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Diffstat (limited to 'libSBRdec/src/lpp_tran.h')
| -rw-r--r-- | libSBRdec/src/lpp_tran.h | 339 |
1 files changed, 186 insertions, 153 deletions
diff --git a/libSBRdec/src/lpp_tran.h b/libSBRdec/src/lpp_tran.h index 003a547..51b4395 100644 --- a/libSBRdec/src/lpp_tran.h +++ b/libSBRdec/src/lpp_tran.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 @@ -79,164 +90,186 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ /*! \file - \brief Low Power Profile Transposer, + \brief Low Power Profile Transposer */ -#ifndef _LPP_TRANS_H -#define _LPP_TRANS_H +#ifndef LPP_TRAN_H +#define LPP_TRAN_H #include "sbrdecoder.h" +#include "hbe.h" #include "qmf.h" /* Common */ -#define QMF_OUT_SCALE 8 +#define QMF_OUT_SCALE 8 + +/* + Frequency scales +*/ /* Env-Adjust */ -#define MAX_NOISE_ENVELOPES 2 -#define MAX_NOISE_COEFFS 5 -#define MAX_NUM_NOISE_VALUES (MAX_NOISE_ENVELOPES * MAX_NOISE_COEFFS) -#define MAX_NUM_LIMITERS 12 +#define MAX_NOISE_ENVELOPES 2 +#define MAX_NOISE_COEFFS 5 +#define MAX_NUM_NOISE_VALUES (MAX_NOISE_ENVELOPES * MAX_NOISE_COEFFS) +#define MAX_NUM_LIMITERS 12 /* Set MAX_ENVELOPES to the largest value of all supported BSFORMATs by overriding MAX_ENVELOPES in the correct order: */ -#define MAX_ENVELOPES_HEAAC 5 -#define MAX_ENVELOPES MAX_ENVELOPES_HEAAC +#define MAX_ENVELOPES_LEGACY 5 +#define MAX_ENVELOPES_USAC 8 +#define MAX_ENVELOPES MAX_ENVELOPES_USAC -#define MAX_FREQ_COEFFS 48 -#define MAX_FREQ_COEFFS_FS44100 35 -#define MAX_FREQ_COEFFS_FS48000 32 +#define MAX_FREQ_COEFFS_DUAL_RATE 48 +#define MAX_FREQ_COEFFS_QUAD_RATE 56 +#define MAX_FREQ_COEFFS MAX_FREQ_COEFFS_QUAD_RATE +#define MAX_FREQ_COEFFS_FS44100 35 +#define MAX_FREQ_COEFFS_FS48000 32 -#define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS) +#define MAX_NUM_ENVELOPE_VALUES (MAX_ENVELOPES * MAX_FREQ_COEFFS) -#define MAX_GAIN_EXP 34 +#define MAX_GAIN_EXP 34 /* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_EXP) example: 34=99dB */ -#define MAX_GAIN_CONCEAL_EXP 1 -/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_CONCEAL_EXP) in concealment case (0dB) */ +#define MAX_GAIN_CONCEAL_EXP 1 +/* Maximum gain will be sqrt(0.5 * 2^MAX_GAIN_CONCEAL_EXP) in concealment case + * (0dB) */ /* LPP Transposer */ -#define LPC_ORDER 2 +#define LPC_ORDER 2 -#define MAX_INVF_BANDS MAX_NOISE_COEFFS +#define MAX_INVF_BANDS MAX_NOISE_COEFFS -#define MAX_NUM_PATCHES 6 -#define SHIFT_START_SB 1 /*!< lowest subband of source range */ +#define MAX_NUM_PATCHES 6 +#define SHIFT_START_SB 1 /*!< lowest subband of source range */ -typedef enum -{ +typedef enum { INVF_OFF = 0, INVF_LOW_LEVEL, INVF_MID_LEVEL, INVF_HIGH_LEVEL, INVF_SWITCHED /* not a real choice but used here to control behaviour */ -} -INVF_MODE; - +} INVF_MODE; /** parameter set for one single patch */ typedef struct { - UCHAR sourceStartBand; /*!< first band in lowbands where to take the samples from */ - UCHAR sourceStopBand; /*!< first band in lowbands which is not included in the patch anymore */ - UCHAR guardStartBand; /*!< first band in highbands to be filled with zeros in order to - reduce interferences between patches */ - UCHAR targetStartBand; /*!< first band in highbands to be filled with whitened lowband signal */ - UCHAR targetBandOffs; /*!< difference between 'startTargetBand' and 'startSourceBand' */ - UCHAR numBandsInPatch; /*!