webrtc/media/engine/webrtcvoiceengine.cc

/*
 *  Copyright (c) 2004 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.
 */

#ifdef HAVE_WEBRTC_VOICE

#include "webrtc/media/engine/webrtcvoiceengine.h"

#include <algorithm>
#include <cstdio>
#include <functional>
#include <string>
#include <vector>

#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/arraysize.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/race_checker.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/base/trace_event.h"
#include "webrtc/media/base/audiosource.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/base/streamparams.h"
#include "webrtc/media/engine/adm_helpers.h"
#include "webrtc/media/engine/apm_helpers.h"
#include "webrtc/media/engine/payload_type_mapper.h"
#include "webrtc/media/engine/webrtcmediaengine.h"
#include "webrtc/media/engine/webrtcvoe.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/include/field_trial.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/voice_engine/transmit_mixer.h"

namespace cricket {
namespace {

constexpr size_t kMaxUnsignaledRecvStreams = 1;

const int kDefaultTraceFilter = webrtc::kTraceNone | webrtc::kTraceTerseInfo |
                                webrtc::kTraceWarning | webrtc::kTraceError |
                                webrtc::kTraceCritical;
const int kElevatedTraceFilter = kDefaultTraceFilter | webrtc::kTraceStateInfo |
                                 webrtc::kTraceInfo;

constexpr int kNackRtpHistoryMs = 5000;

// Check to verify that the define for the intelligibility enhancer is properly
// set.
#if !defined(WEBRTC_INTELLIGIBILITY_ENHANCER) || \
    (WEBRTC_INTELLIGIBILITY_ENHANCER != 0 &&     \
     WEBRTC_INTELLIGIBILITY_ENHANCER != 1)
#error "Set WEBRTC_INTELLIGIBILITY_ENHANCER to either 0 or 1"
#endif

// Codec parameters for Opus.
// draft-spittka-payload-rtp-opus-03

// Recommended bitrates:
// 8-12 kb/s for NB speech,
// 16-20 kb/s for WB speech,
// 28-40 kb/s for FB speech,
// 48-64 kb/s for FB mono music, and
// 64-128 kb/s for FB stereo music.
// The current implementation applies the following values to mono signals,
// and multiplies them by 2 for stereo.
const int kOpusBitrateNbBps = 12000;
const int kOpusBitrateWbBps = 20000;
const int kOpusBitrateFbBps = 32000;

// Opus bitrate should be in the range between 6000 and 510000.
const int kOpusMinBitrateBps = 6000;
const int kOpusMaxBitrateBps = 510000;

// iSAC bitrate should be <= 56000.
const int kIsacMaxBitrateBps = 56000;

// Default audio dscp value.
// See http://tools.ietf.org/html/rfc2474 for details.
// See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;

// Constants from voice_engine_defines.h.
const int kMinTelephoneEventCode = 0;           // RFC4733 (Section 2.3.1)
const int kMaxTelephoneEventCode = 255;
const int kMinTelephoneEventDuration = 100;
const int kMaxTelephoneEventDuration = 60000;   // Actual limit is 2^16

const int kMinPayloadType = 0;
const int kMaxPayloadType = 127;

class ProxySink : public webrtc::AudioSinkInterface {
 public:
  ProxySink(AudioSinkInterface* sink) : sink_(sink) { RTC_DCHECK(sink); }

  void OnData(const Data& audio) override { sink_->OnData(audio); }

 private:
  webrtc::AudioSinkInterface* sink_;
};

bool ValidateStreamParams(const StreamParams& sp) {
  if (sp.ssrcs.empty()) {
    LOG(LS_ERROR) << "No SSRCs in stream parameters: " << sp.ToString();
    return false;
  }
  if (sp.ssrcs.size() > 1) {
    LOG(LS_ERROR) << "Multiple SSRCs in stream parameters: " << sp.ToString();
    return false;
  }
  return true;
}

// Dumps an AudioCodec in RFC 2327-ish format.
std::string ToString(const AudioCodec& codec) {
  std::stringstream ss;
  ss << codec.name << "/" << codec.clockrate << "/" << codec.channels
     << " (" << codec.id << ")";
  return ss.str();
}

bool IsCodec(const AudioCodec& codec, const char* ref_name) {
  return (_stricmp(codec.name.c_str(), ref_name) == 0);
}

bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
  return (_stricmp(codec.plname, ref_name) == 0);
}

bool FindCodec(const std::vector<AudioCodec>& codecs,
               const AudioCodec& codec,
               AudioCodec* found_codec) {
  for (const AudioCodec& c : codecs) {
    if (c.Matches(codec)) {
      if (found_codec != NULL) {
        *found_codec = c;
      }
      return true;
    }
  }
  return false;
}

bool VerifyUniquePayloadTypes(const std::vector<AudioCodec>& codecs) {
  if (codecs.empty()) {
    return true;
  }
  std::vector<int> payload_types;
  for (const AudioCodec& codec : codecs) {
    payload_types.push_back(codec.id);
  }
  std::sort(payload_types.begin(), payload_types.end());
  auto it = std::unique(payload_types.begin(), payload_types.end());
  return it == payload_types.end();
}

// Return true if codec.params[feature] == "1", false otherwise.
bool IsCodecFeatureEnabled(const AudioCodec& codec, const char* feature) {
  int value;
  return codec.GetParam(feature, &value) && value == 1;
}

rtc::Optional<std::string> GetAudioNetworkAdaptorConfig(
    const AudioOptions& options) {
  if (options.audio_network_adaptor && *options.audio_network_adaptor &&
      options.audio_network_adaptor_config) {
    // Turn on audio network adaptor only when |options_.audio_network_adaptor|
    // equals true and |options_.audio_network_adaptor_config| has a value.
    return options.audio_network_adaptor_config;
  }
  return rtc::Optional<std::string>();
}

// Returns integer parameter params[feature] if it is defined. Returns
// |default_value| otherwise.
int GetCodecFeatureInt(const AudioCodec& codec,
                       const char* feature,
                       int default_value) {
  int value = 0;
  if (codec.GetParam(feature, &value)) {
    return value;
  }
  return default_value;
}

// Use params[kCodecParamMaxAverageBitrate] if it is defined, use codec.bitrate
// otherwise. If the value (either from params or codec.bitrate) <=0, use the
// default configuration. If the value is beyond feasible bit rate of Opus,
// clamp it. Returns the Opus bit rate for operation.
int GetOpusBitrate(const AudioCodec& codec, int max_playback_rate) {
  int bitrate = 0;
  bool use_param = true;
  if (!codec.GetParam(kCodecParamMaxAverageBitrate, &bitrate)) {
    bitrate = codec.bitrate;
    use_param = false;
  }
  if (bitrate <= 0) {
    if (max_playback_rate <= 8000) {
      bitrate = kOpusBitrateNbBps;
    } else if (max_playback_rate <= 16000) {
      bitrate = kOpusBitrateWbBps;
    } else {
      bitrate = kOpusBitrateFbBps;
    }

    if (IsCodecFeatureEnabled(codec, kCodecParamStereo)) {
      bitrate *= 2;
    }
  } else if (bitrate < kOpusMinBitrateBps || bitrate > kOpusMaxBitrateBps) {
    bitrate = (bitrate < kOpusMinBitrateBps) ? kOpusMinBitrateBps
                                             : kOpusMaxBitrateBps;
    std::string rate_source =
        use_param ? "Codec parameter \"maxaveragebitrate\"" :
            "Supplied Opus bitrate";
    LOG(LS_WARNING) << rate_source
                    << " is invalid and is replaced by: "
                    << bitrate;
  }
  return bitrate;
}

void GetOpusConfig(const AudioCodec& codec,
                   webrtc::CodecInst* voe_codec,
                   bool* enable_codec_fec,
                   int* max_playback_rate,
                   bool* enable_codec_dtx,
                   int* min_ptime_ms,
                   int* max_ptime_ms) {
  *enable_codec_fec = IsCodecFeatureEnabled(codec, kCodecParamUseInbandFec);
  *enable_codec_dtx = IsCodecFeatureEnabled(codec, kCodecParamUseDtx);
  *max_playback_rate = GetCodecFeatureInt(codec, kCodecParamMaxPlaybackRate,
                                          kOpusDefaultMaxPlaybackRate);
  *max_ptime_ms =
      GetCodecFeatureInt(codec, kCodecParamMaxPTime, kOpusDefaultMaxPTime);
  *min_ptime_ms =
      GetCodecFeatureInt(codec, kCodecParamMinPTime, kOpusDefaultMinPTime);
  if (*max_ptime_ms < *min_ptime_ms) {
    // If min ptime or max ptime defined by codec parameter is wrong, we use
    // the default values.
    *max_ptime_ms = kOpusDefaultMaxPTime;
    *min_ptime_ms = kOpusDefaultMinPTime;
  }

  // If OPUS, change what we send according to the "stereo" codec
  // parameter, and not the "channels" parameter.  We set
  // voe_codec.channels to 2 if "stereo=1" and 1 otherwise.  If
  // the bitrate is not specified, i.e. is <= zero, we set it to the
  // appropriate default value for mono or stereo Opus.
  voe_codec->channels = IsCodecFeatureEnabled(codec, kCodecParamStereo) ? 2 : 1;
  voe_codec->rate = GetOpusBitrate(codec, *max_playback_rate);
}

webrtc::AudioState::Config MakeAudioStateConfig(
    VoEWrapper* voe_wrapper,
    rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
  webrtc::AudioState::Config config;
  config.voice_engine = voe_wrapper->engine();
  if (audio_mixer) {
    config.audio_mixer = audio_mixer;
  } else {
    config.audio_mixer = webrtc::AudioMixerImpl::Create();
  }
  return config;
}

class WebRtcVoiceCodecs final {
 public:
  // TODO(solenberg): Do this filtering once off-line, add a simple AudioCodec
  // list and add a test which verifies VoE supports the listed codecs.
  static std::vector<AudioCodec> SupportedSendCodecs() {
    std::vector<AudioCodec> result;
    // Iterate first over our preferred codecs list, so that the results are
    // added in order of preference.
    for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
      const CodecPref* pref = &kCodecPrefs[i];
      for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
        // Change the sample rate of G722 to 8000 to match SDP.
        MaybeFixupG722(&voe_codec, 8000);
        // Skip uncompressed formats.
        if (IsCodec(voe_codec, kL16CodecName)) {
          continue;
        }

        if (!IsCodec(voe_codec, pref->name) ||
            pref->clockrate != voe_codec.plfreq ||
            pref->channels != voe_codec.channels) {
          // Not a match.
          continue;
        }

        AudioCodec codec(pref->payload_type, voe_codec.plname, voe_codec.plfreq,
                         voe_codec.rate, voe_codec.channels);
        LOG(LS_INFO) << "Adding supported codec: " << ToString(codec);
        if (IsCodec(codec, kIsacCodecName)) {
          // Indicate auto-bitrate in signaling.
          codec.bitrate = 0;
        }
        if (IsCodec(codec, kOpusCodecName)) {
          // Only add fmtp parameters that differ from the spec.
          if (kPreferredMinPTime != kOpusDefaultMinPTime) {
            codec.params[kCodecParamMinPTime] =
                rtc::ToString(kPreferredMinPTime);
          }
          if (kPreferredMaxPTime != kOpusDefaultMaxPTime) {
            codec.params[kCodecParamMaxPTime] =
                rtc::ToString(kPreferredMaxPTime);
          }
          codec.SetParam(kCodecParamUseInbandFec, 1);
          codec.AddFeedbackParam(
              FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));

