Merge "Fix audio preprocessing library wrapper"
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index e9642f7..11e94e8 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -35,7 +35,7 @@
record.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright liblog libutils libbinder
+ libstagefright liblog libutils libbinder libstagefright_foundation
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
@@ -59,7 +59,7 @@
recordvideo.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright liblog libutils libbinder
+ libstagefright liblog libutils libbinder libstagefright_foundation
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
@@ -84,7 +84,7 @@
audioloop.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright liblog libutils libbinder
+ libstagefright liblog libutils libbinder libstagefright_foundation
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
diff --git a/cmds/stagefright/SineSource.cpp b/cmds/stagefright/SineSource.cpp
index 021f636..14b4306 100644
--- a/cmds/stagefright/SineSource.cpp
+++ b/cmds/stagefright/SineSource.cpp
@@ -3,7 +3,7 @@
#include <math.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index 858681f..a6362a4 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp
@@ -2,10 +2,10 @@
#include <binder/ProcessState.h>
#include <media/mediarecorder.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AMRWriter.h>
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/AudioSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXClient.h>
@@ -24,7 +24,7 @@
android::ProcessState::self()->startThreadPool();
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
#if 0
sp<MediaSource> source = new SineSource(kSampleRate, kNumChannels);
@@ -82,7 +82,7 @@
delete player;
player = NULL;
#elif 0
- CHECK_EQ(decoder->start(), OK);
+ CHECK_EQ(decoder->start(), (status_t)OK);
MediaBuffer *buffer;
while (decoder->read(&buffer) == OK) {
@@ -95,7 +95,7 @@
buffer = NULL;
}
- CHECK_EQ(decoder->stop(), OK);
+ CHECK_EQ(decoder->stop(), (status_t)OK);
#endif
#endif
diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index 7703058..45c3f7b 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp
@@ -17,11 +17,11 @@
#include "SineSource.h"
#include <binder/ProcessState.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/CameraSource.h>
#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaExtractor.h>
@@ -183,7 +183,7 @@
return 1;
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
status_t err = OK;
@@ -231,14 +231,14 @@
sp<MPEG4Writer> writer = new MPEG4Writer("/sdcard/output.mp4");
writer->addSource(encoder);
writer->setMaxFileDuration(kDurationUs);
- CHECK_EQ(OK, writer->start());
+ CHECK_EQ((status_t)OK, writer->start());
while (!writer->reachedEOS()) {
fprintf(stderr, ".");
usleep(100000);
}
err = writer->stop();
#else
- CHECK_EQ(OK, encoder->start());
+ CHECK_EQ((status_t)OK, encoder->start());
MediaBuffer *buffer;
while (encoder->read(&buffer) == OK) {
@@ -272,7 +272,7 @@
for (int i = 0; i < 100; ++i) {
MediaBuffer *buffer;
status_t err = source->read(&buffer);
- CHECK_EQ(err, OK);
+ CHECK_EQ(err, (status_t)OK);
printf("got a frame, data=%p, size=%d\n",
buffer->data(), buffer->range_length());
@@ -299,7 +299,7 @@
android::ProcessState::self()->startThreadPool();
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
const int32_t kSampleRate = 22050;
const int32_t kNumChannels = 2;
diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index c402286..3bd1fe2 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp
@@ -17,9 +17,9 @@
#include "SineSource.h"
#include <binder/ProcessState.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MPEG4Writer.h>
@@ -243,7 +243,7 @@
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
status_t err = OK;
sp<MediaSource> source =
@@ -283,7 +283,7 @@
sp<MPEG4Writer> writer = new MPEG4Writer(fileName);
writer->addSource(encoder);
int64_t start = systemTime();
- CHECK_EQ(OK, writer->start());
+ CHECK_EQ((status_t)OK, writer->start());
while (!writer->reachedEOS()) {
}
err = writer->stop();
diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
index 7b0b443..02dfc1b 100644
--- a/include/media/AudioEffect.h
+++ b/include/media/AudioEffect.h
@@ -108,7 +108,8 @@
* Returned value
* *descriptor updated with effect descriptor
*/
- static status_t getEffectDescriptor(effect_uuid_t *uuid, effect_descriptor_t *descriptor);
+ static status_t getEffectDescriptor(const effect_uuid_t *uuid,
+ effect_descriptor_t *descriptor) /*const*/;
/*
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index c8c5dba..ca57f9e 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -344,7 +344,6 @@
virtual status_t readyToRun() { return NO_ERROR; }
virtual void onFirstRef() {}
AudioRecord& mReceiver;
- Mutex mLock;
};
bool processAudioBuffer(const sp<ClientRecordThread>& thread);
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 74a1e62..da99620 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -55,8 +55,10 @@
static status_t getMasterMute(bool* mute);
// set/get stream volume on specified output
- static status_t setStreamVolume(audio_stream_type_t stream, float value, int output);
- static status_t getStreamVolume(audio_stream_type_t stream, float* volume, int output);
+ static status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output);
+ static status_t getStreamVolume(audio_stream_type_t stream, float* volume,
+ audio_io_handle_t output);
// mute/unmute stream
static status_t setStreamMute(audio_stream_type_t stream, bool mute);
@@ -217,7 +219,7 @@
// indicate a change in the configuration of an output or input: keeps the cached
// values for output/input parameters upto date in client process
- virtual void ioConfigChanged(int event, int ioHandle, void *param2);
+ virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, void *param2);
};
class AudioPolicyServiceClient: public IBinder::DeathRecipient
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 02c85cd..11db81b 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -273,18 +273,18 @@
* left and right volumes. Levels must be >= 0.0 and <= 1.0.
*/
status_t setVolume(float left, float right);
- void getVolume(float* left, float* right);
+ void getVolume(float* left, float* right) const;
/* Set the send level for this track. An auxiliary effect should be attached
* to the track with attachEffect(). Level must be >= 0.0 and <= 1.0.
*/
status_t setAuxEffectSendLevel(float level);
- void getAuxEffectSendLevel(float* level);
+ void getAuxEffectSendLevel(float* level) const;
/* Set sample rate for this track, mostly used for games' sound effects
*/
status_t setSampleRate(int sampleRate);
- uint32_t getSampleRate();
+ uint32_t getSampleRate() const;
/* Enables looping and sets the start and end points of looping.
*
@@ -299,7 +299,7 @@
* (loopEnd-loopStart) <= framecount()
*/
status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);
- status_t getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount);
+ status_t getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount) const;
/* Sets marker position. When playback reaches the number of frames specified, a callback with
* event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
@@ -315,7 +315,7 @@
* - INVALID_OPERATION: the AudioTrack has no callback installed.
*/
status_t setMarkerPosition(uint32_t marker);
- status_t getMarkerPosition(uint32_t *marker);
+ status_t getMarkerPosition(uint32_t *marker) const;
/* Sets position update period. Every time the number of frames specified has been played,
@@ -333,7 +333,7 @@
* - INVALID_OPERATION: the AudioTrack has no callback installed.
*/
status_t setPositionUpdatePeriod(uint32_t updatePeriod);
- status_t getPositionUpdatePeriod(uint32_t *updatePeriod);
+ status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const;
/* Sets playback head position within AudioTrack buffer. The new position is specified
* in number of frames.
@@ -384,7 +384,7 @@
* Returned value:
* AudioTrack session ID.
*/
- int getSessionId();
+ int getSessionId() const;
/* Attach track auxiliary output to specified effect. Use effectId = 0
* to detach track from effect.
@@ -446,7 +446,6 @@
virtual status_t readyToRun();
virtual void onFirstRef();
AudioTrack& mReceiver;
- Mutex mLock;
};
bool processAudioBuffer(const sp<AudioTrackThread>& thread);
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index 8ae13cc..df83995 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -109,7 +109,7 @@
// *pHandle: updated with the effect handle.
//
////////////////////////////////////////////////////////////////////////////////
-int EffectCreate(effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle);
+int EffectCreate(const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle);
////////////////////////////////////////////////////////////////////////////////
//
@@ -151,7 +151,7 @@
// *pDescriptor: updated with the effect descriptor.
//
////////////////////////////////////////////////////////////////////////////////
-int EffectGetDescriptor(effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor);
+int EffectGetDescriptor(const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor);
////////////////////////////////////////////////////////////////////////////////
//
@@ -167,7 +167,7 @@
// 1 if uuid is equal to EFFECT_UUID_NULL.
//
////////////////////////////////////////////////////////////////////////////////
-int EffectIsNullUuid(effect_uuid_t *pEffectUuid);
+int EffectIsNullUuid(const effect_uuid_t *pEffectUuid);
#if __cplusplus
} // extern "C"
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 760595c..433ce7c 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -27,6 +27,7 @@
#include <media/IAudioTrack.h>
#include <media/IAudioRecord.h>
#include <media/IAudioFlingerClient.h>
+#include <system/audio.h>
#include <hardware/audio_effect.h>
#include <media/IEffect.h>
#include <media/IEffectClient.h>
@@ -53,13 +54,13 @@
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
int *sessionId,
status_t *status) = 0;
virtual sp<IAudioRecord> openRecord(
pid_t pid,
- int input,
+ audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
@@ -71,11 +72,11 @@
/* query the audio hardware state. This state never changes,
* and therefore can be cached.
*/
- virtual uint32_t sampleRate(int output) const = 0;
- virtual int channelCount(int output) const = 0;
- virtual audio_format_t format(int output) const = 0;
- virtual size_t frameCount(int output) const = 0;
- virtual uint32_t latency(int output) const = 0;
+ virtual uint32_t sampleRate(audio_io_handle_t output) const = 0;
+ virtual int channelCount(audio_io_handle_t output) const = 0;
+ virtual audio_format_t format(audio_io_handle_t output) const = 0;
+ virtual size_t frameCount(audio_io_handle_t output) const = 0;
+ virtual uint32_t latency(audio_io_handle_t output) const = 0;
/* set/get the audio hardware state. This will probably be used by
* the preference panel, mostly.
@@ -89,10 +90,12 @@
/* set/get stream type state. This will probably be used by
* the preference panel, mostly.
*/
- virtual status_t setStreamVolume(audio_stream_type_t stream, float value, int output) = 0;
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output) = 0;
virtual status_t setStreamMute(audio_stream_type_t stream, bool muted) = 0;
- virtual float streamVolume(audio_stream_type_t stream, int output) const = 0;
+ virtual float streamVolume(audio_stream_type_t stream,
+ audio_io_handle_t output) const = 0;
virtual bool streamMute(audio_stream_type_t stream) const = 0;
// set audio mode
@@ -102,63 +105,68 @@
virtual status_t setMicMute(bool state) = 0;
virtual bool getMicMute() const = 0;
- virtual status_t setParameters(int ioHandle, const String8& keyValuePairs) = 0;
- virtual String8 getParameters(int ioHandle, const String8& keys) = 0;
+ virtual status_t setParameters(audio_io_handle_t ioHandle,
+ const String8& keyValuePairs) = 0;
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const = 0;
// register a current process for audio output change notifications
virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0;
// retrieve the audio recording buffer size
- virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) = 0;
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const = 0;
- virtual int openOutput(uint32_t *pDevices,
+ virtual audio_io_handle_t openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags) = 0;
- virtual int openDuplicateOutput(int output1, int output2) = 0;
- virtual status_t closeOutput(int output) = 0;
- virtual status_t suspendOutput(int output) = 0;
- virtual status_t restoreOutput(int output) = 0;
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2) = 0;
+ virtual status_t closeOutput(audio_io_handle_t output) = 0;
+ virtual status_t suspendOutput(audio_io_handle_t output) = 0;
+ virtual status_t restoreOutput(audio_io_handle_t output) = 0;
- virtual int openInput(uint32_t *pDevices,
+ virtual audio_io_handle_t openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
audio_in_acoustics_t acoustics) = 0;
- virtual status_t closeInput(int input) = 0;
+ virtual status_t closeInput(audio_io_handle_t input) = 0;
- virtual status_t setStreamOutput(audio_stream_type_t stream, int output) = 0;
+ virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) = 0;
virtual status_t setVoiceVolume(float volume) = 0;
- virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) = 0;
+ virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const = 0;
- virtual unsigned int getInputFramesLost(int ioHandle) = 0;
+ virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
virtual int newAudioSessionId() = 0;
virtual void acquireAudioSessionId(int audioSession) = 0;
virtual void releaseAudioSessionId(int audioSession) = 0;
- virtual status_t queryNumberEffects(uint32_t *numEffects) = 0;
+ virtual status_t queryNumberEffects(uint32_t *numEffects) const = 0;
- virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) = 0;
+ virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) const = 0;
- virtual status_t getEffectDescriptor(effect_uuid_t *pEffectUUID, effect_descriptor_t *pDescriptor) = 0;
+ virtual status_t getEffectDescriptor(const effect_uuid_t *pEffectUUID,
+ effect_descriptor_t *pDescriptor) const = 0;
virtual sp<IEffect> createEffect(pid_t pid,
effect_descriptor_t *pDesc,
const sp<IEffectClient>& client,
int32_t priority,
- int output,
+ audio_io_handle_t output,
int sessionId,
status_t *status,
int *id,
int *enabled) = 0;
- virtual status_t moveEffects(int session, int srcOutput, int dstOutput) = 0;
+ virtual status_t moveEffects(int session, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput) = 0;
};
diff --git a/include/media/IAudioFlingerClient.h b/include/media/IAudioFlingerClient.h
index aa0cdcf..f3b4df1 100644
--- a/include/media/IAudioFlingerClient.h
+++ b/include/media/IAudioFlingerClient.h
@@ -21,6 +21,7 @@
#include <utils/RefBase.h>
#include <binder/IInterface.h>
#include <utils/KeyedVector.h>
+#include <system/audio.h>
namespace android {
@@ -32,7 +33,7 @@
DECLARE_META_INTERFACE(AudioFlingerClient);
// Notifies a change of audio input/output configuration.
- virtual void ioConfigChanged(int event, int ioHandle, void *param2) = 0;
+ virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, void *param2) = 0;
};
diff --git a/include/media/stagefright/MediaDebug.h b/include/media/stagefright/MediaDebug.h
deleted file mode 100644
index 2ca9667..0000000
--- a/include/media/stagefright/MediaDebug.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (C) 2009 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.
- */
-
-#ifndef MEDIA_DEBUG_H_
-
-#define MEDIA_DEBUG_H_
-
-#include <cutils/log.h>
-
-#define LITERAL_TO_STRING_INTERNAL(x) #x
-#define LITERAL_TO_STRING(x) LITERAL_TO_STRING_INTERNAL(x)
-
-#define CHECK_EQ(x,y) \
- LOG_ALWAYS_FATAL_IF( \
- (x) != (y), \
- __FILE__ ":" LITERAL_TO_STRING(__LINE__) " " #x " != " #y)
-
-#define CHECK(x) \
- LOG_ALWAYS_FATAL_IF( \
- !(x), \
- __FILE__ ":" LITERAL_TO_STRING(__LINE__) " " #x)
-
-#endif // MEDIA_DEBUG_H_
diff --git a/include/media/stagefright/OMXCodec.h b/include/media/stagefright/OMXCodec.h
index 4c30e04..e541c18 100644
--- a/include/media/stagefright/OMXCodec.h
+++ b/include/media/stagefright/OMXCodec.h
@@ -335,7 +335,7 @@
status_t applyRotation();
status_t waitForBufferFilled_l();
- int64_t retrieveDecodingTimeUs(bool isCodecSpecific);
+ int64_t getDecodingTimeUs();
status_t parseAVCCodecSpecificData(
const void *data, size_t size,
diff --git a/include/media/stagefright/SurfaceMediaSource.h b/include/media/stagefright/SurfaceMediaSource.h
index d0940bb..54baab6 100644
--- a/include/media/stagefright/SurfaceMediaSource.h
+++ b/include/media/stagefright/SurfaceMediaSource.h
@@ -58,7 +58,7 @@
// For the MediaSource interface for use by StageFrightRecorder:
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t read(
MediaBuffer **buffer, const ReadOptions *options = NULL);
virtual sp<MetaData> getFormat();
@@ -359,6 +359,8 @@
Condition mFrameAvailableCondition;
Condition mFrameCompleteCondition;
+ status_t reset();
+
// Avoid copying and equating and default constructor
DISALLOW_IMPLICIT_CONSTRUCTORS(SurfaceMediaSource);
};
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index dd97ce4..23226c0 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -54,6 +54,7 @@
#define CBLK_RESTORED_ON 0x0040 // track has been restored after invalidation
#define CBLK_RESTORED_OFF 0x0040 // by AudioFlinger
+// Important: do not add any virtual methods, including ~
struct audio_track_cblk_t
{
diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c
index 9f6599f..59cd9e3 100644
--- a/media/libeffects/factory/EffectsFactory.c
+++ b/media/libeffects/factory/EffectsFactory.c
@@ -53,8 +53,8 @@
static lib_entry_t *getLibrary(const char *path);
static void resetEffectEnumeration();
static uint32_t updateNumEffects();
-static int findEffect(effect_uuid_t *type,
- effect_uuid_t *uuid,
+static int findEffect(const effect_uuid_t *type,
+ const effect_uuid_t *uuid,
lib_entry_t **lib,
effect_descriptor_t **desc);
static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len);
@@ -236,7 +236,7 @@
return ret;
}
-int EffectGetDescriptor(effect_uuid_t *uuid, effect_descriptor_t *pDescriptor)
+int EffectGetDescriptor(const effect_uuid_t *uuid, effect_descriptor_t *pDescriptor)
{
lib_entry_t *l = NULL;
effect_descriptor_t *d = NULL;
@@ -257,7 +257,7 @@
return ret;
}
-int EffectCreate(effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle)
+int EffectCreate(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle)
{
list_elem_t *e = gLibraryList;
lib_entry_t *l = NULL;
@@ -372,7 +372,7 @@
return ret;
}
-int EffectIsNullUuid(effect_uuid_t *uuid)
+int EffectIsNullUuid(const effect_uuid_t *uuid)
{
if (memcmp(uuid, EFFECT_UUID_NULL, sizeof(effect_uuid_t))) {
return 0;
@@ -628,8 +628,8 @@
return cnt;
}
-int findEffect(effect_uuid_t *type,
- effect_uuid_t *uuid,
+int findEffect(const effect_uuid_t *type,
+ const effect_uuid_t *uuid,
lib_entry_t **lib,
effect_descriptor_t **desc)
{
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 108d36a..3714283 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -195,7 +195,7 @@
return 0;
} /* end EffectQueryEffect */
-extern "C" int EffectCreate(effect_uuid_t *uuid,
+extern "C" int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle){
@@ -471,7 +471,7 @@
} /* end EffectRelease */
-extern "C" int EffectGetDescriptor(effect_uuid_t *uuid,
+extern "C" int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
const effect_descriptor_t *desc = NULL;
diff --git a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
index 09cd5cc..358357e 100755
--- a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
+++ b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
@@ -210,7 +210,7 @@
return 0;
} /* end EffectQueryEffect */
-extern "C" int EffectCreate(effect_uuid_t *uuid,
+extern "C" int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle){
@@ -317,7 +317,7 @@
return 0;
} /* end EffectRelease */
-extern "C" int EffectGetDescriptor(effect_uuid_t *uuid,
+extern "C" int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
int i;
int length = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *);
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index f57179d..098a1a2 100755
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -1072,7 +1072,7 @@
return sInitStatus;
}
-const effect_descriptor_t *PreProc_GetDescriptor(effect_uuid_t *uuid)
+const effect_descriptor_t *PreProc_GetDescriptor(const effect_uuid_t *uuid)
{
size_t i;
for (i = 0; i < PREPROC_NUM_EFFECTS; i++) {
@@ -1568,7 +1568,7 @@
return 0;
}
-int PreProcessingLib_Create(effect_uuid_t *uuid,
+int PreProcessingLib_Create(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pInterface)
@@ -1620,7 +1620,7 @@
return Session_ReleaseEffect(fx->session, fx);
}
-int PreProcessingLib_GetDescriptor(effect_uuid_t *uuid,
+int PreProcessingLib_GetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
if (pDescriptor == NULL || uuid == NULL){
diff --git a/media/libeffects/testlibs/EffectEqualizer.cpp b/media/libeffects/testlibs/EffectEqualizer.cpp
index 5241660..35a4a61 100644
--- a/media/libeffects/testlibs/EffectEqualizer.cpp
+++ b/media/libeffects/testlibs/EffectEqualizer.cpp
@@ -140,7 +140,7 @@
return 0;
} /* end EffectQueryNext */
-extern "C" int EffectCreate(effect_uuid_t *uuid,
+extern "C" int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle) {
@@ -195,7 +195,7 @@
return 0;
} /* end EffectRelease */
-extern "C" int EffectGetDescriptor(effect_uuid_t *uuid,
+extern "C" int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
if (pDescriptor == NULL || uuid == NULL){
diff --git a/media/libeffects/testlibs/EffectReverb.c b/media/libeffects/testlibs/EffectReverb.c
index ebb72c1..8351712 100644
--- a/media/libeffects/testlibs/EffectReverb.c
+++ b/media/libeffects/testlibs/EffectReverb.c
@@ -111,7 +111,7 @@
return 0;
}
-int EffectCreate(effect_uuid_t *uuid,
+int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle) {
@@ -182,7 +182,7 @@
return 0;
}
-int EffectGetDescriptor(effect_uuid_t *uuid,
+int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
int i;
int length = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *);
diff --git a/media/libeffects/testlibs/EffectReverb.h b/media/libeffects/testlibs/EffectReverb.h
index 5137074..1fb14a7 100644
--- a/media/libeffects/testlibs/EffectReverb.h
+++ b/media/libeffects/testlibs/EffectReverb.h
@@ -303,12 +303,12 @@
int EffectQueryNumberEffects(uint32_t *pNumEffects);
int EffectQueryEffect(uint32_t index,
effect_descriptor_t *pDescriptor);
-int EffectCreate(effect_uuid_t *effectUID,
+int EffectCreate(const effect_uuid_t *effectUID,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle);
int EffectRelease(effect_handle_t handle);
-int EffectGetDescriptor(effect_uuid_t *uuid,
+int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor);
static int Reverb_Process(effect_handle_t self,
diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp
index 5d70a9b..51c8b68 100644
--- a/media/libeffects/visualizer/EffectVisualizer.cpp
+++ b/media/libeffects/visualizer/EffectVisualizer.cpp
@@ -190,7 +190,7 @@
return 0;
}
-int VisualizerLib_Create(effect_uuid_t *uuid,
+int VisualizerLib_Create(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle) {
@@ -240,7 +240,7 @@
return 0;
}
-int VisualizerLib_GetDescriptor(effect_uuid_t *uuid,
+int VisualizerLib_GetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
if (pDescriptor == NULL || uuid == NULL){
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index a242846..f9f997f 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -159,7 +159,7 @@
mCblk->buffer = (uint8_t *)mCblk + bufOffset;
iEffect->asBinder()->linkToDeath(mIEffectClient);
- ALOGV("set() %p OK effect: %s id: %d status %d enabled %d, ", this, mDescriptor.name, mId, mStatus, mEnabled);
+ ALOGV("set() %p OK effect: %s id: %d status %d enabled %d", this, mDescriptor.name, mId, mStatus, mEnabled);
return mStatus;
}
@@ -412,7 +412,8 @@
return af->queryEffect(index, descriptor);
}
-status_t AudioEffect::getEffectDescriptor(effect_uuid_t *uuid, effect_descriptor_t *descriptor)
+status_t AudioEffect::getEffectDescriptor(const effect_uuid_t *uuid,
+ effect_descriptor_t *descriptor) /*const*/
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index c96bc76..b74b3e3 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -293,7 +293,6 @@
return WOULD_BLOCK;
}
}
- t->mLock.lock();
}
AutoMutex lock(mLock);
@@ -334,10 +333,6 @@
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
-
return ret;
}
@@ -347,10 +342,6 @@
ALOGV("stop");
- if (t != 0) {
- t->mLock.lock();
- }
-
AutoMutex lock(mLock);
if (mActive == 1) {
mActive = 0;
@@ -367,10 +358,6 @@
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
-
return NO_ERROR;
}
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index 110a294..ec4c044 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -121,7 +121,8 @@
return NO_ERROR;
}
-status_t AudioSystem::setStreamVolume(audio_stream_type_t stream, float value, int output)
+status_t AudioSystem::setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -139,7 +140,8 @@
return NO_ERROR;
}
-status_t AudioSystem::getStreamVolume(audio_stream_type_t stream, float* volume, int output)
+status_t AudioSystem::getStreamVolume(audio_stream_type_t stream, float* volume,
+ audio_io_handle_t output)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -402,7 +404,8 @@
ALOGW("AudioFlinger server died!");
}
-void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, int ioHandle, void *param2) {
+void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle_t ioHandle,
+ void *param2) {
ALOGV("ioConfigChanged() event %d", event);
OutputDescriptor *desc;
audio_stream_type_t stream;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 8c33f41..087d7b2 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -345,7 +345,6 @@
return;
}
}
- t->mLock.lock();
}
AutoMutex lock(mLock);
@@ -396,9 +395,6 @@
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
}
void AudioTrack::stop()
@@ -406,9 +402,6 @@
sp<AudioTrackThread> t = mAudioTrackThread;
ALOGV("stop %p", this);
- if (t != 0) {
- t->mLock.lock();
- }
AutoMutex lock(mLock);
if (mActive) {
@@ -434,9 +427,6 @@
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
}
bool AudioTrack::stopped() const
@@ -506,7 +496,7 @@
return NO_ERROR;
}
-void AudioTrack::getVolume(float* left, float* right)
+void AudioTrack::getVolume(float* left, float* right) const
