webrtc Android源码分析一

nativeCreateVideoSource

初始化

PeerConnectionFactory(pc/peerconnectionfactory) 创建PeerConnection方法中:

rtc::scoped_refptr<PeerConnectionInterface>
PeerConnectionFactory::CreatePeerConnection(const PeerConnectionInterface::RTCConfiguration& configuration,std::unique_ptr<cricket::PortAllocator> allocator,std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,PeerConnectionObserver* observer) {RTC_DCHECK(signaling_thread_->IsCurrent());if (!cert_generator.get()) {// No certificate generator specified, use the default one.cert_generator.reset(new rtc::RTCCertificateGenerator(signaling_thread_, network_thread_));}if (!allocator) {allocator.reset(new cricket::BasicPortAllocator(default_network_manager_.get(), default_socket_factory_.get()));}network_thread_->Invoke<void>(RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::SetNetworkIgnoreMask,allocator.get(), options_.network_ignore_mask));rtc::scoped_refptr<PeerConnection> pc(new rtc::RefCountedObject<PeerConnection>(this));if (!pc->Initialize(configuration, std::move(allocator),std::move(cert_generator), observer)) {return nullptr;}return PeerConnectionProxy::Create(signaling_thread(), pc);
}

构造PeerConnection对象pc,并调用初始化方法Initialize,Initialize中:

ool PeerConnection::Initialize(const PeerConnectionInterface::RTCConfiguration& configuration,std::unique_ptr<cricket::PortAllocator> allocator,std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,PeerConnectionObserver* observer) {TRACE_EVENT0("webrtc", "PeerConnection::Initialize");if (!allocator) {LOG(LS_ERROR) << "PeerConnection initialized without a PortAllocator? "<< "This shouldn't happen if using PeerConnectionFactory.";return false;}if (!observer) {// TODO(deadbeef): Why do we do this?LOG(LS_ERROR) << "PeerConnection initialized without a "<< "PeerConnectionObserver";return false;}observer_ = observer;port_allocator_ = std::move(allocator);// The port allocator lives on the network thread and should be initialized// there.if (!network_thread()->Invoke<bool>(RTC_FROM_HERE, rtc::Bind(&PeerConnection::InitializePortAllocator_n,this, configuration))) {return false;}// Call must be constructed on the worker thread.factory_->worker_thread()->Invoke<void>(RTC_FROM_HERE, rtc::Bind(&PeerConnection::CreateCall_w,this));session_.reset(new WebRtcSession(call_.get(), factory_->channel_manager(), configuration.media_config,event_log_.get(),factory_->network_thread(),factory_->worker_thread(), factory_->signaling_thread(),port_allocator_.get(),std::unique_ptr<cricket::TransportController>(factory_->CreateTransportController(port_allocator_.get(),configuration.redetermine_role_on_ice_restart)),
#ifdef HAVE_SCTPstd::unique_ptr<cricket::SctpTransportInternalFactory>(new cricket::SctpTransportFactory(factory_->network_thread()))
#elsenullptr
#endif));stats_.reset(new StatsCollector(this));stats_collector_ = RTCStatsCollector::Create(this);// Initialize the WebRtcSession. It creates transport channels etc.if (!session_->Initialize(factory_->options(), std::move(cert_generator),configuration)) {return false;}// Register PeerConnection as receiver of local ice candidates.// All the callbacks will be posted to the application from PeerConnection.session_->RegisterIceObserver(this);session_->SignalState.connect(this, &PeerConnection::OnSessionStateChange);session_->SignalVoiceChannelCreated.connect(this, &PeerConnection::OnVoiceChannelCreated);session_->SignalVoiceChannelDestroyed.connect(this, &PeerConnection::OnVoiceChannelDestroyed);session_->SignalVideoChannelCreated.connect(this, &PeerConnection::OnVideoChannelCreated);session_->SignalVideoChannelDestroyed.connect(this, &PeerConnection::OnVideoChannelDestroyed);session_->SignalDataChannelCreated.connect(this, &PeerConnection::OnDataChannelCreated);session_->SignalDataChannelDestroyed.connect(this, &PeerConnection::OnDataChannelDestroyed);session_->SignalDataChannelOpenMessage.connect(this, &PeerConnection::OnDataChannelOpenMessage);configuration_ = configuration;return true;
}

