webrtc/modules/audio_coding/main/source/acm_isac.cc

/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include "acm_codec_database.h"
#include "acm_common_defs.h"
#include "acm_isac.h"
#include "acm_neteq.h"
#include "trace.h"
#include "webrtc_neteq.h"
#include "webrtc_neteq_help_macros.h"

#ifdef WEBRTC_CODEC_ISAC
#include "acm_isac_macros.h"
#include "isac.h"
#endif

#ifdef WEBRTC_CODEC_ISACFX
#include "acm_isac_macros.h"
#include "isacfix.h"
#endif

namespace webrtc {

// we need this otherwise we cannot use forward declaration
// in the header file
#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
struct ACMISACInst {
  ACM_ISAC_STRUCT *inst;
};
#endif

#define ISAC_MIN_RATE 10000
#define ISAC_MAX_RATE 56000

// How the scaling is computed. iSAC computes a gain based on the
// bottleneck. It follows the following expression for that
//
// G(BN_kbps) = pow(10, (a + b * BN_kbps + c * BN_kbps * BN_kbps) / 20.0)
//              / 3.4641;
//
// Where for 30 ms framelength we have,
//
// a = -23; b = 0.48; c = 0;
//
// As the default encoder is operating at 32kbps we have the scale as
//
// S(BN_kbps) = G(BN_kbps) / G(32);

#define ISAC_NUM_SUPPORTED_RATES 9
const WebRtc_UWord16 isacSuportedRates[ISAC_NUM_SUPPORTED_RATES] = {
    32000,    30000,    26000,   23000,   21000,
    19000,    17000,   15000,    12000
};

const float isacScale[ISAC_NUM_SUPPORTED_RATES] = {
     1.0f,    0.8954f,  0.7178f, 0.6081f, 0.5445f,
     0.4875f, 0.4365f,  0.3908f, 0.3311f
};

// Tables for bandwidth estimates
#define NR_ISAC_BANDWIDTHS 24
const WebRtc_Word32 isacRatesWB[NR_ISAC_BANDWIDTHS] = {
    10000, 11100, 12300, 13700, 15200, 16900,
    18800, 20900, 23300, 25900, 28700, 31900,
    10100, 11200, 12400, 13800, 15300, 17000,
    18900, 21000, 23400, 26000, 28800, 32000
};

const WebRtc_Word32 isacRatesSWB[NR_ISAC_BANDWIDTHS] = {
    10000, 11000, 12400, 13800, 15300, 17000,
    18900, 21000, 23200, 25400, 27600, 29800,
    32000, 34100, 36300, 38500, 40700, 42900,
    45100, 47300, 49500, 51700, 53900, 56000,
};

#if (!defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX))

ACMISAC::ACMISAC(WebRtc_Word16 /* codecID */)
    : _codecInstPtr(NULL),
      _isEncInitialized(false),
      _isacCodingMode(CHANNEL_INDEPENDENT),
      _enforceFrameSize(false),
      _isacCurrentBN(32000),
      _samplesIn10MsAudio(160) {  // Initiates to 16 kHz mode.
  // Initiate decoder parameters for the 32 kHz mode.
  memset(&_decoderParams32kHz, 0, sizeof(WebRtcACMCodecParams));
  _decoderParams32kHz.codecInstant.pltype = -1;

  return;
}

ACMISAC::~ACMISAC() {
  return;
}

ACMGenericCodec* ACMISAC::CreateInstance(void) {
  return NULL;
}

WebRtc_Word16 ACMISAC::InternalEncode(WebRtc_UWord8* /* bitstream */,
                                      WebRtc_Word16* /* bitStreamLenByte */) {
  return -1;
}

WebRtc_Word16 ACMISAC::DecodeSafe(WebRtc_UWord8* /* bitStream */,
                                  WebRtc_Word16 /* bitStreamLenByte */,
                                  WebRtc_Word16* /* audio */,
                                  WebRtc_Word16* /* audioSamples */,
                                  WebRtc_Word8* /* speechType */) {
  return 0;
}

