libDRCdec/src/drcGainDec_process.cpp - platform/external/aac - Git at Google


            
            /* -----------------------------------------------------------------------------

            
            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

            
            ----------------------------------------------------------------------------- */


            /************************* MPEG-D DRC decoder library **************************


               Author(s):


               Description:


            *******************************************************************************/


            #include "drcDec_types.h"

            
            #include "drcDec_gainDecoder.h"

            
            #include "drcGainDec_process.h"


            #define E_TGAINSTEP 12


            static DRC_ERROR _prepareLnbIndex(ACTIVE_DRC* pActiveDrc,

            
                                              const int channelOffset,

            
                                              const int drcChannelOffset,

            
                                              const int numChannelsProcessed,

            
                                              const int lnbPointer) {

            
              int g, c;

            
              DRC_INSTRUCTIONS_UNI_DRC* pInst = pActiveDrc->pInst;


              /* channelOffset: start index of physical channels

            
                 numChannelsProcessed: number of processed channels, physical channels and

            
                 DRC channels channelOffset + drcChannelOffset: start index of DRC channels,

            
                    i.e. the channel order referenced in pInst.sequenceIndex */


              /* sanity checks */

            
              if ((channelOffset + numChannelsProcessed) > 8) return DE_NOT_OK;


              if ((channelOffset + drcChannelOffset + numChannelsProcessed) > 8)

            
                return DE_NOT_OK;


              if ((channelOffset + drcChannelOffset) < 0) return DE_NOT_OK;


              /* prepare lnbIndexForChannel, a map of indices from each channel to its

            
               * corresponding linearNodeBuffer instance */

            
              for (c = channelOffset; c < channelOffset + numChannelsProcessed; c++) {

            
                if (pInst->drcSetId > 0) {

            
                  int drcChannel = c + drcChannelOffset;

            
                  /* fallback for configuration with more physical channels than DRC

            
                     channels: reuse DRC gain of first channel. This is necessary for HE-AAC

            
                     mono with stereo output */

            
                  if (drcChannel >= pInst->drcChannelCount) drcChannel = 0;

            
                  g = pActiveDrc->channelGroupForChannel[drcChannel];

            
                  if ((g >= 0) && !pActiveDrc->channelGroupIsParametricDrc[g]) {

            
                    pActiveDrc->lnbIndexForChannel[c][lnbPointer] =

            
                        pActiveDrc->activeDrcOffset + pActiveDrc->gainElementForGroup[g];

            
                  }

            
                }

            
              }


              return DE_OK;

            
            }


            static DRC_ERROR _interpolateDrcGain(

            
                const GAIN_INTERPOLATION_TYPE gainInterpolationType,

            
                const SHORT timePrev,  /* time0 */

            
                const SHORT tGainStep, /* time1 - time0 */

            
                const SHORT start, const SHORT stop, const SHORT stepsize,

            
                const FIXP_DBL gainLeft, const FIXP_DBL gainRight, const FIXP_DBL slopeLeft,

            
                const FIXP_DBL slopeRight, FIXP_DBL* buffer) {

            
              int n, n_buf;

            
              int start_modulo, start_offset;


              if (tGainStep < 0) {

            
                return DE_NOT_OK;

            
              }

            
              if (tGainStep == 0) {

            
                return DE_OK;

            
              }


              /* get start index offset and buffer index for downsampled interpolation */

            
              /* start_modulo = (start+timePrev)%stepsize; */ /* stepsize is a power of 2 */

            
              start_modulo = (start + timePrev) & (stepsize - 1);

            
              start_offset = (start_modulo ? stepsize - start_modulo : 0);

            
              /* n_buf = (start + timePrev + start_offset)/stepsize; */

            
              n_buf = (start + timePrev + start_offset) >> (15 - fixnormz_S(stepsize));


              { /* gainInterpolationType == GIT_LINEAR */

            
                LONG a;

            
                /* runs = ceil((stop - start - start_offset)/stepsize). This works for

            
                 * stepsize = 2^N only. */

            
                INT runs = (INT)(stop - start - start_offset + stepsize - 1) >>

            
                           (30 - CountLeadingBits(stepsize));

            
                INT n_min = fMin(

            
                    fMin(CntLeadingZeros(gainRight), CntLeadingZeros(gainLeft)) - 1, 8);

            
                a = (LONG)((gainRight << n_min) - (gainLeft << n_min)) / tGainStep;

            
                LONG a_step = a * stepsize;

            
                n = start + start_offset;

            
                a = a * n + (LONG)(gainLeft << n_min);

            
                buffer += n_buf;

            
            #if defined(FUNCTION_interpolateDrcGain_func1)

            
                interpolateDrcGain_func1(buffer, a, a_step, n_min, runs);

            
            #else

            
                a -= a_step;

            
                n_min = 8 - n_min;

            
                for (int i = 0; i < runs; i++) {

            
                  a += a_step;

            
                  buffer[i] = fMultDiv2(buffer[i], (FIXP_DBL)a) << n_min;

            
                }

            
            #endif /* defined(FUNCTION_interpolateDrcGain_func1) */

            
              }

            
              return DE_OK;