调用CreateCall_w创建call对象:

void PeerConnection::CreateCall_w() {RTC_DCHECK(!call_);const int kMinBandwidthBps = 30000;const int kStartBandwidthBps = 300000;const int kMaxBandwidthBps = 2000000;webrtc::Call::Config call_config(event_log_.get());call_config.audio_state =factory_->channel_manager() ->media_engine()->GetAudioState();call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;call_.reset(webrtc::Call::Create(call_config));
}

使用call对象以及PeerConnectionFactory中channel_manager(PeerConnectionFactory中Initialize中创建)构造WebRtcSession对象session_,调用Initialize方法初始化session_,初始化session_槽函数等.session_初始化方法中创建WebRtcSessionDescriptionFactory对象webrtc_session_desc_factory_.

创建Channel

WebRtcSession::SetLocalDescription:

bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc,std::string* err_desc) {RTC_DCHECK(signaling_thread()->IsCurrent());// Takes the ownership of |desc| regardless of the result.std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);// Validate SDP.if (!ValidateSessionDescription(desc, cricket::CS_LOCAL, err_desc)) {return false;}// Update the initial_offerer flag if this session is the initial_offerer.Action action = GetAction(desc->type());if (state() == STATE_INIT && action == kOffer) {initial_offerer_ = true;transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);}if (action == kAnswer) {current_local_description_.reset(desc_temp.release());pending_local_description_.reset(nullptr);current_remote_description_.reset(pending_remote_description_.release());} else {pending_local_description_.reset(desc_temp.release());}// Transport and Media channels will be created only when offer is set.if (action == kOffer && !CreateChannels(local_description()->description())) {// TODO(mallinath) - Handle CreateChannel failure, as new local description// is applied. Restore back to old description.return BadLocalSdp(desc->type(), kCreateChannelFailed, err_desc);}// Remove unused channels if MediaContentDescription is rejected.RemoveUnusedChannels(local_description()->description());if (!UpdateSessionState(action, cricket::CS_LOCAL, err_desc)) {return false;}if (remote_description()) {// Now that we have a local description, we can push down remote candidates.UseCandidatesInSessionDescription(remote_description());}pending_ice_restarts_.clear();if (error() != ERROR_NONE) {return BadLocalSdp(desc->type(), GetSessionErrorMsg(), err_desc);}return true;
}

action为offer时CreateChannel创建channels:

bool WebRtcSession::CreateChannels(const SessionDescription* desc) {const cricket::ContentGroup* bundle_group = nullptr;if (bundle_policy_ == PeerConnectionInterface::kBundlePolicyMaxBundle) {bundle_group = desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);if (!bundle_group) {LOG(LS_WARNING) << "max-bundle specified without BUNDLE specified";return false;}}// Creating the media channels and transport proxies.const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(desc);if (voice && !voice->rejected && !voice_channel_) {if (!CreateVoiceChannel(voice,GetBundleTransportName(voice, bundle_group))) {LOG(LS_ERROR) << "Failed to create voice channel.";return false;}}const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);if (video && !video->rejected && !video_channel_) {if (!CreateVideoChannel(video,GetBundleTransportName(video, bundle_group))) {LOG(LS_ERROR) << "Failed to create video channel.";return false;}}const cricket::ContentInfo* data = cricket::GetFirstDataContent(desc);if (data_channel_type_ != cricket::DCT_NONE && data && !data->rejected &&!rtp_data_channel_ && !sctp_transport_) {if (!CreateDataChannel(data, GetBundleTransportName(data, bundle_group))) {LOG(LS_ERROR) << "Failed to create data channel.";return false;}}return true;
}

CreateChannels中创建三个Channel,其中CreateVideoChannel创建视频Channel:

