talk/media/webrtc/webrtcvoiceengine.cc

/*
 * libjingle
 * Copyright 2004 Google Inc.
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions are met:
 *
 *  1. Redistributions of source code must retain the above copyright notice,
 *     this list of conditions and the following disclaimer.
 *  2. Redistributions in binary form must reproduce the above copyright notice,
 *     this list of conditions and the following disclaimer in the documentation
 *     and/or other materials provided with the distribution.
 *  3. The name of the author may not be used to endorse or promote products
 *     derived from this software without specific prior written permission.
 *
 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 */

#ifdef HAVE_CONFIG_H
#include <config.h>
#endif

#ifdef HAVE_WEBRTC_VOICE

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

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

#include "talk/media/base/audioframe.h"
#include "talk/media/base/audiorenderer.h"
#include "talk/media/base/constants.h"
#include "talk/media/base/streamparams.h"
#include "talk/media/webrtc/webrtcvoe.h"
#include "webrtc/base/base64.h"
#include "webrtc/base/byteorder.h"
#include "webrtc/base/common.h"
#include "webrtc/base/helpers.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/stringencode.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/common.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/system_wrappers/interface/field_trial.h"

namespace cricket {
namespace {

const int kMaxNumPacketSize = 6;
struct CodecPref {
  const char* name;
  int clockrate;
  int channels;
  int payload_type;
  bool is_multi_rate;
  int packet_sizes_ms[kMaxNumPacketSize];
};
// Note: keep the supported packet sizes in ascending order.
const CodecPref kCodecPrefs[] = {
  { kOpusCodecName,   48000, 2, 111, true,  { 10, 20, 40, 60 } },
  { kIsacCodecName,   16000, 1, 103, true,  { 30, 60 } },
  { kIsacCodecName,   32000, 1, 104, true,  { 30 } },
  // 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, { } },
  { kRedCodecName,    8000,  1, 127, false, { } },
  { kDtmfCodecName,   8000,  1, 126, false, { } },
};

// For Linux/Mac, using the default device is done by specifying index 0 for
// VoE 4.0 and not -1 (which was the case for VoE 3.5).
//
// On Windows Vista and newer, Microsoft introduced the concept of "Default
// Communications Device". This means that there are two types of default
// devices (old Wave Audio style default and Default Communications Device).
//
// On Windows systems which only support Wave Audio style default, uses either
// -1 or 0 to select the default device.
//
// On Windows systems which support both "Default Communication Device" and
// old Wave Audio style default, use -1 for Default Communications Device and
// -2 for Wave Audio style default, which is what we want to use for clips.
// It's not clear yet whether the -2 index is handled properly on other OSes.

#ifdef WIN32
const int kDefaultAudioDeviceId = -1;
#else
const int kDefaultAudioDeviceId = 0;
#endif

// Parameter used for NACK.
// This value is equivalent to 5 seconds of audio data at 20 ms per packet.
const int kNackMaxPackets = 250;

// 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 kOpusBitrateNb = 12000;
const int kOpusBitrateWb = 20000;
const int kOpusBitrateFb = 32000;

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

// 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;

// Ensure we open the file in a writeable path on ChromeOS and Android. This
// workaround can be removed when it's possible to specify a filename for audio
// option based AEC dumps.
//
// TODO(grunell): Use a string in the options instead of hardcoding it here
// and let the embedder choose the filename (crbug.com/264223).
//
// NOTE(ajm): Don't use hardcoded paths on platforms not explicitly specified
// below.
#if defined(CHROMEOS)
const char kAecDumpByAudioOptionFilename[] = "/tmp/audio.aecdump";
#elif defined(ANDROID)
const char kAecDumpByAudioOptionFilename[] = "/sdcard/audio.aecdump";
#else
const char kAecDumpByAudioOptionFilename[] = "audio.aecdump";
#endif

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();
}

std::string ToString(const webrtc::CodecInst& codec) {
  std::stringstream ss;
  ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels
     << " (" << codec.pltype << ")";
  return ss.str();
}

void LogMultiline(rtc::LoggingSeverity sev, char* text) {
  const char* delim = "\r\n";
  for (char* tok = strtok(text, delim); tok; tok = strtok(NULL, delim)) {
    LOG_V(sev) << tok;
  }
}

// Severity is an integer because it comes is assumed to be from command line.
int SeverityToFilter(int severity) {
  int filter = webrtc::kTraceNone;
  switch (severity) {
    case rtc::LS_VERBOSE:
      filter |= webrtc::kTraceAll;
      FALLTHROUGH();
    case rtc::LS_INFO:
      filter |= (webrtc::kTraceStateInfo | webrtc::kTraceInfo);
      FALLTHROUGH();
    case rtc::LS_WARNING:
      filter |= (webrtc::kTraceTerseInfo | webrtc::kTraceWarning);
      FALLTHROUGH();
    case rtc::LS_ERROR:
      filter |= (webrtc::kTraceError | webrtc::kTraceCritical);
  }
  return filter;
}

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

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();
}

bool IsNackEnabled(const AudioCodec& codec) {
  return codec.HasFeedbackParam(FeedbackParam(kRtcpFbParamNack,
                                              kParamValueEmpty));
}

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;
}

// 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.
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;
}

// 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;
}

// 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 = kOpusBitrateNb;
    } else if (max_playback_rate <= 16000) {
      bitrate = kOpusBitrateWb;
    } else {
      bitrate = kOpusBitrateFb;
    }

    if (IsCodecFeatureEnabled(codec, kCodecParamStereo)) {
      bitrate *= 2;
    }
  } else if (bitrate < kOpusMinBitrate || bitrate > kOpusMaxBitrate) {
    bitrate = (bitrate < kOpusMinBitrate) ? kOpusMinBitrate : kOpusMaxBitrate;
    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;
}

// Returns kOpusDefaultPlaybackRate if params[kCodecParamMaxPlaybackRate] is not
// defined. Returns the value of params[kCodecParamMaxPlaybackRate] otherwise.
int GetOpusMaxPlaybackRate(const AudioCodec& codec) {
  int value;
  if (codec.GetParam(kCodecParamMaxPlaybackRate, &value)) {
    return value;
  }
  return kOpusDefaultMaxPlaybackRate;
}

void GetOpusConfig(const AudioCodec& codec, webrtc::CodecInst* voe_codec,
                          bool* enable_codec_fec, int* max_playback_rate,
                          bool* enable_codec_dtx) {
  *enable_codec_fec = IsCodecFeatureEnabled(codec, kCodecParamUseInbandFec);
  *enable_codec_dtx = IsCodecFeatureEnabled(codec, kCodecParamUseDtx);
  *max_playback_rate = GetOpusMaxPlaybackRate(codec);

  // 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);
}

// 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.
void MaybeFixupG722(webrtc::CodecInst* voe_codec, int new_plfreq) {
  if (IsCodec(*voe_codec, kG722CodecName)) {
    // If the ASSERT 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;
  }
}

// Gets the default set of options applied to the engine. Historically, these
// were supplied as a combination of flags from the channel manager (ec, agc,
// ns, and highpass) and the rest hardcoded in InitInternal.
AudioOptions GetDefaultEngineOptions() {
  AudioOptions options;
  options.echo_cancellation.Set(true);
  options.auto_gain_control.Set(true);
  options.noise_suppression.Set(true);
  options.highpass_filter.Set(true);
  options.stereo_swapping.Set(false);
  options.audio_jitter_buffer_max_packets.Set(50);
  options.audio_jitter_buffer_fast_accelerate.Set(false);
  options.typing_detection.Set(true);
  options.adjust_agc_delta.Set(0);
  options.experimental_agc.Set(false);
  options.extended_filter_aec.Set(false);
  options.delay_agnostic_aec.Set(false);
  options.experimental_ns.Set(false);
  options.aec_dump.Set(false);
  return options;
}

std::string GetEnableString(bool enable) {
  return enable ? "enable" : "disable";
}
} // namespace {

WebRtcVoiceEngine::WebRtcVoiceEngine()
    : voe_wrapper_(new VoEWrapper()),
      tracing_(new VoETraceWrapper()),
      adm_(NULL),
      log_filter_(SeverityToFilter(kDefaultLogSeverity)),
      is_dumping_aec_(false) {
  Construct();
}

WebRtcVoiceEngine::WebRtcVoiceEngine(VoEWrapper* voe_wrapper,
                                     VoETraceWrapper* tracing)
    : voe_wrapper_(voe_wrapper),
      tracing_(tracing),
      adm_(NULL),
      log_filter_(SeverityToFilter(kDefaultLogSeverity)),
      is_dumping_aec_(false) {
  Construct();
}

void WebRtcVoiceEngine::Construct() {
  SetTraceFilter(log_filter_);
  initialized_ = false;
  LOG(LS_VERBOSE) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
  SetTraceOptions("");
  if (tracing_->SetTraceCallback(this) == -1) {
    LOG_RTCERR0(SetTraceCallback);
  }
  if (voe_wrapper_->base()->RegisterVoiceEngineObserver(*this) == -1) {
    LOG_RTCERR0(RegisterVoiceEngineObserver);
  }
  // Clear the default agc state.
  memset(&default_agc_config_, 0, sizeof(default_agc_config_));

  // Load our audio codec list.
  ConstructCodecs();

  // Load our RTP Header extensions.
  rtp_header_extensions_.push_back(
      RtpHeaderExtension(kRtpAudioLevelHeaderExtension,
                         kRtpAudioLevelHeaderExtensionDefaultId));
  rtp_header_extensions_.push_back(
      RtpHeaderExtension(kRtpAbsoluteSenderTimeHeaderExtension,
                         kRtpAbsoluteSenderTimeHeaderExtensionDefaultId));
  if (webrtc::field_trial::FindFullName("WebRTC-SendSideBwe") == "Enabled") {
    rtp_header_extensions_.push_back(RtpHeaderExtension(
        kRtpTransportSequenceNumberHeaderExtension,
        kRtpTransportSequenceNumberHeaderExtensionDefaultId));
  }
  options_ = GetDefaultEngineOptions();
}

void WebRtcVoiceEngine::ConstructCodecs() {
  LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
  int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
  for (int i = 0; i < ncodecs; ++i) {
    webrtc::CodecInst voe_codec;
    if (GetVoeCodec(i, &voe_codec)) {
      // Skip uncompressed formats.
      if (IsCodec(voe_codec, kL16CodecName)) {
        continue;
      }

      const CodecPref* pref = NULL;
      for (size_t j = 0; j < ARRAY_SIZE(kCodecPrefs); ++j) {
        if (IsCodec(voe_codec, kCodecPrefs[j].name) &&
            kCodecPrefs[j].clockrate == voe_codec.plfreq &&
            kCodecPrefs[j].channels == voe_codec.channels) {
          pref = &kCodecPrefs[j];
          break;
        }
      }

      if (pref) {
        // Use the payload type that we've configured in our pref table;
        // use the offset in our pref table to determine the sort order.
        AudioCodec codec(pref->payload_type, voe_codec.plname, voe_codec.plfreq,
                         voe_codec.rate, voe_codec.channels,
                         ARRAY_SIZE(kCodecPrefs) - (pref - kCodecPrefs));
        LOG(LS_INFO) << 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);