< number of consecutive bands in this one patch */ + UCHAR sourceStartBand; /*!< first band in lowbands where to take the samples + from */ + UCHAR + sourceStopBand; /*!< first band in lowbands which is not included in the + patch anymore */ + UCHAR guardStartBand; /*!< first band in highbands to be filled with zeros in + order to reduce interferences between patches */ + UCHAR + targetStartBand; /*!< first band in highbands to be filled with whitened + lowband signal */ + UCHAR targetBandOffs; /*!< difference between 'startTargetBand' and + 'startSourceBand' */ + UCHAR numBandsInPatch; /*!< number of consecutive bands in this one patch */ } PATCH_PARAM; - /** whitening factors for different levels of whitening need to be initialized corresponding to crossover frequency */ typedef struct { - FIXP_DBL off; /*!< bw factor for signal OFF */ - FIXP_DBL transitionLevel; - FIXP_DBL lowLevel; /*!< bw factor for signal LOW_LEVEL */ - FIXP_DBL midLevel; /*!< bw factor for signal MID_LEVEL */ - FIXP_DBL highLevel; /*!< bw factor for signal HIGH_LEVEL */ + FIXP_DBL off; /*!< bw factor for signal OFF */ + FIXP_DBL transitionLevel; + FIXP_DBL lowLevel; /*!< bw factor for signal LOW_LEVEL */ + FIXP_DBL midLevel; /*!< bw factor for signal MID_LEVEL */ + FIXP_DBL highLevel; /*!< bw factor for signal HIGH_LEVEL */ } WHITENING_FACTORS; - -/*! The transposer settings are calculated on a header reset and are shared by both channels. */ +/*! The transposer settings are calculated on a header reset and are shared by + * both channels. */ typedef struct { - UCHAR nCols; /*!< number subsamples of a codec frame */ - UCHAR noOfPatches; /*!< number of patches */ - UCHAR lbStartPatching; /*!< first band of lowbands that will be patched */ - UCHAR lbStopPatching; /*!< first band that won't be patched anymore*/ - UCHAR bwBorders[MAX_NUM_NOISE_VALUES]; /*!< spectral bands with different inverse filtering levels */ - - PATCH_PARAM patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */ - WHITENING_FACTORS whFactors; /*!< the pole moving factors for certain whitening levels as indicated - in the bitstream depending on the crossover frequency */ - UCHAR overlap; /*!< Overlap size */ + UCHAR nCols; /*!< number subsamples of a codec frame */ + UCHAR noOfPatches; /*!< number of patches */ + UCHAR lbStartPatching; /*!< first band of lowbands that will be patched */ + UCHAR lbStopPatching; /*!< first band that won't be patched anymore*/ + UCHAR bwBorders[MAX_NUM_NOISE_VALUES]; /*!< spectral bands with different + inverse filtering levels */ + + PATCH_PARAM + patchParam[MAX_NUM_PATCHES]; /*!< new parameter set for patching */ + WHITENING_FACTORS + whFactors; /*!< the pole moving factors for certain + whitening levels as indicated in the bitstream + depending on the crossover frequency */ + UCHAR overlap; /*!< Overlap size */ } TRANSPOSER_SETTINGS; +typedef struct { + TRANSPOSER_SETTINGS *pSettings; /*!< Common settings for both channels */ + FIXP_DBL + bwVectorOld[MAX_NUM_PATCHES]; /*!< pole moving factors of past frame */ + FIXP_DBL lpcFilterStatesRealLegSBR[LPC_ORDER + (3 * (4))][( + 32)]; /*!< pointer array to save filter states */ -typedef struct -{ - TRANSPOSER_SETTINGS *pSettings; /*!< Common settings for both channels */ - FIXP_DBL bwVectorOld[MAX_NUM_PATCHES]; /*!