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diff --git a/fdk-aac/libFDK/src/qmf.cpp b/fdk-aac/libFDK/src/qmf.cpp new file mode 100644 index 0000000..6fca043 --- /dev/null +++ b/fdk-aac/libFDK/src/qmf.cpp @@ -0,0 +1,1135 @@ +/* ----------------------------------------------------------------------------- +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 +----------------------------------------------------------------------------- */ + +/******************* Library for basic calculation routines ******************** + + 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 #qmf_pfilt640 (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 "FDK_trigFcts.h" +#include "fixpoint_math.h" +#include "dct.h" + +#define QSSCALE (0) +#define FX_DBL2FX_QSS(x) (x) +#define FX_QSS2FX_DBL(x) (x) + +/* moved to qmf_pcm.h: -> qmfSynPrototypeFirSlot */ +/* moved to qmf_pcm.h: -> qmfSynPrototypeFirSlot_NonSymmetric */ +/* moved to qmf_pcm.h: -> qmfSynthesisFilteringSlot */ + +#ifndef FUNCTION_qmfAnaPrototypeFirSlot +/*! + \brief Perform Analysis Prototype Filtering on a single slot of input data. +*/ +static void qmfAnaPrototypeFirSlot( + FIXP_DBL *analysisBuffer, + INT no_channels, /*!< Number channels of analysis filter */ + const FIXP_PFT *p_filter, INT p_stride, /*!< Stride of analysis filter */ + FIXP_QAS *RESTRICT pFilterStates) { + INT k; + + FIXP_DBL accu; + const FIXP_PFT *RESTRICT p_flt = p_filter; + FIXP_DBL *RESTRICT pData_0 = analysisBuffer + 2 * no_channels - 1; + FIXP_DBL *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 filters 127..64 0..63 */ + for (k = 0; k < no_channels; k++) { + 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++ = (accu << 1); + sta_1 += staStep2; + + p_flt += pfltStep; + 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-- = (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_DBL *analysisBuffer, + int no_channels, /*!< Number channels of analysis filter */ + const FIXP_PFT *p_filter, int p_stride, /*!< Stride 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] = (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_DBL *timeIn, /*!< Time Signal */ + FIXP_DBL *rSubband) /*!< Real Output */ +{ + int i; + int L = anaQmf->no_channels; + int M = L >> 1; + int scale; + FIXP_DBL accu; + + const FIXP_DBL *timeInTmp1 = (FIXP_DBL *)&timeIn[3 * M]; + const FIXP_DBL *timeInTmp2 = timeInTmp1; + FIXP_DBL *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_DBL *timeIn, /*!< Time Signal */ + FIXP_DBL *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 + * + * + */ +#if !defined(FUNCTION_qmfForwardModulationHQ) +static void qmfForwardModulationHQ( + HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ + const FIXP_DBL *RESTRICT timeIn, /*!< Time Signal */ + FIXP_DBL *RESTRICT rSubband, /*!< Real Output */ + FIXP_DBL *RESTRICT iSubband /*!< Imaginary Output */ +) { + int i; + int L = anaQmf->no_channels; + int L2 = L << 1; + int shift = 0; + + /* Time advance by one sample, which is equivalent to the complex + rotation at the end of the analysis. Works only for STD mode. */ + if ((L == 64) && !