          // TODO(hellner): Add ptime, sprop-stereo, and stereo
          // when they can be set to values other than the default.
        }
        result.push_back(codec);
      }
    }
    return result;
  }

  static bool ToCodecInst(const AudioCodec& in,
                          webrtc::CodecInst* out) {
    for (webrtc::CodecInst voe_codec : webrtc::acm2::RentACodec::Database()) {
      // Change the sample rate of G722 to 8000 to match SDP.
      MaybeFixupG722(&voe_codec, 8000);
      AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
                       voe_codec.rate, voe_codec.channels);
      bool multi_rate = IsCodecMultiRate(voe_codec);
      // Allow arbitrary rates for ISAC to be specified.
      if (multi_rate) {
        // Set codec.bitrate to 0 so the check for codec.Matches() passes.
        codec.bitrate = 0;
      }
      if (codec.Matches(in)) {
        if (out) {
          // Fixup the payload type.
          voe_codec.pltype = in.id;

          // Set bitrate if specified.
          if (multi_rate && in.bitrate != 0) {
            voe_codec.rate = in.bitrate;
          }

          // Reset G722 sample rate to 16000 to match WebRTC.
          MaybeFixupG722(&voe_codec, 16000);

          *out = voe_codec;
        }
        return true;
      }
    }
    return false;
  }

  static bool IsCodecMultiRate(const webrtc::CodecInst& codec) {
    for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
      if (IsCodec(codec, kCodecPrefs[i].name) &&
          kCodecPrefs[i].clockrate == codec.plfreq) {
        return kCodecPrefs[i].is_multi_rate;
      }
    }
    return false;
  }

  static int MaxBitrateBps(const webrtc::CodecInst& codec) {
    for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
      if (IsCodec(codec, kCodecPrefs[i].name) &&
          kCodecPrefs[i].clockrate == codec.plfreq) {
        return kCodecPrefs[i].max_bitrate_bps;
      }
    }
    return 0;
  }

  static rtc::ArrayView<const int> GetPacketSizesMs(
      const webrtc::CodecInst& codec) {
    for (size_t i = 0; i < arraysize(kCodecPrefs); ++i) {
      if (IsCodec(codec, kCodecPrefs[i].name)) {
        size_t num_packet_sizes = kMaxNumPacketSize;
        for (int index = 0; index < kMaxNumPacketSize; index++) {
          if (kCodecPrefs[i].packet_sizes_ms[index] == 0) {
            num_packet_sizes = index;
            break;
          }
        }
        return rtc::ArrayView<const int>(kCodecPrefs[i].packet_sizes_ms,
                                         num_packet_sizes);
      }
    }
    return rtc::ArrayView<const int>();
  }

  // If the AudioCodec param kCodecParamPTime is set, then we will set it to
  // codec pacsize if it's valid, or we will pick the next smallest value we
  // support.
  // TODO(Brave): Query supported packet sizes from ACM when the API is ready.
  static bool SetPTimeAsPacketSize(webrtc::CodecInst* codec, int ptime_ms) {
    for (const CodecPref& codec_pref : kCodecPrefs) {
      if ((IsCodec(*codec, codec_pref.name) &&
          codec_pref.clockrate == codec->plfreq) ||
          IsCodec(*codec, kG722CodecName)) {
        int packet_size_ms = SelectPacketSize(codec_pref, ptime_ms);
        if (packet_size_ms) {
          // Convert unit from milli-seconds to samples.
          codec->pacsize = (codec->plfreq / 1000) * packet_size_ms;
          return true;
        }
      }
    }
    return false;
  }

  static const AudioCodec* GetPreferredCodec(
      const std::vector<AudioCodec>& codecs,
      webrtc::CodecInst* out) {
    RTC_DCHECK(out);
    // Select the preferred send codec (the first non-telephone-event/CN codec).
    for (const AudioCodec& codec : codecs) {
      if (IsCodec(codec, kDtmfCodecName) || IsCodec(codec, kCnCodecName)) {
        // Skip telephone-event/CN codecs - they will be handled later.
        continue;
      }

      // We'll use the first codec in the list to actually send audio data.
      // Be sure to use the payload type requested by the remote side.
      // Ignore codecs we don't know about. The negotiation step should prevent
      // this, but double-check to be sure.
      if (!ToCodecInst(codec, out)) {
        LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
        continue;
      }
      return &codec;
    }
    return nullptr;
  }

 private:
  static const int kMaxNumPacketSize = 6;
  struct CodecPref {
    const char* name;
    int clockrate;
    size_t channels;
    int payload_type;
    bool is_multi_rate;
    int packet_sizes_ms[kMaxNumPacketSize];
    int max_bitrate_bps;
  };
  // Note: keep the supported packet sizes in ascending order.
  static const CodecPref kCodecPrefs[14];

  static int SelectPacketSize(const CodecPref& codec_pref, int ptime_ms) {
    int selected_packet_size_ms = codec_pref.packet_sizes_ms[0];
    for (int packet_size_ms : codec_pref.packet_sizes_ms) {
      if (packet_size_ms && packet_size_ms <= ptime_ms) {
        selected_packet_size_ms = packet_size_ms;
      }
    }
    return selected_packet_size_ms;
  }

  // Changes RTP timestamp rate of G722. This is due to the "bug" in the RFC
  // which says that G722 should be advertised as 8 kHz although it is a 16 kHz
  // codec.
  static void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
    if (IsCodec(*voe_codec, kG722CodecName)) {
      // If the DCHECK triggers, the codec definition in WebRTC VoiceEngine
      // has changed, and this special case is no longer needed.
      RTC_DCHECK(voe_codec->plfreq != new_plfreq);
      voe_codec->plfreq = new_plfreq;
    }
  }
};

const WebRtcVoiceCodecs::CodecPref WebRtcVoiceCodecs::kCodecPrefs[14] = {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
    {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60, 120},
     kOpusMaxBitrateBps},
#else
    {kOpusCodecName, 48000, 2, 111, true, {10, 20, 40, 60}, kOpusMaxBitrateBps},
#endif
    {kIsacCodecName, 16000, 1, 103, true, {30, 60}, kIsacMaxBitrateBps},
    {kIsacCodecName, 32000, 1, 104, true, {30}, kIsacMaxBitrateBps},
    // G722 should be advertised as 8000 Hz because of the RFC "bug".
    {kG722CodecName, 8000, 1, 9, false, {10, 20, 30, 40, 50, 60}},
    {kIlbcCodecName, 8000, 1, 102, false, {20, 30, 40, 60}},
    {kPcmuCodecName, 8000, 1, 0, false, {10, 20, 30, 40, 50, 60}},
    {kPcmaCodecName, 8000, 1, 8, false, {10, 20, 30, 40, 50, 60}},
    {kCnCodecName, 32000, 1, 106, false, {}},
    {kCnCodecName, 16000, 1, 105, false, {}},
    {kCnCodecName, 8000, 1, 13, false, {}},
    {kDtmfCodecName, 48000, 1, 110, false, {}},
    {kDtmfCodecName, 32000, 1, 112, false, {}},
    {kDtmfCodecName, 16000, 1, 113, false, {}},
    {kDtmfCodecName, 8000, 1, 126, false, {}}
};

// |max_send_bitrate_bps| is the bitrate from "b=" in SDP.
// |rtp_max_bitrate_bps| is the bitrate from RtpSender::SetParameters.
rtc::Optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
                                      rtc::Optional<int> rtp_max_bitrate_bps,
                                      const webrtc::CodecInst& codec_inst) {
  // If application-configured bitrate is set, take minimum of that and SDP
  // bitrate.
  const int bps = rtp_max_bitrate_bps
                      ? MinPositive(max_send_bitrate_bps, *rtp_max_bitrate_bps)
                      : max_send_bitrate_bps;
  const int codec_rate = codec_inst.rate;

  if (bps <= 0) {
    return rtc::Optional<int>(codec_rate);
  }

  if (codec_inst.pltype == -1) {
    return rtc::Optional<int>(codec_rate);
    ;
  }

  if (WebRtcVoiceCodecs::IsCodecMultiRate(codec_inst)) {
    // If codec is multi-rate then just set the bitrate.
    return rtc::Optional<int>(
        std::min(bps, WebRtcVoiceCodecs::MaxBitrateBps(codec_inst)));
  }

  if (bps < codec_inst.rate) {
    // If codec is not multi-rate and |bps| is less than the fixed bitrate then
    // fail. If codec is not multi-rate and |bps| exceeds or equal the fixed
    // bitrate then ignore.
    LOG(LS_ERROR) << "Failed to set codec " << codec_inst.plname
                  << " to bitrate " << bps << " bps"
                  << ", requires at least " << codec_inst.rate << " bps.";
    return rtc::Optional<int>();
  }
  return rtc::Optional<int>(codec_rate);
}

}  // namespace

bool WebRtcVoiceEngine::ToCodecInst(const AudioCodec& in,
                                    webrtc::CodecInst* out) {
  return WebRtcVoiceCodecs::ToCodecInst(in, out);
}

WebRtcVoiceEngine::WebRtcVoiceEngine(
    webrtc::AudioDeviceModule* adm,
    const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
    rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer)
    : WebRtcVoiceEngine(adm, decoder_factory, audio_mixer, new VoEWrapper()) {
  audio_state_ =
      webrtc::AudioState::Create(MakeAudioStateConfig(voe(), audio_mixer));
}

WebRtcVoiceEngine::WebRtcVoiceEngine(
    webrtc::AudioDeviceModule* adm,
    const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
    rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
    VoEWrapper* voe_wrapper)
    : adm_(adm), decoder_factory_(decoder_factory), voe_wrapper_(voe_wrapper) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
  RTC_DCHECK(voe_wrapper);
  RTC_DCHECK(decoder_factory);

  signal_thread_checker_.DetachFromThread();

  // Load our audio codec list.
  LOG(LS_INFO) << "Supported send codecs in order of preference:";
  send_codecs_ = WebRtcVoiceCodecs::SupportedSendCodecs();
  for (const AudioCodec& codec : send_codecs_) {
    LOG(LS_INFO) << ToString(codec);
  }

  LOG(LS_INFO) << "Supported recv codecs in order of preference:";
  recv_codecs_ = CollectRecvCodecs();
  for (const AudioCodec& codec : recv_codecs_) {
    LOG(LS_INFO) << ToString(codec);
  }

  channel_config_.enable_voice_pacing = true;

  // Temporarily turn logging level up for the Init() call.
  webrtc::Trace::SetTraceCallback(this);
  webrtc::Trace::set_level_filter(kElevatedTraceFilter);
  LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
  RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm_.get(), nullptr,
                                             decoder_factory_));
  webrtc::Trace::set_level_filter(kDefaultTraceFilter);

  // No ADM supplied? Get the default one from VoE.
  if (!adm_) {
    adm_ = voe_wrapper_->base()->audio_device_module();
  }
  RTC_DCHECK(adm_);

  apm_ = voe_wrapper_->base()->audio_processing();
  RTC_DCHECK(apm_);

  transmit_mixer_ = voe_wrapper_->base()->transmit_mixer();
  RTC_DCHECK(transmit_mixer_);

  // Save the default AGC configuration settings. This must happen before
  // calling ApplyOptions or the default will be overwritten.
  default_agc_config_ = webrtc::apm_helpers::GetAgcConfig(apm_);

