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diff --git a/libSBRdec/src/psdec.h b/libSBRdec/src/psdec.h new file mode 100644 index 0000000..e3a0424 --- /dev/null +++ b/libSBRdec/src/psdec.h @@ -0,0 +1,352 @@ + +/* ----------------------------------------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------------------------------------- */ + +/*! + \file + \brief Sbr decoder +*/ +#ifndef __PSDEC_H +#define __PSDEC_H + +#include "sbrdecoder.h" + + + +/* This PS decoder implements the baseline version. So it always uses the */ +/* hybrid filter structure for 20 stereo bands and does not implemet IPD/OPD */ +/* synthesis. The baseline version has to support the complete PS bitstream */ +/* syntax. But IPD/OPD data is ignored and set to 0. If 34 stereo band config */ +/* is used in the bitstream for IIS/ICC the decoded parameters are mapped to */ +/* 20 stereo bands. */ + + +#include "FDK_bitstream.h" + +#include "psdec_hybrid.h" + +#define SCAL_HEADROOM ( 2 ) + +#define PS_EXTENSION_SIZE_BITS ( 4 ) +#define PS_EXTENSION_ESC_COUNT_BITS ( 8 ) + +#define NO_QMF_CHANNELS ( 64 ) +#define MAX_NUM_COL ( 32 ) + + + #define NO_QMF_BANDS_HYBRID20 ( 3 ) + #define NO_SUB_QMF_CHANNELS ( 12 ) + + #define NRG_INT_COEFF ( 0.75f ) + #define INT_FILTER_COEFF (FL2FXCONST_DBL( 1.0f - NRG_INT_COEFF )) + #define PEAK_DECAY_FACTOR (FL2FXCONST_DBL( 0.765928338364649f )) + #define TRANSIENT_IMPACT_FACTOR (FL2FXCONST_DBL( 2.0 / 3.0 )) + + #define NO_SERIAL_ALLPASS_LINKS ( 3 ) + #define MAX_NO_PS_ENV ( 4 + 1 ) /* +1 needed for VAR_BORDER */ + + #define MAX_DELAY_BUFFER_SIZE ( 14 ) + #define NO_DELAY_BUFFER_BANDS ( 35 ) + + #define NO_HI_RES_BINS ( 34 ) + #define NO_MID_RES_BINS ( 20 ) + #define NO_LOW_RES_BINS ( 10 ) + + #define FIRST_DELAY_SB ( 23 ) + #define NO_SAMPLE_DELAY_ALLPASS ( 2 ) + #define NO_DELAY_LENGTH_VECTORS ( 12 ) /* d(m): d(0)=3 + d(1)=4 + d(2)=5 */ + + #define NO_HI_RES_IID_BINS ( NO_HI_RES_BINS ) + #define NO_HI_RES_ICC_BINS ( NO_HI_RES_BINS ) + + #define NO_MID_RES_IID_BINS ( NO_MID_RES_BINS ) + #define NO_MID_RES_ICC_BINS ( NO_MID_RES_BINS ) + + #define NO_LOW_RES_IID_BINS ( NO_LOW_RES_BINS ) + #define NO_LOW_RES_ICC_BINS ( NO_LOW_RES_BINS ) + + #define SUBQMF_GROUPS ( 10 ) + #define QMF_GROUPS ( 12 ) + + #define SUBQMF_GROUPS_HI_RES ( 32 ) + #define QMF_GROUPS_HI_RES ( 18 ) + + #define NO_IID_GROUPS ( SUBQMF_GROUPS + QMF_GROUPS ) + #define NO_IID_GROUPS_HI_RES ( SUBQMF_GROUPS_HI_RES + QMF_GROUPS_HI_RES ) + + #define NO_IID_STEPS ( 7 ) /* 1 .. + 7 */ + #define NO_IID_STEPS_FINE ( 15 ) /* 1 .. +15 */ + #define NO_ICC_STEPS ( 8 ) /* 0 .. + 7 */ + + #define NO_IID_LEVELS ( 2 * NO_IID_STEPS + 1 ) /* - 7 .. + 7 */ + #define NO_IID_LEVELS_FINE ( 2 * NO_IID_STEPS_FINE + 1 ) /* -15 .. +15 */ + #define NO_ICC_LEVELS ( NO_ICC_STEPS ) /* 0 .. + 7 */ + + #define FIXP_SQRT05 ((FIXP_DBL)0x5a827980) /* 1/SQRT2 */ + + struct PS_DEC_COEFFICIENTS { + + FIXP_DBL H11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL H12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL H21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL H22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + + FIXP_DBL DeltaH11r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL DeltaH12r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL DeltaH21r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + FIXP_DBL DeltaH22r[NO_IID_GROUPS]; /*!< coefficients of the sub-subband groups */ + + SCHAR aaIidIndexMapped[MAX_NO_PS_ENV][NO_HI_RES_IID_BINS]; /*!