          // TODO(hellner): Add ptime, sprop-stereo, and stereo
          // when they can be set to values other than the default.
        }
        codecs_.push_back(codec);
      } else {
        LOG(LS_WARNING) << "Unexpected codec: " << ToString(voe_codec);
      }
    }
  }
  // Make sure they are in local preference order.
  std::sort(codecs_.begin(), codecs_.end(), &AudioCodec::Preferable);
}

bool WebRtcVoiceEngine::GetVoeCodec(int index, webrtc::CodecInst* codec) {
  if (voe_wrapper_->codec()->GetCodec(index, *codec) == -1) {
    return false;
  }
  // Change the sample rate of G722 to 8000 to match SDP.
  MaybeFixupG722(codec, 8000);
  return true;
}

WebRtcVoiceEngine::~WebRtcVoiceEngine() {
  LOG(LS_VERBOSE) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
  if (voe_wrapper_->base()->DeRegisterVoiceEngineObserver() == -1) {
    LOG_RTCERR0(DeRegisterVoiceEngineObserver);
  }
  if (adm_) {
    voe_wrapper_.reset();
    adm_->Release();
    adm_ = NULL;
  }

  tracing_->SetTraceCallback(NULL);
}

bool WebRtcVoiceEngine::Init(rtc::Thread* worker_thread) {
  RTC_DCHECK(worker_thread == rtc::Thread::Current());
  LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
  bool res = InitInternal();
  if (res) {
    LOG(LS_INFO) << "WebRtcVoiceEngine::Init Done!";
  } else {
    LOG(LS_ERROR) << "WebRtcVoiceEngine::Init failed";
    Terminate();
  }
  return res;
}

bool WebRtcVoiceEngine::InitInternal() {
  // Temporarily turn logging level up for the Init call
  int old_filter = log_filter_;
  int extended_filter = log_filter_ | SeverityToFilter(rtc::LS_INFO);
  SetTraceFilter(extended_filter);
  SetTraceOptions("");

  // Init WebRtc VoiceEngine.
  if (voe_wrapper_->base()->Init(adm_) == -1) {
    LOG_RTCERR0_EX(Init, voe_wrapper_->error());
    SetTraceFilter(old_filter);
    return false;
  }

  SetTraceFilter(old_filter);
  SetTraceOptions(log_options_);

  // Log the VoiceEngine version info
  char buffer[1024] = "";
  voe_wrapper_->base()->GetVersion(buffer);
  LOG(LS_INFO) << "WebRtc VoiceEngine Version:";
  LogMultiline(rtc::LS_INFO, buffer);

  // Save the default AGC configuration settings. This must happen before
  // calling SetOptions or the default will be overwritten.
  if (voe_wrapper_->processing()->GetAgcConfig(default_agc_config_) == -1) {
    LOG_RTCERR0(GetAgcConfig);
    return false;
  }

  // Set defaults for options, so that ApplyOptions applies them explicitly
  // when we clear option (channel) overrides. External clients can still
  // modify the defaults via SetOptions (on the media engine).
  if (!SetOptions(GetDefaultEngineOptions())) {
    return false;
  }

  // Print our codec list again for the call diagnostic log
  LOG(LS_INFO) << "WebRtc VoiceEngine codecs:";
  for (const AudioCodec& codec : codecs_) {
    LOG(LS_INFO) << ToString(codec);
  }

  // Disable the DTMF playout when a tone is sent.
  // PlayDtmfTone will be used if local playout is needed.
  if (voe_wrapper_->dtmf()->SetDtmfFeedbackStatus(false) == -1) {
    LOG_RTCERR1(SetDtmfFeedbackStatus, false);
  }

  initialized_ = true;
  return true;
}

void WebRtcVoiceEngine::Terminate() {
  LOG(LS_INFO) << "WebRtcVoiceEngine::Terminate";
  initialized_ = false;

  StopAecDump();

  voe_wrapper_->base()->Terminate();
}

VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(webrtc::Call* call,
    const AudioOptions& options) {
  return new WebRtcVoiceMediaChannel(this, options, call);
}

bool WebRtcVoiceEngine::SetOptions(const AudioOptions& options) {
  if (!ApplyOptions(options)) {
    return false;
  }
  options_ = options;
  return true;
}

// AudioOptions defaults are set in InitInternal (for options with corresponding
// MediaEngineInterface flags) and in SetOptions(int) for flagless options.
bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
  LOG(LS_INFO) << "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;
  webrtc::AecmModes aecm_mode = webrtc::kAecmSpeakerphone;
  webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
  webrtc::NsModes ns_mode = webrtc::kNsHighSuppression;
  bool aecm_comfort_noise = false;
  if (options.aecm_generate_comfort_noise.Get(&aecm_comfort_noise)) {
    LOG(LS_VERBOSE) << "Comfort noise explicitly set to "
                    << aecm_comfort_noise << " (default is false).";
  }

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

#if defined(IOS) || defined(ANDROID)
  // Set the AGC mode for iOS as well despite disabling it above, to avoid
  // unsupported configuration errors from webrtc.
  agc_mode = webrtc::kAgcFixedDigital;
  options.typing_detection.Set(false);
  options.experimental_agc.Set(false);
  options.extended_filter_aec.Set(false);
  options.experimental_ns.Set(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(IOS)
  if (options.delay_agnostic_aec.Get(&use_delay_agnostic_aec)) {
    if (use_delay_agnostic_aec) {
      options.echo_cancellation.Set(true);
      options.extended_filter_aec.Set(true);
      ec_mode = webrtc::kEcConference;
    }
  }
#endif

  webrtc::VoEAudioProcessing* voep = voe_wrapper_->processing();

  bool echo_cancellation = false;
  if (options.echo_cancellation.Get(&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 = voe_wrapper_->hw()->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 =
          echo_cancellation && !use_delay_agnostic_aec;
      if (voe_wrapper_->hw()->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.Set(false);
        echo_cancellation = false;
        LOG(LS_INFO) << "Disabling EC since built-in EC will be used instead";
      }
    }
    if (voep->SetEcStatus(echo_cancellation, ec_mode) == -1) {
      LOG_RTCERR2(SetEcStatus, echo_cancellation, ec_mode);
      return false;
    } else {
      LOG(LS_INFO) << "Echo control set to " << echo_cancellation
                   << " with mode " << ec_mode;
    }
#if !defined(ANDROID)
    // TODO(ajm): Remove the error return on Android from webrtc.
    if (voep->SetEcMetricsStatus(echo_cancellation) == -1) {
      LOG_RTCERR1(SetEcMetricsStatus, echo_cancellation);
      return false;
    }
#endif
    if (ec_mode == webrtc::kEcAecm) {
      if (voep->SetAecmMode(aecm_mode, aecm_comfort_noise) != 0) {
        LOG_RTCERR2(SetAecmMode, aecm_mode, aecm_comfort_noise);
        return false;
      }
    }
  }

  bool auto_gain_control = false;
  if (options.auto_gain_control.Get(&auto_gain_control)) {
    const bool built_in_agc = voe_wrapper_->hw()->BuiltInAGCIsAvailable();
    if (built_in_agc) {
      if (voe_wrapper_->hw()->EnableBuiltInAGC(auto_gain_control) == 0 &&
          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.Set(false);
        auto_gain_control = false;
        LOG(LS_INFO) << "Disabling AGC since built-in AGC will be used instead";
      }
    }
    if (voep->SetAgcStatus(auto_gain_control, agc_mode) == -1) {
      LOG_RTCERR2(SetAgcStatus, auto_gain_control, agc_mode);
      return false;
    } else {
      LOG(LS_INFO) << "Auto gain set to " << auto_gain_control << " with mode "
                   << agc_mode;
    }
  }

  if (options.tx_agc_target_dbov.IsSet() ||
      options.tx_agc_digital_compression_gain.IsSet() ||
      options.tx_agc_limiter.IsSet()) {
    // 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.GetWithDefaultIfUnset(
            default_agc_config_.targetLeveldBOv);
    default_agc_config_.digitalCompressionGaindB =
        options.tx_agc_digital_compression_gain.GetWithDefaultIfUnset(
            default_agc_config_.digitalCompressionGaindB);
    default_agc_config_.limiterEnable =
        options.tx_agc_limiter.GetWithDefaultIfUnset(
            default_agc_config_.limiterEnable);
    if (voe_wrapper_->processing()->SetAgcConfig(default_agc_config_) == -1) {
      LOG_RTCERR3(SetAgcConfig,
                  default_agc_config_.targetLeveldBOv,
                  default_agc_config_.digitalCompressionGaindB,
                  default_agc_config_.limiterEnable);
      return false;
    }
  }

  bool noise_suppression = false;
  if (options.noise_suppression.Get(&noise_suppression)) {
    const bool built_in_ns = voe_wrapper_->hw()->BuiltInNSIsAvailable();
    if (built_in_ns) {
      if (voe_wrapper_->hw()->EnableBuiltInNS(noise_suppression) == 0 &&
          noise_suppression) {
        // Disable internal software NS if built-in NS is enabled,
        // i.e., replace the software NS with the built-in NS.
        options.noise_suppression.Set(false);
        noise_suppression = false;
        LOG(LS_INFO) << "Disabling NS since built-in NS will be used instead";
      }
    }
    if (voep->SetNsStatus(noise_suppression, ns_mode) == -1) {
      LOG_RTCERR2(SetNsStatus, noise_suppression, ns_mode);
      return false;
    } else {
      LOG(LS_INFO) << "Noise suppression set to " << noise_suppression
                   << " with mode " << ns_mode;
    }
  }

  bool highpass_filter;
  if (options.highpass_filter.Get(&highpass_filter)) {
    LOG(LS_INFO) << "High pass filter enabled? " << highpass_filter;
    if (voep->EnableHighPassFilter(highpass_filter) == -1) {
      LOG_RTCERR1(SetHighpassFilterStatus, highpass_filter);
      return false;
    }
  }

  bool stereo_swapping;
  if (options.stereo_swapping.Get(&stereo_swapping)) {
    LOG(LS_INFO) << "Stereo swapping enabled? " << stereo_swapping;
    voep->EnableStereoChannelSwapping(stereo_swapping);
    if (voep->IsStereoChannelSwappingEnabled() != stereo_swapping) {
      LOG_RTCERR1(EnableStereoChannelSwapping, stereo_swapping);
      return false;
    }
  }

  int audio_jitter_buffer_max_packets;
  if (options.audio_jitter_buffer_max_packets.Get(
          &audio_jitter_buffer_max_packets)) {
    LOG(LS_INFO) << "NetEq capacity is " << audio_jitter_buffer_max_packets;
    voe_config_.Set<webrtc::NetEqCapacityConfig>(
        new webrtc::NetEqCapacityConfig(audio_jitter_buffer_max_packets));
  }

  bool audio_jitter_buffer_fast_accelerate;
  if (options.audio_jitter_buffer_fast_accelerate.Get(
          &audio_jitter_buffer_fast_accelerate)) {
    LOG(LS_INFO) << "NetEq fast mode? " << audio_jitter_buffer_fast_accelerate;
    voe_config_.Set<webrtc::NetEqFastAccelerate>(
        new webrtc::NetEqFastAccelerate(audio_jitter_buffer_fast_accelerate));
  }

  bool typing_detection;
  if (options.typing_detection.Get(&typing_detection)) {
    LOG(LS_INFO) << "Typing detection is enabled? " << typing_detection;
    if (voep->SetTypingDetectionStatus(typing_detection) == -1) {
      // In case of error, log the info and continue
      LOG_RTCERR1(SetTypingDetectionStatus, typing_detection);
    }
  }

  int adjust_agc_delta;
  if (options.adjust_agc_delta.Get(&adjust_agc_delta)) {
    LOG(LS_INFO) << "Adjust agc delta is " << adjust_agc_delta;
    if (!AdjustAgcLevel(adjust_agc_delta)) {
      return false;
    }
  }

  bool aec_dump;
  if (options.aec_dump.Get(&aec_dump)) {
    LOG(LS_INFO) << "Aec dump is enabled? " << aec_dump;
    if (aec_dump)
      StartAecDump(kAecDumpByAudioOptionFilename);
    else
      StopAecDump();
  }

  webrtc::Config config;

