Upgrade to FDKv2

Bug: 71430241
Test: CTS DecoderTest and DecoderTestAacDrc

original-Change-Id: Iaa20f749b8a04d553b20247cfe1a8930ebbabe30

Apply clang-format also on header files.

original-Change-Id: I14de1ef16bbc79ec0283e745f98356a10efeb2e4

Fixes for MPEG-D DRC

original-Change-Id: If1de2d74bbbac84b3f67de3b88b83f6a23b8a15c

Catch unsupported tw_mdct at an early stage

original-Change-Id: Ied9dd00d754162a0e3ca1ae3e6b854315d818afe

Fixing PVC transition frames

original-Change-Id: Ib75725abe39252806c32d71176308f2c03547a4e

Move qmf bands sanity check

original-Change-Id: Iab540c3013c174d9490d2ae100a4576f51d8dbc4

Initialize scaling variable

original-Change-Id: I3c4087101b70e998c71c1689b122b0d7762e0f9e

Add 16 qmf band configuration to getSlotNrgHQ()

original-Change-Id: I49a5d30f703a1b126ff163df9656db2540df21f1

Always apply byte alignment at the end of the AudioMuxElement

original-Change-Id: I42d560287506d65d4c3de8bfe3eb9a4ebeb4efc7

Setup SBR element only if no parse error exists

original-Change-Id: I1915b73704bc80ab882b9173d6bec59cbd073676

Additional array index check in HCR

original-Change-Id: I18cc6e501ea683b5009f1bbee26de8ddd04d8267

Fix fade-in index selection in concealment module

original-Change-Id: Ibf802ed6ed8c05e9257e1f3b6d0ac1162e9b81c1

Enable explicit backward compatible parser for AAC_LD

original-Change-Id: I27e9c678dcb5d40ed760a6d1e06609563d02482d

Skip spatial specific config in explicit backward compatible ASC

original-Change-Id: Iff7cc365561319e886090cedf30533f562ea4d6e

Update flags description in decoder API

original-Change-Id: I9a5b4f8da76bb652f5580cbd3ba9760425c43830

Add QMF domain reset function

original-Change-Id: I4f89a8a2c0277d18103380134e4ed86996e9d8d6

DRC upgrade v2.1.0

original-Change-Id: I5731c0540139dab220094cd978ef42099fc45b74

Fix integer overflow in sqrtFixp_lookup()

original-Change-Id: I429a6f0d19aa2cc957e0f181066f0ca73968c914

Fix integer overflow in invSqrtNorm2()

original-Change-Id: I84de5cbf9fb3adeb611db203fe492fabf4eb6155

Fix integer overflow in GenerateRandomVector()

original-Change-Id: I3118a641008bd9484d479e5b0b1ee2b5d7d44d74

Fix integer overflow in adjustTimeSlot_EldGrid()

original-Change-Id: I29d503c247c5c8282349b79df940416a512fb9d5

Fix integer overflow in FDKsbrEnc_codeEnvelope()

original-Change-Id: I6b34b61ebb9d525b0c651ed08de2befc1f801449

Follow-up on: Fix integer overflow in adjustTimeSlot_EldGrid()

original-Change-Id: I6f8f578cc7089e5eb7c7b93e580b72ca35ad689a

Fix integer overflow in get_pk_v2()

original-Change-Id: I63375bed40d45867f6eeaa72b20b1f33e815938c

Fix integer overflow in Syn_filt_zero()

original-Change-Id: Ie0c02fdfbe03988f9d3b20d10cd9fe4c002d1279

Fix integer overflow in CFac_CalcFacSignal()

original-Change-Id: Id2d767c40066c591b51768e978eb8af3b803f0c5

Fix integer overflow in FDKaacEnc_FDKaacEnc_calcPeNoAH()

original-Change-Id: Idcbd0f4a51ae2550ed106aa6f3d678d1f9724841

Fix integer overflow in sbrDecoder_calculateGainVec()

original-Change-Id: I7081bcbe29c5cede9821b38d93de07c7add2d507

Fix integer overflow in CLpc_SynthesisLattice()

