diff options
Diffstat (limited to 'libPCMutils/include')
| -rw-r--r-- | libPCMutils/include/limiter.h | 342 | ||||
| -rw-r--r-- | libPCMutils/include/pcm_utils.h | 131 | ||||
| -rw-r--r-- | libPCMutils/include/pcmdmx_lib.h | 460 | ||||
| -rw-r--r-- | libPCMutils/include/pcmutils_lib.h | 334 |
4 files changed, 786 insertions, 481 deletions
diff --git a/libPCMutils/include/limiter.h b/libPCMutils/include/limiter.h index 0d3d701..fab7226 100644 --- a/libPCMutils/include/limiter.h +++ b/libPCMutils/include/limiter.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 @@ -79,34 +90,53 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ -/************************ FDK PCM postprocessor module ********************* +/**************************** PCM utility library ****************************** Author(s): Matthias Neusinger + Description: Hard limiter for clipping prevention *******************************************************************************/ -#ifndef _LIMITER_H_ -#define _LIMITER_H_ - +#ifndef LIMITER_H +#define LIMITER_H #include "common_fix.h" +#include "FDK_audio.h" -#define TDL_ATTACK_DEFAULT_MS (15) /* default attack time in ms */ -#define TDL_RELEASE_DEFAULT_MS (50) /* default release time in ms */ - -#define TDL_GAIN_SCALING (15) /* scaling of gain value. */ +#define TDL_ATTACK_DEFAULT_MS (15) /* default attack time in ms */ +#define TDL_RELEASE_DEFAULT_MS (50) /* default release time in ms */ +#define TDL_GAIN_SCALING (15) /* scaling of gain value. */ #ifdef __cplusplus extern "C" { #endif +struct TDLimiter { + unsigned int attack; + FIXP_DBL attackConst, releaseConst; + unsigned int attackMs, releaseMs, maxAttackMs; + FIXP_DBL threshold; + unsigned int channels, maxChannels; + UINT sampleRate, maxSampleRate; + FIXP_DBL cor, max; + FIXP_DBL* maxBuf; + FIXP_DBL* delayBuf; + unsigned int maxBufIdx, delayBufIdx; + FIXP_DBL smoothState0; + FIXP_DBL minGain; + + FIXP_DBL additionalGainPrev; + FIXP_DBL additionalGainFilterState; + FIXP_DBL additionalGainFilterState1; +}; typedef enum { TDLIMIT_OK = 0, + TDLIMIT_UNKNOWN = -1, __error_codes_start = -100, @@ -119,115 +149,133 @@ typedef enum { struct TDLimiter; typedef struct TDLimiter* TDLimiterPtr; -/****************************************************************************** -* createLimiter * -* maxAttackMs: maximum and initial attack/lookahead time in milliseconds * -* releaseMs: release time in milliseconds (90% time constant) * -* threshold: limiting threshold * -* maxChannels: maximum and initial number of channels * -* maxSampleRate: maximum and initial sampling rate in Hz * -* returns: limiter handle * -******************************************************************************/ -TDLimiterPtr createLimiter(unsigned int maxAttackMs, - unsigned int releaseMs, - INT_PCM threshold, - unsigned int maxChannels, - unsigned int maxSampleRate); +#define PCM_LIM LONG +#define FIXP_DBL2PCM_LIM(x) (x) +#define PCM_LIM2FIXP_DBL(x) (x) +#define PCM_LIM_BITS 32 +#define FIXP_PCM_LIM FIXP_DBL +#define SAMPLE_BITS_LIM DFRACT_BITS /****************************************************************************** -* resetLimiter * -* limiter: limiter handle * -* returns: error code * -******************************************************************************/ -TDLIMITER_ERROR resetLimiter(TDLimiterPtr limiter); - + * pcmLimiter_Reset * + * limiter: limiter handle * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_Reset(TDLimiterPtr limiter); /****************************************************************************** -* destroyLimiter * -* limiter: limiter handle * -* returns: error code * -******************************************************************************/ -TDLIMITER_ERROR destroyLimiter(TDLimiterPtr limiter); + * pcmLimiter_Destroy * + * limiter: limiter handle * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_Destroy(TDLimiterPtr limiter); /****************************************************************************** -* applyLimiter * -* limiter: limiter handle * -* pGain : pointer to gains to be applied to the signal before limiting, * -* which are downscaled by TDL_GAIN_SCALING bit. * -* These gains are delayed by gain_delay, and smoothed. * -* Smoothing is done by a butterworth lowpass filter with a cutoff * -* frequency which is fixed with respect to the sampling rate. * -* It is a substitute for the smoothing due to windowing and * -* overlap/add, if a gain is applied in frequency domain. * -* gain_scale: pointer to scaling exponents to be applied to the signal before * -* limiting, without delay and without smoothing * -* gain_size: number of elements in pGain, currently restricted to 1 * -* gain_delay: delay [samples] with which the gains in pGain shall be applied * -* gain_delay <= nSamples * -* samples: input/output buffer containing interleaved samples * -* precision of output will be DFRACT_BITS-TDL_GAIN_SCALING bits * -* nSamples: number of samples per channel * -* returns: error code * -******************************************************************************/ -TDLIMITER_ERROR applyLimiter(TDLimiterPtr limiter, - INT_PCM* samples, - FIXP_DBL* pGain, - const INT* gain_scale, - const UINT gain_size, - const UINT gain_delay, - const UINT nSamples); + * pcmLimiter_GetDelay * + * limiter: limiter handle * + * returns: exact delay caused by the limiter in samples per channel * + ******************************************************************************/ +unsigned int pcmLimiter_GetDelay(TDLimiterPtr limiter); /****************************************************************************** -* getLimiterDelay * -* limiter: limiter handle * -* returns: exact delay caused by the limiter in samples * -******************************************************************************/ -unsigned int getLimiterDelay(TDLimiterPtr limiter); + * pcmLimiter_GetMaxGainReduction * + * limiter: limiter handle * + * returns: maximum gain reduction in last processed block in dB * + ******************************************************************************/ +INT pcmLimiter_GetMaxGainReduction(TDLimiterPtr limiter); /****************************************************************************** -* setLimiterNChannels * -* limiter: limiter handle * -* nChannels: number of channels ( <= maxChannels specified on create) * -* returns: error code * -******************************************************************************/ -TDLIMITER_ERROR setLimiterNChannels(TDLimiterPtr limiter, unsigned int nChannels); + * pcmLimiter_SetNChannels * + * limiter: limiter handle * + * nChannels: number of channels ( <= maxChannels specified on create) * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_SetNChannels(TDLimiterPtr limiter, + unsigned int nChannels); /****************************************************************************** -* setLimiterSampleRate * -* limiter: limiter handle * -* sampleRate: sampling rate in Hz ( <= maxSampleRate specified on create) * -* returns: error code * -******************************************************************************/ -TDLIMITER_ERROR setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate); + * pcmLimiter_SetSampleRate * + * limiter: limiter handle * + * sampleRate: sampling rate in Hz ( <= maxSampleRate specified on create) * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_SetSampleRate(TDLimiterPtr limiter, UINT sampleRate); /****************************************************************************** -* setLimiterAttack * -* limiter: limiter handle * -* attackMs: attack time in ms ( <= maxAttackMs specified on create) * -* returns: error code * -******************************************************************************/ -TDLIMITER_ERROR setLimiterAttack(TDLimiterPtr limiter, unsigned int attackMs); + * pcmLimiter_SetAttack * + * limiter: limiter handle * + * attackMs: attack time in ms ( <= maxAttackMs specified on create) * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_SetAttack(TDLimiterPtr limiter, + unsigned int attackMs); /****************************************************************************** -* setLimiterRelease * -* limiter: limiter handle * -* releaseMs: release time in ms * -* returns: error code * -******************************************************************************/ -TDLIMITER_ERROR setLimiterRelease(TDLimiterPtr limiter, unsigned int releaseMs); + * pcmLimiter_SetRelease * + * limiter: limiter handle * + * releaseMs: release time in ms * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_SetRelease(TDLimiterPtr limiter, + unsigned int releaseMs); /****************************************************************************** -* setLimiterThreshold * -* limiter: limiter handle * -* threshold: limiter threshold * -* returns: error code * -******************************************************************************/ -TDLIMITER_ERROR setLimiterThreshold(TDLimiterPtr limiter, INT_PCM threshold); + * pcmLimiter_GetLibInfo * + * info: pointer to an allocated and initialized LIB_INFO structure * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_GetLibInfo(LIB_INFO* info); #ifdef __cplusplus } #endif +/****************************************************************************** + * pcmLimiter_Create * + * maxAttackMs: maximum and initial attack/lookahead time in milliseconds * + * releaseMs: release time in milliseconds (90% time constant) * + * threshold: limiting threshold * + * maxChannels: maximum and initial number of channels * + * maxSampleRate: maximum and initial sampling rate in Hz * + * returns: limiter handle * + ******************************************************************************/ +TDLimiterPtr pcmLimiter_Create(unsigned int maxAttackMs, unsigned int releaseMs, + FIXP_DBL threshold, unsigned int maxChannels, + UINT maxSampleRate); -#endif //#ifndef _LIMITER_H_ +/****************************************************************************** + * pcmLimiter_SetThreshold * + * limiter: limiter handle * + * threshold: limiter threshold * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_SetThreshold(TDLimiterPtr limiter, + FIXP_DBL threshold); + +/****************************************************************************** + * pcmLimiter_Apply * + * limiter: limiter handle * + * pGain : pointer to gains to be applied to the signal before limiting, * + * which are downscaled by TDL_GAIN_SCALING bit. * + * These gains are delayed by gain_delay, and smoothed. * + * Smoothing is done by a butterworth lowpass filter with a cutoff * + * frequency which is fixed with respect to the sampling rate. * + * It is a substitute for the smoothing due to windowing and * + * overlap/add, if a gain is applied in frequency domain. * + * gain_scale: pointer to scaling exponents to be applied to the signal before * + * limiting, without delay and without smoothing * + * gain_size: number of elements in pGain, currently restricted to 1 * + * gain_delay: delay [samples] with which the gains in pGain shall be applied * + * gain_delay <= nSamples * + * samples: input/output buffer containing interleaved samples * + * precision of output will be DFRACT_BITS-TDL_GAIN_SCALING bits * + * nSamples: number of samples per channel * + * returns: error code * + ******************************************************************************/ +TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, + INT_PCM* samplesOut, FIXP_DBL* pGain, + const INT* gain_scale, const UINT gain_size, + const UINT gain_delay, const UINT nSamples); + +#endif /* #ifndef LIMITER_H */ diff --git a/libPCMutils/include/pcm_utils.h b/libPCMutils/include/pcm_utils.h new file mode 100644 index 0000000..073bcfc --- /dev/null +++ b/libPCMutils/include/pcm_utils.h @@ -0,0 +1,131 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© 