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author | The Android Open Source Project <initial-contribution@android.com> | 2012-07-11 10:15:24 -0700 |
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committer | The Android Open Source Project <initial-contribution@android.com> | 2012-07-11 10:15:24 -0700 |
commit | 2228e360595641dd906bf1773307f43d304f5b2e (patch) | |
tree | 57f3d390ebb0782cc0de0fb984c8ea7e45b4f386 /libPCMutils | |
download | fdk-aac-dabplus-2228e360595641dd906bf1773307f43d304f5b2e.tar.gz fdk-aac-dabplus-2228e360595641dd906bf1773307f43d304f5b2e.tar.bz2 fdk-aac-dabplus-2228e360595641dd906bf1773307f43d304f5b2e.zip |
Snapshot 2bda038c163298531d47394bc2c09e1409c5d0db
Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54
Diffstat (limited to 'libPCMutils')
-rw-r--r-- | libPCMutils/include/pcmutils_lib.h | 282 | ||||
-rw-r--r-- | libPCMutils/src/pcmutils_lib.cpp | 1281 |
2 files changed, 1563 insertions, 0 deletions
diff --git a/libPCMutils/include/pcmutils_lib.h b/libPCMutils/include/pcmutils_lib.h new file mode 100644 index 0000000..fd4a26d --- /dev/null +++ b/libPCMutils/include/pcmutils_lib.h @@ -0,0 +1,282 @@ + +/* ----------------------------------------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------------------------------------- */ + +/************************ FDK PCM up/downmixing module ********************* + + Author(s): Christian Griebel + Description: Declares functions to interface with the PCM downmix processing + module. + +*******************************************************************************/ + +#ifndef _PCMUTILS_LIB_H_ +#define _PCMUTILS_LIB_H_ + +#include "machine_type.h" +#include "common_fix.h" +#include "FDK_audio.h" + +/* ------------------------ * + * MODULE SETTINGS: * + * ------------------------ */ +/* #define PCM_UPMIX_ENABLE */ /*!< Generally enable up mixing. */ +#define PCM_DOWNMIX_ENABLE /*!< Generally enable down mixing. */ +#define DVB_MIXDOWN_ENABLE /*!< Enable this to support DVB ancillary data for encoder + assisted downmixing of MPEG-4 AAC and + MPEG-1/2 layer 2 streams. PCM_DOWNMIX_ENABLE has to + be enabled, too! */ +#define MPEG_PCE_MIXDOWN_ENABLE /*!< Enable this to support MPEG matrix mixdown with a + coefficient carried in the PCE. PCM_DOWNMIX_ENABLE + has to be enabled, too! */ +/* #define ARIB_MIXDOWN_ENABLE */ /*!< Enable modifications to the MPEG PCE mixdown method + to fulfill ARIB standard. MPEG_PCE_MIXDOWN_ENABLE has + to be set. */ + +/* ------------------------ * + * ERROR CODES: * + * ------------------------ */ +typedef enum +{ + PCMDMX_OK = 0x0, /*!< No error happened. */ + PCMDMX_OUT_OF_MEMORY = 0x2, /*!< Not enough memory to set up an instance of the module. */ + PCMDMX_UNKNOWN = 0x5, /*!< Error condition is of unknown reason, or from a third + party module. */ + PCMDMX_INVALID_HANDLE, /*!< The given instance handle is not valid. */ + PCMDMX_INVALID_ARGUMENT, /*!< One of the parameters handed over is invalid. */ + PCMDMX_INVALID_CH_CONFIG, /*!< The given channel configuration is not supported and + thus no processing was performed. */ + PCMDMX_INVALID_MODE, /*!< The set configuration/mode is not applicable. */ + PCMDMX_UNKNOWN_PARAM, /*!< The handed parameter is not known/supported. */ + PCMDMX_UNABLE_TO_SET_PARAM, /*!< Unable to set the specific parameter. Most probably + the value ist out of range. */ + PCMDMX_CORRUPT_ANC_DATA /*!< The read ancillary data was corrupt. */ + +} PCMDMX_ERROR; + + +/* ------------------------ * + * RUNTIME PARAMS: * + * ------------------------ */ +typedef enum +{ + DMX_BS_DATA_EXPIRY_FRAME, /*!< The number of frames without new metadata that have to + go by before the bitstream data expires. The value 0 + disables expiry. */ + DMX_BS_DATA_DELAY, /*!< The number of delay frames of the output samples + compared to the bitstream data. */ + NUMBER_OF_OUTPUT_CHANNELS , /*!< The number of output channels (equals the number of + channels processed by the audio output setup). */ + DUAL_CHANNEL_DOWNMIX_MODE /*!< Downmix mode for two channel audio data. */ + +} PCMDMX_PARAM; + + +typedef enum +{ + STEREO_MODE = 0x0, /*!< Leave stereo signals as they are. */ + CH1_MODE = 0x1, /*!< Create a dual mono output signal from channel 1. */ + CH2_MODE = 0x2, /*!< Create a dual mono output signal from channel 2. */ + MIXED_MODE = 0x3 /*!< Create a dual mono output signal by mixing the two channels. */ + +} DUAL_CHANNEL_MODE; + + +/* ------------------------ * + * MODULES INTERFACE: * + * ------------------------ */ +typedef struct PCM_DMX_INSTANCE *HANDLE_PCM_DOWNMIX; + +/* Modules reset flags */ +#define PCMDMX_RESET_PARAMS ( 1 ) +#define PCMDMX_RESET_BS_DATA ( 2 ) +#define PCMDMX_RESET_FULL ( PCMDMX_RESET_PARAMS | PCMDMX_RESET_BS_DATA ) + +#ifdef __cplusplus +extern "C" +{ +#endif + +/** Open and initialize an instance of the PCM downmix module + * @param [out] Pointer to a buffer receiving the handle of the new instance. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_Open ( + HANDLE_PCM_DOWNMIX *pSelf + ); + +/** Set one parameter for one instance of the PCM downmix module. + * @param [in] Handle of PCM downmix instance. + * @param [in] Parameter to be set. + * @param [in] Parameter value. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_SetParam ( + HANDLE_PCM_DOWNMIX self, + PCMDMX_PARAM param, + UINT value + ); + +/** Read the ancillary data transported in DSEs of DVB streams with MPEG-4 content + * @param [in] Handle of PCM downmix instance. + * @param [in] Pointer to ancillary data buffer. + * @param [in] Size of ancillary data. + * @param [in] Flag indicating wheter the ancillary data is from a MPEG-1/2 or an MPEG-4 stream. