntegerValue(second_to_last_item) !=
-             kNumberOfElements - 3);
-    FailTest(second_to_last_item->GetUnsignedItem() != 0);
-    // Pop last and first since they are pushed as unsigned items.
-    FailTest(test_list.PopFront() != 0);
-    FailTest(test_list.PopBack() != 0);
-    // Test Insert. Please note that old iterators are no longer valid at
-    // this point.
-    ListItem* insert_item_last = new ListItem(reinterpret_cast<void*>(&element_array[kNumberOfElements - 2]));
-    FailTest(test_list.Insert(test_list.Last(),insert_item_last) != 0);
-    FailTest(test_list.Insert(NULL,insert_item_last) == 0);
-    ListItem* insert_item_last2 = new ListItem(reinterpret_cast<void*>(&element_array[kNumberOfElements - 2]));
-    FailTest(test_list.Insert(insert_item_last2,NULL) == 0);
-    // test InsertBefore
-    ListItem* insert_item_first = new ListItem(reinterpret_cast<void*>(&element_array[1]));
-    FailTest(test_list.InsertBefore(test_list.First(),insert_item_first) != 0);
-    FailTest(test_list.InsertBefore(NULL,insert_item_first) == 0);
-    ListItem* insert_item_first2 = new ListItem(reinterpret_cast<void*>(&element_array[1]));
-    FailTest(test_list.InsertBefore(insert_item_first2,NULL) == 0);
-    PrintList(test_list);
-    ListSanity(test_list);
-    // Erase the whole list
-    int counter = 0;
-    while (test_list.PopFront() == 0)
-    {
-        FailTest(counter++ > kNumberOfElements);
-    }
-    PrintList(test_list);
-    // Test APIs when list is empty
-    FailTest(test_list.GetSize() != 0);
-    FailTest(test_list.PopFront() != -1);
-    FailTest(test_list.PopBack() != -1);
-    FailTest(!test_list.Empty());
-    FailTest(test_list.First() != NULL);
-    FailTest(test_list.Last() != NULL);
-    FailTest(test_list.Next(NULL) != NULL);
-    FailTest(test_list.Previous(NULL) != NULL);
-    FailTest(test_list.Erase(NULL) != -1);
-    // Test Insert APIs when list is empty
-    ListItem* new_item = new ListItem(reinterpret_cast<void*>(&element_array[0]));
-    FailTest(test_list.Insert(NULL,new_item) != 0);
-    FailTest(test_list.Empty());
-    FailTest(test_list.PopFront() != 0);
-    ListItem* new_item2 = new ListItem(reinterpret_cast<void*>(&element_array[0]));
-    FailTest(test_list.InsertBefore(NULL,new_item2) != 0);
-    FailTest(test_list.Empty());
-
-    printf("Tests passed successfully!\n");
-}
diff --git a/src/system_wrappers/test/map/map.cc b/src/system_wrappers/test/map/map.cc
deleted file mode 100644
index 8a0d3e3..0000000
--- a/src/system_wrappers/test/map/map.cc
+++ /dev/null
@@ -1,112 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "map_wrapper.h"
-
-const int kNumberOfElements = 10;
-
-void FailTest(bool failed)
-{
-    if (failed)
-    {
-        printf("Test failed!\n");
-        printf("Press enter to continue:");
-        getchar();
-        exit(0);
-    }
-}
-
-int GetStoredIntegerValue(MapItem* map_item)
-{
-    void* map_item_pointer = map_item->GetItem();
-    if (map_item_pointer != NULL)
-    {
-        return *(reinterpret_cast<int*>(map_item_pointer));
-    }
-    return static_cast<int>(map_item->GetUnsignedId());
-}
-
-void PrintMap(MapWrapper& map)
-{
-    MapItem* map_item = map.First();
-    printf("Map: ");
-    while (map_item != NULL)
-    {
-      int item_value = GetStoredIntegerValue(map_item);
-        FailTest(item_value < 0);
-        printf(" %d",item_value);
-        map_item = map.Next(map_item);
-    }
-    printf("\n");
-}
-
-int main(int /*argc*/, char* /*argv*/[])
-{
-    int element_array[kNumberOfElements];
-    for (int i = 0; i < kNumberOfElements; i++)
-    {
-        element_array[i] = i;
-    }
-    // Test insert
-    MapWrapper test_map;
-    for (int i = 0; i < kNumberOfElements; i++)
-    {
-        test_map.Insert(i,(void*)&element_array[i]);
-    }
-    // Test Erase1
-    MapItem* remove_item = test_map.Find(2);
-    FailTest(remove_item == NULL);
-    FailTest(test_map.Erase(remove_item) != 0);
-    FailTest(test_map.Find(2) != NULL);
-    remove_item = NULL;
-    FailTest(test_map.Erase(remove_item) != -1);
-    // Test Erase2
-    FailTest(test_map.Erase(1) != 0);
-    FailTest(test_map.Find(1) != NULL);
-    FailTest(test_map.Erase(1) != -1);
-    // Test Size
-    FailTest(test_map.Size() != kNumberOfElements - 2);
-    PrintMap(test_map);
-    // Test First
-    MapItem* first_item = test_map.First();