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. + +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 +----------------------------------------------------------------------------- */ + +/**************************** PCM utility library ****************************** + + Author(s): Alfonso Pino Garcia + + Description: Functions that perform (de)interleaving combined with format +change + +*******************************************************************************/ + +#if !defined(PCM_UTILS_H) +#define PCM_UTILS_H + +#include "common_fix.h" + +void FDK_interleave(const FIXP_DBL *RESTRICT pIn, LONG *RESTRICT pOut, + const UINT channels, const UINT frameSize, + const UINT length); +void FDK_interleave(const FIXP_DBL *RESTRICT pIn, SHORT *RESTRICT pOut, + const UINT channels, const UINT frameSize, + const UINT length); +void FDK_interleave(const FIXP_SGL *RESTRICT pIn, SHORT *RESTRICT pOut, + const UINT channels, const UINT frameSize, + const UINT length); + +void FDK_deinterleave(const LONG *RESTRICT pIn, SHORT *RESTRICT pOut, + const UINT channels, const UINT frameSize, + const UINT length); +void FDK_deinterleave(const LONG *RESTRICT pIn, LONG *RESTRICT pOut, + const UINT channels, const UINT frameSize, + const UINT length); +void FDK_deinterleave(const SHORT *RESTRICT pIn, SHORT *RESTRICT pOut, + const UINT channels, const UINT frameSize, + const UINT length); +void FDK_deinterleave(const SHORT *RESTRICT pIn, LONG *RESTRICT pOut, + const UINT channels, const UINT frameSize, + const UINT length); +#endif /* !defined(PCM_UTILS_H) */ diff --git a/libPCMutils/include/pcmdmx_lib.h b/libPCMutils/include/pcmdmx_lib.h new file mode 100644 index 0000000..d37a851 --- /dev/null +++ b/libPCMutils/include/pcmdmx_lib.h @@ -0,0 +1,460 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© 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. + +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 +----------------------------------------------------------------------------- */ + +/**************************** PCM utility library ****************************** + + Author(s): Christian Griebel + + Description: + +*******************************************************************************/ + +/** + * \file pcmdmx_lib.h + * \brief FDK PCM audio mixdown library interface header file. + + \page INTRO Introduction + + + \section SCOPE Scope + + This document describes the high-level application interface and usage of the + FDK PCM audio mixdown module library developed by the Fraunhofer Institute for + Integrated Circuits (IIS). Depending on the library configuration, the module + can manipulate the number of audio channels of a given PCM signal. It can + create for example a two channel stereo audio signal from a given multi-channel + configuration (e.g. 5.1 channels). + + + \page ABBREV List of abbreviations + + \li \b AAC - Advanced Audio Coding\n + Is an audio coding standard for lossy digital audio compression standardized + by ISO and IEC, as part of the MPEG-2 (ISO/IEC 13818-7:2006) and MPEG-4 + (ISO/IEC 14496-3:2009) specifications. + + \li \b DSE - Data Stream Element\n + A syntactical element of the MPEG-2/4 Advanced Audio Coding bitstream + standardized in ISO/IEC 14496-3:2009. It can convey any kind of data associated + to one program. + + \li \b PCE - Program Config Element\n + A syntactical element of the MPEG-2/4 Advanced Audio Coding bitstream + standardized in ISO/IEC 14496-3:2009 that can define the stream configuration + for a single program. In addition it can comprise simple downmix meta data. + + */ + +#ifndef PCMDMX_LIB_H +#define PCMDMX_LIB_H + +#include "machine_type.h" +#include "common_fix.h" +#include "FDK_audio.h" +#include "FDK_bitstream.h" + +/** + * \enum PCMDMX_ERROR + * + * Error codes that can be returned by module interface functions. + */ +typedef enum { + PCMDMX_OK = 0x0, /*!< No error happened. */ + PCMDMX_UNSUPPORTED = + 0x1, /*!