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_ReadDvbAncData ( + HANDLE_PCM_DOWNMIX self, + UCHAR *pAncDataBuf, + UINT ancDataBytes, + int isMpeg2 + ); + +/** Set the matrix mixdown information extracted from the PCE of an AAC bitstream. + * Note: Call only if matrix_mixdown_idx_present is true. + * @param [in] Handle of PCM downmix instance. + * @param [in] Matrix mixdown index present flag extracted from PCE. + * @param [in] The 2 bit matrix mixdown index extracted from PCE. + * @param [in] The pseudo surround enable flag extracted from PCE. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_SetMatrixMixdownFromPce ( + HANDLE_PCM_DOWNMIX self, + int matrixMixdownPresent, + int matrixMixdownIdx, + int pseudoSurroundEnable + ); + +/** Reset the module. + * @param [in] Handle of PCM downmix instance. + * @param [in] Flags telling which parts of the module shall be reset. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_Reset ( + HANDLE_PCM_DOWNMIX self, + UINT flags + ); + +/** Apply down or up mixing. + * + * \param [in] Handle of PCM downmix module instance. + * \param [inout] Pointer to time buffer with decoded PCM samples. + * \param [in] Pointer where the amount of output samples is returned into. + * \param [inout] Pointer where the amount of output channels is returned into. + * \param [in] Flag which indicates if output time data are writtern interleaved or as subsequent blocks. + * \param [inout] Array were the corresponding channel type for each output audio channel is stored into. + * \param [inout] Array were the corresponding channel type index for each output audio channel is stored into. + * \param [in] Array containing the output channel mapping to be used (From MPEG PCE ordering to whatever is required). + * + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_ApplyFrame ( + HANDLE_PCM_DOWNMIX self, + INT_PCM *pPcmBuf, + UINT frameSize, + INT *nChannels, + + int fInterleaved, + AUDIO_CHANNEL_TYPE channelType[], + UCHAR channelIndices[], + const UCHAR channelMapping[][8] + ); + +/** Close an instance of the PCM downmix module. + * @param [inout] Pointer to a buffer containing the handle of the instance. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_Close ( + HANDLE_PCM_DOWNMIX *pSelf + ); + +/** Get library info for this module. + * @param [out] Pointer to an allocated LIB_INFO structure. + * @returns Returns an error code. + */ +PCMDMX_ERROR pcmDmx_GetLibInfo( LIB_INFO *info ); + + +#ifdef __cplusplus +} +#endif + +#endif /* _PCMUTILS_LIB_H_ */ diff --git a/libPCMutils/src/pcmutils_lib.cpp b/libPCMutils/src/pcmutils_lib.cpp new file mode 100644 index 0000000..4272548 --- /dev/null +++ b/libPCMutils/src/pcmutils_lib.cpp @@ -0,0 +1,1281 @@ + +/* ----------------------------------------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------------------------------------- */ + +/**************************** FDK PCM utils module ************************** + + Author(s): Christian Griebel + Description: Defines functions to interface with the PCM post processing + module. + +*******************************************************************************/ + +#include "pcmutils_lib.h" + +#include "genericStds.h" +#include "fixpoint_math.h" + +/* Decoder library info */ +#define PCMDMX_LIB_VL0 2 +#define PCMDMX_LIB_VL1 2 +#define PCMDMX_LIB_VL2 1 +#define PCMDMX_LIB_TITLE "PCM Downmix Lib" +#define PCMDMX_LIB_BUILD_DATE __DATE__ +#define PCMDMX_LIB_BUILD_TIME __TIME__ + +/* Library settings */ +#define PCM_DMX_MAX_DELAY_FRAMES ( 1 ) +#define PCM_DMX_MAX_CHANNELS ( 8 ) +#define PCM_DMX_MAX_CHANNEL_GROUPS ( 4 ) +#define PCM_DMX_MAX_CHANNELS_PER_GROUP ( 3 ) /* The maximum over all groups */ +#define PCMDMX_DFLT_EXPIRY_FRAME ( 40 ) /* At least 400ms (FL 960 @ 96kHz) */ + +/* Fixed and unique channel group indices. + * The last group index has to be smaller than PCM_DMX_MAX_CHANNEL_GROUPS. */ +#define CH_GROUP_FRONT ( 0 ) +#define CH_GROUP_SIDE ( 1 ) +#define CH_GROUP_REAR ( 2 ) +#define CH_GROUP_LFE ( 3 ) + +/* The ordering of the following fixed channel labels has to be in MPEG-4 style. + * From the center to the back with left and right channel interleaved (starting with left). + * The last channel label index has to be smaller than PCM_DMX_MAX_CHANNELS. */ +#define CENTER_FRONT_CHANNEL ( 0 ) /* C */ +#define LEFT_FRONT_CHANNEL ( 1 ) /* L */ +#define RIGHT_FRONT_CHANNEL ( 2 ) /* R */ +#define LEFT_OUTSIDE_CHANNEL ( 3 ) /* Lo */ +#define RIGHT_OUTSIDE_CHANNEL ( 4 ) /* Ro */ +#define LEFT_REAR_CHANNEL ( 5 ) /* Lr aka left back channel */ +#define RIGHT_REAR_CHANNEL ( 6 ) /* Rr aka right back channel */ +#define LOW_FREQUENCY_CHANNEL ( 7 ) /* Lf */ + +/* More constants */ +#define ANC_DATA_SYNC_BYTE ( 0xBC ) /* ancillary data sync byte. */ +#define ATTENUATION_FACTOR_1 ( FL2FXCONST_SGL(0.70710678f) ) +#define TWO_CHANNEL ( 2 ) + +/* Sanity checks on library setting: */ + +/* List of packed channel modes */ +typedef enum +{ /* CH_MODE_<numFrontCh>_<numOutsideCh>_<numRearCh>_<numLfCh> */ + CH_MODE_UNDEFINED = 0x0000, + /* 1 channel */ + CH_MODE_1_0_0_0 = 0x0001, /* chCfg 1 */ + /* 2 channels */ + CH_MODE_2_0_0_0 = 0x0002, /* chCfg 2 */ + /* 3 channels */ + CH_MODE_3_0_0_0 = 0x0003, /* chCfg 3 */ + CH_MODE_2_0_1_0 = 0x0102, + CH_MODE_2_0_0_1 = 0x1002, + /* 4 channels */ + CH_MODE_3_0_1_0 = 0x0103, /* chCfg 4 */ + CH_MODE_2_0_2_0 = 0x0202, + CH_MODE_2_0_1_1 = 0x1102, + /* 5 channels */ + CH_MODE_3_0_2_0 = 0x0203, /* chCfg 5 */ + CH_MODE_2_0_2_1 = 0x1202, + CH_MODE_3_0_1_1 = 0x1103, + CH_MODE_3_2_0_0 = 0x0023, + /* 6 channels */ + CH_MODE_3_0_2_1 = 0x1203, /* chCfg 