  // Set default engine options.
  {
    AudioOptions options;
    options.echo_cancellation = rtc::Optional<bool>(true);
    options.auto_gain_control = rtc::Optional<bool>(true);
    options.noise_suppression = rtc::Optional<bool>(true);
    options.highpass_filter = rtc::Optional<bool>(true);
    options.stereo_swapping = rtc::Optional<bool>(false);
    options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
    options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
    options.typing_detection = rtc::Optional<bool>(true);
    options.adjust_agc_delta = rtc::Optional<int>(0);
    options.experimental_agc = rtc::Optional<bool>(false);
    options.extended_filter_aec = rtc::Optional<bool>(false);
    options.delay_agnostic_aec = rtc::Optional<bool>(false);
    options.experimental_ns = rtc::Optional<bool>(false);
    options.intelligibility_enhancer = rtc::Optional<bool>(false);
    options.level_control = rtc::Optional<bool>(false);
    options.residual_echo_detector = rtc::Optional<bool>(true);
    bool error = ApplyOptions(options);
    RTC_DCHECK(error);
  }

  // Set default audio devices.
#if !defined(WEBRTC_IOS)
  webrtc::adm_helpers::SetRecordingDevice(adm_);
  apm()->Initialize();
  webrtc::adm_helpers::SetPlayoutDevice(adm_);
#endif  // !WEBRTC_IOS
}

WebRtcVoiceEngine::~WebRtcVoiceEngine() {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
  StopAecDump();
  voe_wrapper_->base()->Terminate();
  webrtc::Trace::SetTraceCallback(nullptr);
}

rtc::scoped_refptr<webrtc::AudioState>
    WebRtcVoiceEngine::GetAudioState() const {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  return audio_state_;
}

VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
    webrtc::Call* call,
    const MediaConfig& config,
    const AudioOptions& options) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  return new WebRtcVoiceMediaChannel(this, config, options, call);
}

bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "WebRtcVoiceEngine::ApplyOptions: " << options_in.ToString();
  AudioOptions options = options_in;  // The options are modified below.

  // kEcConference is AEC with high suppression.
  webrtc::EcModes ec_mode = webrtc::kEcConference;
  if (options.aecm_generate_comfort_noise) {
    LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
                    << *options.aecm_generate_comfort_noise
                    << " (default is false).";
  }

#if defined(WEBRTC_IOS)
  // On iOS, VPIO provides built-in EC, NS and AGC.
  options.echo_cancellation = rtc::Optional<bool>(false);
  options.auto_gain_control = rtc::Optional<bool>(false);
  options.noise_suppression = rtc::Optional<bool>(false);
  LOG(LS_INFO)
      << "Always disable AEC, NS and AGC on iOS. Use built-in instead.";
#elif defined(ANDROID)
  ec_mode = webrtc::kEcAecm;
#endif

#if defined(WEBRTC_IOS) || defined(ANDROID)
  options.typing_detection = rtc::Optional<bool>(false);
  options.experimental_agc = rtc::Optional<bool>(false);
  options.extended_filter_aec = rtc::Optional<bool>(false);
  options.experimental_ns = rtc::Optional<bool>(false);
#endif

  // Delay Agnostic AEC automatically turns on EC if not set except on iOS
  // where the feature is not supported.
  bool use_delay_agnostic_aec = false;
#if !defined(WEBRTC_IOS)
  if (options.delay_agnostic_aec) {
    use_delay_agnostic_aec = *options.delay_agnostic_aec;
    if (use_delay_agnostic_aec) {
      options.echo_cancellation = rtc::Optional<bool>(true);
      options.extended_filter_aec = rtc::Optional<bool>(true);
      ec_mode = webrtc::kEcConference;
    }
  }
#endif

#if (WEBRTC_INTELLIGIBILITY_ENHANCER == 0)
  // Hardcode the intelligibility enhancer to be off.
  options.intelligibility_enhancer = rtc::Optional<bool>(false);
#endif

  if (options.echo_cancellation) {
    // Check if platform supports built-in EC. Currently only supported on
    // Android and in combination with Java based audio layer.
    // TODO(henrika): investigate possibility to support built-in EC also
    // in combination with Open SL ES audio.
    const bool built_in_aec = adm()->BuiltInAECIsAvailable();
    if (built_in_aec) {
      // Built-in EC exists on this device and use_delay_agnostic_aec is not
      // overriding it. Enable/Disable it according to the echo_cancellation
      // audio option.
      const bool enable_built_in_aec =
          *options.echo_cancellation && !use_delay_agnostic_aec;
      if (adm()->EnableBuiltInAEC(enable_built_in_aec) == 0 &&
          enable_built_in_aec) {
        // Disable internal software EC if built-in EC is enabled,
        // i.e., replace the software EC with the built-in EC.
        options.echo_cancellation = rtc::Optional<bool>(false);
        LOG(LS_INFO) << "Disabling EC since built-in EC will be used instead";
      }
    }
    webrtc::apm_helpers::SetEcStatus(
        apm(), *options.echo_cancellation, ec_mode);
#if !defined(ANDROID)
    webrtc::apm_helpers::SetEcMetricsStatus(apm(), *options.echo_cancellation);
#endif
    if (ec_mode == webrtc::kEcAecm) {
      bool cn = options.aecm_generate_comfort_noise.value_or(false);
      webrtc::apm_helpers::SetAecmMode(apm(), cn);
    }
  }

  if (options.auto_gain_control) {
    bool built_in_agc_avaliable = adm()->BuiltInAGCIsAvailable();
    if (built_in_agc_avaliable) {
      if (adm()->EnableBuiltInAGC(*options.auto_gain_control) == 0 &&
          *options.auto_gain_control) {
        // Disable internal software AGC if built-in AGC is enabled,
        // i.e., replace the software AGC with the built-in AGC.
        options.auto_gain_control = rtc::Optional<bool>(false);
        LOG(LS_INFO) << "Disabling AGC since built-in AGC will be used instead";
      }
    }
    webrtc::apm_helpers::SetAgcStatus(apm(), adm(), *options.auto_gain_control);
  }

  if (options.tx_agc_target_dbov || options.tx_agc_digital_compression_gain ||
      options.tx_agc_limiter || options.adjust_agc_delta) {
    // Override default_agc_config_. Generally, an unset option means "leave
    // the VoE bits alone" in this function, so we want whatever is set to be
    // stored as the new "default". If we didn't, then setting e.g.
    // tx_agc_target_dbov would reset digital compression gain and limiter
    // settings.
    // Also, if we don't update default_agc_config_, then adjust_agc_delta
    // would be an offset from the original values, and not whatever was set
    // explicitly.
    default_agc_config_.targetLeveldBOv = options.tx_agc_target_dbov.value_or(
        default_agc_config_.targetLeveldBOv);
    default_agc_config_.digitalCompressionGaindB =
        options.tx_agc_digital_compression_gain.value_or(
            default_agc_config_.digitalCompressionGaindB);
    default_agc_config_.limiterEnable =
        options.tx_agc_limiter.value_or(default_agc_config_.limiterEnable);

    webrtc::AgcConfig config = default_agc_config_;
    if (options.adjust_agc_delta) {
      config.targetLeveldBOv -= *options.adjust_agc_delta;
      LOG(LS_INFO) << "Adjusting AGC level from default -"
                   << default_agc_config_.targetLeveldBOv << "dB to -"
                   << config.targetLeveldBOv << "dB";
    }
    webrtc::apm_helpers::SetAgcConfig(apm_, config);
  }

  if (options.intelligibility_enhancer) {
    intelligibility_enhancer_ = options.intelligibility_enhancer;
  }
  if (intelligibility_enhancer_ && *intelligibility_enhancer_) {
    LOG(LS_INFO) << "Enabling NS when Intelligibility Enhancer is active.";
    options.noise_suppression = intelligibility_enhancer_;
  }

  if (options.noise_suppression) {
    if (adm()->BuiltInNSIsAvailable()) {
      bool builtin_ns =
          *options.noise_suppression &&
          !(intelligibility_enhancer_ && *intelligibility_enhancer_);
      if (adm()->EnableBuiltInNS(builtin_ns) == 0 && builtin_ns) {
        // Disable internal software NS if built-in NS is enabled,
        // i.e., replace the software NS with the built-in NS.
        options.noise_suppression = rtc::Optional<bool>(false);
        LOG(LS_INFO) << "Disabling NS since built-in NS will be used instead";
      }
    }
    webrtc::apm_helpers::SetNsStatus(apm(), *options.noise_suppression);
  }

  if (options.stereo_swapping) {
    LOG(LS_INFO) << "Stereo swapping enabled? " << *options.stereo_swapping;
    transmit_mixer()->EnableStereoChannelSwapping(*options.stereo_swapping);
  }

  if (options.audio_jitter_buffer_max_packets) {
    LOG(LS_INFO) << "NetEq capacity is "
                 << *options.audio_jitter_buffer_max_packets;
    channel_config_.acm_config.neteq_config.max_packets_in_buffer =
        std::max(20, *options.audio_jitter_buffer_max_packets);
  }
  if (options.audio_jitter_buffer_fast_accelerate) {
    LOG(LS_INFO) << "NetEq fast mode? "
                 << *options.audio_jitter_buffer_fast_accelerate;
    channel_config_.acm_config.neteq_config.enable_fast_accelerate =
        *options.audio_jitter_buffer_fast_accelerate;
  }

  if (options.typing_detection) {
    LOG(LS_INFO) << "Typing detection is enabled? "
                 << *options.typing_detection;
    webrtc::apm_helpers::SetTypingDetectionStatus(
        apm(), *options.typing_detection);
  }

  webrtc::Config config;

  if (options.delay_agnostic_aec)
    delay_agnostic_aec_ = options.delay_agnostic_aec;
  if (delay_agnostic_aec_) {
    LOG(LS_INFO) << "Delay agnostic aec is enabled? " << *delay_agnostic_aec_;
    config.Set<webrtc::DelayAgnostic>(
        new webrtc::DelayAgnostic(*delay_agnostic_aec_));
  }

  if (options.extended_filter_aec) {
    extended_filter_aec_ = options.extended_filter_aec;
  }
  if (extended_filter_aec_) {
    LOG(LS_INFO) << "Extended filter aec is enabled? " << *extended_filter_aec_;
    config.Set<webrtc::ExtendedFilter>(
        new webrtc::ExtendedFilter(*extended_filter_aec_));
  }

  if (options.experimental_ns) {
    experimental_ns_ = options.experimental_ns;
  }
  if (experimental_ns_) {
    LOG(LS_INFO) << "Experimental ns is enabled? " << *experimental_ns_;
    config.Set<webrtc::ExperimentalNs>(
        new webrtc::ExperimentalNs(*experimental_ns_));
  }

  if (intelligibility_enhancer_) {
    LOG(LS_INFO) << "Intelligibility Enhancer is enabled? "
                 << *intelligibility_enhancer_;
    config.Set<webrtc::Intelligibility>(
        new webrtc::Intelligibility(*intelligibility_enhancer_));
  }

  if (options.level_control) {
    level_control_ = options.level_control;
  }

  LOG(LS_INFO) << "Level control: "
               << (!!level_control_ ? *level_control_ : -1);
  if (level_control_) {
    apm_config_.level_controller.enabled = *level_control_;
    if (options.level_control_initial_peak_level_dbfs) {
      apm_config_.level_controller.initial_peak_level_dbfs =
          *options.level_control_initial_peak_level_dbfs;
    }
  }

  if (options.highpass_filter) {
    apm_config_.high_pass_filter.enabled = *options.highpass_filter;
  }

  if (options.residual_echo_detector) {
    apm_config_.residual_echo_detector.enabled =
        *options.residual_echo_detector;
  }

  apm()->SetExtraOptions(config);
  apm()->ApplyConfig(apm_config_);

  if (options.recording_sample_rate) {
    LOG(LS_INFO) << "Recording sample rate is "
                 << *options.recording_sample_rate;
    if (adm()->SetRecordingSampleRate(*options.recording_sample_rate)) {
      LOG_RTCERR1(SetRecordingSampleRate, *options.recording_sample_rate);
    }
  }

  if (options.playout_sample_rate) {
    LOG(LS_INFO) << "Playout sample rate is " << *options.playout_sample_rate;
    if (adm()->SetPlayoutSampleRate(*options.playout_sample_rate)) {
      LOG_RTCERR1(SetPlayoutSampleRate, *options.playout_sample_rate);
    }
  }
  return true;
}

// TODO(solenberg): Remove, once AudioMonitor is gone.
int WebRtcVoiceEngine::GetInputLevel() {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  int8_t level = transmit_mixer()->AudioLevel();
  RTC_DCHECK_LE(0, level);
  return level;
}

const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const {
  RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
  return send_codecs_;
}

const std::vector<AudioCodec>& WebRtcVoiceEngine::recv_codecs() const {
  RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
  return recv_codecs_;
}

RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
  RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
  RtpCapabilities capabilities;
  capabilities.header_extensions.push_back(
      webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri,
                           webrtc::RtpExtension::kAudioLevelDefaultId));
  if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
    capabilities.header_extensions.push_back(webrtc::RtpExtension(
        webrtc::RtpExtension::kTransportSequenceNumberUri,
        webrtc::RtpExtension::kTransportSequenceNumberDefaultId));
  }
  return capabilities;
}

int WebRtcVoiceEngine::GetLastEngineError() {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  return voe_wrapper_->error();
}

void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
                              int length) {
  // Note: This callback can happen on any thread!
  rtc::LoggingSeverity sev = rtc::LS_VERBOSE;
  if (level == webrtc::kTraceError || level == webrtc::kTraceCritical)
    sev = rtc::LS_ERROR;
  else if (level == webrtc::kTraceWarning)
    sev = rtc::LS_WARNING;
  else if (level == webrtc::kTraceStateInfo || level == webrtc::kTraceInfo)
    sev = rtc::LS_INFO;
  else if (level == webrtc::kTraceTerseInfo)
    sev = rtc::LS_INFO;

  // Skip past boilerplate prefix text.
  if (length < 72) {
    std::string msg(trace, length);
    LOG(LS_ERROR) << "Malformed webrtc log message: ";
    LOG_V(sev) << msg;
  } else {
    std::string msg(trace + 71, length - 72);
    LOG_V(sev) << "webrtc: " << msg;
  }
}

void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  RTC_DCHECK(channel);
  channels_.push_back(channel);
}

void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  auto it = std::find(channels_.begin(), channels_.end(), channel);
  RTC_DCHECK(it != channels_.end());
  channels_.erase(it);
}

bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
                                     int64_t max_size_bytes) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
  if (!aec_dump_file_stream) {
    LOG(LS_ERROR) << "Could not open AEC dump file stream.";
    if (!rtc::ClosePlatformFile(file))
      LOG(LS_WARNING) << "Could not close file.";
    return false;
  }
  StopAecDump();
  if (apm()->StartDebugRecording(aec_dump_file_stream, max_size_bytes) !=
      webrtc::AudioProcessing::kNoError) {
    LOG_RTCERR0(StartDebugRecording);
    fclose(aec_dump_file_stream);
    return false;
  }
  is_dumping_aec_ = true;
  return true;
}

void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  if (!is_dumping_aec_) {
    // Start dumping AEC when we are not dumping.
    if (apm()->StartDebugRecording(filename.c_str(), -1) !=
        webrtc::AudioProcessing::kNoError) {
      LOG_RTCERR1(StartDebugRecording, filename.c_str());
    } else {
      is_dumping_aec_ = true;
    }
  }
}

void WebRtcVoiceEngine::StopAecDump() {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  if (is_dumping_aec_) {
    // Stop dumping AEC when we are dumping.
    if (apm()->StopDebugRecording() != webrtc::AudioProcessing::kNoError) {
      LOG_RTCERR0(StopDebugRecording);
    }
    is_dumping_aec_ = false;
  }
}

int WebRtcVoiceEngine::CreateVoEChannel() {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  return voe_wrapper_->base()->CreateChannel(channel_config_);
}

webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  RTC_DCHECK(adm_);
  return adm_;
}

webrtc::AudioProcessing* WebRtcVoiceEngine::apm() {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  RTC_DCHECK(apm_);
  return apm_;
}

webrtc::voe::TransmitMixer* WebRtcVoiceEngine::transmit_mixer() {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  RTC_DCHECK(transmit_mixer_);
  return transmit_mixer_;
}

AudioCodecs WebRtcVoiceEngine::CollectRecvCodecs() const {
  PayloadTypeMapper mapper;
  AudioCodecs out;
  const std::vector<webrtc::AudioCodecSpec>& specs =
      decoder_factory_->GetSupportedDecoders();

  // Only generate CN payload types for these clockrates:
  std::map<int, bool, std::greater<int>> generate_cn = {{ 8000,  false },
                                                        { 16000, false },
                                                        { 32000, false }};
  // Only generate telephone-event payload types for these clockrates:
  std::map<int, bool, std::greater<int>> generate_dtmf = {{ 8000,  false },
                                                          { 16000, false },
                                                          { 32000, false },
                                                          { 48000, false }};

  auto map_format = [&mapper](const webrtc::SdpAudioFormat& format,
                              AudioCodecs* out) {
    rtc::Optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
    if (opt_codec) {
      if (out) {
        out->push_back(*opt_codec);
      }
    } else {
      LOG(LS_ERROR) << "Unable to assign payload type to format: " << format;
    }

    return opt_codec;
  };

  for (const auto& spec : specs) {
    // We need to do some extra stuff before adding the main codecs to out.
    rtc::Optional<AudioCodec> opt_codec = map_format(spec.format, nullptr);
    if (opt_codec) {
      AudioCodec& codec = *opt_codec;
      if (spec.supports_network_adaption) {
        codec.AddFeedbackParam(
            FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
      }

      if (spec.allow_comfort_noise) {
        // Generate a CN entry if the decoder allows it and we support the
        // clockrate.
        auto cn = generate_cn.find(spec.format.clockrate_hz);
        if (cn != generate_cn.end()) {
          cn->second = true;
        }
      }

      // Generate a telephone-event entry if we support the clockrate.
      auto dtmf = generate_dtmf.find(spec.format.clockrate_hz);
      if (dtmf != generate_dtmf.end()) {
        dtmf->second = true;
      }

      out.push_back(codec);
    }
  }

  // Add CN codecs after "proper" audio codecs.
  for (const auto& cn : generate_cn) {
    if (cn.second) {
      map_format({kCnCodecName, cn.first, 1}, &out);
    }
  }

  // Add telephone-event codecs last.
  for (const auto& dtmf : generate_dtmf) {
    if (dtmf.second) {
      map_format({kDtmfCodecName, dtmf.first, 1}, &out);
    }
  }

  return out;
}

class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
    : public AudioSource::Sink {
 public:
  WebRtcAudioSendStream(
      int ch,
      webrtc::AudioTransport* voe_audio_transport,
      uint32_t ssrc,
      const std::string& c_name,
      const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec,
      const std::vector<webrtc::RtpExtension>& extensions,
      int max_send_bitrate_bps,
      const rtc::Optional<std::string>& audio_network_adaptor_config,
      webrtc::Call* call,
      webrtc::Transport* send_transport)
      : voe_audio_transport_(voe_audio_transport),
        call_(call),
        config_(send_transport),
        send_side_bwe_with_overhead_(
            webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
        max_send_bitrate_bps_(max_send_bitrate_bps),
        rtp_parameters_(CreateRtpParametersWithOneEncoding()) {
    RTC_DCHECK_GE(ch, 0);
    // TODO(solenberg): Once we're not using FakeWebRtcVoiceEngine anymore:
    // RTC_DCHECK(voe_audio_transport);
    RTC_DCHECK(call);
    config_.rtp.ssrc = ssrc;
    config_.rtp.c_name = c_name;
    config_.voe_channel_id = ch;
    config_.rtp.extensions = extensions;
    config_.audio_network_adaptor_config = audio_network_adaptor_config;
    rtp_parameters_.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
    RecreateAudioSendStream(send_codec_spec);
  }

  ~WebRtcAudioSendStream() override {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    ClearSource();
    call_->DestroyAudioSendStream(stream_);
  }

  void RecreateAudioSendStream(
      const webrtc::AudioSendStream::Config::SendCodecSpec& send_codec_spec) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    send_codec_spec_ = send_codec_spec;
    config_.rtp.nack.rtp_history_ms =
        send_codec_spec_.nack_enabled ? kNackRtpHistoryMs : 0;
    config_.send_codec_spec = send_codec_spec_;
    auto send_rate = ComputeSendBitrate(
        max_send_bitrate_bps_, rtp_parameters_.encodings[0].max_bitrate_bps,
        send_codec_spec.codec_inst);
    if (send_rate) {
      // Apply a send rate that abides by |max_send_bitrate_bps_| and
      // |rtp_parameters_| when possible. Otherwise use the codec rate.
      config_.send_codec_spec.codec_inst.rate = *send_rate;
    }
    RecreateAudioSendStream();
  }

  void RecreateAudioSendStream(
      const std::vector<webrtc::RtpExtension>& extensions) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    config_.rtp.extensions = extensions;
    RecreateAudioSendStream();
  }

  void RecreateAudioSendStream(
      const rtc::Optional<std::string>& audio_network_adaptor_config) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    if (config_.audio_network_adaptor_config == audio_network_adaptor_config) {
      return;
    }
    config_.audio_network_adaptor_config = audio_network_adaptor_config;
    RecreateAudioSendStream();
  }

  bool SetMaxSendBitrate(int bps) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    auto send_rate =
        ComputeSendBitrate(bps, rtp_parameters_.encodings[0].max_bitrate_bps,
                           send_codec_spec_.codec_inst);
    if (!send_rate) {
      return false;
    }

    max_send_bitrate_bps_ = bps;

    if (config_.send_codec_spec.codec_inst.rate != *send_rate) {
      // Recreate AudioSendStream with new bit rate.
      config_.send_codec_spec.codec_inst.rate = *send_rate;
      RecreateAudioSendStream();
    }
    return true;
  }

  bool SendTelephoneEvent(int payload_type, int payload_freq, int event,
                          int duration_ms) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    RTC_DCHECK(stream_);
    return stream_->SendTelephoneEvent(payload_type, payload_freq, event,
                                       duration_ms);
  }

  void SetSend(bool send) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    send_ = send;
    UpdateSendState();
  }

  void SetMuted(bool muted) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    RTC_DCHECK(stream_);
    stream_->SetMuted(muted);
    muted_ = muted;
  }

  bool muted() const {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    return muted_;
  }

  webrtc::AudioSendStream::Stats GetStats() const {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    RTC_DCHECK(stream_);
    return stream_->GetStats();
  }