< The requested feature/service is unavailable. This can + occur if the module was built for a wrong configuration. */ + pcm_dmx_fatal_error_start, + PCMDMX_OUT_OF_MEMORY, /*!< Not enough memory to set up an instance of the + module. */ + pcm_dmx_fatal_error_end, + + 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_OUTPUT_BUFFER_TOO_SMALL /*!< The size of pcm output buffer is too + small. */ + +} PCMDMX_ERROR; + +/** Macro to identify fatal errors. */ +#define PCMDMX_IS_FATAL_ERROR(err) \ + ((((err) >= pcm_dmx_fatal_error_start) && \ + ((err) <= pcm_dmx_fatal_error_end)) \ + ? 1 \ + : 0) + +/** + * \enum PCMDMX_PARAM + * + * Modules dynamic runtime parameters that can be handed to function + * pcmDmx_SetParam() and pcmDmx_GetParam(). + */ +typedef enum { + DMX_PROFILE_SETTING = + 0x01, /*!< Defines which equations, coefficients and default/ + fallback values used for downmixing. See + ::DMX_PROFILE_TYPE type for details. */ + DMX_BS_DATA_EXPIRY_FRAME = + 0x10, /*!< 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 = + 0x11, /*!< The number of delay frames of the output samples + compared to the bitstream data. */ + MIN_NUMBER_OF_OUTPUT_CHANNELS = + 0x20, /*!< The minimum number of output channels. For all + input configurations that have less than the given + channels the module will modify the output + automatically to obtain the given number of output + channels. Mono signals will be duplicated. If more + than two output channels are desired the module + just adds empty channels. The parameter value must + be either -1, 0, 1, 2, 6 or 8. If the value is + greater than zero and exceeds the value of + parameter ::MAX_NUMBER_OF_OUTPUT_CHANNELS the + latter will be set to the same value. Both values + -1 and 0 disable the feature. */ + MAX_NUMBER_OF_OUTPUT_CHANNELS = + 0x21, /*!< The maximum number of output channels. For all + input configurations that have more than the given + channels the module will apply a mixdown + automatically to obtain the given number of output + channels. The value must be either -1, 0, 1, 2, 6 + or 8. If it's greater than zero and lower or equal + than the value of ::MIN_NUMBER_OF_OUTPUT_CHANNELS + parameter the latter will be set to the same value. + The values -1 and 0 disable the feature. */ + DMX_DUAL_CHANNEL_MODE = + 0x30, /*!< Downmix mode for two channel audio data. See type + ::DUAL_CHANNEL_MODE for details. */ + DMX_PSEUDO_SURROUND_MODE = + 0x31 /*!< Defines how module handles pseudo surround + compatible signals. See ::PSEUDO_SURROUND_MODE + type for details. */ +} PCMDMX_PARAM; + +/** + * \enum DMX_PROFILE_TYPE + * + * Valid value list for parameter ::DMX_PROFILE_SETTING. + */ +typedef enum { + DMX_PRFL_STANDARD = + 0x0, /*!< The standard profile creates mixdown signals based on + the advanced downmix metadata (from a DSE), equations + and default values defined in ISO/IEC 14496:3 + Ammendment 4. Any other (legacy) downmix metadata will + be ignored. */ + DMX_PRFL_MATRIX_MIX = + 0x1, /*!< This profile behaves just as the standard profile if + advanced downmix metadata (from a DSE) is available. If + not, the matrix_mixdown information embedded in the + program configuration element (PCE) will be applied. If + neither is the case the module creates a mixdown using + the default coefficients defined in MPEG-4 Ammendment 4. + The profile can be used e.g. to support legacy digital + TV (e.g. DVB) streams. */ + DMX_PRFL_FORCE_MATRIX_MIX = + 0x2, /*!< Similar to the ::DMX_PRFL_MATRIX_MIX profile but if both + the advanced (DSE) and the legacy (PCE) MPEG downmix + metadata are available the latter will be applied. */ + DMX_PRFL_ARIB_JAPAN = + 0x3 /*!< Downmix creation as described in ABNT NBR 15602-2. But + if advanced downmix metadata is available it will be + prefered. */ +} DMX_PROFILE_TYPE; + +/** + * \enum PSEUDO_SURROUND_MODE + * + * Valid value list for parameter ::DMX_PSEUDO_SURROUND_MODE. + */ +typedef enum { + NEVER_DO_PS_DMX = + -1, /*!