WebRTC Audio 接收和发送的关键过程

本文基于 WebRTC 中的示例应用 peerconnection_client 分析 WebRTC Audio 接收和发送的关键过程。首先是发送的过程，然后是接收的过程。

1、创建 webrtc::AudioState

应用程序择机初始化 PeerConnectionFactory：

#0  Init () at webrtc/src/pc/channel_manager.cc:121
#1  Initialize () at webrtc/src/pc/peer_connection_factory.cc:139
#6  webrtc::CreateModularPeerConnectionFactory(webrtc::PeerConnectionFactoryDependencies) () at webrtc/src/pc/peer_connection_factory.cc:55
#7  webrtc::CreatePeerConnectionFactory(rtc::Thread*, rtc::Thread*, rtc::Thread*, rtc::scoped_refptr<webrtc::AudioDeviceModule>, rtc::scoped_refptr<webrtc::AudioEncoderFactory>, rtc::scoped_refptr<webrtc::AudioDecoderFactory>, std::__1::unique_ptr<webrtc::VideoEncoderFactory, std::__1::default_delete<webrtc::VideoEncoderFactory> >, std::__1::unique_ptr<webrtc::VideoDecoderFactory, std::__1::default_delete<webrtc::VideoDecoderFactory> >, rtc::scoped_refptr<webrtc::AudioMixer>, rtc::scoped_refptr<webrtc::AudioProcessing>) () at webrtc/src/api/create_peerconnection_factory.cc:65
#8  InitializePeerConnection () at webrtc/src/examples/peerconnection/client/conductor.cc:132
#9  ConnectToPeer () at webrtc/src/examples/peerconnection/client/conductor.cc:422
#10 OnRowActivated () at webrtc/src/examples/peerconnection/client/linux/main_wnd.cc:433
#11 (anonymous namespace)::OnRowActivatedCallback(_GtkTreeView*, _GtkTreePath*, _GtkTreeViewColumn*, void*) () at webrtc/src/examples/peerconnection/client/linux/main_wnd.cc:70

由 Conductor::InitializePeerConnection() 的代码可知，PeerConnectionFactory 竟然是随同 peer connection一起创建的。

在 webrtc/src/pc/channel_manager.cc 文件里定义的 ChannelManager::Init() 函数中会在另一个线程中起一个 task，调用 media_engine_->Init()，完成媒体引擎的初始化：

bool ChannelManager::Init() {
  RTC_DCHECK(!initialized_);
  if (initialized_) {
    return false;
  }
  RTC_DCHECK(network_thread_);
  RTC_DCHECK(worker_thread_);
  if (!network_thread_->IsCurrent()) {
    // Do not allow invoking calls to other threads on the network thread.
    network_thread_->Invoke<void>(
        RTC_FROM_HERE, [&] { network_thread_->DisallowBlockingCalls(); });
  }
​
  if (media_engine_) {
    initialized_ = worker_thread_->Invoke<bool>(
        RTC_FROM_HERE, [&] { return media_engine_->Init(); });
    RTC_DCHECK(initialized_);
  } else {
    initialized_ = true;
  }
  return initialized_;
}

媒体引擎初始化过程中，将会创建 AudioState：

#0  webrtc::AudioState::Create(webrtc::AudioState::Config const&) () at webrtc/src/audio/audio_state.cc:188
#1  Init () at webrtc/src/media/engine/webrtc_voice_engine.cc:260
#2  cricket::CompositeMediaEngine::Init() () at webrtc/src/media/base/media_engine.cc:155
#3  cricket::ChannelManager::Init()::$_3::operator()() const () at webrtc/src/pc/channel_manager.cc:135

AudioState 伴随着 PeerConnectionFactory 、ChannelManager 和 MediaEngine 的创建及初始化一起创建。

2、创建 WebRTC Call

应用程序根据需要创建 peer connection：

#0  CreatePeerConnection () at webrtc/src/pc/peer_connection_factory.cc:240
#1  webrtc::PeerConnectionFactory::CreatePeerConnection(webrtc::PeerConnectionInterface::RTCConfiguration const&, std::__1::unique_ptr<cricket::PortAllocator, std::__1::default_delete<cricket::PortAllocator> >, std::__1::unique_ptr<rtc::RTCCertificateGeneratorInterface, std::__1::default_delete<rtc::RTCCertificateGeneratorInterface> >, webrtc::PeerConnectionObserver*) () at webrtc/src/pc/peer_connection_factory.cc:233
#7  CreatePeerConnection () at webrtc/src/examples/peerconnection/client/conductor.cc:184
#8  InitializePeerConnection () at webrtc/src/examples/peerconnection/client/conductor.cc:148
#9  ConnectToPeer () at webrtc/src/examples/peerconnection/client/conductor.cc:422
#10 OnRowActivated () at webrtc/src/examples/peerconnection/client/linux/main_wnd.cc:433
#11 (anonymous namespace)::OnRowActivatedCallback(_GtkTreeView*, _GtkTreePath*, _GtkTreeViewColumn*, void*) () at webrtc/src/examples/peerconnection/client/linux/main_wnd.cc:70

PeerConnectionFactory::CreatePeerConnection() 在创建 connection 时，会在另一个线程中起一个task 来创建 Call：

rtc::scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(
    const PeerConnectionInterface::RTCConfiguration& configuration,
    PeerConnectionDependencies dependencies) {
  RTC_DCHECK(signaling_thread_->IsCurrent());
​
  // Set internal defaults if optional dependencies are not set.
  if (!dependencies.cert_generator) {
    dependencies.cert_generator =
        absl::make_unique<rtc::RTCCertificateGenerator>(signaling_thread_,
                                                        network_thread_);
  }
  if (!dependencies.allocator) {
    network_thread_->Invoke<void>(RTC_FROM_HERE, [this, &configuration,
                                                  &dependencies]() {
      dependencies.allocator = absl::make_unique<cricket::BasicPortAllocator>(
          default_network_manager_.get(), default_socket_factory_.get(),
          configuration.turn_customizer);
    });
  }
​
  // TODO(zstein): Once chromium injects its own AsyncResolverFactory, set
  // |dependencies.async_resolver_factory| to a new
  // |rtc::BasicAsyncResolverFactory| if no factory is provided.
​
  network_thread_->Invoke<void>(
      RTC_FROM_HERE,
      rtc::Bind(&cricket::PortAllocator::SetNetworkIgnoreMask,
                dependencies.allocator.get(), options_.network_ignore_mask));
​
  std::unique_ptr<RtcEventLog> event_log =
      worker_thread_->Invoke<std::unique_ptr<RtcEventLog>>(
          RTC_FROM_HERE,
          rtc::Bind(&PeerConnectionFactory::CreateRtcEventLog_w, this));
​
  std::unique_ptr<Call> call = worker_thread_->Invoke<std::unique_ptr<Call>>(
      RTC_FROM_HERE,
      rtc::Bind(&PeerConnectionFactory::CreateCall_w, this, event_log.get()));
​
  rtc::scoped_refptr<PeerConnection> pc(
      new rtc: