mirror of
https://github.com/mstorsjo/fdk-aac.git
synced 2024-12-12 08:26:11 +01:00
5eb9ed1b1f
Bug: 176246647 Test: atest android.media.cts.DecoderTestAacFormat android.media.cts.DecoderTestXheAac android.media.cts.DecoderTestAacDrc Change-Id: Ib8b6973e9c29e13b8ef33d7736be2b977928face
467 lines
17 KiB
C
467 lines
17 KiB
C
/* -----------------------------------------------------------------------------
|
|
Software License for The Fraunhofer FDK AAC Codec Library for Android
|
|
|
|
© Copyright 1995 - 2020 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 transport format encoder library *********************
|
|
|
|
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_USAC_MAX_SPEAKERS (24)
|
|
|
|
#define TP_USAC_MAX_EXT_ELEMENTS ((24))
|
|
|
|
#define TP_USAC_MAX_ELEMENTS ((24) + TP_USAC_MAX_EXT_ELEMENTS)
|
|
|
|
#define TP_USAC_MAX_CONFIG_LEN \
|
|
512 /* next power of two of maximum of escapedValue(hBs, 4, 4, 8) in \
|
|
AudioPreRoll() (285) */
|
|
|
|
#define TPDEC_USAC_NUM_CONFIG_CHANGE_FRAMES \
|
|
(1) /* Number of frames for config change in USAC */
|
|
|
|
enum {
|
|
TPDEC_FLUSH_OFF = 0,
|
|
TPDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1,
|
|
TPDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2,
|
|
TPDEC_USAC_DASH_IPF_FLUSH_ON = 3
|
|
};
|
|
|
|
enum {
|
|
TPDEC_BUILD_UP_OFF = 0,
|
|
TPDEC_RSV60_BUILD_UP_ON = 1,
|
|
TPDEC_RSV60_BUILD_UP_ON_IN_BAND = 2,
|
|
TPDEC_USAC_BUILD_UP_ON = 3,
|
|
TPDEC_RSV60_BUILD_UP_IDLE = 4,
|
|
TPDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5
|
|
};
|
|
|
|
/**
|
|
* ProgramConfig struct.
|
|
*/
|
|
/* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */
|
|
#define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */
|
|
#define PC_LFE_CHANNELS_MAX 4
|
|
#define PC_ASSOCDATA_MAX 8
|
|
#define PC_CCEL_MAX 16 /* CC elements */
|
|
#define PC_COMMENTLENGTH 256
|
|
#define PC_NUM_HEIGHT_LAYER 3
|
|
|
|
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 FrontElementIsCpe[PC_FSB_CHANNELS_MAX];
|
|
UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX];
|
|
UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX];
|
|
|
|
UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX];
|
|
UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX];
|
|
UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX];
|
|
|
|
UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX];
|
|
UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX];
|
|
UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX];
|
|
|
|
UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX];
|
|
|
|
UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX];
|
|
|
|
UCHAR CcElementIsIndSw[PC_CCEL_MAX];
|
|
UCHAR ValidCcElementTagSelect[PC_CCEL_MAX];
|
|
|
|
UCHAR CommentFieldBytes;
|
|
UCHAR Comment[PC_COMMENTLENGTH];
|
|
|
|
/* 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_UNKOWN = -1,
|
|
ASCEXT_SBR = 0x2b7,
|
|
ASCEXT_PS = 0x548,
|
|
ASCEXT_MPS = 0x76a,
|
|
ASCEXT_SAOC = 0x7cb,
|
|
ASCEXT_LDMPS = 0x7cc
|
|
|
|
} TP_ASC_EXTENSION_ID;
|
|
|
|
/**
|
|
* 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;
|
|
|
|
typedef enum {
|
|
ELDEXT_TERM = 0x0, /* Termination tag */
|
|
ELDEXT_SAOC = 0x1, /* SAOC config */
|
|
