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diff --git a/libFDK/src/qmf.cpp b/libFDK/src/qmf.cpp deleted file mode 100644 index 54526dd..0000000 --- a/libFDK/src/qmf.cpp +++ /dev/null @@ -1,1196 +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 ------------------------------------------------------------------------------------------------------------ */ - -/******************************** Fraunhofer IIS *************************** - - Author(s): Markus Lohwasser, Josef Hoepfl, Manuel Jander - Description: QMF filterbank - -******************************************************************************/ - -/*! - \file - \brief Complex qmf analysis/synthesis, - This module contains the qmf filterbank for analysis [ cplxAnalysisQmfFiltering() ] and - synthesis [ cplxSynthesisQmfFiltering() ]. It is a polyphase implementation of a complex - exponential modulated filter bank. The analysis part usually runs at half the sample rate - than the synthesis part. (So called "dual-rate" mode.) - - The coefficients of the prototype filter are specified in #sbr_qmf_64_640 (in sbr_rom.cpp). - Thus only a 64 channel version (32 on the analysis side) with a 640 tap prototype filter - are used. - - \anchor PolyphaseFiltering <h2>About polyphase filtering</h2> - The polyphase implementation of a filterbank requires filtering at the input and output. - This is implemented as part of cplxAnalysisQmfFiltering() and cplxSynthesisQmfFiltering(). - The implementation requires the filter coefficients in a specific structure as described in - #sbr_qmf_64_640_qmf (in sbr_rom.cpp). - - This module comprises the computationally most expensive functions of the SBR decoder. The accuracy of - computations is also important and has a direct impact on the overall sound quality. Therefore a special - test program is available which can be used to only test the filterbank: main_audio.cpp - - This modules also uses scaling of data to provide better SNR on fixed-point processors. See #QMF_SCALE_FACTOR (in sbr_scale.h) for details. - An interesting note: The function getScalefactor() can constitute a significant amount of computational complexity - very much depending on the - bitrate. Since it is a rather small function, effective assembler optimization might be possible. - -*/ - -#include "qmf.h" - - -#include "fixpoint_math.h" -#include "dct.h" - -#ifdef QMFSYN_STATES_16BIT -#define QSSCALE (7) -#define FX_DBL2FX_QSS(x) ((FIXP_QSS) ((x)>>(DFRACT_BITS-QSS_BITS-QSSCALE) )) -#define FX_QSS2FX_DBL(x) ((FIXP_DBL)((LONG)x)<<(DFRACT_BITS-QSS_BITS-QSSCALE)) -#else -#define QSSCALE (0) -#define FX_DBL2FX_QSS(x) (x) -#define FX_QSS2FX_DBL(x) (x) -#endif - - -#if defined(__arm__) -#include "arm/qmf_arm.cpp" - -#endif - -/*! - * \brief Algorithmic scaling in sbrForwardModulation() - * - * The scaling in sbrForwardModulation() is caused by: - * - * \li 1 R_SHIFT in sbrForwardModulation() - * \li 5/6 R_SHIFT in dct3() if using 32/64 Bands - * \li 1 ommited gain of 2.0 in qmfForwardModulation() - */ -#define ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK 7 - -/*! - * \brief Algorithmic scaling in cplxSynthesisQmfFiltering() - * - * The scaling in cplxSynthesisQmfFiltering() is caused by: - * - * \li 5/6 R_SHIFT in dct2() if using 32/64 Bands - * \li 1 ommited gain of 2.0 in qmfInverseModulation() - * \li -6 division by 64 in synthesis filterbank - * \li x bits external influence - */ -#define ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK 1 - - -/*! - \brief Perform Synthesis Prototype Filtering on a single slot of input data. - - The filter takes 2 * qmf->no_channels of input data and - generates qmf->no_channels time domain output samples. -*/ -static -#ifndef FUNCTION_qmfSynPrototypeFirSlot -void qmfSynPrototypeFirSlot( -#else -void qmfSynPrototypeFirSlot_fallback( -#endif - HANDLE_QMF_FILTER_BANK qmf, - FIXP_QMF *RESTRICT realSlot, /*!< Input: Pointer to real Slot */ - FIXP_QMF *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */ - INT_PCM *RESTRICT timeOut, /*!