{
if (left != NULL) {
*left = mVolume[LEFT];
@@ -531,7 +521,7 @@
return NO_ERROR;
}
-void AudioTrack::getAuxEffectSendLevel(float* level)
+void AudioTrack::getAuxEffectSendLevel(float* level) const
{
if (level != NULL) {
*level = mSendLevel;
@@ -553,7 +543,7 @@
return NO_ERROR;
}
-uint32_t AudioTrack::getSampleRate()
+uint32_t AudioTrack::getSampleRate() const
{
AutoMutex lock(mLock);
return mCblk->sampleRate;
@@ -601,7 +591,7 @@
return NO_ERROR;
}
-status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
+status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount) const
{
AutoMutex lock(mLock);
if (loopStart != NULL) {
@@ -631,7 +621,7 @@
return NO_ERROR;
}
-status_t AudioTrack::getMarkerPosition(uint32_t *marker)
+status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
{
if (marker == NULL) return BAD_VALUE;
@@ -652,7 +642,7 @@
return NO_ERROR;
}
-status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
+status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
{
if (updatePeriod == NULL) return BAD_VALUE;
@@ -712,7 +702,7 @@
mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags);
}
-int AudioTrack::getSessionId()
+int AudioTrack::getSessionId() const
{
return mSessionId;
}
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index fc5520f..4507e5d 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -89,7 +89,7 @@
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
int *sessionId,
status_t *status)
{
@@ -104,7 +104,7 @@
data.writeInt32(frameCount);
data.writeInt32(flags);
data.writeStrongBinder(sharedBuffer->asBinder());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
int lSessionId = 0;
if (sessionId != NULL) {
lSessionId = *sessionId;
@@ -129,7 +129,7 @@
virtual sp<IAudioRecord> openRecord(
pid_t pid,
- int input,
+ audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
@@ -142,7 +142,7 @@
sp<IAudioRecord> record;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(pid);
- data.writeInt32(input);
+ data.writeInt32((int32_t) input);
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
@@ -170,47 +170,47 @@
return record;
}
- virtual uint32_t sampleRate(int output) const
+ virtual uint32_t sampleRate(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(SAMPLE_RATE, data, &reply);
return reply.readInt32();
}
- virtual int channelCount(int output) const
+ virtual int channelCount(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(CHANNEL_COUNT, data, &reply);
return reply.readInt32();
}
- virtual audio_format_t format(int output) const
+ virtual audio_format_t format(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(FORMAT, data, &reply);
return (audio_format_t) reply.readInt32();
}
- virtual size_t frameCount(int output) const
+ virtual size_t frameCount(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(FRAME_COUNT, data, &reply);
return reply.readInt32();
}
- virtual uint32_t latency(int output) const
+ virtual uint32_t latency(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(LATENCY, data, &reply);
return reply.readInt32();
}
@@ -249,13 +249,14 @@
return reply.readInt32();
}
- virtual status_t setStreamVolume(audio_stream_type_t stream, float value, int output)
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32((int32_t) stream);
data.writeFloat(value);
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(SET_STREAM_VOLUME, data, &reply);
return reply.readInt32();
}
@@ -270,12 +271,12 @@
return reply.readInt32();
}
- virtual float streamVolume(audio_stream_type_t stream, int output) const
+ virtual float streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32((int32_t) stream);
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(STREAM_VOLUME, data, &reply);
return reply.readFloat();
}
@@ -315,21 +316,21 @@
return reply.readInt32();
}
- virtual status_t setParameters(int ioHandle, const String8& keyValuePairs)
+ virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
data.writeString8(keyValuePairs);
remote()->transact(SET_PARAMETERS, data, &reply);
return reply.readInt32();
}
- virtual String8 getParameters(int ioHandle, const String8& keys)
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
data.writeString8(keys);
remote()->transact(GET_PARAMETERS, data, &reply);
return reply.readString8();
@@ -343,7 +344,7 @@
remote()->transact(REGISTER_CLIENT, data, &reply);
}
- virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount)
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -354,7 +355,7 @@
return reply.readInt32();
}
- virtual int openOutput(uint32_t *pDevices,
+ virtual audio_io_handle_t openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
@@ -376,8 +377,8 @@
data.writeInt32(latency);
data.writeInt32(flags);
remote()->transact(OPEN_OUTPUT, data, &reply);
- int output = reply.readInt32();
- ALOGV("openOutput() returned output, %p", output);
+ audio_io_handle_t output = (audio_io_handle_t) reply.readInt32();
+ ALOGV("openOutput() returned output, %d", output);
devices = reply.readInt32();
if (pDevices) *pDevices = devices;
samplingRate = reply.readInt32();
@@ -391,44 +392,45 @@
return output;
}
- virtual int openDuplicateOutput(int output1, int output2)
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output1);
- data.writeInt32(output2);
+ data.writeInt32((int32_t) output1);
+ data.writeInt32((int32_t) output2);
remote()->transact(OPEN_DUPLICATE_OUTPUT, data, &reply);
- return reply.readInt32();
+ return (audio_io_handle_t) reply.readInt32();
}
- virtual status_t closeOutput(int output)
+ virtual status_t closeOutput(audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(CLOSE_OUTPUT, data, &reply);
return reply.readInt32();
}
- virtual status_t suspendOutput(int output)
+ virtual status_t suspendOutput(audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(SUSPEND_OUTPUT, data, &reply);
return reply.readInt32();
}
- virtual status_t restoreOutput(int output)
+ virtual status_t restoreOutput(audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(RESTORE_OUTPUT, data, &reply);
return reply.readInt32();
}
- virtual int openInput(uint32_t *pDevices,
+ virtual audio_io_handle_t openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
@@ -447,7 +449,7 @@
data.writeInt32(channels);
data.writeInt32((int32_t) acoustics);
remote()->transact(OPEN_INPUT, data, &reply);
- int input = reply.readInt32();
+ audio_io_handle_t input = (audio_io_handle_t) reply.readInt32();
devices = reply.readInt32();
if (pDevices) *pDevices = devices;
samplingRate = reply.readInt32();
@@ -468,12 +470,12 @@
return reply.readInt32();
}
- virtual status_t setStreamOutput(audio_stream_type_t stream, int output)
+ virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32((int32_t) stream);
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(SET_STREAM_OUTPUT, data, &reply);
return reply.readInt32();
}
@@ -487,11 +489,12 @@
return reply.readInt32();
}
- virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
+ virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(GET_RENDER_POSITION, data, &reply);
status_t status = reply.readInt32();
if (status == NO_ERROR) {
@@ -507,11 +510,11 @@
return status;
}
- virtual unsigned int getInputFramesLost(int ioHandle)
+ virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
return reply.readInt32();
}
@@ -544,7 +547,7 @@
remote()->transact(RELEASE_AUDIO_SESSION_ID, data, &reply);
}
- virtual status_t queryNumberEffects(uint32_t *numEffects)
+ virtual status_t queryNumberEffects(uint32_t *numEffects) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -556,13 +559,13 @@
if (status != NO_ERROR) {
return status;
}
- if (numEffects) {
+ if (numEffects != NULL) {
*numEffects = (uint32_t)reply.readInt32();
}
return NO_ERROR;
}
- virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor)
+ virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) const
{
if (pDescriptor == NULL) {
return BAD_VALUE;
@@ -582,7 +585,8 @@
return NO_ERROR;
}
- virtual status_t getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *pDescriptor)
+ virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
+ effect_descriptor_t *pDescriptor) const
{
if (pUuid == NULL || pDescriptor == NULL) {
return BAD_VALUE;
@@ -606,7 +610,7 @@
effect_descriptor_t *pDesc,
const sp<IEffectClient>& client,
int32_t priority,
- int output,
+ audio_io_handle_t output,
int sessionId,
status_t *status,
int *id,
@@ -627,7 +631,7 @@
data.write(pDesc, sizeof(effect_descriptor_t));
data.writeStrongBinder(client->asBinder());
data.writeInt32(priority);
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
data.writeInt32(sessionId);
status_t lStatus = remote()->transact(CREATE_EFFECT, data, &reply);
@@ -653,13 +657,14 @@
return effect;
}
- virtual status_t moveEffects(int session, int srcOutput, int dstOutput)
+ virtual status_t moveEffects(int session, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(session);
- data.writeInt32(srcOutput);
- data.writeInt32(dstOutput);
+ data.writeInt32((int32_t) srcOutput);
+ data.writeInt32((int32_t) dstOutput);
remote()->transact(MOVE_EFFECTS, data, &reply);
return reply.readInt32();
}
@@ -683,7 +688,7 @@
size_t bufferCount = data.readInt32();
uint32_t flags = data.readInt32();
sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
- int output = data.readInt32();
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
int sessionId = data.readInt32();
status_t status;
sp<IAudioTrack> track = createTrack(pid,
@@ -697,7 +702,7 @@
case OPEN_RECORD: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
pid_t pid = data.readInt32();
- int input = data.readInt32();
+ audio_io_handle_t input = (audio_io_handle_t) data.readInt32();
uint32_t sampleRate = data.readInt32();
audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
@@ -714,27 +719,27 @@
} break;
case SAMPLE_RATE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( sampleRate(data.readInt32()) );
+ reply->writeInt32( sampleRate((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case CHANNEL_COUNT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( channelCount(data.readInt32()) );
+ reply->writeInt32( channelCount((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case FORMAT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( format(data.readInt32()) );
+ reply->writeInt32( format((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case FRAME_COUNT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( frameCount(data.readInt32()) );
+ reply->writeInt32( frameCount((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case LATENCY: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( latency(data.readInt32()) );
+ reply->writeInt32( latency((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case SET_MASTER_VOLUME: {
@@ -761,7 +766,7 @@
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
float volume = data.readFloat();
- int output = data.readInt32();
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
reply->writeInt32( setStreamVolume((audio_stream_type_t) stream, volume, output) );
return NO_ERROR;
} break;
@@ -803,14 +808,14 @@
} break;
case SET_PARAMETERS: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int ioHandle = data.readInt32();
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
String8 keyValuePairs(data.readString8());
reply->writeInt32(setParameters(ioHandle, keyValuePairs));
return NO_ERROR;
} break;
case GET_PARAMETERS: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int ioHandle = data.readInt32();
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
String8 keys(data.readString8());
reply->writeString8(getParameters(ioHandle, keys));
return NO_ERROR;
@@ -838,14 +843,14 @@
uint32_t channels = data.readInt32();
uint32_t latency = data.readInt32();
uint32_t flags = data.readInt32();
- int output = openOutput(&devices,
+ audio_io_handle_t output = openOutput(&devices,
&samplingRate,
&format,
&channels,
&latency,
flags);
ALOGV("OPEN_OUTPUT output, %p", output);
- reply->writeInt32(output);
+ reply->writeInt32((int32_t) output);
reply->writeInt32(devices);
reply->writeInt32(samplingRate);
reply->writeInt32(format);
@@ -855,24 +860,24 @@
} break;
case OPEN_DUPLICATE_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int output1 = data.readInt32();
- int output2 = data.readInt32();
- reply->writeInt32(openDuplicateOutput(output1, output2));
+ audio_io_handle_t output1 = (audio_io_handle_t) data.readInt32();
+ audio_io_handle_t output2 = (audio_io_handle_t) data.readInt32();
+ reply->writeInt32((int32_t) openDuplicateOutput(output1, output2));
return NO_ERROR;
} break;
case CLOSE_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(closeOutput(data.readInt32()));
+ reply->writeInt32(closeOutput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case SUSPEND_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(suspendOutput(data.readInt32()));
+ reply->writeInt32(suspendOutput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case RESTORE_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(restoreOutput(data.readInt32()));
+ reply->writeInt32(restoreOutput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case OPEN_INPUT: {
@@ -883,12 +888,12 @@
uint32_t channels = data.readInt32();
audio_in_acoustics_t acoustics = (audio_in_acoustics_t) data.readInt32();
- int input = openInput(&devices,
+ audio_io_handle_t input = openInput(&devices,
&samplingRate,
&format,
&channels,
acoustics);
- reply->writeInt32(input);
+ reply->writeInt32((int32_t) input);
reply->writeInt32(devices);
reply->writeInt32(samplingRate);
reply->writeInt32(format);
@@ -897,13 +902,13 @@
} break;
case CLOSE_INPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(closeInput(data.readInt32()));
+ reply->writeInt32(closeInput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case SET_STREAM_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t stream = data.readInt32();
- int output = data.readInt32();
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
reply->writeInt32(setStreamOutput((audio_stream_type_t) stream, output));
return NO_ERROR;
} break;
@@ -915,7 +920,7 @@
} break;
case GET_RENDER_POSITION: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int output = data.readInt32();
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
uint32_t halFrames;
uint32_t dspFrames;
status_t status = getRenderPosition(&halFrames, &dspFrames, output);
@@ -928,7 +933,7 @@
}
case GET_INPUT_FRAMES_LOST: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int ioHandle = data.readInt32();
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
reply->writeInt32(getInputFramesLost(ioHandle));
return NO_ERROR;
} break;
@@ -988,7 +993,7 @@
data.read(&desc, sizeof(effect_descriptor_t));
sp<IEffectClient> client = interface_cast<IEffectClient>(data.readStrongBinder());
int32_t priority = data.readInt32();
- int output = data.readInt32();
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
int sessionId = data.readInt32();
status_t status;
int id;
@@ -1005,8 +1010,8 @@
case MOVE_EFFECTS: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int session = data.readInt32();
- int srcOutput = data.readInt32();
- int dstOutput = data.readInt32();
+ audio_io_handle_t srcOutput = (audio_io_handle_t) data.readInt32();
+ audio_io_handle_t dstOutput = (audio_io_handle_t) data.readInt32();
reply->writeInt32(moveEffects(session, srcOutput, dstOutput));
return NO_ERROR;
} break;
diff --git a/media/libmedia/IAudioFlingerClient.cpp b/media/libmedia/IAudioFlingerClient.cpp
index 9458bc0..ce28b33 100644
--- a/media/libmedia/IAudioFlingerClient.cpp
+++ b/media/libmedia/IAudioFlingerClient.cpp
@@ -39,12 +39,12 @@
{
}
- void ioConfigChanged(int event, int ioHandle, void *param2)
+ void ioConfigChanged(int event, audio_io_handle_t ioHandle, void *param2)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlingerClient::getInterfaceDescriptor());
data.writeInt32(event);
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
if (event == AudioSystem::STREAM_CONFIG_CHANGED) {
uint32_t stream = *(uint32_t *)param2;
ALOGV("ioConfigChanged stream %d", stream);
@@ -72,7 +72,7 @@
case IO_CONFIG_CHANGED: {
CHECK_INTERFACE(IAudioFlingerClient, data, reply);
int event = data.readInt32();
- int ioHandle = data.readInt32();
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
void *param2 = NULL;
AudioSystem::OutputDescriptor desc;
uint32_t stream;
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index 8456db5..6cb5b82 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -246,14 +246,12 @@
}//while (1)
threadExit:
- if (mAudioTrack) {
+ if (mAudioTrack != NULL) {
mAudioTrack->stop();
mAudioTrack->flush();
}
- if (mAudioBuffer) {
- delete [] mAudioBuffer;
- mAudioBuffer = NULL;
- }
+ delete [] mAudioBuffer;
+ mAudioBuffer = NULL;
mMutex.lock();
mTid = -1;
mCondition.signal();
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index c905762..93ddca8 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -25,7 +25,7 @@
#include <cutils/properties.h>
#include <expat.h>
#include <media/MediaProfiles.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/openmax/OMX_Video.h>
namespace android {
@@ -349,7 +349,7 @@
{
CHECK(!strcmp("quality", atts[0]));
int quality = atoi(atts[1]);
- ALOGV("%s: cameraId=%d, quality=%d\n", __func__, cameraId, quality);
+ ALOGV("%s: cameraId=%d, quality=%d", __func__, cameraId, quality);
ImageEncodingQualityLevels *levels = findImageEncodingQualityLevels(cameraId);
if (levels == NULL) {
diff --git a/media/libmedia/MediaScanner.cpp b/media/libmedia/MediaScanner.cpp
index 79cab74..73d4519 100644
--- a/media/libmedia/MediaScanner.cpp
+++ b/media/libmedia/MediaScanner.cpp
@@ -143,7 +143,7 @@
if (pathRemaining >= 8 /* strlen(".nomedia") */ ) {
strcpy(fileSpot, ".nomedia");
if (access(path, F_OK) == 0) {
- ALOGV("found .nomedia, setting noMedia flag\n");
+ ALOGV("found .nomedia, setting noMedia flag");
noMedia = true;
}
diff --git a/media/libmedia/MediaScannerClient.cpp b/media/libmedia/MediaScannerClient.cpp
index 9fe1820..cdfd477 100644
--- a/media/libmedia/MediaScannerClient.cpp
+++ b/media/libmedia/MediaScannerClient.cpp
@@ -142,12 +142,12 @@
UConverter *conv = ucnv_open(enc, &status);
if (U_FAILURE(status)) {
- ALOGE("could not create UConverter for %s\n", enc);
+ ALOGE("could not create UConverter for %s", enc);
return;
}
UConverter *utf8Conv = ucnv_open("UTF-8", &status);
if (U_FAILURE(status)) {
- ALOGE("could not create UConverter for UTF-8\n");
+ ALOGE("could not create UConverter for UTF-8");
ucnv_close(conv);
return;
}
@@ -181,7 +181,7 @@
ucnv_convertEx(utf8Conv, conv, &target, target + targetLength,
&source, (const char *)dest, NULL, NULL, NULL, NULL, TRUE, TRUE, &status);
if (U_FAILURE(status)) {
- ALOGE("ucnv_convertEx failed: %d\n", status);
+ ALOGE("ucnv_convertEx failed: %d", status);
mValues->setEntry(i, "???");
} else {
// zero terminate
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index e6e989d..6cb10aa 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -800,7 +800,7 @@
////////////////////////////////////////////////////////////////////////////////
ToneGenerator::ToneGenerator(audio_stream_type_t streamType, float volume, bool threadCanCallJava) {
- ALOGV("ToneGenerator constructor: streamType=%d, volume=%f\n", streamType, volume);
+ ALOGV("ToneGenerator constructor: streamType=%d, volume=%f", streamType, volume);
mState = TONE_IDLE;
@@ -829,9 +829,9 @@
}
if (initAudioTrack()) {
- ALOGV("ToneGenerator INIT OK, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("ToneGenerator INIT OK, time: %d", (unsigned int)(systemTime()/1000000));
} else {
- ALOGV("!!!ToneGenerator INIT FAILED!!!\n");
+ ALOGV("!!!ToneGenerator INIT FAILED!!!");
}
}
@@ -853,11 +853,11 @@
//
////////////////////////////////////////////////////////////////////////////////
ToneGenerator::~ToneGenerator() {
- ALOGV("ToneGenerator destructor\n");
+ ALOGV("ToneGenerator destructor");
if (mpAudioTrack != NULL) {
stopTone();
- ALOGV("Delete Track: %p\n", mpAudioTrack);
+ ALOGV("Delete Track: %p", mpAudioTrack);
delete mpAudioTrack;
}
}
@@ -892,7 +892,7 @@
}
}
- ALOGV("startTone\n");
+ ALOGV("startTone");
mLock.lock();
@@ -915,7 +915,7 @@
if (mState == TONE_INIT) {
if (prepareWave()) {
- ALOGV("Immediate start, time %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("Immediate start, time %d", (unsigned int)(systemTime()/1000000));
lResult = true;
mState = TONE_STARTING;
mLock.unlock();
@@ -934,7 +934,7 @@
mState = TONE_IDLE;
}
} else {
- ALOGV("Delayed start\n");
+ ALOGV("Delayed start");
mState = TONE_RESTARTING;
lStatus = mWaitCbkCond.waitRelative(mLock, seconds(3));
if (lStatus == NO_ERROR) {
@@ -949,8 +949,8 @@
}
mLock.unlock();
- ALOGV_IF(lResult, "Tone started, time %d\n", (unsigned int)(systemTime()/1000000));
- ALOGW_IF(!lResult, "Tone start failed!!!, time %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV_IF(lResult, "Tone started, time %d", (unsigned int)(systemTime()/1000000));
+ ALOGW_IF(!lResult, "Tone start failed!!!, time %d", (unsigned int)(systemTime()/1000000));
return lResult;
}
@@ -1017,7 +1017,7 @@
// Open audio track in mono, PCM 16bit, default sampling rate, default buffer size
mpAudioTrack = new AudioTrack();
- ALOGV("Create Track: %p\n", mpAudioTrack);
+ ALOGV("Create Track: %p", mpAudioTrack);
mpAudioTrack->set(mStreamType,
0,
@@ -1045,8 +1045,8 @@
initAudioTrack_exit:
// Cleanup
- if (mpAudioTrack) {
- ALOGV("Delete Track I: %p\n", mpAudioTrack);
+ if (mpAudioTrack != NULL) {
+ ALOGV("Delete Track I: %p", mpAudioTrack);
delete mpAudioTrack;
mpAudioTrack = NULL;
}
@@ -1141,7 +1141,7 @@
if (lpToneGen->mTotalSmp > lpToneGen->mNextSegSmp) {
// Time to go to next sequence segment
- ALOGV("End Segment, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("End Segment, time: %d", (unsigned int)(systemTime()/1000000));
lGenSmp = lReqSmp;
@@ -1156,13 +1156,13 @@
lpWaveGen->getSamples(lpOut, lGenSmp, lWaveCmd);
lFrequency = lpToneDesc->segments[lpToneGen->mCurSegment].waveFreq[++lFreqIdx];
}
- ALOGV("ON->OFF, lGenSmp: %d, lReqSmp: %d\n", lGenSmp, lReqSmp);
+ ALOGV("ON->OFF, lGenSmp: %d, lReqSmp: %d", lGenSmp, lReqSmp);
}
// check if we need to loop and loop for the reqd times
if (lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt) {
if (lpToneGen->mLoopCounter < lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt) {
- ALOGV ("in if loop loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("in if loop loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
@@ -1172,14 +1172,14 @@
// completed loop. go to next segment
lpToneGen->mLoopCounter = 0;
lpToneGen->mCurSegment++;
- ALOGV ("in else loop loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("in else loop loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
}
} else {
lpToneGen->mCurSegment++;
- ALOGV ("Goto next seg loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("Goto next seg loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
@@ -1188,32 +1188,32 @@
// Handle loop if last segment reached
if (lpToneDesc->segments[lpToneGen->mCurSegment].duration == 0) {
- ALOGV("Last Seg: %d\n", lpToneGen->mCurSegment);
+ ALOGV("Last Seg: %d", lpToneGen->mCurSegment);
// Pre increment loop count and restart if total count not reached. Stop sequence otherwise
if (++lpToneGen->mCurCount <= lpToneDesc->repeatCnt) {
- ALOGV("Repeating Count: %d\n", lpToneGen->mCurCount);
+ ALOGV("Repeating Count: %d", lpToneGen->mCurCount);
lpToneGen->mCurSegment = lpToneDesc->repeatSegment;
if (lpToneDesc->segments[lpToneDesc->repeatSegment].waveFreq[0] != 0) {
lWaveCmd = WaveGenerator::WAVEGEN_START;
}
- ALOGV("New segment %d, Next Time: %d\n", lpToneGen->mCurSegment,
+ ALOGV("New segment %d, Next Time: %d", lpToneGen->mCurSegment,
(lpToneGen->mNextSegSmp*1000)/lpToneGen->mSamplingRate);
} else {
lGenSmp = 0;
- ALOGV("End repeat, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("End repeat, time: %d", (unsigned int)(systemTime()/1000000));
}
} else {
- ALOGV("New segment %d, Next Time: %d\n", lpToneGen->mCurSegment,
+ ALOGV("New segment %d, Next Time: %d", lpToneGen->mCurSegment,
(lpToneGen->mNextSegSmp*1000)/lpToneGen->mSamplingRate);
if (lpToneDesc->segments[lpToneGen->mCurSegment].waveFreq[0] != 0) {
// If next segment is not silent, OFF -> ON transition : reset wave generator
lWaveCmd = WaveGenerator::WAVEGEN_START;
- ALOGV("OFF->ON, lGenSmp: %d, lReqSmp: %d\n", lGenSmp, lReqSmp);
+ ALOGV("OFF->ON, lGenSmp: %d, lReqSmp: %d", lGenSmp, lReqSmp);
} else {
lGenSmp = 0;
}
@@ -1251,13 +1251,13 @@
switch (lpToneGen->mState) {
case TONE_RESTARTING:
- ALOGV("Cbk restarting track\n");
+ ALOGV("Cbk restarting track");
if (lpToneGen->prepareWave()) {
lpToneGen->mState = TONE_STARTING;
// must reload lpToneDesc as prepareWave() may change mpToneDesc
lpToneDesc = lpToneGen->mpToneDesc;
} else {
- ALOGW("Cbk restarting prepareWave() failed\n");
+ ALOGW("Cbk restarting prepareWave() failed");
lpToneGen->mState = TONE_IDLE;
lpToneGen->mpAudioTrack->stop();
// Force loop exit
@@ -1266,14 +1266,14 @@
lSignal = true;
break;
case TONE_STOPPING:
- ALOGV("Cbk Stopping\n");
+ ALOGV("Cbk Stopping");
lpToneGen->mState = TONE_STOPPED;
// Force loop exit
lNumSmp = 0;
break;
case TONE_STOPPED:
lpToneGen->mState = TONE_INIT;
- ALOGV("Cbk Stopped track\n");
+ ALOGV("Cbk Stopped track");
lpToneGen->mpAudioTrack->stop();
// Force loop exit
lNumSmp = 0;
@@ -1281,7 +1281,7 @@
lSignal = true;
break;
case TONE_STARTING:
- ALOGV("Cbk starting track\n");
+ ALOGV("Cbk starting track");
lpToneGen->mState = TONE_PLAYING;
lSignal = true;
break;
@@ -1491,7 +1491,7 @@
d0 = 32767;
mA1_Q14 = (short) d0;
- ALOGV("WaveGenerator init, mA1_Q14: %d, mS2_0: %d, mAmplitude_Q15: %d\n",
+ ALOGV("WaveGenerator init, mA1_Q14: %d, mS2_0: %d, mAmplitude_Q15: %d",
mA1_Q14, mS2_0, mAmplitude_Q15);
}
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 4632016..fe519b0 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -24,6 +24,7 @@
#include <binder/IServiceManager.h>
#include <media/IMediaPlayerService.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioSource.h>
#include <media/stagefright/AMRWriter.h>
#include <media/stagefright/AACWriter.h>
@@ -31,7 +32,6 @@
#include <media/stagefright/CameraSourceTimeLapse.h>
#include <media/stagefright/MPEG2TSWriter.h>
#include <media/stagefright/MPEG4Writer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXClient.h>
@@ -241,8 +241,8 @@
status_t StagefrightRecorder::setOutputFile(int fd, int64_t offset, int64_t length) {
ALOGV("setOutputFile: %d, %lld, %lld", fd, offset, length);
// These don't make any sense, do they?