6 */ + CH_MODE_3_2_1_0 = 0x0123, + /* 7 channels */ + CH_MODE_2_2_2_1 = 0x1222, + CH_MODE_3_2_1_1 = 0x1123, + CH_MODE_3_2_2_0 = 0x0223, + /* 8 channels */ + CH_MODE_3_2_2_1 = 0x1222, /* chCfg 7 */ + CH_MODE_3_2_1_2 = 0x2123, + CH_MODE_2_2_2_2 = 0x2222 + +} PCM_DMX_CHANNEL_MODE; + + +/* These are the channel configurations linked to + the number of output channels give by the user: */ +static const PCM_DMX_CHANNEL_MODE outChModeTable[PCM_DMX_MAX_CHANNELS] = +{ + CH_MODE_1_0_0_0, /* 1 channel */ + CH_MODE_2_0_0_0, /* 2 channels */ + CH_MODE_3_0_0_0, /* 3 channels */ + CH_MODE_3_0_1_0, /* 4 channels */ + CH_MODE_3_0_2_0, /* 5 channels */ + CH_MODE_3_0_2_1, /* 6 channels */ + CH_MODE_3_2_2_0, /* 7 channels */ + CH_MODE_3_2_2_1 /* 8 channels */ +}; + +static const FIXP_SGL dvbDownmixFactors[8] = +{ + FL2FXCONST_SGL(1.0f), + FL2FXCONST_SGL(0.841f), + FL2FXCONST_SGL(0.707f), + FL2FXCONST_SGL(0.596f), + FL2FXCONST_SGL(0.500f), + FL2FXCONST_SGL(0.422f), + FL2FXCONST_SGL(0.355f), + FL2FXCONST_SGL(0.0f) +}; + + + /* MPEG matrix mixdown: + Set 1: L' = (1 + 2^-0.5 + A )^-1 * [L + C * 2^-0.5 + A * Ls]; + R' = (1 + 2^-0.5 + A )^-1 * [R + C * 2^-0.5 + A * Rs]; + + Set 2: L' = (1 + 2^-0.5 + 2A )^-1 * [L + C * 2^-0.5 - A * (Ls + Rs)]; + R' = (1 + 2^-0.5 + 2A )^-1 * [R + C * 2^-0.5 + A * (Ls + Rs)]; + + M = (3 + 2A)^-1 * [L + C + R + A*(Ls + Rs)]; + */ + static const FIXP_SGL mpegMixDownIdx2Coef[4] = + { + FL2FXCONST_SGL(0.70710678f), + FL2FXCONST_SGL(0.5f), + FL2FXCONST_SGL(0.35355339f), + FL2FXCONST_SGL(0.0f) + }; + + static const FIXP_SGL mpegMixDownIdx2PreFact[4] = + { + FL2FXCONST_SGL(0.4142135623730950f), + FL2FXCONST_SGL(0.4530818393219728f), + FL2FXCONST_SGL(0.4852813742385703f), + FL2FXCONST_SGL(0.5857864376269050f) + }; + + typedef struct + { + USHORT matrixMixdownIdx; /*!< MPEG mixdown index extracted from PCE. */ + USHORT pseudoSurroundEnable; /*!< Pseudo surround enable flag extracted from PCE. */ + USHORT mixdownAvailable; /*!< Will be set to 1 if we found a valid coefficient. */ + + } MPEG_MIXDOWN_INFO; + + +typedef struct +{ + FIXP_SGL centerMixLevelValue; /*!< DVB mixdown level for the center channel extracted from anc data. */ + FIXP_SGL surroundMixLevelValue; /*!< DVB mixdown level for back channels extracted from anc data. */ + + UCHAR mixLevelsAvail; /*!< Will be set to 1 if we found a valid coefficient. */ + +} DVB_MIXDOWN_LEVELS; + + +/* Modules main data structure: */ +struct PCM_DMX_INSTANCE +{ + DVB_MIXDOWN_LEVELS dvbMixDownLevels[PCM_DMX_MAX_DELAY_FRAMES+1]; + MPEG_MIXDOWN_INFO mpegMixDownInfo[PCM_DMX_MAX_DELAY_FRAMES+1]; + DUAL_CHANNEL_MODE dualChannelMode; + UINT expiryFrame; + UINT expiryCount; + SHORT numOutputChannels; + UCHAR applyProcessing; + UCHAR frameDelay; +}; + +/* Memory allocation macro */ +C_ALLOC_MEM_STATIC(PcmDmxInstance, struct PCM_DMX_INSTANCE, 1) + + +/** Evaluate a given channel configuration and extract a packed channel mode and generate a channel offset table + * This function is the inverse to the getChannelDescription() routine. + * @param [in] The total number of channels of the given configuration. + * @param [in] Array holding the corresponding channel types for each channel. + * @param [in] Array holding the corresponding channel type indices for each channel. + * @param [in] Array containing the channel mapping to be used (From MPEG PCE ordering to whatever is required). + * @param [out] Array where the buffer offsets for each channel are stored into. + * @returns Returns the packed channel mode. + **/ +static +PCM_DMX_CHANNEL_MODE getChannelMode ( + const INT numChannels, /* in */ + const AUDIO_CHANNEL_TYPE channelType[], /* in */ + const UCHAR channelIndices[], /* in */ + const UCHAR channelMapping[PCM_DMX_MAX_CHANNELS], /* in */ + UCHAR offsetTable[PCM_DMX_MAX_CHANNELS] /* out */ + ) +{ + UINT chMode = CH_MODE_UNDEFINED; + UCHAR chIdx[PCM_DMX_MAX_CHANNEL_GROUPS][PCM_DMX_MAX_CHANNELS_PER_GROUP]; + UCHAR numChInGrp[PCM_DMX_MAX_CHANNEL_GROUPS]; + int ch, grpIdx, err = 0; + + FDK_ASSERT(channelType != NULL); + FDK_ASSERT(channelIndices != NULL); + FDK_ASSERT(channelMapping != NULL); + FDK_ASSERT(offsetTable != NULL); + + /* For details see ISO/IEC 13818-7:2005(E), 8.5.3 Channel configuration */ + FDKmemclear(numChInGrp, PCM_DMX_MAX_CHANNEL_GROUPS*sizeof(UCHAR)); + FDKmemset(offsetTable, 255, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR)); + + /* Categorize channels */ + for (ch = 0; ch < numChannels; ch += 1) { + int i = 0, j, chGrpIdx = channelIndices[ch]; + + switch (channelType[ch]) { + case ACT_FRONT: + case ACT_FRONT_TOP: + grpIdx = CH_GROUP_FRONT; + break; + case ACT_SIDE: + case ACT_SIDE_TOP: + grpIdx = CH_GROUP_SIDE; + break; + case ACT_BACK: + case ACT_BACK_TOP: + grpIdx = CH_GROUP_REAR; + break; + case ACT_LFE: + grpIdx = CH_GROUP_LFE; + break; + default: + err = -1; + continue; + } + + if (numChInGrp[grpIdx] < PCM_DMX_MAX_CHANNELS_PER_GROUP) { + /* Sort channels by index */ + while ( (i < numChInGrp[grpIdx]) && (chGrpIdx > channelIndices[chIdx[grpIdx][i]]) ) { + i += 1; + } + for (j = numChInGrp[grpIdx]; j > i; j -= 1) { + chIdx[grpIdx][j] = chIdx[grpIdx][j-1]; + } + chIdx[grpIdx][i] = ch; + numChInGrp[grpIdx] += 1; + } + } + + /* Compose channel offset table */ + + /* Non-symmetric channels */ + if (numChInGrp[CH_GROUP_FRONT] & 0x1) { + /* Odd number of front channels -> we have a center channel. + In MPEG-4 the center has the index 0. */ + offsetTable[CENTER_FRONT_CHANNEL] = channelMapping[chIdx[CH_GROUP_FRONT][0]]; + } + + for (grpIdx = 0; grpIdx < PCM_DMX_MAX_CHANNEL_GROUPS; grpIdx += 1) { + int chMapPos, maxChannels = 0; + ch = 0; + + switch (grpIdx) { + case