< Time domain data */ - int stride - ) -{ - FIXP_QSS* FilterStates = (FIXP_QSS*)qmf->FilterStates; - int no_channels = qmf->no_channels; - const FIXP_PFT *p_Filter = qmf->p_filter; - int p_stride = qmf->p_stride; - int j; - FIXP_QSS *RESTRICT sta = FilterStates; - const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm; - int scale = ((DFRACT_BITS-SAMPLE_BITS)-1-qmf->outScalefactor); - - p_flt = p_Filter+p_stride*QMF_NO_POLY; /* 5-ter von 330 */ - p_fltm = p_Filter+(qmf->FilterSize/2)-p_stride*QMF_NO_POLY; /* 5 + (320 - 2*5) = 315-ter von 330 */ - - FDK_ASSERT(SAMPLE_BITS-1-qmf->outScalefactor >= 0); // (DFRACT_BITS-SAMPLE_BITS)-1-qmf->outScalefactor >= 0); - - for (j = no_channels-1; j >= 0; j--) { /* ---- läuft ueber alle Linien eines Slots ---- */ - FIXP_QMF imag = imagSlot[j]; // no_channels-1 .. 0 - FIXP_QMF real = realSlot[j]; // ~~"~~ - { - INT_PCM tmp; - FIXP_DBL Are = FX_QSS2FX_DBL(sta[0]) + fMultDiv2( p_fltm[0] , real); - - if (qmf->outGain!=(FIXP_DBL)0x80000000) { - Are = fMult(Are,qmf->outGain); - } - - #if SAMPLE_BITS > 16 - tmp = (INT_PCM)(SATURATE_SHIFT(fAbs(Are), scale, SAMPLE_BITS)); - #else - tmp = (INT_PCM)(SATURATE_RIGHT_SHIFT(fAbs(Are), scale, SAMPLE_BITS)); - #endif - if (Are < (FIXP_QMF)0) { - tmp = -tmp; - } - timeOut[ (j)*stride ] = tmp; - } - - sta[0] = sta[1] + FX_DBL2FX_QSS(fMultDiv2( p_flt [4] , imag )); - sta[1] = sta[2] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[1] , real )); - sta[2] = sta[3] + FX_DBL2FX_QSS(fMultDiv2( p_flt [3] , imag )); - sta[3] = sta[4] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[2] , real )); - sta[4] = sta[5] + FX_DBL2FX_QSS(fMultDiv2( p_flt [2] , imag )); - sta[5] = sta[6] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[3] , real )); - sta[6] = sta[7] + FX_DBL2FX_QSS(fMultDiv2( p_flt [1] , imag )); - sta[7] = sta[8] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[4] , real )); - sta[8] = FX_DBL2FX_QSS(fMultDiv2( p_flt [0] , imag )); - - p_flt += (p_stride*QMF_NO_POLY); - p_fltm -= (p_stride*QMF_NO_POLY); - sta += 9; // = (2*QMF_NO_POLY-1); - } -} - -#ifndef FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric -/*! - \brief Perform Synthesis Prototype Filtering on a single slot of input data. - - The filter takes 2 * qmf->no_channels of input data and - generates qmf->no_channels time domain output samples. -*/ -static -void qmfSynPrototypeFirSlot_NonSymmetric( - HANDLE_QMF_FILTER_BANK qmf, - FIXP_QMF *RESTRICT realSlot, /*!< Input: Pointer to real Slot */ - FIXP_QMF *RESTRICT imagSlot, /*!< Input: Pointer to imag Slot */ - INT_PCM *RESTRICT timeOut, /*!< Time domain data */ - int stride - ) -{ - FIXP_QSS* FilterStates = (FIXP_QSS*)qmf->FilterStates; - int no_channels = qmf->no_channels; - const FIXP_PFT *p_Filter = qmf->p_filter; - int p_stride = qmf->p_stride; - int j; - FIXP_QSS *RESTRICT sta = FilterStates; - const FIXP_PFT *RESTRICT p_flt, *RESTRICT p_fltm; - int scale = ((DFRACT_BITS-SAMPLE_BITS)-1-qmf->outScalefactor); - - p_flt = p_Filter; /*!< Pointer to first half of filter coefficients */ - p_fltm = &p_flt[qmf->FilterSize/2]; /* at index 320, overall 640 coefficients */ - - FDK_ASSERT(SAMPLE_BITS-1-qmf->outScalefactor >= 0); // (DFRACT_BITS-SAMPLE_BITS)-1-qmf->outScalefactor >= 0); - - for (j = no_channels-1; j >= 0; j--) { /* ---- läuft ueber alle Linien eines Slots ---- */ - - FIXP_QMF imag = imagSlot[j]; // no_channels-1 .. 0 - FIXP_QMF real = realSlot[j]; // ~~"~~ - { - INT_PCM tmp; - FIXP_QMF Are = sta[0] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[4] , real )); - - #if SAMPLE_BITS > 16 - tmp = (INT_PCM)(SATURATE_SHIFT(fAbs(Are), scale, SAMPLE_BITS)); - #else - tmp = (INT_PCM)(SATURATE_RIGHT_SHIFT(fAbs(Are), scale, SAMPLE_BITS)); - #endif - if (Are < (FIXP_QMF)0) { - tmp = -tmp; - } - timeOut[j*stride] = tmp; - } - - sta[0] = sta[1] + FX_DBL2FX_QSS(fMultDiv2( p_flt [4] , imag )); - sta[1] = sta[2] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[3] , real )); - sta[2] = sta[3] + FX_DBL2FX_QSS(fMultDiv2( p_flt [3] , imag )); - - sta[3] = sta[4] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[2] , real )); - sta[4] = sta[5] + FX_DBL2FX_QSS(fMultDiv2( p_flt [2] , imag )); - sta[5] = sta[6] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[1] , real )); - sta[6] = sta[7] + FX_DBL2FX_QSS(fMultDiv2( p_flt [1] , imag )); - - sta[7] = sta[8] + FX_DBL2FX_QSS(fMultDiv2( p_fltm[0] , real )); - sta[8] = FX_DBL2FX_QSS(fMultDiv2( p_flt [0] , imag )); - - p_flt += (p_stride*QMF_NO_POLY); - p_fltm += (p_stride*QMF_NO_POLY); - sta += 9; // = (2*QMF_NO_POLY-1); - } - -} -#endif /* FUNCTION_qmfSynPrototypeFirSlot_NonSymmetric */ - -#ifndef FUNCTION_qmfAnaPrototypeFirSlot -/*! - \brief Perform Analysis Prototype Filtering on a single slot of input data. -*/ -static -void qmfAnaPrototypeFirSlot( FIXP_QMF *analysisBuffer, - int no_channels, /*!