From 0e5af65c467b2423a0b857ae3ad98c91acc1e190 Mon Sep 17 00:00:00 2001 From: "Matthias P. Braendli" Date: Mon, 11 Nov 2019 11:38:02 +0100 Subject: Include patched FDK-AAC in the repository The initial idea was to get the DAB+ patch into upstream, but since that follows the android source releases, there is no place for a custom DAB+ patch there. So instead of having to maintain a patched fdk-aac that has to have the same .so version as the distribution package on which it is installed, we prefer having a separate fdk-aac-dab library to avoid collision. At that point, there's no reason to keep fdk-aac in a separate repository, as odr-audioenc is the only tool that needs DAB+ encoding support. Including it here simplifies installation, and makes it consistent with toolame-dab, also shipped in this repository. DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop, welle.io and others) can be done using upstream FDK-AAC. --- fdk-aac/libSBRdec/src/psdec.cpp | 722 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 722 insertions(+) create mode 100644 fdk-aac/libSBRdec/src/psdec.cpp (limited to 'fdk-aac/libSBRdec/src/psdec.cpp') diff --git a/fdk-aac/libSBRdec/src/psdec.cpp b/fdk-aac/libSBRdec/src/psdec.cpp new file mode 100644 index 0000000..b31b310 --- /dev/null +++ b/fdk-aac/libSBRdec/src/psdec.cpp @@ -0,0 +1,722 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/**************************** SBR decoder library ****************************** + + Author(s): + + Description: + +*******************************************************************************/ + +/*! + \file + \brief parametric stereo decoder +*/ + +#include "psdec.h" + +#include "FDK_bitbuffer.h" + +#include "sbr_rom.h" +#include "sbr_ram.h" + +#include "FDK_tools_rom.h" + +#include "genericStds.h" + +#include "FDK_trigFcts.h" + +/********************************************************************/ +/* MLQUAL DEFINES */ +/********************************************************************/ + +#define FRACT_ZERO FRACT_BITS - 1 +/********************************************************************/ + +SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d); + +/***** HELPERS *****/ + +/***************************************************************************/ +/*! + \brief Creates one instance of the PS_DEC struct + + \return Error info + +****************************************************************************/ +int CreatePsDec(HANDLE_PS_DEC *h_PS_DEC, /*!< pointer to the module state */ + int aacSamplesPerFrame) { + SBR_ERROR errorInfo = SBRDEC_OK; + HANDLE_PS_DEC h_ps_d; + int i; + + if (*h_PS_DEC == NULL) { + /* Get ps dec ram */ + h_ps_d = GetRam_ps_dec(); + if (h_ps_d == NULL) { + goto bail; + } + } else { + /* Reset an open instance */ + h_ps_d = *h_PS_DEC; + } + + /* + * Create Analysis Hybrid filterbank. + */ + FDKhybridAnalysisOpen(&h_ps_d->specificTo.mpeg.hybridAnalysis, + h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx, + sizeof(h_ps_d->specificTo.mpeg.pHybridAnaStatesLFdmx), + NULL, 0); + + /* initialisation */ + switch (aacSamplesPerFrame) { + case 960: + h_ps_d->noSubSamples = 30; /* col */ + break; + case 1024: + h_ps_d->noSubSamples = 32; /* col */ + break; + default: + h_ps_d->noSubSamples = -1; + break; + } + + if (h_ps_d->noSubSamples > MAX_NUM_COL || h_ps_d->noSubSamples <= 0) { + goto bail; + } + h_ps_d->noChannels = NO_QMF_CHANNELS; /* row */ + + h_ps_d->psDecodedPrv = 0; + h_ps_d->procFrameBased = -1; + for (i = 0; i < (1) + 1; i++) { + h_ps_d->bPsDataAvail[i] = ppt_none; + } + { + int error; + error = FDKdecorrelateOpen(&(h_ps_d->specificTo.mpeg.apDecor), + h_ps_d->specificTo.mpeg.decorrBufferCplx, + (2 * ((825) + (373)))); + if (error) goto bail; + } + + for (i = 0; i < (1) + 1; i++) { + FDKmemclear(&h_ps_d->bsData[i].mpeg, sizeof(MPEG_PS_BS_DATA)); + } + + errorInfo = ResetPsDec(h_ps_d); + + if (errorInfo != SBRDEC_OK) goto bail; + + *h_PS_DEC = h_ps_d; + + return 0; + +bail: + if (h_ps_d != NULL) { + DeletePsDec(&h_ps_d); + } + + return -1; +} /*END CreatePsDec */ + +/***************************************************************************/ +/*! + \brief Delete one instance of the PS_DEC struct + + \return Error info + +****************************************************************************/ +int DeletePsDec(HANDLE_PS_DEC *h_PS_DEC) /*!< pointer to the module state */ +{ + if (*h_PS_DEC == NULL) { + return -1; + } + + { + HANDLE_PS_DEC h_ps_d = *h_PS_DEC; + FDKdecorrelateClose(&(h_ps_d->specificTo.mpeg.apDecor)); + } + + FreeRam_ps_dec(h_PS_DEC); + + return 0; +} /*END DeletePsDec */ + +/***************************************************************************/ +/*! + \brief resets some values of the PS handle to default states + + \return + +****************************************************************************/ +SBR_ERROR ResetPsDec(HANDLE_PS_DEC h_ps_d) /*!< pointer to the module state */ +{ + SBR_ERROR errorInfo = SBRDEC_OK; + INT i; + + /* explicitly init state variables to safe values (until first ps header + * arrives) */ + + h_ps_d->specificTo.mpeg.lastUsb = 0; + + /* + * Initialize Analysis Hybrid filterbank. + */ + FDKhybridAnalysisInit(&h_ps_d->specificTo.mpeg.hybridAnalysis, THREE_TO_TEN, + NO_QMF_BANDS_HYBRID20, NO_QMF_BANDS_HYBRID20, 1); + + /* + * Initialize Synthesis Hybrid filterbank. + */ + for (i = 0; i < 2; i++) { + FDKhybridSynthesisInit(&h_ps_d->specificTo.mpeg.hybridSynthesis[i], + THREE_TO_TEN, NO_QMF_CHANNELS, NO_QMF_CHANNELS); + } + { + INT error; + error = FDKdecorrelateInit(&h_ps_d->specificTo.mpeg.apDecor, 71, DECORR_PS, + DUCKER_AUTOMATIC, 0, 0, 0, 0, 1, /* isLegacyPS */ + 1); + if (error) return SBRDEC_NOT_INITIALIZED; + } + + for (i = 0; i < NO_IID_GROUPS; i++) { + h_ps_d->specificTo.mpeg.h11rPrev[i] = FL2FXCONST_DBL(0.5f); + h_ps_d->specificTo.mpeg.h12rPrev[i] = FL2FXCONST_DBL(0.5f); + } + + FDKmemclear(h_ps_d->specificTo.mpeg.h21rPrev, + sizeof(h_ps_d->specificTo.mpeg.h21rPrev)); + FDKmemclear(h_ps_d->specificTo.mpeg.h22rPrev, + sizeof(h_ps_d->specificTo.mpeg.h22rPrev)); + + return errorInfo; +} + +/***************************************************************************/ +/*! + \brief Feed delaylines when parametric stereo is switched on. + \return +****************************************************************************/ +void PreparePsProcessing(HANDLE_PS_DEC h_ps_d, + const FIXP_DBL *const *const rIntBufferLeft, + const FIXP_DBL *const *const iIntBufferLeft, + const int scaleFactorLowBand) { + if (h_ps_d->procFrameBased == + 1) /* If we have switched from frame to slot based processing */ + { /* fill hybrid delay buffer. */ + int i, j; + + for (i = 0; i < HYBRID_FILTER_DELAY; i++) { + FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20]; + FIXP_DBL hybridOutputData[2][NO_SUB_QMF_CHANNELS]; + + for (j = 0; j < NO_QMF_BANDS_HYBRID20; j++) { + qmfInputData[0][j] = + scaleValue(rIntBufferLeft[i][j], scaleFactorLowBand); + qmfInputData[1][j] = + scaleValue(iIntBufferLeft[i][j], scaleFactorLowBand); + } + + FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis, + qmfInputData[0], qmfInputData[1], + hybridOutputData[0], hybridOutputData[1]); + } + h_ps_d->procFrameBased = 0; /* switch to slot based processing. */ + + } /* procFrameBased==1 */ +} + +void initSlotBasedRotation( + HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ + int env, int usb) { + INT group = 0; + INT bin = 0; + INT noIidSteps, noFactors; + + FIXP_SGL invL; + FIXP_DBL ScaleL, ScaleR; + FIXP_DBL Alpha, Beta, AlphasValue; + FIXP_DBL h11r, h12r, h21r, h22r; + + const FIXP_DBL *PScaleFactors; + + if (h_ps_d->bsData[h_ps_d->processSlot].mpeg.bFineIidQ) { + PScaleFactors = ScaleFactorsFine; /* values are shiftet right by one */ + noIidSteps = NO_IID_STEPS_FINE; + noFactors = NO_IID_LEVELS_FINE; + } else { + PScaleFactors = ScaleFactors; /* values are shiftet right by one */ + noIidSteps = NO_IID_STEPS; + noFactors = NO_IID_LEVELS; + } + + /* dequantize and decode */ + for (group = 0; group < NO_IID_GROUPS; group++) { + bin = bins2groupMap20[group]; + + /*! +

type 'A' rotation

+ mixing procedure R_a, used in baseline version
+ + Scale-factor vectors c1 and c2 are precalculated in initPsTables () and + stored in scaleFactors[] and scaleFactorsFine[] = pScaleFactors []. From the + linearized IID parameters (intensity differences), two scale factors are + calculated. They are used to obtain the coefficients h11... h22. + */ + + /* ScaleR and ScaleL are scaled by 1 shift right */ + + ScaleL = ScaleR = 0; + if (noIidSteps + h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] >= 0 && noIidSteps + h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] < noFactors) + ScaleR = PScaleFactors[noIidSteps + h_ps_d->specificTo.mpeg.pCoef + ->aaIidIndexMapped[env][bin]]; + if (noIidSteps - h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] >= 0 && noIidSteps - h_ps_d->specificTo.mpeg.pCoef->aaIidIndexMapped[env][bin] < noFactors) + ScaleL = PScaleFactors[noIidSteps - h_ps_d->specificTo.mpeg.pCoef + ->aaIidIndexMapped[env][bin]]; + + AlphasValue = 0; + if (h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin] >= 0) + AlphasValue = Alphas[h_ps_d->specificTo.mpeg.pCoef->aaIccIndexMapped[env][bin]]; + Beta = fMult( + fMult(AlphasValue, + (ScaleR - ScaleL)), + FIXP_SQRT05); + Alpha = + AlphasValue >> 1; + + /* Alpha and Beta are now both scaled by 2 shifts right */ + + /* calculate the coefficients h11... h22 from scale-factors and ICC + * parameters */ + + /* h values are scaled by 1 shift right */ + { + FIXP_DBL trigData[4]; + + inline_fixp_cos_sin(Beta + Alpha, Beta - Alpha, 2, trigData); + h11r = fMult(ScaleL, trigData[0]); + h12r = fMult(ScaleR, trigData[2]); + h21r = fMult(ScaleL, trigData[1]); + h22r = fMult(ScaleR, trigData[3]); + } + /*****************************************************************************************/ + /* Interpolation of the matrices H11... H22: */ + /* */ + /* H11(k,n) = H11(k,n[e]) + (n-n[e]) * (H11(k,n[e+1] - H11(k,n[e])) / + * (n[e+1] - n[e]) */ + /* ... */ + /*****************************************************************************************/ + + /* invL = 1/(length of envelope) */ + invL = FX_DBL2FX_SGL(GetInvInt( + h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env + 1] - + h_ps_d->bsData[h_ps_d->processSlot].mpeg.aEnvStartStop[env])); + + h_ps_d->specificTo.mpeg.pCoef->H11r[group] = + h_ps_d->specificTo.mpeg.h11rPrev[group]; + h_ps_d->specificTo.mpeg.pCoef->H12r[group] = + h_ps_d->specificTo.mpeg.h12rPrev[group]; + h_ps_d->specificTo.mpeg.pCoef->H21r[group] = + h_ps_d->specificTo.mpeg.h21rPrev[group]; + h_ps_d->specificTo.mpeg.pCoef->H22r[group] = + h_ps_d->specificTo.mpeg.h22rPrev[group]; + + h_ps_d->specificTo.mpeg.pCoef->DeltaH11r[group] = + fMult(h11r - h_ps_d->specificTo.mpeg.pCoef->H11r[group], invL); + h_ps_d->specificTo.mpeg.pCoef->DeltaH12r[group] = + fMult(h12r - h_ps_d->specificTo.mpeg.pCoef->H12r[group], invL); + h_ps_d->specificTo.mpeg.pCoef->DeltaH21r[group] = + fMult(h21r - h_ps_d->specificTo.mpeg.pCoef->H21r[group], invL); + h_ps_d->specificTo.mpeg.pCoef->DeltaH22r[group] = + fMult(h22r - h_ps_d->specificTo.mpeg.pCoef->H22r[group], invL); + + /* update prev coefficients for interpolation in next envelope */ + + h_ps_d->specificTo.mpeg.h11rPrev[group] = h11r; + h_ps_d->specificTo.mpeg.h12rPrev[group] = h12r; + h_ps_d->specificTo.mpeg.h21rPrev[group] = h21r; + h_ps_d->specificTo.mpeg.h22rPrev[group] = h22r; + + } /* group loop */ +} + +static const UCHAR groupTable[NO_IID_GROUPS + 1] = { + 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, + 12, 13, 14, 15, 16, 18, 21, 25, 30, 42, 71}; + +static void applySlotBasedRotation( + HANDLE_PS_DEC h_ps_d, /*!< pointer to the module state */ + + FIXP_DBL *mHybridRealLeft, /*!< hybrid values real left */ + FIXP_DBL *mHybridImagLeft, /*!< hybrid values imag left */ + + FIXP_DBL *mHybridRealRight, /*!< hybrid values real right */ + FIXP_DBL *mHybridImagRight /*!< hybrid values imag right */ +) { + INT group; + INT subband; + + /**********************************************************************************************/ + /*! +

Mapping

+ + The number of stereo bands that is actually used depends on the number of + availble parameters for IID and ICC: 
 nr. of IID para.| nr. of ICC para.
+  | nr. of Stereo bands
+   ----------------|------------------|-------------------
+     10,20         |     10,20        |        20
+     10,20         |     34           |        34
+     34            |     10,20        |        34
+     34            |     34           |        34
+
+ In the case the number of parameters for IIS and ICC differs from the number + of stereo bands, a mapping from the lower number to the higher number of + parameters is applied. Index mapping of IID and ICC parameters is already done + in psbitdec.cpp. Further mapping is not needed here in baseline version. + **********************************************************************************************/ + + /************************************************************************************************/ + /*! +

Mixing

+ + To generate the QMF subband signals for the subband samples n = n[e]+1 ,,, + n_[e+1] the parameters at position n[e] and n[e+1] are required as well as the + subband domain signals s_k(n) and d_k(n) for n = n[e]+1... n_[e+1]. n[e] + represents the start position for envelope e. The border positions n[e] are + handled in DecodePS(). + + The stereo sub subband signals are constructed as: + 
+  l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n)
+  r_k(n) = H21(k,n) s_k(n) + H22(k,n) d_k(n)
+
+ In order to obtain the matrices H11(k,n)... H22 (k,n), the vectors h11(b)... + h22(b) need to be calculated first (b: parameter index). Depending on ICC mode + either mixing procedure R_a or R_b is used for that. For both procedures, the + parameters for parameter position n[e+1] is used. + ************************************************************************************************/ + + /************************************************************************************************/ + /*! +

Phase parameters

+ With disabled phase parameters (which is the case in baseline version), the + H-matrices are just calculated by: + + 
+  H11(k,n[e+1] = h11(b(k))
+  (...)