WebRtc_Word16 ACMISAC::InternalInitEncoder(
    WebRtcACMCodecParams* /* codecParams */) {
  return -1;
}

WebRtc_Word16 ACMISAC::InternalInitDecoder(
    WebRtcACMCodecParams* /* codecParams */) {
  return -1;
}

WebRtc_Word16 ACMISAC::InternalCreateDecoder() {
  return -1;
}

void ACMISAC::DestructDecoderSafe() {
  return;
}

WebRtc_Word16 ACMISAC::InternalCreateEncoder() {
  return -1;
}

void ACMISAC::DestructEncoderSafe() {
  return;
}

WebRtc_Word32 ACMISAC::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
                                const CodecInst& /* codecInst */) {
  return -1;
}

void ACMISAC::InternalDestructEncoderInst(void* /* ptrInst */) {
  return;
}

WebRtc_Word16 ACMISAC::DeliverCachedIsacData(
    WebRtc_UWord8* /* bitStream */, WebRtc_Word16* /* bitStreamLenByte */,
    WebRtc_UWord32* /* timestamp */, WebRtcACMEncodingType* /* encodingType */,
    const WebRtc_UWord16 /* isacRate */,
    const WebRtc_UWord8 /* isacBWestimate */) {
  return -1;
}

WebRtc_Word16 ACMISAC::Transcode(WebRtc_UWord8* /* bitStream */,
                                 WebRtc_Word16* /* bitStreamLenByte */,
                                 WebRtc_Word16 /* qBWE */,
                                 WebRtc_Word32 /* scale */,
                                 bool /* isRED */) {
  return -1;
}

WebRtc_Word16 ACMISAC::SetBitRateSafe(WebRtc_Word32 /* bitRate */) {
  return -1;
}

WebRtc_Word32 ACMISAC::GetEstimatedBandwidthSafe() {
  return -1;
}

WebRtc_Word32 ACMISAC::SetEstimatedBandwidthSafe(
    WebRtc_Word32 /* estimatedBandwidth */) {
  return -1;
}

WebRtc_Word32 ACMISAC::GetRedPayloadSafe(WebRtc_UWord8* /* redPayload */,
                                         WebRtc_Word16* /* payloadBytes */) {
  return -1;
}

WebRtc_Word16 ACMISAC::UpdateDecoderSampFreq(WebRtc_Word16 /* codecId */) {
  return -1;
}

WebRtc_Word16 ACMISAC::UpdateEncoderSampFreq(
    WebRtc_UWord16 /* encoderSampFreqHz */) {
  return -1;
}

WebRtc_Word16 ACMISAC::EncoderSampFreq(WebRtc_UWord16& /* sampFreqHz */) {
  return -1;
}

WebRtc_Word32 ACMISAC::ConfigISACBandwidthEstimator(
    const WebRtc_UWord8 /* initFrameSizeMsec */,
    const WebRtc_UWord16 /* initRateBitPerSec */,
    const bool /* enforceFrameSize  */) {
  return -1;
}

WebRtc_Word32 ACMISAC::SetISACMaxPayloadSize(
    const WebRtc_UWord16 /* maxPayloadLenBytes */) {
  return -1;
}

WebRtc_Word32 ACMISAC::SetISACMaxRate(
    const WebRtc_UWord32 /* maxRateBitPerSec */) {
  return -1;
}

void ACMISAC::UpdateFrameLen() {
  return;
}

void ACMISAC::CurrentRate(WebRtc_Word32& /*rateBitPerSec */) {
  return;
}

bool
ACMISAC::DecoderParamsSafe(
    WebRtcACMCodecParams* /* decParams */,
    const WebRtc_UWord8   /* payloadType */)
{
    return false;
}

void
ACMISAC::SaveDecoderParamSafe(
    const WebRtcACMCodecParams* /* codecParams */)
{
    return;
}

WebRtc_Word16 ACMISAC::REDPayloadISAC(const WebRtc_Word32 /* isacRate */,
                                      const WebRtc_Word16 /* isacBwEstimate */,
                                      WebRtc_UWord8* /* payload */,
                                      WebRtc_Word16* /* payloadLenBytes */) {
  return -1;
}