< Number channels of analysis filter */ - const FIXP_PFT *p_filter, - int p_stride, /*!< Stide of analysis filter */ - FIXP_QAS *RESTRICT pFilterStates - ) -{ - int k; - - FIXP_DBL accu; - const FIXP_PFT *RESTRICT p_flt = p_filter; - FIXP_QMF *RESTRICT pData_0 = analysisBuffer + 2*no_channels - 1; - FIXP_QMF *RESTRICT pData_1 = analysisBuffer; - - FIXP_QAS *RESTRICT sta_0 = (FIXP_QAS *)pFilterStates; - FIXP_QAS *RESTRICT sta_1 = (FIXP_QAS *)pFilterStates + (2*QMF_NO_POLY*no_channels) - 1; - int pfltStep = QMF_NO_POLY * (p_stride); - int staStep1 = no_channels<<1; - int staStep2 = (no_channels<<3) - 1; /* Rewind one less */ - - /* FIR filter 0 */ - accu = fMultDiv2( p_flt[0], *sta_1); sta_1 -= staStep1; - accu += fMultDiv2( p_flt[1], *sta_1); sta_1 -= staStep1; - accu += fMultDiv2( p_flt[2], *sta_1); sta_1 -= staStep1; - accu += fMultDiv2( p_flt[3], *sta_1); sta_1 -= staStep1; - accu += fMultDiv2( p_flt[4], *sta_1); - *pData_1++ = FX_DBL2FX_QMF(accu<<1); - sta_1 += staStep2; - - p_flt += pfltStep; - - /* FIR filters 1..63 127..65 */ - for (k=0; k<no_channels-1; k++) - { - accu = fMultDiv2( p_flt[0], *sta_0); sta_0 += staStep1; - accu += fMultDiv2( p_flt[1], *sta_0); sta_0 += staStep1; - accu += fMultDiv2( p_flt[2], *sta_0); sta_0 += staStep1; - accu += fMultDiv2( p_flt[3], *sta_0); sta_0 += staStep1; - accu += fMultDiv2( p_flt[4], *sta_0); - *pData_0-- = FX_DBL2FX_QMF(accu<<1); - sta_0 -= staStep2; - - accu = fMultDiv2( p_flt[0], *sta_1); sta_1 -= staStep1; - accu += fMultDiv2( p_flt[1], *sta_1); sta_1 -= staStep1; - accu += fMultDiv2( p_flt[2], *sta_1); sta_1 -= staStep1; - accu += fMultDiv2( p_flt[3], *sta_1); sta_1 -= staStep1; - accu += fMultDiv2( p_flt[4], *sta_1); - *pData_1++ = FX_DBL2FX_QMF(accu<<1); - sta_1 += staStep2; - - p_flt += pfltStep; - } - - /* FIR filter 64 */ - accu = fMultDiv2( p_flt[0], *sta_0); sta_0 += staStep1; - accu += fMultDiv2( p_flt[1], *sta_0); sta_0 += staStep1; - accu += fMultDiv2( p_flt[2], *sta_0); sta_0 += staStep1; - accu += fMultDiv2( p_flt[3], *sta_0); sta_0 += staStep1; - accu += fMultDiv2( p_flt[4], *sta_0); - *pData_0-- = FX_DBL2FX_QMF(accu<<1); - sta_0 -= staStep2; -} -#endif /* !defined(FUNCTION_qmfAnaPrototypeFirSlot) */ - - -#ifndef FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric -/*! - \brief Perform Analysis Prototype Filtering on a single slot of input data. -*/ -static -void qmfAnaPrototypeFirSlot_NonSymmetric( - FIXP_QMF *analysisBuffer, - int no_channels, /*!< Number channels of analysis filter */ - const FIXP_PFT *p_filter, - int p_stride, /*!< Stide of analysis filter */ - FIXP_QAS *RESTRICT pFilterStates - ) -{ - const FIXP_PFT *RESTRICT p_flt = p_filter; - int p, k; - - for (k = 0; k < 2*no_channels; k++) - { - FIXP_DBL accu = (FIXP_DBL)0; - - p_flt += QMF_NO_POLY * (p_stride - 1); - - /* - Perform FIR-Filter - */ - for (p = 0; p < QMF_NO_POLY; p++) { - accu += fMultDiv2(*p_flt++, pFilterStates[2*no_channels * p]); - } - analysisBuffer[2*no_channels - 1 - k] = FX_DBL2FX_QMF(accu<<1); - pFilterStates++; - } -} -#endif /* FUNCTION_qmfAnaPrototypeFirSlot_NonSymmetric */ - -/*! - * - * \brief Perform real-valued forward modulation of the time domain - * data of timeIn and stores the real part of the subband - * samples in rSubband - * - */ -static void -qmfForwardModulationLP_even( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ - FIXP_QMF *timeIn, /*!< Time Signal */ - FIXP_QMF *rSubband ) /*!< Real Output */ -{ - int i; - int L = anaQmf->no_channels; - int M = L>>1; - int scale; - FIXP_QMF accu; - - const FIXP_QMF *timeInTmp1 = (FIXP_QMF *) &timeIn[3 * M]; - const FIXP_QMF *timeInTmp2 = timeInTmp1; - FIXP_QMF *rSubbandTmp = rSubband; - - rSubband[0] = timeIn[3 * M] >> 1; - - for (i = M-1; i != 0; i--) { - accu = ((*--timeInTmp1) >> 1) + ((*++timeInTmp2) >> 1); - *++rSubbandTmp = accu; - } - - timeInTmp1 = &timeIn[2 * M]; - timeInTmp2 = &timeIn[0]; - rSubbandTmp = &rSubband[M]; - - for (i = L-M; i != 0; i--) { - accu = ((*timeInTmp1--) >> 1) - ((*timeInTmp2++) >> 1); - *rSubbandTmp++ = accu; - } - - dct_III(rSubband, timeIn, L, &scale); -} - -#if !defined(FUNCTION_qmfForwardModulationLP_odd) -static void -qmfForwardModulationLP_odd( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ - const FIXP_QMF *timeIn, /*!