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authorMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
committerMatthias P. Braendli <matthias.braendli@mpb.li>2019-11-11 11:38:02 +0100
commit0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch)
treed07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libFDK/src/mdct.cpp
parentefe406d9724f959c8bc2a31802559ca6d41fd897 (diff)
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Include patched FDK-AAC in the repository
The initial idea was to get the DAB+ patch into upstream, but since that follows the android source releases, there is no place for a custom DAB+ patch there. So instead of having to maintain a patched fdk-aac that has to have the same .so version as the distribution package on which it is installed, we prefer having a separate fdk-aac-dab library to avoid collision. At that point, there's no reason to keep fdk-aac in a separate repository, as odr-audioenc is the only tool that needs DAB+ encoding support. Including it here simplifies installation, and makes it consistent with toolame-dab, also shipped in this repository. DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop, welle.io and others) can be done using upstream FDK-AAC.
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+/* -----------------------------------------------------------------------------
+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): Josef Hoepfl, Manuel Jander, Youliy Ninov, Daniel Hagel
+
+ Description: MDCT/MDST routines
+
+*******************************************************************************/
+
+#include "mdct.h"
+
+#include "FDK_tools_rom.h"
+#include "dct.h"
+#include "fixpoint_math.h"
+
+void mdct_init(H_MDCT hMdct, FIXP_DBL *overlap, INT overlapBufferSize) {
+ hMdct->overlap.freq = overlap;
+ // FDKmemclear(overlap, overlapBufferSize*sizeof(FIXP_DBL));
+ hMdct->prev_fr = 0;
+ hMdct->prev_nr = 0;
+ hMdct->prev_tl = 0;
+ hMdct->ov_size = overlapBufferSize;
+ hMdct->prevAliasSymmetry = 0;
+ hMdct->prevPrevAliasSymmetry = 0;
+ hMdct->pFacZir = NULL;
+ hMdct->pAsymOvlp = NULL;
+}
+
+/*
+This program implements the forward MDCT transform on an input block of data.
+The input block is in a form (A,B,C,D) where A,B,C and D are the respective
+1/4th segments of the block. The program takes the input block and folds it in
+the form:
+(-D-Cr,A-Br). This block is twice shorter and here the 'r' suffix denotes
+flipping of the sequence (reversing the order of the samples). While folding the
+input block in the above mentioned shorter block the program windows the data.
+Because the two operations (windowing and folding) are not implemented
+sequentially, but together the program's structure is not easy to understand.
+Once the output (already windowed) block (-D-Cr,A-Br) is ready it is passed to
+the DCT IV for processing.
+*/
+INT mdct_block(H_MDCT hMdct, const INT_PCM *RESTRICT timeData,
+ const INT noInSamples, FIXP_DBL *RESTRICT mdctData,
+ const INT nSpec, const INT tl, const FIXP_WTP *pRightWindowPart,
+ const INT fr, SHORT *pMdctData_e) {
+ int i, n;
+ /* tl: transform length
+ fl: left window slope length
+ nl: left window slope offset
+ fr: right window slope length
+ nr: right window slope offset
+ See FDK_tools/doc/intern/mdct.tex for more detail. */
+ int fl, nl, nr;
+ const FIXP_WTP *wls, *wrs;
+
+ wrs = pRightWindowPart;
+
+ /* Detect FRprevious / FL mismatches and override parameters accordingly */
+ if (hMdct->prev_fr ==
+ 0) { /* At start just initialize and pass parameters as they are */
+ hMdct->prev_fr = fr;
+ hMdct->prev_wrs = wrs;
+ hMdct->prev_tl = tl;
+ }
+
+ /* Derive NR */
+ nr = (tl - fr) >> 1;
+
+ /* Skip input samples if tl is smaller than block size */
+ timeData += (noInSamples - tl) >> 1;
+
+ /* windowing */
+ for (n = 0; n < nSpec; n++) {
+ /*
+ * MDCT scale:
+ * + 1: fMultDiv2() in windowing.
+ * + 1: Because of factor 1/2 in Princen-Bradley compliant windowed TDAC.
