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/* -----------------------------------------------------------------------------------------------------------
Software License for The Fraunhofer FDK AAC Codec Library for Android
© Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
All rights reserved.
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 ( 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.
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."
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.
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.
Fraunhofer Institute for Integrated Circuits IIS
Attention: Audio and Multimedia Departments - FDK AAC LL
Am Wolfsmantel 33
91058 Erlangen, Germany
----------------------------------------------------------------------------------------------------------- */
/***************************** MPEG-4 AAC Decoder **************************
Author(s): Manuel Jander
Description: MPEG Transport data tables
#ifndef __TP_DATA_H__
#define __TP_DATA_H__
#include "machine_type.h"
#include "FDK_audio.h"
#include "FDK_bitstream.h"
* Configuration
#define TP_GA_ENABLE
/* #define TP_CELP_ENABLE */
/* #define TP_HVXC_ENABLE */
/* #define TP_SLS_ENABLE */
/* #define TP_USAC_ENABLE */
/* #define TP_RSVD50_ENABLE */
#if defined(TP_GA_ENABLE) || defined(TP_SLS_ENABLE)
#define TP_PCE_ENABLE /**< Enable full PCE support */
* ProgramConfig struct.
/* ISO/IEC 14496-3 Table 4.2 Program config element */
#define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */
#define PC_CCEL_MAX 16 /* CC elements */
typedef struct
/* PCE bitstream elements: */
UCHAR ElementInstanceTag;
UCHAR Profile;
UCHAR SamplingFrequencyIndex;
UCHAR NumFrontChannelElements;
UCHAR NumSideChannelElements;
UCHAR NumBackChannelElements;
UCHAR NumLfeChannelElements;
UCHAR NumAssocDataElements;
UCHAR NumValidCcElements;
UCHAR MonoMixdownPresent;
UCHAR MonoMixdownElementNumber;
UCHAR StereoMixdownPresent;
UCHAR StereoMixdownElementNumber;
UCHAR MatrixMixdownIndexPresent;
UCHAR MatrixMixdownIndex;
UCHAR PseudoSurroundEnable;
UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX];
UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX];
UCHAR ValidCcElementTagSelect[PC_CCEL_MAX];
UCHAR CommentFieldBytes;
#endif /* TP_PCE_ENABLE */
/* Helper variables for administration: */
UCHAR isValid; /*!< Flag showing if PCE has been read successfully. */
UCHAR NumChannels; /*!< Amount of audio channels summing all channel elements including LFEs */
UCHAR NumEffectiveChannels; /*!< Amount of audio channels summing only SCEs and CPEs */
UCHAR elCounter;
} CProgramConfig;
typedef enum {
ASCEXT_SBR = 0x2b7,
ASCEXT_PS = 0x548,
ASCEXT_MPS = 0x76a,
ASCEXT_SAOC = 0x7cb,
* GaSpecificConfig struct
typedef struct {
UINT m_frameLengthFlag ;
UINT m_dependsOnCoreCoder ;
UINT m_coreCoderDelay ;
UINT m_extensionFlag ;
UINT m_extensionFlag3 ;
UINT m_layer;
UINT m_numOfSubFrame;
UINT m_layerLength;
} CSGaSpecificConfig;
#endif /* TP_GA_ENABLE */
typedef enum {
ELDEXT_TERM = 0x0, /* Termination tag */
ELDEXT_SAOC = 0x1, /* SAOC config */
ELDEXT_LDSAC = 0x2 /* LD MPEG Surround config */
/* reserved */
typedef struct {
UCHAR m_frameLengthFlag;
UCHAR m_sbrPresentFlag;
UCHAR m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */
UCHAR m_sbrSamplingRate;
UCHAR m_sbrCrcFlag;
} CSEldSpecificConfig;
#endif /* TP_ELD_ENABLE */
* Audio configuration struct, suitable for encoder and decoder configuration.
typedef struct {
/* XYZ Specific Data */
union {
CSGaSpecificConfig m_gaSpecificConfig; /**< General audio specific configuration. */
#endif /* TP_GA_ENABLE */
CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */
#endif /* TP_ELD_ENABLE */
} m_sc;
/* Common ASC parameters */
CProgramConfig m_progrConfigElement; /**< Program configuration. */
#endif /* TP_PCE_ENABLE */
AUDIO_OBJECT_TYPE m_aot; /**< Audio Object Type. */
UINT m_samplingFrequency; /**< Samplerate. */
UINT m_samplesPerFrame; /**< Amount of samples per frame. */
UINT m_directMapping; /**< Document this please !! */
AUDIO_OBJECT_TYPE m_extensionAudioObjectType; /**< Audio object type */
UINT m_extensionSamplingFrequency; /**< Samplerate */
SCHAR m_channelConfiguration; /**< Channel configuration index */
SCHAR m_epConfig; /**< Error protection index */
SCHAR m_vcb11Flag; /**< aacSectionDataResilienceFlag */
SCHAR m_rvlcFlag; /**< aacScalefactorDataResilienceFlag */
SCHAR m_hcrFlag; /**< aacSpectralDataResilienceFlag */
SCHAR m_sbrPresentFlag; /**< Flag indicating the presence of SBR data in the bitstream */
SCHAR m_psPresentFlag; /**< Flag indicating the presence of parametric stereo data in the bitstream */
UCHAR m_samplingFrequencyIndex; /**< Samplerate index */
UCHAR m_extensionSamplingFrequencyIndex; /**< Samplerate index */
SCHAR m_extensionChannelConfiguration; /**< Channel configuration index */
} CSAudioSpecificConfig;
typedef INT (*cbUpdateConfig_t)(void*, const CSAudioSpecificConfig*);
typedef INT (*cbSsc_t)(
const AUDIO_OBJECT_TYPE coreCodec,
const INT samplingFrequency,
const INT muxMode,
const INT configBytes
typedef INT (*cbSbr_t)(
void * self,
const INT sampleRateIn,
const INT sampleRateOut,
const INT samplesPerFrame,
const AUDIO_OBJECT_TYPE coreCodec,
const MP4_ELEMENT_ID elementID,
const INT elementIndex
typedef struct {
cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change notify callback. */
void *cbUpdateConfigData; /*!< User data pointer for Config change notify callback. */
cbSsc_t cbSsc; /*!< Function pointer for SSC parser callback. */
void *cbSscData; /*!< User data pointer for SSC parser callback. */
cbSbr_t cbSbr; /*!< Function pointer for SBR header parser callback. */
void *cbSbrData; /*!< User data pointer for SBR header parser callback. */
} CSTpCallBacks;
static const UINT SamplingRateTable[] =
{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0,
static inline
int getSamplingRateIndex( UINT samplingRate )
UINT sf_index, tableSize=sizeof(SamplingRateTable)/sizeof(UINT);
for (sf_index=0; sf_index<tableSize; sf_index++) {
if( SamplingRateTable[sf_index] == samplingRate ) break;
if (sf_index>tableSize-1) {
return tableSize-1;
return sf_index;
* Get Channel count from channel configuration
static inline int getNumberOfTotalChannels(int channelConfig)
switch (channelConfig) {
case 1: case 2: case 3:
case 4: case 5: case 6:
return channelConfig;
case 7: case 12: case 14:
return 8;
case 11:
return 7;
return 0;
static inline
int getNumberOfEffectiveChannels(const int channelConfig)
{ /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */
const int n[] = {0,1,2,3,4,5,5,7,0,0, 0, 6, 7, 0, 7, 0};
return n[channelConfig];
#endif /* __TP_DATA_H__ */