< Time Signal */ - FIXP_QMF *rSubband ) /*!< Real Output */ -{ - int i; - int L = anaQmf->no_channels; - int M = L>>1; - int shift = (anaQmf->no_channels>>6) + 1; - - for (i = 0; i < M; i++) { - rSubband[M + i] = (timeIn[L - 1 - i]>>1) - (timeIn[i]>>shift); - rSubband[M - 1 - i] = (timeIn[L + i]>>1) + (timeIn[2 * L - 1 - i]>>shift); - } - - dct_IV(rSubband, L, &shift); -} -#endif /* !defined(FUNCTION_qmfForwardModulationLP_odd) */ - - - -/*! - * - * \brief Perform complex-valued forward modulation of the time domain - * data of timeIn and stores the real part of the subband - * samples in rSubband, and the imaginary part in iSubband - * - * Only the lower bands are obtained (upto anaQmf->lsb). For - * a full bandwidth analysis it is required to set both anaQmf->lsb - * and anaQmf->usb to the amount of QMF bands. - * - */ -static void -qmfForwardModulationHQ( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ - const FIXP_QMF *RESTRICT timeIn, /*!< Time Signal */ - FIXP_QMF *RESTRICT rSubband, /*!< Real Output */ - FIXP_QMF *RESTRICT iSubband /*!< Imaginary Output */ - ) -{ - int i; - int L = anaQmf->no_channels; - int L2 = L<<1; - int shift = 0; - - for (i = 0; i < L; i+=2) { - FIXP_QMF x0, x1, y0, y1; - - x0 = timeIn[i] >> 1; - x1 = timeIn[i+1] >> 1; - y0 = timeIn[L2 - 1 - i] >> 1; - y1 = timeIn[L2 - 2 - i] >> 1; - - rSubband[i] = x0 - y0; - rSubband[i+1] = x1 - y1; - iSubband[i] = x0 + y0; - iSubband[i+1] = x1 + y1; - } - - dct_IV(rSubband, L, &shift); - dst_IV(iSubband, L, &shift); - - { - { - const FIXP_QTW *RESTRICT sbr_t_cos; - const FIXP_QTW *RESTRICT sbr_t_sin; - sbr_t_cos = anaQmf->t_cos; - sbr_t_sin = anaQmf->t_sin; - - for (i = 0; i < anaQmf->lsb; i++) { - cplxMult(&iSubband[i], &rSubband[i], iSubband[i], rSubband[i], sbr_t_cos[i], sbr_t_sin[i]); - } - } - } -} - -/* - * \brief Perform one QMF slot analysis of the time domain data of timeIn - * with specified stride and stores the real part of the subband - * samples in rSubband, and the imaginary part in iSubband - * - * Only the lower bands are obtained (upto anaQmf->lsb). For - * a full bandwidth analysis it is required to set both anaQmf->lsb - * and anaQmf->usb to the amount of QMF bands. - */ -void -qmfAnalysisFilteringSlot( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ - FIXP_QMF *qmfReal, /*!< Low and High band, real */ - FIXP_QMF *qmfImag, /*!< Low and High band, imag */ - const INT_PCM *RESTRICT timeIn, /*!< Pointer to input */ - const int stride, /*!< stride factor of input */ - FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ - ) -{ - int i; - int offset = anaQmf->no_channels*(QMF_NO_POLY*2-1); - /* - Feed time signal into oldest anaQmf->no_channels states - */ - { - FIXP_QAS *RESTRICT FilterStatesAnaTmp = ((FIXP_QAS*)anaQmf->FilterStates)+offset; - - /* Feed and scale actual time in slot */ - for(i=anaQmf->no_channels>>1; i!=0; i--) { - /* Place INT_PCM value left aligned in scaledTimeIn */ -#if (QAS_BITS==SAMPLE_BITS) - *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn; timeIn += stride; - *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn; timeIn += stride; -#elif (QAS_BITS>SAMPLE_BITS) - *FilterStatesAnaTmp++ = (FIXP_QAS)((*timeIn)<<(QAS_BITS-SAMPLE_BITS)); timeIn += stride; - *FilterStatesAnaTmp++ = (FIXP_QAS)((*timeIn)<<(QAS_BITS-SAMPLE_BITS)); timeIn += stride; -#else - *FilterStatesAnaTmp++ = (FIXP_QAS)((*timeIn)>>(SAMPLE_BITS-QAS_BITS)); timeIn += stride; - *FilterStatesAnaTmp++ = (FIXP_QAS)((*timeIn)>>(SAMPLE_BITS-QAS_BITS)); timeIn += stride; -#endif - } - } - - if (anaQmf->flags & QMF_FLAG_NONSYMMETRIC) { - qmfAnaPrototypeFirSlot_NonSymmetric( - pWorkBuffer, - anaQmf->no_channels, - anaQmf->p_filter, - anaQmf->p_stride, - (FIXP_QAS*)anaQmf->FilterStates - ); - } else { - qmfAnaPrototypeFirSlot( pWorkBuffer, - anaQmf->no_channels, - anaQmf->p_filter, - anaQmf->p_stride, - (FIXP_QAS*)anaQmf->FilterStates - ); - } - - if (anaQmf->flags & QMF_FLAG_LP) { - if (anaQmf->flags & QMF_FLAG_CLDFB) - qmfForwardModulationLP_odd( anaQmf, - pWorkBuffer, - qmfReal ); - else - qmfForwardModulationLP_even( anaQmf, - pWorkBuffer, - qmfReal ); - - } else { - qmfForwardModulationHQ( anaQmf, - pWorkBuffer, - qmfReal, - qmfImag - ); - } - /* - Shift filter states - - Should be realized with modulo adressing on a DSP instead of a true buffer shift - */ - FDKmemmove ((FIXP_QAS*)anaQmf->FilterStates, (FIXP_QAS*)anaQmf->FilterStates+anaQmf->no_channels, offset*sizeof(FIXP_QAS)); -} - - -/*! - * - * \brief Perform complex-valued subband filtering of the time domain - * data of timeIn and stores the real part of the subband - * samples in rAnalysis, and the imaginary part in iAnalysis - * The qmf coefficient table is symmetric. The symmetry is expoited by - * shrinking the coefficient table to half the size. The addressing mode - * takes care of the symmetries. - * - * Only the lower bands are obtained (upto anaQmf->lsb). For - * a full bandwidth analysis it is required to set both anaQmf->lsb - * and anaQmf->usb to the amount of QMF bands. - * - * \sa PolyphaseFiltering - */ - -void -qmfAnalysisFiltering( HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ - FIXP_QMF **qmfReal, /*!< Pointer to real subband slots */ - FIXP_QMF **qmfImag, /*!< Pointer to imag subband slots */ - QMF_SCALE_FACTOR *scaleFactor, - const INT_PCM *timeIn, /*!< Time signal */ - const int stride, - FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ - ) -{ - int i; - int no_channels = anaQmf->no_channels; - - scaleFactor->lb_scale = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK; - scaleFactor->lb_scale -= anaQmf->filterScale; - - for (i = 0; i < anaQmf->no_col; i++) - { - FIXP_QMF *qmfImagSlot = NULL; - - if (!(anaQmf->flags & QMF_FLAG_LP)) { - qmfImagSlot = qmfImag[i]; - } - - qmfAnalysisFilteringSlot( anaQmf, qmfReal[i], qmfImagSlot, timeIn , stride, pWorkBuffer ); - - timeIn += no_channels*stride; - - } /* no_col loop i */ -} - -/*! - * - * \brief Perform low power inverse modulation of the subband - * samples stored in rSubband (real part) and iSubband (imaginary - * part) and stores the result in pWorkBuffer. - * - */ -inline -static void -qmfInverseModulationLP_even( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ - const FIXP_QMF *qmfReal, /*!< Pointer to qmf real subband slot (input) */ - const int scaleFactorLowBand, /*!< Scalefactor for Low band */ - const int scaleFactorHighBand, /*!< Scalefactor for High band */ - FIXP_QMF *pTimeOut /*!< Pointer to qmf subband slot (output)*/ - ) -{ - int i; - int L = synQmf->no_channels; - int M = L>>1; - int scale; - FIXP_QMF tmp; - FIXP_QMF *RESTRICT tReal = pTimeOut; - FIXP_QMF *RESTRICT tImag = pTimeOut + L; - - /* Move input to output vector with offset */ - scaleValues(&tReal[0], &qmfReal[0], synQmf->lsb, scaleFactorLowBand); - scaleValues(&tReal[0+synQmf->lsb], &qmfReal[0+synQmf->lsb], synQmf->usb-synQmf->lsb, scaleFactorHighBand); - FDKmemclear(&tReal[0+synQmf->usb], (L-synQmf->usb)*sizeof(FIXP_QMF)); - - /* Dct type-2 transform */ - dct_II(tReal, tImag, L, &scale); - - /* Expand output and replace inplace the output buffers */ - tImag[0] = tReal[M]; - tImag[M] = (FIXP_QMF)0; - tmp = tReal [0]; - tReal [0] = tReal[M]; - tReal [M] = tmp; - - for (i = 1; i < M/2; i++) { - /* Imag */ - tmp = tReal[L - i]; - tImag[M - i] = tmp; - tImag[i + M] = -tmp; - - tmp = tReal[M + i]; - tImag[i] = tmp; - tImag[L - i] = -tmp; - - /* Real */ - tReal [M + i] = tReal[i]; - tReal [L - i] = tReal[M - i]; - tmp = tReal[i]; - tReal[i] = tReal [M - i]; - tReal [M - i] = tmp; - - } - /* Remaining odd terms */ - tmp = tReal[M + M/2]; - tImag[M/2] = tmp; - tImag[M/2 + M] = -tmp; - - tReal [M + M/2] = tReal[M/2]; -} - -inline -static void -qmfInverseModulationLP_odd( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ - const FIXP_QMF *qmfReal, /*!< Pointer to qmf real subband slot (input) */ - const int scaleFactorLowBand, /*!< Scalefactor for Low band */ - const int scaleFactorHighBand, /*!< Scalefactor for High band */ - FIXP_QMF *pTimeOut /*!