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

  extended_filter_aec_.SetFrom(options.extended_filter_aec);
  bool extended_filter;
  if (extended_filter_aec_.Get(&extended_filter)) {
    LOG(LS_INFO) << "Extended filter aec is enabled? " << extended_filter;
    config.Set<webrtc::ExtendedFilter>(
        new webrtc::ExtendedFilter(extended_filter));
  }

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

  // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
  // returns NULL on audio_processing().
  webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
  if (audioproc) {
    audioproc->SetExtraOptions(config);
  }

  uint32_t recording_sample_rate;
  if (options.recording_sample_rate.Get(&recording_sample_rate)) {
    LOG(LS_INFO) << "Recording sample rate is " << recording_sample_rate;
    if (voe_wrapper_->hw()->SetRecordingSampleRate(recording_sample_rate)) {
      LOG_RTCERR1(SetRecordingSampleRate, recording_sample_rate);
    }
  }

  uint32_t playout_sample_rate;
  if (options.playout_sample_rate.Get(&playout_sample_rate)) {
    LOG(LS_INFO) << "Playout sample rate is " << playout_sample_rate;
    if (voe_wrapper_->hw()->SetPlayoutSampleRate(playout_sample_rate)) {
      LOG_RTCERR1(SetPlayoutSampleRate, playout_sample_rate);
    }
  }

  return true;
}

// TODO(juberti): Refactor this so that the core logic can be used to set the
// soundclip device. At that time, reinstate the soundclip pause/resume code.
bool WebRtcVoiceEngine::SetDevices(const Device* in_device,
                                   const Device* out_device) {
#if !defined(IOS)
  int in_id = in_device ? rtc::FromString<int>(in_device->id) :
      kDefaultAudioDeviceId;
  int out_id = out_device ? rtc::FromString<int>(out_device->id) :
      kDefaultAudioDeviceId;
  // The device manager uses -1 as the default device, which was the case for
  // VoE 3.5. VoE 4.0, however, uses 0 as the default in Linux and Mac.
#ifndef WIN32
  if (-1 == in_id) {
    in_id = kDefaultAudioDeviceId;
  }
  if (-1 == out_id) {
    out_id = kDefaultAudioDeviceId;
  }
#endif

  std::string in_name = (in_id != kDefaultAudioDeviceId) ?
      in_device->name : "Default device";
  std::string out_name = (out_id != kDefaultAudioDeviceId) ?
      out_device->name : "Default device";
  LOG(LS_INFO) << "Setting microphone to (id=" << in_id << ", name=" << in_name
            << ") and speaker to (id=" << out_id << ", name=" << out_name
            << ")";

  // Must also pause all audio playback and capture.
  bool ret = true;
  for (WebRtcVoiceMediaChannel* channel : channels_) {
    if (!channel->PausePlayout()) {
      LOG(LS_WARNING) << "Failed to pause playout";
      ret = false;
    }
    if (!channel->PauseSend()) {
      LOG(LS_WARNING) << "Failed to pause send";
      ret = false;
    }
  }

  // Find the recording device id in VoiceEngine and set recording device.
  if (!FindWebRtcAudioDeviceId(true, in_name, in_id, &in_id)) {
    ret = false;
  }
  if (ret) {
    if (voe_wrapper_->hw()->SetRecordingDevice(in_id) == -1) {
      LOG_RTCERR2(SetRecordingDevice, in_name, in_id);
      ret = false;
    }
    webrtc::AudioProcessing* ap = voe()->base()->audio_processing();
    if (ap)
      ap->Initialize();
  }

  // Find the playout device id in VoiceEngine and set playout device.
  if (!FindWebRtcAudioDeviceId(false, out_name, out_id, &out_id)) {
    LOG(LS_WARNING) << "Failed to find VoiceEngine device id for " << out_name;
    ret = false;
  }
  if (ret) {
    if (voe_wrapper_->hw()->SetPlayoutDevice(out_id) == -1) {
      LOG_RTCERR2(SetPlayoutDevice, out_name, out_id);
      ret = false;
    }
  }

  // Resume all audio playback and capture.
  for (WebRtcVoiceMediaChannel* channel : channels_) {
    if (!channel->ResumePlayout()) {
      LOG(LS_WARNING) << "Failed to resume playout";
      ret = false;
    }
    if (!channel->ResumeSend()) {
      LOG(LS_WARNING) << "Failed to resume send";
      ret = false;
    }
  }

  if (ret) {
    LOG(LS_INFO) << "Set microphone to (id=" << in_id <<" name=" << in_name
                 << ") and speaker to (id="<< out_id << " name=" << out_name
                 << ")";
  }

  return ret;
#else
  return true;
#endif  // !IOS
}

bool WebRtcVoiceEngine::FindWebRtcAudioDeviceId(
  bool is_input, const std::string& dev_name, int dev_id, int* rtc_id) {
  // In Linux, VoiceEngine uses the same device dev_id as the device manager.
#if defined(LINUX) || defined(ANDROID)
  *rtc_id = dev_id;
  return true;
#else
  // In Windows and Mac, we need to find the VoiceEngine device id by name
  // unless the input dev_id is the default device id.
  if (kDefaultAudioDeviceId == dev_id) {
    *rtc_id = dev_id;
    return true;
  }

  // Get the number of VoiceEngine audio devices.
  int count = 0;
  if (is_input) {
    if (-1 == voe_wrapper_->hw()->GetNumOfRecordingDevices(count)) {
      LOG_RTCERR0(GetNumOfRecordingDevices);
      return false;
    }
  } else {
    if (-1 == voe_wrapper_->hw()->GetNumOfPlayoutDevices(count)) {
      LOG_RTCERR0(GetNumOfPlayoutDevices);
      return false;
    }
  }

  for (int i = 0; i < count; ++i) {
    char name[128];
    char guid[128];
    if (is_input) {
      voe_wrapper_->hw()->GetRecordingDeviceName(i, name, guid);
      LOG(LS_VERBOSE) << "VoiceEngine microphone " << i << ": " << name;
    } else {
      voe_wrapper_->hw()->GetPlayoutDeviceName(i, name, guid);
      LOG(LS_VERBOSE) << "VoiceEngine speaker " << i << ": " << name;
    }

    std::string webrtc_name(name);
    if (dev_name.compare(0, webrtc_name.size(), webrtc_name) == 0) {
      *rtc_id = i;
      return true;
    }
  }
  LOG(LS_WARNING) << "VoiceEngine cannot find device: " << dev_name;
  return false;
#endif
}

bool WebRtcVoiceEngine::GetOutputVolume(int* level) {
  unsigned int ulevel;
  if (voe_wrapper_->volume()->GetSpeakerVolume(ulevel) == -1) {
    LOG_RTCERR1(GetSpeakerVolume, level);
    return false;
  }
  *level = ulevel;
  return true;
}

bool WebRtcVoiceEngine::SetOutputVolume(int level) {
  RTC_DCHECK(level >= 0 && level <= 255);
  if (voe_wrapper_->volume()->SetSpeakerVolume(level) == -1) {
    LOG_RTCERR1(SetSpeakerVolume, level);
    return false;
  }
  return true;
}

int WebRtcVoiceEngine::GetInputLevel() {
  unsigned int ulevel;
  return (voe_wrapper_->volume()->GetSpeechInputLevel(ulevel) != -1) ?
      static_cast<int>(ulevel) : -1;
}

const std::vector<AudioCodec>& WebRtcVoiceEngine::codecs() {
  return codecs_;
}

bool WebRtcVoiceEngine::FindCodec(const AudioCodec& in) {
  return FindWebRtcCodec(in, NULL);
}

// Get the VoiceEngine codec that matches |in|, with the supplied settings.
bool WebRtcVoiceEngine::FindWebRtcCodec(const AudioCodec& in,
                                        webrtc::CodecInst* out) {
  int ncodecs = voe_wrapper_->codec()->NumOfCodecs();
  for (int i = 0; i < ncodecs; ++i) {
    webrtc::CodecInst voe_codec;
    if (GetVoeCodec(i, &voe_codec)) {
      AudioCodec codec(voe_codec.pltype, voe_codec.plname, voe_codec.plfreq,
                       voe_codec.rate, voe_codec.channels, 0);
      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);

          // Apply codec-specific settings.
          if (IsCodec(codec, kIsacCodecName)) {
            // If ISAC and an explicit bitrate is not specified,
            // enable auto bitrate adjustment.
            voe_codec.rate = (in.bitrate > 0) ? in.bitrate : -1;
          }
          *out = voe_codec;
        }
        return true;
      }
    }
  }
  return false;
}
const std::vector<RtpHeaderExtension>&
WebRtcVoiceEngine::rtp_header_extensions() const {
  return rtp_header_extensions_;
}

void WebRtcVoiceEngine::SetLogging(int min_sev, const char* filter) {
  // if min_sev == -1, we keep the current log level.
  if (min_sev >= 0) {
    SetTraceFilter(SeverityToFilter(min_sev));
  }
  log_options_ = filter;
  SetTraceOptions(initialized_ ? log_options_ : "");
}

int WebRtcVoiceEngine::GetLastEngineError() {
  return voe_wrapper_->error();
}

void WebRtcVoiceEngine::SetTraceFilter(int filter) {
  log_filter_ = filter;
  tracing_->SetTraceFilter(filter);
}

// We suppport three different logging settings for VoiceEngine:
// 1. Observer callback that goes into talk diagnostic logfile.
//    Use --logfile and --loglevel
//
// 2. Encrypted VoiceEngine log for debugging VoiceEngine.
//    Use --voice_loglevel --voice_logfilter "tracefile file_name"
//
// 3. EC log and dump for debugging QualityEngine.
//    Use --voice_loglevel --voice_logfilter "recordEC file_name"
//
// For more details see: "https://sites.google.com/a/google.com/wavelet/Home/
//    Magic-Flute--RTC-Engine-/Magic-Flute-Command-Line-Parameters"
void WebRtcVoiceEngine::SetTraceOptions(const std::string& options) {
  // Set encrypted trace file.
  std::vector<std::string> opts;
  rtc::tokenize(options, ' ', '"', '"', &opts);
  std::vector<std::string>::iterator tracefile =
      std::find(opts.begin(), opts.end(), "tracefile");
  if (tracefile != opts.end() && ++tracefile != opts.end()) {
    // Write encrypted debug output (at same loglevel) to file
    // EncryptedTraceFile no longer supported.
    if (tracing_->SetTraceFile(tracefile->c_str()) == -1) {
      LOG_RTCERR1(SetTraceFile, *tracefile);
    }
  }