+ */
+ INT mdctData_e = 1 + 1;
+
+ /* Derive left parameters */
+ wls = hMdct->prev_wrs;
+ fl = hMdct->prev_fr;
+ nl = (tl - fl) >> 1;
+
+ /* Here we implement a simplified version of what happens after the this
+ piece of code (see the comments below). We implement the folding of A and B
+ segments to (A-Br) but A is zero, because in this part of the MDCT sequence
+ the window coefficients with which A must be multiplied are zero. */
+ for (i = 0; i < nl; i++) {
+#if SAMPLE_BITS == DFRACT_BITS /* SPC_BITS and DFRACT_BITS should be equal. */
+ mdctData[(tl / 2) + i] = -((FIXP_DBL)timeData[tl - i - 1] >> (1));
+#else
+ mdctData[(tl / 2) + i] = -(FIXP_DBL)timeData[tl - i - 1]
+ << (DFRACT_BITS - SAMPLE_BITS - 1); /* 0(A)-Br */
+#endif
+ }
+
+ /* Implements the folding and windowing of the left part of the sequence,
+ that is segments A and B. The A segment is multiplied by the respective left
+ window coefficient and placed in a temporary variable.
+
+ tmp0 = fMultDiv2((FIXP_PCM)timeData[i+nl], pLeftWindowPart[i].v.im);
+
+ After this the B segment taken in reverse order is multiplied by the left
+ window and subtracted from the previously derived temporary variable, so
+ that finally we implement the A-Br operation. This output is written to the
+ right part of the MDCT output : (-D-Cr,A-Br).
+
+ mdctData[(tl/2)+i+nl] = fMultSubDiv2(tmp0, (FIXP_PCM)timeData[tl-nl-i-1],
+ pLeftWindowPart[i].v.re);//A*window-Br*window
+
+ The (A-Br) data is written to the output buffer (mdctData) without being
+ flipped. */
+ for (i = 0; i < fl / 2; i++) {
+ FIXP_DBL tmp0;
+ tmp0 = fMultDiv2((FIXP_PCM)timeData[i + nl], wls[i].v.im); /* a*window */
+ mdctData[(tl / 2) + i + nl] =
+ fMultSubDiv2(tmp0, (FIXP_PCM)timeData[tl - nl - i - 1],
+ wls[i].v.re); /* A*window-Br*window */
+ }
+
+ /* Right window slope offset */
+ /* Here we implement a simplified version of what happens after the this
+ piece of code (see the comments below). We implement the folding of C and D
+ segments to (-D-Cr) but D is zero, because in this part of the MDCT sequence
+ the window coefficients with which D must be multiplied are zero. */
+ for (i = 0; i < nr; i++) {
+#if SAMPLE_BITS == \
+ DFRACT_BITS /* This should be SPC_BITS instead of DFRACT_BITS. */
+ mdctData[(tl / 2) - 1 - i] = -((FIXP_DBL)timeData[tl + i] >> (1));
+#else
+ mdctData[(tl / 2) - 1 - i] =
+ -(FIXP_DBL)timeData[tl + i]
+ << (DFRACT_BITS - SAMPLE_BITS - 1); /* -C flipped at placing */
+#endif
+ }
+
+ /* Implements the folding and windowing of the right part of the sequence,
+ that is, segments C and D. The C segment is multiplied by the respective
+ right window coefficient and placed in a temporary variable.
+
+ tmp1 = fMultDiv2((FIXP_PCM)timeData[tl+nr+i], pRightWindowPart[i].v.re);
+
+ After this the D segment taken in reverse order is multiplied by the right
+ window and added from the previously derived temporary variable, so that we
+ get (C+Dr) operation. This output is negated to get (-C-Dr) and written to
+ the left part of the MDCT output while being reversed (flipped) at the same
+ time, so that from (-C-Dr) we get (-D-Cr)=> (-D-Cr,A-Br).
