/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android © Copyright 1995 - 2019 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

About polyphase filtering

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 */ /*! * * \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 = 0; 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 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 = 0; FIXP_DBL tmp; FIXP_DBL *RESTRICT tReal = pTimeOut; FIXP_DBL *RESTRICT tImag = pTimeOut + L; /* Move input to output vector with offset */ scaleValuesSaturate(&tReal[0], &qmfReal[0], synQmf->lsb, scaleFactorLowBand); scaleValuesSaturate(&tReal[0 + synQmf->lsb], &qmfReal[0 + synQmf->lsb], synQmf->usb - synQmf->lsb, 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 */ scaleValuesSaturate(pTimeOut + M, qmfReal, synQmf->lsb, scaleFactorLowBand); scaleValuesSaturate(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->usb; i++) { cplxMultDiv2(&tImag[i], &tReal[i], qmfImag[i], qmfReal[i], synQmf->t_cos[i], synQmf->t_sin[i]); } scaleValuesSaturate(&tReal[0], synQmf->lsb, scaleFactorLowBand + 1); scaleValuesSaturate(&tReal[0 + synQmf->lsb], synQmf->usb - synQmf->lsb, scaleFactorHighBand + 1); scaleValuesSaturate(&tImag[0], synQmf->lsb, scaleFactorLowBand + 1); scaleValuesSaturate(&tImag[0 + synQmf->lsb], synQmf->usb - synQmf->lsb, scaleFactorHighBand + 1); } if ((synQmf->flags & QMF_FLAG_CLDFB) == 0) { scaleValuesSaturate(&tReal[0], &qmfReal[0], synQmf->lsb, scaleFactorLowBand); scaleValuesSaturate(&tReal[0 + synQmf->lsb], &qmfReal[0 + synQmf->lsb], synQmf->usb - synQmf->lsb, scaleFactorHighBand); scaleValuesSaturate(&tImag[0], &qmfImag[0], synQmf->lsb, scaleFactorLowBand); scaleValuesSaturate(&tImag[0 + synQmf->lsb], &qmfImag[0 + synQmf->lsb], synQmf->usb - synQmf->lsb, 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 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; } #define INT_PCM_QMFOUT INT_PCM #define SAMPLE_BITS_QMFOUT SAMPLE_BITS #include "qmf_pcm.h" #if SAMPLE_BITS == 16 /* also create a 32 bit output version */ #undef INT_PCM_QMFOUT #undef SAMPLE_BITS_QMFOUT #undef QMF_PCM_H #undef FIXP_QAS #undef QAS_BITS #undef INT_PCM_QMFIN #define INT_PCM_QMFOUT LONG #define SAMPLE_BITS_QMFOUT 32 #define FIXP_QAS FIXP_DBL #define QAS_BITS 32 #define INT_PCM_QMFIN LONG #include "qmf_pcm.h" #endif