< Ignore any metadata and do never create a pseudo surround + compatible downmix. (Default) */ + AUTO_PS_DMX = 0, /*!< Create a pseudo surround compatible downmix only if + signalled in bitstreams meta data. */ + FORCE_PS_DMX = + 1 /*!< Always create a pseudo surround compatible downmix. + CAUTION: This can lead to excessive signal cancellations + and signal level differences for non-compatible signals. */ +} PSEUDO_SURROUND_MODE; + +/** + * \enum DUAL_CHANNEL_MODE + * + * Valid value list for parameter ::DMX_DUAL_CHANNEL_MODE. + */ +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; + +#define DMX_PCM FIXP_DBL +#define DMX_PCMF FIXP_DBL +#define DMX_PCM_BITS DFRACT_BITS +#define FX_DMX2FX_PCM(x) FX_DBL2FX_PCM((FIXP_DBL)(x)) + +/* ------------------------ * + * MODULES INTERFACE: * + * ------------------------ */ +typedef struct PCM_DMX_INSTANCE *HANDLE_PCM_DOWNMIX; + +/*! \addtogroup pcmDmxResetFlags Modules reset flags + * Macros that can be used as parameter for function pcmDmx_Reset() to specify + * which parts of the module shall be reset. + * @{ + * + * \def PCMDMX_RESET_PARAMS + * Only reset the user specific parameters that have been modified with + * pcmDmx_SetParam(). + * + * \def PCMDMX_RESET_BS_DATA + * Delete the meta data that has been fed with the appropriate interface + * functions. + * + * \def PCMDMX_RESET_FULL + * Reset the complete module instance to the state after pcmDmx_Open() had been + * called. + */ +#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] pSelf Pointer to a buffer receiving the handle of the new + *instance. + * @returns Returns an error code of type ::PCMDMX_ERROR. + **/ +PCMDMX_ERROR pcmDmx_Open(HANDLE_PCM_DOWNMIX *pSelf); + +/** Set one parameter for a single instance of the PCM downmix module. + * @param[in] self Handle of PCM downmix instance. + * @param[in] param Parameter to be set. Can be one from the ::PCMDMX_PARAM + *list. + * @param[in] value Parameter value. + * @returns Returns an error code of type ::PCMDMX_ERROR. + **/ +PCMDMX_ERROR pcmDmx_SetParam(HANDLE_PCM_DOWNMIX self, const PCMDMX_PARAM param, + const INT value); + +/** Get one parameter value of a single PCM downmix module instance. + * @param[in] self Handle of PCM downmix module instance. + * @param[in] param Parameter to query. Can be one from the ::PCMDMX_PARAM + *list. + * @param[out] pValue Pointer to buffer receiving the parameter value. + * @returns Returns an error code of type ::PCMDMX_ERROR. + **/ +PCMDMX_ERROR pcmDmx_GetParam(HANDLE_PCM_DOWNMIX self, const PCMDMX_PARAM param, + INT *const pValue); + +/** \cond + * Extract relevant downmix meta-data directly from a given bitstream. The + *function can handle both data specified in ETSI TS 101 154 or ISO/IEC + *14496-3:2009/Amd.4:2013. + * @param[in] self Handle of PCM downmix instance. + * @param[in] hBitStream Handle of FDK bitstream buffer. + * @param[in] ancDataBits Length of ancillary data in bits. + * @param[in] isMpeg2 Flag indicating wheter the ancillary data is from a + *MPEG-1/2 or a MPEG-4 stream. + * @returns Returns an error code of type ::PCMDMX_ERROR. + **/ +PCMDMX_ERROR pcmDmx_Parse(HANDLE_PCM_DOWNMIX self, + HANDLE_FDK_BITSTREAM hBitStream, UINT ancDataBits, + int isMpeg2); +/** \endcond */ + +/** Read from a given ancillary data buffer and extract the relevant downmix + *meta-data. The function can handle both data specified in ETSI TS 101 154 or + *ISO/IEC 14496-3:2009/Amd.4:2013. + * @param[in] self Handle of PCM downmix instance. + * @param[in] pAncDataBuf Pointer to ancillary buffer holding the data. + * @param[in] ancDataBytes Size of ancillary data in bytes. + * @param[in] isMpeg2 Flag indicating wheter the ancillary data is from a + *MPEG-1/2 or a MPEG-4 stream. + * @returns Returns an error code of type ::PCMDMX_ERROR. + **/ +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. + * @param[in] self Handle of PCM downmix instance. + * @param[in] matrixMixdownPresent