+
+ mdctData[(tl/2)-nr-i-1] = -fMultAddDiv2(tmp1,
+ (FIXP_PCM)timeData[(tl*2)-nr-i-1], pRightWindowPart[i].v.im);*/
+ for (i = 0; i < fr / 2; i++) {
+ FIXP_DBL tmp1;
+ tmp1 = fMultDiv2((FIXP_PCM)timeData[tl + nr + i],
+ wrs[i].v.re); /* C*window */
+ mdctData[(tl / 2) - nr - i - 1] =
+ -fMultAddDiv2(tmp1, (FIXP_PCM)timeData[(tl * 2) - nr - i - 1],
+ wrs[i].v.im); /* -(C*window+Dr*window) and flip before
+ placing -> -Cr - D */
+ }
+
+ /* We pass the shortened folded data (-D-Cr,A-Br) to the MDCT function */
+ dct_IV(mdctData, tl, &mdctData_e);
+
+ pMdctData_e[n] = (SHORT)mdctData_e;
+
+ timeData += tl;
+ mdctData += tl;
+
+ hMdct->prev_wrs = wrs;
+ hMdct->prev_fr = fr;
+ hMdct->prev_tl = tl;
+ }
+
+ return nSpec * tl;
+}
+
+void imdct_gain(FIXP_DBL *pGain_m, int *pGain_e, int tl) {
+ FIXP_DBL gain_m = *pGain_m;
+ int gain_e = *pGain_e;
+ int log2_tl;
+
+ gain_e += -MDCT_OUTPUT_GAIN - MDCT_OUT_HEADROOM + 1;
+ if (tl == 0) {
+ /* Dont regard the 2/N factor from the IDCT. It is compensated for somewhere
+ * else. */
+ *pGain_e = gain_e;
+ return;
+ }
+
+ log2_tl = DFRACT_BITS - 1 - fNormz((FIXP_DBL)tl);
+ gain_e += -log2_tl;
+
+ FDK_ASSERT(log2_tl - 2 >= 0);
+ FDK_ASSERT(log2_tl - 2 < 8*sizeof(int));
+
+ /* Detect non-radix 2 transform length and add amplitude compensation factor
+ which cannot be included into the exponent above */
+ switch ((tl) >> (log2_tl - 2)) {
+ case 0x7: /* 10 ms, 1/tl = 1.0/(FDKpow(2.0, -log2_tl) *
+ 0.53333333333333333333) */
+ if (gain_m == (FIXP_DBL)0) {
+ gain_m = FL2FXCONST_DBL(0.53333333333333333333f);
+ } else {
+ gain_m = fMult(gain_m, FL2FXCONST_DBL(0.53333333333333333333f));
+ }
+ break;
+ case 0x6: /* 3/4 of radix 2, 1/tl = 1.0/(FDKpow(2.0, -log2_tl) * 2.0/3.0) */
+ if (gain_m == (FIXP_DBL)0) {
+ gain_m = FL2FXCONST_DBL(2.0 / 3.0f);
+ } else {
+ gain_m = fMult(gain_m, FL2FXCONST_DBL(2.0 / 3.0f));
+ }
+ break;
+ case 0x5: /* 0.8 of radix 2 (e.g. tl 160), 1/tl = 1.0/(FDKpow(2.0, -log2_tl)
+ * 0.8/1.5) */
+ if (gain_m == (FIXP_DBL)0) {
+ gain_m = FL2FXCONST_DBL(0.53333333333333333333f);
+ } else {
+ gain_m = fMult(gain_m, FL2FXCONST_DBL(0.53333333333333333333f));
+ }
+ break;
+ case 0x4:
+ /* radix 2, nothing to do. */
+ break;
+ default:
+ /* unsupported */
+ FDK_ASSERT(0);
+ break;
+ }
+
+ *pGain_m = gain_m;
+ *pGain_e = gain_e;
+}
+
+INT imdct_drain(H_MDCT hMdct, FIXP_DBL *output, INT nrSamplesRoom) {
+ int buffered_samples = 0;
+
+ if (nrSamplesRoom > 0) {
+ buffered_samples = hMdct->ov_offset;
+
+ FDK_ASSERT(buffered_samples <= nrSamplesRoom);
+
+ if (buffered_samples > 0) {
+ FDKmemcpy(output, hMdct->overlap.time,
+ buffered_samples * sizeof(FIXP_DBL));
+ hMdct->ov_offset = 0;
+ }
+ }
+ return buffered_samples;
+}
+
+INT imdct_copy_ov_and_nr(H_MDCT hMdct, FIXP_DBL *pTimeData, INT nrSamples) {
+ FIXP_DBL *pOvl;
+ int nt, nf, i;
+
+ nt = fMin(hMdct->ov_offset, nrSamples);
+ nrSamples -= nt;
+ nf = fMin(hMdct->prev_nr, nrSamples);
+ FDKmemcpy(pTimeData, hMdct->overlap.time, nt * sizeof(FIXP_DBL));
+ pTimeData += nt;
+
+ pOvl = hMdct->overlap.freq + hMdct->ov_size - 1;
+ if (hMdct->prevPrevAliasSymmetry == 0) {
+ for (i = 0; i < nf; i++) {
+ FIXP_DBL x = -(*pOvl--);
+ *pTimeData = IMDCT_SCALE_DBL(x);
+ pTimeData++;
+ }
+ } else {
+ for (i = 0; i < nf; i++) {
+ FIXP_DBL x = (*pOvl--);
