| |
| /* ----------------------------------------------------------------------------------------------------------- |
| Software License for The Fraunhofer FDK AAC Codec Library for Android |
| |
| © Copyright 1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. |
| All rights reserved. |
| |
| 1. INTRODUCTION |
| The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements |
| the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. |
| This FDK AAC Codec software is intended to be used on a wide variety of Android devices. |
| |
| AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual |
| audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by |
| independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part |
| of the MPEG specifications. |
| |
| Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) |
| may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners |
| individually for the purpose of encoding or decoding bit streams in products that are compliant with |
| the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license |
| these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec |
| software may already be covered under those patent licenses when it is used for those licensed purposes only. |
| |
| Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, |
| are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional |
| applications information and documentation. |
| |
| 2. COPYRIGHT LICENSE |
| |
| Redistribution and use in source and binary forms, with or without modification, are permitted without |
| payment of copyright license fees provided that you satisfy the following conditions: |
| |
| You must retain the complete text of this software license in redistributions of the FDK AAC Codec or |
| your modifications thereto in source code form. |
| |
| You must retain the complete text of this software license in the documentation and/or other materials |
| provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. |
| You must make available free of charge copies of the complete source code of the FDK AAC Codec and your |
| modifications thereto to recipients of copies in binary form. |
| |
| The name of Fraunhofer may not be used to endorse or promote products derived from this library without |
| prior written permission. |
| |
| You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec |
| software or your modifications thereto. |
| |
| Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software |
| and the date of any change. For modified versions of the FDK AAC Codec, the term |
| "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term |
| "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." |
| |
| 3. NO PATENT LICENSE |
| |
| NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, |
| ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with |
| respect to this software. |
| |
| You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized |
| by appropriate patent licenses. |
| |
| 4. DISCLAIMER |
| |
| This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors |
| "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties |
| of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR |
| CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, |
| including but not limited to procurement of substitute goods or services; loss of use, data, or profits, |
| or business interruption, however caused and on any theory of liability, whether in contract, strict |
| liability, or tort (including negligence), arising in any way out of the use of this software, even if |
| advised of the possibility of such damage. |
| |
| 5. CONTACT INFORMATION |
| |
| Fraunhofer Institute for Integrated Circuits IIS |
| Attention: Audio and Multimedia Departments - FDK AAC LL |
| Am Wolfsmantel 33 |
| 91058 Erlangen, Germany |
| |
| www.iis.fraunhofer.de/amm |
| amm-info@iis.fraunhofer.de |
| ----------------------------------------------------------------------------------------------------------- */ |
| |
| /***************************** MPEG Audio Encoder *************************** |
| |
| Initial Authors: Markus Multrus |
| Contents/Description: PS Wrapper, Downmix header file |
| |
| ******************************************************************************/ |
| |
| #ifndef __INCLUDED_PS_MAIN_H |
| #define __INCLUDED_PS_MAIN_H |
| |
| /* Includes ******************************************************************/ |
| #include "sbr_def.h" |
| #include "qmf.h" |
| #include "ps_encode.h" |
| #include "FDK_bitstream.h" |
| #include "FDK_hybrid.h" |
| |
| |
| /* Data Types ****************************************************************/ |
| typedef enum { |
| PSENC_STEREO_BANDS_INVALID = 0, |
| PSENC_STEREO_BANDS_10 = 10, |
| PSENC_STEREO_BANDS_20 = 20 |
| |
| } PSENC_STEREO_BANDS_CONFIG; |
| |
| typedef enum { |
| PSENC_NENV_1 = 1, |
| PSENC_NENV_2 = 2, |
| PSENC_NENV_4 = 4, |
| PSENC_NENV_DEFAULT = PSENC_NENV_2, |
| PSENC_NENV_MAX = PSENC_NENV_4 |
| |