ELDEXT_LDSAC = 0x2, /* LD MPEG Surround config */
|
|
ELDEXT_DOWNSCALEINFO = 0x3 /* ELD sample rate adaptation */
|
|
/* reserved */
|
|
} ASC_ELD_EXT_TYPE;
|
|
|
|
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;
|
|
UINT m_downscaledSamplingFrequency;
|
|
|
|
} CSEldSpecificConfig;
|
|
|
|
typedef struct {
|
|
USAC_EXT_ELEMENT_TYPE usacExtElementType;
|
|
USHORT usacExtElementConfigLength;
|
|
USHORT usacExtElementDefaultLength;
|
|
UCHAR usacExtElementPayloadFrag;
|
|
UCHAR usacExtElementHasAudioPreRoll;
|
|
} CSUsacExtElementConfig;
|
|
|
|
typedef struct {
|
|
MP4_ELEMENT_ID usacElementType;
|
|
UCHAR m_noiseFilling;
|
|
UCHAR m_harmonicSBR;
|
|
UCHAR m_interTes;
|
|
UCHAR m_pvc;
|
|
UCHAR m_stereoConfigIndex;
|
|
CSUsacExtElementConfig extElement;
|
|
} CSUsacElementConfig;
|
|
|
|
typedef struct {
|
|
UCHAR m_frameLengthFlag;
|
|
UCHAR m_coreSbrFrameLengthIndex;
|
|
UCHAR m_sbrRatioIndex;
|
|
UCHAR m_nUsacChannels; /* number of audio channels signaled in
|
|
UsacDecoderConfig() / rsv603daDecoderConfig() via
|
|
numElements and usacElementType */
|
|
UCHAR m_channelConfigurationIndex;
|
|
UINT m_usacNumElements;
|
|
CSUsacElementConfig element[TP_USAC_MAX_ELEMENTS];
|
|
|
|
UCHAR numAudioChannels;
|
|
UCHAR m_usacConfigExtensionPresent;
|
|
UCHAR elementLengthPresent;
|
|
UCHAR UsacConfig[TP_USAC_MAX_CONFIG_LEN];
|
|
USHORT UsacConfigBits;
|
|
} CSUsacConfig;
|
|
|
|
/**
|
|
* Audio configuration struct, suitable for encoder and decoder configuration.
|
|
*/
|
|
typedef struct {
|
|
/* XYZ Specific Data */
|
|
union {
|
|
CSGaSpecificConfig
|
|
m_gaSpecificConfig; /**< General audio specific configuration. */
|
|
CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */
|
|
CSUsacConfig m_usacConfig; /**< USAC specific configuration */
|
|
} m_sc;
|
|
|
|
/* Common ASC parameters */
|
|
CProgramConfig m_progrConfigElement; /**< Program configuration. */
|
|
|
|
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 */
|
|
|
|
UCHAR
|
|
configMode; /**< The flag indicates if the callback shall work in memory
|
|
allocation mode or in config change detection mode */
|
|
UCHAR AacConfigChanged; /**< The flag will be set if at least one aac config
|
|
parameter has changed that requires a memory
|
|
reconfiguration, otherwise it will be cleared */
|
|
UCHAR SbrConfigChanged; /**< The flag will be set if at least one sbr config
|
|
parameter has changed that requires a memory
|
|
reconfiguration, otherwise it will be cleared */
|
|
UCHAR SacConfigChanged; /**< The flag will be set if at least one sac config
|
|
parameter has changed that requires a memory
|
|
reconfiguration, otherwise it will be cleared */
|
|
|
|
UCHAR
|
|
config[TP_USAC_MAX_CONFIG_LEN]; /**< Configuration stored as bitstream */
|
|
UINT configBits; /**< Configuration length in bits */
|
|
|
|
} CSAudioSpecificConfig;
|
|
|
|
typedef struct {
|
|
SCHAR flushCnt; /**< Flush frame counter */
|
|
UCHAR flushStatus; /**< Flag indicates flush mode: on|off */
|
|
SCHAR buildUpCnt; /**< Build up frame counter */
|
|
UCHAR buildUpStatus; /**< Flag indicates build up mode: on|off */
|
|
UCHAR cfgChanged; /**< Flag indicates that the config changed and the decoder
|
|
needs to be initialized again via callback. Make sure
|