  // Starts the sending by setting ourselves as a sink to the AudioSource to
  // get data callbacks.
  // This method is called on the libjingle worker thread.
  // TODO(xians): Make sure Start() is called only once.
  void SetSource(AudioSource* source) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    RTC_DCHECK(source);
    if (source_) {
      RTC_DCHECK(source_ == source);
      return;
    }
    source->SetSink(this);
    source_ = source;
    UpdateSendState();
  }

  // Stops sending by setting the sink of the AudioSource to nullptr. No data
  // callback will be received after this method.
  // This method is called on the libjingle worker thread.
  void ClearSource() {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    if (source_) {
      source_->SetSink(nullptr);
      source_ = nullptr;
    }
    UpdateSendState();
  }

  // AudioSource::Sink implementation.
  // This method is called on the audio thread.
  void OnData(const void* audio_data,
              int bits_per_sample,
              int sample_rate,
              size_t number_of_channels,
              size_t number_of_frames) override {
    RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
    RTC_DCHECK(voe_audio_transport_);
    voe_audio_transport_->PushCaptureData(config_.voe_channel_id, audio_data,
                                          bits_per_sample, sample_rate,
                                          number_of_channels, number_of_frames);
  }

  // Callback from the |source_| when it is going away. In case Start() has
  // never been called, this callback won't be triggered.
  void OnClose() override {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    // Set |source_| to nullptr to make sure no more callback will get into
    // the source.
    source_ = nullptr;
    UpdateSendState();
  }

  // Accessor to the VoE channel ID.
  int channel() const {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    return config_.voe_channel_id;
  }

  const webrtc::RtpParameters& rtp_parameters() const {
    return rtp_parameters_;
  }

  bool ValidateRtpParameters(const webrtc::RtpParameters& rtp_parameters) {
    if (rtp_parameters.encodings.size() != 1) {
      LOG(LS_ERROR)
          << "Attempted to set RtpParameters without exactly one encoding";
      return false;
    }
    if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
      LOG(LS_ERROR) << "Attempted to set RtpParameters with modified SSRC";
      return false;
    }
    return true;
  }

  bool SetRtpParameters(const webrtc::RtpParameters& parameters) {
    if (!ValidateRtpParameters(parameters)) {
      return false;
    }
    auto send_rate = ComputeSendBitrate(max_send_bitrate_bps_,
                                        parameters.encodings[0].max_bitrate_bps,
                                        send_codec_spec_.codec_inst);
    if (!send_rate) {
      return false;
    }

    rtp_parameters_ = parameters;

    // parameters.encodings[0].encodings[0].max_bitrate_bps could have changed.
    if (config_.send_codec_spec.codec_inst.rate != *send_rate) {
      // Recreate AudioSendStream with new bit rate.
      config_.send_codec_spec.codec_inst.rate = *send_rate;
      RecreateAudioSendStream();
    } else {
      // parameters.encodings[0].active could have changed.
      UpdateSendState();
    }
    return true;
  }

 private:
  void UpdateSendState() {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    RTC_DCHECK(stream_);
    RTC_DCHECK_EQ(1UL, rtp_parameters_.encodings.size());
    if (send_ && source_ != nullptr && rtp_parameters_.encodings[0].active) {
      stream_->Start();
    } else {  // !send || source_ = nullptr
      stream_->Stop();
    }
  }

  void RecreateAudioSendStream() {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    if (stream_) {
      call_->DestroyAudioSendStream(stream_);
      stream_ = nullptr;
    }
    RTC_DCHECK(!stream_);
    if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
      config_.min_bitrate_bps = kOpusMinBitrateBps;
      config_.max_bitrate_bps = kOpusBitrateFbBps;
      // TODO(mflodman): Keep testing this and set proper values.
      // Note: This is an early experiment currently only supported by Opus.
      if (send_side_bwe_with_overhead_) {
        auto packet_sizes_ms = WebRtcVoiceCodecs::GetPacketSizesMs(
            config_.send_codec_spec.codec_inst);
        if (!packet_sizes_ms.empty()) {
          int max_packet_size_ms =
              *std::max_element(packet_sizes_ms.begin(), packet_sizes_ms.end());
          int min_packet_size_ms =
              *std::min_element(packet_sizes_ms.begin(), packet_sizes_ms.end());

          // Audio network adaptor will just use 20ms and 60ms frame lengths.
          // The adaptor will only be active for the Opus encoder.
          if (config_.audio_network_adaptor_config &&
              IsCodec(config_.send_codec_spec.codec_inst, kOpusCodecName)) {
#if WEBRTC_OPUS_SUPPORT_120MS_PTIME
            max_packet_size_ms = 120;
#else
            max_packet_size_ms = 60;
#endif
            min_packet_size_ms = 20;
          }

          // OverheadPerPacket = Ipv4(20B) + UDP(8B) + SRTP(10B) + RTP(12)
          constexpr int kOverheadPerPacket = 20 + 8 + 10 + 12;

          int min_overhead_bps =
              kOverheadPerPacket * 8 * 1000 / max_packet_size_ms;

          int max_overhead_bps =
              kOverheadPerPacket * 8 * 1000 / min_packet_size_ms;

          config_.min_bitrate_bps = kOpusMinBitrateBps + min_overhead_bps;
          config_.max_bitrate_bps = kOpusBitrateFbBps + max_overhead_bps;
        }
      }
    }
    stream_ = call_->CreateAudioSendStream(config_);
    RTC_CHECK(stream_);
    UpdateSendState();
  }

  rtc::ThreadChecker worker_thread_checker_;
  rtc::RaceChecker audio_capture_race_checker_;
  webrtc::AudioTransport* const voe_audio_transport_ = nullptr;
  webrtc::Call* call_ = nullptr;
  webrtc::AudioSendStream::Config config_;
  const bool send_side_bwe_with_overhead_;
  // The stream is owned by WebRtcAudioSendStream and may be reallocated if
  // configuration changes.
  webrtc::AudioSendStream* stream_ = nullptr;

  // Raw pointer to AudioSource owned by LocalAudioTrackHandler.
  // PeerConnection will make sure invalidating the pointer before the object
  // goes away.
  AudioSource* source_ = nullptr;
  bool send_ = false;
  bool muted_ = false;
  int max_send_bitrate_bps_;
  webrtc::RtpParameters rtp_parameters_;
  webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_;

  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
};

class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
 public:
  WebRtcAudioReceiveStream(
      int ch,
      uint32_t remote_ssrc,
      uint32_t local_ssrc,
      bool use_transport_cc,
      bool use_nack,
      const std::string& sync_group,
      const std::vector<webrtc::RtpExtension>& extensions,
      webrtc::Call* call,
      webrtc::Transport* rtcp_send_transport,
      const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
      const std::map<int, webrtc::SdpAudioFormat>& decoder_map)
      : call_(call), config_() {
    RTC_DCHECK_GE(ch, 0);
    RTC_DCHECK(call);
    config_.rtp.remote_ssrc = remote_ssrc;
    config_.rtp.local_ssrc = local_ssrc;
    config_.rtp.transport_cc = use_transport_cc;
    config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
    config_.rtp.extensions = extensions;
    config_.rtcp_send_transport = rtcp_send_transport;
    config_.voe_channel_id = ch;
    config_.sync_group = sync_group;
    config_.decoder_factory = decoder_factory;
    config_.decoder_map = decoder_map;
    RecreateAudioReceiveStream();
  }

  ~WebRtcAudioReceiveStream() {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    call_->DestroyAudioReceiveStream(stream_);
  }

  void RecreateAudioReceiveStream(uint32_t local_ssrc) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    config_.rtp.local_ssrc = local_ssrc;
    RecreateAudioReceiveStream();
  }

  void RecreateAudioReceiveStream(bool use_transport_cc, bool use_nack) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    config_.rtp.transport_cc = use_transport_cc;
    config_.rtp.nack.rtp_history_ms = use_nack ? kNackRtpHistoryMs : 0;
    RecreateAudioReceiveStream();
  }

  void RecreateAudioReceiveStream(
      const std::vector<webrtc::RtpExtension>& extensions) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    config_.rtp.extensions = extensions;
    RecreateAudioReceiveStream();
  }

  // Set a new payload type -> decoder map. The new map must be a superset of
  // the old one.
  void RecreateAudioReceiveStream(
      const std::map<int, webrtc::SdpAudioFormat>& decoder_map) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    RTC_DCHECK([&] {
      for (const auto& item : config_.decoder_map) {
        auto it = decoder_map.find(item.first);
        if (it == decoder_map.end() || *it != item) {
          return false;  // The old map isn't a subset of the new map.
        }
      }
      return true;
    }());
    config_.decoder_map = decoder_map;
    RecreateAudioReceiveStream();
  }

  void MaybeRecreateAudioReceiveStream(const std::string& sync_group) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    if (config_.sync_group != sync_group) {
      config_.sync_group = sync_group;
      RecreateAudioReceiveStream();
    }
  }

  webrtc::AudioReceiveStream::Stats GetStats() const {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    RTC_DCHECK(stream_);
    return stream_->GetStats();
  }

  int GetOutputLevel() const {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    RTC_DCHECK(stream_);
    return stream_->GetOutputLevel();
  }

  int channel() const {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    return config_.voe_channel_id;
  }

  void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    stream_->SetSink(std::move(sink));
  }

  void SetOutputVolume(double volume) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    stream_->SetGain(volume);
  }

  void SetPlayout(bool playout) {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    RTC_DCHECK(stream_);
    if (playout) {
      LOG(LS_INFO) << "Starting playout for channel #" << channel();
      stream_->Start();
    } else {
      LOG(LS_INFO) << "Stopping playout for channel #" << channel();
      stream_->Stop();
    }
    playout_ = playout;
  }

 private:
  void RecreateAudioReceiveStream() {
    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
    if (stream_) {
      call_->DestroyAudioReceiveStream(stream_);
    }
    stream_ = call_->CreateAudioReceiveStream(config_);
    RTC_CHECK(stream_);
    SetPlayout(playout_);
  }

  rtc::ThreadChecker worker_thread_checker_;
  webrtc::Call* call_ = nullptr;
  webrtc::AudioReceiveStream::Config config_;
  // The stream is owned by WebRtcAudioReceiveStream and may be reallocated if
  // configuration changes.
  webrtc::AudioReceiveStream* stream_ = nullptr;
  bool playout_ = false;

  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
};

WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
                                                 const MediaConfig& config,
                                                 const AudioOptions& options,
                                                 webrtc::Call* call)
    : VoiceMediaChannel(config), engine_(engine), call_(call) {
  LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
  RTC_DCHECK(call);
  engine->RegisterChannel(this);
  SetOptions(options);
}

WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel";
  // TODO(solenberg): Should be able to delete the streams directly, without
  //                  going through RemoveNnStream(), once stream objects handle
  //                  all (de)configuration.
  while (!send_streams_.empty()) {
    RemoveSendStream(send_streams_.begin()->first);
  }
  while (!recv_streams_.empty()) {
    RemoveRecvStream(recv_streams_.begin()->first);
  }
  engine()->UnregisterChannel(this);
}

rtc::DiffServCodePoint WebRtcVoiceMediaChannel::PreferredDscp() const {
  return kAudioDscpValue;
}

bool WebRtcVoiceMediaChannel::SetSendParameters(
    const AudioSendParameters& params) {
  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSendParameters");
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendParameters: "
               << params.ToString();
  // TODO(pthatcher): Refactor this to be more clean now that we have
  // all the information at once.

