| // Copyright 2015 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "remoting/protocol/webrtc_connection_to_client.h" |
| |
| #include <utility> |
| |
| #include "base/bind.h" |
| #include "base/location.h" |
| #include "jingle/glue/thread_wrapper.h" |
| #include "net/base/io_buffer.h" |
| #include "remoting/codec/video_encoder.h" |
| #include "remoting/codec/video_encoder_verbatim.h" |
| #include "remoting/codec/video_encoder_vpx.h" |
| #include "remoting/protocol/audio_writer.h" |
| #include "remoting/protocol/clipboard_stub.h" |
| #include "remoting/protocol/host_control_dispatcher.h" |
| #include "remoting/protocol/host_event_dispatcher.h" |
| #include "remoting/protocol/host_stub.h" |
| #include "remoting/protocol/input_stub.h" |
| #include "remoting/protocol/transport_context.h" |
| #include "remoting/protocol/webrtc_transport.h" |
| #include "remoting/protocol/webrtc_video_capturer_adapter.h" |
| #include "remoting/protocol/webrtc_video_stream.h" |
| #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
| #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h" |
| #include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h" |
| #include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" |
| |
| namespace remoting { |
| namespace protocol { |
| |
| const char kStreamLabel[] = "screen_stream"; |
| const char kVideoLabel[] = "screen_video"; |
| |
| // Currently the network thread is also used as worker thread for webrtc. |
| // |
| // TODO(sergeyu): Figure out if we would benefit from using a separate |
| // thread as a worker thread. |
| WebrtcConnectionToClient::WebrtcConnectionToClient( |
| scoped_ptr<protocol::Session> session, |
| scoped_refptr<protocol::TransportContext> transport_context) |
| : transport_(jingle_glue::JingleThreadWrapper::current(), |
| transport_context, |
| this), |
| session_(std::move(session)), |
| control_dispatcher_(new HostControlDispatcher()), |
| event_dispatcher_(new HostEventDispatcher()) { |
| session_->SetEventHandler(this); |
| session_->SetTransport(&transport_); |
| } |
| |
| WebrtcConnectionToClient::~WebrtcConnectionToClient() {} |
| |
| void WebrtcConnectionToClient::SetEventHandler( |
| ConnectionToClient::EventHandler* event_handler) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| event_handler_ = event_handler; |
| } |
| |
| protocol::Session* WebrtcConnectionToClient::session() { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| return session_.get(); |
| } |
| |
| void WebrtcConnectionToClient::Disconnect(ErrorCode error) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| |
| // This should trigger OnConnectionClosed() event and this object |
| // may be destroyed as the result. |
| session_->Close(error); |
| } |
| |
| void WebrtcConnectionToClient::OnInputEventReceived(int64_t timestamp) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| event_handler_->OnInputEventReceived(this, timestamp); |
| } |
| |
| scoped_ptr<VideoStream> WebrtcConnectionToClient::StartVideoStream( |
| scoped_ptr<webrtc::DesktopCapturer> desktop_capturer) { |
| scoped_ptr<WebrtcVideoCapturerAdapter> video_capturer_adapter( |
| new WebrtcVideoCapturerAdapter(std::move(desktop_capturer))); |
| |
| // Set video stream constraints. |
| webrtc::FakeConstraints video_constraints; |
| video_constraints.AddMandatory( |
| webrtc::MediaConstraintsInterface::kMinFrameRate, 5); |
| |
| rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track = |
| transport_.peer_connection_factory()->CreateVideoTrack( |
| kVideoLabel, |
| transport_.peer_connection_factory()->CreateVideoSource( |
| video_capturer_adapter.release(), &video_constraints)); |
| |
| rtc::scoped_refptr<webrtc::MediaStreamInterface> video_stream = |
| transport_.peer_connection_factory()->CreateLocalMediaStream( |
| kStreamLabel); |
| |
| if (!video_stream->AddTrack(video_track) || |
| !transport_.peer_connection()->AddStream(video_stream)) { |
| return nullptr; |
| } |
| |
| return make_scoped_ptr( |
| new WebrtcVideoStream(transport_.peer_connection(), video_stream)); |
| } |
| |
| AudioStub* WebrtcConnectionToClient::audio_stub() { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| return nullptr; |
| } |
| |
| // Return pointer to ClientStub. |
| ClientStub* WebrtcConnectionToClient::client_stub() { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| return control_dispatcher_.get(); |
| } |
| |
| void WebrtcConnectionToClient::set_clipboard_stub( |
| protocol::ClipboardStub* clipboard_stub) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| control_dispatcher_->set_clipboard_stub(clipboard_stub); |
| } |
| |
| void WebrtcConnectionToClient::set_host_stub(protocol::HostStub* host_stub) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| control_dispatcher_->set_host_stub(host_stub); |
| } |
| |
| void WebrtcConnectionToClient::set_input_stub(protocol::InputStub* input_stub) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| event_dispatcher_->set_input_stub(input_stub); |
| } |
| |
| void WebrtcConnectionToClient::OnSessionStateChange(Session::State state) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| |
| DCHECK(event_handler_); |
| switch(state) { |
| case Session::INITIALIZING: |
| case Session::CONNECTING: |
| case Session::ACCEPTING: |
| case Session::ACCEPTED: |
| // Don't care about these events. |
| break; |
| case Session::AUTHENTICATING: |
| event_handler_->OnConnectionAuthenticating(this); |
| break; |
| case Session::AUTHENTICATED: { |
| // Initialize channels. |
| control_dispatcher_->Init(transport_.GetStreamChannelFactory(), this); |
| |
| event_dispatcher_->Init(transport_.GetStreamChannelFactory(), this); |
| event_dispatcher_->set_on_input_event_callback(base::Bind( |
| &ConnectionToClient::OnInputEventReceived, base::Unretained(this))); |
| |
| // Notify the handler after initializing the channels, so that |
| // ClientSession can get a client clipboard stub. |
| event_handler_->OnConnectionAuthenticated(this); |
| break; |
| } |
| |
| case Session::CLOSED: |
| case Session::FAILED: |
| control_dispatcher_.reset(); |
| event_dispatcher_.reset(); |
| event_handler_->OnConnectionClosed( |
| this, state == Session::CLOSED ? OK : session_->error()); |
| break; |
| } |
| } |
| |
| void WebrtcConnectionToClient::OnWebrtcTransportConnected() { |
| event_handler_->OnConnectionChannelsConnected(this); |
| } |
| |
| void WebrtcConnectionToClient::OnWebrtcTransportError(ErrorCode error) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| Disconnect(error); |
| } |
| |
| void WebrtcConnectionToClient::OnChannelInitialized( |
| ChannelDispatcherBase* channel_dispatcher) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| } |
| |
| void WebrtcConnectionToClient::OnChannelError( |
| ChannelDispatcherBase* channel_dispatcher, |
| ErrorCode error) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| |
| LOG(ERROR) << "Failed to connect channel " |
| << channel_dispatcher->channel_name(); |
| Disconnect(error); |
| } |
| |
| } // namespace protocol |
| } // namespace remoting |