#else     //===================== Actual Implementation =======================

#ifdef WEBRTC_CODEC_ISACFX

enum IsacSamplingRate {
  kIsacWideband = 16,
  kIsacSuperWideband = 32
};

static float ACMISACFixTranscodingScale(WebRtc_UWord16 rate) {
  // find the scale for transcoding, the scale is rounded
  // downward
  float scale = -1;
  for (WebRtc_Word16 n = 0; n < ISAC_NUM_SUPPORTED_RATES; n++) {
    if (rate >= isacSuportedRates[n]) {
      scale = isacScale[n];
      break;
    }
  }
  return scale;
}

static void ACMISACFixGetSendBitrate(ACM_ISAC_STRUCT* inst,
                                     WebRtc_Word32* bottleNeck) {
  *bottleNeck = WebRtcIsacfix_GetUplinkBw(inst);
}

static WebRtc_Word16 ACMISACFixGetNewBitstream(ACM_ISAC_STRUCT* inst,
                                               WebRtc_Word16 BWEIndex,
                                               WebRtc_Word16 /* jitterIndex */,
                                               WebRtc_Word32 rate,
                                               WebRtc_Word16* bitStream,
                                               bool isRED) {
  if (isRED) {
    // RED not supported with iSACFIX
    return -1;
  }
  float scale = ACMISACFixTranscodingScale((WebRtc_UWord16) rate);
  return WebRtcIsacfix_GetNewBitStream(inst, BWEIndex, scale, bitStream);
}

static WebRtc_Word16 ACMISACFixGetSendBWE(ACM_ISAC_STRUCT* inst,
                                          WebRtc_Word16* rateIndex,
                                          WebRtc_Word16* /* dummy */) {
  WebRtc_Word16 localRateIndex;
  WebRtc_Word16 status = WebRtcIsacfix_GetDownLinkBwIndex(inst,
                                                          &localRateIndex);
  if (status < 0) {
    return -1;
  } else {
    *rateIndex = localRateIndex;
    return 0;
  }
}

static WebRtc_Word16 ACMISACFixControlBWE(ACM_ISAC_STRUCT* inst,
                                          WebRtc_Word32 rateBPS,
                                          WebRtc_Word16 frameSizeMs,
                                          WebRtc_Word16 enforceFrameSize) {
  return WebRtcIsacfix_ControlBwe(inst, (WebRtc_Word16) rateBPS, frameSizeMs,
                                  enforceFrameSize);
}

static WebRtc_Word16 ACMISACFixControl(ACM_ISAC_STRUCT* inst,
                                       WebRtc_Word32 rateBPS,
                                       WebRtc_Word16 frameSizeMs) {
  return WebRtcIsacfix_Control(inst, (WebRtc_Word16) rateBPS, frameSizeMs);
}

static IsacSamplingRate ACMISACFixGetEncSampRate(ACM_ISAC_STRUCT* /* inst */) {
  return kIsacWideband;
}

static IsacSamplingRate ACMISACFixGetDecSampRate(ACM_ISAC_STRUCT* /* inst */) {
  return kIsacWideband;
}

#endif

ACMISAC::ACMISAC(WebRtc_Word16 codecID)
    : _isEncInitialized(false),
      _isacCodingMode(CHANNEL_INDEPENDENT),
      _enforceFrameSize(false),
      _isacCurrentBN(32000),
      _samplesIn10MsAudio(160) {  // Initiates to 16 kHz mode.
  _codecID = codecID;

  // Create codec instance.
  _codecInstPtr = new ACMISACInst;
  if (_codecInstPtr == NULL) {
    return;
  }
  _codecInstPtr->inst = NULL;

  // Initiate decoder parameters for the 32 kHz mode.
  memset(&_decoderParams32kHz, 0, sizeof(WebRtcACMCodecParams));
  _decoderParams32kHz.codecInstant.pltype = -1;

  // TODO(tlegrand): Check if the following is really needed, now that
  // ACMGenericCodec has been updated to initialize this value.
  // Initialize values that can be used uninitialized otherwise
  _decoderParams.codecInstant.pltype = -1;
}

