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+
+/* -----------------------------------------------------------------------------------------------------------
+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;
+}
+