  // Allow trace options to override the trace filter. We default
  // it to log_filter_ (as a translation of libjingle log levels)
  // elsewhere, but this allows clients to explicitly set webrtc
  // log levels.
  std::vector<std::string>::iterator tracefilter =
      std::find(opts.begin(), opts.end(), "tracefilter");
  if (tracefilter != opts.end() && ++tracefilter != opts.end()) {
    if (!tracing_->SetTraceFilter(rtc::FromString<int>(*tracefilter))) {
      LOG_RTCERR1(SetTraceFilter, *tracefilter);
    }
  }

  // Set AEC dump file
  std::vector<std::string>::iterator recordEC =
      std::find(opts.begin(), opts.end(), "recordEC");
  if (recordEC != opts.end()) {
    ++recordEC;
    if (recordEC != opts.end())
      StartAecDump(recordEC->c_str());
    else
      StopAecDump();
  }
}

void WebRtcVoiceEngine::Print(webrtc::TraceLevel level, const char* trace,
                              int length) {
  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::CallbackOnError(int channel_id, int err_code) {
  RTC_DCHECK(channel_id == -1);
  LOG(LS_WARNING) << "VoiceEngine error " << err_code << " reported on channel "
                  << channel_id << ".";
  rtc::CritScope lock(&channels_cs_);
  for (WebRtcVoiceMediaChannel* channel : channels_) {
    channel->OnError(err_code);
  }
}

void WebRtcVoiceEngine::RegisterChannel(WebRtcVoiceMediaChannel* channel) {
  RTC_DCHECK(channel != NULL);
  rtc::CritScope lock(&channels_cs_);
  channels_.push_back(channel);
}

void WebRtcVoiceEngine::UnregisterChannel(WebRtcVoiceMediaChannel* channel) {
  rtc::CritScope lock(&channels_cs_);
  auto it = std::find(channels_.begin(), channels_.end(), channel);
  if (it != channels_.end()) {
    channels_.erase(it);
  }
}

// Adjusts the default AGC target level by the specified delta.
// NB: If we start messing with other config fields, we'll want
// to save the current webrtc::AgcConfig as well.
bool WebRtcVoiceEngine::AdjustAgcLevel(int delta) {
  webrtc::AgcConfig config = default_agc_config_;
  config.targetLeveldBOv -= delta;

  LOG(LS_INFO) << "Adjusting AGC level from default -"
               << default_agc_config_.targetLeveldBOv << "dB to -"
               << config.targetLeveldBOv << "dB";

  if (voe_wrapper_->processing()->SetAgcConfig(config) == -1) {
    LOG_RTCERR1(SetAgcConfig, config.targetLeveldBOv);
    return false;
  }
  return true;
}

bool WebRtcVoiceEngine::SetAudioDeviceModule(webrtc::AudioDeviceModule* adm) {
  if (initialized_) {
    LOG(LS_WARNING) << "SetAudioDeviceModule can not be called after Init.";
    return false;
  }
  if (adm_) {
    adm_->Release();
    adm_ = NULL;
  }
  if (adm) {
    adm_ = adm;
    adm_->AddRef();
  }
  return true;
}

bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
  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 (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) !=
      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) {
  if (!is_dumping_aec_) {
    // Start dumping AEC when we are not dumping.
    if (voe_wrapper_->processing()->StartDebugRecording(
        filename.c_str()) != webrtc::AudioProcessing::kNoError) {
      LOG_RTCERR1(StartDebugRecording, filename.c_str());
    } else {
      is_dumping_aec_ = true;
    }
  }
}

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

bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
  return voe_wrapper_->codec()->GetEventLog()->StartLogging(file);
}

void WebRtcVoiceEngine::StopRtcEventLog() {
  voe_wrapper_->codec()->GetEventLog()->StopLogging();
}

int WebRtcVoiceEngine::CreateVoEChannel() {
  return voe_wrapper_->base()->CreateChannel(voe_config_);
}

class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
    : public AudioRenderer::Sink {
 public:
  WebRtcVoiceChannelRenderer(int ch,
                             webrtc::AudioTransport* voe_audio_transport)
      : channel_(ch),
        voe_audio_transport_(voe_audio_transport),
        renderer_(NULL) {}
  ~WebRtcVoiceChannelRenderer() override { Stop(); }

  // Starts the rendering by setting a sink to the renderer to get data
  // callback.
  // This method is called on the libjingle worker thread.
  // TODO(xians): Make sure Start() is called only once.
  void Start(AudioRenderer* renderer) {
    rtc::CritScope lock(&lock_);
    RTC_DCHECK(renderer != NULL);
    if (renderer_ != NULL) {
      RTC_DCHECK(renderer_ == renderer);
      return;
    }
    renderer->SetSink(this);
    renderer_ = renderer;
  }

  // Stops rendering by setting the sink of the renderer to NULL. No data
  // callback will be received after this method.
  // This method is called on the libjingle worker thread.
  void Stop() {
    rtc::CritScope lock(&lock_);
    if (renderer_ != NULL) {
      renderer_->SetSink(NULL);
      renderer_ = NULL;
    }
  }

  // AudioRenderer::Sink implementation.
  // This method is called on the audio thread.
  void OnData(const void* audio_data,
              int bits_per_sample,
              int sample_rate,
              int number_of_channels,
              size_t number_of_frames) override {
    voe_audio_transport_->OnData(channel_,
                                 audio_data,
                                 bits_per_sample,
                                 sample_rate,
                                 number_of_channels,
                                 number_of_frames);
  }

  // Callback from the |renderer_| when it is going away. In case Start() has
  // never been called, this callback won't be triggered.
  void OnClose() override {
    rtc::CritScope lock(&lock_);
    // Set |renderer_| to NULL to make sure no more callback will get into
    // the renderer.
    renderer_ = NULL;
  }

  // Accessor to the VoE channel ID.
  int channel() const { return channel_; }

 private:
  const int channel_;
  webrtc::AudioTransport* const voe_audio_transport_;

  // Raw pointer to AudioRenderer owned by LocalAudioTrackHandler.
  // PeerConnection will make sure invalidating the pointer before the object
  // goes away.
  AudioRenderer* renderer_;

  // Protects |renderer_| in Start(), Stop() and OnClose().
  rtc::CriticalSection lock_;
};

// WebRtcVoiceMediaChannel
WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
                                                 const AudioOptions& options,
                                                 webrtc::Call* call)
    : engine_(engine),
      send_bitrate_setting_(false),
      send_bitrate_bps_(0),
      options_(),
      dtmf_allowed_(false),
      desired_playout_(false),
      nack_enabled_(false),
      playout_(false),
      typing_noise_detected_(false),
      desired_send_(SEND_NOTHING),
      send_(SEND_NOTHING),
      call_(call) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel";
  RTC_DCHECK(nullptr != call);
  engine->RegisterChannel(this);
  SetOptions(options);
}

WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel";

  // Remove any remaining send streams.
  while (!send_channels_.empty()) {
    RemoveSendStream(send_channels_.begin()->first);
  }

  // Remove any remaining receive streams.
  while (!receive_channels_.empty()) {
    RemoveRecvStream(receive_channels_.begin()->first);
  }
  RTC_DCHECK(receive_streams_.empty());

  // Unregister ourselves from the engine.
  engine()->UnregisterChannel(this);
}

bool WebRtcVoiceMediaChannel::SetSendParameters(
    const AudioSendParameters& params) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  // TODO(pthatcher): Refactor this to be more clean now that we have
  // all the information at once.
  return (SetSendCodecs(params.codecs) &&
          SetSendRtpHeaderExtensions(params.extensions) &&
          SetMaxSendBandwidth(params.max_bandwidth_bps) &&
          SetOptions(params.options));
}

bool WebRtcVoiceMediaChannel::SetRecvParameters(
    const AudioRecvParameters& params) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  // TODO(pthatcher): Refactor this to be more clean now that we have
  // all the information at once.
  return (SetRecvCodecs(params.codecs) &&
          SetRecvRtpHeaderExtensions(params.extensions));
}

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

  // Check if DSCP value is changed from previous.
  bool dscp_option_changed = (options_.dscp != options.dscp);

  // 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 (send_ != SEND_NOTHING) {
    if (!engine()->ApplyOptions(options_)) {
      LOG(LS_WARNING) <<
          "Failed to apply engine options during channel SetOptions.";
      return false;
    }
  }

  if (dscp_option_changed) {
    rtc::DiffServCodePoint dscp = rtc::DSCP_DEFAULT;
    if (options_.dscp.GetWithDefaultIfUnset(false))
      dscp = kAudioDscpValue;
    if (MediaChannel::SetDscp(dscp) != 0) {
      LOG(LS_WARNING) << "Failed to set DSCP settings for audio channel";
    }
  }

  RecreateAudioReceiveStreams();

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

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

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

  bool result = SetRecvCodecsInternal(new_codecs);
  if (result) {
    recv_codecs_ = codecs;
  }

  if (desired_playout_ && !playout_) {
    ResumePlayout();
  }
  return result;
}

bool WebRtcVoiceMediaChannel::SetSendCodecs(
    int channel, const std::vector<AudioCodec>& codecs) {
  // Disable VAD, FEC, and RED unless we know the other side wants them.
  engine()->voe()->codec()->SetVADStatus(channel, false);
  engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
  engine()->voe()->rtp()->SetREDStatus(channel, false);
  engine()->voe()->codec()->SetFECStatus(channel, false);

  // Scan through the list to figure out the codec to use for sending, along
  // with the proper configuration for VAD and DTMF.
  bool found_send_codec = false;
  webrtc::CodecInst send_codec;
  memset(&send_codec, 0, sizeof(send_codec));

  bool nack_enabled = nack_enabled_;
  bool enable_codec_fec = false;
  bool enable_opus_dtx = false;
  int opus_max_playback_rate = 0;

  // Set send codec (the first non-telephone-event/CN codec)
  for (const AudioCodec& 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;
    if (!engine()->FindWebRtcCodec(codec, &voe_codec)) {
      LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
      continue;
    }

    if (IsCodec(codec, kDtmfCodecName) || IsCodec(codec, kCnCodecName)) {
      // Skip telephone-event/CN codec, which 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.
    // "red", for RED audio, is a special case where the actual codec to be
    // used is specified in params.
    if (IsCodec(codec, kRedCodecName)) {
      // Parse out the RED parameters. If we fail, just ignore RED;
      // we don't support all possible params/usage scenarios.
      if (!GetRedSendCodec(codec, codecs, &send_codec)) {
        continue;
      }

      // Enable redundant encoding of the specified codec. Treat any
      // failure as a fatal internal error.
      LOG(LS_INFO) << "Enabling RED on channel " << channel;
      if (engine()->voe()->rtp()->SetREDStatus(channel, true, codec.id) == -1) {
        LOG_RTCERR3(SetREDStatus, channel, true, codec.id);
        return false;
      }
    } else {
      send_codec = voe_codec;
      nack_enabled = IsNackEnabled(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, &enable_codec_fec,
                      &opus_max_playback_rate, &enable_opus_dtx);
      }