(anaQmf->flags & (QMF_FLAG_CLDFB | QMF_FLAG_MPSLDFB))) { + FIXP_DBL x, y; + + /*rSubband[0] = u[1] + u[0]*/ + /*iSubband[0] = u[1] - u[0]*/ + x = timeIn[1] >> 1; + y = timeIn[0]; + rSubband[0] = x + (y >> 1); + iSubband[0] = x - (y >> 1); + + /*rSubband[n] = u[n+1] - u[2M-n], n=1,...,M-1*/ + /*iSubband[n] = u[n+1] + u[2M-n], n=1,...,M-1*/ + for (i = 1; i < L; i++) { + x = timeIn[i + 1] >> 1; /*u[n+1] */ + y = timeIn[L2 - i]; /*u[2M-n] */ + rSubband[i] = x - (y >> 1); + iSubband[i] = x + (y >> 1); + } + } else { + for (i = 0; i < L; i += 2) { + FIXP_DBL x0, x1, y0, y1; + + x0 = timeIn[i + 0] >> 1; + x1 = timeIn[i + 1] >> 1; + y0 = timeIn[L2 - 1 - i]; + y1 = timeIn[L2 - 2 - i]; + + rSubband[i + 0] = x0 - (y0 >> 1); + rSubband[i + 1] = x1 - (y1 >> 1); + iSubband[i + 0] = x0 + (y0 >> 1); + iSubband[i + 1] = x1 + (y1 >> 1); + } + } + + dct_IV(rSubband, L, &shift); + dst_IV(iSubband, L, &shift); + + /* Do the complex rotation except for the case of 64 bands (in STD mode). */ + if ((L != 64) || (anaQmf->flags & (QMF_FLAG_CLDFB | QMF_FLAG_MPSLDFB))) { + if (anaQmf->flags & QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION) { + FIXP_DBL iBand; + for (i = 0; i < fMin(anaQmf->lsb, L); i += 2) { + iBand = rSubband[i]; + rSubband[i] = -iSubband[i]; + iSubband[i] = iBand; + + iBand = -rSubband[i + 1]; + rSubband[i + 1] = iSubband[i + 1]; + iSubband[i + 1] = iBand; + } + } else { + const FIXP_QTW *sbr_t_cos; + const FIXP_QTW *sbr_t_sin; + const int len = L; /* was len = fMin(anaQmf->lsb, L) but in case of USAC + the signal above lsb is actually needed in some + cases (HBE?) */ + sbr_t_cos = anaQmf->t_cos; + sbr_t_sin = anaQmf->t_sin; + + for (i = 0; i < len; i++) { + cplxMult(&iSubband[i], &rSubband[i], iSubband[i], rSubband[i], + sbr_t_cos[i], sbr_t_sin[i]); + } + } + } +} +#endif /* FUNCTION_qmfForwardModulationHQ */ + +/* + * \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 + * + * Note: anaQmf->lsb can be greater than anaQmf->no_channels in case + * of implicit resampling (USAC with reduced 3/4 core frame length). + */ +#if (SAMPLE_BITS != DFRACT_BITS) && (QAS_BITS == DFRACT_BITS) +void qmfAnalysisFilteringSlot( + HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ + FIXP_DBL *qmfReal, /*!< Low and High band, real */ + FIXP_DBL *qmfImag, /*!< Low and High band, imag */ + const LONG *RESTRICT timeIn, /*!< Pointer to input */ + const int stride, /*!< stride factor of input */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ +) { + int offset = anaQmf->no_channels * (QMF_NO_POLY * 2 - 1); + /* + Feed time signal into oldest anaQmf->no_channels states + */ + { + FIXP_DBL *FilterStatesAnaTmp = ((FIXP_DBL *)anaQmf->FilterStates) + offset; + + /* Feed and scale actual time in slot */ + for (int i = anaQmf->no_channels >> 1; i != 0; i--) { + /* Place INT_PCM value left aligned in scaledTimeIn */ + + *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn; + timeIn += stride; + *FilterStatesAnaTmp++ = (FIXP_QAS)*timeIn; + timeIn += stride; + } + } + + 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(anaQmf->FilterStates, + (FIXP_QAS *)anaQmf->FilterStates + anaQmf->no_channels, + offset * sizeof(FIXP_QAS)); +} +#endif + +void qmfAnalysisFilteringSlot( + HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Synthesis Bank */ + FIXP_DBL *qmfReal, /*!< Low and High band, real */ + FIXP_DBL *qmfImag, /*!< Low and High band, imag */ + const INT_PCM *RESTRICT timeIn, /*!< Pointer to input */ + const int stride, /*!< stride factor of input */ + FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ +) { + int offset = anaQmf->no_channels * (QMF_NO_POLY * 2 - 1); + /* + Feed time signal into oldest anaQmf->no_channels states + */ + { + FIXP_QAS *FilterStatesAnaTmp = ((FIXP_QAS *)anaQmf->FilterStates) + offset; + + /* Feed and scale actual time in slot */ + for (int 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(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. + * + * + * \sa PolyphaseFiltering + */ +#if (SAMPLE_BITS != DFRACT_BITS) && (QAS_BITS == DFRACT_BITS) +void qmfAnalysisFiltering( + HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ + FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */ + FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */ + QMF_SCALE_FACTOR *scaleFactor, const LONG *timeIn, /*!