文章目录
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- [一 webrtc 各个模块创建的时机](#一 webrtc 各个模块创建的时机)
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- [1 CreatePeerConnectionFactory](#1 CreatePeerConnectionFactory)
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- [1.1.1 创建的模块](#1.1.1 创建的模块)
- [1.1.2 关键代码](#1.1.2 关键代码)
- 2PeerConnectionFactory::Initialize
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- [1.2.1 创建的模块](#1.2.1 创建的模块)
- [1.2.2 关键代码](#1.2.2 关键代码)
- [3 CreatePeerConnection](#3 CreatePeerConnection)
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- [1.3.1 创建的模块](#1.3.1 创建的模块)
- [1.3.2 关键代码](#1.3.2 关键代码)
- [4 PeerConnection::Initialize](#4 PeerConnection::Initialize)
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- [1.4.1 创建的模块](#1.4.1 创建的模块)
- [1.4.2 关键代码](#1.4.2 关键代码)
- 5PeerConnection::AddTransceiver
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- [1.5.1 创建的模块](#1.5.1 创建的模块)
- [1.5.2 关键代码](#1.5.2 关键代码)
- 6PeerConnection::SetLocalDescription
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一 webrtc 各个模块创建的时机
1 CreatePeerConnectionFactory
1.1.1 创建的模块
c
复制代码
TaskQueueFactory
CallFactory
CompositeMediaEngine
WebRtcVoiceEngine
WebRtcVideoEngine
1.1.2 关键代码
c
复制代码
/*
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
c
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BasicNetworkManager
BasicPacketSocketFactory
ChannelManager
1.2.2 关键代码
c
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/*
// 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
c
复制代码
Call
RtpTransportControllerSend
BasicPortAllocator
PeerConnection
1.3.2 关键代码
c
复制代码
/*
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
c
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JsepTransportController
WebRtcSessionDescriptionFactory
1.4.2 关键代码
c
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// 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
c
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RtpSenderInternal
RtpReceiverInternal
RtpTransceiver
1.5.2 关键代码
c
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/*
PeerConnection::CreateSender RtpSenderInternal
PeerConnection::CreateReceiver RtpReceiverInternal
PeerConnection::CreateAndAddTransceiver RtpTransceiver
*/
6PeerConnection::SetLocalDescription
6.1PushdownTransportDescription
1.6.1.1 创建的模块
c
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DtlsTransportInternal
RtpTransport
SrtpTransport
DtlsSrtpTransport
RtpTransportInternal
1.6.1.2 关键代码
c
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/*
创建：
PeerConnection::PushdownTransportDescription
transport_controller_->SetLocalDescription
JsepTransportController::ApplyDescription_n
JsepTransportController::MaybeCreateJsepTransport
*/
6.2UpdateTransceiversAndDataChannels
1.6.2.1 创建的模块
c
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VoiceMediaChannel
VoiceChannel
或
VideoMediaChannel
VideoChannel
1.6.2.2 关键代码
c
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/*
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
c
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WebRtcAudioSendStream
webrtc::AudioSendStream
ChannelSend
WebRtcVideoSendStream
VideoSendStream
1.6.3.2 关键代码
c
复制代码
/*
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
*/