< The mapped IID index for all envelopes and all IID bins */ + SCHAR aaIccIndexMapped[MAX_NO_PS_ENV][NO_HI_RES_ICC_BINS]; /*!< The mapped ICC index for all envelopes and all ICC bins */ + + }; + + + + +typedef enum { + ppt_none = 0, + ppt_mpeg = 1, + ppt_drm = 2 +} PS_PAYLOAD_TYPE; + + +typedef struct { + UCHAR bPsHeaderValid; /*!< set if new header is available from bitstream */ + + UCHAR bEnableIid; /*!< One bit denoting the presence of IID parameters */ + UCHAR bEnableIcc; /*!< One bit denoting the presence of ICC parameters */ + UCHAR bEnableExt; /*!< The PS extension layer is enabled using the enable_ext bit. + If it is set to %1 the IPD and OPD parameters are sent. + If it is disabled, i.e. %0, the extension layer is skipped. */ + + UCHAR modeIid; /*!< The configuration of IID parameters (number of bands and + quantisation grid, iid_quant) is determined by iid_mode. */ + UCHAR modeIcc; /*!< The configuration of Inter-channel Coherence parameters + (number of bands and quantisation grid) is determined by + icc_mode. */ + + UCHAR freqResIid; /*!< 0=low, 1=mid or 2=high frequency resolution for iid */ + UCHAR freqResIcc; /*!< 0=low, 1=mid or 2=high frequency resolution for icc */ + + UCHAR bFineIidQ; /*!< Use fine Iid quantisation. */ + + UCHAR bFrameClass; /*!< The frame_class bit determines whether the parameter + positions of the current frame are uniformly spaced + accross the frame or they are defined using the positions + described by border_position. */ + + UCHAR noEnv; /*!< The number of envelopes per frame */ + UCHAR aEnvStartStop[MAX_NO_PS_ENV+1]; /*!< In case of variable parameter spacing the parameter + positions are determined by border_position */ + + SCHAR abIidDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for IID, 0 => freq */ + SCHAR abIccDtFlag[MAX_NO_PS_ENV]; /*!< Deltacoding time/freq flag for ICC, 0 => freq */ + + SCHAR aaIidIndex[MAX_NO_PS_ENV][NO_HI_RES_IID_BINS]; /*!< The IID index for all envelopes and all IID bins */ + SCHAR aaIccIndex[MAX_NO_PS_ENV][NO_HI_RES_ICC_BINS]; /*!< The ICC index for all envelopes and all ICC bins */ + +} MPEG_PS_BS_DATA; + + + +struct PS_DEC { + + SCHAR noSubSamples; + SCHAR noChannels; + + SCHAR procFrameBased; /*!< Helper to detected switching from frame based to slot based + processing */ + + PS_PAYLOAD_TYPE bPsDataAvail[(1)+1]; /*!< set if new data available from bitstream */ + UCHAR psDecodedPrv; /*!< set if PS has been processed in the last frame */ + + /* helpers for frame delay line */ + UCHAR bsLastSlot; /*!< Index of last read slot. */ + UCHAR bsReadSlot; /*!< Index of current read slot for additional delay. */ + UCHAR processSlot; /*!< Index of current slot for processing (need for add. delay). */ + + + INT rescal; + INT sf_IntBuffer; + + union { /* Bitstream data */ + MPEG_PS_BS_DATA mpeg; /*!< Struct containing all MPEG specific PS data from bitstream. */ + } bsData[(1)+1]; + + shouldBeUnion { /* Static data */ + struct { + SCHAR aIidPrevFrameIndex[NO_HI_RES_IID_BINS]; /*!< The IID index for previous frame */ + SCHAR aIccPrevFrameIndex[NO_HI_RES_ICC_BINS]; /*!< The ICC index for previous frame */ + + UCHAR delayBufIndex; /*!< Pointer to where the latest sample is in buffer */ + UCHAR noSampleDelay; /*!< How many QMF samples delay is used. */ + UCHAR lastUsb; /*!< uppermost WMF delay band of last frame */ + + UCHAR aDelayRBufIndexSer[NO_SERIAL_ALLPASS_LINKS]; /*!< Delay buffer for reverb filter */ + UCHAR aDelayBufIndexDelayQmf[NO_QMF_CHANNELS-FIRST_DELAY_SB]; /*!< Delay buffer for ICC group 20 & 21 */ + + SCHAR scaleFactorPsDelayBuffer; /*!