      // Set packet size if the AudioCodec param kCodecParamPTime is set.
      int ptime_ms = 0;
      if (codec.GetParam(kCodecParamPTime, &ptime_ms)) {
        if (!SetPTimeAsPacketSize(&send_codec, ptime_ms)) {
          LOG(LS_WARNING) << "Failed to set packet size for codec "
                          << send_codec.plname;
          return false;
        }
      }
    }
    found_send_codec = true;
    break;
  }

  if (nack_enabled_ != nack_enabled) {
    SetNack(channel, nack_enabled);
    nack_enabled_ = nack_enabled;
  }

  if (!found_send_codec) {
    LOG(LS_WARNING) << "Received empty list of codecs.";
    return false;
  }

  // Set the codec immediately, since SetVADStatus() depends on whether
  // the current codec is mono or stereo.
  if (!SetSendCodec(channel, send_codec))
    return false;

  // FEC should be enabled after SetSendCodec.
  if (enable_codec_fec) {
    LOG(LS_INFO) << "Attempt to enable codec internal FEC on channel "
                 << channel;
    if (engine()->voe()->codec()->SetFECStatus(channel, true) == -1) {
      // Enable codec internal FEC. Treat any failure as fatal internal error.
      LOG_RTCERR2(SetFECStatus, channel, true);
      return false;
    }
  }

  if (IsCodec(send_codec, kOpusCodecName)) {
    // DTX and maxplaybackrate should be set after SetSendCodec. Because current
    // send codec has to be Opus.

    // Set Opus internal DTX.
    LOG(LS_INFO) << "Attempt to "
                 << GetEnableString(enable_opus_dtx)
                 << " Opus DTX on channel "
                 << channel;
    if (engine()->voe()->codec()->SetOpusDtx(channel, enable_opus_dtx)) {
      LOG_RTCERR2(SetOpusDtx, channel, enable_opus_dtx);
      return false;
    }

    // If opus_max_playback_rate <= 0, the default maximum playback rate
    // (48 kHz) will be used.
    if (opus_max_playback_rate > 0) {
      LOG(LS_INFO) << "Attempt to set maximum playback rate to "
                   << opus_max_playback_rate
                   << " Hz on channel "
                   << channel;
      if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
          channel, opus_max_playback_rate) == -1) {
        LOG_RTCERR2(SetOpusMaxPlaybackRate, channel, opus_max_playback_rate);
        return false;
      }
    }
  }

  // Always update the |send_codec_| to the currently set send codec.
  send_codec_.reset(new webrtc::CodecInst(send_codec));

  if (send_bitrate_setting_) {
    SetSendBitrateInternal(send_bitrate_bps_);
  }

  // Loop through the codecs list again to config the telephone-event/CN codec.
  for (const AudioCodec& 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;
    if (!engine()->FindWebRtcCodec(codec, &voe_codec)) {
      LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
      continue;
    }

    // Find the DTMF telephone event "codec" and tell VoiceEngine channels
    // about it.
    if (IsCodec(codec, kDtmfCodecName)) {
      if (engine()->voe()->dtmf()->SetSendTelephoneEventPayloadType(
              channel, codec.id) == -1) {
        LOG_RTCERR2(SetSendTelephoneEventPayloadType, channel, codec.id);
        return false;
      }
    } else if (IsCodec(codec, kCnCodecName)) {
      // Turn voice activity detection/comfort noise on if supported.
      // Set the wideband CN payload type appropriately.
      // (narrowband always uses the static payload type 13).
      webrtc::PayloadFrequencies cn_freq;
      switch (codec.clockrate) {
        case 8000:
          cn_freq = webrtc::kFreq8000Hz;
          break;
        case 16000:
          cn_freq = webrtc::kFreq16000Hz;
          break;
        case 32000:
          cn_freq = webrtc::kFreq32000Hz;
          break;
        default:
          LOG(LS_WARNING) << "CN frequency " << codec.clockrate
                          << " not supported.";
          continue;
      }
      // Set the CN payloadtype and the VAD status.
      // The CN payload type for 8000 Hz clockrate is fixed at 13.
      if (cn_freq != webrtc::kFreq8000Hz) {
        if (engine()->voe()->codec()->SetSendCNPayloadType(
                channel, codec.id, cn_freq) == -1) {
          LOG_RTCERR3(SetSendCNPayloadType, channel, codec.id, cn_freq);
          // TODO(ajm): This failure condition will be removed from VoE.
          // Restore the return here when we update to a new enough webrtc.
          //
          // Not returning false because the SetSendCNPayloadType will fail if
          // the channel is already sending.
          // This can happen if the remote description is applied twice, for
          // example in the case of ROAP on top of JSEP, where both side will
          // send the offer.
        }
      }
      // Only turn on VAD if we have a CN payload type that matches the
      // clockrate for the codec we are going to use.
      if (codec.clockrate == send_codec.plfreq && send_codec.channels != 2) {
        // TODO(minyue): If CN frequency == 48000 Hz is allowed, consider the
        // interaction between VAD and Opus FEC.
        LOG(LS_INFO) << "Enabling VAD";
        if (engine()->voe()->codec()->SetVADStatus(channel, true) == -1) {
          LOG_RTCERR2(SetVADStatus, channel, true);
          return false;
        }
      }
    }
  }
  return true;
}

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

  dtmf_allowed_ = false;
  for (const AudioCodec& codec : codecs) {
    // Find the DTMF telephone event "codec".
    if (IsCodec(codec, kDtmfCodecName)) {
      dtmf_allowed_ = true;
    }
  }

  // Cache the codecs in order to configure the channel created later.
  send_codecs_ = codecs;
  for (const auto& ch : send_channels_) {
    if (!SetSendCodecs(ch.second->channel(), codecs)) {
      return false;
    }
  }

  // Set nack status on receive channels and update |nack_enabled_|.
  for (const auto& ch : receive_channels_) {
    SetNack(ch.second->channel(), nack_enabled_);
  }

  return true;
}

void WebRtcVoiceMediaChannel::SetNack(int channel, bool nack_enabled) {
  if (nack_enabled) {
    LOG(LS_INFO) << "Enabling NACK for channel " << channel;
    engine()->voe()->rtp()->SetNACKStatus(channel, true, kNackMaxPackets);
  } else {
    LOG(LS_INFO) << "Disabling NACK for channel " << channel;
    engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
  }
}

bool WebRtcVoiceMediaChannel::SetSendCodec(
    int channel, const webrtc::CodecInst& send_codec) {
  LOG(LS_INFO) << "Send channel " << channel <<  " selected voice codec "
               << ToString(send_codec) << ", bitrate=" << send_codec.rate;

  webrtc::CodecInst current_codec;
  if (engine()->voe()->codec()->GetSendCodec(channel, current_codec) == 0 &&
      (send_codec == current_codec)) {
    // Codec is already configured, we can return without setting it again.
    return true;
  }

  if (engine()->voe()->codec()->SetSendCodec(channel, send_codec) == -1) {
    LOG_RTCERR2(SetSendCodec, channel, ToString(send_codec));
    return false;
  }
  return true;
}

bool WebRtcVoiceMediaChannel::SetRecvRtpHeaderExtensions(
    const std::vector<RtpHeaderExtension>& extensions) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  if (receive_extensions_ == extensions) {
    return true;
  }

  for (const auto& ch : receive_channels_) {
    if (!SetChannelRecvRtpHeaderExtensions(ch.second->channel(), extensions)) {
      return false;
    }
  }

  receive_extensions_ = extensions;

  // Recreate AudioReceiveStream:s.
  {
    std::vector<webrtc::RtpExtension> exts;

    const RtpHeaderExtension* audio_level_extension =
        FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
    if (audio_level_extension) {
      exts.push_back({
          kRtpAudioLevelHeaderExtension, audio_level_extension->id});
    }

    const RtpHeaderExtension* send_time_extension =
        FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
    if (send_time_extension) {
      exts.push_back({
          kRtpAbsoluteSenderTimeHeaderExtension, send_time_extension->id});
    }

    recv_rtp_extensions_.swap(exts);
    RecreateAudioReceiveStreams();
  }

  return true;
}

bool WebRtcVoiceMediaChannel::SetChannelRecvRtpHeaderExtensions(
    int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
  const RtpHeaderExtension* audio_level_extension =
      FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);
  if (!SetHeaderExtension(
      &webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus, channel_id,
      audio_level_extension)) {
    return false;
  }

  const RtpHeaderExtension* send_time_extension =
      FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
  if (!SetHeaderExtension(
      &webrtc::VoERTP_RTCP::SetReceiveAbsoluteSenderTimeStatus, channel_id,
      send_time_extension)) {
    return false;
  }

  return true;
}

bool WebRtcVoiceMediaChannel::SetSendRtpHeaderExtensions(
    const std::vector<RtpHeaderExtension>& extensions) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  if (send_extensions_ == extensions) {
    return true;
  }

  for (const auto& ch : send_channels_) {
    if (!SetChannelSendRtpHeaderExtensions(ch.second->channel(), extensions)) {
      return false;
    }
  }

  send_extensions_ = extensions;
  return true;
}

bool WebRtcVoiceMediaChannel::SetChannelSendRtpHeaderExtensions(
    int channel_id, const std::vector<RtpHeaderExtension>& extensions) {
  const RtpHeaderExtension* audio_level_extension =
      FindHeaderExtension(extensions, kRtpAudioLevelHeaderExtension);

  if (!SetHeaderExtension(
      &webrtc::VoERTP_RTCP::SetSendAudioLevelIndicationStatus, channel_id,
      audio_level_extension)) {
    return false;
  }

  const RtpHeaderExtension* send_time_extension =
      FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
  if (!SetHeaderExtension(
      &webrtc::VoERTP_RTCP::SetSendAbsoluteSenderTimeStatus, channel_id,
      send_time_extension)) {
    return false;
  }

  return true;
}

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

bool WebRtcVoiceMediaChannel::PausePlayout() {
  return ChangePlayout(false);
}

bool WebRtcVoiceMediaChannel::ResumePlayout() {
  return ChangePlayout(desired_playout_);
}

bool WebRtcVoiceMediaChannel::ChangePlayout(bool playout) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  if (playout_ == playout) {
    return true;
  }

  for (const auto& ch : receive_channels_) {
    if (!SetPlayout(ch.second->channel(), playout)) {
      LOG(LS_ERROR) << "SetPlayout " << playout << " on channel "
                    << ch.second->channel() << " failed";
      return false;
    }
  }
  playout_ = playout;
  return true;
}

bool WebRtcVoiceMediaChannel::SetSend(SendFlags send) {
  desired_send_ = send;
  if (!send_channels_.empty())
    return ChangeSend(desired_send_);
  return true;
}

bool WebRtcVoiceMediaChannel::PauseSend() {
  return ChangeSend(SEND_NOTHING);
}

bool WebRtcVoiceMediaChannel::ResumeSend() {
  return ChangeSend(desired_send_);
}

bool WebRtcVoiceMediaChannel::ChangeSend(SendFlags send) {
  if (send_ == send) {
    return true;
  }