            
            }


            static DRC_ERROR _processNodeSegments(

            
                const int frameSize, const GAIN_INTERPOLATION_TYPE gainInterpolationType,

            
                const int nNodes, const NODE_LIN* pNodeLin, const int offset,

            
                const SHORT stepsize,

            
                const NODE_LIN nodePrevious, /* the last node of the previous frame */

            
                const FIXP_DBL channelGain, FIXP_DBL* buffer) {

            
              DRC_ERROR err = DE_OK;

            
              SHORT timePrev, duration, start, stop, time;

            
              int n;

            
              FIXP_DBL gainLin = FL2FXCONST_DBL(1.0f / (float)(1 << 7)), gainLinPrev;

            
              FIXP_DBL slopeLin = (FIXP_DBL)0, slopeLinPrev = (FIXP_DBL)0;


              timePrev = nodePrevious.time + offset;

            
              gainLinPrev = nodePrevious.gainLin;

            
              for (n = 0; n < nNodes; n++) {

            
                time = pNodeLin[n].time + offset;

            
                duration = time - timePrev;

            
                gainLin = pNodeLin[n].gainLin;

            
                if (channelGain != FL2FXCONST_DBL(1.0f / (float)(1 << 8)))

            
                  gainLin =

            
                      SATURATE_LEFT_SHIFT(fMultDiv2(gainLin, channelGain), 9, DFRACT_BITS);


                if ((timePrev >= (frameSize - 1)) ||

            
                    (time < 0)) { /* This segment (between previous and current node) lies

            
                                     outside of this audio frame */

            
                  timePrev = time;

            
                  gainLinPrev = gainLin;

            
                  slopeLinPrev = slopeLin;

            
                  continue;

            
                }


                /* start and stop are the boundaries of the region of this segment that lie

            
                   within this audio frame. Their values are relative to the beginning of

            
                   this segment. stop is the first sample that isn't processed any more. */

            
                start = fMax(-timePrev, 1);

            
                stop = fMin(time, (SHORT)(frameSize - 1)) - timePrev + 1;


                err = _interpolateDrcGain(gainInterpolationType, timePrev, duration, start,

            
                                          stop, stepsize, gainLinPrev, gainLin,

            
                                          slopeLinPrev, slopeLin, buffer);

            
                if (err) return err;


                timePrev = time;

            
                gainLinPrev = gainLin;

            
              }

            
              return err;

            
            }


            /* process DRC on time-domain signal */

            
            DRC_ERROR

            
            processDrcTime(HANDLE_DRC_GAIN_DECODER hGainDec, const int activeDrcIndex,

            
                           const int delaySamples, const int channelOffset,

            
                           const int drcChannelOffset, const int numChannelsProcessed,

            
                           const int timeDataChannelOffset, FIXP_DBL* deinterleavedAudio) {

            
              DRC_ERROR err = DE_OK;

            
              int c, b, i;

            
              ACTIVE_DRC* pActiveDrc = &(hGainDec->activeDrc[activeDrcIndex]);

            
              DRC_GAIN_BUFFERS* pDrcGainBuffers = &(hGainDec->drcGainBuffers);

            
              int lnbPointer = pDrcGainBuffers->lnbPointer, lnbIx;

            
              LINEAR_NODE_BUFFER* pLinearNodeBuffer = pDrcGainBuffers->linearNodeBuffer;

            
              LINEAR_NODE_BUFFER* pDummyLnb = &(pDrcGainBuffers->dummyLnb);

            
              int offset = 0;


              if (hGainDec->delayMode == DM_REGULAR_DELAY) {

            
                offset = hGainDec->frameSize;

            
              }


              if ((delaySamples + offset) >

            
                  (NUM_LNB_FRAMES - 2) *

            
                      hGainDec->frameSize) /* if delaySamples is too big, NUM_LNB_FRAMES

            
                                              should be increased */

            
                return DE_NOT_OK;


              err = _prepareLnbIndex(pActiveDrc, channelOffset, drcChannelOffset,

            
                                     numChannelsProcessed, lnbPointer);

            
              if (err) return err;


              deinterleavedAudio +=

            
                  channelOffset * timeDataChannelOffset; /* apply channelOffset */


              /* signal processing loop */

            
              for (c = channelOffset; c < channelOffset + numChannelsProcessed; c++) {

            
                if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex)

            
                  pDrcGainBuffers->channelGain[c][lnbPointer] = hGainDec->channelGain[c];


                b = 0;

            
                {

            
                  LINEAR_NODE_BUFFER *pLnb, *pLnbPrevious;

            
                  NODE_LIN nodePrevious;

            
                  int lnbPointerDiff;

            
                  FIXP_DBL channelGain;

            
                  /* get pointer to oldest linearNodes */

            
                  lnbIx = lnbPointer + 1 - NUM_LNB_FRAMES;

            
                  while (lnbIx < 0) lnbIx += NUM_LNB_FRAMES;


                  if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex)

            
                    channelGain = pDrcGainBuffers->channelGain[c][lnbIx];

            
                  else

            
                    channelGain = FL2FXCONST_DBL(1.0f / (float)(1 << 8));


                  /* Loop over all node buffers in linearNodeBuffer.