original-Change-Id: I4a95ddc18de150102352d4a1845f06094764c881

Fix integer overflow in Pred_Lt4()

original-Change-Id: I4dbd012b2de7d07c3e70a47b92e3bfae8dbc750a

Fix integer overflow in FDKsbrEnc_InitSbrFastTransientDetector()

original-Change-Id: I788cbec1a4a00f44c2f3a72ad7a4afa219807d04

Fix unsigned integer overflow in FDKaacEnc_WriteBitstream()

original-Change-Id: I68fc75166e7d2cd5cd45b18dbe3d8c2a92f1822a

Fix unsigned integer overflow in FDK_MetadataEnc_Init()

original-Change-Id: Ie8d025f9bcdb2442c704bd196e61065c03c10af4

Fix overflow in pseudo random number generators

original-Change-Id: I3e2551ee01356297ca14e3788436ede80bd5513c

Fix unsigned integer overflow in sbrDecoder_Parse()

original-Change-Id: I3f231b2f437e9c37db4d5b964164686710eee971

Fix unsigned integer overflow in longsub()

original-Change-Id: I73c2bc50415cac26f1f5a29e125bbe75f9180a6e

Fix unsigned integer overflow in CAacDecoder_DecodeFrame()

original-Change-Id: Ifce2db4b1454b46fa5f887e9d383f1cc43b291e4

Fix overflow at CLpdChannelStream_Read()

original-Change-Id: Idb9d822ce3a4272e4794b643644f5434e2d4bf3f

Fix unsigned integer overflow in Hcr_State_BODY_SIGN_ESC__ESC_WORD()

original-Change-Id: I1ccf77c0015684b85534c5eb97162740a870b71c

Fix unsigned integer overflow in UsacConfig_Parse()

original-Change-Id: Ie6d27f84b6ae7eef092ecbff4447941c77864d9f

Fix unsigned integer overflow in aacDecoder_drcParse()

original-Change-Id: I713f28e883eea3d70b6fa56a7b8f8c22bcf66ca0

Fix unsigned integer overflow in aacDecoder_drcReadCompression()

original-Change-Id: Ia34dfeb88c4705c558bce34314f584965cafcf7a

Fix unsigned integer overflow in CDataStreamElement_Read()

original-Change-Id: Iae896cc1d11f0a893d21be6aa90bd3e60a2c25f0

Fix unsigned integer overflow in transportDec_AdjustEndOfAccessUnit()

original-Change-Id: I64cf29a153ee784bb4a16fdc088baabebc0007dc

Fix unsigned integer overflow in transportDec_GetAuBitsRemaining()

original-Change-Id: I975b3420faa9c16a041874ba0db82e92035962e4

Fix unsigned integer overflow in extractExtendedData()

original-Change-Id: I2a59eb09e2053cfb58dfb75fcecfad6b85a80a8f

Fix signed integer overflow in CAacDecoder_ExtPayloadParse()

original-Change-Id: I4ad5ca4e3b83b5d964f1c2f8c5e7b17c477c7929

Fix unsigned integer overflow in CAacDecoder_DecodeFrame()

original-Change-Id: I29a39df77d45c52a0c9c5c83c1ba81f8d0f25090

Follow-up on: Fix integer overflow in CLpc_SynthesisLattice()

original-Change-Id: I8fb194ffc073a3432a380845be71036a272d388f

Fix signed integer overflow in _interpolateDrcGain()

original-Change-Id: I879ec9ab14005069a7c47faf80e8bc6e03d22e60

Fix unsigned integer overflow in FDKreadBits()

original-Change-Id: I1f47a6a8037ff70375aa8844947d5681bb4287ad

Fix unsigned integer overflow in FDKbyteAlign()

original-Change-Id: Id5f3a11a0c9e50fc6f76ed6c572dbd4e9f2af766

Fix unsigned integer overflow in FDK_get32()

original-Change-Id: I9d33b8e97e3d38cbb80629cb859266ca0acdce96

Fix unsigned integer overflow in FDK_pushBack()

original-Change-Id: Ic87f899bc8c6acf7a377a8ca7f3ba74c3a1e1c19

Fix unsigned integer overflow in FDK_pushForward()

original-Change-Id: I3b754382f6776a34be1602e66694ede8e0b8effc

Fix unsigned integer overflow in ReadPsData()

original-Change-Id: I25361664ba8139e32bbbef2ca8c106a606ce9c37

Fix signed integer overflow in E_UTIL_residu()

original-Change-Id: I8c3abd1f437ee869caa8fb5903ce7d3d641b6aad

REVERT: Follow-up on: Integer overflow in CLpc_SynthesisLattice().