  if (!SetSendCodecs(params.codecs)) {
    return false;
  }

  if (!ValidateRtpExtensions(params.extensions)) {
    return false;
  }
  std::vector<webrtc::RtpExtension> filtered_extensions =
      FilterRtpExtensions(params.extensions,
                          webrtc::RtpExtension::IsSupportedForAudio, true);
  if (send_rtp_extensions_ != filtered_extensions) {
    send_rtp_extensions_.swap(filtered_extensions);
    for (auto& it : send_streams_) {
      it.second->RecreateAudioSendStream(send_rtp_extensions_);
    }
  }

  if (!SetMaxSendBitrate(params.max_bandwidth_bps)) {
    return false;
  }
  return SetOptions(params.options);
}

bool WebRtcVoiceMediaChannel::SetRecvParameters(
    const AudioRecvParameters& params) {
  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetRecvParameters");
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetRecvParameters: "
               << params.ToString();
  // TODO(pthatcher): Refactor this to be more clean now that we have
  // all the information at once.

  if (!SetRecvCodecs(params.codecs)) {
    return false;
  }

  if (!ValidateRtpExtensions(params.extensions)) {
    return false;
  }
  std::vector<webrtc::RtpExtension> filtered_extensions =
      FilterRtpExtensions(params.extensions,
                          webrtc::RtpExtension::IsSupportedForAudio, false);
  if (recv_rtp_extensions_ != filtered_extensions) {
    recv_rtp_extensions_.swap(filtered_extensions);
    for (auto& it : recv_streams_) {
      it.second->RecreateAudioReceiveStream(recv_rtp_extensions_);
    }
  }
  return true;
}

webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpSendParameters(
    uint32_t ssrc) const {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  auto it = send_streams_.find(ssrc);
  if (it == send_streams_.end()) {
    LOG(LS_WARNING) << "Attempting to get RTP send parameters for stream "
                    << "with ssrc " << ssrc << " which doesn't exist.";
    return webrtc::RtpParameters();
  }

  webrtc::RtpParameters rtp_params = it->second->rtp_parameters();
  // Need to add the common list of codecs to the send stream-specific
  // RTP parameters.
  for (const AudioCodec& codec : send_codecs_) {
    rtp_params.codecs.push_back(codec.ToCodecParameters());
  }
  return rtp_params;
}

bool WebRtcVoiceMediaChannel::SetRtpSendParameters(
    uint32_t ssrc,
    const webrtc::RtpParameters& parameters) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  auto it = send_streams_.find(ssrc);
  if (it == send_streams_.end()) {
    LOG(LS_WARNING) << "Attempting to set RTP send parameters for stream "
                    << "with ssrc " << ssrc << " which doesn't exist.";
    return false;
  }

  // TODO(deadbeef): Handle setting parameters with a list of codecs in a
  // different order (which should change the send codec).
  webrtc::RtpParameters current_parameters = GetRtpSendParameters(ssrc);
  if (current_parameters.codecs != parameters.codecs) {
    LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
                  << "is not currently supported.";
    return false;
  }

  // TODO(minyue): The following legacy actions go into
  // |WebRtcAudioSendStream::SetRtpParameters()| which is called at the end,
  // though there are two difference:
  // 1. |WebRtcVoiceMediaChannel::SetChannelSendParameters()| only calls
  // |SetSendCodec| while |WebRtcAudioSendStream::SetRtpParameters()| calls
  // |SetSendCodecs|. The outcome should be the same.
  // 2. AudioSendStream can be recreated.

  // Codecs are handled at the WebRtcVoiceMediaChannel level.
  webrtc::RtpParameters reduced_params = parameters;
  reduced_params.codecs.clear();
  return it->second->SetRtpParameters(reduced_params);
}

webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
    uint32_t ssrc) const {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  auto it = recv_streams_.find(ssrc);
  if (it == recv_streams_.end()) {
    LOG(LS_WARNING) << "Attempting to get RTP receive parameters for stream "
                    << "with ssrc " << ssrc << " which doesn't exist.";
    return webrtc::RtpParameters();
  }

  // TODO(deadbeef): Return stream-specific parameters.
  webrtc::RtpParameters rtp_params = CreateRtpParametersWithOneEncoding();
  for (const AudioCodec& codec : recv_codecs_) {
    rtp_params.codecs.push_back(codec.ToCodecParameters());
  }
  rtp_params.encodings[0].ssrc = rtc::Optional<uint32_t>(ssrc);
  return rtp_params;
}

bool WebRtcVoiceMediaChannel::SetRtpReceiveParameters(
    uint32_t ssrc,
    const webrtc::RtpParameters& parameters) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  auto it = recv_streams_.find(ssrc);
  if (it == recv_streams_.end()) {
    LOG(LS_WARNING) << "Attempting to set RTP receive parameters for stream "
                    << "with ssrc " << ssrc << " which doesn't exist.";
    return false;
  }

  webrtc::RtpParameters current_parameters = GetRtpReceiveParameters(ssrc);
  if (current_parameters != parameters) {
    LOG(LS_ERROR) << "Changing the RTP receive parameters is currently "
                  << "unsupported.";
    return false;
  }
  return true;
}

bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "Setting voice channel options: "
               << options.ToString();

  // We retain all of the existing options, and apply the given ones
  // on top.  This means there is no way to "clear" options such that
  // they go back to the engine default.
  options_.SetAll(options);
  if (!engine()->ApplyOptions(options_)) {
    LOG(LS_WARNING) <<
        "Failed to apply engine options during channel SetOptions.";
    return false;
  }

  rtc::Optional<std::string> audio_network_adatptor_config =
      GetAudioNetworkAdaptorConfig(options_);
  for (auto& it : send_streams_) {
    it.second->RecreateAudioSendStream(audio_network_adatptor_config);
  }

  LOG(LS_INFO) << "Set voice channel options. Current options: "
               << options_.ToString();
  return true;
}

bool WebRtcVoiceMediaChannel::SetRecvCodecs(
    const std::vector<AudioCodec>& codecs) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());

  // Set the payload types to be used for incoming media.
  LOG(LS_INFO) << "Setting receive voice codecs.";

  if (!VerifyUniquePayloadTypes(codecs)) {
    LOG(LS_ERROR) << "Codec payload types overlap.";
    return false;
  }

  std::vector<AudioCodec> new_codecs;
  // Find all new codecs. We allow adding new codecs but don't allow changing
  // the payload type of codecs that is already configured since we might
  // already be receiving packets with that payload type.
  for (const AudioCodec& codec : codecs) {
    AudioCodec old_codec;
    // TODO(solenberg): This isn't strictly correct. It should be possible to
    // add an additional payload type for a codec. That would result in a new
    // decoder object being allocated. What shouldn't work is to remove a PT
    // mapping that was previously configured.
    if (FindCodec(recv_codecs_, codec, &old_codec)) {
      if (old_codec.id != codec.id) {
        LOG(LS_ERROR) << codec.name << " payload type changed.";
        return false;
      }
    } else {
      new_codecs.push_back(codec);
    }
  }
  if (new_codecs.empty()) {
    // There are no new codecs to configure. Already configured codecs are
    // never removed.
    return true;
  }

  // Create a payload type -> SdpAudioFormat map with all the decoders. Fail
  // unless the factory claims to support all decoders.
  std::map<int, webrtc::SdpAudioFormat> decoder_map;
  for (const AudioCodec& codec : codecs) {
    auto format = AudioCodecToSdpAudioFormat(codec);
    if (!IsCodec(codec, "cn") && !IsCodec(codec, "telephone-event") &&
        !engine()->decoder_factory_->IsSupportedDecoder(format)) {
      LOG(LS_ERROR) << "Unsupported codec: " << format;
      return false;
    }
    decoder_map.insert({codec.id, std::move(format)});
  }

  if (playout_) {
    // Receive codecs can not be changed while playing. So we temporarily
    // pause playout.
    ChangePlayout(false);
  }

  decoder_map_ = std::move(decoder_map);
  for (auto& kv : recv_streams_) {
    kv.second->RecreateAudioReceiveStream(decoder_map_);
  }
  recv_codecs_ = codecs;

  if (desired_playout_ && !playout_) {
    ChangePlayout(desired_playout_);
  }
  return true;
}

// Utility function called from SetSendParameters() to extract current send
// codec settings from the given list of codecs (originally from SDP). Both send
// and receive streams may be reconfigured based on the new settings.
bool WebRtcVoiceMediaChannel::SetSendCodecs(
    const std::vector<AudioCodec>& codecs) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  dtmf_payload_type_ = rtc::Optional<int>();
  dtmf_payload_freq_ = -1;

  // Validate supplied codecs list.
  for (const AudioCodec& codec : codecs) {
    // TODO(solenberg): Validate more aspects of input - that payload types
    //                  don't overlap, remove redundant/unsupported codecs etc -
    //                  the same way it is done for RtpHeaderExtensions.
    if (codec.id < kMinPayloadType || codec.id > kMaxPayloadType) {
      LOG(LS_WARNING) << "Codec payload type out of range: " << ToString(codec);
      return false;
    }
  }

  // Find PT of telephone-event codec with lowest clockrate, as a fallback, in
  // case we don't have a DTMF codec with a rate matching the send codec's, or
  // if this function returns early.
  std::vector<AudioCodec> dtmf_codecs;
  for (const AudioCodec& codec : codecs) {
    if (IsCodec(codec, kDtmfCodecName)) {
      dtmf_codecs.push_back(codec);
      if (!dtmf_payload_type_ || codec.clockrate < dtmf_payload_freq_) {
        dtmf_payload_type_ = rtc::Optional<int>(codec.id);
        dtmf_payload_freq_ = codec.clockrate;
      }
    }
  }

  // Scan through the list to figure out the codec to use for sending, along
  // with the proper configuration for VAD, CNG, NACK and Opus-specific
  // parameters.
  // TODO(solenberg): Refactor this logic once we create AudioEncoders here.
  webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec;
  webrtc::Call::Config::BitrateConfig bitrate_config;
  {
    send_codec_spec.nack_enabled = send_codec_spec_.nack_enabled;

    // Find send codec (the first non-telephone-event/CN codec).
    const AudioCodec* codec = WebRtcVoiceCodecs::GetPreferredCodec(
        codecs, &send_codec_spec.codec_inst);
    if (!codec) {
      LOG(LS_WARNING) << "Received empty list of codecs.";
      return false;
    }

    send_codec_spec.transport_cc_enabled = HasTransportCc(*codec);
    send_codec_spec.nack_enabled = HasNack(*codec);
    bitrate_config = GetBitrateConfigForCodec(*codec);

    // For Opus as the send codec, we are to determine inband FEC, maximum
    // playback rate, and opus internal dtx.
    if (IsCodec(*codec, kOpusCodecName)) {
      GetOpusConfig(*codec, &send_codec_spec.codec_inst,
                    &send_codec_spec.enable_codec_fec,
                    &send_codec_spec.opus_max_playback_rate,
                    &send_codec_spec.enable_opus_dtx,
                    &send_codec_spec.min_ptime_ms,
                    &send_codec_spec.max_ptime_ms);
    }

    // Set packet size if the AudioCodec param kCodecParamPTime is set.
    int ptime_ms = 0;
    if (codec->GetParam(kCodecParamPTime, &ptime_ms)) {
      if (!WebRtcVoiceCodecs::SetPTimeAsPacketSize(
          &send_codec_spec.codec_inst, ptime_ms)) {
        LOG(LS_WARNING) << "Failed to set packet size for codec "
                        << send_codec_spec.codec_inst.plname;
        return false;
      }
    }