            
                     All nodes which are not relevant for the current frame are sorted out

            
                     inside _processNodeSegments. */

            
                  for (i = 0; i < NUM_LNB_FRAMES - 1; i++) {

            
                    /* Prepare previous node */

            
                    if (pActiveDrc->lnbIndexForChannel[c][lnbIx] >= 0)

            
                      pLnbPrevious = &(

            
                          pLinearNodeBuffer[pActiveDrc->lnbIndexForChannel[c][lnbIx] + b]);

            
                    else

            
                      pLnbPrevious = pDummyLnb;

            
                    nodePrevious =

            
                        pLnbPrevious->linearNode[lnbIx][pLnbPrevious->nNodes[lnbIx] - 1];

            
                    nodePrevious.time -= hGainDec->frameSize;

            
                    if (channelGain != FL2FXCONST_DBL(1.0f / (float)(1 << 8)))

            
                      nodePrevious.gainLin = SATURATE_LEFT_SHIFT(

            
                          fMultDiv2(nodePrevious.gainLin,

            
                                    pDrcGainBuffers->channelGain[c][lnbIx]),

            
                          9, DFRACT_BITS);


                    /* Prepare current linearNodeBuffer instance */

            
                    lnbIx++;

            
                    if (lnbIx >= NUM_LNB_FRAMES) lnbIx = 0;


                    /* if lnbIndexForChannel changes over time, use the old indices for

            
                     * smooth transitions */

            
                    if (pActiveDrc->lnbIndexForChannel[c][lnbIx] >= 0)

            
                      pLnb = &(

            
                          pLinearNodeBuffer[pActiveDrc->lnbIndexForChannel[c][lnbIx] + b]);

            
                    else /* lnbIndexForChannel = -1 means "no DRC processing", due to

            
                            drcInstructionsIndex < 0, drcSetId < 0 or channel group < 0 */

            
                      pLnb = pDummyLnb;


                    if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex)

            
                      channelGain = pDrcGainBuffers->channelGain[c][lnbIx];


                    /* number of frames of offset with respect to lnbPointer */

            
                    lnbPointerDiff = i - (NUM_LNB_FRAMES - 2);


                    err = _processNodeSegments(

            
                        hGainDec->frameSize, pLnb->gainInterpolationType,

            
                        pLnb->nNodes[lnbIx], pLnb->linearNode[lnbIx],

            
                        lnbPointerDiff * hGainDec->frameSize + delaySamples + offset, 1,

            
                        nodePrevious, channelGain, deinterleavedAudio);

            
                    if (err) return err;

            
                  }

            
                  deinterleavedAudio += timeDataChannelOffset; /* proceed to next channel */

            
                }

            
              }

            
              return DE_OK;

            
            }


            /* process DRC on subband-domain signal */

            
            DRC_ERROR

            
            processDrcSubband(HANDLE_DRC_GAIN_DECODER hGainDec, const int activeDrcIndex,

            
                              const int delaySamples, const int channelOffset,

            
                              const int drcChannelOffset, const int numChannelsProcessed,

            
                              const int processSingleTimeslot,

            
                              FIXP_DBL* deinterleavedAudioReal[],

            
                              FIXP_DBL* deinterleavedAudioImag[]) {

            
              DRC_ERROR err = DE_OK;

            
              int b, c, g, m, m_start, m_stop, s, i;

            
              FIXP_DBL gainSb;

            
              DRC_INSTRUCTIONS_UNI_DRC* pInst = hGainDec->activeDrc[activeDrcIndex].pInst;

            
              DRC_GAIN_BUFFERS* pDrcGainBuffers = &(hGainDec->drcGainBuffers);

            
              ACTIVE_DRC* pActiveDrc = &(hGainDec->activeDrc[activeDrcIndex]);

            
              int activeDrcOffset = pActiveDrc->activeDrcOffset;

            
              int lnbPointer = pDrcGainBuffers->lnbPointer, lnbIx;

            
              LINEAR_NODE_BUFFER* pLinearNodeBuffer = pDrcGainBuffers->linearNodeBuffer;

            
              FIXP_DBL(*subbandGains)[4 * 1024 / 256] = hGainDec->subbandGains;

            
              FIXP_DBL* dummySubbandGains = hGainDec->dummySubbandGains;

            
              SUBBAND_DOMAIN_MODE subbandDomainMode = hGainDec->subbandDomainSupported;

            
              int signalIndex = 0;

            
              int frameSizeSb = 0;

            
              int nDecoderSubbands;

            
              SHORT L = 0; /* L: downsampling factor */

            
              int offset = 0;

            
              FIXP_DBL *audioReal = NULL, *audioImag = NULL;


              if (hGainDec->delayMode == DM_REGULAR_DELAY) {

            
                offset = hGainDec->frameSize;

            
              }


              if ((delaySamples + offset) >

            
                  (NUM_LNB_FRAMES - 2) *

            
                      hGainDec->frameSize) /* if delaySamples is too big, NUM_LNB_FRAMES

            
                                              should be increased */

            
                return DE_NOT_OK;


              switch (subbandDomainMode) {

            
            #if ((1024 / 256) >= (4096 / SUBBAND_DOWNSAMPLING_FACTOR_QMF64))

            
                case SDM_QMF64:

            
                  nDecoderSubbands = SUBBAND_NUM_BANDS_QMF64;

            
                  L = SUBBAND_DOWNSAMPLING_FACTOR_QMF64;

            
                  /* analysisDelay = SUBBAND_ANALYSIS_DELAY_QMF64; */

            
                  break;

            
                case SDM_QMF71:

            
                  nDecoderSubbands = SUBBAND_NUM_BANDS_QMF71;

            
                  L = SUBBAND_DOWNSAMPLING_FACTOR_QMF71;

            
                  /* analysisDelay = SUBBAND_ANALYSIS_DELAY_QMF71; */

            
                  break;

            
            #else

            
                case SDM_QMF64:

            
                case SDM_QMF71:

            
                  /* QMF domain processing is not supported. */

            
                  return DE_NOT_OK;

            
            #endif

            
                case SDM_STFT256:

            
                  nDecoderSubbands = SUBBAND_NUM_BANDS_STFT256;

            
                  L = SUBBAND_DOWNSAMPLING_FACTOR_STFT256;

            
                  /* analysisDelay = SUBBAND_ANALYSIS_DELAY_STFT256; */

            
                  break;

            
                default:

            
                  return DE_NOT_OK;

            
              }


              /* frameSizeSb = hGainDec->frameSize/L; */ /* L is a power of 2 */

            
              frameSizeSb =

            
                  hGainDec->frameSize >> (15 - fixnormz_S(L)); /* timeslots per frame */


              if ((processSingleTimeslot < 0) || (processSingleTimeslot >= frameSizeSb)) {

            
                m_start = 0;

            
                m_stop = frameSizeSb;

            
              } else {

            
                m_start = processSingleTimeslot;

            
                m_stop = m_start + 1;

            
              }


              err = _prepareLnbIndex(pActiveDrc, channelOffset, drcChannelOffset,

            
                                     numChannelsProcessed, lnbPointer);

            
              if (err) return err;


              if (!pActiveDrc->subbandGainsReady) /* only for the first time per frame that

            
                                                     processDrcSubband is called */

            
              {

            
                /* write subbandGains */

            
                for (g = 0; g < pInst->nDrcChannelGroups; g++) {

            
                  b = 0;

            
                  {

            
                    LINEAR_NODE_BUFFER* pLnb =

            
                        &(pLinearNodeBuffer[activeDrcOffset +

            
                                            pActiveDrc->gainElementForGroup[g] + b]);

            
                    NODE_LIN nodePrevious;

            
                    int lnbPointerDiff;


                    for (m = 0; m < frameSizeSb; m++) {

            
                      subbandGains[activeDrcOffset + g][b * frameSizeSb + m] =

            
                          FL2FXCONST_DBL(1.0f / (float)(1 << 7));

            
                    }


                    lnbIx = lnbPointer - (NUM_LNB_FRAMES - 1);

            
                    while (lnbIx < 0) lnbIx += NUM_LNB_FRAMES;


                    /* Loop over all node buffers in linearNodeBuffer.

            
                       All nodes which are not relevant for the current frame are sorted out

            
                       inside _processNodeSegments. */

            
                    for (i = 0; i < NUM_LNB_FRAMES - 1; i++) {

            
                      /* Prepare previous node */

            
                      nodePrevious = pLnb->linearNode[lnbIx][pLnb->nNodes[lnbIx] - 1];

            
                      nodePrevious.time -= hGainDec->frameSize;


                      lnbIx++;

            
                      if (lnbIx >= NUM_LNB_FRAMES) lnbIx = 0;


                      /* number of frames of offset with respect to lnbPointer */

            
                      lnbPointerDiff = i - (NUM_LNB_FRAMES - 2);


                      err = _processNodeSegments(

            
                          hGainDec->frameSize, pLnb->gainInterpolationType,

            
                          pLnb->nNodes[lnbIx], pLnb->linearNode[lnbIx],

            
                          lnbPointerDiff * hGainDec->frameSize + delaySamples + offset -

            
                              (L - 1) / 2,

            
                          L, nodePrevious, FL2FXCONST_DBL(1.0f / (float)(1 << 8)),

            
                          &(subbandGains[activeDrcOffset + g][b * frameSizeSb]));

            
                      if (err) return err;

            
                    }

            
                  }

            
                }

            
                pActiveDrc->subbandGainsReady = 1;

            
              }


              for (c = channelOffset; c < channelOffset + numChannelsProcessed; c++) {

            
                FIXP_DBL* thisSubbandGainsBuffer;

            
                if (pInst->drcSetId > 0)

            
                  g = pActiveDrc->channelGroupForChannel[c + drcChannelOffset];

            
                else

            
                  g = -1;


                audioReal = deinterleavedAudioReal[signalIndex];

            
                if (subbandDomainMode != SDM_STFT256) {

            
                  audioImag = deinterleavedAudioImag[signalIndex];

            
                }


                if ((g >= 0) && !pActiveDrc->channelGroupIsParametricDrc[g]) {

            
                  thisSubbandGainsBuffer = subbandGains[activeDrcOffset + g];

            
                } else {

            
                  thisSubbandGainsBuffer = dummySubbandGains;

            
                }


                for (m = m_start; m < m_stop; m++) {

            
                  INT n_min = 8;

            
                  { /* single-band DRC */

            
                    gainSb = thisSubbandGainsBuffer[m];

            
                    if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex)

            
                      gainSb = SATURATE_LEFT_SHIFT(

            
                          fMultDiv2(gainSb, hGainDec->channelGain[c]), 9, DFRACT_BITS);