| } PSENC_NENV_CONFIG; |
| |
| typedef struct { |
| UINT bitrateFrom; /* inclusive */ |
| UINT bitrateTo; /* exclusive */ |
| PSENC_STEREO_BANDS_CONFIG nStereoBands; |
| PSENC_NENV_CONFIG nEnvelopes; |
| LONG iidQuantErrorThreshold; /* quantization threshold to switch between coarse and fine iid quantization */ |
| |
| } psTuningTable_t; |
| |
| /* Function / Class Declarations *********************************************/ |
| |
| typedef struct T_PARAMETRIC_STEREO { |
| HANDLE_PS_ENCODE hPsEncode; |
| PS_OUT psOut[2]; |
| |
| FIXP_DBL __staticHybridData[HYBRID_READ_OFFSET][MAX_PS_CHANNELS][2][MAX_HYBRID_BANDS]; |
| FIXP_DBL *pHybridData[HYBRID_READ_OFFSET+HYBRID_FRAMESIZE][MAX_PS_CHANNELS][2]; |
| |
| FIXP_QMF qmfDelayLines[2][QMF_MAX_TIME_SLOTS>>1][QMF_CHANNELS]; |
| int qmfDelayScale; |
| |
| INT psDelay; |
| UINT maxEnvelopes; |
| UCHAR dynBandScale[PS_MAX_BANDS]; |
| FIXP_DBL maxBandValue[PS_MAX_BANDS]; |
| SCHAR dmxScale; |
| INT initPS; |
| INT noQmfSlots; |
| INT noQmfBands; |
| |
| FIXP_DBL __staticHybAnaStatesLF[MAX_PS_CHANNELS][2*HYBRID_FILTER_LENGTH*HYBRID_MAX_QMF_BANDS]; |
| FIXP_DBL __staticHybAnaStatesHF[MAX_PS_CHANNELS][2*HYBRID_FILTER_DELAY*(QMF_CHANNELS-HYBRID_MAX_QMF_BANDS)]; |
| FDK_ANA_HYB_FILTER fdkHybAnaFilter[MAX_PS_CHANNELS]; |
| FDK_SYN_HYB_FILTER fdkHybSynFilter; |
| |
| } PARAMETRIC_STEREO; |
| |
| |
| typedef struct T_PSENC_CONFIG { |
| INT frameSize; |
| INT qmfFilterMode; |
| INT sbrPsDelay; |
| PSENC_STEREO_BANDS_CONFIG nStereoBands; |
| PSENC_NENV_CONFIG maxEnvelopes; |
| FIXP_DBL iidQuantErrorThreshold; |
| |
| } PSENC_CONFIG, *HANDLE_PSENC_CONFIG; |
| |
| typedef struct T_PARAMETRIC_STEREO *HANDLE_PARAMETRIC_STEREO; |
| |
| |
| /** |
| * \brief Create a parametric stereo encoder instance. |
| * |
| * \param phParametricStereo A pointer to a parametric stereo handle to be allocated. Initialized on return. |
| * |
| * \return |
| * - PSENC_OK, on succes. |
| * - PSENC_INVALID_HANDLE, PSENC_MEMORY_ERROR, on failure. |
| */ |
| FDK_PSENC_ERROR PSEnc_Create( |
| HANDLE_PARAMETRIC_STEREO *phParametricStereo |
| ); |
| |
| |
| /** |
| * \brief Initialize a parametric stereo encoder instance. |
| * |
| * \param hParametricStereo Meta Data handle. |
| * \param hPsEncConfig Filled parametric stereo configuration structure. |
| * \param noQmfSlots Number of slots within one audio frame. |
| * \param noQmfBands Number of QMF bands. |
| * \param dynamic_RAM Pointer to preallocated workbuffer. |
| * |
| * \return |
| * - PSENC_OK, on succes. |
| * - PSENC_INVALID_HANDLE, PSENC_INIT_ERROR, on failure. |
| */ |
| FDK_PSENC_ERROR PSEnc_Init( |
| HANDLE_PARAMETRIC_STEREO hParametricStereo, |
| const HANDLE_PSENC_CONFIG hPsEncConfig, |
| INT noQmfSlots, |
| INT noQmfBands |
| ,UCHAR *dynamic_RAM |
| ); |
| |
| |
| /** |
| * \brief Destroy parametric stereo encoder instance. |
| * |
| * Deallocate instance and free whole memory. |
| * |
| * \param phParametricStereo Pointer to the parametric stereo handle to be deallocated. |
| * |
| * \return |
| * - PSENC_OK, on succes. |
| * - PSENC_INVALID_HANDLE, on failure. |
| */ |
| FDK_PSENC_ERROR PSEnc_Destroy( |
| HANDLE_PARAMETRIC_STEREO *phParametricStereo |
| ); |
| |
| |
| /** |
| * \brief Apply parametric stereo processing. |
| * |
| * \param hParametricStereo Meta Data handle. |
| * \param samples Pointer to 2 channel audio input signal. |
| * \param timeInStride, Stride factor of input buffer. |
| * \param hQmfAnalysis, Pointer to QMF analysis filterbanks. |
| * \param downmixedRealQmfData Pointer to real QMF buffer to be written to. |
| * \param downmixedImagQmfData Pointer to imag QMF buffer to be written to. |
| * \param downsampledOutSignal Pointer to buffer where to write downmixed timesignal. |
| * \param sbrSynthQmf Pointer to QMF synthesis filterbank. |
| * \param qmfScale Return scaling factor of the qmf data. |
| * \param sendHeader Signal whether to write header data. |
| * |
| * \return |
| * - PSENC_OK, on succes. |
| * - PSENC_INVALID_HANDLE, PSENC_ENCODE_ERROR, on failure. |
| */ |
| FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing( |
| HANDLE_PARAMETRIC_STEREO hParametricStereo, |
| INT_PCM *samples[2], |
| UINT timeInStride, |
| QMF_FILTER_BANK **hQmfAnalysis, |
| FIXP_QMF **RESTRICT downmixedRealQmfData, |
| FIXP_QMF **RESTRICT downmixedImagQmfData, |
| INT_PCM *downsampledOutSignal, |
| HANDLE_QMF_FILTER_BANK sbrSynthQmf, |
| SCHAR *qmfScale, |
| const int sendHeader |
| ); |
| |
| |
| /** |
| * \brief Write parametric stereo bitstream. |
| * |
| * Write ps_data() element to bitstream and return number of written bits. |
| * Returns number of written bits only, if hBitstream == NULL. |
| * |
| * \param hParametricStereo Meta Data handle. |
| * \param hBitstream Bitstream buffer handle. |
| * |
| * \return |
| * - number of written bits. |
| */ |
| INT FDKsbrEnc_PSEnc_WritePSData( |
| HANDLE_PARAMETRIC_STEREO hParametricStereo, |
| HANDLE_FDK_BITSTREAM hBitstream |
| ); |
| |
| #endif /* __INCLUDED_PS_MAIN_H */ |