  // Apply channel specific options.
  if (send == SEND_MICROPHONE) {
    engine()->ApplyOptions(options_);
  }

  // Change the settings on each send channel.
  for (const auto& ch : send_channels_) {
    if (!ChangeSend(ch.second->channel(), send)) {
      return false;
    }
  }

  // Clear up the options after stopping sending. Since we may previously have
  // applied the channel specific options, now apply the original options stored
  // in WebRtcVoiceEngine.
  if (send == SEND_NOTHING) {
    engine()->ApplyOptions(engine()->GetOptions());
  }

  send_ = send;
  return true;
}

bool WebRtcVoiceMediaChannel::ChangeSend(int channel, SendFlags send) {
  if (send == SEND_MICROPHONE) {
    if (engine()->voe()->base()->StartSend(channel) == -1) {
      LOG_RTCERR1(StartSend, channel);
      return false;
    }
  } else {  // SEND_NOTHING
    RTC_DCHECK(send == SEND_NOTHING);
    if (engine()->voe()->base()->StopSend(channel) == -1) {
      LOG_RTCERR1(StopSend, channel);
      return false;
    }
  }

  return true;
}

bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc,
                                           bool enable,
                                           const AudioOptions* options,
                                           AudioRenderer* renderer) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  // TODO(solenberg): The state change should be fully rolled back if any one of
  //                  these calls fail.
  if (!SetLocalRenderer(ssrc, renderer)) {
    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;
  }
  if (engine()->voe()->network()->RegisterExternalTransport(id, *this) == -1) {
    LOG_RTCERR2(RegisterExternalTransport, id, this);
    engine()->voe()->base()->DeleteChannel(id);
    return -1;
  }
  return id;
}

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

bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
  RTC_DCHECK(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;
  }

  // Enable RTCP (for quality stats and feedback messages).
  if (engine()->voe()->rtp()->SetRTCPStatus(channel, true) == -1) {
    LOG_RTCERR2(SetRTCPStatus, channel, 1);
  }

  SetChannelSendRtpHeaderExtensions(channel, send_extensions_);

  // Set the local (send) SSRC.
  if (engine()->voe()->rtp()->SetLocalSSRC(channel, ssrc) == -1) {
    LOG_RTCERR2(SetLocalSSRC, channel, ssrc);
    DeleteChannel(channel);
    return false;
  }

  if (engine()->voe()->rtp()->SetRTCP_CNAME(channel, sp.cname.c_str()) == -1) {
    LOG_RTCERR2(SetRTCP_CNAME, channel, sp.cname);
    DeleteChannel(channel);
    return false;
  }

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

  // Set the current codecs to be used for the new channel. We need to do this
  // after adding the channel to send_channels_, because of how max bitrate is
  // currently being configured by SetSendCodec().
  if (!send_codecs_.empty() && !SetSendCodecs(channel, send_codecs_)) {
    RemoveSendStream(ssrc);
    return false;
  }

  // At this point the channel's local SSRC has been updated. If the channel is
  // the first send channel make sure that all the receive channels are updated
  // with the same SSRC in order to send receiver reports.
  if (send_channels_.size() == 1) {
    receiver_reports_ssrc_ = ssrc;
    for (const auto& ch : receive_channels_) {
      int recv_channel = ch.second->channel();
      if (engine()->voe()->rtp()->SetLocalSSRC(recv_channel, ssrc) != 0) {
        LOG_RTCERR2(SetLocalSSRC, ch.second->channel(), ssrc);
        return false;
      }
      engine()->voe()->base()->AssociateSendChannel(recv_channel, channel);
      LOG(LS_INFO) << "VoiceEngine channel #" << recv_channel
                   << " is associated with channel #" << channel << ".";
    }
  }

  return ChangeSend(channel, desired_send_);
}

bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  ChannelMap::iterator it = send_channels_.find(ssrc);
  if (it == send_channels_.end()) {
    LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
                    << " which doesn't exist.";
    return false;
  }

  int channel = it->second->channel();
  ChangeSend(channel, SEND_NOTHING);

  // Delete the WebRtcVoiceChannelRenderer object connected to the channel,
  // this will disconnect the audio renderer with the send channel.
  delete it->second;
  send_channels_.erase(it);

  // Clean up and delete the send channel.
  LOG(LS_INFO) << "Removing audio send stream " << ssrc
               << " with VoiceEngine channel #" << channel << ".";
  if (!DeleteChannel(channel)) {
    return false;
  }
  if (send_channels_.empty()) {
    ChangeSend(SEND_NOTHING);
  }
  return true;
}

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

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

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

  // Remove the default receive stream if one had been created with this ssrc;
  // we'll recreate it then.
  if (IsDefaultRecvStream(ssrc)) {
    RemoveRecvStream(ssrc);
  }

  if (receive_channels_.find(ssrc) != receive_channels_.end()) {
    LOG(LS_ERROR) << "Stream already exists with ssrc " << ssrc;
    return false;
  }
  RTC_DCHECK(receive_stream_params_.find(ssrc) == receive_stream_params_.end());

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

  webrtc::AudioTransport* audio_transport =
      engine()->voe()->base()->audio_transport();
  WebRtcVoiceChannelRenderer* channel_renderer =
      new WebRtcVoiceChannelRenderer(channel, audio_transport);
  receive_channels_.insert(std::make_pair(ssrc, channel_renderer));
  receive_stream_params_[ssrc] = sp;
  AddAudioReceiveStream(ssrc);

  LOG(LS_INFO) << "New audio stream " << ssrc
               << " registered to VoiceEngine channel #"
               << channel << ".";
  return true;
}

bool WebRtcVoiceMediaChannel::ConfigureRecvChannel(int channel) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());

  int send_channel = GetSendChannelId(receiver_reports_ssrc_);
  if (send_channel != -1) {
    // Associate receive channel with first send channel (so the receive channel
    // can obtain RTT from the send channel)
    engine()->voe()->base()->AssociateSendChannel(channel, send_channel);
    LOG(LS_INFO) << "VoiceEngine channel #" << channel
                 << " is associated with channel #" << send_channel << ".";
  }
  if (engine()->voe()->rtp()->SetLocalSSRC(channel,
                                           receiver_reports_ssrc_) == -1) {
    LOG_RTCERR1(SetLocalSSRC, channel);
    return false;
  }

  // Turn off all supported codecs.
  int ncodecs = engine()->voe()->codec()->NumOfCodecs();
  for (int i = 0; i < ncodecs; ++i) {
    webrtc::CodecInst voe_codec;
    if (engine()->voe()->codec()->GetCodec(i, voe_codec) != -1) {
      voe_codec.pltype = -1;
      if (engine()->voe()->codec()->SetRecPayloadType(
          channel, voe_codec) == -1) {
        LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
        return false;
      }
    }
  }

  // Only enable those configured for this channel.
  for (const auto& codec : recv_codecs_) {
    webrtc::CodecInst voe_codec;
    if (engine()->FindWebRtcCodec(codec, &voe_codec)) {
      voe_codec.pltype = codec.id;
      if (engine()->voe()->codec()->SetRecPayloadType(
          channel, voe_codec) == -1) {
        LOG_RTCERR2(SetRecPayloadType, channel, ToString(voe_codec));
        return false;
      }
    }
  }

  SetNack(channel, nack_enabled_);

  // Set RTP header extension for the new channel.
  if (!SetChannelRecvRtpHeaderExtensions(channel, receive_extensions_)) {
    return false;
  }

  SetPlayout(channel, playout_);
  return true;
}

bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;

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

  RemoveAudioReceiveStream(ssrc);
  receive_stream_params_.erase(ssrc);

  // Delete the WebRtcVoiceChannelRenderer object connected to the channel, this
  // will disconnect the audio renderer with the receive channel.
  // Cache the channel before the deletion.
  const int channel = it->second->channel();
  delete it->second;
  receive_channels_.erase(it);

  // Deregister default channel, if that's the one being destroyed.
  if (IsDefaultRecvStream(ssrc)) {
    default_recv_ssrc_ = -1;
  }

  LOG(LS_INFO) << "Removing audio stream " << ssrc
               << " with VoiceEngine channel #" << channel << ".";
  return DeleteChannel(channel);
}

bool WebRtcVoiceMediaChannel::SetLocalRenderer(uint32_t ssrc,
                                               AudioRenderer* renderer) {
  ChannelMap::iterator it = send_channels_.find(ssrc);
  if (it == send_channels_.end()) {
    if (renderer) {
      // Return an error if trying to set a valid renderer with an invalid ssrc.
      LOG(LS_ERROR) << "SetLocalRenderer failed with ssrc "<< ssrc;
      return false;
    }

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

  if (renderer) {
    it->second->Start(renderer);
  } else {
    it->second->Stop();
  }

  return true;
}

bool WebRtcVoiceMediaChannel::GetActiveStreams(
    AudioInfo::StreamList* actives) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  actives->clear();
  for (const auto& ch : receive_channels_) {
    int level = GetOutputLevel(ch.second->channel());
    if (level > 0) {
      actives->push_back(std::make_pair(ch.first, level));
    }
  }
  return true;
}

int WebRtcVoiceMediaChannel::GetOutputLevel() {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  int highest = 0;
  for (const auto& ch : receive_channels_) {
    highest = std::max(GetOutputLevel(ch.second->channel()), highest);
  }
  return highest;
}

int WebRtcVoiceMediaChannel::GetTimeSinceLastTyping() {
  int ret;
  if (engine()->voe()->processing()->TimeSinceLastTyping(ret) == -1) {
    // In case of error, log the info and continue
    LOG_RTCERR0(TimeSinceLastTyping);
    ret = -1;
  } else {
    ret *= 1000;  // We return ms, webrtc returns seconds.
  }
  return ret;
}

void WebRtcVoiceMediaChannel::SetTypingDetectionParameters(int time_window,
    int cost_per_typing, int reporting_threshold, int penalty_decay,
    int type_event_delay) {
  if (engine()->voe()->processing()->SetTypingDetectionParameters(
          time_window, cost_per_typing,
          reporting_threshold, penalty_decay, type_event_delay) == -1) {
    // In case of error, log the info and continue
    LOG_RTCERR5(SetTypingDetectionParameters, time_window,
                cost_per_typing, reporting_threshold, penalty_decay,
                type_event_delay);
  }
}

bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  if (ssrc == 0) {
    default_recv_volume_ = volume;
    if (default_recv_ssrc_ == -1) {
      return true;
    }
    ssrc = static_cast<uint32_t>(default_recv_ssrc_);
  }
  int ch_id = GetReceiveChannelId(ssrc);
  if (ch_id < 0) {
    LOG(LS_WARNING) << "Cannot find channel for ssrc:" << ssrc;
    return false;
  }

  if (-1 == engine()->voe()->volume()->SetChannelOutputVolumeScaling(ch_id,
                                                                     volume)) {
    LOG_RTCERR2(SetChannelOutputVolumeScaling, ch_id, volume);
    return false;
  }
  LOG(LS_INFO) << "SetOutputVolume to " << volume
               << " for channel " << ch_id << " and ssrc " << ssrc;
  return true;
}

bool WebRtcVoiceMediaChannel::CanInsertDtmf() {
  return dtmf_allowed_;
}

bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc,
                                         int event,
                                         int duration,
                                         int flags) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  if (!dtmf_allowed_) {
    return false;
  }