+ *pTimeData = IMDCT_SCALE_DBL(x);
+ pTimeData++;
+ }
+ }
+
+ return (nt + nf);
+}
+
+void imdct_adapt_parameters(H_MDCT hMdct, int *pfl, int *pnl, int tl,
+ const FIXP_WTP *wls, int noOutSamples) {
+ int fl = *pfl, nl = *pnl;
+ int window_diff, use_current = 0, use_previous = 0;
+ if (hMdct->prev_tl == 0) {
+ hMdct->prev_wrs = wls;
+ hMdct->prev_fr = fl;
+ hMdct->prev_nr = (noOutSamples - fl) >> 1;
+ hMdct->prev_tl = noOutSamples;
+ hMdct->ov_offset = 0;
+ use_current = 1;
+ }
+
+ window_diff = (hMdct->prev_fr - fl) >> 1;
+
+ /* check if the previous window slope can be adjusted to match the current
+ * window slope */
+ if (hMdct->prev_nr + window_diff > 0) {
+ use_current = 1;
+ }
+ /* check if the current window slope can be adjusted to match the previous
+ * window slope */
+ if (nl - window_diff > 0) {
+ use_previous = 1;
+ }
+
+ /* if both is possible choose the larger of both window slope lengths */
+ if (use_current && use_previous) {
+ if (fl < hMdct->prev_fr) {
+ use_current = 0;
+ }
+ }
+ /*
+ * If the previous transform block is big enough, enlarge previous window
+ * overlap, if not, then shrink current window overlap.
+ */
+ if (use_current) {
+ hMdct->prev_nr += window_diff;
+ hMdct->prev_fr = fl;
+ hMdct->prev_wrs = wls;
+ } else {
+ nl -= window_diff;
+ fl = hMdct->prev_fr;
+ }
+
+ *pfl = fl;
+ *pnl = nl;
+}
+
+/*
+This program implements the inverse modulated lapped transform, a generalized
+version of the inverse MDCT transform. Setting none of the MLT_*_ALIAS_FLAG
+flags computes the IMDCT, setting all of them computes the IMDST. Other
+combinations of these flags compute type III transforms used by the RSVD60
+multichannel tool for transitions between MDCT/MDST. The following description
+relates to the IMDCT only.
+
+If we pass the data block (A,B,C,D,E,F) to the FORWARD MDCT it will produce two
+outputs. The first one will be over the (A,B,C,D) part =>(-D-Cr,A-Br) and the
+second one will be over the (C,D,E,F) part => (-F-Er,C-Dr), since there is a
+overlap between consequtive passes of the algorithm. This overlap is over the
+(C,D) segments. The two outputs will be given sequentially to the DCT IV
+algorithm. At the INVERSE MDCT side we get two consecutive outputs from the IDCT
+IV algorithm, namely the same blocks: (-D-Cr,A-Br) and (-F-Er,C-Dr). The first
+of them lands in the Overlap buffer and the second is in the working one, which,
+one algorithm pass later will substitute the one residing in the overlap
+register. The IMDCT algorithm has to produce the C and D segments from the two
+buffers. In order to do this we take the left part of the overlap
+buffer(-D-Cr,A-Br), namely (-D-Cr) and add it appropriately to the right part of
+the working buffer (-F-Er,C-Dr), namely (C-Dr), so that we get first the C
+segment and later the D segment. We do this in the following way: From the right
+part of the working buffer(C-Dr) we subtract the flipped left part of the
+overlap buffer(-D-Cr):
+
+Result = (C-Dr) - flipped(-D-Cr) = C -Dr + Dr + C = 2C
+We divide by two and get the C segment. What we did is adding the right part of
+the first frame to the left part of the second one. While applying these
+operation we multiply the respective segments with the appropriate window
+functions.