< Scale factor for ps delay buffer */ + + /* hybrid filter bank delay lines */ + FIXP_DBL aaQmfDelayBufReal[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)]; + FIXP_DBL aaQmfDelayBufImag[(NO_QMF_CHANNELS-FIRST_DELAY_SB) + (MAX_DELAY_BUFFER_SIZE-1)*(NO_DELAY_BUFFER_BANDS-FIRST_DELAY_SB)]; + + FIXP_DBL *pAaRealDelayBufferQmf[MAX_DELAY_BUFFER_SIZE]; /*!< Real part delay buffer */ + FIXP_DBL *pAaImagDelayBufferQmf[MAX_DELAY_BUFFER_SIZE]; /*!< Imaginary part delay buffer */ + + FIXP_DBL aaRealDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Real part delay buffer */ + FIXP_DBL aaImagDelayBufferQmf[NO_SAMPLE_DELAY_ALLPASS][FIRST_DELAY_SB]; /*!< Imaginary part delay buffer*/ + + FIXP_DBL aaRealDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS]; /*!< Real part delay buffer */ + FIXP_DBL aaImagDelayBufferSubQmf[NO_SAMPLE_DELAY_ALLPASS][NO_SUB_QMF_CHANNELS]; /*!< Imaginary part delay buffer */ + + FIXP_DBL aaaRealDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS]; /*!< Real part delay buffer */ + FIXP_DBL aaaImagDelayRBufferSerQmf[FIRST_DELAY_SB][NO_DELAY_LENGTH_VECTORS]; /*!< Imaginary part delay buffer */ + + FIXP_DBL aaaRealDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS]; /*!< Real part delay buffer */ + FIXP_DBL aaaImagDelayRBufferSerSubQmf[NO_SUB_QMF_CHANNELS][NO_DELAY_LENGTH_VECTORS]; /*!< Imaginary part delay buffer */ + + HYBRID hybrid; /*!< hybrid filter bank struct 1 or 2. */ + + FIXP_DBL aPrevNrgBin[NO_MID_RES_BINS]; /*!< energy of previous frame */ + FIXP_DBL aPrevPeakDiffBin[NO_MID_RES_BINS]; /*!< peak difference of previous frame */ + FIXP_DBL aPeakDecayFastBin[NO_MID_RES_BINS]; /*!< Saved max. peak decay value per bin */ + SCHAR aPowerPrevScal[NO_MID_RES_BINS]; /*!< Last power value (each bin) of previous frame */ + + FIXP_DBL h11rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ + FIXP_DBL h12rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ + FIXP_DBL h21rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ + FIXP_DBL h22rPrev[NO_IID_GROUPS]; /*!< previous calculated h(xy) coefficients */ + + PS_DEC_COEFFICIENTS coef; /*!< temporal coefficients (reusable scratch memory) */ + + } mpeg; + + } specificTo; + + +}; + +typedef struct PS_DEC *HANDLE_PS_DEC; + + +int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, int aacSamplesPerFrame); + +int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC); + +void +scalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */ + FIXP_DBL **fixpQmfReal, /* qmf filterbank values */ + FIXP_DBL **fixpQmfImag, /* qmf filterbank values */ + int lsb, /* sbr start subband */ + int scaleFactorLowBandSplitLow, + int scaleFactorLowBandSplitHigh, + SCHAR *scaleFactorLowBand_lb, + SCHAR *scaleFactorLowBand_hb, + int scaleFactorHighBands, + INT *scaleFactorHighBand, + INT noCols); + +void +rescalFilterBankValues( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */ + FIXP_DBL **QmfBufferReal, /* qmf filterbank values */ + FIXP_DBL **QmfBufferImag, /* qmf filterbank values */ + int lsb, /* sbr start subband */ + INT noCols); + + +void +initSlotBasedRotation( HANDLE_PS_DEC h_ps_d, + int env, + int usb); + +void +ApplyPsSlot( HANDLE_PS_DEC h_ps_d, /* parametric stereo decoder handle */ + FIXP_DBL **rIntBufferLeft, /* real values of left qmf timeslot */ + FIXP_DBL **iIntBufferLeft, /* imag values of left qmf timeslot */ + FIXP_DBL *rIntBufferRight, /* real values of right qmf timeslot */ + FIXP_DBL *iIntBufferRight); /* imag values of right qmf timeslot */ + + + +#endif /* __PSDEC_H */ |