  // Send the event.
  if (flags & cricket::DF_SEND) {
    int channel = -1;
    if (ssrc == 0) {
      if (send_channels_.size() > 0) {
        channel = send_channels_.begin()->second->channel();
      }
    } else {
      channel = GetSendChannelId(ssrc);
    }
    if (channel == -1) {
      LOG(LS_WARNING) << "InsertDtmf - The specified ssrc "
                      << ssrc << " is not in use.";
      return false;
    }
    // Send DTMF using out-of-band DTMF. ("true", as 3rd arg)
    if (engine()->voe()->dtmf()->SendTelephoneEvent(
            channel, event, true, duration) == -1) {
      LOG_RTCERR4(SendTelephoneEvent, channel, event, true, duration);
      return false;
    }
  }

  // Play the event.
  if (flags & cricket::DF_PLAY) {
    // Play DTMF tone locally.
    if (engine()->voe()->dtmf()->PlayDtmfTone(event, duration) == -1) {
      LOG_RTCERR2(PlayDtmfTone, event, duration);
      return false;
    }
  }

  return true;
}

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

  uint32_t ssrc = 0;
  if (!GetRtpSsrc(packet->data(), packet->size(), &ssrc)) {
    return;
  }

  if (receive_channels_.empty()) {
    // Create new channel, which will be the default receive channel.
    StreamParams sp;
    sp.ssrcs.push_back(ssrc);
    LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc << ".";
    if (!AddRecvStream(sp)) {
      LOG(LS_WARNING) << "Could not create default receive stream.";
      return;
    }
    default_recv_ssrc_ = ssrc;
    SetOutputVolume(default_recv_ssrc_, default_recv_volume_);
  }

  // Forward packet to Call. If the SSRC is unknown we'll return after this.
  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                              packet_time.not_before);
  webrtc::PacketReceiver::DeliveryStatus delivery_result =
      call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
          reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
          webrtc_packet_time);
  if (webrtc::PacketReceiver::DELIVERY_OK != delivery_result) {
    return;
  }

  // Find the channel to send this packet to. It must exist since webrtc::Call
  // was able to demux the packet.
  int channel = GetReceiveChannelId(ssrc);
  RTC_DCHECK(channel != -1);

  // Pass it off to the decoder.
  engine()->voe()->network()->ReceivedRTPPacket(
      channel, packet->data(), packet->size(), webrtc_packet_time);
}

void WebRtcVoiceMediaChannel::OnRtcpReceived(
    rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
  RTC_DCHECK(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,
      reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
      webrtc_packet_time);

  // Sending channels need all RTCP packets with feedback information.
  // Even sender reports can contain attached report blocks.
  // Receiving channels need sender reports in order to create
  // correct receiver reports.
  int type = 0;
  if (!GetRtcpType(packet->data(), packet->size(), &type)) {
    LOG(LS_WARNING) << "Failed to parse type from received RTCP packet";
    return;
  }

  // If it is a sender report, find the receive channel that is listening.
  if (type == kRtcpTypeSR) {
    uint32_t ssrc = 0;
    if (!GetRtcpSsrc(packet->data(), packet->size(), &ssrc)) {
      return;
    }
    int recv_channel_id = GetReceiveChannelId(ssrc);
    if (recv_channel_id != -1) {
      engine()->voe()->network()->ReceivedRTCPPacket(
          recv_channel_id, packet->data(), packet->size());
    }
  }

  // SR may continue RR and any RR entry may correspond to any one of the send
  // channels. So all RTCP packets must be forwarded all send channels. VoE
  // will filter out RR internally.
  for (const auto& ch : send_channels_) {
    engine()->voe()->network()->ReceivedRTCPPacket(
        ch.second->channel(), packet->data(), packet->size());
  }
}

bool WebRtcVoiceMediaChannel::MuteStream(uint32_t ssrc, bool muted) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  int channel = GetSendChannelId(ssrc);
  if (channel == -1) {
    LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
    return false;
  }
  if (engine()->voe()->volume()->SetInputMute(channel, muted) == -1) {
    LOG_RTCERR2(SetInputMute, channel, muted);
    return false;
  }
  // 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& ch : send_channels_) {
    if (!all_muted) {
      break;
    }
    if (engine()->voe()->volume()->GetInputMute(ch.second->channel(),
                                                all_muted)) {
      LOG_RTCERR1(GetInputMute, ch.second->channel());
      return false;
    }
  }

  webrtc::AudioProcessing* ap = engine()->voe()->base()->audio_processing();
  if (ap) {
    ap->set_output_will_be_muted(all_muted);
  }
  return true;
}

// TODO(minyue): SetMaxSendBandwidth() is subject to be renamed to
// SetMaxSendBitrate() in future.
bool WebRtcVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
  LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetMaxSendBandwidth.";
  return SetSendBitrateInternal(bps);
}

bool WebRtcVoiceMediaChannel::SetSendBitrateInternal(int bps) {
  LOG(LS_INFO) << "WebRtcVoiceMediaChannel::SetSendBitrateInternal.";

  send_bitrate_setting_ = true;
  send_bitrate_bps_ = bps;

  if (!send_codec_) {
    LOG(LS_INFO) << "The send codec has not been set up yet. "
                 << "The send bitrate setting will be applied later.";
    return true;
  }

  // Bitrate is auto by default.
  // TODO(bemasc): Fix this so that if SetMaxSendBandwidth(50) is followed by
  // SetMaxSendBandwith(0), the second call removes the previous limit.
  if (bps <= 0)
    return true;

  webrtc::CodecInst codec = *send_codec_;
  bool is_multi_rate = IsCodecMultiRate(codec);

  if (is_multi_rate) {
    // If codec is multi-rate then just set the bitrate.
    codec.rate = bps;
    for (const auto& ch : send_channels_) {
      if (!SetSendCodec(ch.second->channel(), codec)) {
        LOG(LS_INFO) << "Failed to set codec " << codec.plname
                     << " to bitrate " << bps << " bps.";
        return false;
      }
    }
    return true;
  } else {
    // 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.
    if (bps < codec.rate) {
      LOG(LS_INFO) << "Failed to set codec " << codec.plname
                   << " to bitrate " << bps << " bps"
                   << ", requires at least " << codec.rate << " bps.";
      return false;
    }
    return true;
  }
}

bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());

  bool echo_metrics_on = false;
  // These can take on valid negative values, so use the lowest possible level
  // as default rather than -1.
  int echo_return_loss = -100;
  int echo_return_loss_enhancement = -100;
  // These can also be negative, but in practice -1 is only used to signal
  // insufficient data, since the resolution is limited to multiples of 4 ms.
  int echo_delay_median_ms = -1;
  int echo_delay_std_ms = -1;
  if (engine()->voe()->processing()->GetEcMetricsStatus(
          echo_metrics_on) != -1 && echo_metrics_on) {
    // TODO(ajm): we may want to use VoECallReport::GetEchoMetricsSummary
    // here, but it appears to be unsuitable currently. Revisit after this is
    // investigated: http://b/issue?id=5666755
    int erl, erle, rerl, anlp;
    if (engine()->voe()->processing()->GetEchoMetrics(
            erl, erle, rerl, anlp) != -1) {
      echo_return_loss = erl;
      echo_return_loss_enhancement = erle;
    }

    int median, std;
    float dummy;
    if (engine()->voe()->processing()->GetEcDelayMetrics(
        median, std, dummy) != -1) {
      echo_delay_median_ms = median;
      echo_delay_std_ms = std;
    }
  }

  webrtc::CallStatistics cs;
  unsigned int ssrc;
  webrtc::CodecInst codec;
  unsigned int level;

  for (const auto& ch : send_channels_) {
    const int channel = ch.second->channel();

    // Fill in the sender info, based on what we know, and what the
    // remote side told us it got from its RTCP report.
    VoiceSenderInfo sinfo;

    if (engine()->voe()->rtp()->GetRTCPStatistics(channel, cs) == -1 ||
        engine()->voe()->rtp()->GetLocalSSRC(channel, ssrc) == -1) {
      continue;
    }

    sinfo.add_ssrc(ssrc);
    sinfo.codec_name = send_codec_.get() ? send_codec_->plname : "";
    sinfo.bytes_sent = cs.bytesSent;
    sinfo.packets_sent = cs.packetsSent;
    // RTT isn't known until a RTCP report is received. Until then, VoiceEngine
    // returns 0 to indicate an error value.
    sinfo.rtt_ms = (cs.rttMs > 0) ? cs.rttMs : -1;

    // Get data from the last remote RTCP report. Use default values if no data
    // available.
    sinfo.fraction_lost = -1.0;
    sinfo.jitter_ms = -1;
    sinfo.packets_lost = -1;
    sinfo.ext_seqnum = -1;
    std::vector<webrtc::ReportBlock> receive_blocks;
    if (engine()->voe()->rtp()->GetRemoteRTCPReportBlocks(
            channel, &receive_blocks) != -1 &&
        engine()->voe()->codec()->GetSendCodec(channel, codec) != -1) {
      for (const webrtc::ReportBlock& block : receive_blocks) {
        // Lookup report for send ssrc only.
        if (block.source_SSRC == sinfo.ssrc()) {
          // Convert Q8 to floating point.
          sinfo.fraction_lost = static_cast<float>(block.fraction_lost) / 256;
          // Convert samples to milliseconds.
          if (codec.plfreq / 1000 > 0) {
            sinfo.jitter_ms = block.interarrival_jitter / (codec.plfreq / 1000);
          }
          sinfo.packets_lost = block.cumulative_num_packets_lost;
          sinfo.ext_seqnum = block.extended_highest_sequence_number;
          break;
        }
      }
    }

    // Local speech level.
    sinfo.audio_level = (engine()->voe()->volume()->
        GetSpeechInputLevelFullRange(level) != -1) ? level : -1;

    // TODO(xians): We are injecting the same APM logging to all the send
    // channels here because there is no good way to know which send channel
    // is using the APM. The correct fix is to allow the send channels to have
    // their own APM so that we can feed the correct APM logging to different
    // send channels. See issue crbug/264611 .
    sinfo.echo_return_loss = echo_return_loss;
    sinfo.echo_return_loss_enhancement = echo_return_loss_enhancement;
    sinfo.echo_delay_median_ms = echo_delay_median_ms;
    sinfo.echo_delay_std_ms = echo_delay_std_ms;
    // TODO(ajm): Re-enable this metric once we have a reliable implementation.
    sinfo.aec_quality_min = -1;
    sinfo.typing_noise_detected = typing_noise_detected_;

    info->senders.push_back(sinfo);
  }

  // Get the SSRC and stats for each receiver.
  for (const auto& ch : receive_channels_) {
    int ch_id = ch.second->channel();
    memset(&cs, 0, sizeof(cs));
    if (engine()->voe()->rtp()->GetRemoteSSRC(ch_id, ssrc) != -1 &&
        engine()->voe()->rtp()->GetRTCPStatistics(ch_id, cs) != -1 &&
        engine()->voe()->codec()->GetRecCodec(ch_id, codec) != -1) {
      VoiceReceiverInfo rinfo;
      rinfo.add_ssrc(ssrc);
      rinfo.bytes_rcvd = cs.bytesReceived;
      rinfo.packets_rcvd = cs.packetsReceived;
      // The next four fields are from the most recently sent RTCP report.
      // Convert Q8 to floating point.
      rinfo.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
      rinfo.packets_lost = cs.cumulativeLost;
      rinfo.ext_seqnum = cs.extendedMax;
      rinfo.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
      if (codec.pltype != -1) {
        rinfo.codec_name = codec.plname;
      }
      // Convert samples to milliseconds.
      if (codec.plfreq / 1000 > 0) {
        rinfo.jitter_ms = cs.jitterSamples / (codec.plfreq / 1000);
      }

