/* -----------------------------------------------------------------------------

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 decoder 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;

}

Unexecuted instantiation: tpenc_adif.cpp:getSamplingRateIndex(unsigned int, unsigned int)

Unexecuted instantiation: tpenc_adts.cpp:getSamplingRateIndex(unsigned int, unsigned int)

Unexecuted instantiation: tpenc_asc.cpp:getSamplingRateIndex(unsigned int, unsigned int)

/*

 * 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;

  }

}

Unexecuted instantiation: tpenc_adif.cpp:getNumberOfTotalChannels(int)

Unexecuted instantiation: tpenc_adts.cpp:getNumberOfTotalChannels(int)

Unexecuted instantiation: tpenc_asc.cpp:getNumberOfTotalChannels(int)

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];

}

Unexecuted instantiation: tpenc_adif.cpp:getNumberOfEffectiveChannels(int)

Unexecuted instantiation: tpenc_adts.cpp:getNumberOfEffectiveChannels(int)

Unexecuted instantiation: tpenc_asc.cpp:getNumberOfEffectiveChannels(int)

#endif /* TP_DATA_H */