            
                    /* normalize gainSb for keeping signal precision */

            
                    n_min = fMin(CntLeadingZeros(gainSb) - 1, n_min);

            
                    gainSb <<= n_min;

            
                    n_min = 8 - n_min;

            
                    if (subbandDomainMode ==

            
                        SDM_STFT256) { /* For STFT filterbank, real and imaginary parts are

            
                                          interleaved. */

            
                      for (s = 0; s < nDecoderSubbands; s++) {

            
                        *audioReal = fMultDiv2(*audioReal, gainSb) << n_min;

            
                        audioReal++;

            
                        *audioReal = fMultDiv2(*audioReal, gainSb) << n_min;

            
                        audioReal++;

            
                      }

            
                    } else {

            
                      for (s = 0; s < nDecoderSubbands; s++) {

            
                        *audioReal = fMultDiv2(*audioReal, gainSb) << n_min;

            
                        audioReal++;

            
                        *audioImag = fMultDiv2(*audioImag, gainSb) << n_min;

            
                        audioImag++;

            
                      }

            
                    }

            
                  }

            
                }

            
                signalIndex++;

            
              }

            
              return DE_OK;

            
            }
	/* -----------------------------------------------------------------------------
	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
	----------------------------------------------------------------------------- */

	/*********************** MPEG-D DRC decoder library ************************

	Author(s):

	Description:

	*******************************************************************************/

	#include "drcDec_types.h"
	#include "drcDec_gainDecoder.h"
	#include "drcGainDec_process.h"

	#define E_TGAINSTEP 12

	static DRC_ERROR _prepareLnbIndex(ACTIVE_DRC* pActiveDrc,
	const int channelOffset,
	const int drcChannelOffset,
	const int numChannelsProcessed,
	const int lnbPointer) {
	int g, c;
	DRC_INSTRUCTIONS_UNI_DRC* pInst = pActiveDrc->pInst;

	/* channelOffset: start index of physical channels
	numChannelsProcessed: number of processed channels, physical channels and
	DRC channels channelOffset + drcChannelOffset: start index of DRC channels,
	i.e. the channel order referenced in pInst.sequenceIndex */

	/* sanity checks */
	if ((channelOffset + numChannelsProcessed) > 8) return DE_NOT_OK;

	if ((channelOffset + drcChannelOffset + numChannelsProcessed) > 8)
	return DE_NOT_OK;

	if ((channelOffset + drcChannelOffset) < 0) return DE_NOT_OK;

	/* prepare lnbIndexForChannel, a map of indices from each channel to its
	* corresponding linearNodeBuffer instance */
	for (c = channelOffset; c < channelOffset + numChannelsProcessed; c++) {
	if (pInst->drcSetId > 0) {
	int drcChannel = c + drcChannelOffset;
	/* fallback for configuration with more physical channels than DRC
	channels: reuse DRC gain of first channel. This is necessary for HE-AAC
	mono with stereo output */
	if (drcChannel >= pInst->drcChannelCount) drcChannel = 0;
	g = pActiveDrc->channelGroupForChannel[drcChannel];
	if ((g >= 0) && !pActiveDrc->channelGroupIsParametricDrc[g]) {
	pActiveDrc->lnbIndexForChannel[c][lnbPointer] =
	pActiveDrc->activeDrcOffset + pActiveDrc->gainElementForGroup[g];
	}
	}
	}

	return DE_OK;
	}

	static DRC_ERROR _interpolateDrcGain(
	const GAIN_INTERPOLATION_TYPE gainInterpolationType,
	const SHORT timePrev, /* time0 */
	const SHORT tGainStep, /* time1 - time0 */
	const SHORT start, const SHORT stop, const SHORT stepsize,
	const FIXP_DBL gainLeft, const FIXP_DBL gainRight, const FIXP_DBL slopeLeft,
	const FIXP_DBL slopeRight, FIXP_DBL* buffer) {
	int n, n_buf;
	int start_modulo, start_offset;

	if (tGainStep < 0) {
	return DE_NOT_OK;
	}
	if (tGainStep == 0) {
	return DE_OK;
	}

	/* get start index offset and buffer index for downsampled interpolation */
	/* start_modulo = (start+timePrev)%stepsize; / / stepsize is a power of 2 */
	start_modulo = (start + timePrev) & (stepsize - 1);
	start_offset = (start_modulo ? stepsize - start_modulo : 0);
	/* n_buf = (start + timePrev + start_offset)/stepsize; */
	n_buf = (start + timePrev + start_offset) >> (15 - fixnormz_S(stepsize));

	{ /* gainInterpolationType == GIT_LINEAR */
	LONG a;
	/* runs = ceil((stop - start - start_offset)/stepsize). This works for
	* stepsize = 2^N only. */
	INT runs = (INT)(stop - start - start_offset + stepsize - 1) >>
	(30 - CountLeadingBits(stepsize));
	INT n_min = fMin(
	fMin(CntLeadingZeros(gainRight), CntLeadingZeros(gainLeft)) - 1, 8);
	a = (LONG)((gainRight << n_min) - (gainLeft << n_min)) / tGainStep;
	LONG a_step = a * stepsize;
	n = start + start_offset;
	a = a * n + (LONG)(gainLeft << n_min);
	buffer += n_buf;
	#if defined(FUNCTION_interpolateDrcGain_func1)
	interpolateDrcGain_func1(buffer, a, a_step, n_min, runs);
	#else
	a -= a_step;
	n_min = 8 - n_min;
	for (int i = 0; i < runs; i++) {
	a += a_step;
	buffer[i] = fMultDiv2(buffer[i], (FIXP_DBL)a) << n_min;
	}
	#endif /* defined(FUNCTION_interpolateDrcGain_func1) */
	}
	return DE_OK;
	}