- CHECK_EQ(offset, 0);
- CHECK_EQ(length, 0);
+ CHECK_EQ(offset, 0ll);
+ CHECK_EQ(length, 0ll);
if (fd < 0) {
ALOGE("Invalid file descriptor: %d", fd);
@@ -734,7 +734,7 @@
}
status_t StagefrightRecorder::start() {
- CHECK(mOutputFd >= 0);
+ CHECK_GE(mOutputFd, 0);
if (mWriter != NULL) {
ALOGE("File writer is not avaialble");
@@ -837,7 +837,7 @@
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
sp<MediaSource> audioEncoder =
OMXCodec::Create(client.interface(), encMeta,
@@ -850,9 +850,9 @@
status_t StagefrightRecorder::startAACRecording() {
// FIXME:
// Add support for OUTPUT_FORMAT_AAC_ADIF
- CHECK(mOutputFormat == OUTPUT_FORMAT_AAC_ADTS);
+ CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_AAC_ADTS);
- CHECK(mAudioEncoder == AUDIO_ENCODER_AAC);
+ CHECK_EQ(mAudioEncoder, AUDIO_ENCODER_AAC);
CHECK(mAudioSource != AUDIO_SOURCE_CNT);
mWriter = new AACWriter(mOutputFd);
@@ -1386,7 +1386,7 @@
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
uint32_t encoder_flags = 0;
if (mIsMetaDataStoredInVideoBuffers) {
diff --git a/media/libstagefright/AACExtractor.cpp b/media/libstagefright/AACExtractor.cpp
index 33f22f2..4d1072f 100644
--- a/media/libstagefright/AACExtractor.cpp
+++ b/media/libstagefright/AACExtractor.cpp
@@ -23,9 +23,9 @@
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/AMRExtractor.cpp b/media/libstagefright/AMRExtractor.cpp
index 5a28347..03dcbf9 100644
--- a/media/libstagefright/AMRExtractor.cpp
+++ b/media/libstagefright/AMRExtractor.cpp
@@ -20,9 +20,9 @@
#include "include/AMRExtractor.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/AMRWriter.cpp b/media/libstagefright/AMRWriter.cpp
index 59b4ca7..ca85640 100644
--- a/media/libstagefright/AMRWriter.cpp
+++ b/media/libstagefright/AMRWriter.cpp
@@ -14,9 +14,9 @@
* limitations under the License.
*/
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AMRWriter.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index 8073af8..85b7979 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -1360,7 +1360,7 @@
mAudioTrack = source;
}
-void AwesomePlayer::addTextSource(sp<MediaSource> source) {
+void AwesomePlayer::addTextSource(const sp<MediaSource>& source) {
Mutex::Autolock autoLock(mTimedTextLock);
CHECK(source != NULL);
@@ -1609,7 +1609,7 @@
mSeekTimeUs,
mSeeking == SEEK_VIDEO_ONLY
? MediaSource::ReadOptions::SEEK_NEXT_SYNC
- : MediaSource::ReadOptions::SEEK_CLOSEST_SYNC);
+ : MediaSource::ReadOptions::SEEK_CLOSEST);
}
for (;;) {
status_t err = mVideoSource->read(&mVideoBuffer, &options);
diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp
index 228659c..ed1d5f4 100755
--- a/media/libstagefright/CameraSource.cpp
+++ b/media/libstagefright/CameraSource.cpp
@@ -20,8 +20,8 @@
#include <OMX_Component.h>
#include <binder/IPCThreadState.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/CameraSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -114,7 +114,7 @@
ALOGE("Uknown color format (%s), please add it to "
"CameraSource::getColorFormat", colorFormat);
- CHECK_EQ(0, "Unknown color format");
+ CHECK(!"Unknown color format");
}
CameraSource *CameraSource::Create() {
@@ -517,7 +517,7 @@
// This CHECK is good, since we just passed the lock/unlock
// check earlier by calling mCamera->setParameters().
- CHECK_EQ(OK, mCamera->setPreviewDisplay(mSurface));
+ CHECK_EQ((status_t)OK, mCamera->setPreviewDisplay(mSurface));
// By default, do not store metadata in video buffers
mIsMetaDataStoredInVideoBuffers = false;
@@ -566,7 +566,8 @@
if (mCameraFlags & FLAGS_HOT_CAMERA) {
mCamera->unlock();
mCamera.clear();
- CHECK_EQ(OK, mCameraRecordingProxy->startRecording(new ProxyListener(this)));
+ CHECK_EQ((status_t)OK,
+ mCameraRecordingProxy->startRecording(new ProxyListener(this)));
} else {
mCamera->setListener(new CameraSourceListener(this));
mCamera->startRecording();
@@ -718,7 +719,7 @@
return;
}
}
- CHECK_EQ(0, "signalBufferReturned: bogus buffer");
+ CHECK(!"signalBufferReturned: bogus buffer");
}
status_t CameraSource::read(
diff --git a/media/libstagefright/CameraSourceTimeLapse.cpp b/media/libstagefright/CameraSourceTimeLapse.cpp
index 83d67b9..26ce7ae 100644
--- a/media/libstagefright/CameraSourceTimeLapse.cpp
+++ b/media/libstagefright/CameraSourceTimeLapse.cpp
@@ -20,9 +20,9 @@
#include <binder/IPCThreadState.h>
#include <binder/MemoryBase.h>
#include <binder/MemoryHeapBase.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/CameraSource.h>
#include <media/stagefright/CameraSourceTimeLapse.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MetaData.h>
#include <camera/Camera.h>
#include <camera/CameraParameters.h>
diff --git a/media/libstagefright/DRMExtractor.cpp b/media/libstagefright/DRMExtractor.cpp
index afc4a80..524c3aa 100644
--- a/media/libstagefright/DRMExtractor.cpp
+++ b/media/libstagefright/DRMExtractor.cpp
@@ -23,6 +23,7 @@
#include <arpa/inet.h>
#include <utils/String8.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/Utils.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaSource.h>
@@ -30,7 +31,6 @@
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <drm/drm_framework_common.h>
#include <utils/Errors.h>
diff --git a/media/libstagefright/FileSource.cpp b/media/libstagefright/FileSource.cpp
index 01f53e4..73c8d03 100644
--- a/media/libstagefright/FileSource.cpp
+++ b/media/libstagefright/FileSource.cpp
@@ -14,8 +14,8 @@
* limitations under the License.
*/
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/FileSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <sys/types.h>
#include <unistd.h>
#include <sys/types.h>
diff --git a/media/libstagefright/JPEGSource.cpp b/media/libstagefright/JPEGSource.cpp
index e818115..bafa4b2 100644
--- a/media/libstagefright/JPEGSource.cpp
+++ b/media/libstagefright/JPEGSource.cpp
@@ -18,10 +18,10 @@
#define LOG_TAG "JPEGSource"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/JPEGSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -59,7 +59,7 @@
mWidth(0),
mHeight(0),
mOffset(0) {
- CHECK_EQ(parseJPEG(), OK);
+ CHECK_EQ(parseJPEG(), (status_t)OK);
CHECK(mSource->getSize(&mSize) == OK);
}
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 068660b..7ebbe1d 100755
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -23,10 +23,10 @@
#include <pthread.h>
#include <sys/prctl.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MPEG4Writer.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MetaData.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
@@ -70,6 +70,10 @@
status_t dump(int fd, const Vector<String16>& args) const;
private:
+ enum {
+ kMaxCttsOffsetTimeUs = 1000000LL, // 1 second
+ };
+
MPEG4Writer *mOwner;
sp<MetaData> mMeta;
sp<MediaSource> mSource;
@@ -137,11 +141,12 @@
: sampleCount(count), sampleDuration(timescaledDur) {}
uint32_t sampleCount;
- int32_t sampleDuration; // time scale based
+ uint32_t sampleDuration; // time scale based
};
- bool mHasNegativeCttsDeltaDuration;
size_t mNumCttsTableEntries;
List<CttsTableEntry> mCttsTableEntries;
+ int64_t mMinCttsOffsetTimeUs;
+ int64_t mMaxCttsOffsetTimeUs;
// Sequence parameter set or picture parameter set
struct AVCParamSet {
@@ -172,6 +177,8 @@
// Update the audio track's drift information.
void updateDriftTime(const sp<MetaData>& meta);
+ int32_t getStartTimeOffsetScaledTime() const;
+
static void *ThreadWrapper(void *me);
status_t threadEntry();
@@ -471,7 +478,7 @@
!param->findInt32(kKeyTimeScale, &mTimeScale)) {
mTimeScale = 1000;
}
- CHECK(mTimeScale > 0);
+ CHECK_GT(mTimeScale, 0);
ALOGV("movie time scale: %d", mTimeScale);
mStreamableFile = true;
@@ -490,7 +497,7 @@
}
mEstimatedMoovBoxSize = estimateMoovBoxSize(bitRate);
}
- CHECK(mEstimatedMoovBoxSize >= 8);
+ CHECK_GE(mEstimatedMoovBoxSize, 8);
lseek64(mFd, mFreeBoxOffset, SEEK_SET);
writeInt32(mEstimatedMoovBoxSize);
write("free", 4);
@@ -684,7 +691,7 @@
mWriteMoovBoxToMemory = false;
if (mStreamableFile) {
- CHECK(mMoovBoxBufferOffset + 8 <= mEstimatedMoovBoxSize);
+ CHECK_LE(mMoovBoxBufferOffset + 8, mEstimatedMoovBoxSize);
// Moov box
lseek64(mFd, mFreeBoxOffset, SEEK_SET);
@@ -856,7 +863,7 @@
mOffset += length + 4;
} else {
- CHECK(length < 65536);
+ CHECK_LT(length, 65536);
uint8_t x = length >> 8;
::write(mFd, &x, 1);
@@ -1085,7 +1092,7 @@
void MPEG4Writer::setStartTimestampUs(int64_t timeUs) {
ALOGI("setStartTimestampUs: %lld", timeUs);
- CHECK(timeUs >= 0);
+ CHECK_GE(timeUs, 0ll);
Mutex::Autolock autoLock(mLock);
if (mStartTimestampUs < 0 || mStartTimestampUs > timeUs) {
mStartTimestampUs = timeUs;
@@ -1186,9 +1193,6 @@
if (mIsAudio) {
return;
}
- if (duration < 0 && !mHasNegativeCttsDeltaDuration) {
- mHasNegativeCttsDeltaDuration = true;
- }
CttsTableEntry cttsEntry(sampleCount, duration);
mCttsTableEntries.push_back(cttsEntry);
++mNumCttsTableEntries;
@@ -1218,7 +1222,7 @@
mTimeScale = timeScale;
}
- CHECK(mTimeScale > 0);
+ CHECK_GT(mTimeScale, 0);
}
void MPEG4Writer::Track::getCodecSpecificDataFromInputFormatIfPossible() {
@@ -1299,7 +1303,7 @@
}
}
- CHECK("Received a chunk for a unknown track" == 0);
+ CHECK(!"Received a chunk for a unknown track");
}
void MPEG4Writer::writeChunkToFile(Chunk* chunk) {
@@ -1509,7 +1513,6 @@
mMdatSizeBytes = 0;
mMaxChunkDurationUs = 0;
- mHasNegativeCttsDeltaDuration = false;
pthread_create(&mThread, &attr, ThreadWrapper, this);
pthread_attr_destroy(&attr);
@@ -1833,29 +1836,18 @@
int32_t nChunks = 0;
int32_t nZeroLengthFrames = 0;
int64_t lastTimestampUs = 0; // Previous sample time stamp
- int64_t lastCttsTimeUs = 0; // Previous sample time stamp
int64_t lastDurationUs = 0; // Between the previous two samples
int64_t currDurationTicks = 0; // Timescale based ticks
int64_t lastDurationTicks = 0; // Timescale based ticks
int32_t sampleCount = 1; // Sample count in the current stts table entry
- int64_t currCttsDurTicks = 0; // Timescale based ticks
- int64_t lastCttsDurTicks = 0; // Timescale based ticks
- int32_t cttsSampleCount = 1; // Sample count in the current ctts table entry
- uint32_t previousSampleSize = 0; // Size of the previous sample
+ uint32_t previousSampleSize = 0; // Size of the previous sample
int64_t previousPausedDurationUs = 0;
int64_t timestampUs = 0;
- int64_t cttsDeltaTimeUs = 0;
- bool hasBFrames = false;
+ int64_t cttsOffsetTimeUs = 0;
+ int64_t currCttsOffsetTimeTicks = 0; // Timescale based ticks
+ int64_t lastCttsOffsetTimeTicks = -1; // Timescale based ticks
+ int32_t cttsSampleCount = 0; // Sample count in the current ctts table entry
-#if 1
- // XXX: Samsung's video encoder's output buffer timestamp
- // is not correct. see bug 4724339
- char value[PROPERTY_VALUE_MAX];
- if (property_get("rw.media.record.hasb", value, NULL) &&
- (!strcasecmp(value, "true") || !strcasecmp(value, "1"))) {
- hasBFrames = true;
- }
-#endif
if (mIsAudio) {
prctl(PR_SET_NAME, (unsigned long)"AudioTrackEncoding", 0, 0, 0);
} else {
@@ -1897,7 +1889,7 @@
(const uint8_t *)buffer->data()
+ buffer->range_offset(),
buffer->range_length());
- CHECK_EQ(OK, err);
+ CHECK_EQ((status_t)OK, err);
} else if (mIsMPEG4) {
mCodecSpecificDataSize = buffer->range_length();
mCodecSpecificData = malloc(mCodecSpecificDataSize);
@@ -1963,32 +1955,64 @@
if (mResumed) {
int64_t durExcludingEarlierPausesUs = timestampUs - previousPausedDurationUs;
- CHECK(durExcludingEarlierPausesUs >= 0);
+ CHECK_GE(durExcludingEarlierPausesUs, 0ll);
int64_t pausedDurationUs = durExcludingEarlierPausesUs - mTrackDurationUs;
- CHECK(pausedDurationUs >= lastDurationUs);
+ CHECK_GE(pausedDurationUs, lastDurationUs);
previousPausedDurationUs += pausedDurationUs - lastDurationUs;
mResumed = false;
}
timestampUs -= previousPausedDurationUs;
- CHECK(timestampUs >= 0);
- if (!mIsAudio && hasBFrames) {
+ CHECK_GE(timestampUs, 0ll);
+ if (!mIsAudio) {
/*
* Composition time: timestampUs
* Decoding time: decodingTimeUs
- * Composition time delta = composition time - decoding time
- *
- * We save picture decoding time stamp delta in stts table entries,
- * and composition time delta duration in ctts table entries.
+ * Composition time offset = composition time - decoding time
*/
int64_t decodingTimeUs;
CHECK(meta_data->findInt64(kKeyDecodingTime, &decodingTimeUs));
decodingTimeUs -= previousPausedDurationUs;
- int64_t timeUs = decodingTimeUs;
- cttsDeltaTimeUs = timestampUs - decodingTimeUs;
+ cttsOffsetTimeUs =
+ timestampUs + kMaxCttsOffsetTimeUs - decodingTimeUs;
+ CHECK_GE(cttsOffsetTimeUs, 0ll);
timestampUs = decodingTimeUs;
- ALOGV("decoding time: %lld and ctts delta time: %lld",
- timestampUs, cttsDeltaTimeUs);
+ ALOGV("decoding time: %lld and ctts offset time: %lld",
+ timestampUs, cttsOffsetTimeUs);
+
+ // Update ctts box table if necessary
+ currCttsOffsetTimeTicks =
+ (cttsOffsetTimeUs * mTimeScale + 500000LL) / 1000000LL;
+ CHECK_LE(currCttsOffsetTimeTicks, 0x0FFFFFFFFLL);
+ if (mNumSamples == 0) {
+ // Force the first ctts table entry to have one single entry
+ // so that we can do adjustment for the initial track start
+ // time offset easily in writeCttsBox().
+ lastCttsOffsetTimeTicks = currCttsOffsetTimeTicks;
+ addOneCttsTableEntry(1, currCttsOffsetTimeTicks);
+ cttsSampleCount = 0; // No sample in ctts box is pending
+ } else {
+ if (currCttsOffsetTimeTicks != lastCttsOffsetTimeTicks) {
+ addOneCttsTableEntry(cttsSampleCount, lastCttsOffsetTimeTicks);
+ lastCttsOffsetTimeTicks = currCttsOffsetTimeTicks;
+ cttsSampleCount = 1; // One sample in ctts box is pending
+ } else {
+ ++cttsSampleCount;
+ }
+ }
+
+ // Update ctts time offset range
+ if (mNumSamples == 0) {
+ mMinCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ mMaxCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ } else {
+ if (currCttsOffsetTimeTicks > mMaxCttsOffsetTimeUs) {
+ mMaxCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ } else if (currCttsOffsetTimeTicks < mMinCttsOffsetTimeUs) {
+ mMinCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ }
+ }
+
}
if (mIsRealTimeRecording) {
@@ -1997,7 +2021,7 @@
}
}
- CHECK(timestampUs >= 0);
+ CHECK_GE(timestampUs, 0ll);
ALOGV("%s media time stamp: %lld and previous paused duration %lld",
mIsAudio? "Audio": "Video", timestampUs, previousPausedDurationUs);
if (timestampUs > mTrackDurationUs) {
@@ -2012,6 +2036,7 @@
currDurationTicks =
((timestampUs * mTimeScale + 500000LL) / 1000000LL -
(lastTimestampUs * mTimeScale + 500000LL) / 1000000LL);
+ CHECK_GE(currDurationTicks, 0ll);
mSampleSizes.push_back(sampleSize);
++mNumSamples;
@@ -2020,25 +2045,12 @@
// Force the first sample to have its own stts entry so that
// we can adjust its value later to maintain the A/V sync.
if (mNumSamples == 3 || currDurationTicks != lastDurationTicks) {
- ALOGV("%s lastDurationUs: %lld us, currDurationTicks: %lld us",
- mIsAudio? "Audio": "Video", lastDurationUs, currDurationTicks);
addOneSttsTableEntry(sampleCount, lastDurationTicks);
sampleCount = 1;
} else {
++sampleCount;
}
- if (!mIsAudio) {
- currCttsDurTicks =
- ((cttsDeltaTimeUs * mTimeScale + 500000LL) / 1000000LL -
- (lastCttsTimeUs * mTimeScale + 500000LL) / 1000000LL);
- if (currCttsDurTicks != lastCttsDurTicks) {
- addOneCttsTableEntry(cttsSampleCount, lastCttsDurTicks);
- cttsSampleCount = 1;
- } else {
- ++cttsSampleCount;
- }
- }
}
if (mSamplesHaveSameSize) {
if (mNumSamples >= 2 && previousSampleSize != sampleSize) {
@@ -2052,11 +2064,6 @@
lastDurationTicks = currDurationTicks;
lastTimestampUs = timestampUs;
- if (!mIsAudio) {
- lastCttsDurTicks = currCttsDurTicks;
- lastCttsTimeUs = cttsDeltaTimeUs;
- }
-
if (isSync != 0) {
addOneStssTableEntry(mNumSamples);
}
@@ -2125,10 +2132,8 @@
if (mNumSamples == 1) {
lastDurationUs = 0; // A single sample's duration
lastDurationTicks = 0;
- lastCttsDurTicks = 0;
} else {
++sampleCount; // Count for the last sample
- ++cttsSampleCount;
}
if (mNumSamples <= 2) {
@@ -2140,7 +2145,14 @@
addOneSttsTableEntry(sampleCount, lastDurationTicks);
}
- addOneCttsTableEntry(cttsSampleCount, lastCttsDurTicks);
+ // The last ctts box may not have been written yet, and this
+ // is to make sure that we write out the last ctts box.
+ if (currCttsOffsetTimeTicks == lastCttsOffsetTimeTicks) {
+ if (cttsSampleCount > 0) {
+ addOneCttsTableEntry(cttsSampleCount, lastCttsOffsetTimeTicks);
+ }
+ }
+
mTrackDurationUs += lastDurationUs;
mReachedEOS = true;
@@ -2406,7 +2418,7 @@
mOwner->writeInt16(0x18); // depth
mOwner->writeInt16(-1); // predefined
- CHECK(23 + mCodecSpecificDataSize < 128);
+ CHECK_LT(23 + mCodecSpecificDataSize, 128);
if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_MPEG4, mime)) {
writeMp4vEsdsBox();
@@ -2465,10 +2477,10 @@
void MPEG4Writer::Track::writeMp4aEsdsBox() {
mOwner->beginBox("esds");
CHECK(mCodecSpecificData);
- CHECK(mCodecSpecificDataSize > 0);
+ CHECK_GT(mCodecSpecificDataSize, 0);
// Make sure all sizes encode to a single byte.
- CHECK(mCodecSpecificDataSize + 23 < 128);
+ CHECK_LT(mCodecSpecificDataSize + 23, 128);
mOwner->writeInt32(0); // version=0, flags=0
mOwner->writeInt8(0x03); // ES_DescrTag
@@ -2502,7 +2514,7 @@
void MPEG4Writer::Track::writeMp4vEsdsBox() {
CHECK(mCodecSpecificData);
- CHECK(mCodecSpecificDataSize > 0);
+ CHECK_GT(mCodecSpecificDataSize, 0);
mOwner->beginBox("esds");
mOwner->writeInt32(0); // version=0, flags=0
@@ -2662,7 +2674,7 @@
void MPEG4Writer::Track::writeAvccBox() {
CHECK(mCodecSpecificData);
- CHECK(mCodecSpecificDataSize >= 5);
+ CHECK_GE(mCodecSpecificDataSize, 5);
// Patch avcc's lengthSize field to match the number
// of bytes we use to indicate the size of a nal unit.
@@ -2690,23 +2702,26 @@
mOwner->endBox(); // pasp
}
+int32_t MPEG4Writer::Track::getStartTimeOffsetScaledTime() const {
+ int64_t trackStartTimeOffsetUs = 0;
+ int64_t moovStartTimeUs = mOwner->getStartTimestampUs();
+ if (mStartTimestampUs != moovStartTimeUs) {
+ CHECK_GT(mStartTimestampUs, moovStartTimeUs);
+ trackStartTimeOffsetUs = mStartTimestampUs - moovStartTimeUs;
+ }
+ return (trackStartTimeOffsetUs * mTimeScale + 500000LL) / 1000000LL;
+}
+
void MPEG4Writer::Track::writeSttsBox() {
mOwner->beginBox("stts");
mOwner->writeInt32(0); // version=0, flags=0
mOwner->writeInt32(mNumSttsTableEntries);
// Compensate for small start time difference from different media tracks
- int64_t trackStartTimeOffsetUs = 0;
- int64_t moovStartTimeUs = mOwner->getStartTimestampUs();
- if (mStartTimestampUs != moovStartTimeUs) {
- CHECK(mStartTimestampUs > moovStartTimeUs);
- trackStartTimeOffsetUs = mStartTimestampUs - moovStartTimeUs;
- }
List<SttsTableEntry>::iterator it = mSttsTableEntries.begin();
CHECK(it != mSttsTableEntries.end() && it->sampleCount == 1);
mOwner->writeInt32(it->sampleCount);
- int32_t dur = (trackStartTimeOffsetUs * mTimeScale + 500000LL) / 1000000LL;
- mOwner->writeInt32(dur + it->sampleDuration);
+ mOwner->writeInt32(getStartTimeOffsetScaledTime() + it->sampleDuration);
int64_t totalCount = 1;
while (++it != mSttsTableEntries.end()) {
@@ -2714,7 +2729,7 @@
mOwner->writeInt32(it->sampleDuration);
totalCount += it->sampleCount;
}
- CHECK(totalCount == mNumSamples);
+ CHECK_EQ(totalCount, mNumSamples);
mOwner->endBox(); // stts
}
@@ -2723,6 +2738,11 @@
return;
}
+ // There is no B frame at all
+ if (mMinCttsOffsetTimeUs == mMaxCttsOffsetTimeUs) {
+ return;
+ }
+
// Do not write ctts box when there is no need to have it.
if ((mNumCttsTableEntries == 1 &&
mCttsTableEntries.begin()->sampleDuration == 0) ||
@@ -2730,24 +2750,29 @@
return;
}
- ALOGV("ctts box has %d entries", mNumCttsTableEntries);
+ ALOGD("ctts box has %d entries with range [%lld, %lld]",
+ mNumCttsTableEntries, mMinCttsOffsetTimeUs, mMaxCttsOffsetTimeUs);
mOwner->beginBox("ctts");
- if (mHasNegativeCttsDeltaDuration) {
- mOwner->writeInt32(0x00010000); // version=1, flags=0
- } else {
- mOwner->writeInt32(0); // version=0, flags=0
- }
+ // Version 1 allows to use negative offset time value, but
+ // we are sticking to version 0 for now.
+ mOwner->writeInt32(0); // version=0, flags=0
mOwner->writeInt32(mNumCttsTableEntries);
- int64_t totalCount = 0;
- for (List<CttsTableEntry>::iterator it = mCttsTableEntries.begin();
- it != mCttsTableEntries.end(); ++it) {
+ // Compensate for small start time difference from different media tracks
+ List<CttsTableEntry>::iterator it = mCttsTableEntries.begin();
+ CHECK(it != mCttsTableEntries.end() && it->sampleCount == 1);
+ mOwner->writeInt32(it->sampleCount);
+ mOwner->writeInt32(getStartTimeOffsetScaledTime() +
+ it->sampleDuration - mMinCttsOffsetTimeUs);
+
+ int64_t totalCount = 1;
+ while (++it != mCttsTableEntries.end()) {
mOwner->writeInt32(it->sampleCount);
- mOwner->writeInt32(it->sampleDuration);
+ mOwner->writeInt32(it->sampleDuration - mMinCttsOffsetTimeUs);
totalCount += it->sampleCount;
}
- CHECK(totalCount == mNumSamples);
+ CHECK_EQ(totalCount, mNumSamples);
mOwner->endBox(); // ctts
}
diff --git a/media/libstagefright/MediaBuffer.cpp b/media/libstagefright/MediaBuffer.cpp
index 96271e4..11b80bf 100644
--- a/media/libstagefright/MediaBuffer.cpp
+++ b/media/libstagefright/MediaBuffer.cpp
@@ -22,8 +22,8 @@
#include <stdlib.h>
#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MetaData.h>
#include <ui/GraphicBuffer.h>
@@ -157,7 +157,7 @@
}
MediaBuffer::~MediaBuffer() {
- CHECK_EQ(mObserver, NULL);
+ CHECK(mObserver == NULL);
if (mOwnsData && mData != NULL) {
free(mData);
@@ -188,7 +188,7 @@
}
MediaBuffer *MediaBuffer::clone() {
- CHECK_EQ(mGraphicBuffer, NULL);
+ CHECK(mGraphicBuffer == NULL);
MediaBuffer *buffer = new MediaBuffer(mData, mSize);
buffer->set_range(mRangeOffset, mRangeLength);
diff --git a/media/libstagefright/MediaBufferGroup.cpp b/media/libstagefright/MediaBufferGroup.cpp
index c8d05f4..80aae51 100644
--- a/media/libstagefright/MediaBufferGroup.cpp
+++ b/media/libstagefright/MediaBufferGroup.cpp
@@ -17,9 +17,9 @@
#define LOG_TAG "MediaBufferGroup"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
namespace android {
diff --git a/media/libstagefright/MediaSourceSplitter.cpp b/media/libstagefright/MediaSourceSplitter.cpp
index 8af0694..3b64ded 100644
--- a/media/libstagefright/MediaSourceSplitter.cpp
+++ b/media/libstagefright/MediaSourceSplitter.cpp
@@ -18,8 +18,8 @@
#define LOG_TAG "MediaSourceSplitter"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaSourceSplitter.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MetaData.h>
diff --git a/media/libstagefright/MetaData.cpp b/media/libstagefright/MetaData.cpp
index 884f3b4..66dec90 100644
--- a/media/libstagefright/MetaData.cpp
+++ b/media/libstagefright/MetaData.cpp
@@ -17,7 +17,7 @@
#include <stdlib.h>
#include <string.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MetaData.h>
namespace android {
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 381320b..470f750 100755
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -2187,7 +2187,7 @@
}
}
-int64_t OMXCodec::retrieveDecodingTimeUs(bool isCodecSpecific) {
+int64_t OMXCodec::getDecodingTimeUs() {
CHECK(mIsEncoder && mIsVideo);
if (mDecodingTimeList.empty()) {
@@ -2199,12 +2199,7 @@
List<int64_t>::iterator it = mDecodingTimeList.begin();
int64_t timeUs = *it;
-
- // If the output buffer is codec specific configuration,
- // do not remove the decoding time from the list.
- if (!isCodecSpecific) {
- mDecodingTimeList.erase(it);
- }
+ mDecodingTimeList.erase(it);
return timeUs;
}
@@ -2384,7 +2379,7 @@
}
if (mIsEncoder && mIsVideo) {
- int64_t decodingTimeUs = retrieveDecodingTimeUs(isCodecSpecific);
+ int64_t decodingTimeUs = isCodecSpecific? 0: getDecodingTimeUs();
buffer->meta_data()->setInt64(kKeyDecodingTime, decodingTimeUs);
}
diff --git a/media/libstagefright/OggExtractor.cpp b/media/libstagefright/OggExtractor.cpp
index 73efc27..5e79e78 100644
--- a/media/libstagefright/OggExtractor.cpp
+++ b/media/libstagefright/OggExtractor.cpp
@@ -21,10 +21,10 @@
#include "include/OggExtractor.h"
#include <cutils/properties.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/SampleIterator.cpp b/media/libstagefright/SampleIterator.cpp
index 81ec5c1..eae721b 100644
--- a/media/libstagefright/SampleIterator.cpp
+++ b/media/libstagefright/SampleIterator.cpp
@@ -22,8 +22,8 @@
#include <arpa/inet.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/Utils.h>
#include "include/SampleTable.h"
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index 43bfd9e..35f9c1f 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -20,10 +20,10 @@
#include "include/StagefrightMetadataRetriever.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/ColorConverter.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/FileSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaExtractor.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXCodec.h>
@@ -37,7 +37,7 @@
ALOGV("StagefrightMetadataRetriever()");
DataSource::RegisterDefaultSniffers();
- CHECK_EQ(mClient.connect(), OK);
+ CHECK_EQ(mClient.connect(), (status_t)OK);
}
StagefrightMetadataRetriever::~StagefrightMetadataRetriever() {
@@ -169,7 +169,7 @@
|| (buffer != NULL && buffer->range_length() == 0));
if (err != OK) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
ALOGV("decoding frame failed.");
decoder->stop();
diff --git a/media/libstagefright/SurfaceMediaSource.cpp b/media/libstagefright/SurfaceMediaSource.cpp
index 2233d1b..aa047d6 100644
--- a/media/libstagefright/SurfaceMediaSource.cpp
+++ b/media/libstagefright/SurfaceMediaSource.cpp
@@ -16,14 +16,14 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "SurfaceMediaSource"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/SurfaceMediaSource.h>
-#include <ui/GraphicBuffer.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/openmax/OMX_IVCommon.h>
#include <media/stagefright/MetadataBufferType.h>
+#include <ui/GraphicBuffer.h>
#include <surfaceflinger/ISurfaceComposer.h>
#include <surfaceflinger/SurfaceComposerClient.h>
#include <surfaceflinger/IGraphicBufferAlloc.h>
@@ -60,7 +60,7 @@
SurfaceMediaSource::~SurfaceMediaSource() {
ALOGV("SurfaceMediaSource::~SurfaceMediaSource");
if (!mStopped) {
- stop();
+ reset();
}
}
@@ -716,9 +716,9 @@
}
-status_t SurfaceMediaSource::stop()
+status_t SurfaceMediaSource::reset()
{
- ALOGV("Stop");
+ ALOGV("Reset");
Mutex::Autolock lock(mMutex);
// TODO: Add waiting on mFrameCompletedCondition here?
@@ -855,7 +855,7 @@
}
if (!foundBuffer) {
- CHECK_EQ(0, "signalBufferReturned: bogus buffer");
+ CHECK(!"signalBufferReturned: bogus buffer");
}
}
diff --git a/media/libstagefright/ThrottledSource.cpp b/media/libstagefright/ThrottledSource.cpp
index 88e07b0..b1fcafd 100644
--- a/media/libstagefright/ThrottledSource.cpp
+++ b/media/libstagefright/ThrottledSource.cpp
@@ -16,7 +16,7 @@
#include "include/ThrottledSource.h"
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
namespace android {
diff --git a/media/libstagefright/TimedEventQueue.cpp b/media/libstagefright/TimedEventQueue.cpp
index 12c9c36..f4b5d4f 100644
--- a/media/libstagefright/TimedEventQueue.cpp
+++ b/media/libstagefright/TimedEventQueue.cpp
@@ -31,7 +31,7 @@
#include <sys/prctl.h>
#include <sys/time.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#ifdef ANDROID_SIMULATOR
#include <jni.h>
diff --git a/media/libstagefright/VideoSourceDownSampler.cpp b/media/libstagefright/VideoSourceDownSampler.cpp
index 1b66990..90a42c9 100644
--- a/media/libstagefright/VideoSourceDownSampler.cpp
+++ b/media/libstagefright/VideoSourceDownSampler.cpp
@@ -17,9 +17,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "VideoSourceDownSampler"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/VideoSourceDownSampler.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/YUVImage.h>
#include <media/stagefright/YUVCanvas.h>
diff --git a/media/libstagefright/WAVExtractor.cpp b/media/libstagefright/WAVExtractor.cpp
index 0bcaf08..501f480 100644
--- a/media/libstagefright/WAVExtractor.cpp
+++ b/media/libstagefright/WAVExtractor.cpp
@@ -20,9 +20,9 @@
#include "include/WAVExtractor.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
@@ -217,7 +217,7 @@
kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_G711_ALAW);
break;
default:
- CHECK_EQ(mWaveFormat, WAVE_FORMAT_MULAW);
+ CHECK_EQ(mWaveFormat, (uint16_t)WAVE_FORMAT_MULAW);
mTrackMeta->setCString(
kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_G711_MLAW);
break;
@@ -362,7 +362,7 @@
// Convert 8-bit unsigned samples to 16-bit signed.
MediaBuffer *tmp;
- CHECK_EQ(mGroup->acquire_buffer(&tmp), OK);
+ CHECK_EQ(mGroup->acquire_buffer(&tmp), (status_t)OK);
// The new buffer holds the sample number of samples, but each
// one is 2 bytes wide.
diff --git a/media/libstagefright/WVMExtractor.cpp b/media/libstagefright/WVMExtractor.cpp
index 1e4e049..c7ad513 100644
--- a/media/libstagefright/WVMExtractor.cpp
+++ b/media/libstagefright/WVMExtractor.cpp
@@ -21,6 +21,7 @@
#include <arpa/inet.h>
#include <utils/String8.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/Utils.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaSource.h>
@@ -28,7 +29,6 @@
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <dlfcn.h>
#include <utils/Errors.h>
diff --git a/media/libstagefright/codecs/aacenc/AACEncoder.cpp b/media/libstagefright/codecs/aacenc/AACEncoder.cpp
index 2b8633d..8b5007e 100644
--- a/media/libstagefright/codecs/aacenc/AACEncoder.cpp
+++ b/media/libstagefright/codecs/aacenc/AACEncoder.cpp
@@ -22,8 +22,8 @@
#include "voAAC.h"
#include "cmnMemory.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -114,8 +114,8 @@
ALOGV("setAudioSpecificConfigData: %d hz, %d bps, and %d channels",
mSampleRate, mBitRate, mChannels);
- int32_t index;
- CHECK_EQ(OK, getSampleRateTableIndex(mSampleRate, index));
+ int32_t index = 0;
+ CHECK_EQ((status_t)OK, getSampleRateTableIndex(mSampleRate, index));
if (mChannels > 2 || mChannels <= 0) {
ALOGE("Unsupported number of channels(%d)", mChannels);
return UNKNOWN_ERROR;
@@ -142,7 +142,7 @@
mBufferGroup = new MediaBufferGroup;
mBufferGroup->add_buffer(new MediaBuffer(2048));
- CHECK_EQ(OK, initCheck());
+ CHECK_EQ((status_t)OK, initCheck());
mNumInputSamples = 0;
mAnchorTimeUs = 0;
@@ -183,7 +183,7 @@
mSource->stop();
if (mEncoderHandle) {
- CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+ CHECK_EQ((VO_U32)VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
mEncoderHandle = NULL;
}
delete mApiHandle;
@@ -223,7 +223,7 @@
CHECK(options == NULL || !options->getSeekTo(&seekTimeUs, &mode));
MediaBuffer *buffer;
- CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), OK);
+ CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), (status_t)OK);
uint8_t *outPtr = (uint8_t *)buffer->data();
bool readFromSource = false;
int64_t wallClockTimeUs = -1;
@@ -255,7 +255,7 @@
}
size_t align = mInputBuffer->range_length() % sizeof(int16_t);
- CHECK_EQ(align, 0);
+ CHECK_EQ(align, (size_t)0);
int64_t timeUs;
if (mInputBuffer->meta_data()->findInt64(kKeyDriftTime, &timeUs)) {
diff --git a/media/libstagefright/codecs/amrnb/enc/AMRNBEncoder.cpp b/media/libstagefright/codecs/amrnb/enc/AMRNBEncoder.cpp
index 3afbc4f..27d7e4d 100644
--- a/media/libstagefright/codecs/amrnb/enc/AMRNBEncoder.cpp
+++ b/media/libstagefright/codecs/amrnb/enc/AMRNBEncoder.cpp
@@ -18,8 +18,8 @@
#include "gsmamr_enc.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -210,7 +210,7 @@
}
MediaBuffer *buffer;
- CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), OK);
+ CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), (status_t)OK);
uint8_t *outPtr = (uint8_t *)buffer->data();
diff --git a/media/libstagefright/codecs/amrwbenc/AMRWBEncoder.cpp b/media/libstagefright/codecs/amrwbenc/AMRWBEncoder.cpp
index 60b1163..7fd3a95 100644
--- a/media/libstagefright/codecs/amrwbenc/AMRWBEncoder.cpp
+++ b/media/libstagefright/codecs/amrwbenc/AMRWBEncoder.cpp
@@ -22,8 +22,8 @@
#include "voAMRWB.h"
#include "cmnMemory.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -134,7 +134,7 @@
// The largest buffer size is header + 477 bits
mBufferGroup->add_buffer(new MediaBuffer(1024));
- CHECK_EQ(OK, initCheck());
+ CHECK_EQ((status_t)OK, initCheck());
mNumFramesOutput = 0;
@@ -163,7 +163,7 @@
mBufferGroup = NULL;
- CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+ CHECK_EQ((VO_U32)VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
mEncoderHandle = NULL;
delete mApiHandle;
@@ -222,7 +222,7 @@
}
size_t align = mInputBuffer->range_length() % sizeof(int16_t);
- CHECK_EQ(align, 0);
+ CHECK_EQ(align, (size_t)0);
int64_t timeUs;
if (mInputBuffer->meta_data()->findInt64(kKeyDriftTime, &timeUs)) {
@@ -271,7 +271,7 @@
CHECK(VO_ERR_NONE == mApiHandle->SetInputData(mEncoderHandle,&inputData));
MediaBuffer *buffer;
- CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), OK);
+ CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), (status_t)OK);
uint8_t *outPtr = (uint8_t *)buffer->data();
VO_CODECBUFFER outputData;
diff --git a/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp b/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp
index e202a2b..7533f07 100644
--- a/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp
+++ b/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp
@@ -24,8 +24,8 @@
#include "avcenc_int.h"
#include "OMX_Video.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -417,7 +417,7 @@
*out = NULL;
MediaBuffer *outputBuffer;
- CHECK_EQ(OK, mGroup->acquire_buffer(&outputBuffer));
+ CHECK_EQ((status_t)OK, mGroup->acquire_buffer(&outputBuffer));
uint8_t *outPtr = (uint8_t *) outputBuffer->data();
uint32_t dataLength = outputBuffer->size();
@@ -557,9 +557,9 @@
encoderStatus = PVAVCEncodeNAL(mHandle, outPtr, &dataLength, &type);
if (encoderStatus == AVCENC_SUCCESS) {
outputBuffer->meta_data()->setInt32(kKeyIsSyncFrame, mIsIDRFrame);
- CHECK_EQ(NULL, PVAVCEncGetOverrunBuffer(mHandle));
+ CHECK(NULL == PVAVCEncGetOverrunBuffer(mHandle));
} else if (encoderStatus == AVCENC_PICTURE_READY) {
- CHECK_EQ(NULL, PVAVCEncGetOverrunBuffer(mHandle));
+ CHECK(NULL == PVAVCEncGetOverrunBuffer(mHandle));
if (mIsIDRFrame) {
outputBuffer->meta_data()->setInt32(kKeyIsSyncFrame, mIsIDRFrame);
mIsIDRFrame = 0;
diff --git a/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp b/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp
index d538603..20b0f8d 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp
@@ -23,8 +23,8 @@
#include "mp4enc_api.h"
#include "OMX_Video.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -379,7 +379,7 @@
*out = NULL;
MediaBuffer *outputBuffer;
- CHECK_EQ(OK, mGroup->acquire_buffer(&outputBuffer));
+ CHECK_EQ((status_t)OK, mGroup->acquire_buffer(&outputBuffer));
uint8_t *outPtr = (uint8_t *) outputBuffer->data();
int32_t dataLength = outputBuffer->size();
@@ -467,7 +467,7 @@
mInputBuffer = NULL;
return UNKNOWN_ERROR;
}
- CHECK_EQ(NULL, PVGetOverrunBuffer(mHandle));
+ CHECK(NULL == PVGetOverrunBuffer(mHandle));
if (hintTrack.CodeType == 0) { // I-frame serves as sync frame
outputBuffer->meta_data()->setInt32(kKeyIsSyncFrame, 1);
}
diff --git a/media/libstagefright/colorconversion/ColorConverter.cpp b/media/libstagefright/colorconversion/ColorConverter.cpp
index f3ef3de..597167f 100644
--- a/media/libstagefright/colorconversion/ColorConverter.cpp
+++ b/media/libstagefright/colorconversion/ColorConverter.cpp
@@ -18,8 +18,8 @@
#define LOG_TAG "ColorConverter"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/ColorConverter.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaErrors.h>
namespace android {
diff --git a/media/libstagefright/id3/Android.mk b/media/libstagefright/id3/Android.mk
index 23c8e44..ff35d4a 100644
--- a/media/libstagefright/id3/Android.mk
+++ b/media/libstagefright/id3/Android.mk
@@ -16,7 +16,7 @@
testid3.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright libutils libbinder
+ libstagefright libutils libbinder libstagefright_foundation
LOCAL_STATIC_LIBRARIES := \
libstagefright_id3
diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp
index 6dde9d8..2e92926 100644
--- a/media/libstagefright/id3/ID3.cpp
+++ b/media/libstagefright/id3/ID3.cpp
@@ -20,8 +20,8 @@
#include "../include/ID3.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/Utils.h>
#include <utils/String8.h>
#include <byteswap.h>
diff --git a/media/libstagefright/id3/testid3.cpp b/media/libstagefright/id3/testid3.cpp
index 0741045..bc4572c 100644
--- a/media/libstagefright/id3/testid3.cpp
+++ b/media/libstagefright/id3/testid3.cpp
@@ -23,7 +23,7 @@
#include <binder/ProcessState.h>
#include <media/stagefright/FileSource.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#define MAXPATHLEN 256
@@ -70,7 +70,7 @@
void scanFile(const char *path) {
sp<FileSource> file = new FileSource(path);
- CHECK_EQ(file->initCheck(), OK);
+ CHECK_EQ(file->initCheck(), (status_t)OK);
ID3 tag(file);
if (!tag.isValid()) {
diff --git a/media/libstagefright/include/AwesomePlayer.h b/media/libstagefright/include/AwesomePlayer.h
index a7a3d47..4c7bfa6 100644
--- a/media/libstagefright/include/AwesomePlayer.h
+++ b/media/libstagefright/include/AwesomePlayer.h
@@ -258,7 +258,7 @@
void setVideoSource(sp<MediaSource> source);
status_t initVideoDecoder(uint32_t flags = 0);
- void addTextSource(sp<MediaSource> source);
+ void addTextSource(const sp<MediaSource>& source);
void onStreamDone();
diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index 4fbf47e..a0db719 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -93,7 +93,10 @@
void advance();
void reset();
- void seek(int64_t seekTimeUs, bool seekToKeyFrame);
+
+ void seek(
+ int64_t seekTimeUs, bool seekToKeyFrame,
+ int64_t *actualFrameTimeUs);
const mkvparser::Block *block() const;
int64_t blockTimeUs() const;
@@ -303,22 +306,52 @@
} while (!eos() && block()->GetTrackNumber() != mTrackNum);
}
-void BlockIterator::seek(int64_t seekTimeUs, bool seekToKeyFrame) {
+void BlockIterator::seek(
+ int64_t seekTimeUs, bool seekToKeyFrame,
+ int64_t *actualFrameTimeUs) {
Mutex::Autolock autoLock(mExtractor->mLock);
- mCluster = mExtractor->mSegment->FindCluster(seekTimeUs * 1000ll);
+ *actualFrameTimeUs = -1ll;
+
+ int64_t seekTimeNs = seekTimeUs * 1000ll;
+
+ mCluster = mExtractor->mSegment->FindCluster(seekTimeNs);
mBlockEntry = NULL;
mBlockEntryIndex = 0;
- do {
- advance_l();
- }
- while (!eos() && block()->GetTrackNumber() != mTrackNum);
+ long prevKeyFrameBlockEntryIndex = -1;
- if (seekToKeyFrame) {
- while (!eos() && !mBlockEntry->GetBlock()->IsKey()) {
- advance_l();
+ for (;;) {
+ advance_l();
+
+ if (eos()) {
+ break;
}
+
+ if (block()->GetTrackNumber() != mTrackNum) {
+ continue;
+ }
+
+ if (block()->IsKey()) {
+ prevKeyFrameBlockEntryIndex = mBlockEntryIndex - 1;
+ }
+
+ int64_t timeNs = block()->GetTime(mCluster);
+
+ if (timeNs >= seekTimeNs) {
+ *actualFrameTimeUs = (timeNs + 500ll) / 1000ll;
+ break;
+ }
+ }
+
+ if (eos()) {
+ return;
+ }
+
+ if (seekToKeyFrame && !block()->IsKey()) {
+ CHECK_GE(prevKeyFrameBlockEntryIndex, 0);
+ mBlockEntryIndex = prevKeyFrameBlockEntryIndex;
+ advance_l();
}
}
@@ -397,6 +430,8 @@
MediaBuffer **out, const ReadOptions *options) {
*out = NULL;
+ int64_t targetSampleTimeUs = -1ll;
+
int64_t seekTimeUs;
ReadOptions::SeekMode mode;
if (options && options->getSeekTo(&seekTimeUs, &mode)
@@ -406,10 +441,14 @@
// Apparently keyframe indication in audio tracks is unreliable,
// fortunately in all our currently supported audio encodings every
// frame is effectively a keyframe.