< Pointer to qmf subband slot (output)*/ - ) -{ - int i; - int L = synQmf->no_channels; - int M = L>>1; - int shift = 0; - - /* Move input to output vector with offset */ - scaleValues(pTimeOut+M, qmfReal, synQmf->lsb, scaleFactorLowBand); - scaleValues(pTimeOut+M+synQmf->lsb, qmfReal+synQmf->lsb, synQmf->usb-synQmf->lsb, scaleFactorHighBand); - FDKmemclear(pTimeOut+M+synQmf->usb, (L-synQmf->usb)*sizeof(FIXP_QMF)); - - dct_IV(pTimeOut+M, L, &shift); - for (i = 0; i < M; i++) { - pTimeOut[i] = pTimeOut[L - 1 - i]; - pTimeOut[2 * L - 1 - i] = -pTimeOut[L + i]; - } -} - - -/*! - * - * \brief Perform complex-valued inverse modulation of the subband - * samples stored in rSubband (real part) and iSubband (imaginary - * part) and stores the result in pWorkBuffer. - * - */ -inline -static void -qmfInverseModulationHQ( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ - const FIXP_QMF *qmfReal, /*!< Pointer to qmf real subband slot */ - const FIXP_QMF *qmfImag, /*!< Pointer to qmf imag subband slot */ - const int scaleFactorLowBand, /*!< Scalefactor for Low band */ - const int scaleFactorHighBand,/*!< Scalefactor for High band */ - FIXP_QMF *pWorkBuffer /*!< WorkBuffer (output) */ - ) -{ - int i; - int L = synQmf->no_channels; - int M = L>>1; - int shift = 0; - FIXP_QMF *RESTRICT tReal = pWorkBuffer; - FIXP_QMF *RESTRICT tImag = pWorkBuffer+L; - - if (synQmf->flags & QMF_FLAG_CLDFB){ - for (i = 0; i < synQmf->lsb; i++) { - cplxMult(&tImag[i], &tReal[i], - scaleValue(qmfImag[i],scaleFactorLowBand), scaleValue(qmfReal[i],scaleFactorLowBand), - synQmf->t_cos[i], synQmf->t_sin[i]); - } - for (; i < synQmf->usb; i++) { - cplxMult(&tImag[i], &tReal[i], - scaleValue(qmfImag[i],scaleFactorHighBand), scaleValue(qmfReal[i],scaleFactorHighBand), - synQmf->t_cos[i], synQmf->t_sin[i]); - } - } - - if ( (synQmf->flags & QMF_FLAG_CLDFB) == 0) { - scaleValues(&tReal[0], &qmfReal[0], synQmf->lsb, scaleFactorLowBand); - scaleValues(&tReal[0+synQmf->lsb], &qmfReal[0+synQmf->lsb], synQmf->usb-synQmf->lsb, scaleFactorHighBand); - scaleValues(&tImag[0], &qmfImag[0], synQmf->lsb, scaleFactorLowBand); - scaleValues(&tImag[0+synQmf->lsb], &qmfImag[0+synQmf->lsb], synQmf->usb-synQmf->lsb, scaleFactorHighBand); - } - - FDKmemclear(&tReal[synQmf->usb], (synQmf->no_channels-synQmf->usb)*sizeof(FIXP_QMF)); - FDKmemclear(&tImag[synQmf->usb], (synQmf->no_channels-synQmf->usb)*sizeof(FIXP_QMF)); - - dct_IV(tReal, L, &shift); - dst_IV(tImag, L, &shift); - - if (synQmf->flags & QMF_FLAG_CLDFB){ - for (i = 0; i < M; i++) { - FIXP_QMF r1, i1, r2, i2; - r1 = tReal[i]; - i2 = tImag[L - 1 - i]; - r2 = tReal[L - i - 1]; - i1 = tImag[i]; - - tReal[i] = (r1 - i1)>>1; - tImag[L - 1 - i] = -(r1 + i1)>>1; - tReal[L - i - 1] = (r2 - i2)>>1; - tImag[i] = -(r2 + i2)>>1; - } - } else - { - /* The array accesses are negative to compensate the missing minus sign in the low and hi band gain. */ - /* 26 cycles on ARM926 */ - for (i = 0; i < M; i++) { - FIXP_QMF r1, i1, r2, i2; - r1 = -tReal[i]; - i2 = -tImag[L - 1 - i]; - r2 = -tReal[L - i - 1]; - i1 = -tImag[i]; - - tReal[i] = (r1 - i1)>>1; - tImag[L - 1 - i] = -(r1 + i1)>>1; - tReal[L - i - 1] = (r2 - i2)>>1; - tImag[i] = -(r2 + i2)>>1; - } - } -} - -void qmfSynthesisFilteringSlot( HANDLE_QMF_FILTER_BANK synQmf, - const FIXP_QMF *realSlot, - const FIXP_QMF *imagSlot, - const int scaleFactorLowBand, - const int scaleFactorHighBand, - INT_PCM *timeOut, - const int stride, - FIXP_QMF *pWorkBuffer) -{ - if (!(synQmf->flags & QMF_FLAG_LP)) - qmfInverseModulationHQ ( synQmf, - realSlot, - imagSlot, - scaleFactorLowBand, - scaleFactorHighBand, - pWorkBuffer - ); - else - { - if (synQmf->flags & QMF_FLAG_CLDFB) { - qmfInverseModulationLP_odd ( synQmf, - realSlot, - scaleFactorLowBand, - scaleFactorHighBand, - pWorkBuffer - ); - } else { - qmfInverseModulationLP_even ( synQmf, - realSlot, - scaleFactorLowBand, - scaleFactorHighBand, - pWorkBuffer - ); - } - } - - if (synQmf->flags & QMF_FLAG_NONSYMMETRIC) { - qmfSynPrototypeFirSlot_NonSymmetric ( - synQmf, - pWorkBuffer, - pWorkBuffer+synQmf->no_channels, - timeOut, - stride - ); - } else { - qmfSynPrototypeFirSlot ( synQmf, - pWorkBuffer, - pWorkBuffer+synQmf->no_channels, - timeOut, - stride - ); - } -} - - -/*! - * - * - * \brief Perform complex-valued subband synthesis of the - * low band and the high band and store the - * time domain data in timeOut - * - * First step: Calculate the proper scaling factor of current - * spectral data in qmfReal/qmfImag, old spectral data in the overlap - * range and filter states. - * - * Second step: Perform Frequency-to-Time mapping with inverse - * Modulation slot-wise. - * - * Third