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diff --git a/libFDK/src/qmf.cpp b/libFDK/src/qmf.cpp new file mode 100644 index 0000000..e410f55 --- /dev/null +++ b/libFDK/src/qmf.cpp @@ -0,0 +1,1188 @@ + +/* ----------------------------------------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------------------------------------- */ + +/******************************** 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; + 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) { + 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) { + 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; +} + |