|
that memory is freed before initialization. */
|
|
UCHAR contentChanged; /**< Flag indicates that the content changed i.e. a
|
|
right truncation occured before */
|
|
UCHAR forceCfgChange; /**< Flag indicates if config change has to be forced
|
|
even if new config is the same */
|
|
} CCtrlCFGChange;
|
|
|
|
typedef INT (*cbUpdateConfig_t)(void *, const CSAudioSpecificConfig *,
|
|
const UCHAR configMode, UCHAR *configChanged);
|
|
typedef INT (*cbFreeMem_t)(void *, const CSAudioSpecificConfig *);
|
|
typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *);
|
|
typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM,
|
|
const AUDIO_OBJECT_TYPE coreCodec,
|
|
const INT samplingRate, const INT frameSize,
|
|
const INT numChannels, const INT stereoConfigIndex,
|
|
const INT coreSbrFrameLengthIndex, const INT configBytes,
|
|
const UCHAR configMode, UCHAR *configChanged);
|
|
|
|
typedef INT (*cbSbr_t)(void *self, HANDLE_FDK_BITSTREAM hBs,
|
|
const INT sampleRateIn, const INT sampleRateOut,
|
|
const INT samplesPerFrame,
|
|
const AUDIO_OBJECT_TYPE coreCodec,
|
|
const MP4_ELEMENT_ID elementID, const INT elementIndex,
|
|
const UCHAR harmonicSbr, const UCHAR stereoConfigIndex,
|
|
const UCHAR configMode, UCHAR *configChanged,
|
|
const INT downscaleFactor);
|
|
|
|
typedef INT (*cbUsac_t)(void *self, HANDLE_FDK_BITSTREAM hBs);
|
|
|
|
typedef INT (*cbUniDrc_t)(void *self, HANDLE_FDK_BITSTREAM hBs,
|
|
const INT fullPayloadLength, const INT payloadType,
|
|
const INT subStreamIndex, const INT payloadStart,
|
|
const AUDIO_OBJECT_TYPE);
|
|
|
|
typedef struct {
|
|
cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change
|
|
notify callback. */
|
|
void *cbUpdateConfigData; /*!< User data pointer for Config change notify
|
|
callback. */
|
|
cbFreeMem_t cbFreeMem; /*!< Function pointer for free memory callback. */
|
|
void *cbFreeMemData; /*!< User data pointer for free memory callback. */
|
|
cbCtrlCFGChange_t cbCtrlCFGChange; /*!< Function pointer for config change
|
|
control callback. */
|
|
void *cbCtrlCFGChangeData; /*!< User data pointer for config change control
|
|
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. */
|
|
cbUsac_t cbUsac;
|
|
void *cbUsacData;
|
|
cbUniDrc_t cbUniDrc; /*!< Function pointer for uniDrcConfig and
|
|
loudnessInfoSet parser callback. */
|
|
void *cbUniDrcData; /*!< User data pointer for uniDrcConfig and
|
|
loudnessInfoSet parser callback. */
|
|
} CSTpCallBacks;
|
|
|
|
static const UINT SamplingRateTable[] = {
|
|
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025,
|
|
8000, 7350, 0, 0, 57600, 51200, 40000, 38400, 34150, 28800, 25600,
|
|
20000, 19200, 17075, 14400, 12800, 9600, 0, 0, 0, 0};
|
|
|
|
static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) {
|
|
UINT sf_index;
|
|
UINT tableSize = (1 << nBits) - 1;
|
|
|
|
for (sf_index = 0; sf_index < tableSize; sf_index++) {
|
|
if (SamplingRateTable[sf_index] == samplingRate) break;
|
|
}
|
|
|
|
if (sf_index > tableSize) {
|
|
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;
|
|
case 13:
|
|
return 24;
|
|
default:
|
|
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, 22, 7, 0};
|
|
return n[channelConfig];
|
|
}
|
|
|
|
#endif /* TP_DATA_H */
|