+  b(k): parameter index according to mapping table
+
+ +

Processing of the samples in the sub subbands

+ this loop includes the interpolation of the coefficients Hxx + ************************************************************************************************/ + + /******************************************************/ + /* construct stereo sub subband signals according to: */ + /* */ + /* l_k(n) = H11(k,n) s_k(n) + H21(k,n) d_k(n) */ + /* r_k(n) = H12(k,n) s_k(n) + H22(k,n) d_k(n) */ + /******************************************************/ + PS_DEC_COEFFICIENTS *pCoef = h_ps_d->specificTo.mpeg.pCoef; + + for (group = 0; group < NO_IID_GROUPS; group++) { + pCoef->H11r[group] += pCoef->DeltaH11r[group]; + pCoef->H12r[group] += pCoef->DeltaH12r[group]; + pCoef->H21r[group] += pCoef->DeltaH21r[group]; + pCoef->H22r[group] += pCoef->DeltaH22r[group]; + + const int start = groupTable[group]; + const int stop = groupTable[group + 1]; + for (subband = start; subband < stop; subband++) { + FIXP_DBL tmpLeft = + fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridRealLeft[subband]), + pCoef->H21r[group], mHybridRealRight[subband]); + FIXP_DBL tmpRight = + fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridRealLeft[subband]), + pCoef->H22r[group], mHybridRealRight[subband]); + mHybridRealLeft[subband] = tmpLeft; + mHybridRealRight[subband] = tmpRight; + + tmpLeft = + fMultAdd(fMultDiv2(pCoef->H11r[group], mHybridImagLeft[subband]), + pCoef->H21r[group], mHybridImagRight[subband]); + tmpRight = + fMultAdd(fMultDiv2(pCoef->H12r[group], mHybridImagLeft[subband]), + pCoef->H22r[group], mHybridImagRight[subband]); + mHybridImagLeft[subband] = tmpLeft; + mHybridImagRight[subband] = tmpRight; + } /* subband */ + } +} + +/***************************************************************************/ +/*! + \brief Applies IID, ICC, IPD and OPD parameters to the current frame. + + \return none + +****************************************************************************/ +void ApplyPsSlot( + HANDLE_PS_DEC h_ps_d, /*!< handle PS_DEC*/ + FIXP_DBL **rIntBufferLeft, /*!< real bands left qmf channel (38x64) */ + FIXP_DBL **iIntBufferLeft, /*!< imag bands left qmf channel (38x64) */ + FIXP_DBL *rIntBufferRight, /*!< real bands right qmf channel (38x64) */ + FIXP_DBL *iIntBufferRight, /*!< imag bands right qmf channel (38x64) */ + const int scaleFactorLowBand_no_ov, const int scaleFactorLowBand, + const int scaleFactorHighBand, const int lsb, const int usb) { +/*! +The 64-band QMF representation of the monaural signal generated by the SBR tool +is used as input of the PS tool. After the PS processing, the outputs of the +left and right hybrid synthesis filterbanks are used to generate the stereo +output signal. + +
+
+           -------------            ----------            -------------
+          | Hybrid      | M_n[k,m] |          | L_n[k,m] | Hybrid      | l[n]
+ m[n] --->| analysis    |--------->|          |--------->| synthesis   |----->
+           -------------           | Stereo   |           -------------
+                 |                 | recon-   |
+                 |                 | stuction |
+                \|/                |          |
+           -------------           |          |
+          | De-         | D_n[k,m] |          |
+          | correlation |--------->|          |
+           -------------           |          |           -------------
+                                   |          | R_n[k,m] | Hybrid      | r[n]
+                                   |          |--------->| synthesis   |----->
+ IID, ICC ------------------------>|          |          | filter bank |
+(IPD, OPD)                          ----------            -------------
+
+m[n]:      QMF represantation of the mono input
+M_n[k,m]:  (sub-)sub-band domain signals of the mono input
+D_n[k,m]:  decorrelated (sub-)sub-band domain signals
+L_n[k,m]:  (sub-)sub-band domain signals of the left output
+R_n[k,m]:  (sub-)sub-band domain signals of the right output
+l[n],r[n]: left/right output signals
+
+