ACMISAC::~ACMISAC() {
  if (_codecInstPtr != NULL) {
    if (_codecInstPtr->inst != NULL) {
      ACM_ISAC_FREE(_codecInstPtr->inst);
      _codecInstPtr->inst = NULL;
    }
    delete _codecInstPtr;
    _codecInstPtr = NULL;
  }
  return;
}

ACMGenericCodec* ACMISAC::CreateInstance(void) {
  return NULL;
}

WebRtc_Word16 ACMISAC::InternalEncode(WebRtc_UWord8* bitstream,
                                      WebRtc_Word16* bitStreamLenByte) {
  // ISAC takes 10ms audio everytime we call encoder, therefor,
  // it should be treated like codecs with 'basic coding block'
  // non-zero, and the following 'while-loop' should not be necessary.
  // However, due to a mistake in the codec the frame-size might change
  // at the first 10ms pushed in to iSAC if the bit-rate is low, this is
  // sort of a bug in iSAC. to address this we treat iSAC as the
  // following.
  if (_codecInstPtr == NULL) {
    return -1;
  }
  *bitStreamLenByte = 0;
  while ((*bitStreamLenByte == 0) && (_inAudioIxRead < _frameLenSmpl)) {
    if (_inAudioIxRead > _inAudioIxWrite) {
      // something is wrong.
      WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
          "The actual fram-size of iSAC appears to be larger that expected. "
          "All audio pushed in but no bit-stream is generated.");
      return -1;
    }
    *bitStreamLenByte = ACM_ISAC_ENCODE(_codecInstPtr->inst,
                                        &_inAudio[_inAudioIxRead],
                                        (WebRtc_Word16*) bitstream);
    // increment the read index this tell the caller that how far
    // we have gone forward in reading the audio buffer
    _inAudioIxRead += _samplesIn10MsAudio;
  }
  if (*bitStreamLenByte == 0) {
    WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _uniqueID,
        "ISAC Has encoded the whole frame but no bit-stream is generated.");
  }

  // a packet is generated iSAC, is set in adaptive mode may change
  // the frame length and we like to update the bottleneck value as
  // well, although updating bottleneck is not crucial
  if ((*bitStreamLenByte > 0) && (_isacCodingMode == ADAPTIVE)) {
    //_frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
    ACM_ISAC_GETSENDBITRATE(_codecInstPtr->inst, &_isacCurrentBN);
  }
  UpdateFrameLen();
  return *bitStreamLenByte;
}

WebRtc_Word16 ACMISAC::DecodeSafe(WebRtc_UWord8* /* bitStream */,
                                  WebRtc_Word16 /* bitStreamLenByte */,
                                  WebRtc_Word16* /* audio */,
                                  WebRtc_Word16* /* audioSample */,
                                  WebRtc_Word8* /* speechType */) {
  return 0;
}

WebRtc_Word16 ACMISAC::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
  // if rate is set to -1 then iSAC has to be in adaptive mode
  if (codecParams->codecInstant.rate == -1) {
    _isacCodingMode = ADAPTIVE;
  }

  // sanity check that rate is in acceptable range
  else if ((codecParams->codecInstant.rate >= ISAC_MIN_RATE) &&
      (codecParams->codecInstant.rate <= ISAC_MAX_RATE)) {
    _isacCodingMode = CHANNEL_INDEPENDENT;
    _isacCurrentBN = codecParams->codecInstant.rate;
  } else {
    return -1;
  }

  // we need to set the encoder sampling frequency.
  if (UpdateEncoderSampFreq((WebRtc_UWord16) codecParams->codecInstant.plfreq)
      < 0) {
    return -1;
  }
  if (ACM_ISAC_ENCODERINIT(_codecInstPtr->inst, _isacCodingMode) < 0) {
    return -1;
  }

  // apply the frame-size and rate if operating in
  // channel-independent mode
  if (_isacCodingMode == CHANNEL_INDEPENDENT) {
    if (ACM_ISAC_CONTROL(_codecInstPtr->inst, codecParams->codecInstant.rate,
        codecParams->codecInstant.pacsize /
        (codecParams->codecInstant.plfreq / 1000)) < 0) {
      return -1;
    }
  } else {
    // We need this for adaptive case and has to be called
    // after initialization
    ACM_ISAC_GETSENDBITRATE(_codecInstPtr->inst, &_isacCurrentBN);
  }
  _frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
  return 0;
}

WebRtc_Word16 ACMISAC::InternalInitDecoder(WebRtcACMCodecParams* codecParams) {
  if (_codecInstPtr == NULL) {
    return -1;
  }

  // set decoder sampling frequency.
  if (codecParams->codecInstant.plfreq == 32000) {
    UpdateDecoderSampFreq(ACMCodecDB::kISACSWB);
  } else {
    UpdateDecoderSampFreq(ACMCodecDB::kISAC);
  }

  // in a one-way communication we may never register send-codec.
  // However we like that the BWE to work properly so it has to
  // be initialized. The BWE is initialized when iSAC encoder is initialized.
  // Therefore, we need this.
  if (!_encoderInitialized) {
    // Since we don't require a valid rate or a valid packet size when
    // initializing the decoder, we set valid values before initializing encoder
    codecParams->codecInstant.rate = kIsacWbDefaultRate;
    codecParams->codecInstant.pacsize = kIsacPacSize960;
    if (InternalInitEncoder(codecParams) < 0) {
      return -1;
    }
    _encoderInitialized = true;
  }

  return ACM_ISAC_DECODERINIT(_codecInstPtr->inst);
}

WebRtc_Word16 ACMISAC::InternalCreateDecoder() {
  if (_codecInstPtr == NULL) {
    return -1;
  }
  WebRtc_Word16 status = ACM_ISAC_CREATE(&(_codecInstPtr->inst));

  // specific to codecs with one instance for encoding and decoding
  _encoderInitialized = false;
  if (status < 0) {
    _encoderExist = false;
  } else {
    _encoderExist = true;
  }
  return status;
}

void ACMISAC::DestructDecoderSafe() {
  // codec with shared instance cannot delete.
  _decoderInitialized = false;
  return;
}

WebRtc_Word16 ACMISAC::InternalCreateEncoder() {
  if (_codecInstPtr == NULL) {
    return -1;
  }
  WebRtc_Word16 status = ACM_ISAC_CREATE(&(_codecInstPtr->inst));

  // specific to codecs with one instance for encoding and decoding
  _decoderInitialized = false;
  if (status < 0) {
    _decoderExist = false;
  } else {
    _decoderExist = true;
  }
  return status;
}

void ACMISAC::DestructEncoderSafe() {
  // codec with shared instance cannot delete.
  _encoderInitialized = false;
  return;
}

WebRtc_Word32 ACMISAC::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
                                const CodecInst& codecInst) {
  // Sanity checks
  if (_codecInstPtr == NULL) {
    return -1;
  }
  if (!_decoderInitialized || !_decoderExist) {
    return -1;
  }
  // Fill up the structure by calling
  // "SET_CODEC_PAR" & "SET_ISAC_FUNCTION."
  // Then call NetEQ to add the codec to it's
  // database.
  if (codecInst.plfreq == 16000) {
    SET_CODEC_PAR((codecDef), kDecoderISAC, codecInst.pltype,
                  _codecInstPtr->inst, 16000);
#ifdef WEBRTC_CODEC_ISAC
    SET_ISAC_FUNCTIONS((codecDef));
#else
    SET_ISACfix_FUNCTIONS((codecDef));
#endif
  } else {
#ifdef WEBRTC_CODEC_ISAC
    SET_CODEC_PAR((codecDef), kDecoderISACswb, codecInst.pltype,
        _codecInstPtr->inst, 32000);
    SET_ISACSWB_FUNCTIONS((codecDef));
#else
    return -1;
#endif
  }
  return 0;
}

void ACMISAC::InternalDestructEncoderInst(void* ptrInst) {
  if (ptrInst != NULL) {
    ACM_ISAC_FREE((ACM_ISAC_STRUCT *) ptrInst);
  }
  return;
}