< pole moving factors of past frame */ - FIXP_DBL lpcFilterStatesReal[LPC_ORDER][(32)]; /*!< pointer array to save filter states */ - FIXP_DBL lpcFilterStatesImag[LPC_ORDER][(32)]; /*!< pointer array to save filter states */ -} -SBR_LPP_TRANS; - -typedef SBR_LPP_TRANS *HANDLE_SBR_LPP_TRANS; - + FIXP_DBL lpcFilterStatesImagLegSBR[LPC_ORDER + (3 * (4))][( + 32)]; /*!< pointer array to save filter states */ -void lppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, - QMF_SCALE_FACTOR *sbrScaleFactor, - FIXP_DBL **qmfBufferReal, + FIXP_DBL lpcFilterStatesRealHBE[LPC_ORDER + (3 * (4))][( + 64)]; /*!< pointer array to save filter states */ + FIXP_DBL lpcFilterStatesImagHBE[LPC_ORDER + (3 * (4))][( + 64)]; /*!< pointer array to save filter states */ +} SBR_LPP_TRANS; - FIXP_DBL *degreeAlias, - FIXP_DBL **qmfBufferImag, - const int useLP, - const int timeStep, - const int firstSlotOffset, - const int lastSlotOffset, - const int nInvfBands, - INVF_MODE *sbr_invf_mode, - INVF_MODE *sbr_invf_mode_prev - ); +typedef SBR_LPP_TRANS *HANDLE_SBR_LPP_TRANS; +void lppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, + QMF_SCALE_FACTOR *sbrScaleFactor, FIXP_DBL **qmfBufferReal, + + FIXP_DBL *degreeAlias, FIXP_DBL **qmfBufferImag, + const int useLP, const int fPreWhitening, + const int v_k_master0, const int timeStep, + const int firstSlotOffset, const int lastSlotOffset, + const int nInvfBands, INVF_MODE *sbr_invf_mode, + INVF_MODE *sbr_invf_mode_prev); + +void lppTransposerHBE( + HANDLE_SBR_LPP_TRANS hLppTrans, /*!< Handle of lpp transposer */ + HANDLE_HBE_TRANSPOSER hQmfTransposer, + QMF_SCALE_FACTOR *sbrScaleFactor, /*!< Scaling factors */ + FIXP_DBL **qmfBufferReal, /*!< Pointer to pointer to real part of subband + samples (source) */ + FIXP_DBL **qmfBufferImag, /*!< Pointer to pointer to imaginary part of + subband samples (source) */ + const int timeStep, /*!< Time step of envelope */ + const int firstSlotOffs, /*!< Start position in time */ + const int lastSlotOffs, /*!< Number of overlap-slots into next frame */ + const int nInvfBands, /*!< Number of bands for inverse filtering */ + INVF_MODE *sbr_invf_mode, /*!< Current inverse filtering modes */ + INVF_MODE *sbr_invf_mode_prev /*!< Previous inverse filtering modes */ +); SBR_ERROR -createLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, - TRANSPOSER_SETTINGS *pSettings, - const int highBandStartSb, - UCHAR *v_k_master, - const int numMaster, - const int usb, - const int timeSlots, - const int nCols, - UCHAR *noiseBandTable, - const int noNoiseBands, - UINT fs, - const int chan, - const int overlap); - +createLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, + TRANSPOSER_SETTINGS *pSettings, const int highBandStartSb, + UCHAR *v_k_master, const int numMaster, const int usb, + const int timeSlots, const int nCols, UCHAR *noiseBandTable, + const int noNoiseBands, UINT fs, const int chan, + const int overlap); SBR_ERROR -resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans, - UCHAR highBandStartSb, - UCHAR *v_k_master, - UCHAR numMaster, - UCHAR *noiseBandTable, - UCHAR noNoiseBands, - UCHAR usb, - UINT fs); - - - -#endif /* _LPP_TRANS_H */ +resetLppTransposer(HANDLE_SBR_LPP_TRANS hLppTrans, UCHAR highBandStartSb, + UCHAR *v_k_master, UCHAR numMaster, UCHAR *noiseBandTable, + UCHAR noNoiseBands, UCHAR usb, UINT fs); +#endif /* LPP_TRAN_H */ |