+*/ +#define NO_HYBRID_DATA_BANDS (71) + + int i; + FIXP_DBL qmfInputData[2][NO_QMF_BANDS_HYBRID20]; + FIXP_DBL *hybridData[2][2]; + C_ALLOC_SCRATCH_START(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS); + + hybridData[0][0] = + pHybridData + 0 * NO_HYBRID_DATA_BANDS; /* left real hybrid data */ + hybridData[0][1] = + pHybridData + 1 * NO_HYBRID_DATA_BANDS; /* left imag hybrid data */ + hybridData[1][0] = + pHybridData + 2 * NO_HYBRID_DATA_BANDS; /* right real hybrid data */ + hybridData[1][1] = + pHybridData + 3 * NO_HYBRID_DATA_BANDS; /* right imag hybrid data */ + + /*! + Hybrid analysis filterbank: + The lower 3 (5) of the 64 QMF subbands are further split to provide better + frequency resolution. for PS processing. For the 10 and 20 stereo bands + configuration, the QMF band H_0(w) is split up into 8 (sub-) sub-bands and the + QMF bands H_1(w) and H_2(w) are spit into 2 (sub-) 4th. (See figures 8.20 + and 8.22 of ISO/IEC 14496-3:2001/FDAM 2:2004(E) ) + */ + + /* + * Hybrid analysis. + */ + + /* Get qmf input data and apply descaling */ + for (i = 0; i < NO_QMF_BANDS_HYBRID20; i++) { + qmfInputData[0][i] = scaleValue(rIntBufferLeft[HYBRID_FILTER_DELAY][i], + scaleFactorLowBand_no_ov); + qmfInputData[1][i] = scaleValue(iIntBufferLeft[HYBRID_FILTER_DELAY][i], + scaleFactorLowBand_no_ov); + } + + /* LF - part */ + FDKhybridAnalysisApply(&h_ps_d->specificTo.mpeg.hybridAnalysis, + qmfInputData[0], qmfInputData[1], hybridData[0][0], + hybridData[0][1]); + + /* HF - part */ + /* bands up to lsb */ + scaleValues(&hybridData[0][0][NO_SUB_QMF_CHANNELS - 2], + &rIntBufferLeft[0][NO_QMF_BANDS_HYBRID20], + lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand); + scaleValues(&hybridData[0][1][NO_SUB_QMF_CHANNELS - 2], + &iIntBufferLeft[0][NO_QMF_BANDS_HYBRID20], + lsb - NO_QMF_BANDS_HYBRID20, scaleFactorLowBand); + + /* bands from lsb to usb */ + scaleValues(&hybridData[0][0][lsb + (NO_SUB_QMF_CHANNELS - 2 - + NO_QMF_BANDS_HYBRID20)], + &rIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand); + scaleValues(&hybridData[0][1][lsb + (NO_SUB_QMF_CHANNELS - 2 - + NO_QMF_BANDS_HYBRID20)], + &iIntBufferLeft[0][lsb], usb - lsb, scaleFactorHighBand); + + /* bands from usb to NO_SUB_QMF_CHANNELS which should be zero for non-overlap + slots but can be non-zero for overlap slots */ + FDKmemcpy( + &hybridData[0][0] + [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)], + &rIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb)); + FDKmemcpy( + &hybridData[0][1] + [usb + (NO_SUB_QMF_CHANNELS - 2 - NO_QMF_BANDS_HYBRID20)], + &iIntBufferLeft[0][usb], sizeof(FIXP_DBL) * (NO_QMF_CHANNELS - usb)); + + /*! + Decorrelation: + By means of all-pass filtering and delaying, the (sub-)sub-band samples s_k(n) + are converted into de-correlated (sub-)sub-band samples d_k(n). + - k: frequency in hybrid spectrum + - n: time index + */ + + FDKdecorrelateApply(&h_ps_d->specificTo.mpeg.apDecor, + &hybridData[0][0][0], /* left real hybrid data */ + &hybridData[0][1][0], /* left imag hybrid data */ + &hybridData[1][0][0], /* right real hybrid data */ + &hybridData[1][1][0], /* right imag hybrid data */ + 0 /* startHybBand */ + ); + + /*! + Stereo Processing: + The sets of (sub-)sub-band samples s_k(n) and d_k(n) are processed according + to the stereo cues which are defined per stereo band. + */ + + applySlotBasedRotation(h_ps_d, + &hybridData[0][0][0], /* left real hybrid data */ + &hybridData[0][1][0], /* left imag hybrid data */ + &hybridData[1][0][0], /* right real hybrid data */ + &hybridData[1][1][0] /* right imag hybrid data */ + ); + + /*! + Hybrid synthesis filterbank: + The stereo processed hybrid subband signals l_k(n) and r_k(n) are fed into the + hybrid synthesis filterbanks which are identical to the 64 complex synthesis + filterbank of the SBR tool. The input to the filterbank are slots of 64 QMF + samples. For each slot the filterbank outputs one block of 64 samples of one + reconstructed stereo channel. The hybrid synthesis filterbank is computed + seperatly for the left and right channel. + */ + + /* + * Hybrid synthesis. + */ + for (i = 0; i < 2; i++) { + FDKhybridSynthesisApply( + &h_ps_d->specificTo.mpeg.hybridSynthesis[i], + hybridData[i][0], /* real hybrid data */ + hybridData[i][1], /* imag hybrid data */ + (i == 0) ? rIntBufferLeft[0] + : rIntBufferRight, /* output real qmf buffer */ + (i == 0) ? iIntBufferLeft[0] + : iIntBufferRight /* output imag qmf buffer */ + ); + } + + /* free temporary hybrid qmf values of one timeslot */ + C_ALLOC_SCRATCH_END(pHybridData, FIXP_DBL, 4 * NO_HYBRID_DATA_BANDS); + +} /* END ApplyPsSlot */ -- cgit v1.2.3