< Time signal */ + const int timeIn_e, const int stride, + FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ +) { + int i; + int no_channels = anaQmf->no_channels; + + scaleFactor->lb_scale = + -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - timeIn_e; + scaleFactor->lb_scale -= anaQmf->filterScale; + + for (i = 0; i < anaQmf->no_col; i++) { + FIXP_DBL *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 */ +} +#endif + +void qmfAnalysisFiltering( + HANDLE_QMF_FILTER_BANK anaQmf, /*!< Handle of Qmf Analysis Bank */ + FIXP_DBL **qmfReal, /*!< Pointer to real subband slots */ + FIXP_DBL **qmfImag, /*!< Pointer to imag subband slots */ + QMF_SCALE_FACTOR *scaleFactor, const INT_PCM *timeIn, /*!< Time signal */ + const int timeIn_e, const int stride, + FIXP_DBL *pWorkBuffer /*!< pointer to temporal working buffer */ +) { + int i; + int no_channels = anaQmf->no_channels; + + scaleFactor->lb_scale = + -ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK - timeIn_e; + scaleFactor->lb_scale -= anaQmf->filterScale; + + for (i = 0; i < anaQmf->no_col; i++) { + FIXP_DBL *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_DBL *qmfReal, /*!< Pointer to qmf real subband slot (input) */ + const int scaleFactorLowBand, /*!< Scalefactor for Low band */ + const int scaleFactorHighBand, /*!< Scalefactor for High band */ + FIXP_DBL *pTimeOut /*!< Pointer to qmf subband slot (output)*/ +) { + int i; + int L = synQmf->no_channels; + int M = L >> 1; + int scale; + FIXP_DBL tmp; + FIXP_DBL *RESTRICT tReal = pTimeOut; + FIXP_DBL *RESTRICT tImag = pTimeOut + L; + + /* Move input to output vector with offset */ + scaleValues(&tReal[0], &qmfReal[0], synQmf->lsb, (int)scaleFactorLowBand); + scaleValues(&tReal[0 + synQmf->lsb], &qmfReal[0 + synQmf->lsb], + synQmf->usb - synQmf->lsb, (int)scaleFactorHighBand); + FDKmemclear(&tReal[0 + synQmf->usb], (L - synQmf->usb) * sizeof(FIXP_DBL)); + + /* 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_DBL)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_DBL *qmfReal, /*!< Pointer to qmf real subband slot (input) */ + const int scaleFactorLowBand, /*!< Scalefactor for Low band */ + const int scaleFactorHighBand, /*!< Scalefactor for High band */ + FIXP_DBL *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_DBL)); + + 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]; + } +} + +#ifndef FUNCTION_qmfInverseModulationHQ +/*! + * + * \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_DBL *qmfReal, /*!< Pointer to qmf real subband slot */ + const FIXP_DBL *qmfImag, /*!< Pointer to qmf imag subband slot */ + const int scaleFactorLowBand, /*!< Scalefactor for Low band */ + const int scaleFactorHighBand, /*!< Scalefactor for High band */ + FIXP_DBL *pWorkBuffer /*!