+
+In order to get the D segment we do the following:
+From the negated second part of the working buffer(C-Dr) we subtract the flipped
+first part of the overlap buffer (-D-Cr):
+
+Result= - (C -Dr) - flipped(-D-Cr)= -C +Dr +Dr +C = 2Dr.
+After dividing by two and flipping we get the D segment.What we did is adding
+the right part of the first frame to the left part of the second one. While
+applying these operation we multiply the respective segments with the
+appropriate window functions.
+
+Once we have obtained the C and D segments the overlap buffer is emptied and the
+current buffer is sent in it, so that the E and F segments are available for
+decoding in the next algorithm pass.*/
+INT imlt_block(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *spectrum,
+ const SHORT scalefactor[], const INT nSpec,
+ const INT noOutSamples, const INT tl, const FIXP_WTP *wls,
+ INT fl, const FIXP_WTP *wrs, const INT fr, FIXP_DBL gain,
+ int flags) {
+ FIXP_DBL *pOvl;
+ FIXP_DBL *pOut0 = output, *pOut1;
+ INT nl, nr;
+ int w, i, nrSamples = 0, specShiftScale, transform_gain_e = 0;
+ int currAliasSymmetry = (flags & MLT_FLAG_CURR_ALIAS_SYMMETRY);
+
+ /* Derive NR and NL */
+ nr = (tl - fr) >> 1;
+ nl = (tl - fl) >> 1;
+
+ /* Include 2/N IMDCT gain into gain factor and exponent. */
+ imdct_gain(&gain, &transform_gain_e, tl);
+
+ /* Detect FRprevious / FL mismatches and override parameters accordingly */
+ if (hMdct->prev_fr != fl) {
+ imdct_adapt_parameters(hMdct, &fl, &nl, tl, wls, noOutSamples);
+ }
+
+ pOvl = hMdct->overlap.freq + hMdct->ov_size - 1;
+
+ if (noOutSamples > nrSamples) {
+ /* Purge buffered output. */
+ for (i = 0; i < hMdct->ov_offset; i++) {
+ *pOut0 = hMdct->overlap.time[i];
+ pOut0++;
+ }
+ nrSamples = hMdct->ov_offset;
+ hMdct->ov_offset = 0;
+ }
+
+ for (w = 0; w < nSpec; w++) {
+ FIXP_DBL *pSpec, *pCurr;
+ const FIXP_WTP *pWindow;
+
+ /* Detect FRprevious / FL mismatches and override parameters accordingly */
+ if (hMdct->prev_fr != fl) {
+ imdct_adapt_parameters(hMdct, &fl, &nl, tl, wls, noOutSamples);
+ }
+
+ specShiftScale = transform_gain_e;
+
+ /* Setup window pointers */
+ pWindow = hMdct->prev_wrs;
+
+ /* Current spectrum */
+ pSpec = spectrum + w * tl;
+
+ /* DCT IV of current spectrum. */
+ if (currAliasSymmetry == 0) {
+ if (hMdct->prevAliasSymmetry == 0) {
+ dct_IV(pSpec, tl, &specShiftScale);
+ } else {
+ FIXP_DBL _tmp[1024 + ALIGNMENT_DEFAULT / sizeof(FIXP_DBL)];
+ FIXP_DBL *tmp = (FIXP_DBL *)ALIGN_PTR(_tmp);
+ C_ALLOC_ALIGNED_REGISTER(tmp, sizeof(_tmp));
+ dct_III(pSpec, tmp, tl, &specShiftScale);
+ C_ALLOC_ALIGNED_UNREGISTER(tmp);