bool WebRtcSession::CreateVideoChannel(const cricket::ContentInfo* content,const std::string* bundle_transport) {bool require_rtcp_mux =rtcp_mux_policy_ == PeerConnectionInterface::kRtcpMuxPolicyRequire;std::string transport_name =bundle_transport ? *bundle_transport : content->name;cricket::DtlsTransportInternal* rtp_dtls_transport =transport_controller_->CreateDtlsTransport(transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;if (!require_rtcp_mux) {rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);}video_channel_.reset(channel_manager_->CreateVideoChannel(call_, media_config_, rtp_dtls_transport, rtcp_dtls_transport,transport_controller_->signaling_thread(), content->name, SrtpRequired(),video_options_));if (!video_channel_) {transport_controller_->DestroyDtlsTransport(transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);if (rtcp_dtls_transport) {transport_controller_->DestroyDtlsTransport(transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);}return false;}video_channel_->SignalRtcpMuxFullyActive.connect(this, &WebRtcSession::DestroyRtcpTransport_n);video_channel_->SignalDtlsSrtpSetupFailure.connect(this, &WebRtcSession::OnDtlsSrtpSetupFailure);SignalVideoChannelCreated();video_channel_->SignalSentPacket.connect(this,&WebRtcSession::OnSentPacket_w);return true;
}

调用channel_manager的CreateVideoChannel创建BaseChannel基类的cricket::VideoChannel.
VideoChannel需要传入VideoMediaChannel作为构造参数:

//pc/channelmanager.h/cc
VideoChannel* ChannelManager::CreateVideoChannel_w(webrtc::Call* call,const cricket::MediaConfig& media_config,DtlsTransportInternal* rtp_dtls_transport,DtlsTransportInternal* rtcp_dtls_transport,rtc::PacketTransportInternal* rtp_packet_transport,rtc::PacketTransportInternal* rtcp_packet_transport,rtc::Thread* signaling_thread,const std::string& content_name,bool srtp_required,const VideoOptions& options) {RTC_DCHECK(initialized_);RTC_DCHECK(worker_thread_ == rtc::Thread::Current());RTC_DCHECK(nullptr != call);VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel(call, media_config, options);if (media_channel == NULL) {return NULL;}VideoChannel* video_channel = new VideoChannel(worker_thread_, network_thread_, signaling_thread, media_channel,content_name, rtcp_packet_transport == nullptr, srtp_required);if (!video_channel->Init_w(rtp_dtls_transport, rtcp_dtls_transport,rtp_packet_transport, rtcp_packet_transport)) {delete video_channel;return NULL;}video_channels_.push_back(video_channel);return video_channel;
}

VideoMediaChannel实例media_channel由MediaEngineInterface对象media_engine创建,media_engine由ChannelManager构造方法传入并初始化,ChannelManager由PeerConnectionFactory创建,在PeerConnection初始化方法中,media_engine被创建:

//pc/peerconnectionfactory.cc
bool PeerConnectionFactory::Initialize() {RTC_DCHECK(signaling_thread_->IsCurrent());rtc::InitRandom(rtc::Time32());default_network_manager_.reset(new rtc::BasicNetworkManager());if (!default_network_manager_) {return false;}default_socket_factory_.reset(new rtc::BasicPacketSocketFactory(network_thread_));if (!default_socket_factory_) {return false;}std::unique_ptr<cricket::MediaEngineInterface> media_engine =worker_thread_->Invoke<std::unique_ptr<cricket::MediaEngineInterface>>(RTC_FROM_HERE,rtc::Bind(&PeerConnectionFactory::CreateMediaEngine_w, this));channel_manager_.reset(new cricket::ChannelManager(std::move(media_engine), worker_thread_, network_thread_));channel_manager_->SetVideoRtxEnabled(true);if (!channel_manager_->Init()) {return false;}return true;
}std::unique_ptr<cricket::MediaEngineInterface>
PeerConnectionFactory::CreateMediaEngine_w() {RTC_DCHECK(worker_thread_ == rtc::Thread::Current());return std::unique_ptr<cricket::MediaEngineInterface>(cricket::WebRtcMediaEngineFactory::Create(default_adm_.get(), audio_encoder_factory_,audio_decoder_factory_,video_encoder_factory_.get(), video_decoder_factory_.get(),external_audio_mixer_));
}

WebRtcMediaEngine2继承自CompositeMediaEngine,CompositeMediaEngine父类MediaEngineInterface有WebRtcVoiceEngine voice与WebRtcVideoEngine2 video两个对象

WebRtcVideoEngine2

WebRtcVideoEngine2定义在media/engine/webrtcvideoengine2.h下,用于创建WebRtcVideoChannel2(定义在同一头文件),WebRtcVideoChannel2定义了WebRtcVideoSendStream与WebRtcVideoReceiveStream两个内部类.

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