CH_GROUP_FRONT: + chMapPos = LEFT_FRONT_CHANNEL; + maxChannels = 3; + ch = numChInGrp[grpIdx] & 0x1; + break; + case CH_GROUP_SIDE: + chMapPos = LEFT_OUTSIDE_CHANNEL; + maxChannels = 2; + break; + case CH_GROUP_REAR: + chMapPos = LEFT_REAR_CHANNEL; + maxChannels = 2; + break; + case CH_GROUP_LFE: + chMapPos = LOW_FREQUENCY_CHANNEL; + maxChannels = 1; + break; + default: + err = -1; + continue; + } + + for ( ; ch < numChInGrp[grpIdx]; ch += 1) { + if (ch < maxChannels) { + offsetTable[chMapPos] = channelMapping[chIdx[grpIdx][ch]]; + chMapPos += 1; + } else { + err = -1; + } + } + } + + if (err == 0) { + /* Compose the channel mode */ + chMode = (numChInGrp[CH_GROUP_LFE] & 0xF) << 12 + | (numChInGrp[CH_GROUP_REAR] & 0xF) << 8 + | (numChInGrp[CH_GROUP_SIDE] & 0xF) << 4 + | (numChInGrp[CH_GROUP_FRONT] & 0xF); + } + + return (PCM_DMX_CHANNEL_MODE)chMode; +} + + +/** Generate a channel offset table and complete channel description for a given (packed) channel mode. + * This function is the inverse to the getChannelMode() routine. + * @param [in] The total number of channels of the given configuration. + * @param [in] Array containing the channel mapping to be used (From MPEG PCE ordering to whatever is required). + * @param [out] Array where corresponding channel types for each channels are stored into. + * @param [out] Array where corresponding channel type indices for each output channel are stored into. + * @param [out] Array where the buffer offsets for each channel are stored into. + * @returns None. + **/ +void getChannelDescription ( + const PCM_DMX_CHANNEL_MODE chMode, /* in */ + const UCHAR channelMapping[][PCM_DMX_MAX_CHANNELS], /* in */ + AUDIO_CHANNEL_TYPE channelType[], /* out */ + UCHAR channelIndices[], /* out */ + UCHAR offsetTable[PCM_DMX_MAX_CHANNELS] /* out */ + ) +{ + const UCHAR *pChannelMap; + int grpIdx, ch = 0, numChannels = 0; + UCHAR numChInGrp[PCM_DMX_MAX_CHANNEL_GROUPS]; + + FDK_ASSERT(channelType != NULL); + FDK_ASSERT(channelIndices != NULL); + FDK_ASSERT(channelMapping != NULL); + FDK_ASSERT(offsetTable != NULL); + + /* Init output arrays */ + FDKmemclear(channelType, PCM_DMX_MAX_CHANNELS*sizeof(AUDIO_CHANNEL_TYPE)); + FDKmemclear(channelIndices, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR)); + FDKmemset(offsetTable, 255, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR)); + + /* Extract the number of channels per group */ + numChInGrp[CH_GROUP_FRONT] = chMode & 0xF; + numChInGrp[CH_GROUP_SIDE] = (chMode >> 4) & 0xF; + numChInGrp[CH_GROUP_REAR] = (chMode >> 8) & 0xF; + numChInGrp[CH_GROUP_LFE] = (chMode >> 12) & 0xF; + + /* Summerize to get the total number of channels */ + for (grpIdx = 0; grpIdx < PCM_DMX_MAX_CHANNEL_GROUPS; grpIdx += 1) { + numChannels += numChInGrp[grpIdx]; + } + + /* Get the appropriate channel map */ + pChannelMap = channelMapping[numChannels-1]; + + /* Compose channel offset table */ + + /* Non-symmetric channels */ + if (numChInGrp[CH_GROUP_FRONT] & 0x1) { + /* Odd number of front channels -> we have a center channel. + In MPEG-4 the center has the index 0. */ + offsetTable[CENTER_FRONT_CHANNEL] = pChannelMap[0]; + channelType[0] = ACT_FRONT; + ch += 1; + } + + for (grpIdx = 0; grpIdx < PCM_DMX_MAX_CHANNEL_GROUPS; grpIdx += 1) { + AUDIO_CHANNEL_TYPE type; + int chMapPos, maxChannels = 0; + int chIdx = 0; + + switch (grpIdx) { + case CH_GROUP_FRONT: + type = ACT_FRONT; + chMapPos = LEFT_FRONT_CHANNEL; + maxChannels = 3; + chIdx = numChInGrp[grpIdx] & 0x1; + break; + case CH_GROUP_SIDE: + type = ACT_SIDE; + chMapPos = LEFT_OUTSIDE_CHANNEL; + maxChannels = 2; + break; + case CH_GROUP_REAR: + type = ACT_BACK; + chMapPos = LEFT_REAR_CHANNEL; + maxChannels = 2; + break; + case CH_GROUP_LFE: + type = ACT_LFE; + chMapPos = LOW_FREQUENCY_CHANNEL; + maxChannels = 1; + break; + default: + break; + } + + for ( ; (chIdx < numChInGrp[grpIdx]) && (chIdx < maxChannels); chIdx += 1) { + offsetTable[chMapPos] = pChannelMap[ch]; + channelType[ch] = type; + channelIndices[ch] = chIdx; + chMapPos += 1; + ch += 1; + } + } +} + + +/** Open and initialize an instance of the PCM downmix module + * @param [out] Pointer to a buffer receiving the handle of the new instance. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_Open ( + HANDLE_PCM_DOWNMIX *pSelf + ) +{ + HANDLE_PCM_DOWNMIX self; + + if (pSelf == NULL) { + return (PCMDMX_INVALID_HANDLE); + } + + *pSelf = NULL; + + self = (HANDLE_PCM_DOWNMIX) GetPcmDmxInstance( 0 ); + if (self == NULL) { + return (PCMDMX_OUT_OF_MEMORY); + } + + /* Reset the full instance */ + pcmDmx_Reset( self, PCMDMX_RESET_FULL ); + + *pSelf = self; + + return (PCMDMX_OK); +} + + +/** Reset all static values like e.g. mixdown coefficients. + * @param [in] Handle of PCM downmix module instance. + * @param [in] Flags telling which parts of the module shall be reset. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_Reset ( + HANDLE_PCM_DOWNMIX self, + UINT flags + ) +{ + if (self == NULL) { return (PCMDMX_INVALID_HANDLE); } + + if (flags & PCMDMX_RESET_PARAMS) { + self->dualChannelMode = STEREO_MODE; + self->numOutputChannels = 0; + self->applyProcessing = 0; + self->frameDelay = 0; + self->expiryFrame = PCMDMX_DFLT_EXPIRY_FRAME; + } + + if (flags & PCMDMX_RESET_BS_DATA) { + int slot; + for (slot = 0; slot <= PCM_DMX_MAX_DELAY_FRAMES; slot += 1) { + self->dvbMixDownLevels[slot].centerMixLevelValue = dvbDownmixFactors[2]; /* 0.707 */ + self->dvbMixDownLevels[slot].surroundMixLevelValue = dvbDownmixFactors[0]; /* 1.000 */ + self->dvbMixDownLevels[slot].mixLevelsAvail = 0; + + self->mpegMixDownInfo[slot].mixdownAvailable = 0; + } + /* Reset expiry counter */ + self->expiryCount = 0; + } + + return (PCMDMX_OK); +} + + +/** Set one parameter for one instance of the PCM downmix module. + * @param [in] Handle of PCM downmix module instance. + * @param [in] Parameter to be set. + * @param [in] Parameter value. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_SetParam ( + HANDLE_PCM_DOWNMIX self, + PCMDMX_PARAM param, + UINT value + ) +{ + switch (param) + { + case DMX_BS_DATA_EXPIRY_FRAME: + if (self == NULL) + return (PCMDMX_INVALID_HANDLE); + self->expiryFrame = value; + break; + + case DMX_BS_DATA_DELAY: + if (value > PCM_DMX_MAX_DELAY_FRAMES) { + return (PCMDMX_UNABLE_TO_SET_PARAM); + } + if (self == NULL) { + return (PCMDMX_INVALID_HANDLE); + } + self->frameDelay = value; + break; + + case NUMBER_OF_OUTPUT_CHANNELS: + switch ((int)value) { /* supported output channels */ + case -1: case 0: case 1: case 2: + case 6: case 8: + break; + default: + return (PCMDMX_UNABLE_TO_SET_PARAM); + } + if (self == NULL) + return (PCMDMX_INVALID_HANDLE); + if ((int)value > 0) { + self->numOutputChannels = (int)value; + self->applyProcessing = 1; + } else { + self->numOutputChannels = 0; + self->applyProcessing = 0; + } + break; + + case DUAL_CHANNEL_DOWNMIX_MODE: + switch ((DUAL_CHANNEL_MODE)value) { + case STEREO_MODE: + case CH1_MODE: + case CH2_MODE: + case MIXED_MODE: + break; + default: + return (PCMDMX_UNABLE_TO_SET_PARAM); + } + if (self == NULL) + return (PCMDMX_INVALID_HANDLE); + self->dualChannelMode = (DUAL_CHANNEL_MODE)value; + self->applyProcessing = 1; + break; + + default: + return (PCMDMX_UNKNOWN_PARAM); + } + + return (PCMDMX_OK); +} + + +/** Read the ancillary data transported in DSEs of DVB streams with MPEG-4 content + * @param [in] Handle of PCM downmix module instance. + * @param [in] Pointer to ancillary data buffer. + * @param [in] Size of ancillary data. + * @param [in] Flag indicating wheter the DVB ancillary data is from an MPEG-1/2 or an MPEG-4 stream. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_ReadDvbAncData ( + HANDLE_PCM_DOWNMIX self, + UCHAR *pAncDataBuf, + UINT ancDataBytes, + int isMpeg2 + ) +{ + DVB_MIXDOWN_LEVELS *pDownmixLevels = &self->dvbMixDownLevels[0]; + + int offset = (isMpeg2) ? 2 : 0; + UCHAR ancDataStatus; + + if (self == NULL) { return (PCMDMX_INVALID_HANDLE); } + + /* sanity checks */ + if (pAncDataBuf == NULL || ancDataBytes < (UCHAR)(3+offset)) { + return (PCMDMX_CORRUPT_ANC_DATA); + } + + /* check sync word */ + if (pAncDataBuf[offset] != ANC_DATA_SYNC_BYTE) { + return (PCMDMX_CORRUPT_ANC_DATA); + } + + offset += 2; + ancDataStatus = pAncDataBuf[offset++]; + + if (isMpeg2) { + /* skip advanced_dynamic_range_control */ + if (ancDataStatus & 0x80) offset += 3; + /* skip dialog_normalization */ + if (ancDataStatus & 0x40) offset += 1; + /* skip reproduction_level */ + if (ancDataStatus & 0x20) offset += 1; + } + else { + /* check reserved bits */ + if (ancDataStatus & 0xE8) { return (PCMDMX_CORRUPT_ANC_DATA); } + } + + /* downmix_levels_MPEGX */ + if (ancDataStatus & 0x10) + { + int foundNewData = 0; + UCHAR downmixData = pAncDataBuf[offset++]; + + if (downmixData & 0x80) { /* center_mix_level_on */ + pDownmixLevels->centerMixLevelValue = + dvbDownmixFactors[(downmixData >> 4) & 0x07]; + foundNewData = 1; + } else { + pDownmixLevels->centerMixLevelValue = dvbDownmixFactors[0]; + if (downmixData & 0x70) { return (PCMDMX_CORRUPT_ANC_DATA); } + } + + if (downmixData & 0x08) { /* surround_mix_level_on */ + pDownmixLevels->surroundMixLevelValue = + dvbDownmixFactors[downmixData & 0x07]; + foundNewData = 1; + } else { + pDownmixLevels->surroundMixLevelValue = dvbDownmixFactors[0]; + if (downmixData & 0x07) { return (PCMDMX_CORRUPT_ANC_DATA); } + } + + pDownmixLevels->mixLevelsAvail = foundNewData; + } + + /* Reset expiry counter */ + self->expiryCount = 0; + + return (PCMDMX_OK); +} + +/** Set the matrix mixdown information extracted from the PCE of an AAC bitstream. + * Note: Call only if matrix_mixdown_idx_present is true. + * @param [in] Handle of PCM downmix module instance. + * @param [in] The 2 bit matrix mixdown index extracted from PCE. + * @param [in] The pseudo surround enable flag extracted from PCE. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_SetMatrixMixdownFromPce ( + HANDLE_PCM_DOWNMIX self, + int matrixMixdownPresent, + int matrixMixdownIdx, + int pseudoSurroundEnable + ) +{ + MPEG_MIXDOWN_INFO *pMpegMixDownInfo; + + if (self == NULL) { + return (PCMDMX_INVALID_HANDLE); + } + + pMpegMixDownInfo = &self->mpegMixDownInfo[0]; + + if (matrixMixdownPresent) { + pMpegMixDownInfo->matrixMixdownIdx = matrixMixdownIdx & 0x03; + pMpegMixDownInfo->pseudoSurroundEnable = pseudoSurroundEnable; + } + + pMpegMixDownInfo->mixdownAvailable = matrixMixdownPresent; + /* Reset expiry counter */ + self->expiryCount = 0; + + return (PCMDMX_OK); +} + + +/** Apply down or up mixing. + * @param [in] Handle of PCM downmix module instance. + * @param [inout] Pointer to time buffer. Depending on interface configuration, the content of pTimeData is ignored, + * and the internal QMF buffer will be used as input data source. Otherwise, the MPEG Surround processing is + * applied to the timesignal pTimeData. For both variants, the resulting MPEG Surround signal is written into pTimeData. + * @param [in] Pointer where the amount of output samples is returned into. + * @param [inout] Pointer where the amount of output channels is returned into. + * @param [in] Flag which indicates if output time data are writtern interleaved or as subsequent blocks. + * @param [inout] Array where the corresponding channel type for each output audio channel is stored