	static DRC_ERROR _processNodeSegments(
	const int frameSize, const GAIN_INTERPOLATION_TYPE gainInterpolationType,
	const int nNodes, const NODE_LIN* pNodeLin, const int offset,
	const SHORT stepsize,
	const NODE_LIN nodePrevious, /* the last node of the previous frame */
	const FIXP_DBL channelGain, FIXP_DBL* buffer) {
	DRC_ERROR err = DE_OK;
	SHORT timePrev, duration, start, stop, time;
	int n;
	FIXP_DBL gainLin = FL2FXCONST_DBL(1.0f / (float)(1 << 7)), gainLinPrev;
	FIXP_DBL slopeLin = (FIXP_DBL)0, slopeLinPrev = (FIXP_DBL)0;

	timePrev = nodePrevious.time + offset;
	gainLinPrev = nodePrevious.gainLin;
	for (n = 0; n < nNodes; n++) {
	time = pNodeLin[n].time + offset;
	duration = time - timePrev;
	gainLin = pNodeLin[n].gainLin;
	if (channelGain != FL2FXCONST_DBL(1.0f / (float)(1 << 8)))
	gainLin =
	SATURATE_LEFT_SHIFT(fMultDiv2(gainLin, channelGain), 9, DFRACT_BITS);

	if ((timePrev >= (frameSize - 1)) \|\|
	(time < 0)) { /* This segment (between previous and current node) lies
	outside of this audio frame */
	timePrev = time;
	gainLinPrev = gainLin;
	slopeLinPrev = slopeLin;
	continue;
	}

	/* start and stop are the boundaries of the region of this segment that lie
	within this audio frame. Their values are relative to the beginning of
	this segment. stop is the first sample that isn't processed any more. */
	start = fMax(-timePrev, 1);
	stop = fMin(time, (SHORT)(frameSize - 1)) - timePrev + 1;

	err = _interpolateDrcGain(gainInterpolationType, timePrev, duration, start,
	stop, stepsize, gainLinPrev, gainLin,
	slopeLinPrev, slopeLin, buffer);
	if (err) return err;

	timePrev = time;
	gainLinPrev = gainLin;
	}
	return err;
	}

	/* process DRC on time-domain signal */
	DRC_ERROR
	processDrcTime(HANDLE_DRC_GAIN_DECODER hGainDec, const int activeDrcIndex,
	const int delaySamples, const int channelOffset,
	const int drcChannelOffset, const int numChannelsProcessed,
	const int timeDataChannelOffset, FIXP_DBL* deinterleavedAudio) {
	DRC_ERROR err = DE_OK;
	int c, b, i;
	ACTIVE_DRC* pActiveDrc = &(hGainDec->activeDrc[activeDrcIndex]);
	DRC_GAIN_BUFFERS* pDrcGainBuffers = &(hGainDec->drcGainBuffers);
	int lnbPointer = pDrcGainBuffers->lnbPointer, lnbIx;
	LINEAR_NODE_BUFFER* pLinearNodeBuffer = pDrcGainBuffers->linearNodeBuffer;
	LINEAR_NODE_BUFFER* pDummyLnb = &(pDrcGainBuffers->dummyLnb);
	int offset = 0;

	if (hGainDec->delayMode == DM_REGULAR_DELAY) {
	offset = hGainDec->frameSize;
	}

	if ((delaySamples + offset) >
	(NUM_LNB_FRAMES - 2) *
	hGainDec->frameSize) /* if delaySamples is too big, NUM_LNB_FRAMES
	should be increased */
	return DE_NOT_OK;

	err = _prepareLnbIndex(pActiveDrc, channelOffset, drcChannelOffset,
	numChannelsProcessed, lnbPointer);
	if (err) return err;

	deinterleavedAudio +=
	channelOffset * timeDataChannelOffset; /* apply channelOffset */

	/* signal processing loop */
	for (c = channelOffset; c < channelOffset + numChannelsProcessed; c++) {
	if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex)
	pDrcGainBuffers->channelGain[c][lnbPointer] = hGainDec->channelGain[c];

	b = 0;
	{
	LINEAR_NODE_BUFFER pLnb, pLnbPrevious;
	NODE_LIN nodePrevious;
	int lnbPointerDiff;
	FIXP_DBL channelGain;
	/* get pointer to oldest linearNodes */
	lnbIx = lnbPointer + 1 - NUM_LNB_FRAMES;
	while (lnbIx < 0) lnbIx += NUM_LNB_FRAMES;

	if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex)
	channelGain = pDrcGainBuffers->channelGain[c][lnbIx];
	else
	channelGain = FL2FXCONST_DBL(1.0f / (float)(1 << 8));