WebRtc_Word16 ACMISAC::Transcode(WebRtc_UWord8* bitStream,
                                 WebRtc_Word16* bitStreamLenByte,
                                 WebRtc_Word16 qBWE, WebRtc_Word32 rate,
                                 bool isRED) {
  WebRtc_Word16 jitterInfo = 0;
  // transcode from a higher rate to lower rate sanity check
  if (_codecInstPtr == NULL) {
    return -1;
  }

  *bitStreamLenByte = ACM_ISAC_GETNEWBITSTREAM(_codecInstPtr->inst, qBWE,
                                               jitterInfo, rate,
                                               (WebRtc_Word16*) bitStream,
                                               (isRED) ? 1 : 0);

  if (*bitStreamLenByte < 0) {
    // error happened
    *bitStreamLenByte = 0;
    return -1;
  } else {
    return *bitStreamLenByte;
  }
}

WebRtc_Word16 ACMISAC::SetBitRateSafe(WebRtc_Word32 bitRate) {
  if (_codecInstPtr == NULL) {
    return -1;
  }
  WebRtc_UWord16 encoderSampFreq;
  EncoderSampFreq(encoderSampFreq);
  bool reinit = false;
  // change the BN of iSAC
  if (bitRate == -1) {
    // ADAPTIVE MODE
    // Check if it was already in adaptive mode
    if (_isacCodingMode != ADAPTIVE) {
      // was not in adaptive, then set the mode to adaptive
      // and flag for re-initialization
      _isacCodingMode = ADAPTIVE;
      reinit = true;
    }
  }
  // Sanity check if the rate valid
  else if ((bitRate >= ISAC_MIN_RATE) && (bitRate <= ISAC_MAX_RATE)) {
    //check if it was in channel-independent mode before
    if (_isacCodingMode != CHANNEL_INDEPENDENT) {
      // was not in channel independent, set the mode to
      // channel-independent and flag for re-initialization
      _isacCodingMode = CHANNEL_INDEPENDENT;
      reinit = true;
    }
    // store the bottleneck
    _isacCurrentBN = (WebRtc_UWord16) bitRate;
  } else {
    // invlaid rate
    return -1;
  }

  WebRtc_Word16 status = 0;
  if (reinit) {
    // initialize and check if it is successful
    if (ACM_ISAC_ENCODERINIT(_codecInstPtr->inst, _isacCodingMode) < 0) {
      // failed initialization
      return -1;
    }
  }
  if (_isacCodingMode == CHANNEL_INDEPENDENT) {

    status = ACM_ISAC_CONTROL(
        _codecInstPtr->inst, _isacCurrentBN,
        (encoderSampFreq == 32000) ? 30 : (_frameLenSmpl / 16));
    if (status < 0) {
      status = -1;
    }
  }

  // Update encoder parameters
  _encoderParams.codecInstant.rate = bitRate;

  UpdateFrameLen();
  return status;
}

WebRtc_Word32 ACMISAC::GetEstimatedBandwidthSafe() {
  WebRtc_Word16 bandwidthIndex = 0;
  WebRtc_Word16 delayIndex = 0;
  IsacSamplingRate sampRate;

  // Get bandwidth information
  ACM_ISAC_GETSENDBWE(_codecInstPtr->inst, &bandwidthIndex, &delayIndex);

  // Validy check of index
  if ((bandwidthIndex < 0) || (bandwidthIndex >= NR_ISAC_BANDWIDTHS)) {
    return -1;
  }

  // Check sample frequency
  sampRate = ACM_ISAC_GETDECSAMPRATE(_codecInstPtr->inst);
  if (sampRate == kIsacWideband) {
    return isacRatesWB[bandwidthIndex];
  } else {
    return isacRatesSWB[bandwidthIndex];
  }
}

WebRtc_Word32 ACMISAC::SetEstimatedBandwidthSafe(
    WebRtc_Word32 estimatedBandwidth) {
  IsacSamplingRate sampRate;
  WebRtc_Word16 bandwidthIndex;