Matrix mixdown index present flag extracted + *from PCE. + * @param[in] matrixMixdownIdx The 2 bit matrix mixdown index extracted + *from PCE. + * @param[in] pseudoSurroundEnable The pseudo surround enable flag extracted + *from PCE. + * @returns Returns an error code of type + *::PCMDMX_ERROR. + **/ +PCMDMX_ERROR pcmDmx_SetMatrixMixdownFromPce(HANDLE_PCM_DOWNMIX self, + int matrixMixdownPresent, + int matrixMixdownIdx, + int pseudoSurroundEnable); + +/** Reset the module. + * @param[in] self Handle of PCM downmix instance. + * @param[in] flags Flags telling which parts of the module shall be reset. + * See \ref pcmDmxResetFlags for details. + * @returns Returns an error code of type ::PCMDMX_ERROR. + **/ +PCMDMX_ERROR pcmDmx_Reset(HANDLE_PCM_DOWNMIX self, UINT flags); + +/** Create a mixdown, bypass or extend the output signal depending on the + *modules settings and the respective given input configuration. + * + * \param[in] self Handle of PCM downmix module instance. + * \param[in,out] pPcmBuf Pointer to time buffer with PCM samples. + * \param[in] pcmBufSize Size of pPcmBuf buffer. + * \param[in] frameSize The I/O block size which is the number of samples per channel. + * \param[in,out] nChannels Pointer to buffer that holds the number of input channels and + * where the amount of output channels is written + *to. + * \param[in] fInterleaved Input and output samples are processed interleaved. + * \param[in,out] channelType Array were the corresponding channel type for each output audio + * channel is stored into. + * \param[in,out] channelIndices Array were the corresponding channel type index for each output + * audio channel is stored into. + * \param[in] mapDescr Pointer to a FDK channel mapping descriptor that contains the + * channel mapping to be used. + * \param[out] pDmxOutScale Pointer on a field receiving the scale factor that has to be + * applied on all samples afterwards. If the + *handed pointer is NULL the final scaling is done internally. + * @returns Returns an error code of type ::PCMDMX_ERROR. + **/ +PCMDMX_ERROR pcmDmx_ApplyFrame(HANDLE_PCM_DOWNMIX self, DMX_PCM *pPcmBuf, + const int pcmBufSize, UINT frameSize, + INT *nChannels, INT fInterleaved, + AUDIO_CHANNEL_TYPE channelType[], + UCHAR channelIndices[], + const FDK_channelMapDescr *const mapDescr, + INT *pDmxOutScale); + +/** Close an instance of the PCM downmix module. + * @param[in,out] pSelf Pointer to a buffer containing the handle of the + *instance. + * @returns Returns an error code of type ::PCMDMX_ERROR. + **/ +PCMDMX_ERROR pcmDmx_Close(HANDLE_PCM_DOWNMIX *pSelf); + +/** Get library info for this module. + * @param[out] info Pointer to an allocated LIB_INFO structure. + * @returns Returns an error code of type ::PCMDMX_ERROR. + */ +PCMDMX_ERROR pcmDmx_GetLibInfo(LIB_INFO *info); + +#ifdef __cplusplus +} +#endif + +#endif /* PCMDMX_LIB_H */ diff --git a/libPCMutils/include/pcmutils_lib.h b/libPCMutils/include/pcmutils_lib.h deleted file mode 100644 index e7e6a41..0000000 --- a/libPCMutils/include/pcmutils_lib.h +++ /dev/null @@ -1,334 +0,0 @@ - -/* ----------------------------------------------------------------------------------------------------------- -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. - - 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" -#include "FDK_bitstream.h" - - -/* ------------------------ * - * ERROR CODES: * - * ------------------------ */ -typedef enum -{ - PCMDMX_OK = 0x0, /*!< No error happened. */ - - pcm_dmx_fatal_error_start, - 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. */ - pcm_dmx_fatal_error_end, - - 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; - -/** Macro to identify fatal errors. */ -#define PCMDMX_IS_FATAL_ERROR(err) ( (((err)>=pcm_dmx_fatal_error_start) && ((err)<=pcm_dmx_fatal_error_end)) ? 1 : 0) - -/* ------------------------ * - * 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. */ - MIN_NUMBER_OF_OUTPUT_CHANNELS, /*!