step: Perform FIR-filter slot-wise. To save space for filter - * states, the MAC operations are executed directly on the filter states - * instead of accumulating several products in the accumulator. The - * buffer shift at the end of the function should be replaced by a - * modulo operation, which is available on some DSPs. - * - * Last step: Copy the upper part of the spectral data to the overlap buffer. - * - * The qmf coefficient table is symmetric. The symmetry is exploited by - * shrinking the coefficient table to half the size. The addressing mode - * takes care of the symmetries. If the #define #QMFTABLE_FULL is set, - * coefficient addressing works on the full table size. The code will be - * slightly faster and slightly more compact. - * - * Workbuffer requirement: 2 x sizeof(**QmfBufferReal) * synQmf->no_channels - */ -void -qmfSynthesisFiltering( HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ - FIXP_QMF **QmfBufferReal, /*!< Low and High band, real */ - FIXP_QMF **QmfBufferImag, /*!< Low and High band, imag */ - const QMF_SCALE_FACTOR *scaleFactor, - const INT ov_len, /*!< split Slot of overlap and actual slots */ - INT_PCM *timeOut, /*!< Pointer to output */ - const INT stride, /*!< stride factor of output */ - FIXP_QMF *pWorkBuffer /*!< pointer to temporal working buffer */ - ) -{ - int i; - int L = synQmf->no_channels; - SCHAR scaleFactorHighBand; - SCHAR scaleFactorLowBand_ov, scaleFactorLowBand_no_ov; - - /* adapt scaling */ - scaleFactorHighBand = -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - scaleFactor->hb_scale; - scaleFactorLowBand_ov = - ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - scaleFactor->ov_lb_scale; - scaleFactorLowBand_no_ov = - ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - scaleFactor->lb_scale; - - for (i = 0; i < synQmf->no_col; i++) /* ----- no_col loop ----- */ - { - const FIXP_DBL *QmfBufferImagSlot = NULL; - - SCHAR scaleFactorLowBand = (i<ov_len) ? scaleFactorLowBand_ov : scaleFactorLowBand_no_ov; - - if (!(synQmf->flags & QMF_FLAG_LP)) - QmfBufferImagSlot = QmfBufferImag[i]; - - qmfSynthesisFilteringSlot( synQmf, - QmfBufferReal[i], - QmfBufferImagSlot, - scaleFactorLowBand, - scaleFactorHighBand, - timeOut+(i*L*stride), - stride, - pWorkBuffer); - } /* no_col loop i */ - -} - - -/*! - * - * \brief Create QMF filter bank instance - * - * \return 0 if successful - * - */ -static int -qmfInitFilterBank (HANDLE_QMF_FILTER_BANK h_Qmf, /*!< Handle to return */ - void *pFilterStates, /*!< Handle to filter states */ - int noCols, /*!< Number of timeslots per frame */ - int lsb, /*!< Lower end of QMF frequency range */ - int usb, /*!< Upper end of QMF frequency range */ - int no_channels, /*!< Number of channels (bands) */ - UINT flags) /*!< flags */ -{ - FDKmemclear(h_Qmf,sizeof(QMF_FILTER_BANK)); - - if (flags & QMF_FLAG_MPSLDFB) - { - return -1; - } - - if ( !(flags & QMF_FLAG_MPSLDFB) && (flags & QMF_FLAG_CLDFB) ) - { - flags |= QMF_FLAG_NONSYMMETRIC; - h_Qmf->filterScale = QMF_CLDFB_PFT_SCALE; - - h_Qmf->p_stride = 1; - switch (no_channels) { - case 64: - h_Qmf->t_cos = qmf_phaseshift_cos64_cldfb; - h_Qmf->t_sin = qmf_phaseshift_sin64_cldfb; - h_Qmf->p_filter = qmf_cldfb_640; - h_Qmf->FilterSize = 640; - break; - case 32: - h_Qmf->t_cos = qmf_phaseshift_cos32_cldfb; - h_Qmf->t_sin = qmf_phaseshift_sin32_cldfb; - h_Qmf->p_filter = qmf_cldfb_320; - h_Qmf->FilterSize = 320; - break; - default: - return -1; - } - } - - if ( !(flags & QMF_FLAG_MPSLDFB) && ((flags & QMF_FLAG_CLDFB) == 0) ) - { - switch (no_channels) { - case 64: - h_Qmf->p_filter = qmf_64; - h_Qmf->t_cos = qmf_phaseshift_cos64; - h_Qmf->t_sin = qmf_phaseshift_sin64; - h_Qmf->p_stride = 1; - h_Qmf->FilterSize = 640; - h_Qmf->filterScale = 0; - break; - case 32: - h_Qmf->p_filter = qmf_64; - if (flags & QMF_FLAG_DOWNSAMPLED) { - h_Qmf->t_cos = qmf_phaseshift_cos_downsamp32; - h_Qmf->t_sin = qmf_phaseshift_sin_downsamp32; - } - else { - h_Qmf->t_cos = qmf_phaseshift_cos32; - h_Qmf->t_sin = qmf_phaseshift_sin32; - } - h_Qmf->p_stride = 2; - h_Qmf->FilterSize = 640; - h_Qmf->filterScale = 0; - break; - default: - return -1; - } - } - - h_Qmf->flags = flags; - - h_Qmf->no_channels = no_channels; - h_Qmf->no_col = noCols; - - h_Qmf->lsb = lsb; - h_Qmf->usb = fMin(usb, h_Qmf->no_channels); - - h_Qmf->FilterStates = (void*)pFilterStates; - - h_Qmf->outScalefactor = ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK + ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK + h_Qmf->filterScale; - - if ( (h_Qmf->p_stride == 2) - || ((flags & QMF_FLAG_CLDFB) && (no_channels == 32)) ) { - h_Qmf->outScalefactor -= 1; - } - h_Qmf->outGain = (FIXP_DBL)0x80000000; /* default init value will be not applied */ - - return (0); -} - -/*! - * - * \brief Adjust synthesis qmf filter states - * - * \return void - * - */ -static inline void -qmfAdaptFilterStates (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Filter Bank */ - int scaleFactorDiff) /*!< Scale factor difference to be applied */ -{ - if (synQmf == NULL || synQmf->FilterStates == NULL) { - return; - } - scaleValues((FIXP_QSS*)synQmf->FilterStates, synQmf->no_channels*(QMF_NO_POLY*2 - 1), scaleFactorDiff); -} - -/*! - * - * \brief Create QMF filter bank instance - * - * Only the lower bands are obtained (upto anaQmf->lsb). For - * a full bandwidth analysis it is required to set both anaQmf->lsb - * and anaQmf->usb to the amount of QMF bands. - * - * \return 0 if succesful - * - */ -int -qmfInitAnalysisFilterBank (HANDLE_QMF_FILTER_BANK h_Qmf, /*!< Returns handle */ - FIXP_QAS *pFilterStates, /*!< Handle to filter states */ - int noCols, /*!< Number of timeslots per frame */ - int lsb, /*!< lower end of QMF */ - int usb, /*!< upper end of QMF */ - int no_channels, /*!< Number of channels (bands) */ - int flags) /*!< Low Power flag */ -{ - int err = qmfInitFilterBank(h_Qmf, pFilterStates, noCols, lsb, usb, no_channels, flags); - if ( !(flags & QMF_FLAG_KEEP_STATES) && (h_Qmf->FilterStates != NULL) ) { - FDKmemclear(h_Qmf->FilterStates, (2*QMF_NO_POLY-1)*h_Qmf->no_channels*sizeof(FIXP_QAS)); - } - - return err; -} - -/*! - * - * \brief Create QMF filter bank instance - * - * Only the lower bands are obtained (upto anaQmf->lsb). For - * a full bandwidth analysis it is required to set both anaQmf->lsb - * and anaQmf->usb to the amount of QMF bands. - * - * \return 0 if succesful - * - */ -int -qmfInitSynthesisFilterBank (HANDLE_QMF_FILTER_BANK h_Qmf, /*!< Returns handle */ - FIXP_QSS *pFilterStates, /*!< Handle to filter states */ - int noCols, /*!< Number of timeslots per frame */ - int lsb, /*!< lower end of QMF */ - int usb, /*!< upper end of QMF */ - int no_channels, /*!< Number of channels (bands) */ - int flags) /*!< Low Power flag */ -{ - int oldOutScale = h_Qmf->outScalefactor; - int err = qmfInitFilterBank(h_Qmf, pFilterStates, noCols, lsb, usb, no_channels, flags); - if ( h_Qmf->FilterStates != NULL ) { - if ( !(flags & QMF_FLAG_KEEP_STATES) ) { - FDKmemclear(h_Qmf->FilterStates, (2*QMF_NO_POLY-1)*h_Qmf->no_channels*sizeof(FIXP_QSS)); - } else { - qmfAdaptFilterStates(h_Qmf, oldOutScale-h_Qmf->outScalefactor); - } - } - return err; -} - - - - -/*! - * - * \brief Change scale factor for output data and adjust qmf filter states - * - * \return void - * - */ -void -qmfChangeOutScalefactor (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ - int outScalefactor /*!< New scaling factor for output data */ - ) -{ - if (synQmf == NULL || synQmf->FilterStates == NULL) { - return; - } - - /* Add internal filterbank scale */ - outScalefactor += ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK + ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK + synQmf->filterScale; - - if ( (synQmf->p_stride == 2) - || ((synQmf->flags & QMF_FLAG_CLDFB) && (synQmf->no_channels == 32)) ) { - outScalefactor -= 1; - } - - /* adjust filter states when scale factor has been changed */ - if (synQmf->outScalefactor != outScalefactor) - { - int diff; - - if (outScalefactor > (SAMPLE_BITS - 1)) { - outScalefactor = SAMPLE_BITS - 1; - } else if (outScalefactor < (1 - SAMPLE_BITS)) { - outScalefactor = 1 - SAMPLE_BITS; - } - - diff = synQmf->outScalefactor - outScalefactor; - - qmfAdaptFilterStates(synQmf, diff); - - /* save new scale factor */ - synQmf->outScalefactor = outScalefactor; - } -} - -/*! - * - * \brief Change gain for output data - * - * \return void - * - */ -void -qmfChangeOutGain (HANDLE_QMF_FILTER_BANK synQmf, /*!< Handle of Qmf Synthesis Bank */ - FIXP_DBL outputGain /*!< New gain for output data */ - ) -{ - synQmf->outGain = outputGain; -} - |