ADM与播放声音

播放声音的流程

重要流程

源码分析-音频播放流程

在h:\webrtc-20210315\webrtc-20210315\webrtc\webrtc-checkout\src\media\engine\webrtc_voice_engine.cc
WebRtcVoiceMediaChannel::AddRecvStream 打断点调试

WebRtcVoiceMediaChannel::AddRecvStream

bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
  TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::AddRecvStream");
  RTC_DCHECK_RUN_ON(worker_thread_);
***
  // Create a new channel for receiving audio data.
    // 创建一个新的通道去接收音频数据
    // 键值<ssrc, WebRtcAudioReceiveStream>
  recv_streams_.insert(std::make_pair(
      ssrc, new WebRtcAudioReceiveStream(
                ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
                recv_nack_enabled_, sp.stream_ids(), recv_rtp_extensions_,
                call_, this, engine()->decoder_factory_, decoder_map_,
                codec_pair_id_, engine()->audio_jitter_buffer_max_packets_,
                engine()->audio_jitter_buffer_fast_accelerate_,
                engine()->audio_jitter_buffer_min_delay_ms_,
                engine()->audio_jitter_buffer_enable_rtx_handling_,
                unsignaled_frame_decryptor_, crypto_options_, nullptr)));
  recv_streams_[ssrc]->SetPlayout(playout_);
  return true;
}

调用堆栈

recvstreams[ssrc]->SetPlayout(playout_);

SetPlayout(bool playout)

  void SetPlayout(bool playout) {
    RTC_DCHECK_RUN_ON(&worker_thread_checker_);
    RTC_DCHECK(stream_);
    if (playout) {
      stream_->Start();
    } else {
      stream_->Stop();
    }
  }

stream->Start();
stream->Stop();

AudioReceiveStream::Start

void AudioReceiveStream::Start() {
  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
  if (playing_) {
    return;
  }
  channel_receive_->StartPlayout();
  playing_ = true;
  audio_state()->AddReceivingStream(this);
}
void AudioReceiveStream::Stop() {
  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
  if (!playing_) {
    return;
  }
  channel_receive_->StopPlayout();
  playing_ = false;
  audio_state()->RemoveReceivingStream(this);
}

audio_state()->AddReceivingStream(this);

AudioState::AddReceivingStream

void AudioState::AddReceivingStream(webrtc::AudioReceiveStream* stream) {
  RTC_DCHECK(thread_checker_.IsCurrent());
  RTC_DCHECK_EQ(0, receiving_streams_.count(stream));
    // 首先将该流插入到自己的管理中
  receiving_streams_.insert(stream);
    // 然后添加到audio_mixer的源中，后面就可以用它来进行混音了
  if (!config_.audio_mixer->AddSource(
          static_cast<internal::AudioReceiveStream*>(stream))) {
    RTC_DLOG(LS_ERROR) << "Failed to add source to mixer.";
  }
  // Make sure playback is initialized; start playing if enabled.
  UpdateNullAudioPollerState();
  // 判断是否正在播放，没有则调用adm->InitPlayout
  auto* adm = config_.audio_device_module.get();
  if (!adm->Playing()) {
    if (adm->InitPlayout() == 0) {
      if (playout_enabled_) {
        adm->StartPlayout();
      }
    } else {
      RTC_DLOG_F(LS_ERROR) << "Failed to initialize playout.";
    }
  }
}

adm->InitPlayout

AudioDeviceModuleImpl::InitPlayout

int32_t AudioDeviceModuleImpl::InitPlayout() {
  RTC_LOG(INFO) << __FUNCTION__;
  CHECKinitialized_();
  if (PlayoutIsInitialized()) {
    return 0;
  }
    // 又回到 AudioDeviceWindowsCore::InitPlayout
  int32_t result = audio_device_->InitPlayout();
  RTC_LOG(INFO) << "output: " << result;
  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
                        static_cast<int>(result == 0));
  return result;
}