-    FailTest(first_item == NULL);
-    FailTest(GetStoredIntegerValue(first_item) != 0);
-    // Test Last
-    MapItem* last_item = test_map.Last();
-    FailTest(last_item == NULL);
-    FailTest(GetStoredIntegerValue(last_item) != 9);
-    // Test Next
-    MapItem* second_item = test_map.Next(first_item);
-    FailTest(second_item == NULL);
-    FailTest(GetStoredIntegerValue(second_item) != 3);
-    FailTest(test_map.Next(last_item) != NULL);
-    // Test Previous
-    MapItem* second_to_last_item = test_map.Previous(last_item);
-    FailTest(second_to_last_item == NULL);
-    FailTest(GetStoredIntegerValue(second_to_last_item) != 8);
-    FailTest(test_map.Previous(first_item) != NULL);
-    // Test Find (only improper usage untested)
-    FailTest(test_map.Find(kNumberOfElements + 2) != NULL);
-    // Test GetId
-    FailTest(*(reinterpret_cast<int*>(second_to_last_item->GetItem())) !=
-         second_to_last_item->GetId());
-    FailTest(second_to_last_item->GetUnsignedId() !=
-             static_cast<unsigned int>(second_to_last_item->GetId()));
-    // Test SetItem
-    int swapped_item = kNumberOfElements;
-    last_item->SetItem(reinterpret_cast<void*>(&swapped_item));
-    FailTest(GetStoredIntegerValue(last_item) !=
-             swapped_item);
-
-    printf("Tests passed successfully!\n");
-}
diff --git a/src/typedefs.h b/src/typedefs.h
deleted file mode 100644
index ae71690..0000000
--- a/src/typedefs.h
+++ /dev/null
@@ -1,107 +0,0 @@
-/*
- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- *
- * This file contains type definitions used in all WebRtc APIs.
- * 
- */
-
-/* Reserved words definitions */
-#define WEBRTC_EXTERN extern
-#define G_CONST const
-#define WEBRTC_INLINE extern __inline
-
-#ifndef WEBRTC_TYPEDEFS_H
-#define WEBRTC_TYPEDEFS_H
-
-/* Define WebRtc preprocessor identifiers based on the current build platform */
-#if defined(WIN32)
-    // Windows & Windows Mobile
-    #if !defined(WEBRTC_TARGET_PC)
-        #define WEBRTC_TARGET_PC
-    #endif
-#elif defined(__APPLE__)
-    // Mac OS X
-    #if defined(__LITTLE_ENDIAN__ ) //TODO: is this used?
-        #if !defined(WEBRTC_TARGET_MAC_INTEL)
-            #define WEBRTC_TARGET_MAC_INTEL
-        #endif  
-    #else
-        #if !defined(WEBRTC_TARGET_MAC)
-            #define WEBRTC_TARGET_MAC
-        #endif  
-    #endif
-#else
-    // Linux etc.
-    #if !defined(WEBRTC_TARGET_PC)
-        #define WEBRTC_TARGET_PC
-    #endif
-#endif
-
-#if defined(WEBRTC_TARGET_PC)
-
-#if !defined(_MSC_VER)
-  #include <stdint.h>
-#else
-    // Define C99 equivalent types.
-    // Since MSVC doesn't include these headers, we have to write our own
-    // version to provide a compatibility layer between MSVC and the WebRTC
-    // headers.
-    typedef signed char         int8_t;
-    typedef signed short        int16_t;
-    typedef signed int          int32_t;
-    typedef signed long long    int64_t;
-    typedef unsigned char       uint8_t;
-    typedef unsigned short      uint16_t;
-    typedef unsigned int        uint32_t;
-    typedef unsigned long long  uint64_t;
-#endif
-
-#if defined(WIN32)
-    typedef __int64             WebRtc_Word64;
-    typedef unsigned __int64    WebRtc_UWord64;
-#else
-    typedef int64_t             WebRtc_Word64;
-    typedef uint64_t            WebRtc_UWord64;
-#endif
-    typedef int32_t             WebRtc_Word32;
-    typedef uint32_t            WebRtc_UWord32;
-    typedef int16_t             WebRtc_Word16;
-    typedef uint16_t            WebRtc_UWord16;
-    typedef char                WebRtc_Word8;
-    typedef uint8_t             WebRtc_UWord8;
-
-    /* Define endian for the platform */
-    #define WEBRTC_LITTLE_ENDIAN
-
-#elif defined(WEBRTC_TARGET_MAC_INTEL)
-    #include <stdint.h>
-
-    typedef int64_t             WebRtc_Word64;
-    typedef uint64_t            WebRtc_UWord64;
-    typedef int32_t             WebRtc_Word32;
-    typedef uint32_t            WebRtc_UWord32;
-    typedef int16_t             WebRtc_Word16;
-    typedef char                WebRtc_Word8;
-    typedef uint16_t            WebRtc_UWord16;
-    typedef uint8_t             WebRtc_UWord8;
-
-    /* Define endian for the platform */
-    #define WEBRTC_LITTLE_ENDIAN
-
-#else
-
-    #error "No platform defined for WebRtc type definitions (webrtc_typedefs.h)"
-
-#endif
-
-
-#endif // WEBRTC_TYPEDEFS_H