webrtc 关键模块创建的时机

一 webrtc 各个模块创建的时机

1 CreatePeerConnectionFactory

1.1.1 创建的模块

TaskQueueFactory
CallFactory
CompositeMediaEngine
WebRtcVoiceEngine
WebRtcVideoEngine

1.1.2 关键代码

/*
api/create_peerconnection_factory.h
CreatePeerConnectionFactory 函数

PeerConnectionFactoryDependencies dependencies;
dependencies.task_queue_factory = CreateDefaultTaskQueueFactory();
dependencies.call_factory = CreateCallFactory();
创建：

dependencies.media_engine = cricket::CreateMediaEngine(std::move(media_dependencies));
创建了：
CompositeMediaEngine
WebRtcVoiceEngine
WebRtcVideoEngine
*/

2PeerConnectionFactory::Initialize

1.2.1 创建的模块

BasicNetworkManager
BasicPacketSocketFactory
ChannelManager

1.2.2 关键代码

/*
// 关键代码
default_network_manager_.reset(new rtc::BasicNetworkManager());	
default_socket_factory_.reset(
      new rtc::BasicPacketSocketFactory(network_thread_));
	  
channel_manager_ = std::make_unique<cricket::ChannelManager>(
	  std::move(media_engine_), std::make_unique<cricket::RtpDataEngine>(),
	  worker_thread_, network_thread_);

// 收集音频编码器
channelManager::Init -> media_engine_->Init  -> voice().Init();
*/

3 CreatePeerConnection

1.3.1 创建的模块

Call
RtpTransportControllerSend
BasicPortAllocator
PeerConnection

1.3.2 关键代码

/*
PeerConnectionFactory::CreatePeerConnection

创建call
    PeerConnectionFactory::CreateCall_w
    CallFactory::CreateCall
    Call::Create
创建：RtpTransportControllerSend
创建 BasicPortAllocator
*/

4 PeerConnection::Initialize

1.4.1 创建的模块

JsepTransportController
WebRtcSessionDescriptionFactory

1.4.2 关键代码

// 创建JsepTransportController
transport_controller_.reset(new JsepTransportController(
      signaling_thread(), network_thread(), port_allocator_.get(),
      async_resolver_factory_.get(), config));
// 创建WebRtcSessionDescriptionFactory
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
  signaling_thread(), channel_manager(), this, session_id(),
  std::move(dependencies.cert_generator), certificate, &ssrc_generator_));

5PeerConnection::AddTransceiver

1.5.1 创建的模块

RtpSenderInternal
RtpReceiverInternal
RtpTransceiver

1.5.2 关键代码

/*
PeerConnection::CreateSender              RtpSenderInternal
PeerConnection::CreateReceiver            RtpReceiverInternal
PeerConnection::CreateAndAddTransceiver   RtpTransceiver
*/

6PeerConnection::SetLocalDescription

6.1PushdownTransportDescription

1.6.1.1 创建的模块

DtlsTransportInternal 
RtpTransport 
SrtpTransport
DtlsSrtpTransport 
RtpTransportInternal

1.6.1.2 关键代码

/*
创建：
PeerConnection::PushdownTransportDescription  
transport_controller_->SetLocalDescription
JsepTransportController::ApplyDescription_n 
JsepTransportController::MaybeCreateJsepTransport
*/

6.2UpdateTransceiversAndDataChannels

1.6.2.1 创建的模块

VoiceMediaChannel
VoiceChannel
或
VideoMediaChannel
VideoChannel

1.6.2.2 关键代码

/*
创建：
VoiceMediaChannel
VoiceChannel

PeerConnection::UpdateTransceiversAndDataChannels		
PeerConnection::UpdateTransceiverChannel
PeerConnection::CreateVoiceChannel
channel_manager()->CreateVoiceChannel
	
关键代码：			
创建：VoiceMediaChannel
VoiceMediaChannel* media_channel = media_engine_->voice().CreateMediaChannel(call, media_config, options, crypto_options);
创建：VoiceChannel    成员有：VoiceMediaChannel  rtp_transport（MaybeCreateJsepTransport 创建的rtp_transport）

或

创建：
VideoMediaChannel
VideoChannel

PeerConnection::UpdateTransceiversAndDataChannels		
PeerConnection::UpdateTransceiverChannel
PeerConnection::CreateVideoChannel	  
channel_manager()->CreateVideoChannel
	
关键代码：			
创建 VideoMediaChannel
VideoMediaChannel* media_channel = media_engine_->video().CreateMediaChannel
创建：VideoChannel 成员有：VideoMediaChannel  rtp_transport（MaybeCreateJsepTransport 创建的rtp_transport）
*/

6.3 UpdateSessionState

1.6.3.1 创建的模块

WebRtcAudioSendStream
webrtc::AudioSendStream
ChannelSend
WebRtcVideoSendStream
VideoSendStream

1.6.3.2 关键代码

/*
创建 WebRtcAudioSendStream
创建 webrtc::AudioSendStream
创建：ChannelSend

PeerConnection::UpdateSessionState
PeerConnection::PushdownMediaDescription
BaseChannel::SetLocalContent
VoiceChannel::SetLocalContent_w
BaseChannel::UpdateLocalStreams_w
WebRtcVoiceMediaChannel::AddSendStream
WebRtcVoiceMediaChannel::WebRtcAudioSendStream
AudioSendStream::AudioSendStream  创建 webrtc::AudioSendStream
voe::CreateChannelSend 创建：ChannelSend

或

创建：WebRtcVideoSendStream
创建： VideoSendStream

PeerConnection::UpdateSessionState
PeerConnection::PushdownMediaDescription
BaseChannel::SetLocalContent
VoiceChannel::SetLocalContent_w
BaseChannel::UpdateLocalStreams_w
WebRtcVideoChannel::AddSendStream
WebRtcVideoChannel::WebRtcVideoSendStream
备注：
1 创建 WebRtcVideoSendStream
2 创建 webrtc::VideoSendStream::Config，把自己传递给 config.send_transport

VideoChannel::SetRemoteContent_w
WebRtcVideoChannel::SetSendParameters
WebRtcVideoChannel::ApplyChangedParams
WebRtcVideoChannel::WebRtcVideoSendStream::SetSendParameter
WebRtcVideoChannel::WebRtcVideoSendStream::SetCodec
WebRtcVideoChannel::WebRtcVideoSendStream::RecreateWebRtcStream
Call::CreateVideoSendStream (参数有webrtc::VideoSendStream::Config)
创建 VideoSendStream
创建 VideoSendStreamImpl（参数有webrtc::VideoSendStream::Config）
创建 rtp_video_sender 其中 参数send_transport 就是 WebRtcVideoChannel
备注：将来分片rtp数据就是通过WebRtcVideoChannel 发送出去
*/