into. + * @param [inout] Array where the corresponding channel type index for each output audio channel is stored into. + * @param [in] Array containing the output channel mapping to be used (From MPEG PCE ordering to whatever is required). + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_ApplyFrame ( + HANDLE_PCM_DOWNMIX self, + INT_PCM *pPcmBuf, + UINT frameSize, + INT *nChannels, + + int fInterleaved, + AUDIO_CHANNEL_TYPE channelType[], + UCHAR channelIndices[], + const UCHAR channelMapping[][8] + ) +{ + PCMDMX_ERROR errorStatus = PCMDMX_OK; + DUAL_CHANNEL_MODE dualChannelMode; + PCM_DMX_CHANNEL_MODE inChMode; + int numOutChannels; + int numInChannels = *nChannels; + int slot; + UCHAR inOffsetTable[PCM_DMX_MAX_CHANNELS]; + + MPEG_MIXDOWN_INFO mpegMixDownInfo; + DVB_MIXDOWN_LEVELS dvbMixDownLevels; + + if (self == NULL) { return (PCMDMX_INVALID_HANDLE); } + + if ( (self->expiryFrame > 0) + && (++self->expiryCount > self->expiryFrame) ) + { /* The metadata read from bitstream is too old. */ + errorStatus = pcmDmx_Reset(self, PCMDMX_RESET_BS_DATA); + } + + FDKmemcpy(&mpegMixDownInfo, &self->mpegMixDownInfo[self->frameDelay], sizeof(MPEG_MIXDOWN_INFO)); + /* Maintain delay line */ + for (slot = self->frameDelay; slot > 0; slot -= 1) { + FDKmemcpy(&self->mpegMixDownInfo[slot], &self->mpegMixDownInfo[slot-1], sizeof(MPEG_MIXDOWN_INFO)); + } + FDKmemcpy(&dvbMixDownLevels, &self->dvbMixDownLevels[self->frameDelay], sizeof(DVB_MIXDOWN_LEVELS)); + /* Maintain delay line */ + for (slot = self->frameDelay; slot > 0; slot -= 1) { + FDKmemcpy(&self->dvbMixDownLevels[slot], &self->dvbMixDownLevels[slot-1], sizeof(DVB_MIXDOWN_LEVELS)); + } + + if (self->applyProcessing == 0) { return (errorStatus); } + + if (pPcmBuf == NULL) { return (PCMDMX_INVALID_ARGUMENT); } + if (frameSize == 0) { return (PCMDMX_INVALID_ARGUMENT); } + if (numInChannels == 0) { return (PCMDMX_INVALID_ARGUMENT); } + + if (self->numOutputChannels <= 0) { + numOutChannels = numInChannels; + } else { + numOutChannels = self->numOutputChannels; + } + dualChannelMode = self->dualChannelMode; + + /* Analyse input channel configuration and get channel offset + * table that can be accessed with the fixed channel labels. */ + inChMode = getChannelMode( + numInChannels, + channelType, + channelIndices, + channelMapping[numInChannels], + inOffsetTable + ); + if (inChMode == CH_MODE_UNDEFINED) { + /* We don't need to restore because the channel + configuration has not been changed. Just exit. */ + return (PCMDMX_INVALID_CH_CONFIG); + } + + /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - */ + if ( numInChannels > numOutChannels ) + { /* Apply downmix */ + INT_PCM *pInCF, *pInLF, *pInRF, *pInLO, *pInRO, *pInLR, *pInRR, *pOutL, *pOutR; + FIXP_SGL flev, clev, slev; + + UINT sample; + int inStride, outStride, offset; + int useGuidedDownMix = 0; + UCHAR outOffsetTable[PCM_DMX_MAX_CHANNELS]; + + /* Set I/O strides and offsets */ + if (fInterleaved) { + inStride = numInChannels; + outStride = TWO_CHANNEL; /* The output of STAGE ONE is always STEREO !!! + STAGE TWO creates a downmix to mono if required. */ + offset = 1; /* Channel specific offset factor */ + } else { + inStride = 1; + outStride = 1; + offset = frameSize; /* Channel specific offset factor */ + } + + /* Get channel description and channel mapping for this + * stages number of output channels (always STEREO). */ + getChannelDescription( + CH_MODE_2_0_0_0, + channelMapping, + channelType, + channelIndices, + outOffsetTable + ); + /* Now there is no way back because we modified the channel configuration! */ + + /* Set channel pointer for input */ + pInCF = &pPcmBuf[inOffsetTable[CENTER_FRONT_CHANNEL]*offset]; + pInLF = &pPcmBuf[inOffsetTable[LEFT_FRONT_CHANNEL]*offset]; + pInRF = &pPcmBuf[inOffsetTable[RIGHT_FRONT_CHANNEL]*offset]; + pInLO = &pPcmBuf[inOffsetTable[LEFT_OUTSIDE_CHANNEL]*offset]; + pInRO = &pPcmBuf[inOffsetTable[RIGHT_OUTSIDE_CHANNEL]*offset]; + pInLR = &pPcmBuf[inOffsetTable[LEFT_REAR_CHANNEL]*offset]; + pInRR = &pPcmBuf[inOffsetTable[RIGHT_REAR_CHANNEL]*offset]; + + /* Set channel pointer for output + Caution: Different channel mapping compared to input */ + pOutL = &pPcmBuf[outOffsetTable[LEFT_FRONT_CHANNEL]*offset]; /* LEFT_FRONT_CHANNEL */ + pOutR = &pPcmBuf[outOffsetTable[RIGHT_FRONT_CHANNEL]*offset]; /* RIGHT_FRONT_CHANNEL */ + + /* Set downmix levels: */ + flev = ATTENUATION_FACTOR_1; /* 0.707 */ + clev = ATTENUATION_FACTOR_1; /* 0.707 */ + slev = ATTENUATION_FACTOR_1; /* 0.707 */ + + if ( dvbMixDownLevels.mixLevelsAvail ) { + clev = dvbMixDownLevels.centerMixLevelValue; + slev = dvbMixDownLevels.surroundMixLevelValue; + useGuidedDownMix = 1; + } + + /* FIRST STAGE: + Always downmix to 2 channel output: */ + switch ( inChMode ) + { + case CH_MODE_2_0_0_0: + case CH_MODE_2_0_0_1: + /* 2/0 input: */ + switch (dualChannelMode) + { + case CH1_MODE: /* L' = 0.707 * Ch1; R' = 0.707 * Ch1 */ + for (sample = 0; sample < frameSize; sample++) { + *pOutL = *pOutR = + (INT_PCM)SATURATE_RIGHT_SHIFT(fMult((FIXP_PCM)*pInLF, flev), DFRACT_BITS-SAMPLE_BITS, SAMPLE_BITS); + + pInLF += inStride; + pOutL += outStride; pOutR += outStride; + } + break; + + case CH2_MODE: /* L' = 0.707 * Ch2; R' = 0.707 * Ch2 */ + for (sample = 0; sample < frameSize; sample++) { + *pOutL = *pOutR = + (INT_PCM)SATURATE_RIGHT_SHIFT(fMult((FIXP_PCM)*pInRF, flev), DFRACT_BITS-SAMPLE_BITS, SAMPLE_BITS); + + pInRF += inStride; + pOutL += outStride; pOutR += outStride; + } + break; + case MIXED_MODE: /* L' = 0.5*Ch1 + 0.5*Ch2; R' = 0.5*Ch1 + 0.5*Ch2 */ + for (sample = 0; sample < frameSize; sample++) { + *pOutL = *pOutR = (*pInLF >> 