- mBlockIter.seek(seekTimeUs, !mIsAudio);
+ int64_t actualFrameTimeUs;
+ mBlockIter.seek(seekTimeUs, !mIsAudio, &actualFrameTimeUs);
+
+ if (mode == ReadOptions::SEEK_CLOSEST) {
+ targetSampleTimeUs = actualFrameTimeUs;
+ }
}
-again:
while (mPendingFrames.empty()) {
status_t err = readBlock();
@@ -424,6 +463,11 @@
mPendingFrames.erase(mPendingFrames.begin());
if (mType != AVC) {
+ if (targetSampleTimeUs >= 0ll) {
+ frame->meta_data()->setInt64(
+ kKeyTargetTime, targetSampleTimeUs);
+ }
+
*out = frame;
return OK;
@@ -506,6 +550,11 @@
frame->release();
frame = NULL;
+ if (targetSampleTimeUs >= 0ll) {
+ buffer->meta_data()->setInt64(
+ kKeyTargetTime, targetSampleTimeUs);
+ }
+
*out = buffer;
return OK;
@@ -610,36 +659,41 @@
return mIsLiveStreaming;
}
-static void addESDSFromAudioSpecificInfo(
- const sp<MetaData> &meta, const void *asi, size_t asiSize) {
+static void addESDSFromCodecPrivate(
+ const sp<MetaData> &meta,
+ bool isAudio, const void *priv, size_t privSize) {
static const uint8_t kStaticESDS[] = {
0x03, 22,
0x00, 0x00, // ES_ID
0x00, // streamDependenceFlag, URL_Flag, OCRstreamFlag
0x04, 17,
- 0x40, // Audio ISO/IEC 14496-3
+ 0x40, // ObjectTypeIndication
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00,
0x05,
- // AudioSpecificInfo (with size prefix) follows
+ // CodecSpecificInfo (with size prefix) follows
};
// Make sure all sizes can be coded in a single byte.
- CHECK(asiSize + 22 - 2 < 128);
- size_t esdsSize = sizeof(kStaticESDS) + asiSize + 1;
+ CHECK(privSize + 22 - 2 < 128);
+ size_t esdsSize = sizeof(kStaticESDS) + privSize + 1;
uint8_t *esds = new uint8_t[esdsSize];
memcpy(esds, kStaticESDS, sizeof(kStaticESDS));
uint8_t *ptr = esds + sizeof(kStaticESDS);
- *ptr++ = asiSize;
- memcpy(ptr, asi, asiSize);
+ *ptr++ = privSize;
+ memcpy(ptr, priv, privSize);
// Increment by codecPrivateSize less 2 bytes that are accounted for
// already in lengths of 22/17
- esds[1] += asiSize - 2;
- esds[6] += asiSize - 2;
+ esds[1] += privSize - 2;
+ esds[6] += privSize - 2;
+
+ // Set ObjectTypeIndication.
+ esds[7] = isAudio ? 0x40 // Audio ISO/IEC 14496-3
+ : 0x20; // Visual ISO/IEC 14496-2
meta->setData(kKeyESDS, 0, esds, esdsSize);
@@ -707,9 +761,21 @@
if (!strcmp("V_MPEG4/ISO/AVC", codecID)) {
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
meta->setData(kKeyAVCC, 0, codecPrivate, codecPrivateSize);
+ } else if (!strcmp("V_MPEG4/ISO/ASP", codecID)) {
+ if (codecPrivateSize > 0) {
+ meta->setCString(
+ kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG4);
+ addESDSFromCodecPrivate(
+ meta, false, codecPrivate, codecPrivateSize);
+ } else {
+ ALOGW("%s is detected, but does not have configuration.",
+ codecID);
+ continue;
+ }
} else if (!strcmp("V_VP8", codecID)) {
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_VPX);
} else {
+ ALOGW("%s is not supported.", codecID);
continue;
}
@@ -727,13 +793,16 @@
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AAC);
CHECK(codecPrivateSize >= 2);
- addESDSFromAudioSpecificInfo(
- meta, codecPrivate, codecPrivateSize);
+ addESDSFromCodecPrivate(
+ meta, true, codecPrivate, codecPrivateSize);
} else if (!strcmp("A_VORBIS", codecID)) {
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_VORBIS);
addVorbisCodecInfo(meta, codecPrivate, codecPrivateSize);
+ } else if (!strcmp("A_MPEG/L3", codecID)) {
+ meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
} else {
+ ALOGW("%s is not supported.", codecID);
continue;
}
diff --git a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
index 03033f5..e1589b4 100644
--- a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
@@ -22,8 +22,8 @@
#include "include/LiveSession.h"
#include "include/NuCachedSource2.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/omx/OMX.cpp b/media/libstagefright/omx/OMX.cpp
index ace883c..f11fcd2 100644
--- a/media/libstagefright/omx/OMX.cpp
+++ b/media/libstagefright/omx/OMX.cpp
@@ -25,7 +25,7 @@
#include "../include/OMXNodeInstance.h"
#include <binder/IMemory.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <utils/threads.h>
#include "OMXMaster.h"
@@ -102,7 +102,7 @@
if (status != WOULD_BLOCK) {
// Other than join to self, the only other error return codes are
// whatever readyToRun() returns, and we don't override that
- CHECK_EQ(status, NO_ERROR);
+ CHECK_EQ(status, (status_t)NO_ERROR);
}
}
diff --git a/media/libstagefright/omx/OMXComponentBase.cpp b/media/libstagefright/omx/OMXComponentBase.cpp
index 35227a0..7d11dce 100644
--- a/media/libstagefright/omx/OMXComponentBase.cpp
+++ b/media/libstagefright/omx/OMXComponentBase.cpp
@@ -18,7 +18,7 @@
#include <stdlib.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
namespace android {
@@ -33,7 +33,7 @@
OMXComponentBase::~OMXComponentBase() {}
void OMXComponentBase::setComponentHandle(OMX_COMPONENTTYPE *handle) {
- CHECK_EQ(mComponentHandle, NULL);
+ CHECK(mComponentHandle == NULL);
mComponentHandle = handle;
}
diff --git a/media/libstagefright/omx/OMXMaster.cpp b/media/libstagefright/omx/OMXMaster.cpp
index d698939..6b6d0ab 100644
--- a/media/libstagefright/omx/OMXMaster.cpp
+++ b/media/libstagefright/omx/OMXMaster.cpp
@@ -24,7 +24,7 @@
#include <dlfcn.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
namespace android {
diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index 8938e33..099c4f5 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -24,8 +24,8 @@
#include <OMX_Component.h>
#include <binder/IMemory.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/HardwareAPI.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaErrors.h>
namespace android {
@@ -91,11 +91,11 @@
}
OMXNodeInstance::~OMXNodeInstance() {
- CHECK_EQ(mHandle, NULL);
+ CHECK(mHandle == NULL);
}
void OMXNodeInstance::setHandle(OMX::node_id node_id, OMX_HANDLETYPE handle) {
- CHECK_EQ(mHandle, NULL);
+ CHECK(mHandle == NULL);
mNodeID = node_id;
mHandle = handle;
}
diff --git a/media/libstagefright/omx/tests/Android.mk b/media/libstagefright/omx/tests/Android.mk
index 41c08be..0c0a70c 100644
--- a/media/libstagefright/omx/tests/Android.mk
+++ b/media/libstagefright/omx/tests/Android.mk
@@ -5,7 +5,7 @@
OMXHarness.cpp \
LOCAL_SHARED_LIBRARIES := \
- libstagefright libbinder libmedia libutils
+ libstagefright libbinder libmedia libutils libstagefright_foundation
LOCAL_C_INCLUDES := \
$(JNI_H_INCLUDE) \
diff --git a/media/libstagefright/omx/tests/OMXHarness.cpp b/media/libstagefright/omx/tests/OMXHarness.cpp
index 8faf544..fab1771 100644
--- a/media/libstagefright/omx/tests/OMXHarness.cpp
+++ b/media/libstagefright/omx/tests/OMXHarness.cpp
@@ -26,9 +26,9 @@
#include <binder/IServiceManager.h>
#include <binder/MemoryDealer.h>
#include <media/IMediaPlayerService.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaExtractor.h>
@@ -155,7 +155,7 @@
if (err == TIMED_OUT) {
return err;
}
- CHECK_EQ(err, OK);
+ CHECK_EQ(err, (status_t)OK);
}
}
@@ -317,7 +317,7 @@
EXPECT_SUCCESS(err, "allocatePortBuffers(input)");
err = dequeueMessageForNode(node, &msg, DEFAULT_TIMEOUT);
- CHECK_EQ(err, TIMED_OUT);
+ CHECK_EQ(err, (status_t)TIMED_OUT);
Vector<Buffer> outputBuffers;
err = allocatePortBuffers(dealer, node, 1, &outputBuffers);
@@ -412,7 +412,7 @@
// Make sure node doesn't just transition to loaded before we are done
// freeing all input and output buffers.
err = dequeueMessageForNode(node, &msg, DEFAULT_TIMEOUT);
- CHECK_EQ(err, TIMED_OUT);
+ CHECK_EQ(err, (status_t)TIMED_OUT);
for (size_t i = 0; i < inputBuffers.size(); ++i) {
err = mOMX->freeBuffer(node, 0, inputBuffers[i].mID);
@@ -420,7 +420,7 @@
}
err = dequeueMessageForNode(node, &msg, DEFAULT_TIMEOUT);
- CHECK_EQ(err, TIMED_OUT);
+ CHECK_EQ(err, (status_t)TIMED_OUT);
for (size_t i = 0; i < outputBuffers.size(); ++i) {
err = mOMX->freeBuffer(node, 1, outputBuffers[i].mID);
@@ -584,7 +584,7 @@
return UNKNOWN_ERROR;
}
- CHECK_EQ(seekSource->start(), OK);
+ CHECK_EQ(seekSource->start(), (status_t)OK);
sp<MediaSource> codec = OMXCodec::Create(
mOMX, source->getFormat(), false /* createEncoder */,
@@ -592,7 +592,7 @@
CHECK(codec != NULL);
- CHECK_EQ(codec->start(), OK);
+ CHECK_EQ(codec->start(), (status_t)OK);
int64_t durationUs;
CHECK(source->getFormat()->findInt64(kKeyDuration, &durationUs));
@@ -638,7 +638,7 @@
requestedSeekTimeUs, MediaSource::ReadOptions::SEEK_NEXT_SYNC);
if (seekSource->read(&buffer, &options) != OK) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
actualSeekTimeUs = -1;
} else {
CHECK(buffer != NULL);
@@ -659,7 +659,7 @@
err = codec->read(&buffer, &options);
options.clearSeekTo();
if (err == INFO_FORMAT_CHANGED) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
continue;
}
if (err == OK) {
@@ -670,7 +670,7 @@
continue;
}
} else {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
}
break;
@@ -679,7 +679,7 @@
if (requestedSeekTimeUs < 0) {
// Linear read.
if (err != OK) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
} else {
CHECK(buffer != NULL);
buffer->release();
@@ -694,8 +694,8 @@
"We attempted to seek beyond EOS and expected "
"ERROR_END_OF_STREAM to be returned, but instead "
"we found some other error.");
- CHECK_EQ(err, ERROR_END_OF_STREAM);
- CHECK_EQ(buffer, NULL);
+ CHECK_EQ(err, (status_t)ERROR_END_OF_STREAM);
+ CHECK(buffer == NULL);
} else {
EXPECT(err == OK,
"Expected a valid buffer to be returned from "
@@ -715,7 +715,7 @@
buffer->release();
buffer = NULL;
- CHECK_EQ(codec->stop(), OK);
+ CHECK_EQ(codec->stop(), (status_t)OK);
return UNKNOWN_ERROR;
}
@@ -725,7 +725,7 @@
}
}
- CHECK_EQ(codec->stop(), OK);
+ CHECK_EQ(codec->stop(), (status_t)OK);
return OK;
}
@@ -841,7 +841,7 @@
srand(seed);
sp<Harness> h = new Harness;
- CHECK_EQ(h->initCheck(), OK);
+ CHECK_EQ(h->initCheck(), (status_t)OK);
if (argc == 0) {
h->testAll();
diff --git a/media/libstagefright/tests/SurfaceMediaSource_test.cpp b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
index 76b507f..d7cec04 100644
--- a/media/libstagefright/tests/SurfaceMediaSource_test.cpp
+++ b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
@@ -35,7 +35,7 @@
#include <binder/ProcessState.h>
#include <ui/FramebufferNativeWindow.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
@@ -475,7 +475,7 @@
mr->setVideoFrameRate(fps);
mr->prepare();
ALOGV("Starting MediaRecorder...");
- CHECK_EQ(OK, mr->start());
+ CHECK_EQ((status_t)OK, mr->start());
return mr;
}
@@ -757,7 +757,7 @@
ASSERT_EQ(NO_ERROR, native_window_api_disconnect(mANW.get(), NATIVE_WINDOW_API_CPU));
ALOGV("Stopping MediaRecorder...");
- CHECK_EQ(OK, mr->stop());
+ CHECK_EQ((status_t)OK, mr->stop());
mr.clear();
close(fd);
}
@@ -886,7 +886,7 @@
mEglSurface = EGL_NO_SURFACE;
ALOGV("Stopping MediaRecorder...");
- CHECK_EQ(OK, mr->stop());
+ CHECK_EQ((status_t)OK, mr->stop());
mr.clear();
close(fd);
}
@@ -929,7 +929,7 @@
mEglSurface = EGL_NO_SURFACE;
ALOGV("Stopping MediaRecorder...");
- CHECK_EQ(OK, mr->stop());
+ CHECK_EQ((status_t)OK, mr->stop());
mr.clear();
close(fd);
}
diff --git a/media/libstagefright/timedtext/TimedTextInBandSource.cpp b/media/libstagefright/timedtext/TimedTextInBandSource.cpp
index f2c4d54..afb73fb 100644
--- a/media/libstagefright/timedtext/TimedTextInBandSource.cpp
+++ b/media/libstagefright/timedtext/TimedTextInBandSource.cpp
@@ -19,8 +19,8 @@
#include <utils/Log.h>
#include <binder/Parcel.h>
+#include <media/stagefright/foundation/ADebug.h> // CHECK_XX macro
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h> // CHECK_XX macro
#include <media/stagefright/MediaDefs.h> // for MEDIA_MIMETYPE_xxx
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/timedtext/TimedTextPlayer.cpp b/media/libstagefright/timedtext/TimedTextPlayer.cpp
index 8c2df88..bf7cbf6 100644
--- a/media/libstagefright/timedtext/TimedTextPlayer.cpp
+++ b/media/libstagefright/timedtext/TimedTextPlayer.cpp
@@ -18,8 +18,8 @@
#define LOG_TAG "TimedTextPlayer"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaErrors.h>
#include <media/MediaPlayerInterface.h>
diff --git a/media/libstagefright/yuv/YUVCanvas.cpp b/media/libstagefright/yuv/YUVCanvas.cpp
index 38aa779..4c9fee8 100644
--- a/media/libstagefright/yuv/YUVCanvas.cpp
+++ b/media/libstagefright/yuv/YUVCanvas.cpp
@@ -17,7 +17,7 @@
#define LOG_NDEBUG 0
#define LOG_TAG "YUVCanvas"
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/YUVCanvas.h>
#include <media/stagefright/YUVImage.h>
#include <ui/Rect.h>
diff --git a/media/libstagefright/yuv/YUVImage.cpp b/media/libstagefright/yuv/YUVImage.cpp
index 0d67c96..7b9000b 100644
--- a/media/libstagefright/yuv/YUVImage.cpp
+++ b/media/libstagefright/yuv/YUVImage.cpp
@@ -17,9 +17,9 @@
#define LOG_NDEBUG 0
#define LOG_TAG "YUVImage"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/YUVImage.h>
#include <ui/Rect.h>
-#include <media/stagefright/MediaDebug.h>
namespace android {
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 52834db..157405a 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -6,9 +6,10 @@
AudioFlinger.cpp \
AudioMixer.cpp.arm \
AudioResampler.cpp.arm \
- AudioResamplerSinc.cpp.arm \
- AudioResamplerCubic.cpp.arm \
- AudioPolicyService.cpp
+ AudioPolicyService.cpp \
+ ServiceUtilities.cpp
+# AudioResamplerSinc.cpp.arm
+# AudioResamplerCubic.cpp.arm
LOCAL_C_INCLUDES := \
system/media/audio_effects/include \
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index f71ba0a..5c964b2 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -48,6 +48,7 @@
#include "AudioMixer.h"
#include "AudioFlinger.h"
+#include "ServiceUtilities.h"
#include <media/EffectsFactoryApi.h>
#include <audio_effects/effect_visualizer.h>
@@ -101,20 +102,6 @@
// ----------------------------------------------------------------------------
-static bool recordingAllowed() {
- if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
- bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
- if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO");
- return ok;
-}
-
-static bool settingsAllowed() {
- if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
- bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
- if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
- return ok;
-}
-
// To collect the amplifier usage
static void addBatteryData(uint32_t params) {
sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
@@ -188,29 +175,32 @@
mod->name, mod->id);
mAudioHwDevs.push(dev);
- if (!mPrimaryHardwareDev) {
+ if (mPrimaryHardwareDev == NULL) {
mPrimaryHardwareDev = dev;
ALOGI("Using '%s' (%s.%s) as the primary audio interface",
mod->name, mod->id, audio_interfaces[i]);
}
}
- mHardwareStatus = AUDIO_HW_INIT;
-
- if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
+ if (mPrimaryHardwareDev == NULL) {
ALOGE("Primary audio interface not found");
- return;
+ // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck()
}
+ // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the
+ // primary HW dev is selected can change so these conditions might not always be equivalent.