  // Check sample frequency and choose appropriate table
  sampRate = ACM_ISAC_GETENCSAMPRATE(_codecInstPtr->inst);

  if (sampRate == kIsacWideband) {
    // Search through the WB rate table to find the index
    bandwidthIndex = NR_ISAC_BANDWIDTHS / 2 - 1;
    for (int i = 0; i < (NR_ISAC_BANDWIDTHS / 2); i++) {
      if (estimatedBandwidth == isacRatesWB[i]) {
        bandwidthIndex = i;
        break;
      } else if (estimatedBandwidth
          == isacRatesWB[i + NR_ISAC_BANDWIDTHS / 2]) {
        bandwidthIndex = i + NR_ISAC_BANDWIDTHS / 2;
        break;
      } else if (estimatedBandwidth < isacRatesWB[i]) {
        bandwidthIndex = i;
        break;
      }
    }
  } else {
    // Search through the SWB rate table to find the index
    bandwidthIndex = NR_ISAC_BANDWIDTHS - 1;
    for (int i = 0; i < NR_ISAC_BANDWIDTHS; i++) {
      if (estimatedBandwidth <= isacRatesSWB[i]) {
        bandwidthIndex = i;
        break;
      }
    }
  }

  // Set iSAC Bandwidth Estimate
  ACM_ISAC_SETBWE(_codecInstPtr->inst, bandwidthIndex);

  return 0;
}

WebRtc_Word32 ACMISAC::GetRedPayloadSafe(
#if (!defined(WEBRTC_CODEC_ISAC))
    WebRtc_UWord8* /* redPayload */, WebRtc_Word16* /* payloadBytes */) {
  return -1;
#else
    WebRtc_UWord8* redPayload, WebRtc_Word16* payloadBytes) {
  WebRtc_Word16 bytes = WebRtcIsac_GetRedPayload(_codecInstPtr->inst,
                                                 (WebRtc_Word16*)redPayload);
  if (bytes < 0) {
    return -1;
  }
  *payloadBytes = bytes;
  return 0;
#endif
}

WebRtc_Word16 ACMISAC::UpdateDecoderSampFreq(
#ifdef WEBRTC_CODEC_ISAC
    WebRtc_Word16 codecId) {
  if (ACMCodecDB::kISAC == codecId) {
    return WebRtcIsac_SetDecSampRate(_codecInstPtr->inst, kIsacWideband);
  } else if (ACMCodecDB::kISACSWB == codecId) {
    return WebRtcIsac_SetDecSampRate(_codecInstPtr->inst, kIsacSuperWideband);
  } else {
    return -1;
  }
#else
    WebRtc_Word16 /* codecId */) {
  return 0;
#endif
}

WebRtc_Word16 ACMISAC::UpdateEncoderSampFreq(
#ifdef WEBRTC_CODEC_ISAC
    WebRtc_UWord16 encoderSampFreqHz) {
  WebRtc_UWord16 currentSampRateHz;
  EncoderSampFreq(currentSampRateHz);

  if (currentSampRateHz != encoderSampFreqHz) {
    if ((encoderSampFreqHz != 16000) && (encoderSampFreqHz != 32000)) {
      return -1;
    } else {
      _inAudioIxRead = 0;
      _inAudioIxWrite = 0;
      _inTimestampIxWrite = 0;
      if (encoderSampFreqHz == 16000) {
        if (WebRtcIsac_SetEncSampRate(_codecInstPtr->inst, kIsacWideband) < 0) {
          return -1;
        }
        _samplesIn10MsAudio = 160;
      } else {
        if (WebRtcIsac_SetEncSampRate(_codecInstPtr->inst, kIsacSuperWideband)
            < 0) {
          return -1;
        }
        _samplesIn10MsAudio = 320;
      }
      _frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
      _encoderParams.codecInstant.pacsize = _frameLenSmpl;
      _encoderParams.codecInstant.plfreq = encoderSampFreqHz;
      return 0;
    }
  }
#else
    WebRtc_UWord16 /* codecId */) {
#endif
  return 0;
}