    // Loop through the codecs list again to find the CN codec.
    // TODO(solenberg): Break out into a separate function?
    for (const AudioCodec& cn_codec : codecs) {
      // Ignore codecs we don't know about. The negotiation step should prevent
      // this, but double-check to be sure.
      webrtc::CodecInst voe_codec = {0};
      if (!WebRtcVoiceEngine::ToCodecInst(cn_codec, &voe_codec)) {
        LOG(LS_WARNING) << "Unknown codec " << ToString(cn_codec);
        continue;
      }

      if (IsCodec(cn_codec, kCnCodecName) &&
          cn_codec.clockrate == codec->clockrate) {
        // Turn voice activity detection/comfort noise on if supported.
        // Set the wideband CN payload type appropriately.
        // (narrowband always uses the static payload type 13).
        int cng_plfreq = -1;
        switch (cn_codec.clockrate) {
          case 8000:
          case 16000:
          case 32000:
            cng_plfreq = cn_codec.clockrate;
            break;
          default:
            LOG(LS_WARNING) << "CN frequency " << cn_codec.clockrate
                            << " not supported.";
            continue;
        }
        send_codec_spec.cng_payload_type = cn_codec.id;
        send_codec_spec.cng_plfreq = cng_plfreq;
        break;
      }
    }

    // Find the telephone-event PT exactly matching the preferred send codec.
    for (const AudioCodec& dtmf_codec : dtmf_codecs) {
      if (dtmf_codec.clockrate == codec->clockrate) {
        dtmf_payload_type_ = rtc::Optional<int>(dtmf_codec.id);
        dtmf_payload_freq_ = dtmf_codec.clockrate;
        break;
      }
    }
  }

  if (send_codec_spec_ != send_codec_spec) {
    send_codec_spec_ = std::move(send_codec_spec);
    // Apply new settings to all streams.
    for (const auto& kv : send_streams_) {
      kv.second->RecreateAudioSendStream(send_codec_spec_);
    }
  } else {
    // If the codec isn't changing, set the start bitrate to -1 which means
    // "unchanged" so that BWE isn't affected.
    bitrate_config.start_bitrate_bps = -1;
  }
  call_->SetBitrateConfig(bitrate_config);

  // Check if the transport cc feedback or NACK status has changed on the
  // preferred send codec, and in that case reconfigure all receive streams.
  if (recv_transport_cc_enabled_ != send_codec_spec_.transport_cc_enabled ||
      recv_nack_enabled_ != send_codec_spec_.nack_enabled) {
    LOG(LS_INFO) << "Recreate all the receive streams because the send "
                    "codec has changed.";
    recv_transport_cc_enabled_ = send_codec_spec_.transport_cc_enabled;
    recv_nack_enabled_ = send_codec_spec_.nack_enabled;
    for (auto& kv : recv_streams_) {
      kv.second->RecreateAudioReceiveStream(recv_transport_cc_enabled_,
                                            recv_nack_enabled_);
    }
  }

  send_codecs_ = codecs;
  return true;
}

void WebRtcVoiceMediaChannel::SetPlayout(bool playout) {
  desired_playout_ = playout;
  return ChangePlayout(desired_playout_);
}

void WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::ChangePlayout");
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  if (playout_ == playout) {
    return;
  }

  for (const auto& kv : recv_streams_) {
    kv.second->SetPlayout(playout);
  }
  playout_ = playout;
}

void WebRtcVoiceMediaChannel::SetSend(bool send) {
  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSend");
  if (send_ == send) {
    return;
  }

  // Apply channel specific options, and initialize the ADM for recording (this
  // may take time on some platforms, e.g. Android).
  if (send) {
    engine()->ApplyOptions(options_);

    // InitRecording() may return an error if the ADM is already recording.
    if (!engine()->adm()->RecordingIsInitialized() &&
        !engine()->adm()->Recording()) {
      if (engine()->adm()->InitRecording() != 0) {
        LOG(LS_WARNING) << "Failed to initialize recording";
      }
    }
  }

  // Change the settings on each send channel.
  for (auto& kv : send_streams_) {
    kv.second->SetSend(send);
  }

  send_ = send;
}

bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc,
                                           bool enable,
                                           const AudioOptions* options,
                                           AudioSource* source) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  // TODO(solenberg): The state change should be fully rolled back if any one of
  //                  these calls fail.
  if (!SetLocalSource(ssrc, source)) {
    return false;
  }
  if (!MuteStream(ssrc, !enable)) {
    return false;
  }
  if (enable && options) {
    return SetOptions(*options);
  }
  return true;
}

int WebRtcVoiceMediaChannel::CreateVoEChannel() {
  int id = engine()->CreateVoEChannel();
  if (id == -1) {
    LOG_RTCERR0(CreateVoEChannel);
    return -1;
  }

  return id;
}

bool WebRtcVoiceMediaChannel::DeleteVoEChannel(int channel) {
  if (engine()->voe()->base()->DeleteChannel(channel) == -1) {
    LOG_RTCERR1(DeleteChannel, channel);
    return false;
  }
  return true;
}

bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddSendStream");
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "AddSendStream: " << sp.ToString();

  uint32_t ssrc = sp.first_ssrc();
  RTC_DCHECK(0 != ssrc);

  if (GetSendChannelId(ssrc) != -1) {
    LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
    return false;
  }

  // Create a new channel for sending audio data.
  int channel = CreateVoEChannel();
  if (channel == -1) {
    return false;
  }

  // Save the channel to send_streams_, so that RemoveSendStream() can still
  // delete the channel in case failure happens below.
  webrtc::AudioTransport* audio_transport =
      engine()->voe()->base()->audio_transport();

  rtc::Optional<std::string> audio_network_adaptor_config =
      GetAudioNetworkAdaptorConfig(options_);
  WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
      channel, audio_transport, ssrc, sp.cname, send_codec_spec_,
      send_rtp_extensions_, max_send_bitrate_bps_, audio_network_adaptor_config,
      call_, this);
  send_streams_.insert(std::make_pair(ssrc, stream));

  // At this point the stream's local SSRC has been updated. If it is the first
  // send stream, make sure that all the receive streams are updated with the
  // same SSRC in order to send receiver reports.
  if (send_streams_.size() == 1) {
    receiver_reports_ssrc_ = ssrc;
    for (const auto& kv : recv_streams_) {
      // TODO(solenberg): Allow applications to set the RTCP SSRC of receive
      // streams instead, so we can avoid recreating the streams here.
      kv.second->RecreateAudioReceiveStream(ssrc);
    }
  }

  send_streams_[ssrc]->SetSend(send_);
  return true;
}

bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveSendStream");
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "RemoveSendStream: " << ssrc;

  auto it = send_streams_.find(ssrc);
  if (it == send_streams_.end()) {
    LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
                    << " which doesn't exist.";
    return false;
  }

  it->second->SetSend(false);

  // TODO(solenberg): If we're removing the receiver_reports_ssrc_ stream, find
  // the first active send stream and use that instead, reassociating receive
  // streams.

  // Clean up and delete the send stream+channel.
  int channel = it->second->channel();
  LOG(LS_INFO) << "Removing audio send stream " << ssrc
               << " with VoiceEngine channel #" << channel << ".";
  delete it->second;
  send_streams_.erase(it);
  if (!DeleteVoEChannel(channel)) {
    return false;
  }
  if (send_streams_.empty()) {
    SetSend(false);
  }
  return true;
}

bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddRecvStream");
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();

  if (!ValidateStreamParams(sp)) {
    return false;
  }

  const uint32_t ssrc = sp.first_ssrc();
  if (ssrc == 0) {
    LOG(LS_WARNING) << "AddRecvStream with ssrc==0 is not supported.";
    return false;
  }

  // If this stream was previously received unsignaled, we promote it, possibly
  // recreating the AudioReceiveStream, if sync_label has changed.
  if (MaybeDeregisterUnsignaledRecvStream(ssrc)) {
    recv_streams_[ssrc]->MaybeRecreateAudioReceiveStream(sp.sync_label);
    return true;
  }

  if (GetReceiveChannelId(ssrc) != -1) {
    LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
    return false;
  }

  // Create a new channel for receiving audio data.
  const int channel = CreateVoEChannel();
  if (channel == -1) {
    return false;
  }

  recv_streams_.insert(std::make_pair(
      ssrc,
      new WebRtcAudioReceiveStream(
          channel, ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
          recv_nack_enabled_, sp.sync_label, recv_rtp_extensions_, call_, this,
          engine()->decoder_factory_, decoder_map_)));
  recv_streams_[ssrc]->SetPlayout(playout_);

  return true;
}

bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveRecvStream");
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;

  const auto it = recv_streams_.find(ssrc);
  if (it == recv_streams_.end()) {
    LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
                    << " which doesn't exist.";
    return false;
  }

  MaybeDeregisterUnsignaledRecvStream(ssrc);

  const int channel = it->second->channel();

  // Clean up and delete the receive stream+channel.
  LOG(LS_INFO) << "Removing audio receive stream " << ssrc
               << " with VoiceEngine channel #" << channel << ".";
  it->second->SetRawAudioSink(nullptr);
  delete it->second;
  recv_streams_.erase(it);
  return DeleteVoEChannel(channel);
}

bool WebRtcVoiceMediaChannel::SetLocalSource(uint32_t ssrc,
                                             AudioSource* source) {
  auto it = send_streams_.find(ssrc);
  if (it == send_streams_.end()) {
    if (source) {
      // Return an error if trying to set a valid source with an invalid ssrc.
      LOG(LS_ERROR) << "SetLocalSource failed with ssrc " << ssrc;
      return false;
    }

    // The channel likely has gone away, do nothing.
    return true;
  }

  if (source) {
    it->second->SetSource(source);
  } else {
    it->second->ClearSource();
  }

  return true;
}

// TODO(solenberg): Remove, once AudioMonitor is gone.
bool WebRtcVoiceMediaChannel::GetActiveStreams(
    AudioInfo::StreamList* actives) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  actives->clear();
  for (const auto& ch : recv_streams_) {
    int level = ch.second->GetOutputLevel();
    if (level > 0) {
      actives->push_back(std::make_pair(ch.first, level));
    }
  }
  return true;
}

// TODO(solenberg): Remove, once AudioMonitor is gone.
int WebRtcVoiceMediaChannel::GetOutputLevel() {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  int highest = 0;
  for (const auto& ch : recv_streams_) {
    highest = std::max(ch.second->GetOutputLevel(), highest);
  }
  return highest;
}

bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  std::vector<uint32_t> ssrcs(1, ssrc);
  if (ssrc == 0) {
    default_recv_volume_ = volume;
    ssrcs = unsignaled_recv_ssrcs_;
  }
  for (uint32_t ssrc : ssrcs) {
    const auto it = recv_streams_.find(ssrc);
    if (it == recv_streams_.end()) {
      LOG(LS_WARNING) << "SetOutputVolume: no recv stream " << ssrc;
      return false;
    }
    it->second->SetOutputVolume(volume);
    LOG(LS_INFO) << "SetOutputVolume() to " << volume
                 << " for recv stream with ssrc " << ssrc;
  }
  return true;
}

bool WebRtcVoiceMediaChannel::CanInsertDtmf() {
  return dtmf_payload_type_ ? true : false;
}

bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, int event,
                                         int duration) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "WebRtcVoiceMediaChannel::InsertDtmf";
  if (!dtmf_payload_type_) {
    return false;
  }