+ }
+ } else {
+ if (hMdct->prevAliasSymmetry == 0) {
+ FIXP_DBL _tmp[1024 + ALIGNMENT_DEFAULT / sizeof(FIXP_DBL)];
+ FIXP_DBL *tmp = (FIXP_DBL *)ALIGN_PTR(_tmp);
+ C_ALLOC_ALIGNED_REGISTER(tmp, sizeof(_tmp));
+ dst_III(pSpec, tmp, tl, &specShiftScale);
+ C_ALLOC_ALIGNED_UNREGISTER(tmp);
+ } else {
+ dst_IV(pSpec, tl, &specShiftScale);
+ }
+ }
+
+ /* Optional scaling of time domain - no yet windowed - of current spectrum
+ */
+ /* and de-scale current spectrum signal (time domain, no yet windowed) */
+ if (gain != (FIXP_DBL)0) {
+ for (i = 0; i < tl; i++) {
+ pSpec[i] = fMult(pSpec[i], gain);
+ }
+ }
+
+ {
+ int loc_scale =
+ fixmin_I(scalefactor[w] + specShiftScale, (INT)DFRACT_BITS - 1);
+ DWORD_ALIGNED(pSpec);
+ scaleValuesSaturate(pSpec, tl, loc_scale);
+ }
+
+ if (noOutSamples <= nrSamples) {
+ /* Divert output first half to overlap buffer if we already got enough
+ * output samples. */
+ pOut0 = hMdct->overlap.time + hMdct->ov_offset;
+ hMdct->ov_offset += hMdct->prev_nr + fl / 2;
+ } else {
+ /* Account output samples */
+ nrSamples += hMdct->prev_nr + fl / 2;
+ }
+
+ /* NR output samples 0 .. NR. -overlap[TL/2..TL/2-NR] */
+ if ((hMdct->pFacZir != 0) && (hMdct->prev_nr == fl / 2)) {
+ /* In the case of ACELP -> TCX20 -> FD short add FAC ZIR on nr signal part
+ */
+ for (i = 0; i < hMdct->prev_nr; i++) {
+ FIXP_DBL x = -(*pOvl--);
+ *pOut0 = IMDCT_SCALE_DBL(x + hMdct->pFacZir[i]);
+ pOut0++;
+ }
+ hMdct->pFacZir = NULL;
+ } else {
+ /* Here we implement a simplified version of what happens after the this
+ piece of code (see the comments below). We implement the folding of C and
+ D segments from (-D-Cr) but D is zero, because in this part of the MDCT
+ sequence the window coefficients with which D must be multiplied are zero.
+ "pOut0" writes sequentially the C block from left to right. */
+ if (hMdct->prevPrevAliasSymmetry == 0) {
+ for (i = 0; i < hMdct->prev_nr; i++) {
+ FIXP_DBL x = -(*pOvl--);
+ *pOut0 = IMDCT_SCALE_DBL(x);
+ pOut0++;
+ }
+ } else {
+ for (i = 0; i < hMdct->prev_nr; i++) {
+ FIXP_DBL x = *pOvl--;
+ *pOut0 = IMDCT_SCALE_DBL(x);
+ pOut0++;
+ }
+ }
+ }
+
+ if (noOutSamples <= nrSamples) {
+ /* Divert output second half to overlap buffer if we already got enough
+ * output samples. */
+ pOut1 = hMdct->overlap.time + hMdct->ov_offset + fl / 2 - 1;
+ hMdct->ov_offset += fl / 2 + nl;
+ } else {
+ pOut1 = pOut0 + (fl - 1);
+ nrSamples += fl / 2 + nl;
+ }
+
+ /* output samples before window crossing point NR .. TL/2.
+ * -overlap[TL/2-NR..TL/2-NR-FL/2] + current[NR..TL/2] */
+ /* output samples after window crossing point TL/2 .. TL/2+FL/2.