+ // When that happens, re-visit all the code that assumes this.
+
+ AutoMutex lock(mHardwareLock);
+
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
mHardwareStatus = AUDIO_HW_INIT;
rc = dev->init_check(dev);
+ mHardwareStatus = AUDIO_HW_IDLE;
if (rc == 0) {
- AutoMutex lock(mHardwareLock);
-
- mMode = AUDIO_MODE_NORMAL;
+ mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value
mHardwareStatus = AUDIO_HW_SET_MODE;
dev->set_mode(dev, mMode);
mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
@@ -220,17 +210,8 @@
}
}
-status_t AudioFlinger::initCheck() const
-{
- Mutex::Autolock _l(mLock);
- if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
- return NO_INIT;
- return NO_ERROR;
-}
-
AudioFlinger::~AudioFlinger()
{
- int num_devs = mAudioHwDevs.size();
while (!mRecordThreads.isEmpty()) {
// closeInput() will remove first entry from mRecordThreads
@@ -241,11 +222,10 @@
closeOutput(mPlaybackThreads.keyAt(0));
}
- for (int i = 0; i < num_devs; i++) {
- audio_hw_device_t *dev = mAudioHwDevs[i];
- audio_hw_device_close(dev);
+ for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
+ // no mHardwareLock needed, as there are no other references to this
+ audio_hw_device_close(mAudioHwDevs[i]);
}
- mAudioHwDevs.clear();
}
audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
@@ -328,7 +308,7 @@
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
{
- if (!checkCallingPermission(String16("android.permission.DUMP"))) {
+ if (!dumpAllowed()) {
dumpPermissionDenial(fd, args);
} else {
// get state of hardware lock
@@ -371,6 +351,18 @@
return NO_ERROR;
}
+sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
+{
+ // If pid is already in the mClients wp<> map, then use that entry
+ // (for which promote() is always != 0), otherwise create a new entry and Client.
+ sp<Client> client = mClients.valueFor(pid).promote();
+ if (client == 0) {
+ client = new Client(this, pid);
+ mClients.add(pid, client);
+ }
+
+ return client;
+}
// IAudioFlinger interface
@@ -384,14 +376,13 @@
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
int *sessionId,
status_t *status)
{
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
- wp<Client> wclient;
status_t lStatus;
int lSessionId;
@@ -413,14 +404,7 @@
goto Exit;
}
- wclient = mClients.valueFor(pid);
-
- if (wclient != NULL) {
- client = wclient.promote();
- } else {
- client = new Client(this, pid);
- mClients.add(pid, client);
- }
+ client = registerPid_l(pid);
ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
@@ -477,7 +461,7 @@
return trackHandle;
}
-uint32_t AudioFlinger::sampleRate(int output) const
+uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -488,7 +472,7 @@
return thread->sampleRate();
}
-int AudioFlinger::channelCount(int output) const
+int AudioFlinger::channelCount(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -499,7 +483,7 @@
return thread->channelCount();
}
-audio_format_t AudioFlinger::format(int output) const
+audio_format_t AudioFlinger::format(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -510,7 +494,7 @@
return thread->format();
}
-size_t AudioFlinger::frameCount(int output) const
+size_t AudioFlinger::frameCount(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -521,7 +505,7 @@
return thread->frameCount();
}
-uint32_t AudioFlinger::latency(int output) const
+uint32_t AudioFlinger::latency(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -622,6 +606,7 @@
}
bool state = AUDIO_MODE_INVALID;
+ AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
mHardwareStatus = AUDIO_HW_IDLE;
@@ -655,7 +640,8 @@
return masterMute_l();
}
-status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, int output)
+status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output)
{
// check calling permissions
if (!settingsAllowed()) {
@@ -710,7 +696,7 @@
return NO_ERROR;
}
-float AudioFlinger::streamVolume(audio_stream_type_t stream, int output) const
+float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return 0.0f;
@@ -740,11 +726,11 @@
return mStreamTypes[stream].mute;
}
-status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
+status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
{
status_t result;
- ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
+ ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d",
ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
// check calling permissions
if (!settingsAllowed()) {
@@ -809,16 +795,15 @@
}
}
}
- if (thread != NULL) {
- result = thread->setParameters(keyValuePairs);
- return result;
+ if (thread != 0) {
+ return thread->setParameters(keyValuePairs);
}
return BAD_VALUE;
}
-String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
+String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
{
-// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
+// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d",
// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
if (ioHandle == 0) {
@@ -846,17 +831,21 @@
return String8("");
}
-size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount)
+size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return 0;
}
- return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
+ size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return size;
}
-unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
+unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
{
if (ioHandle == 0) {
return 0;
@@ -891,7 +880,8 @@
return ret;
}
-status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
+status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const
{
status_t status;
@@ -910,7 +900,7 @@
Mutex::Autolock _l(mLock);
- int pid = IPCThreadState::self()->getCallingPid();
+ pid_t pid = IPCThreadState::self()->getCallingPid();
if (mNotificationClients.indexOfKey(pid) < 0) {
sp<NotificationClient> notificationClient = new NotificationClient(this,
client,
@@ -966,7 +956,7 @@
}
// audioConfigChanged_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
+void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2)
{
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
@@ -985,8 +975,8 @@
// ----------------------------------------------------------------------------
-AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device,
- type_t type)
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+ uint32_t device, type_t type)
: Thread(false),
mType(type),
mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0),
@@ -994,7 +984,7 @@
mChannelCount(0),
mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
mParamStatus(NO_ERROR),
- mStandby(false), mId(id), mExiting(false),
+ mStandby(false), mId(id),
mDevice(device),
mDeathRecipient(new PMDeathRecipient(this))
{
@@ -1013,40 +1003,26 @@
void AudioFlinger::ThreadBase::exit()
{
- // keep a strong ref on ourself so that we won't get
- // destroyed in the middle of requestExitAndWait()
- sp <ThreadBase> strongMe = this;
-
ALOGV("ThreadBase::exit");
{
+ // This lock prevents the following race in thread (uniprocessor for illustration):
+ // if (!exitPending()) {
+ // // context switch from here to exit()
+ // // exit() calls requestExit(), what exitPending() observes
+ // // exit() calls signal(), which is dropped since no waiters
+ // // context switch back from exit() to here
+ // mWaitWorkCV.wait(...);
+ // // now thread is hung
+ // }
AutoMutex lock(mLock);
- mExiting = true;
requestExit();
mWaitWorkCV.signal();
}
+ // When Thread::requestExitAndWait is made virtual and this method is renamed to
+ // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
requestExitAndWait();
}
-uint32_t AudioFlinger::ThreadBase::sampleRate() const
-{
- return mSampleRate;
-}
-
-int AudioFlinger::ThreadBase::channelCount() const
-{
- return (int)mChannelCount;
-}
-
-audio_format_t AudioFlinger::ThreadBase::format() const
-{
- return mFormat;
-}
-
-size_t AudioFlinger::ThreadBase::frameCount() const
-{
- return mFrameCount;
-}
-
status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
{
status_t status;
@@ -1248,8 +1224,7 @@
void AudioFlinger::ThreadBase::setEffectSuspended_l(
const effect_uuid_t *type, bool suspend, int sessionId)
{
- sp<EffectChain> chain;
- chain = getEffectChain_l(sessionId);
+ sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
if (type != NULL) {
chain->setEffectSuspended_l(type, suspend);
@@ -1377,7 +1352,7 @@
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
- int id,
+ audio_io_handle_t id,
uint32_t device,
type_t type)
: ThreadBase(audioFlinger, id, device, type),
@@ -1541,12 +1516,11 @@
// all tracks in same audio session must share the same routing strategy otherwise
// conflicts will happen when tracks are moved from one output to another by audio policy
// manager
- uint32_t strategy =
- AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
+ uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> t = mTracks[i];
if (t != 0) {
- uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type());
+ uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
if (sessionId == t->sessionId() && strategy != actual) {
ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
strategy, actual);
@@ -1568,7 +1542,7 @@
if (chain != 0) {
ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
- chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
+ chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
chain->incTrackCnt();
}
@@ -1611,16 +1585,6 @@
return NO_ERROR;
}
-float AudioFlinger::PlaybackThread::masterVolume() const
-{
- return mMasterVolume;
-}
-
-bool AudioFlinger::PlaybackThread::masterMute() const
-{
- return mMasterMute;
-}
-
status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
{
mStreamTypes[stream].volume = value;
@@ -1808,7 +1772,7 @@
sp<Track> track = mTracks[i];
if (sessionId == track->sessionId() &&
!(track->mCblk->flags & CBLK_INVALID_MSK)) {
- return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
+ return AudioSystem::getStrategyForStream(track->streamType());
}
}
return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
@@ -1853,7 +1817,7 @@
// ----------------------------------------------------------------------------
AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
- int id, uint32_t device, type_t type)
+ audio_io_handle_t id, uint32_t device, type_t type)
: PlaybackThread(audioFlinger, output, id, device, type),
mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)),
mPrevMixerStatus(MIXER_IDLE)
@@ -1947,7 +1911,7 @@
if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
mSuspended)) {
if (!mStandby) {
- ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
+ ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended);
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
@@ -1961,9 +1925,9 @@
releaseWakeLock_l();
// wait until we have something to do...
- ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
+ ALOGV("MixerThread %p TID %d going to sleep", this, gettid());
mWaitWorkCV.wait(mLock);
- ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
+ ALOGV("MixerThread %p TID %d waking up", this, gettid());
acquireWakeLock_l();
mPrevMixerStatus = MIXER_IDLE;
@@ -2189,7 +2153,7 @@
// compute volume for this track
uint32_t vl, vr, va;
if (track->isMuted() || track->isPausing() ||
- mStreamTypes[track->type()].mute) {
+ mStreamTypes[track->streamType()].mute) {
vl = vr = va = 0;
if (track->isPausing()) {
track->setPaused();
@@ -2197,7 +2161,7 @@
} else {
// read original volumes with volume control
- float typeVolume = mStreamTypes[track->type()].volume;
+ float typeVolume = mStreamTypes[track->streamType()].volume;
float v = masterVolume * typeVolume;
uint32_t vlr = cblk->getVolumeLR();
vl = vlr & 0xFFFF;
@@ -2358,7 +2322,7 @@
size_t size = mTracks.size();
for (size_t i = 0; i < size; i++) {
sp<Track> t = mTracks[i];
- if (t->type() == streamType) {
+ if (t->streamType() == streamType) {
android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
t->mCblk->cv.signal();
}
@@ -2520,7 +2484,8 @@
}
// ----------------------------------------------------------------------------
-AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
+AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
+ AudioStreamOut* output, audio_io_handle_t id, uint32_t device)
: PlaybackThread(audioFlinger, output, id, device, DIRECT)
// mLeftVolFloat, mRightVolFloat
// mLeftVolShort, mRightVolShort
@@ -2531,21 +2496,6 @@
{
}
-static inline
-int32_t mul(int16_t in, int16_t v)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- asm( "smulbb %[out], %[in], %[v] \n"
- : [out]"=r"(out)
- : [in]"%r"(in), [v]"r"(v)
- : );
- return out;
-#else
- return in * int32_t(v);
-#endif
-}
-
void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
{
// Do not apply volume on compressed audio
@@ -2664,7 +2614,7 @@
mSuspended)) {
// wait until we have something to do...
if (!mStandby) {
- ALOGV("Audio hardware entering standby, mixer %p\n", this);
+ ALOGV("Audio hardware entering standby, mixer %p", this);
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
@@ -2677,9 +2627,9 @@
if (exitPending()) break;
releaseWakeLock_l();
- ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
+ ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid());
mWaitWorkCV.wait(mLock);
- ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
+ ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid());
acquireWakeLock_l();
if (!mMasterMute) {
@@ -2730,13 +2680,13 @@
// compute volume for this track
float left, right;
if (track->isMuted() || mMasterMute || track->isPausing() ||
- mStreamTypes[track->type()].mute) {
+ mStreamTypes[track->streamType()].mute) {
left = right = 0;
if (track->isPausing()) {
track->setPaused();
}
} else {
- float typeVolume = mStreamTypes[track->type()].volume;
+ float typeVolume = mStreamTypes[track->streamType()].volume;
float v = mMasterVolume * typeVolume;
uint32_t vlr = cblk->getVolumeLR();
float v_clamped = v * (vlr & 0xFFFF);
@@ -3002,7 +2952,7 @@
// ----------------------------------------------------------------------------
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
- AudioFlinger::MixerThread* mainThread, int id)
+ AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
: MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING),
mWaitTimeMs(UINT_MAX)
{
@@ -3014,7 +2964,6 @@
for (size_t i = 0; i < mOutputTracks.size(); i++) {
mOutputTracks[i]->destroy();
}
- mOutputTracks.clear();
}
bool AudioFlinger::DuplicatingThread::threadLoop()
@@ -3073,9 +3022,9 @@
if (exitPending()) break;
releaseWakeLock_l();
- ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
+ ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid());
mWaitWorkCV.wait(mLock);
- ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
+ ALOGV("DuplicatingThread %p TID %d waking up", this, gettid());
acquireWakeLock_l();
mPrevMixerStatus = MIXER_IDLE;
@@ -3206,7 +3155,7 @@
mWaitTimeMs = UINT_MAX;
for (size_t i = 0; i < mOutputTracks.size(); i++) {
sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
- if (strong != NULL) {
+ if (strong != 0) {
uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
if (waitTimeMs < mWaitTimeMs) {
mWaitTimeMs = waitTimeMs;
@@ -3259,7 +3208,6 @@
// mBufferEnd
mFrameCount(0),
mState(IDLE),
- mClientTid(-1),
mFormat(format),
mFlags(flags & ~SYSTEM_FLAGS_MASK),
mSessionId(sessionId)
@@ -3324,15 +3272,19 @@
AudioFlinger::ThreadBase::TrackBase::~TrackBase()
{
if (mCblk != NULL) {
- mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
- if (mClient == NULL) {
+ if (mClient == 0) {
delete mCblk;
+ } else {
+ mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
}
}
- mCblkMemory.clear(); // and free the shared memory
- if (mClient != NULL) {
+ mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
+ if (mClient != 0) {
// Client destructor must run with AudioFlinger mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
+ // If the client's reference count drops to zero, the associated destructor
+ // must run with AudioFlinger lock held. Thus the explicit clear() rather than
+ // relying on the automatic clear() at end of scope.
mClient.clear();
}
}
@@ -3368,23 +3320,10 @@
ALOGV("TrackBase::reset");
}
-sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
-{
- return mCblkMemory;
-}
-
int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
return (int)mCblk->sampleRate;
}
-int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
- return (const int)mChannelCount;
-}
-
-uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
- return mChannelMask;
-}
-
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
size_t frameSize = cblk->frameSize;
@@ -3428,7 +3367,7 @@
mName = playbackThread->getTrackName_l();
mMainBuffer = playbackThread->mixBuffer();
}
- ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid());
if (mName < 0) {
ALOGE("no more track names available");
}
@@ -3465,9 +3404,7 @@
if (thread != 0) {
if (!isOutputTrack()) {
if (mState == ACTIVE || mState == RESUMING) {
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
@@ -3486,7 +3423,7 @@
uint32_t vlr = mCblk->getVolumeLR();
snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
mName - AudioMixer::TRACK0,
- (mClient == NULL) ? getpid() : mClient->pid(),
+ (mClient == 0) ? getpid_cached : mClient->pid(),
mStreamType,
mFormat,
mChannelMask,
@@ -3560,7 +3497,7 @@
status_t AudioFlinger::PlaybackThread::Track::start()
{
status_t status = NO_ERROR;
- ALOGV("start(%d), calling thread %d session %d",
+ ALOGV("start(%d), calling pid %d session %d",
mName, IPCThreadState::self()->getCallingPid(), mSessionId);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
@@ -3578,9 +3515,7 @@
if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
thread->mLock.unlock();
- status = AudioSystem::startOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
@@ -3602,7 +3537,7 @@
void AudioFlinger::PlaybackThread::Track::stop()
{
- ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
@@ -3618,9 +3553,7 @@
}
if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
thread->mLock.unlock();
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
@@ -3631,7 +3564,7 @@
void AudioFlinger::PlaybackThread::Track::pause()
{
- ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
@@ -3640,9 +3573,7 @@
ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
if (!isOutputTrack()) {
thread->mLock.unlock();
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
@@ -3813,7 +3744,7 @@
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n",
- (mClient == NULL) ? getpid() : mClient->pid(),
+ (mClient == 0) ? getpid_cached : mClient->pid(),
mFormat,
mChannelMask,
mSessionId,
@@ -4160,7 +4091,7 @@
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
- int input,
+ audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
@@ -4172,7 +4103,6 @@
sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
- wp<Client> wclient;
status_t lStatus;
RecordThread *thread;
size_t inFrameCount;
@@ -4193,13 +4123,7 @@
goto Exit;
}
- wclient = mClients.valueFor(pid);
- if (wclient != NULL) {
- client = wclient.promote();
- } else {
- client = new Client(this, pid);
- mClients.add(pid, client);
- }
+ client = registerPid_l(pid);
// If no audio session id is provided, create one here
if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
@@ -4277,7 +4201,7 @@
AudioStreamIn *input,
uint32_t sampleRate,
uint32_t channels,
- int id,
+ audio_io_handle_t id,
uint32_t device) :
ThreadBase(audioFlinger, id, device, RECORD),
mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
@@ -4528,7 +4452,7 @@
track = new RecordTrack(this, client, sampleRate,
format, channelMask, frameCount, flags, sessionId);
- if (track->getCblk() == NULL) {
+ if (track->getCblk() == 0) {
lStatus = NO_MEMORY;
goto Exit;
}
@@ -4584,7 +4508,7 @@
ALOGV("Signal record thread");
mWaitWorkCV.signal();
// do not wait for mStartStopCond if exiting
- if (mExiting) {
+ if (exitPending()) {
mActiveTrack.clear();
status = INVALID_OPERATION;
goto startError;
@@ -4611,7 +4535,7 @@
if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
mActiveTrack->mState = TrackBase::PAUSING;
// do not wait for mStartStopCond if exiting
- if (mExiting) {
+ if (exitPending()) {
return;
}
mStartStopCond.wait(mLock);
@@ -4631,7 +4555,6 @@
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
- pid_t pid = 0;
snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
result.append(buffer);
@@ -4938,7 +4861,7 @@
// ----------------------------------------------------------------------------
-int AudioFlinger::openOutput(uint32_t *pDevices,
+audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
@@ -4984,7 +4907,7 @@
mHardwareStatus = AUDIO_HW_IDLE;
if (outStream != NULL) {
AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
- int id = nextUniqueId();
+ audio_io_handle_t id = nextUniqueId();
if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
(format != AUDIO_FORMAT_PCM_16_BIT) ||
@@ -5010,7 +4933,8 @@
return 0;
}
-int AudioFlinger::openDuplicateOutput(int output1, int output2)
+audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2)
{
Mutex::Autolock _l(mLock);
MixerThread *thread1 = checkMixerThread_l(output1);
@@ -5021,7 +4945,7 @@
return 0;
}
- int id = nextUniqueId();
+ audio_io_handle_t id = nextUniqueId();
DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
thread->addOutputTrack(thread2);
mPlaybackThreads.add(id, thread);
@@ -5030,7 +4954,7 @@
return id;
}
-status_t AudioFlinger::closeOutput(int output)
+status_t AudioFlinger::closeOutput(audio_io_handle_t output)
{
// keep strong reference on the playback thread so that
// it is not destroyed while exit() is executed
@@ -5057,6 +4981,8 @@
mPlaybackThreads.removeItem(output);
}
thread->exit();
+ // The thread entity (active unit of execution) is no longer running here,
+ // but the ThreadBase container still exists.
if (thread->type() != ThreadBase::DUPLICATING) {
AudioStreamOut *out = thread->clearOutput();
@@ -5068,7 +4994,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::suspendOutput(int output)
+status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -5083,7 +5009,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::restoreOutput(int output)
+status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -5099,7 +5025,7 @@
return NO_ERROR;
}
-int AudioFlinger::openInput(uint32_t *pDevices,
+audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
audio_format_t *pFormat,
uint32_t *pChannels,
@@ -5155,7 +5081,7 @@
if (inStream != NULL) {
AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
- int id = nextUniqueId();
+ audio_io_handle_t id = nextUniqueId();
// Start record thread
// RecorThread require both input and output device indication to forward to audio
// pre processing modules
@@ -5182,7 +5108,7 @@
return 0;
}
-status_t AudioFlinger::closeInput(int input)
+status_t AudioFlinger::closeInput(audio_io_handle_t input)
{
// keep strong reference on the record thread so that
// it is not destroyed while exit() is executed
@@ -5200,6 +5126,8 @@
mRecordThreads.removeItem(input);
}
thread->exit();
+ // The thread entity (active unit of execution) is no longer running here,
+ // but the ThreadBase container still exists.