< The minimum number of output channels. For all input - configurations that have less than the given channels the - module will modify the output automatically to obtain the - given number of output channels. Mono signals will be - duplicated. If more than two output channels are desired - the module just adds empty channels. The parameter value - must be either -1, 0, 1, 2, 6 or 8. If the value is - greater than zero and exceeds the value of parameter - MAX_NUMBER_OF_OUTPUT_CHANNELS the latter will be set to - the same value. Both values -1 and 0 disable the feature. */ - MAX_NUMBER_OF_OUTPUT_CHANNELS, /*!< The maximum number of output channels. For all input - configurations that have more than the given channels the - module will apply a mixdown automatically to obtain the - given number of output channels. The value must be either - -1, 0, 1, 2, 6 or 8. If it is greater than zero and lower - or equal than the value of MIN_NUMBER_OF_OUTPUT_CHANNELS - parameter the latter will be set to the same value. - The values -1 and 0 disable the feature. */ - DMX_DUAL_CHANNEL_MODE, /*!< Downmix mode for two channel audio data. */ - DMX_PSEUDO_SURROUND_MODE /*!< Defines how module handles pseudo surround compatible - signals. See PSEUDO_SURROUND_MODE type for details. */ -} PCMDMX_PARAM; - -/* Parameter value types */ -typedef enum -{ - NEVER_DO_PS_DMX = -1, /*!< Never create a pseudo surround compatible downmix. */ - AUTO_PS_DMX = 0, /*!< Create a pseudo surround compatible downmix only if - signalled in bitstreams meta data. (Default) */ - FORCE_PS_DMX = 1 /*!< Always create a pseudo surround compatible downmix. - CAUTION: This can lead to excessive signal cancellations - and signal level differences for non-compatible signals. */ -} PSEUDO_SURROUND_MODE; - -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, - const PCMDMX_PARAM param, - const INT value - ); - -/** Get one parameter value of one PCM downmix module instance. - * @param [in] Handle of PCM downmix module instance. - * @param [in] Parameter to be set. - * @param [out] Pointer to buffer receiving the parameter value. - * @returns Returns an error code. - **/ -PCMDMX_ERROR pcmDmx_GetParam ( - HANDLE_PCM_DOWNMIX self, - const PCMDMX_PARAM param, - INT * const pValue - ); - -/** Read downmix meta-data directly from a given bitstream. - * @param [in] Handle of PCM downmix instance. - * @param [in] Handle of FDK bitstream buffer. - * @param [in] Length of ancillary data in bits. - * @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_Parse ( - HANDLE_PCM_DOWNMIX self, - HANDLE_FDK_BITSTREAM hBitStream, - UINT ancDataBits, - int isMpeg2 - ); - -/** Read downmix meta-data from a given data buffer. - * @param [in] Handle of PCM downmix instance. - * @param [in] Pointer to ancillary data buffer. - * @param [in] Size of ancillary data in bytes. - * @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. - * @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 - ); - -/** Create a mixdown, bypass or extend the output signal depending on the modules settings and the - * respective given input configuration. - * - * \param [in] Handle of PCM downmix module instance. - * \param [inout] Pointer to time buffer with decoded PCM samples. - * \param [in] The I/O block size which is the number of samples per channel. - * \param [inout] Pointer to buffer that holds the number of input channels and where the - * amount of output channels is written to. - * \param [in] Flag which indicates if output time data is 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). - * \param [out] Pointer on a field receiving the scale factor that has to be applied on all - * samples afterwards. If the handed pointer is NULL the final scaling is done - * internally. - * @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], - INT *pDmxOutScale - ); - -/** 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_ */ |