1) + (*pInRF >> 1); + + pInLF += inStride; pInRF += inStride; + pOutL += outStride; pOutR += outStride; + } + break; + default: + case STEREO_MODE: + /* nothing to do */ + break; + } + break; + + case CH_MODE_3_0_0_0: + /* 3/0 input: L' = L + 0.707*C; R' = R + 0.707*C; */ + for (sample = 0; sample < frameSize; sample++) + { + FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev); +#if (SAMPLE_BITS == 32) + /* left channel */ + *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>1)+tCF, 1, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>1)+tCF, 1, SAMPLE_BITS); +#else + /* left channel */ + *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>1)+tCF, DFRACT_BITS-SAMPLE_BITS-1, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>1)+tCF, DFRACT_BITS-SAMPLE_BITS-1, SAMPLE_BITS); +#endif + pInLF += inStride; pInRF += inStride; pInCF += inStride; + pOutL += outStride; pOutR += outStride; + } + break; + + /* 2/1 input: not supported! + case CH_MODE_2_0_1_0: */ + + case CH_MODE_3_0_1_0: + if (useGuidedDownMix) { + /* 3/1 input: L' = L + clev*C + 0.707*slev*S; R' = R + clev*C + 0.707*slev*S; */ + slev = FX_DBL2FX_SGL(fMult(flev, slev)); /* 0.707*slef */ + + for (sample = 0; sample < frameSize; sample++) + { + FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev) >> 1; + FIXP_DBL tLR = fMultDiv2((FIXP_PCM)*pInLR, slev) >> 1; +#if (SAMPLE_BITS == 32) + /* left channel */ + *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF+tLR, 2, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tLR, 2, SAMPLE_BITS); +#else + /* left channel */ + *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF-tLR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tLR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS); +#endif + pInLF += inStride; pInRF += inStride; pInCF += inStride; pInLR += inStride; + pOutL += outStride; pOutR += outStride; + } + } else { + /* 3/1 input: L' = L + 0.707*C - 0.707*S; R' = R + 0.707*C + 0.707*S */ + for (sample = 0; sample < frameSize; sample++) + { + FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev) >> 1; + FIXP_DBL tLR = fMultDiv2((FIXP_PCM)*pInLR, slev) >> 1; +#if (SAMPLE_BITS == 32) + /* left channel */ + *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF-tLR, 2, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tLR, 2, SAMPLE_BITS); +#else + /* left channel */ + *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF-tLR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tLR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS); +#endif + pInLF += inStride; pInRF += inStride; pInCF += inStride; pInLR += inStride; + pOutL += outStride; pOutR += outStride; + } + } + break; + + /* 2/2 input: not supported! + case CH_MODE_2_0_2_0: */ + + case CH_MODE_3_0_2_0: /* 5.0ch input */ + case CH_MODE_3_0_2_1: /* 5.1ch input */ + if (useGuidedDownMix) { + /* 3/2 input: L' = L + clev*C + slev*Ls; R' = R + clev*C + slev*Rs; */ + for (sample = 0; sample < frameSize; sample++) + { + FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev) >> 1; + FIXP_DBL tLR = fMultDiv2((FIXP_PCM)*pInLR, slev) >> 1; + FIXP_DBL tRR = fMultDiv2((FIXP_PCM)*pInRR, slev) >> 1; +#if (SAMPLE_BITS == 32) + /* left channel */ + *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF+tLR, 2, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tRR, 2, SAMPLE_BITS); +#else + /* left channel */ + *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF+tLR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tRR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS); +#endif + pInLF += inStride; pInRF += inStride; pInCF += inStride; pInLR += inStride; pInRR += inStride; + pOutL += outStride; pOutR += outStride; + } + } + else if (mpegMixDownInfo.mixdownAvailable) { + /* 3/2 input: L' = (1.707+A)^-1 * [L+0.707*C+A*Ls]; R'= (1.707+A)^-1 * [R+0.707*C+A*Rs]; */ + FIXP_SGL mtrxMixDwnCoef = mpegMixDownIdx2Coef[mpegMixDownInfo.matrixMixdownIdx]; + FIXP_SGL mtrxMixDwnPreFact = mpegMixDownIdx2PreFact[mpegMixDownInfo.matrixMixdownIdx]; + clev = FX_DBL2FX_SGL(fMult(mtrxMixDwnPreFact, flev /* 0.707 */)); + flev = mtrxMixDwnPreFact; + slev = FX_DBL2FX_SGL(fMult(mtrxMixDwnPreFact, mtrxMixDwnCoef)); + + for (sample = 0; sample < frameSize; sample++) + { + FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev); + FIXP_DBL tLF = fMultDiv2((FIXP_PCM)*pInLF, flev); + FIXP_DBL tRF = fMultDiv2((FIXP_PCM)*pInRF, flev); + FIXP_DBL tLR = fMultDiv2((FIXP_PCM)*pInLR, slev); + FIXP_DBL tRR = fMultDiv2((FIXP_PCM)*pInRR, slev); + +#if (SAMPLE_BITS == 32) + /* left channel */ + *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT(tLF+tCF+tLR, 1, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT(tRF+tCF+tRR, 1, SAMPLE_BITS); +#else + /* left channel */ + *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT(tLF+tCF+tLR, DFRACT_BITS-SAMPLE_BITS-1, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT(tRF+tCF+tRR, DFRACT_BITS-SAMPLE_BITS-1, SAMPLE_BITS); +#endif + + pInLF += inStride; pInRF += inStride; pInCF += inStride; pInLR += inStride; pInRR += inStride; + pOutL += outStride; pOutR += outStride; + } + } + else { + /* 3/2 input: L' = L + 0.707*C - 0.707*Ls - 0.707*Rs; R' = R + 0.707*C + 0.707*Ls + 0.707*Rs */ + for (sample = 0; sample < frameSize; sample++) + { + FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev) >> 2; + FIXP_DBL tLR = fMultDiv2((FIXP_PCM)*pInLR, slev) >> 2; + FIXP_DBL tRR = fMultDiv2((FIXP_PCM)*pInRR, slev) >> 2; +#if (SAMPLE_BITS == 32) + /* left channel */ + *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>3)+tCF-tLR-tRR, 3, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>3)+tCF+tLR+tRR, 