  // Figure out which WebRtcAudioSendStream to send the event on.
  auto it = ssrc != 0 ? send_streams_.find(ssrc) : send_streams_.begin();
  if (it == send_streams_.end()) {
    LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
    return false;
  }
  if (event < kMinTelephoneEventCode ||
      event > kMaxTelephoneEventCode) {
    LOG(LS_WARNING) << "DTMF event code " << event << " out of range.";
    return false;
  }
  if (duration < kMinTelephoneEventDuration ||
      duration > kMaxTelephoneEventDuration) {
    LOG(LS_WARNING) << "DTMF event duration " << duration << " out of range.";
    return false;
  }
  RTC_DCHECK_NE(-1, dtmf_payload_freq_);
  return it->second->SendTelephoneEvent(*dtmf_payload_type_, dtmf_payload_freq_,
                                        event, duration);
}

void WebRtcVoiceMediaChannel::OnPacketReceived(
    rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());

  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                              packet_time.not_before);
  webrtc::PacketReceiver::DeliveryStatus delivery_result =
      call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
                                       packet->cdata(), packet->size(),
                                       webrtc_packet_time);
  if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
    return;
  }

  // Create an unsignaled receive stream for this previously not received ssrc.
  // If there already is N unsignaled receive streams, delete the oldest.
  // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5208
  uint32_t ssrc = 0;
  if (!GetRtpSsrc(packet->cdata(), packet->size(), &ssrc)) {
    return;
  }
  RTC_DCHECK(std::find(unsignaled_recv_ssrcs_.begin(),
      unsignaled_recv_ssrcs_.end(), ssrc) == unsignaled_recv_ssrcs_.end());

  // Add new stream.
  StreamParams sp;
  sp.ssrcs.push_back(ssrc);
  LOG(LS_INFO) << "Creating unsignaled receive stream for SSRC=" << ssrc;
  if (!AddRecvStream(sp)) {
    LOG(LS_WARNING) << "Could not create unsignaled receive stream.";
    return;
  }
  unsignaled_recv_ssrcs_.push_back(ssrc);
  RTC_HISTOGRAM_COUNTS_LINEAR(
      "WebRTC.Audio.NumOfUnsignaledStreams", unsignaled_recv_ssrcs_.size(), 1,
      100, 101);

  // Remove oldest unsignaled stream, if we have too many.
  if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) {
    uint32_t remove_ssrc = unsignaled_recv_ssrcs_.front();
    LOG(LS_INFO) << "Removing unsignaled receive stream with SSRC="
                 << remove_ssrc;
    RemoveRecvStream(remove_ssrc);
  }
  RTC_DCHECK_GE(kMaxUnsignaledRecvStreams, unsignaled_recv_ssrcs_.size());

  SetOutputVolume(ssrc, default_recv_volume_);

  // The default sink can only be attached to one stream at a time, so we hook
  // it up to the *latest* unsignaled stream we've seen, in order to support the
  // case where the SSRC of one unsignaled stream changes.
  if (default_sink_) {
    for (uint32_t drop_ssrc : unsignaled_recv_ssrcs_) {
      auto it = recv_streams_.find(drop_ssrc);
      it->second->SetRawAudioSink(nullptr);
    }
    std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
        new ProxySink(default_sink_.get()));
    SetRawAudioSink(ssrc, std::move(proxy_sink));
  }

  delivery_result = call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
                                                     packet->cdata(),
                                                     packet->size(),
                                                     webrtc_packet_time);
  RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result);
}

void WebRtcVoiceMediaChannel::OnRtcpReceived(
    rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());

  // Forward packet to Call as well.
  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                              packet_time.not_before);
  call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
      packet->cdata(), packet->size(), webrtc_packet_time);
}

void WebRtcVoiceMediaChannel::OnNetworkRouteChanged(
    const std::string& transport_name,
    const rtc::NetworkRoute& network_route) {
  call_->OnNetworkRouteChanged(transport_name, network_route);
}

bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  const auto it = send_streams_.find(ssrc);
  if (it == send_streams_.end()) {
    LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
    return false;
  }
  it->second->SetMuted(muted);

  // TODO(solenberg):
  // We set the AGC to mute state only when all the channels are muted.
  // This implementation is not ideal, instead we should signal the AGC when
  // the mic channel is muted/unmuted. We can't do it today because there
  // is no good way to know which stream is mapping to the mic channel.
  bool all_muted = muted;
  for (const auto& kv : send_streams_) {
    all_muted = all_muted && kv.second->muted();
  }
  engine()->apm()->set_output_will_be_muted(all_muted);

  return true;
}

bool WebRtcVoiceMediaChannel::SetMaxSendBitrate(int bps) {
  LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBitrate.";
  max_send_bitrate_bps_ = bps;
  bool success = true;
  for (const auto& kv : send_streams_) {
    if (!kv.second->SetMaxSendBitrate(max_send_bitrate_bps_)) {
      success = false;
    }
  }
  return success;
}

void WebRtcVoiceMediaChannel::OnReadyToSend(bool ready) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
  call_->SignalChannelNetworkState(
      webrtc::MediaType::AUDIO,
      ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
}

void WebRtcVoiceMediaChannel::OnTransportOverheadChanged(
    int transport_overhead_per_packet) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO,
                                    transport_overhead_per_packet);
}

bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::GetStats");
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  RTC_DCHECK(info);

  // Get SSRC and stats for each sender.
  RTC_DCHECK_EQ(info->senders.size(), 0U);
  for (const auto& stream : send_streams_) {
    webrtc::AudioSendStream::Stats stats = stream.second->GetStats();
    VoiceSenderInfo sinfo;
    sinfo.add_ssrc(stats.local_ssrc);
    sinfo.bytes_sent = stats.bytes_sent;
    sinfo.packets_sent = stats.packets_sent;
    sinfo.packets_lost = stats.packets_lost;
    sinfo.fraction_lost = stats.fraction_lost;
    sinfo.codec_name = stats.codec_name;
    sinfo.codec_payload_type = stats.codec_payload_type;
    sinfo.ext_seqnum = stats.ext_seqnum;
    sinfo.jitter_ms = stats.jitter_ms;
    sinfo.rtt_ms = stats.rtt_ms;
    sinfo.audio_level = stats.audio_level;
    sinfo.aec_quality_min = stats.aec_quality_min;
    sinfo.echo_delay_median_ms = stats.echo_delay_median_ms;
    sinfo.echo_delay_std_ms = stats.echo_delay_std_ms;
    sinfo.echo_return_loss = stats.echo_return_loss;
    sinfo.echo_return_loss_enhancement = stats.echo_return_loss_enhancement;
    sinfo.residual_echo_likelihood = stats.residual_echo_likelihood;
    sinfo.residual_echo_likelihood_recent_max =
        stats.residual_echo_likelihood_recent_max;
    sinfo.typing_noise_detected = (send_ ? stats.typing_noise_detected : false);
    info->senders.push_back(sinfo);
  }

  // Get SSRC and stats for each receiver.
  RTC_DCHECK_EQ(info->receivers.size(), 0U);
  for (const auto& stream : recv_streams_) {
    webrtc::AudioReceiveStream::Stats stats = stream.second->GetStats();
    VoiceReceiverInfo rinfo;
    rinfo.add_ssrc(stats.remote_ssrc);
    rinfo.bytes_rcvd = stats.bytes_rcvd;
    rinfo.packets_rcvd = stats.packets_rcvd;
    rinfo.packets_lost = stats.packets_lost;
    rinfo.fraction_lost = stats.fraction_lost;
    rinfo.codec_name = stats.codec_name;
    rinfo.codec_payload_type = stats.codec_payload_type;
    rinfo.ext_seqnum = stats.ext_seqnum;
    rinfo.jitter_ms = stats.jitter_ms;
    rinfo.jitter_buffer_ms = stats.jitter_buffer_ms;
    rinfo.jitter_buffer_preferred_ms = stats.jitter_buffer_preferred_ms;
    rinfo.delay_estimate_ms = stats.delay_estimate_ms;
    rinfo.audio_level = stats.audio_level;
    rinfo.expand_rate = stats.expand_rate;
    rinfo.speech_expand_rate = stats.speech_expand_rate;
    rinfo.secondary_decoded_rate = stats.secondary_decoded_rate;
    rinfo.accelerate_rate = stats.accelerate_rate;
    rinfo.preemptive_expand_rate = stats.preemptive_expand_rate;
    rinfo.decoding_calls_to_silence_generator =
        stats.decoding_calls_to_silence_generator;
    rinfo.decoding_calls_to_neteq = stats.decoding_calls_to_neteq;
    rinfo.decoding_normal = stats.decoding_normal;
    rinfo.decoding_plc = stats.decoding_plc;
    rinfo.decoding_cng = stats.decoding_cng;
    rinfo.decoding_plc_cng = stats.decoding_plc_cng;
    rinfo.decoding_muted_output = stats.decoding_muted_output;
    rinfo.capture_start_ntp_time_ms = stats.capture_start_ntp_time_ms;
    info->receivers.push_back(rinfo);
  }

  // Get codec info
  for (const AudioCodec& codec : send_codecs_) {
    webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
    info->send_codecs.insert(
        std::make_pair(codec_params.payload_type, std::move(codec_params)));
  }
  for (const AudioCodec& codec : recv_codecs_) {
    webrtc::RtpCodecParameters codec_params = codec.ToCodecParameters();
    info->receive_codecs.insert(
        std::make_pair(codec_params.payload_type, std::move(codec_params)));
  }

  return true;
}

void WebRtcVoiceMediaChannel::SetRawAudioSink(
    uint32_t ssrc,
    std::unique_ptr<webrtc::AudioSinkInterface> sink) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::SetRawAudioSink: ssrc:" << ssrc
                  << " " << (sink ? "(ptr)" : "NULL");
  if (ssrc == 0) {
    if (!unsignaled_recv_ssrcs_.empty()) {
      std::unique_ptr<webrtc::AudioSinkInterface> proxy_sink(
          sink ? new ProxySink(sink.get()) : nullptr);
      SetRawAudioSink(unsignaled_recv_ssrcs_.back(), std::move(proxy_sink));
    }
    default_sink_ = std::move(sink);
    return;
  }
  const auto it = recv_streams_.find(ssrc);
  if (it == recv_streams_.end()) {
    LOG(LS_WARNING) << "SetRawAudioSink: no recv stream " << ssrc;
    return;
  }
  it->second->SetRawAudioSink(std::move(sink));
}

int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  const auto it = recv_streams_.find(ssrc);
  if (it != recv_streams_.end()) {
    return it->second->channel();
  }
  return -1;
}

int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  const auto it = send_streams_.find(ssrc);
  if (it != send_streams_.end()) {
    return it->second->channel();
  }
  return -1;
}

bool WebRtcVoiceMediaChannel::
    MaybeDeregisterUnsignaledRecvStream(uint32_t ssrc) {
  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
  auto it = std::find(unsignaled_recv_ssrcs_.begin(),
                      unsignaled_recv_ssrcs_.end(),
                      ssrc);
  if (it != unsignaled_recv_ssrcs_.end()) {
    unsignaled_recv_ssrcs_.erase(it);
    return true;
  }
  return false;
}
}  // namespace cricket

#endif  // HAVE_WEBRTC_VOICE