AudioStreamIn *in = thread->clearInput();
assert(in != NULL);
@@ -5210,7 +5138,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output)
+status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
MixerThread *dstThread = checkMixerThread_l(output);
@@ -5246,7 +5174,7 @@
void AudioFlinger::acquireAudioSessionId(int audioSession)
{
Mutex::Autolock _l(mLock);
- int caller = IPCThreadState::self()->getCallingPid();
+ pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("acquiring %d from %d", audioSession, caller);
int num = mAudioSessionRefs.size();
for (int i = 0; i< num; i++) {
@@ -5264,7 +5192,7 @@
void AudioFlinger::releaseAudioSessionId(int audioSession)
{
Mutex::Autolock _l(mLock);
- int caller = IPCThreadState::self()->getCallingPid();
+ pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("releasing %d from %d", audioSession, caller);
int num = mAudioSessionRefs.size();
for (int i = 0; i< num; i++) {
@@ -5348,7 +5276,7 @@
}
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
+AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
{
PlaybackThread *thread = NULL;
if (mPlaybackThreads.indexOfKey(output) >= 0) {
@@ -5358,7 +5286,7 @@
}
// checkMixerThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
+AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
{
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread != NULL) {
@@ -5370,7 +5298,7 @@
}
// checkRecordThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
+AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
{
RecordThread *thread = NULL;
if (mRecordThreads.indexOfKey(input) >= 0) {
@@ -5413,19 +5341,20 @@
// ----------------------------------------------------------------------------
-status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
+status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
{
Mutex::Autolock _l(mLock);
return EffectQueryNumberEffects(numEffects);
}
-status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
+status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
{
Mutex::Autolock _l(mLock);
return EffectQueryEffect(index, descriptor);
}
-status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
+status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
+ effect_descriptor_t *descriptor) const
{
Mutex::Autolock _l(mLock);
return EffectGetDescriptor(pUuid, descriptor);
@@ -5436,7 +5365,7 @@
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
- int io,
+ audio_io_handle_t io,
int sessionId,
status_t *status,
int *id,
@@ -5445,10 +5374,8 @@
status_t lStatus = NO_ERROR;
sp<EffectHandle> handle;
effect_descriptor_t desc;
- sp<Client> client;
- wp<Client> wclient;
- ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
+ ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
pid, effectClient.get(), priority, sessionId, io);
if (pDesc == NULL) {
@@ -5464,7 +5391,7 @@
// Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
// that can only be created by audio policy manager (running in same process)
- if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
+ if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
@@ -5599,14 +5526,7 @@
}
}
- wclient = mClients.valueFor(pid);
-
- if (wclient != NULL) {
- client = wclient.promote();
- } else {
- client = new Client(this, pid);
- mClients.add(pid, client);
- }
+ sp<Client> client = registerPid_l(pid);
// create effect on selected output thread
handle = thread->createEffect_l(client, effectClient, priority, sessionId,
@@ -5623,7 +5543,8 @@
return handle;
}
-status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
+status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput)
{
ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
sessionId, srcOutput, dstOutput);
@@ -5674,7 +5595,7 @@
// transfer all effects one by one so that new effect chain is created on new thread with
// correct buffer sizes and audio parameters and effect engines reconfigured accordingly
- int dstOutput = dstThread->id();
+ audio_io_handle_t dstOutput = dstThread->id();
sp<EffectChain> dstChain;
uint32_t strategy = 0; // prevent compiler warning
sp<EffectModule> effect = chain->getEffectFromId_l(0);
@@ -5830,13 +5751,8 @@
sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
{
- sp<EffectModule> effect;
-
sp<EffectChain> chain = getEffectChain_l(sessionId);
- if (chain != 0) {
- effect = chain->getEffectFromId_l(effectId);
- }
- return effect;
+ return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
}
// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
@@ -5921,16 +5837,13 @@
sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
{
- sp<EffectChain> chain;
-
size_t size = mEffectChains.size();
for (size_t i = 0; i < size; i++) {
if (mEffectChains[i]->sessionId() == sessionId) {
- chain = mEffectChains[i];
- break;
+ return mEffectChains[i];
}
}
- return chain;
+ return 0;
}
void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
@@ -5944,13 +5857,13 @@
void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
const wp<EffectHandle>& handle,
- bool unpiniflast) {
+ bool unpinIfLast) {
Mutex::Autolock _l(mLock);
ALOGV("disconnectEffect() %p effect %p", this, effect.get());
// delete the effect module if removing last handle on it
if (effect->removeHandle(handle) == 0) {
- if (!effect->isPinned() || unpiniflast) {
+ if (!effect->isPinned() || unpinIfLast) {
removeEffect_l(effect);
AudioSystem::unregisterEffect(effect->id());
}
@@ -6263,23 +6176,19 @@
sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
{
Mutex::Autolock _l(mLock);
- sp<EffectHandle> handle;
- if (mHandles.size() != 0) {
- handle = mHandles[0].promote();
- }
- return handle;
+ return mHandles.size() != 0 ? mHandles[0].promote() : 0;
}
-void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
+void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast)
{
- ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
+ ALOGV("disconnect() %p handle %p", this, handle.unsafe_get());
// keep a strong reference on this EffectModule to avoid calling the
// destructor before we exit
sp<EffectModule> keep(this);
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
- thread->disconnectEffect(keep, handle, unpiniflast);
+ thread->disconnectEffect(keep, handle, unpinIfLast);
}
}
}
@@ -6622,7 +6531,7 @@
return NO_ERROR;
}
-bool AudioFlinger::EffectModule::isEnabled()
+bool AudioFlinger::EffectModule::isEnabled() const
{
switch (mState) {
case RESTART:
@@ -6638,7 +6547,7 @@
}
}
-bool AudioFlinger::EffectModule::isProcessEnabled()
+bool AudioFlinger::EffectModule::isProcessEnabled() const
{
switch (mState) {
case RESTART:
@@ -6951,13 +6860,13 @@
disconnect(true);
}
-void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
+void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast)
{
- ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
+ ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false");
if (mEffect == 0) {
return;
}
- mEffect->disconnect(this, unpiniflast);
+ mEffect->disconnect(this, unpinIfLast);
if (mHasControl && mEnabled) {
sp<ThreadBase> thread = mEffect->thread().promote();
@@ -6970,9 +6879,11 @@
mEffect.clear();
if (mClient != 0) {
if (mCblk != NULL) {
+ // unlike ~TrackBase(), mCblk is never a local new, so don't delete
mCblk->~effect_param_cblk_t(); // destroy our shared-structure.
}
- mCblkMemory.clear(); // and free the shared memory
+ mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
+ // Client destructor must run with AudioFlinger mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
mClient.clear();
}
@@ -7054,10 +6965,6 @@
return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
}
-sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
- return mCblkMemory;
-}
-
void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
{
ALOGV("setControl %p control %d", this, hasControl);
@@ -7102,7 +7009,7 @@
bool locked = mCblk != NULL && tryLock(mCblk->lock);
snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n",
- (mClient == NULL) ? getpid() : mClient->pid(),
+ (mClient == 0) ? getpid_cached : mClient->pid(),
mPriority,
mHasControl,
!locked,
@@ -7144,48 +7051,42 @@
// getEffectFromDesc_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
{
- sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
- effect = mEffects[i];
- break;
+ return mEffects[i];
}
}
- return effect;
+ return 0;
}
// getEffectFromId_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
{
- sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
// by convention, return first effect if id provided is 0 (0 is never a valid id)
if (id == 0 || mEffects[i]->id() == id) {
- effect = mEffects[i];
- break;
+ return mEffects[i];
}
}
- return effect;
+ return 0;
}
// getEffectFromType_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
const effect_uuid_t *type)
{
- sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
- effect = mEffects[i];
- break;
+ return mEffects[i];
}
}
- return effect;
+ return 0;
}
// Must be called with EffectChain::mLock locked
@@ -7626,12 +7527,8 @@
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
const effect_uuid_t *type)
{
- sp<EffectModule> effect;
- effect = getEffectFromType_l(type);
- if (effect != 0 && !effect->isEnabled()) {
- effect.clear();
- }
- return effect;
+ sp<EffectModule> effect = getEffectFromType_l(type);
+ return effect != 0 && effect->isEnabled() ? effect : 0;
}
void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 3f3188c..fdcd916 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -67,7 +67,7 @@
virtual status_t dump(int fd, const Vector<String16>& args);
- // IAudioFlinger interface
+ // IAudioFlinger interface, in binder opcode order
virtual sp<IAudioTrack> createTrack(
pid_t pid,
audio_stream_type_t streamType,
@@ -77,117 +77,13 @@
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
int *sessionId,
status_t *status);
- virtual uint32_t sampleRate(int output) const;
- virtual int channelCount(int output) const;
- virtual audio_format_t format(int output) const;
- virtual size_t frameCount(int output) const;
- virtual uint32_t latency(int output) const;
-
- virtual status_t setMasterVolume(float value);
- virtual status_t setMasterMute(bool muted);
-
- virtual float masterVolume() const;
- virtual bool masterMute() const;
-
- virtual status_t setStreamVolume(audio_stream_type_t stream, float value, int output);
- virtual status_t setStreamMute(audio_stream_type_t stream, bool muted);
-
- virtual float streamVolume(audio_stream_type_t stream, int output) const;
- virtual bool streamMute(audio_stream_type_t stream) const;
-
- virtual status_t setMode(audio_mode_t mode);
-
- virtual status_t setMicMute(bool state);
- virtual bool getMicMute() const;
-
- virtual status_t setParameters(int ioHandle, const String8& keyValuePairs);
- virtual String8 getParameters(int ioHandle, const String8& keys);
-
- virtual void registerClient(const sp<IAudioFlingerClient>& client);
-
- virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount);
- virtual unsigned int getInputFramesLost(int ioHandle);
-
- virtual int openOutput(uint32_t *pDevices,
- uint32_t *pSamplingRate,
- audio_format_t *pFormat,
- uint32_t *pChannels,
- uint32_t *pLatencyMs,
- uint32_t flags);
-
- virtual int openDuplicateOutput(int output1, int output2);
-
- virtual status_t closeOutput(int output);
-
- virtual status_t suspendOutput(int output);
-
- virtual status_t restoreOutput(int output);
-
- virtual int openInput(uint32_t *pDevices,
- uint32_t *pSamplingRate,
- audio_format_t *pFormat,
- uint32_t *pChannels,
- audio_in_acoustics_t acoustics);
-
- virtual status_t closeInput(int input);
-
- virtual status_t setStreamOutput(audio_stream_type_t stream, int output);
-
- virtual status_t setVoiceVolume(float volume);
-
- virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output);
-
- virtual int newAudioSessionId();
-
- virtual void acquireAudioSessionId(int audioSession);
-
- virtual void releaseAudioSessionId(int audioSession);
-
- virtual status_t queryNumberEffects(uint32_t *numEffects);
-
- virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor);
-
- virtual status_t getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor);
-
- virtual sp<IEffect> createEffect(pid_t pid,
- effect_descriptor_t *pDesc,
- const sp<IEffectClient>& effectClient,
- int32_t priority,
- int io,
- int sessionId,
- status_t *status,
- int *id,
- int *enabled);
-
- virtual status_t moveEffects(int sessionId, int srcOutput, int dstOutput);
-
- enum hardware_call_state {
- AUDIO_HW_IDLE = 0,
- AUDIO_HW_INIT,
- AUDIO_HW_OUTPUT_OPEN,
- AUDIO_HW_OUTPUT_CLOSE,
- AUDIO_HW_INPUT_OPEN,
- AUDIO_HW_INPUT_CLOSE,
- AUDIO_HW_STANDBY,
- AUDIO_HW_SET_MASTER_VOLUME,
- AUDIO_HW_GET_ROUTING,
- AUDIO_HW_SET_ROUTING,
- AUDIO_HW_GET_MODE,
- AUDIO_HW_SET_MODE,
- AUDIO_HW_GET_MIC_MUTE,
- AUDIO_HW_SET_MIC_MUTE,
- AUDIO_SET_VOICE_VOLUME,
- AUDIO_SET_PARAMETER,
- };
-
- // record interface
virtual sp<IAudioRecord> openRecord(
pid_t pid,
- int input,
+ audio_io_handle_t input,
uint32_t sampleRate,
audio_format_t format,
uint32_t channelMask,
@@ -196,22 +92,116 @@
int *sessionId,
status_t *status);
+ virtual uint32_t sampleRate(audio_io_handle_t output) const;
+ virtual int channelCount(audio_io_handle_t output) const;
+ virtual audio_format_t format(audio_io_handle_t output) const;
+ virtual size_t frameCount(audio_io_handle_t output) const;
+ virtual uint32_t latency(audio_io_handle_t output) const;
+
+ virtual status_t setMasterVolume(float value);
+ virtual status_t setMasterMute(bool muted);
+
+ virtual float masterVolume() const;
+ virtual bool masterMute() const;
+
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output);
+ virtual status_t setStreamMute(audio_stream_type_t stream, bool muted);
+
+ virtual float streamVolume(audio_stream_type_t stream,
+ audio_io_handle_t output) const;
+ virtual bool streamMute(audio_stream_type_t stream) const;
+
+ virtual status_t setMode(audio_mode_t mode);
+
+ virtual status_t setMicMute(bool state);
+ virtual bool getMicMute() const;
+
+ virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
+
+ virtual void registerClient(const sp<IAudioFlingerClient>& client);
+
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const;
+
+ virtual audio_io_handle_t openOutput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ uint32_t *pChannels,
+ uint32_t *pLatencyMs,
+ uint32_t flags);
+
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2);
+
+ virtual status_t closeOutput(audio_io_handle_t output);
+
+ virtual status_t suspendOutput(audio_io_handle_t output);
+
+ virtual status_t restoreOutput(audio_io_handle_t output);
+
+ virtual audio_io_handle_t openInput(uint32_t *pDevices,
+ uint32_t *pSamplingRate,
+ audio_format_t *pFormat,
+ uint32_t *pChannels,
+ audio_in_acoustics_t acoustics);
+
+ virtual status_t closeInput(audio_io_handle_t input);
+
+ virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output);
+
+ virtual status_t setVoiceVolume(float volume);
+
+ virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const;
+
+ virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const;
+
+ virtual int newAudioSessionId();
+
+ virtual void acquireAudioSessionId(int audioSession);
+
+ virtual void releaseAudioSessionId(int audioSession);
+
+ virtual status_t queryNumberEffects(uint32_t *numEffects) const;
+
+ virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
+
+ virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
+ effect_descriptor_t *descriptor) const;
+
+ virtual sp<IEffect> createEffect(pid_t pid,
+ effect_descriptor_t *pDesc,
+ const sp<IEffectClient>& effectClient,
+ int32_t priority,
+ audio_io_handle_t io,
+ int sessionId,
+ status_t *status,
+ int *id,
+ int *enabled);
+
+ virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput);
+
virtual status_t onTransact(
uint32_t code,
const Parcel& data,
Parcel* reply,
uint32_t flags);
- audio_mode_t getMode() const { return mMode; }
-
- bool btNrecIsOff() { return mBtNrecIsOff; }
+ // end of IAudioFlinger interface
private:
+ audio_mode_t getMode() const { return mMode; }
+
+ bool btNrecIsOff() const { return mBtNrecIsOff; }
AudioFlinger();
virtual ~AudioFlinger();
- status_t initCheck() const;
+ // call in any IAudioFlinger method that accesses mPrimaryHardwareDev
+ status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; }
+
virtual void onFirstRef();
audio_hw_device_t* findSuitableHwDev_l(uint32_t devices);
void purgeStaleEffects_l();
@@ -228,7 +218,7 @@
virtual ~Client();
sp<MemoryDealer> heap() const;
pid_t pid() const { return mPid; }
- sp<AudioFlinger> audioFlinger() { return mAudioFlinger; }
+ sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; }
private:
Client(const Client&);
@@ -285,7 +275,7 @@
RECORD // Thread class is RecordThread
};
- ThreadBase (const sp<AudioFlinger>& audioFlinger, int id, uint32_t device, type_t type);
+ ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type);
virtual ~ThreadBase();
status_t dumpBase(int fd, const Vector<String16>& args);
@@ -322,13 +312,13 @@
uint32_t flags,
const sp<IMemory>& sharedBuffer,
int sessionId);
- ~TrackBase();
+ virtual ~TrackBase();
virtual status_t start() = 0;
virtual void stop() = 0;
- sp<IMemory> getCblk() const;
+ sp<IMemory> getCblk() const { return mCblkMemory; }
audio_track_cblk_t* cblk() const { return mCblk; }
- int sessionId() { return mSessionId; }
+ int sessionId() const { return mSessionId; }
protected:
friend class ThreadBase;
@@ -348,11 +338,11 @@
return mFormat;
}
- int channelCount() const ;
+ int channelCount() const { return mChannelCount; }
- uint32_t channelMask() const;
+ uint32_t channelMask() const { return mChannelMask; }
- int sampleRate() const;
+ int sampleRate() const; // FIXME inline after cblk sr moved
void* getBuffer(uint32_t offset, uint32_t frames) const;
@@ -376,7 +366,6 @@
uint32_t mFrameCount;
// we don't really need a lock for these
track_state mState;
- int mClientTid;
const audio_format_t mFormat;
uint32_t mFlags;
const int mSessionId;
@@ -409,11 +398,13 @@
virtual status_t initCheck() const = 0;
type_t type() const { return mType; }
- uint32_t sampleRate() const;
- int channelCount() const;
- audio_format_t format() const;
- size_t frameCount() const;
+ uint32_t sampleRate() const { return mSampleRate; }
+ int channelCount() const { return mChannelCount; }
+ audio_format_t format() const { return mFormat; }
+ size_t frameCount() const { return mFrameCount; }
void wakeUp() { mWaitWorkCV.broadcast(); }
+ // Should be "virtual status_t requestExitAndWait()" and override same
+ // method in Thread, but Thread::requestExitAndWait() is not yet virtual.
void exit();
virtual bool checkForNewParameters_l() = 0;
virtual status_t setParameters(const String8& keyValuePairs);
@@ -422,7 +413,7 @@
void sendConfigEvent(int event, int param = 0);
void sendConfigEvent_l(int event, int param = 0);
void processConfigEvents();
- int id() const { return mId;}
+ audio_io_handle_t id() const { return mId;}
bool standby() { return mStandby; }
uint32_t device() { return mDevice; }
virtual audio_stream_t* stream() = 0;
@@ -437,7 +428,7 @@
status_t *status);
void disconnectEffect(const sp< EffectModule>& effect,
const wp<EffectHandle>& handle,
- bool unpiniflast);
+ bool unpinIfLast);
// return values for hasAudioSession (bit field)
enum effect_state {
@@ -544,8 +535,7 @@
status_t mParamStatus;
Vector<ConfigEvent> mConfigEvents;
bool mStandby;
- int mId;
- bool mExiting;
+ const audio_io_handle_t mId;
Vector< sp<EffectChain> > mEffectChains;
uint32_t mDevice; // output device for PlaybackThread
// input + output devices for RecordThread
@@ -559,6 +549,18 @@
KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions;
};
+ struct stream_type_t {
+ stream_type_t()
+ : volume(1.0f),
+ mute(false),
+ valid(true)
+ {
+ }
+ float volume;
+ bool mute;
+ bool valid;
+ };
+
// --- PlaybackThread ---
class PlaybackThread : public ThreadBase {
public:
@@ -581,7 +583,7 @@
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
- ~Track();
+ virtual ~Track();
void dump(char* buffer, size_t size);
virtual status_t start();
@@ -595,15 +597,15 @@
return mName;
}
- audio_stream_type_t type() const {
+ audio_stream_type_t streamType() const {
return mStreamType;
}
status_t attachAuxEffect(int EffectId);
void setAuxBuffer(int EffectId, int32_t *buffer);
- int32_t *auxBuffer() { return mAuxBuffer; }
+ int32_t *auxBuffer() const { return mAuxBuffer; }
void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; }
- int16_t *mainBuffer() { return mMainBuffer; }
- int auxEffectId() { return mAuxEffectId; }
+ int16_t *mainBuffer() const { return mMainBuffer; }
+ int auxEffectId() const { return mAuxEffectId; }
protected:
@@ -617,7 +619,7 @@
Track& operator = (const Track&);
virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
- bool isMuted() { return mMute; }
+ bool isMuted() const { return mMute; }
bool isPausing() const {
return mState == PAUSING;
}
@@ -664,14 +666,14 @@
audio_format_t format,
uint32_t channelMask,
int frameCount);
- ~OutputTrack();
+ virtual ~OutputTrack();
virtual status_t start();
virtual void stop();
bool write(int16_t* data, uint32_t frames);
- bool bufferQueueEmpty() { return (mBufferQueue.size() == 0) ? true : false; }
- bool isActive() { return mActive; }
- const wp<ThreadBase>& thread() { return mThread; }
+ bool bufferQueueEmpty() const { return (mBufferQueue.size() == 0) ? true : false; }
+ bool isActive() const { return mActive; }
+ const wp<ThreadBase>& thread() const { return mThread; }
private:
@@ -691,8 +693,8 @@
DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
}; // end of OutputTrack
- PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id,
- uint32_t device, type_t type);
+ PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ audio_io_handle_t id, uint32_t device, type_t type);
virtual ~PlaybackThread();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -708,8 +710,8 @@
virtual status_t setMasterVolume(float value);
virtual status_t setMasterMute(bool muted);
- virtual float masterVolume() const;
- virtual bool masterMute() const;
+ virtual float masterVolume() const { return mMasterVolume; }
+ virtual bool masterMute() const { return mMasterMute; }
virtual status_t setStreamVolume(audio_stream_type_t stream, float value);
virtual status_t setStreamMute(audio_stream_type_t stream, bool muted);
@@ -738,7 +740,7 @@
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
- int16_t *mixBuffer() { return mMixBuffer; };
+ int16_t *mixBuffer() const { return mMixBuffer; };
virtual void detachAuxEffect_l(int effectId);
status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
@@ -753,18 +755,6 @@
void setStreamValid(audio_stream_type_t streamType, bool valid);
- struct stream_type_t {
- stream_type_t()
- : volume(1.0f),
- mute(false),
- valid(true)
- {
- }
- float volume;
- bool mute;
- bool valid;
- };
-
protected:
int16_t* mMixBuffer;
int mSuspended;
@@ -817,7 +807,7 @@
public:
MixerThread (const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
- int id,
+ audio_io_handle_t id,
uint32_t device,
type_t type = MIXER);
virtual ~MixerThread();
@@ -844,8 +834,9 @@
class DirectOutputThread : public PlaybackThread {
public:
- DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device);
- ~DirectOutputThread();
+ DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ audio_io_handle_t id, uint32_t device);
+ virtual ~DirectOutputThread();
// Thread virtuals
virtual bool threadLoop();
@@ -870,8 +861,9 @@
class DuplicatingThread : public MixerThread {
public:
- DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, int id);
- ~DuplicatingThread();
+ DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
+ audio_io_handle_t id);
+ virtual ~DuplicatingThread();
// Thread virtuals
virtual bool threadLoop();
@@ -889,13 +881,16 @@
uint32_t mWaitTimeMs;
};
- PlaybackThread *checkPlaybackThread_l(int output) const;
- MixerThread *checkMixerThread_l(int output) const;
- RecordThread *checkRecordThread_l(int input) const;
- float streamVolumeInternal(audio_stream_type_t stream) const { return mStreamTypes[stream].volume; }
- void audioConfigChanged_l(int event, int ioHandle, void *param2);
+ PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
+ MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
+ RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
+ float streamVolumeInternal(audio_stream_type_t stream) const
+ { return mStreamTypes[stream].volume; }
+ void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2);
+ // allocate an audio_io_handle_t, session ID, or effect ID
uint32_t nextUniqueId();
+
status_t moveEffectChain_l(int sessionId,
AudioFlinger::PlaybackThread *srcThread,
AudioFlinger::PlaybackThread *dstThread,
@@ -946,7 +941,7 @@
int frameCount,
uint32_t flags,
int sessionId);
- ~RecordTrack();
+ virtual ~RecordTrack();
virtual status_t start();
virtual void stop();
@@ -973,9 +968,9 @@
AudioStreamIn *input,
uint32_t sampleRate,
uint32_t channels,
- int id,
+ audio_io_handle_t id,
uint32_t device);
- ~RecordThread();
+ virtual ~RecordThread();
virtual bool threadLoop();
virtual status_t readyToRun();
@@ -1064,7 +1059,7 @@
effect_descriptor_t *desc,
int id,
int sessionId);
- ~EffectModule();
+ virtual ~EffectModule();
enum effect_state {
IDLE,
@@ -1076,7 +1071,7 @@
DESTROYED
};
- int id() { return mId; }
+ int id() const { return mId; }
void process();
void updateState();
status_t command(uint32_t cmdCode,
@@ -1094,12 +1089,12 @@
uint32_t status() {
return mStatus;
}
- int sessionId() {
+ int sessionId() const {
return mSessionId;
}
status_t setEnabled(bool enabled);
- bool isEnabled();
- bool isProcessEnabled();
+ bool isEnabled() const;
+ bool isProcessEnabled() const;
void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; }
int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; }
@@ -1110,7 +1105,7 @@
const wp<ThreadBase>& thread() { return mThread; }
status_t addHandle(const sp<EffectHandle>& handle);
- void disconnect(const wp<EffectHandle>& handle, bool unpiniflast);
+ void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast);
size_t removeHandle (const wp<EffectHandle>& handle);
effect_descriptor_t& desc() { return mDescriptor; }
@@ -1126,7 +1121,7 @@
sp<EffectHandle> controlHandle();
- bool isPinned() { return mPinned; }
+ bool isPinned() const { return mPinned; }
void unPin() { mPinned = false; }
status_t dump(int fd, const Vector<String16>& args);
@@ -1186,8 +1181,10 @@
uint32_t *replySize,
void *pReplyData);
virtual void disconnect();
- virtual void disconnect(bool unpiniflast);
- virtual sp<IMemory> getCblk() const;
+ private:
+ void disconnect(bool unpinIfLast);
+ public:
+ virtual sp<IMemory> getCblk() const { return mCblkMemory; }
virtual status_t onTransact(uint32_t code, const Parcel& data,
Parcel* reply, uint32_t flags);
@@ -1203,13 +1200,13 @@
uint32_t replySize,
void *pReplyData);
void setEnabled(bool enabled);
- bool enabled() { return mEnabled; }
+ bool enabled() const { return mEnabled; }
// Getters
- int id() { return mEffect->id(); }
- int priority() { return mPriority; }
- bool hasControl() { return mHasControl; }
- sp<EffectModule> effect() { return mEffect; }
+ int id() const { return mEffect->id(); }
+ int priority() const { return mPriority; }
+ bool hasControl() const { return mHasControl; }
+ sp<EffectModule> effect() const { return mEffect; }
void dump(char* buffer, size_t size);
@@ -1221,7 +1218,7 @@
sp<EffectModule> mEffect; // pointer to controlled EffectModule
sp<IEffectClient> mEffectClient; // callback interface for client notifications
- sp<Client> mClient; // client for shared memory allocation
+ /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect()
sp<IMemory> mCblkMemory; // shared memory for control block
effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory
uint8_t* mBuffer; // pointer to parameter area in shared memory
@@ -1241,7 +1238,7 @@
class EffectChain: public RefBase {
public:
EffectChain(const wp<ThreadBase>& wThread, int sessionId);
- ~EffectChain();
+ virtual ~EffectChain();
// special key used for an entry in mSuspendedEffects keyed vector
// corresponding to a suspend all request.