< WorkBuffer (output) */ +) { + int i; + int L = synQmf->no_channels; + int M = L >> 1; + int shift = 0; + FIXP_DBL *RESTRICT tReal = pWorkBuffer; + FIXP_DBL *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, (int)scaleFactorLowBand); + scaleValues(&tReal[0 + synQmf->lsb], &qmfReal[0 + synQmf->lsb], + synQmf->usb - synQmf->lsb, (int)scaleFactorHighBand); + scaleValues(&tImag[0], &qmfImag[0], synQmf->lsb, (int)scaleFactorLowBand); + scaleValues(&tImag[0 + synQmf->lsb], &qmfImag[0 + synQmf->lsb], + synQmf->usb - synQmf->lsb, (int)scaleFactorHighBand); + } + + FDKmemclear(&tReal[synQmf->usb], + (synQmf->no_channels - synQmf->usb) * sizeof(FIXP_DBL)); + FDKmemclear(&tImag[synQmf->usb], + (synQmf->no_channels - synQmf->usb) * sizeof(FIXP_DBL)); + + dct_IV(tReal, L, &shift); + dst_IV(tImag, L, &shift); + + if (synQmf->flags & QMF_FLAG_CLDFB) { + for (i = 0; i < M; i++) { + FIXP_DBL 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_DBL 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; + } + } +} +#endif /* #ifndef FUNCTION_qmfInverseModulationHQ */ + +/*! + * + * \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 */ + int synflag) /*!< 1: synthesis; 0: analysis */ +{ + FDKmemclear(h_Qmf, sizeof(QMF_FILTER_BANK)); + + if (flags & QMF_FLAG_MPSLDFB) { + flags |= QMF_FLAG_NONSYMMETRIC; + flags |= QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION; + + h_Qmf->t_cos = NULL; + h_Qmf->t_sin = NULL; + h_Qmf->filterScale = QMF_MPSLDFB_PFT_SCALE; + h_Qmf->p_stride = 1; + + switch (no_channels) { + case 64: + h_Qmf->p_filter = qmf_mpsldfb_640; + h_Qmf->FilterSize = 640; + break; + case 32: + h_Qmf->p_filter = qmf_mpsldfb_320; + h_Qmf->FilterSize = 320; + break; + default: + 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 = (synflag) ? qmf_phaseshift_cos32_cldfb_syn + : qmf_phaseshift_cos32_cldfb_ana; + h_Qmf->t_sin = qmf_phaseshift_sin32_cldfb; + h_Qmf->p_filter = qmf_cldfb_320; + h_Qmf->FilterSize = 320; + break; + case 16: + h_Qmf->t_cos = (synflag) ? qmf_phaseshift_cos16_cldfb_syn + : qmf_phaseshift_cos16_cldfb_ana; + h_Qmf->t_sin = qmf_phaseshift_sin16_cldfb; + h_Qmf->p_filter = qmf_cldfb_160; + h_Qmf->FilterSize = 160; + break; + case 8: + h_Qmf->t_cos = (synflag) ? qmf_phaseshift_cos8_cldfb_syn + : qmf_phaseshift_cos8_cldfb_ana; + h_Qmf->t_sin = qmf_phaseshift_sin8_cldfb; + h_Qmf->p_filter = qmf_cldfb_80; + h_Qmf->FilterSize = 80; + break; + default: + return -1; + } + } + + if (!(flags & QMF_FLAG_MPSLDFB) && ((flags & QMF_FLAG_CLDFB) == 0)) { + switch (no_channels) { + case 64: + h_Qmf->p_filter = qmf_pfilt640; + 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 40: + if (synflag) { + break; + } else { + h_Qmf->p_filter = qmf_pfilt400; /* Scaling factor 0.8 */ + h_Qmf->t_cos = qmf_phaseshift_cos40; + h_Qmf->t_sin = qmf_phaseshift_sin40; + h_Qmf->filterScale = 1; + h_Qmf->p_stride = 1; + h_Qmf->FilterSize = no_channels * 10; + } + break; + case 32: + h_Qmf->p_filter = qmf_pfilt640; + 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; + case 20: + h_Qmf->p_filter = qmf_pfilt200; + h_Qmf->p_stride = 1; + h_Qmf->FilterSize = 200; + h_Qmf->filterScale = 0; + break; + case 12: + h_Qmf->p_filter = qmf_pfilt120; + h_Qmf->p_stride = 1; + h_Qmf->FilterSize = 120; + h_Qmf->filterScale = 0; + break; + case 8: + h_Qmf->p_filter = qmf_pfilt640; + h_Qmf->p_stride = 8; + h_Qmf->FilterSize = 640; + h_Qmf->filterScale = 0; + break; + case 16: + h_Qmf->p_filter = qmf_pfilt640; + h_Qmf->t_cos = qmf_phaseshift_cos16; + h_Qmf->t_sin = qmf_phaseshift_sin16; + h_Qmf->p_stride = 4; + h_Qmf->FilterSize = 640; + h_Qmf->filterScale = 0; + break; + case 24: + h_Qmf->p_filter = qmf_pfilt240; + h_Qmf->t_cos = qmf_phaseshift_cos24; + h_Qmf->t_sin = qmf_phaseshift_sin24; + h_Qmf->p_stride = 1; + h_Qmf->FilterSize = 240; + h_Qmf->filterScale = 1; + break; + default: + return -1; + } + } + + h_Qmf->synScalefactor = h_Qmf->filterScale; + // DCT|DST dependency + switch (no_channels) { + case 128: + h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK + 1; + break; + case 40: { + h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK - 1; + } break; + case 64: + h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK; + break; + case 8: + h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK - 3; + break; + case 12: + h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK; + break; + case 20: + h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK + 1; + break; + case 32: + h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK - 1; + break; + case 16: + h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK - 2; + break; + case 24: + h_Qmf->synScalefactor += ALGORITHMIC_SCALING_IN_SYNTHESIS_FILTERBANK - 1; + break; + default: + return -1; + } + + h_Qmf->flags = flags; + + h_Qmf->no_channels = no_channels; + h_Qmf->no_col = noCols; + + h_Qmf->lsb = fMin(lsb, h_Qmf->no_channels); + h_Qmf->usb = synflag + ? fMin(usb, h_Qmf->no_channels) + : usb; /* was: h_Qmf->usb = fMin(usb, h_Qmf->no_channels); */ + + h_Qmf->FilterStates = (void *)pFilterStates; + + h_Qmf->outScalefactor = + (ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK + h_Qmf->filterScale) + + h_Qmf->synScalefactor; + + h_Qmf->outGain_m = + (FIXP_DBL)0x80000000; /* default init value will be not applied */ + h_Qmf->outGain_e = 0; + + 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; + } + if (scaleFactorDiff > 0) { + scaleValuesSaturate((FIXP_QSS *)synQmf->FilterStates, + synQmf->no_channels * (QMF_NO_POLY * 2 - 1), + scaleFactorDiff); + } else { + scaleValues((FIXP_QSS *)synQmf->FilterStates, + synQmf->no_channels * (QMF_NO_POLY * 2 - 1), scaleFactorDiff); + } +} + +/*! + * + * \brief Create QMF filter bank instance + * + * + * \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, 0); + 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)); + } + + FDK_ASSERT(h_Qmf->no_channels >= h_Qmf->lsb); + + return err; +} + +/*! + * + * \brief Create QMF filter bank instance + * + * + * \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, 1); + 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); + } + } + + FDK_ASSERT(h_Qmf->no_channels >= h_Qmf->lsb); + FDK_ASSERT(h_Qmf->no_channels >= h_Qmf->usb); + + 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) { + return; + } + + /* Add internal filterbank scale */ + outScalefactor += + (ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK + synQmf->filterScale) + + synQmf->synScalefactor; + + /* adjust filter states when scale factor has been changed */ + if (synQmf->outScalefactor != outScalefactor) { + int diff; + + diff = synQmf->outScalefactor - outScalefactor; + + qmfAdaptFilterStates(synQmf, diff); + + /* save new scale factor */ + synQmf->outScalefactor = outScalefactor; + } +} + +/*! + * + * \brief Get scale factor change which was set by qmfChangeOutScalefactor() + * + * \return scaleFactor + * + */ +int qmfGetOutScalefactor( + HANDLE_QMF_FILTER_BANK synQmf) /*!< Handle of Qmf Synthesis Bank */ +{ + int scaleFactor = synQmf->outScalefactor + ? (synQmf->outScalefactor - + (ALGORITHMIC_SCALING_IN_ANALYSIS_FILTERBANK + + synQmf->filterScale + synQmf->synScalefactor)) + : 0; + return scaleFactor; +} + +/*! + * + * \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 (mantissa) */ + int outputGainScale /*!< New gain for output data (exponent) */ +) { + synQmf->outGain_m = outputGain; + synQmf->outGain_e = outputGainScale; +} + +/* When QMF_16IN_32OUT is set, synthesis functions for 16 and 32 bit parallel + * output is compiled */ +#define INT_PCM_QMFOUT INT_PCM +#define SAMPLE_BITS_QMFOUT SAMPLE_BITS +#include "qmf_pcm.h" |