+ * -overlap[0..FL/2] - current[TL/2..FL/2] */
+ pCurr = pSpec + tl - fl / 2;
+ DWORD_ALIGNED(pCurr);
+ C_ALLOC_ALIGNED_REGISTER(pWindow, fl);
+ DWORD_ALIGNED(pWindow);
+ C_ALLOC_ALIGNED_UNREGISTER(pWindow);
+
+ if (hMdct->prevPrevAliasSymmetry == 0) {
+ if (hMdct->prevAliasSymmetry == 0) {
+ if (!hMdct->pAsymOvlp) {
+ for (i = 0; i < fl / 2; i++) {
+ FIXP_DBL x0, x1;
+ cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow[i]);
+ *pOut0 = IMDCT_SCALE_DBL_LSH1(x0);
+ *pOut1 = IMDCT_SCALE_DBL_LSH1(-x1);
+ pOut0++;
+ pOut1--;
+ }
+ } else {
+ FIXP_DBL *pAsymOvl = hMdct->pAsymOvlp + fl / 2 - 1;
+ for (i = 0; i < fl / 2; i++) {
+ FIXP_DBL x0, x1;
+ x1 = -fMultDiv2(*pCurr, pWindow[i].v.re) +
+ fMultDiv2(*pAsymOvl, pWindow[i].v.im);
+ x0 = fMultDiv2(*pCurr, pWindow[i].v.im) -
+ fMultDiv2(*pOvl, pWindow[i].v.re);
+ pCurr++;
+ pOvl--;
+ pAsymOvl--;
+ *pOut0++ = IMDCT_SCALE_DBL_LSH1(x0);
+ *pOut1-- = IMDCT_SCALE_DBL_LSH1(x1);
+ }
+ hMdct->pAsymOvlp = NULL;
+ }
+ } else { /* prevAliasingSymmetry == 1 */
+ for (i = 0; i < fl / 2; i++) {
+ FIXP_DBL x0, x1;
+ cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow[i]);
+ *pOut0 = IMDCT_SCALE_DBL_LSH1(x0);
+ *pOut1 = IMDCT_SCALE_DBL_LSH1(x1);
+ pOut0++;
+ pOut1--;
+ }
+ }
+ } else { /* prevPrevAliasingSymmetry == 1 */
+ if (hMdct->prevAliasSymmetry == 0) {
+ for (i = 0; i < fl / 2; i++) {
+ FIXP_DBL x0, x1;
+ cplxMultDiv2(&x1, &x0, *pCurr++, *pOvl--, pWindow[i]);
+ *pOut0 = IMDCT_SCALE_DBL_LSH1(x0);
+ *pOut1 = IMDCT_SCALE_DBL_LSH1(-x1);
+ pOut0++;
+ pOut1--;
+ }
+ } else { /* prevAliasingSymmetry == 1 */
+ for (i = 0; i < fl / 2; i++) {
+ FIXP_DBL x0, x1;
+ cplxMultDiv2(&x1, &x0, *pCurr++, *pOvl--, pWindow[i]);
+ *pOut0 = IMDCT_SCALE_DBL_LSH1(x0);
+ *pOut1 = IMDCT_SCALE_DBL_LSH1(x1);
+ pOut0++;
+ pOut1--;
+ }
+ }
+ }
+
+ if (hMdct->pFacZir != 0) {
+ /* add FAC ZIR of previous ACELP -> mdct transition */
+ FIXP_DBL *pOut = pOut0 - fl / 2;
+ FDK_ASSERT(fl / 2 <= 128);
+ for (i = 0; i < fl / 2; i++) {
+ pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]);
+ }
+ hMdct->pFacZir = NULL;
+ }
+ pOut0 += (fl / 2) + nl;
+
+ /* NL output samples TL/2+FL/2..TL. - current[FL/2..0] */
+ pOut1 += (fl / 2) + 1;
+ pCurr = pSpec + tl - fl / 2 - 1;
+ /* Here we implement a simplified version of what happens above the this
+ piece of code (see the comments above). We implement the folding of C and D
+ segments from (C-Dr) but C is zero, because in this part of the MDCT
+ sequence the window coefficients with which C must be multiplied are zero.
+ "pOut1" writes sequentially the D block from left to right. */
+ if (hMdct->prevAliasSymmetry == 0) {
+ for (i = 0; i < nl; i++) {
+ FIXP_DBL x = -(*pCurr--);
+ *pOut1++ = IMDCT_SCALE_DBL(x);
+ }
+ } else {
+ for (i = 0; i < nl; i++) {
+ FIXP_DBL x = *pCurr--;
+ *pOut1++ = IMDCT_SCALE_DBL(x);
+ }
+ }
+
+ /* Set overlap source pointer for next window pOvl = pSpec + tl/2 - 1; */
+ pOvl = pSpec + tl / 2 - 1;
+
+ /* Previous window values. */
+ hMdct->prev_nr = nr;
+ hMdct->prev_fr = fr;
+ hMdct->prev_tl = tl;
+ hMdct->prev_wrs = wrs;
+
+ /* Previous aliasing symmetry */
+ hMdct->prevPrevAliasSymmetry = hMdct->prevAliasSymmetry;
+ hMdct->prevAliasSymmetry = currAliasSymmetry;
+ }
+
+ /* Save overlap */
+
+ pOvl = hMdct->overlap.freq + hMdct->ov_size - tl / 2;
+ FDKmemcpy(pOvl, &spectrum[(nSpec - 1) * tl], (tl / 2) * sizeof(FIXP_DBL));
+
+ return nrSamples;
+}