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-rw-r--r--libMpegTPEnc/include/tp_data.h487
1 files changed, 301 insertions, 186 deletions
diff --git a/libMpegTPEnc/include/tp_data.h b/libMpegTPEnc/include/tp_data.h
index c6e89b5..6f032d1 100644
--- a/libMpegTPEnc/include/tp_data.h
+++ b/libMpegTPEnc/include/tp_data.h
@@ -1,74 +1,85 @@
-
-/* -----------------------------------------------------------------------------------------------------------
+/* -----------------------------------------------------------------------------
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.
+© Copyright 1995 - 2018 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.
+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:
+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 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
+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.
+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.
+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."
+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.
+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.
+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.
+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
@@ -79,17 +90,18 @@ Am Wolfsmantel 33
www.iis.fraunhofer.de/amm
amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------------------------------------ */
+----------------------------------------------------------------------------- */
+
+/******************* MPEG transport format encoder library *********************
-/***************************** MPEG-4 AAC Decoder **************************
+ Author(s): Manuel Jander
- Author(s): Manuel Jander
Description: MPEG Transport data tables
-******************************************************************************/
+*******************************************************************************/
-#ifndef __TP_DATA_H__
-#define __TP_DATA_H__
+#ifndef TP_DATA_H
+#define TP_DATA_H
#include "machine_type.h"
#include "FDK_audio.h"
@@ -98,17 +110,35 @@ amm-info@iis.fraunhofer.de
/*
* Configuration
*/
-#define TP_GA_ENABLE
-/* #define TP_CELP_ENABLE */
-/* #define TP_HVXC_ENABLE */
-/* #define TP_SLS_ENABLE */
-#define TP_ELD_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 */
-#endif
+
+#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.
@@ -116,13 +146,12 @@ amm-info@iis.fraunhofer.de
/* 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_ASSOCDATA_MAX 8
+#define PC_CCEL_MAX 16 /* CC elements */
+#define PC_COMMENTLENGTH 256
+#define PC_NUM_HEIGHT_LAYER 3
-typedef struct
-{
-#ifdef TP_PCE_ENABLE
+typedef struct {
/* PCE bitstream elements: */
UCHAR ElementInstanceTag;
UCHAR Profile;
@@ -165,54 +194,50 @@ typedef struct
UCHAR CommentFieldBytes;
UCHAR Comment[PC_COMMENTLENGTH];
-#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 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
+ ASCEXT_SBR = 0x2b7,
+ ASCEXT_PS = 0x548,
+ ASCEXT_MPS = 0x76a,
+ ASCEXT_SAOC = 0x7cb,
+ ASCEXT_LDMPS = 0x7cc
} TP_ASC_EXTENSION_ID;
-#ifdef TP_GA_ENABLE
/**
* GaSpecificConfig struct
*/
typedef struct {
- UINT m_frameLengthFlag ;
- UINT m_dependsOnCoreCoder ;
- UINT m_coreCoderDelay ;
+ UINT m_frameLengthFlag;
+ UINT m_dependsOnCoreCoder;
+ UINT m_coreCoderDelay;
- UINT m_extensionFlag ;
- UINT m_extensionFlag3 ;
+ UINT m_extensionFlag;
+ UINT m_extensionFlag3;
UINT m_layer;
UINT m_numOfSubFrame;
UINT m_layerLength;
} CSGaSpecificConfig;
-#endif /* TP_GA_ENABLE */
-
-
-
-
-#ifdef TP_ELD_ENABLE
typedef enum {
- ELDEXT_TERM = 0x0, /* Termination tag */
- ELDEXT_SAOC = 0x1, /* SAOC config */
- ELDEXT_LDSAC = 0x2 /* LD MPEG Surround config */
+ 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;
@@ -220,103 +245,186 @@ typedef struct {
UCHAR m_frameLengthFlag;
UCHAR m_sbrPresentFlag;
- UCHAR m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */
+ UCHAR
+ m_useLdQmfTimeAlign; /* Use LD-MPS QMF in SBR to achive time alignment */
UCHAR m_sbrSamplingRate;
UCHAR m_sbrCrcFlag;
+ UINT m_downscaledSamplingFrequency;
} CSEldSpecificConfig;
-#endif /* TP_ELD_ENABLE */
+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 {
-#ifdef TP_GA_ENABLE
- CSGaSpecificConfig m_gaSpecificConfig; /**< General audio specific configuration. */
-#endif /* TP_GA_ENABLE */
-#ifdef TP_ELD_ENABLE
- CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */
-#endif /* TP_ELD_ENABLE */
+ CSGaSpecificConfig
+ m_gaSpecificConfig; /**< General audio specific configuration. */
+ CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */
+ CSUsacConfig m_usacConfig; /**< USAC specific configuration */
} m_sc;
-
- /* Common ASC parameters */
-#ifdef TP_PCE_ENABLE
- 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 */
+ /* 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 INT (*cbUpdateConfig_t)(void*, const CSAudioSpecificConfig*);
-typedef INT (*cbSsc_t)(
- void*, HANDLE_FDK_BITSTREAM,
- const AUDIO_OBJECT_TYPE coreCodec,
- const INT samplingFrequency,
- const INT muxMode,
- const INT configBytes
- );
-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
- );
-
-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. */
+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 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,
- 0
-};
+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 sf_index, tableSize=sizeof(SamplingRateTable)/sizeof(UINT);
+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;
+ for (sf_index = 0; sf_index < tableSize; sf_index++) {
+ if (SamplingRateTable[sf_index] == samplingRate) break;
}
- if (sf_index>tableSize-1) {
- return tableSize-1;
+ if (sf_index > tableSize) {
+ return tableSize - 1;
}
return sf_index;
@@ -325,26 +433,33 @@ int getSamplingRateIndex( UINT samplingRate )
/*
* Get Channel count from channel configuration
*/
-static inline int getNumberOfTotalChannels(int channelConfig)
-{
+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;
- default:
- return 0;
+ 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, 0, 7, 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__ */
+#endif /* TP_DATA_H */