前言

代码分析

H:\webrtc-20210315\webrtc-20210315\webrtc\webrtc-checkout\src\media\engine\webrtc_voice_engine.cc

WebRtcVoiceEngine::Init

void WebRtcVoiceEngine::Init() {
  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
  RTC_LOG(LS_INFO) << "WebRtcVoiceEngine::Init";
  //1、 TaskQueue expects to be created/destroyed on the same thread.
  low_priority_worker_queue_.reset(
      new rtc::TaskQueue(task_queue_factory_->CreateTaskQueue(
          "rtc-low-prio", webrtc::TaskQueueFactory::Priority::LOW)));
  //2、 收集音频编码器
  RTC_LOG(LS_VERBOSE) << "Supported send codecs in order of preference:";
  send_codecs_ = CollectCodecs(encoder_factory_->GetSupportedEncoders());
  for (const AudioCodec& codec : send_codecs_) {
    RTC_LOG(LS_VERBOSE) << ToString(codec);
  }
  //3、 收集音频解码器
  RTC_LOG(LS_VERBOSE) << "Supported recv codecs in order of preference:";
  recv_codecs_ = CollectCodecs(decoder_factory_->GetSupportedDecoders());
  for (const AudioCodec& codec : recv_codecs_) {
    RTC_LOG(LS_VERBOSE) << ToString(codec);
  }
 // 4、创建ADM
#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
  // No ADM supplied? Create a default one.
  if (!adm_) {
    adm_ = webrtc::AudioDeviceModule::Create(
        webrtc::AudioDeviceModule::kPlatformDefaultAudio, task_queue_factory_);
  }
#endif  // WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE
  RTC_CHECK(adm());
  // 5、 初始化adm
  webrtc::adm_helpers::Init(adm());
  //6、 Set up AudioState.
  {
    webrtc::AudioState::Config config;
    if (audio_mixer_) {
      config.audio_mixer = audio_mixer_;
    } else {
      config.audio_mixer = webrtc::AudioMixerImpl::Create();
    }
    config.audio_processing = apm_;
    config.audio_device_module = adm_;
    if (audio_frame_processor_)
      config.async_audio_processing_factory =
          new rtc::RefCountedObject<webrtc::AsyncAudioProcessing::Factory>(
              *audio_frame_processor_, *task_queue_factory_);
    audio_state_ = webrtc::AudioState::Create(config);
  }
  //7、 Connect the ADM to our audio path.
  adm()->RegisterAudioCallback(audio_state()->audio_transport());
  //8、 Set default engine options.
  {
    AudioOptions options;
    options.echo_cancellation = true;
    options.auto_gain_control = true;
#if defined(WEBRTC_IOS)
    // On iOS, VPIO provides built-in NS.
    options.noise_suppression = false;
    options.typing_detection = false;
#else
    options.noise_suppression = true;
    options.typing_detection = true;
#endif
    options.experimental_ns = false;
    options.highpass_filter = true;
    options.stereo_swapping = false;
    options.audio_jitter_buffer_max_packets = 200;
    options.audio_jitter_buffer_fast_accelerate = false;
    options.audio_jitter_buffer_min_delay_ms = 0;
    options.audio_jitter_buffer_enable_rtx_handling = false;
    options.experimental_agc = false;
    options.residual_echo_detector = true;
    bool error = ApplyOptions(options);
    RTC_DCHECK(error);
  }
  initialized_ = true;
}

—》
sendcodecs = CollectCodecs(encoderfactory->GetSupportedEncoders());
recvcodecs = CollectCodecs(decoderfactory->GetSupportedDecoders());

WebRtcVoiceEngine::CollectCodecs

std::vector<AudioCodec> WebRtcVoiceEngine::CollectCodecs(
    const std::vector<webrtc::AudioCodecSpec>& specs) const {
  PayloadTypeMapper mapper;
  std::vector<AudioCodec> out;
  // Only generate CN payload types for these clockrates:
  std::map<int, bool, std::greater<int>> generate_cn = {
      {8000, false}, {16000, false}, {32000, false}};
  // Only generate telephone-event payload types for these clockrates:
  std::map<int, bool, std::greater<int>> generate_dtmf = {
      {8000, false}, {16000, false}, {32000, false}, {48000, false}};
  auto map_format = [&mapper](const webrtc::SdpAudioFormat& format,
                              std::vector<AudioCodec>* out) {
    absl::optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
    if (opt_codec) {
      if (out) {
        out->push_back(*opt_codec);
      }
    } else {
      RTC_LOG(LS_ERROR) << "Unable to assign payload type to format: "
                        << rtc::ToString(format);
    }
    return opt_codec;
  };
  for (const auto& spec : specs) {
    // We need to do some extra stuff before adding the main codecs to out.
      // 调用上面定义的map_format函数，来将format格式转为AudioCodec
    absl::optional<AudioCodec> opt_codec = map_format(spec.format, nullptr);
    if (opt_codec) {
      AudioCodec& codec = *opt_codec;
      if (spec.info.supports_network_adaption) {
        codec.AddFeedbackParam(
            FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
      }
      if (spec.info.allow_comfort_noise) {
        // Generate a CN entry if the decoder allows it and we support the
        // clockrate.
        auto cn = generate_cn.find(spec.format.clockrate_hz);
        if (cn != generate_cn.end()) {
          cn->second = true;
        }
      }
      // Generate a telephone-event entry if we support the clockrate.
      auto dtmf = generate_dtmf.find(spec.format.clockrate_hz);
      if (dtmf != generate_dtmf.end()) {
        dtmf->second = true;
      }
      out.push_back(codec);
      if (codec.name == kOpusCodecName && audio_red_for_opus_trial_enabled_) {
        map_format({kRedCodecName, 48000, 2}, &out);
      }
    }
  }
  // Add CN codecs after "proper" audio codecs.
  for (const auto& cn : generate_cn) {
    if (cn.second) {
      map_format({kCnCodecName, cn.first, 1}, &out);
    }
  }
  // Add telephone-event codecs last.
  for (const auto& dtmf : generate_dtmf) {
    if (dtmf.second) {
      map_format({kDtmfCodecName, dtmf.first, 1}, &out);
    }
  }
  return out;
}

返回out的值，可以看到有14个编码器