	/* Loop over all node buffers in linearNodeBuffer.
	All nodes which are not relevant for the current frame are sorted out
	inside _processNodeSegments. */
	for (i = 0; i < NUM_LNB_FRAMES - 1; i++) {
	/* Prepare previous node */
	if (pActiveDrc->lnbIndexForChannel[c][lnbIx] >= 0)
	pLnbPrevious = &(
	pLinearNodeBuffer[pActiveDrc->lnbIndexForChannel[c][lnbIx] + b]);
	else
	pLnbPrevious = pDummyLnb;
	nodePrevious =
	pLnbPrevious->linearNode[lnbIx][pLnbPrevious->nNodes[lnbIx] - 1];
	nodePrevious.time -= hGainDec->frameSize;
	if (channelGain != FL2FXCONST_DBL(1.0f / (float)(1 << 8)))
	nodePrevious.gainLin = SATURATE_LEFT_SHIFT(
	fMultDiv2(nodePrevious.gainLin,
	pDrcGainBuffers->channelGain[c][lnbIx]),
	9, DFRACT_BITS);

	/* Prepare current linearNodeBuffer instance */
	lnbIx++;
	if (lnbIx >= NUM_LNB_FRAMES) lnbIx = 0;

	/* if lnbIndexForChannel changes over time, use the old indices for
	* smooth transitions */
	if (pActiveDrc->lnbIndexForChannel[c][lnbIx] >= 0)
	pLnb = &(
	pLinearNodeBuffer[pActiveDrc->lnbIndexForChannel[c][lnbIx] + b]);
	else /* lnbIndexForChannel = -1 means "no DRC processing", due to
	drcInstructionsIndex < 0, drcSetId < 0 or channel group < 0 */
	pLnb = pDummyLnb;

	if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex)
	channelGain = pDrcGainBuffers->channelGain[c][lnbIx];

	/* number of frames of offset with respect to lnbPointer */
	lnbPointerDiff = i - (NUM_LNB_FRAMES - 2);

	err = _processNodeSegments(
	hGainDec->frameSize, pLnb->gainInterpolationType,
	pLnb->nNodes[lnbIx], pLnb->linearNode[lnbIx],
	lnbPointerDiff * hGainDec->frameSize + delaySamples + offset, 1,
	nodePrevious, channelGain, deinterleavedAudio);
	if (err) return err;
	}
	deinterleavedAudio += timeDataChannelOffset; /* proceed to next channel */
	}
	}
	return DE_OK;
	}

	/* process DRC on subband-domain signal */
	DRC_ERROR
	processDrcSubband(HANDLE_DRC_GAIN_DECODER hGainDec, const int activeDrcIndex,
	const int delaySamples, const int channelOffset,
	const int drcChannelOffset, const int numChannelsProcessed,
	const int processSingleTimeslot,
	FIXP_DBL* deinterleavedAudioReal[],
	FIXP_DBL* deinterleavedAudioImag[]) {
	DRC_ERROR err = DE_OK;
	int b, c, g, m, m_start, m_stop, s, i;
	FIXP_DBL gainSb;
	DRC_INSTRUCTIONS_UNI_DRC* pInst = hGainDec->activeDrc[activeDrcIndex].pInst;
	DRC_GAIN_BUFFERS* pDrcGainBuffers = &(hGainDec->drcGainBuffers);
	ACTIVE_DRC* pActiveDrc = &(hGainDec->activeDrc[activeDrcIndex]);
	int activeDrcOffset = pActiveDrc->activeDrcOffset;
	int lnbPointer = pDrcGainBuffers->lnbPointer, lnbIx;
	LINEAR_NODE_BUFFER* pLinearNodeBuffer = pDrcGainBuffers->linearNodeBuffer;
	FIXP_DBL(subbandGains)[4 1024 / 256] = hGainDec->subbandGains;
	FIXP_DBL* dummySubbandGains = hGainDec->dummySubbandGains;
	SUBBAND_DOMAIN_MODE subbandDomainMode = hGainDec->subbandDomainSupported;
	int signalIndex = 0;
	int frameSizeSb = 0;
	int nDecoderSubbands;
	SHORT L = 0; /* L: downsampling factor */
	int offset = 0;
	FIXP_DBL audioReal = NULL, audioImag = NULL;

	if (hGainDec->delayMode == DM_REGULAR_DELAY) {
	offset = hGainDec->frameSize;
	}

	if ((delaySamples + offset) >
	(NUM_LNB_FRAMES - 2) *
	hGainDec->frameSize) /* if delaySamples is too big, NUM_LNB_FRAMES
	should be increased */
	return DE_NOT_OK;

	switch (subbandDomainMode) {
	#if ((1024 / 256) >= (4096 / SUBBAND_DOWNSAMPLING_FACTOR_QMF64))
	case SDM_QMF64:
	nDecoderSubbands = SUBBAND_NUM_BANDS_QMF64;
	L = SUBBAND_DOWNSAMPLING_FACTOR_QMF64;
	/* analysisDelay = SUBBAND_ANALYSIS_DELAY_QMF64; */
	break;
	case SDM_QMF71:
	nDecoderSubbands = SUBBAND_NUM_BANDS_QMF71;
	L = SUBBAND_DOWNSAMPLING_FACTOR_QMF71;
	/* analysisDelay = SUBBAND_ANALYSIS_DELAY_QMF71; */
	break;
	#else
	case SDM_QMF64:
	case SDM_QMF71:
	/* QMF domain processing is not supported. */
	return DE_NOT_OK;
	#endif
	case SDM_STFT256:
	nDecoderSubbands = SUBBAND_NUM_BANDS_STFT256;
	L = SUBBAND_DOWNSAMPLING_FACTOR_STFT256;
	/* analysisDelay = SUBBAND_ANALYSIS_DELAY_STFT256; */
	break;
	default:
	return DE_NOT_OK;
	}