      // Get jitter buffer and total delay (alg + jitter + playout) stats.
      webrtc::NetworkStatistics ns;
      if (engine()->voe()->neteq() &&
          engine()->voe()->neteq()->GetNetworkStatistics(
              ch_id, ns) != -1) {
        rinfo.jitter_buffer_ms = ns.currentBufferSize;
        rinfo.jitter_buffer_preferred_ms = ns.preferredBufferSize;
        rinfo.expand_rate =
            static_cast<float>(ns.currentExpandRate) / (1 << 14);
        rinfo.speech_expand_rate =
            static_cast<float>(ns.currentSpeechExpandRate) / (1 << 14);
        rinfo.secondary_decoded_rate =
            static_cast<float>(ns.currentSecondaryDecodedRate) / (1 << 14);
        rinfo.accelerate_rate =
            static_cast<float>(ns.currentAccelerateRate) / (1 << 14);
        rinfo.preemptive_expand_rate =
            static_cast<float>(ns.currentPreemptiveRate) / (1 << 14);
      }

      webrtc::AudioDecodingCallStats ds;
      if (engine()->voe()->neteq() &&
          engine()->voe()->neteq()->GetDecodingCallStatistics(
              ch_id, &ds) != -1) {
        rinfo.decoding_calls_to_silence_generator =
            ds.calls_to_silence_generator;
        rinfo.decoding_calls_to_neteq = ds.calls_to_neteq;
        rinfo.decoding_normal = ds.decoded_normal;
        rinfo.decoding_plc = ds.decoded_plc;
        rinfo.decoding_cng = ds.decoded_cng;
        rinfo.decoding_plc_cng = ds.decoded_plc_cng;
      }

      if (engine()->voe()->sync()) {
        int jitter_buffer_delay_ms = 0;
        int playout_buffer_delay_ms = 0;
        engine()->voe()->sync()->GetDelayEstimate(
            ch_id, &jitter_buffer_delay_ms, &playout_buffer_delay_ms);
        rinfo.delay_estimate_ms = jitter_buffer_delay_ms +
            playout_buffer_delay_ms;
      }

      // Get speech level.
      rinfo.audio_level = (engine()->voe()->volume()->
          GetSpeechOutputLevelFullRange(ch_id, level) != -1) ? level : -1;
      info->receivers.push_back(rinfo);
    }
  }

  return true;
}

void WebRtcVoiceMediaChannel::OnError(int error) {
  if (send_ == SEND_NOTHING) {
    return;
  }
  if (error == VE_TYPING_NOISE_WARNING) {
    typing_noise_detected_ = true;
  } else if (error == VE_TYPING_NOISE_OFF_WARNING) {
    typing_noise_detected_ = false;
  }
}

int WebRtcVoiceMediaChannel::GetOutputLevel(int channel) {
  unsigned int ulevel = 0;
  int ret = engine()->voe()->volume()->GetSpeechOutputLevel(channel, ulevel);
  return (ret == 0) ? static_cast<int>(ulevel) : -1;
}

int WebRtcVoiceMediaChannel::GetReceiveChannelId(uint32_t ssrc) const {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  ChannelMap::const_iterator it = receive_channels_.find(ssrc);
  if (it != receive_channels_.end()) {
    return it->second->channel();
  }
  return -1;
}

int WebRtcVoiceMediaChannel::GetSendChannelId(uint32_t ssrc) const {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  ChannelMap::const_iterator it = send_channels_.find(ssrc);
  if (it != send_channels_.end()) {
    return it->second->channel();
  }
  return -1;
}

bool WebRtcVoiceMediaChannel::GetRedSendCodec(const AudioCodec& red_codec,
    const std::vector<AudioCodec>& all_codecs, webrtc::CodecInst* send_codec) {
  // Get the RED encodings from the parameter with no name. This may
  // change based on what is discussed on the Jingle list.
  // The encoding parameter is of the form "a/b"; we only support where
  // a == b. Verify this and parse out the value into red_pt.
  // If the parameter value is absent (as it will be until we wire up the
  // signaling of this message), use the second codec specified (i.e. the
  // one after "red") as the encoding parameter.
  int red_pt = -1;
  std::string red_params;
  CodecParameterMap::const_iterator it = red_codec.params.find("");
  if (it != red_codec.params.end()) {
    red_params = it->second;
    std::vector<std::string> red_pts;
    if (rtc::split(red_params, '/', &red_pts) != 2 ||
        red_pts[0] != red_pts[1] ||
        !rtc::FromString(red_pts[0], &red_pt)) {
      LOG(LS_WARNING) << "RED params " << red_params << " not supported.";
      return false;
    }
  } else if (red_codec.params.empty()) {
    LOG(LS_WARNING) << "RED params not present, using defaults";
    if (all_codecs.size() > 1) {
      red_pt = all_codecs[1].id;
    }
  }

  // Try to find red_pt in |codecs|.
  for (const AudioCodec& codec : all_codecs) {
    if (codec.id == red_pt) {
      // If we find the right codec, that will be the codec we pass to
      // SetSendCodec, with the desired payload type.
      if (engine()->FindWebRtcCodec(codec, send_codec)) {
        return true;
      } else {
        break;
      }
    }
  }
  LOG(LS_WARNING) << "RED params " << red_params << " are invalid.";
  return false;
}

bool WebRtcVoiceMediaChannel::SetPlayout(int channel, bool playout) {
  if (playout) {
    LOG(LS_INFO) << "Starting playout for channel #" << channel;
    if (engine()->voe()->base()->StartPlayout(channel) == -1) {
      LOG_RTCERR1(StartPlayout, channel);
      return false;
    }
  } else {
    LOG(LS_INFO) << "Stopping playout for channel #" << channel;
    engine()->voe()->base()->StopPlayout(channel);
  }
  return true;
}

// Convert VoiceEngine error code into VoiceMediaChannel::Error enum.
VoiceMediaChannel::Error
    WebRtcVoiceMediaChannel::WebRtcErrorToChannelError(int err_code) {
  switch (err_code) {
    case 0:
      return ERROR_NONE;
    case VE_CANNOT_START_RECORDING:
    case VE_MIC_VOL_ERROR:
    case VE_GET_MIC_VOL_ERROR:
    case VE_CANNOT_ACCESS_MIC_VOL:
      return ERROR_REC_DEVICE_OPEN_FAILED;
    case VE_SATURATION_WARNING:
      return ERROR_REC_DEVICE_SATURATION;
    case VE_REC_DEVICE_REMOVED:
      return ERROR_REC_DEVICE_REMOVED;
    case VE_RUNTIME_REC_WARNING:
    case VE_RUNTIME_REC_ERROR:
      return ERROR_REC_RUNTIME_ERROR;
    case VE_CANNOT_START_PLAYOUT:
    case VE_SPEAKER_VOL_ERROR:
    case VE_GET_SPEAKER_VOL_ERROR:
    case VE_CANNOT_ACCESS_SPEAKER_VOL:
      return ERROR_PLAY_DEVICE_OPEN_FAILED;
    case VE_RUNTIME_PLAY_WARNING:
    case VE_RUNTIME_PLAY_ERROR:
      return ERROR_PLAY_RUNTIME_ERROR;
    case VE_TYPING_NOISE_WARNING:
      return ERROR_REC_TYPING_NOISE_DETECTED;
    default:
      return VoiceMediaChannel::ERROR_OTHER;
  }
}

bool WebRtcVoiceMediaChannel::SetHeaderExtension(ExtensionSetterFunction setter,
    int channel_id, const RtpHeaderExtension* extension) {
  bool enable = false;
  int id = 0;
  std::string uri;
  if (extension) {
    enable = true;
    id = extension->id;
    uri = extension->uri;
  }
  if ((engine()->voe()->rtp()->*setter)(channel_id, enable, id) != 0) {
    LOG_RTCERR4(*setter, uri, channel_id, enable, id);
    return false;
  }
  return true;
}

void WebRtcVoiceMediaChannel::RecreateAudioReceiveStreams() {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  for (const auto& it : receive_channels_) {
    RemoveAudioReceiveStream(it.first);
  }
  for (const auto& it : receive_channels_) {
    AddAudioReceiveStream(it.first);
  }
}

void WebRtcVoiceMediaChannel::AddAudioReceiveStream(uint32_t ssrc) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  WebRtcVoiceChannelRenderer* channel = receive_channels_[ssrc];
  RTC_DCHECK(channel != nullptr);
  RTC_DCHECK(receive_streams_.find(ssrc) == receive_streams_.end());
  webrtc::AudioReceiveStream::Config config;
  config.rtp.remote_ssrc = ssrc;
  // Only add RTP extensions if we support combined A/V BWE.
  config.rtp.extensions = recv_rtp_extensions_;
  config.combined_audio_video_bwe =
      options_.combined_audio_video_bwe.GetWithDefaultIfUnset(false);
  config.voe_channel_id = channel->channel();
  config.sync_group = receive_stream_params_[ssrc].sync_label;
  webrtc::AudioReceiveStream* s = call_->CreateAudioReceiveStream(config);
  receive_streams_.insert(std::make_pair(ssrc, s));
}

void WebRtcVoiceMediaChannel::RemoveAudioReceiveStream(uint32_t ssrc) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  auto stream_it = receive_streams_.find(ssrc);
  if (stream_it != receive_streams_.end()) {
    call_->DestroyAudioReceiveStream(stream_it->second);
    receive_streams_.erase(stream_it);
  }
}

bool WebRtcVoiceMediaChannel::SetRecvCodecsInternal(
    const std::vector<AudioCodec>& new_codecs) {
  RTC_DCHECK(thread_checker_.CalledOnValidThread());
  for (const AudioCodec& codec : new_codecs) {
    webrtc::CodecInst voe_codec;
    if (engine()->FindWebRtcCodec(codec, &voe_codec)) {
      LOG(LS_INFO) << ToString(codec);
      voe_codec.pltype = codec.id;
      for (const auto& ch : receive_channels_) {
        if (engine()->voe()->codec()->SetRecPayloadType(
                ch.second->channel(), voe_codec) == -1) {
          LOG_RTCERR2(SetRecPayloadType, ch.second->channel(),
                      ToString(voe_codec));
          return false;
        }
      }
    } else {
      LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
      return false;
    }
  }
  return true;
}

}  // namespace cricket

#endif  // HAVE_WEBRTC_VOICE