WebRtc_Word16 ACMISAC::EncoderSampFreq(WebRtc_UWord16& sampFreqHz) {
  IsacSamplingRate sampRate;
  sampRate = ACM_ISAC_GETENCSAMPRATE(_codecInstPtr->inst);
  if (sampRate == kIsacSuperWideband) {
    sampFreqHz = 32000;
  } else {
    sampFreqHz = 16000;
  }
  return 0;
}

WebRtc_Word32 ACMISAC::ConfigISACBandwidthEstimator(
    const WebRtc_UWord8 initFrameSizeMsec,
    const WebRtc_UWord16 initRateBitPerSec, const bool enforceFrameSize) {
  WebRtc_Word16 status;
  {
    WebRtc_UWord16 sampFreqHz;
    EncoderSampFreq(sampFreqHz);
    // TODO(turajs): at 32kHz we hardcode calling with 30ms and enforce
    // the frame-size otherwise we might get error. Revise if
    // control-bwe is changed.
    if (sampFreqHz == 32000) {
      status = ACM_ISAC_CONTROL_BWE(_codecInstPtr->inst, initRateBitPerSec, 30,
                                    1);
    } else {
      status = ACM_ISAC_CONTROL_BWE(_codecInstPtr->inst, initRateBitPerSec,
                                    initFrameSizeMsec,
                                    enforceFrameSize ? 1 : 0);
    }
  }
  if (status < 0) {
    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
                 "Couldn't config iSAC BWE.");
    return -1;
  }
  UpdateFrameLen();
  ACM_ISAC_GETSENDBITRATE(_codecInstPtr->inst, &_isacCurrentBN);
  return 0;
}

WebRtc_Word32 ACMISAC::SetISACMaxPayloadSize(
    const WebRtc_UWord16 maxPayloadLenBytes) {
  return ACM_ISAC_SETMAXPAYLOADSIZE(_codecInstPtr->inst, maxPayloadLenBytes);
}

WebRtc_Word32 ACMISAC::SetISACMaxRate(const WebRtc_UWord32 maxRateBitPerSec) {
  return ACM_ISAC_SETMAXRATE(_codecInstPtr->inst, maxRateBitPerSec);
}

void ACMISAC::UpdateFrameLen() {
  _frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
  _encoderParams.codecInstant.pacsize = _frameLenSmpl;
}

void ACMISAC::CurrentRate(WebRtc_Word32& rateBitPerSec) {
  if (_isacCodingMode == ADAPTIVE) {
    ACM_ISAC_GETSENDBITRATE(_codecInstPtr->inst, &rateBitPerSec);
  }
}

bool ACMISAC::DecoderParamsSafe(WebRtcACMCodecParams* decParams,
                                const WebRtc_UWord8 payloadType) {
  if (_decoderInitialized) {
    if (payloadType == _decoderParams.codecInstant.pltype) {
      memcpy(decParams, &_decoderParams, sizeof(WebRtcACMCodecParams));
      return true;
    }
    if (payloadType == _decoderParams32kHz.codecInstant.pltype) {
      memcpy(decParams, &_decoderParams32kHz, sizeof(WebRtcACMCodecParams));
      return true;
    }
  }
  return false;
}

void ACMISAC::SaveDecoderParamSafe(const WebRtcACMCodecParams* codecParams) {
  // set decoder sampling frequency.
  if (codecParams->codecInstant.plfreq == 32000) {
    memcpy(&_decoderParams32kHz, codecParams, sizeof(WebRtcACMCodecParams));
  } else {
    memcpy(&_decoderParams, codecParams, sizeof(WebRtcACMCodecParams));
  }
}

WebRtc_Word16 ACMISAC::REDPayloadISAC(const WebRtc_Word32 isacRate,
                                      const WebRtc_Word16 isacBwEstimate,
                                      WebRtc_UWord8* payload,
                                      WebRtc_Word16* payloadLenBytes) {
  WebRtc_Word16 status;
  ReadLockScoped rl(_codecWrapperLock);
  status = Transcode(payload, payloadLenBytes, isacBwEstimate, isacRate, true);
  return status;
}

#endif

}  // namespace webrtc