	/* frameSizeSb = hGainDec->frameSize/L; / / L is a power of 2 */
	frameSizeSb =
	hGainDec->frameSize >> (15 - fixnormz_S(L)); /* timeslots per frame */

	if ((processSingleTimeslot < 0) \|\| (processSingleTimeslot >= frameSizeSb)) {
	m_start = 0;
	m_stop = frameSizeSb;
	} else {
	m_start = processSingleTimeslot;
	m_stop = m_start + 1;
	}

	err = _prepareLnbIndex(pActiveDrc, channelOffset, drcChannelOffset,
	numChannelsProcessed, lnbPointer);
	if (err) return err;

	if (!pActiveDrc->subbandGainsReady) /* only for the first time per frame that
	processDrcSubband is called */
	{
	/* write subbandGains */
	for (g = 0; g < pInst->nDrcChannelGroups; g++) {
	b = 0;
	{
	LINEAR_NODE_BUFFER* pLnb =
	&(pLinearNodeBuffer[activeDrcOffset +
	pActiveDrc->gainElementForGroup[g] + b]);
	NODE_LIN nodePrevious;
	int lnbPointerDiff;

	for (m = 0; m < frameSizeSb; m++) {
	subbandGains[activeDrcOffset + g][b * frameSizeSb + m] =
	FL2FXCONST_DBL(1.0f / (float)(1 << 7));
	}

	lnbIx = lnbPointer - (NUM_LNB_FRAMES - 1);
	while (lnbIx < 0) lnbIx += NUM_LNB_FRAMES;

	/* Loop over all node buffers in linearNodeBuffer.
	All nodes which are not relevant for the current frame are sorted out
	inside _processNodeSegments. */
	for (i = 0; i < NUM_LNB_FRAMES - 1; i++) {
	/* Prepare previous node */
	nodePrevious = pLnb->linearNode[lnbIx][pLnb->nNodes[lnbIx] - 1];
	nodePrevious.time -= hGainDec->frameSize;

	lnbIx++;
	if (lnbIx >= NUM_LNB_FRAMES) lnbIx = 0;

	/* number of frames of offset with respect to lnbPointer */
	lnbPointerDiff = i - (NUM_LNB_FRAMES - 2);

	err = _processNodeSegments(
	hGainDec->frameSize, pLnb->gainInterpolationType,
	pLnb->nNodes[lnbIx], pLnb->linearNode[lnbIx],
	lnbPointerDiff * hGainDec->frameSize + delaySamples + offset -
	(L - 1) / 2,
	L, nodePrevious, FL2FXCONST_DBL(1.0f / (float)(1 << 8)),
	&(subbandGains[activeDrcOffset + g][b * frameSizeSb]));
	if (err) return err;
	}
	}
	}
	pActiveDrc->subbandGainsReady = 1;
	}

	for (c = channelOffset; c < channelOffset + numChannelsProcessed; c++) {
	FIXP_DBL* thisSubbandGainsBuffer;
	if (pInst->drcSetId > 0)
	g = pActiveDrc->channelGroupForChannel[c + drcChannelOffset];
	else
	g = -1;

	audioReal = deinterleavedAudioReal[signalIndex];
	if (subbandDomainMode != SDM_STFT256) {
	audioImag = deinterleavedAudioImag[signalIndex];
	}

	if ((g >= 0) && !pActiveDrc->channelGroupIsParametricDrc[g]) {
	thisSubbandGainsBuffer = subbandGains[activeDrcOffset + g];
	} else {
	thisSubbandGainsBuffer = dummySubbandGains;
	}

	for (m = m_start; m < m_stop; m++) {
	INT n_min = 8;
	{ /* single-band DRC */
	gainSb = thisSubbandGainsBuffer[m];
	if (activeDrcIndex == hGainDec->channelGainActiveDrcIndex)
	gainSb = SATURATE_LEFT_SHIFT(
	fMultDiv2(gainSb, hGainDec->channelGain[c]), 9, DFRACT_BITS);
	/* normalize gainSb for keeping signal precision */
	n_min = fMin(CntLeadingZeros(gainSb) - 1, n_min);
	gainSb <<= n_min;
	n_min = 8 - n_min;
	if (subbandDomainMode ==
	SDM_STFT256) { /* For STFT filterbank, real and imaginary parts are
	interleaved. */
	for (s = 0; s < nDecoderSubbands; s++) {
	audioReal = fMultDiv2(audioReal, gainSb) << n_min;
	audioReal++;
	audioReal = fMultDiv2(audioReal, gainSb) << n_min;
	audioReal++;
	}
	} else {
	for (s = 0; s < nDecoderSubbands; s++) {
	audioReal = fMultDiv2(audioReal, gainSb) << n_min;
	audioReal++;
	audioImag = fMultDiv2(audioImag, gainSb) << n_min;
	audioImag++;
	}
	}
	}
	}
	signalIndex++;
	}
	return DE_OK;
	}