3, SAMPLE_BITS); +#else + /* left channel */ + *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>3)+tCF-tLR-tRR, DFRACT_BITS-SAMPLE_BITS-3, SAMPLE_BITS); + /* right channel */ + *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>3)+tCF+tLR+tRR, DFRACT_BITS-SAMPLE_BITS-3, SAMPLE_BITS); +#endif + pInLF += inStride; pInRF += inStride; pInCF += inStride; pInLR += inStride; pInRR += inStride; + pOutL += outStride; pOutR += outStride; + } + } + break; + + default: + errorStatus = PCMDMX_INVALID_MODE; + break; + } + + /* SECOND STAGE: + If desired create a mono donwmix: + Note: Input are always two channels! */ + if (numOutChannels == 1) + { + INT_PCM *pOutC; + FIXP_SGL mlev; + + if (useGuidedDownMix) mlev = FL2FXCONST_SGL(1.0f); else mlev = flev; + + /* Output of STAGE ONE = Input of STAGE TWO */ + FDKmemcpy(inOffsetTable, outOffsetTable, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR)); + + /* Set I/O strides and offsets */ + inStride = outStride; /* output from STAGE ONE */ + outStride = numOutChannels; /* final output */ + + /* Get channel description and channel mapping for this + * stages number of output channels (always MONO). */ + getChannelDescription( + CH_MODE_1_0_0_0, + channelMapping, + channelType, + channelIndices, + outOffsetTable + ); + + /* Set input channel pointer. */ + pInLF = &pPcmBuf[inOffsetTable[LEFT_FRONT_CHANNEL]*offset]; + pInRF = &pPcmBuf[inOffsetTable[RIGHT_FRONT_CHANNEL]*offset]; + + /* Set output channel pointer */ + pOutC = &pPcmBuf[outOffsetTable[CENTER_FRONT_CHANNEL]*offset]; + + /* C' = 0.707*L + 0.707*R */ + for (sample = 0; sample < frameSize; sample++) { +#if (SAMPLE_BITS == 32) + *pOutC = + (INT_PCM)SATURATE_LEFT_SHIFT(fMultDiv2((FIXP_PCM)*pInLF,mlev)+fMultDiv2((FIXP_PCM)*pInRF,mlev), 1, SAMPLE_BITS); +#else + *pOutC = + (INT_PCM)SATURATE_RIGHT_SHIFT(fMultDiv2((FIXP_PCM)*pInLF,mlev)+fMultDiv2((FIXP_PCM)*pInRF,mlev), DFRACT_BITS-SAMPLE_BITS-1, SAMPLE_BITS); +#endif + + pInLF += inStride; pInRF += inStride; + pOutC += 1; + } + /* Finished STAGE TWO */ + } + + /* Update the number of output channels */ + *nChannels = self->numOutputChannels; + + } /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - */ + else + if ( numInChannels == numOutChannels ) + { + /* Don't need to change the channel description here */ + + switch (numInChannels) + { + case 2: + { /* Set up channel pointer */ + INT_PCM *pInLF, *pInRF, *pOutL, *pOutR; + FIXP_SGL flev; + + UINT sample; + int inStride, outStride, offset; + + if (fInterleaved) { + inStride = numInChannels; + outStride = 2; /* fixed !!! (below stereo is donwmixed to mono if required */ + offset = 1; /* Channel specific offset factor */ + } else { + inStride = 1; + outStride = 1; + offset = frameSize; /* Channel specific offset factor */ + } + + /* Set input channel pointer */ + pInLF = &pPcmBuf[inOffsetTable[LEFT_FRONT_CHANNEL]*offset]; + pInRF = &pPcmBuf[inOffsetTable[RIGHT_FRONT_CHANNEL]*offset]; + + /* Set output channel pointer (same as input) */ + pOutL = pInLF; + pOutR = pInRF; + + /* Set downmix levels: */ + flev = ATTENUATION_FACTOR_1; /* 0.707 */ + /* 2/0 input: */ + switch (dualChannelMode) + { + case CH1_MODE: /* L' = 0.707 * Ch1; R' = 0.707 * Ch1 */ + for (sample = 0; sample < frameSize; sample++) { + *pOutL = *pOutR = + (INT_PCM)SATURATE_RIGHT_SHIFT(fMult((FIXP_PCM)*pInLF, flev), DFRACT_BITS-SAMPLE_BITS, SAMPLE_BITS); + + pInLF += inStride; + pOutL += outStride; pOutR += outStride; + } + break; + case CH2_MODE: /* L' = 0.707 * Ch2; R' = 0.707 * Ch2 */ + for (sample = 0; sample < frameSize; sample++) { + *pOutL = *pOutR = + (INT_PCM)SATURATE_RIGHT_SHIFT(fMult((FIXP_PCM)*pInRF, flev), DFRACT_BITS-SAMPLE_BITS, SAMPLE_BITS); + + pInRF += inStride; + pOutL += outStride; pOutR += outStride; + } + break; + case MIXED_MODE: /* L' = 0.5*Ch1 + 0.5*Ch2; R' = 0.5*Ch1 + 0.5*Ch2 */ + for (sample = 0; sample < frameSize; sample++) { + *pOutL = *pOutR = (*pInLF >> 1) + (*pInRF >> 1); + + pInLF += inStride; pInRF += inStride; + pOutL += outStride; pOutR += outStride; + } + break; + default: + case STEREO_MODE: + /* nothing to do */ + break; + } + } + break; + + default: + /* nothing to do */ + break; + } + } + + return (errorStatus); +} + + +/** Close an instance of the PCM downmix module. + * @param [inout] Pointer to a buffer containing the handle of the instance. + * @returns Returns an error code. + **/ +PCMDMX_ERROR pcmDmx_Close ( + HANDLE_PCM_DOWNMIX *pSelf + ) +{ + if (pSelf == NULL) { + return (PCMDMX_INVALID_HANDLE); + } + + FreePcmDmxInstance( pSelf ); + *pSelf = NULL; + + return (PCMDMX_OK); +} + + +/** Get library info for this module. + * @param [out] Pointer to an allocated LIB_INFO structure. + * @returns Returns an error code. + */ +PCMDMX_ERROR pcmDmx_GetLibInfo( LIB_INFO *info ) +{ + int i; + + if (info == NULL) { + return PCMDMX_INVALID_ARGUMENT; + } + + /* Search for next free tab */ + for (i = 0; i < FDK_MODULE_LAST; i++) { + if (info[i].module_id == FDK_NONE) break; + } + if (i == FDK_MODULE_LAST) { + return PCMDMX_UNKNOWN; + } + info += i; + + /* Add the library info */ + info->module_id = FDK_PCMDMX; + info->version = LIB_VERSION(PCMDMX_LIB_VL0, PCMDMX_LIB_VL1, PCMDMX_LIB_VL2); + LIB_VERSION_STRING(info); + info->build_date = PCMDMX_LIB_BUILD_DATE; + info->build_time = PCMDMX_LIB_BUILD_TIME; + info->title = PCMDMX_LIB_TITLE; + + /* Set flags */ + info->flags = 0 + | CAPF_DMX_BLIND /* At least blind downmixing is possible */ + | CAPF_DMX_PCE /* Guided downmix with data from MPEG-2/4 Program Config Elements (PCE). */ + | CAPF_DMX_DVB /* Guided downmix with data from DVB ancillary data fields. */ + ; + + return PCMDMX_OK; +} + + + |