@@ -1263,7 +1260,7 @@
status_t addEffect_l(const sp<EffectModule>& handle);
size_t removeEffect_l(const sp<EffectModule>& handle);
- int sessionId() { return mSessionId; }
+ int sessionId() const { return mSessionId; }
void setSessionId(int sessionId) { mSessionId = sessionId; }
sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor);
@@ -1277,26 +1274,26 @@
mInBuffer = buffer;
mOwnInBuffer = ownsBuffer;
}
- int16_t *inBuffer() {
+ int16_t *inBuffer() const {
return mInBuffer;
}
void setOutBuffer(int16_t *buffer) {
mOutBuffer = buffer;
}
- int16_t *outBuffer() {
+ int16_t *outBuffer() const {
return mOutBuffer;
}
void incTrackCnt() { android_atomic_inc(&mTrackCnt); }
void decTrackCnt() { android_atomic_dec(&mTrackCnt); }
- int32_t trackCnt() { return mTrackCnt;}
+ int32_t trackCnt() const { return mTrackCnt;}
void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt);
mTailBufferCount = mMaxTailBuffers; }
void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); }
- int32_t activeTrackCnt() { return mActiveTrackCnt;}
+ int32_t activeTrackCnt() const { return mActiveTrackCnt;}
- uint32_t strategy() { return mStrategy; }
+ uint32_t strategy() const { return mStrategy; }
void setStrategy(uint32_t strategy)
{ mStrategy = strategy; }
@@ -1393,22 +1390,45 @@
mutable Mutex mLock;
- DefaultKeyedVector< pid_t, wp<Client> > mClients;
+ DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client()
mutable Mutex mHardwareLock;
- audio_hw_device_t* mPrimaryHardwareDev;
+
+ // These two fields are immutable after onFirstRef(), so no lock needed to access
+ audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL
Vector<audio_hw_device_t*> mAudioHwDevs;
+
+ enum hardware_call_state {
+ AUDIO_HW_IDLE = 0,
+ AUDIO_HW_INIT,
+ AUDIO_HW_OUTPUT_OPEN,
+ AUDIO_HW_OUTPUT_CLOSE,
+ AUDIO_HW_INPUT_OPEN,
+ AUDIO_HW_INPUT_CLOSE,
+ AUDIO_HW_STANDBY,
+ AUDIO_HW_SET_MASTER_VOLUME,
+ AUDIO_HW_GET_ROUTING,
+ AUDIO_HW_SET_ROUTING,
+ AUDIO_HW_GET_MODE,
+ AUDIO_HW_SET_MODE,
+ AUDIO_HW_GET_MIC_MUTE,
+ AUDIO_HW_SET_MIC_MUTE,
+ AUDIO_SET_VOICE_VOLUME,
+ AUDIO_SET_PARAMETER,
+ AUDIO_HW_GET_INPUT_BUFFER_SIZE,
+ };
+
mutable hardware_call_state mHardwareStatus; // for dump only
- DefaultKeyedVector< int, sp<PlaybackThread> > mPlaybackThreads;
- PlaybackThread::stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
+ DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads;
+ stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
// both are protected by mLock
float mMasterVolume;
bool mMasterMute;
- DefaultKeyedVector< int, sp<RecordThread> > mRecordThreads;
+ DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads;
DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients;
volatile int32_t mNextUniqueId;
@@ -1419,6 +1439,10 @@
float masterVolume_l() const { return mMasterVolume; }
bool masterMute_l() const { return mMasterMute; }
+
+private:
+ sp<Client> registerPid_l(pid_t pid); // always returns non-0
+
};
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 0b9f8ba..191520a 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -68,7 +68,7 @@
// t->prevAuxLevel
// t->frameCount
t->channelCount = 2;
- t->enabled = 0;
+ t->enabled = false;
t->format = 16;
t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
t->bufferProvider = NULL;
@@ -121,8 +121,8 @@
assert(uint32_t(name) < MAX_NUM_TRACKS);
ALOGV("deleteTrackName(%d)", name);
track_t& track(mState.tracks[ name ]);
- if (track.enabled != 0) {
- track.enabled = 0;
+ if (track.enabled) {
+ track.enabled = false;
invalidateState(1<<name);
}
if (track.resampler != NULL) {
@@ -143,8 +143,8 @@
assert(uint32_t(name) < MAX_NUM_TRACKS);
track_t& track = mState.tracks[name];
- if (track.enabled != 1) {
- track.enabled = 1;
+ if (!track.enabled) {
+ track.enabled = true;
ALOGV("enable(%d)", name);
invalidateState(1 << name);
}
@@ -156,8 +156,8 @@
assert(uint32_t(name) < MAX_NUM_TRACKS);
track_t& track = mState.tracks[name];
- if (track.enabled != 0) {
- track.enabled = 0;
+ if (track.enabled) {
+ track.enabled = false;
ALOGV("disable(%d)", name);
invalidateState(1 << name);
}
@@ -296,18 +296,6 @@
return false;
}
-bool AudioMixer::track_t::doesResample() const
-{
- return resampler != NULL;
-}
-
-void AudioMixer::track_t::resetResampler()
-{
- if (resampler != NULL) {
- resampler->reset();
- }
-}
-
inline
void AudioMixer::track_t::adjustVolumeRamp(bool aux)
{
@@ -327,20 +315,11 @@
}
}
-size_t AudioMixer::track_t::getUnreleasedFrames()
-{
- if (resampler != NULL) {
- return resampler->getUnreleasedFrames();
- }
- return 0;
-}
-
-size_t AudioMixer::getUnreleasedFrames(int name)
+size_t AudioMixer::getUnreleasedFrames(int name) const
{
name -= TRACK0;
if (uint32_t(name) < MAX_NUM_TRACKS) {
- track_t& track(mState.tracks[name]);
- return track.getUnreleasedFrames();
+ return mState.tracks[name].getUnreleasedFrames();
}
return 0;
}
@@ -383,9 +362,9 @@
// compute everything we need...
int countActiveTracks = 0;
- int all16BitsStereoNoResample = 1;
- int resampling = 0;
- int volumeRamp = 0;
+ bool all16BitsStereoNoResample = true;
+ bool resampling = false;
+ bool volumeRamp = false;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
@@ -402,7 +381,7 @@
}
if (t.volumeInc[0]|t.volumeInc[1]) {
- volumeRamp = 1;
+ volumeRamp = true;
} else if (!t.doesResample() && t.volumeRL == 0) {
n |= NEEDS_MUTE_ENABLED;
}
@@ -412,16 +391,16 @@
t.hook = track__nop;
} else {
if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
- all16BitsStereoNoResample = 0;
+ all16BitsStereoNoResample = false;
}
if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
- all16BitsStereoNoResample = 0;
- resampling = 1;
+ all16BitsStereoNoResample = false;
+ resampling = true;
t.hook = track__genericResample;
} else {
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
t.hook = track__16BitsMono;
- all16BitsStereoNoResample = 0;
+ all16BitsStereoNoResample = false;
}
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_2){
t.hook = track__16BitsStereo;
@@ -469,7 +448,7 @@
// Now that the volume ramp has been done, set optimal state and
// track hooks for subsequent mixer process
if (countActiveTracks) {
- int allMuted = 1;
+ bool allMuted = true;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
@@ -480,7 +459,7 @@
t.needs |= NEEDS_MUTE_ENABLED;
t.hook = track__nop;
} else {
- allMuted = 0;
+ allMuted = false;
}
}
if (allMuted) {
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 84f6330..c709686 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -33,7 +33,7 @@
public:
AudioMixer(size_t frameCount, uint32_t sampleRate);
- ~AudioMixer();
+ /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed
static const uint32_t MAX_NUM_TRACKS = 32;
static const uint32_t MAX_NUM_CHANNELS = 2;
@@ -83,7 +83,7 @@
uint32_t trackNames() const { return mTrackNames; }
- size_t getUnreleasedFrames(int name);
+ size_t getUnreleasedFrames(int name) const;
private:
@@ -153,10 +153,11 @@
int32_t* auxBuffer;
bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
- bool doesResample() const;
- void resetResampler();
+ bool doesResample() const { return resampler != NULL; }
+ void resetResampler() { if (resampler != NULL) resampler->reset(); }
void adjustVolumeRamp(bool aux);
- size_t getUnreleasedFrames();
+ size_t getUnreleasedFrames() const { return resampler != NULL ?
+ resampler->getUnreleasedFrames() : 0; };
};
// pad to 32-bytes to fill cache line
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 1dddbb3..21b5811 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -30,6 +30,7 @@
#include <utils/String16.h>
#include <utils/threads.h>
#include "AudioPolicyService.h"
+#include "ServiceUtilities.h"
#include <cutils/properties.h>
#include <hardware_legacy/power.h>
#include <media/AudioEffect.h>
@@ -49,13 +50,6 @@
static const int kDumpLockRetries = 50;
static const int kDumpLockSleepUs = 20000;
-static bool checkPermission() {
- if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
- bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
- if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
- return ok;
-}
-
namespace {
extern struct audio_policy_service_ops aps_ops;
};
@@ -157,7 +151,7 @@
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (!audio_is_output_device(device) && !audio_is_input_device(device)) {
@@ -190,7 +184,7 @@
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(state) >= AUDIO_MODE_CNT) {
@@ -213,7 +207,7 @@
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) {
@@ -388,7 +382,7 @@
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
@@ -405,7 +399,7 @@
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
@@ -542,7 +536,7 @@
}
void AudioPolicyService::binderDied(const wp<IBinder>& who) {
- ALOGW("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(),
+ ALOGW("binderDied() %p, tid %d, calling pid %d", who.unsafe_get(), gettid(),
IPCThreadState::self()->getCallingPid());
}
@@ -578,7 +572,7 @@
status_t AudioPolicyService::dump(int fd, const Vector<String16>& args)
{
- if (!checkCallingPermission(String16("android.permission.DUMP"))) {
+ if (!dumpAllowed()) {
dumpPermissionDenial(fd);
} else {
bool locked = tryLock(mLock);
@@ -588,10 +582,10 @@
}
dumpInternals(fd);
- if (mAudioCommandThread != NULL) {
+ if (mAudioCommandThread != 0) {
mAudioCommandThread->dump(fd);
}
- if (mTonePlaybackThread != NULL) {
+ if (mTonePlaybackThread != 0) {
mTonePlaybackThread->dump(fd);
}
@@ -819,7 +813,7 @@
status_t AudioPolicyService::AudioCommandThread::volumeCommand(audio_stream_type_t stream,
float volume,
- int output,
+ audio_io_handle_t output,
int delayMs)
{
status_t status = NO_ERROR;
@@ -849,7 +843,7 @@
return status;
}
-status_t AudioPolicyService::AudioCommandThread::parametersCommand(int ioHandle,
+status_t AudioPolicyService::AudioCommandThread::parametersCommand(audio_io_handle_t ioHandle,
const char *keyValuePairs,
int delayMs)
{
@@ -1019,7 +1013,7 @@
const char *keyValuePairs,
int delayMs)
{
- mAudioCommandThread->parametersCommand((int)ioHandle, keyValuePairs,
+ mAudioCommandThread->parametersCommand(ioHandle, keyValuePairs,
delayMs);
}
@@ -1029,7 +1023,7 @@
int delayMs)
{
return (int)mAudioCommandThread->volumeCommand(stream, volume,
- (int)output, delayMs);
+ output, delayMs);
}
int AudioPolicyService::startTone(audio_policy_tone_t tone,
@@ -1362,7 +1356,7 @@
audio_policy_output_flags_t flags)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return 0;
}
@@ -1376,7 +1370,7 @@
audio_io_handle_t output2)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return 0;
}
@@ -1386,7 +1380,7 @@
static int aps_close_output(void *service, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
return af->closeOutput(output);
@@ -1395,7 +1389,7 @@
static int aps_suspend_output(void *service, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return PERMISSION_DENIED;
}
@@ -1406,7 +1400,7 @@
static int aps_restore_output(void *service, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return PERMISSION_DENIED;
}
@@ -1422,7 +1416,7 @@
audio_in_acoustics_t acoustics)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return 0;
}
@@ -1434,7 +1428,7 @@
static int aps_close_input(void *service, audio_io_handle_t input)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
return af->closeInput(input);
@@ -1444,7 +1438,7 @@
audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
return af->setStreamOutput(stream, output);
@@ -1455,10 +1449,10 @@
audio_io_handle_t dst_output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
- return af->moveEffects(session, (int)src_output, (int)dst_output);
+ return af->moveEffects(session, src_output, dst_output);
}
static char * aps_get_parameters(void *service, audio_io_handle_t io_handle,
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index 62219e5..fdaf576 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -174,8 +174,10 @@
void startToneCommand(ToneGenerator::tone_type type,
audio_stream_type_t stream);
void stopToneCommand();
- status_t volumeCommand(audio_stream_type_t stream, float volume, int output, int delayMs = 0);
- status_t parametersCommand(int ioHandle, const char *keyValuePairs, int delayMs = 0);
+ status_t volumeCommand(audio_stream_type_t stream, float volume,
+ audio_io_handle_t output, int delayMs = 0);
+ status_t parametersCommand(audio_io_handle_t ioHandle,
+ const char *keyValuePairs, int delayMs = 0);
status_t voiceVolumeCommand(float volume, int delayMs = 0);
void insertCommand_l(AudioCommand *command, int delayMs = 0);
@@ -207,12 +209,12 @@
public:
audio_stream_type_t mStream;
float mVolume;
- int mIO;
+ audio_io_handle_t mIO;
};
class ParametersData {
public:
- int mIO;
+ audio_io_handle_t mIO;
String8 mKeyValuePairs;
};
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index feacd96..9486b9c 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -23,8 +23,10 @@
#include <cutils/log.h>
#include <cutils/properties.h>
#include "AudioResampler.h"
+#if 0
#include "AudioResamplerSinc.h"
#include "AudioResamplerCubic.h"
+#endif
#ifdef __arm__
#include <machine/cpu-features.h>
@@ -99,6 +101,7 @@
ALOGV("Create linear Resampler");
resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
break;
+#if 0
case MED_QUALITY:
ALOGV("Create cubic Resampler");
resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
@@ -107,6 +110,7 @@
ALOGV("Create sinc Resampler");
resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
break;
+#endif
}
// initialize resampler
@@ -130,12 +134,6 @@
mVolume[0] = mVolume[1] = 0;
mBuffer.frameCount = 0;
- // save format for quick lookup
- if (inChannelCount == 1) {
- mFormat = MONO_16_BIT;
- } else {
- mFormat = STEREO_16_BIT;
- }
}
AudioResampler::~AudioResampler() {
@@ -190,7 +188,7 @@
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+ // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
while (outputIndex < outputSampleCount) {
@@ -203,7 +201,7 @@
goto resampleStereo16_exit;
}
- // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+ // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
if (mBuffer.frameCount > inputIndex) break;
inputIndex -= mBuffer.frameCount;
@@ -217,7 +215,7 @@
// handle boundary case
while (inputIndex == 0) {
- // ALOGE("boundary case\n");
+ // ALOGE("boundary case");
out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
Advance(&inputIndex, &phaseFraction, phaseIncrement);
@@ -226,7 +224,7 @@
}
// process input samples
- // ALOGE("general case\n");
+ // ALOGE("general case");
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
if (inputIndex + 2 < mBuffer.frameCount) {
@@ -248,7 +246,7 @@
Advance(&inputIndex, &phaseFraction, phaseIncrement);
}
- // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
@@ -265,7 +263,7 @@
}
}
- // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
resampleStereo16_exit:
// save state
@@ -286,7 +284,7 @@
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+ // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one
@@ -298,7 +296,7 @@
mPhaseFraction = phaseFraction;
goto resampleMono16_exit;
}
- // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+ // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
if (mBuffer.frameCount > inputIndex) break;
inputIndex -= mBuffer.frameCount;
@@ -310,7 +308,7 @@
// handle boundary case
while (inputIndex == 0) {
- // ALOGE("boundary case\n");
+ // ALOGE("boundary case");
int32_t sample = Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vl * sample;
out[outputIndex++] += vr * sample;
@@ -320,7 +318,7 @@
}
// process input samples
- // ALOGE("general case\n");
+ // ALOGE("general case");
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
if (inputIndex + 2 < mBuffer.frameCount) {
@@ -343,7 +341,7 @@
}
- // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
@@ -359,7 +357,7 @@
}
}
- // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
resampleMono16_exit:
// save state
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index ffa690a..c23016e 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -54,7 +54,7 @@
AudioBufferProvider* provider) = 0;
virtual void reset();
- virtual size_t getUnreleasedFrames() { return mInputIndex; }
+ virtual size_t getUnreleasedFrames() const { return mInputIndex; }
protected:
// number of bits for phase fraction - 30 bits allows nearly 2x downsampling
@@ -66,16 +66,15 @@
// multiplier to calculate fixed point phase increment
static const double kPhaseMultiplier = 1L << kNumPhaseBits;
- enum format {MONO_16_BIT, STEREO_16_BIT};
AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
// prevent copying
AudioResampler(const AudioResampler&);
AudioResampler& operator=(const AudioResampler&);
- int32_t mBitDepth;
- int32_t mChannelCount;
- int32_t mSampleRate;
+ const int32_t mBitDepth;
+ const int32_t mChannelCount;
+ const int32_t mSampleRate;
int32_t mInSampleRate;
AudioBufferProvider::Buffer mBuffer;
union {
@@ -83,7 +82,6 @@
uint32_t mVolumeRL;
};
int16_t mTargetVolume[2];
- format mFormat;
size_t mInputIndex;
int32_t mPhaseIncrement;
uint32_t mPhaseFraction;
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 47205ba..c0e760e 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -99,7 +99,7 @@
if (mBuffer.raw == NULL)
goto save_state; // ugly, but efficient
in = mBuffer.i16;
- // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+ // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
// advance sample state
@@ -133,7 +133,7 @@
provider->getNextBuffer(&mBuffer);
if (mBuffer.raw == NULL)
return;
- // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+ // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
int16_t *in = mBuffer.i16;
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index d012433..7a27b81 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -199,33 +199,32 @@
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- AudioBufferProvider::Buffer& buffer(mBuffer);
while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one
- while (buffer.frameCount == 0) {
- buffer.frameCount = inFrameCount;
- provider->getNextBuffer(&buffer);
- if (buffer.raw == NULL) {
+ while (mBuffer.frameCount == 0) {
+ mBuffer.frameCount = inFrameCount;
+ provider->getNextBuffer(&mBuffer);
+ if (mBuffer.raw == NULL) {
goto resample_exit;
}
const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
if (phaseIndex == 1) {
// read one frame
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
} else if (phaseIndex == 2) {
// read 2 frames
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
inputIndex++;
if (inputIndex >= mBuffer.frameCount) {
inputIndex -= mBuffer.frameCount;
- provider->releaseBuffer(&buffer);
+ provider->releaseBuffer(&mBuffer);
} else {
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
}
}
}
- int16_t *in = buffer.i16;
- const size_t frameCount = buffer.frameCount;
+ int16_t *in = mBuffer.i16;
+ const size_t frameCount = mBuffer.frameCount;
// Always read-in the first samples from the input buffer
int16_t* head = impulse + halfNumCoefs*CHANNELS;
@@ -264,7 +263,7 @@
// if done with buffer, save samples
if (inputIndex >= frameCount) {
inputIndex -= frameCount;
- provider->releaseBuffer(&buffer);
+ provider->releaseBuffer(&mBuffer);
}
}
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index 0e1bc44..f0a07b8 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -31,7 +31,7 @@
public:
AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate);
- ~AudioResamplerSinc();
+ virtual ~AudioResamplerSinc();
virtual void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
diff --git a/services/audioflinger/ServiceUtilities.cpp b/services/audioflinger/ServiceUtilities.cpp
new file mode 100644
index 0000000..6a58852
--- /dev/null
+++ b/services/audioflinger/ServiceUtilities.cpp
@@ -0,0 +1,55 @@
+/*
+ * Copyright (C) 2012 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.
+ */
+
+#include <binder/IPCThreadState.h>
+#include <binder/IServiceManager.h>
+#include <binder/PermissionCache.h>
+#include "ServiceUtilities.h"
+
+namespace android {
+
+// This optimization assumes mediaserver process doesn't fork, which it doesn't
+const pid_t getpid_cached = getpid();
+
+bool recordingAllowed() {
+ if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
+ static const String16 sRecordAudio("android.permission.RECORD_AUDIO");
+ // don't use PermissionCache; this is not a system permission
+ bool ok = checkCallingPermission(sRecordAudio);
+ if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO");
+ return ok;
+}
+
+bool settingsAllowed() {
+ if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
+ static const String16 sAudioSettings("android.permission.MODIFY_AUDIO_SETTINGS");
+ // don't use PermissionCache; this is not a system permission
+ bool ok = checkCallingPermission(sAudioSettings);
+ if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
+ return ok;
+}
+
+bool dumpAllowed() {
+ // don't optimize for same pid, since mediaserver never dumps itself
+ static const String16 sDump("android.permission.DUMP");
+ // OK to use PermissionCache; this is a system permission
+ bool ok = PermissionCache::checkCallingPermission(sDump);
+ // convention is for caller to dump an error message to fd instead of logging here
+ //if (!ok) ALOGE("Request requires android.permission.DUMP");
+ return ok;
+}
+
+} // namespace android
diff --git a/services/audioflinger/ServiceUtilities.h b/services/audioflinger/ServiceUtilities.h
new file mode 100644
index 0000000..f77ec5b
--- /dev/null
+++ b/services/audioflinger/ServiceUtilities.h
@@ -0,0 +1,27 @@
+/*
+ * Copyright (C) 2012 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.
+ */
+
+#include <unistd.h>
+
+namespace android {
+
+extern const pid_t getpid_cached;
+
+bool recordingAllowed();
+bool settingsAllowed();
+bool dumpAllowed();
+
+}