original-Change-Id: I3d340099acb0414795c8dfbe6362bc0a8f045f9b

Follow-up on: Fix integer overflow in CLpc_SynthesisLattice()

original-Change-Id: I4aedb8b3a187064e9f4d985175aa55bb99cc7590

Follow-up on: Fix unsigned integer overflow in aacDecoder_drcParse()

original-Change-Id: I2aa2e13916213bf52a67e8b0518e7bf7e57fb37d

Fix integer overflow in acelp

original-Change-Id: Ie6390c136d84055f8b728aefbe4ebef6e029dc77

Fix unsigned integer overflow in aacDecoder_UpdateBitStreamCounters()

original-Change-Id: I391ffd97ddb0b2c184cba76139bfb356a3b4d2e2

Adjust concealment default settings

original-Change-Id: I6a95db935a327c47df348030bcceafcb29f54b21

Saturate estimatedStartPos

original-Change-Id: I27be2085e0ae83ec9501409f65e003f6bcba1ab6

Negative shift exponent in _interpolateDrcGain()

original-Change-Id: I18edb26b26d002aafd5e633d4914960f7a359c29

Negative shift exponent in calculateICC()

original-Change-Id: I3dcd2ae98d2eb70ee0d59750863cbb2a6f4f8aba

Too large shift exponent in FDK_put()

original-Change-Id: Ib7d9aaa434d2d8de4a13b720ca0464b31ca9b671

Too large shift exponent in CalcInvLdData()

original-Change-Id: I43e6e78d4cd12daeb1dcd5d82d1798bdc2550262

Member access within null pointer of type SBR_CHANNEL

original-Change-Id: Idc5e4ea8997810376d2f36bbdf628923b135b097

Member access within null pointer of type CpePersistentData

original-Change-Id: Ib6c91cb0d37882768e5baf63324e429589de0d9d

Member access within null pointer FDKaacEnc_psyMain()

original-Change-Id: I7729b7f4479970531d9dc823abff63ca52e01997

Member access within null pointer FDKaacEnc_GetPnsParam()

original-Change-Id: I9aa3b9f3456ae2e0f7483dbd5b3dde95fc62da39

Member access within null pointer FDKsbrEnc_EnvEncodeFrame()

original-Change-Id: I67936f90ea714e90b3e81bc0dd1472cc713eb23a

Add HCR sanity check

original-Change-Id: I6c1d9732ebcf6af12f50b7641400752f74be39f7

Fix memory issue for HBE edge case with 8:3 SBR

original-Change-Id: I11ea58a61e69fbe8bf75034b640baee3011e63e9

Additional SBR parametrization sanity check for ELD

original-Change-Id: Ie26026fbfe174c2c7b3691f6218b5ce63e322140

Add MPEG-D DRC channel layout check

original-Change-Id: Iea70a74f171b227cce636a9eac4ba662777a2f72

Additional out-of-bounds checks in MPEG-D DRC

original-Change-Id: Ife4a8c3452c6fde8a0a09e941154a39a769777d4

Change-Id: Ic63cb2f628720f54fe9b572b0cb528e2599c624e

1 files changed, 532 insertions, 0 deletions

diff --git a/libDRCdec/src/drcGainDec_process.cpp b/libDRCdec/src/drcGainDec_process.cpp
new file mode 100644
index 0000000..70c9533
--- /dev/null
+++ b/libDRCdec/src/drcGainDec_process.cpp
@@ -0,0 +1,532 @@
+/* -----------------------------------------------------------------------------
+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;
+}