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1 // Copyright 2015 The Chromium Authors. All rights reserved. | 1 // Copyright 2015 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include "remoting/protocol/webrtc_connection_to_client.h" | 5 #include "remoting/protocol/webrtc_connection_to_client.h" |
6 | 6 |
7 #include <utility> | 7 #include <utility> |
8 | 8 |
9 #include "base/bind.h" | 9 #include "base/bind.h" |
10 #include "base/location.h" | 10 #include "base/location.h" |
| 11 #include "jingle/glue/thread_wrapper.h" |
11 #include "net/base/io_buffer.h" | 12 #include "net/base/io_buffer.h" |
12 #include "remoting/codec/video_encoder.h" | 13 #include "remoting/codec/video_encoder.h" |
13 #include "remoting/codec/video_encoder_verbatim.h" | 14 #include "remoting/codec/video_encoder_verbatim.h" |
14 #include "remoting/codec/video_encoder_vpx.h" | 15 #include "remoting/codec/video_encoder_vpx.h" |
15 #include "remoting/protocol/audio_writer.h" | 16 #include "remoting/protocol/audio_writer.h" |
16 #include "remoting/protocol/clipboard_stub.h" | 17 #include "remoting/protocol/clipboard_stub.h" |
17 #include "remoting/protocol/host_control_dispatcher.h" | 18 #include "remoting/protocol/host_control_dispatcher.h" |
18 #include "remoting/protocol/host_event_dispatcher.h" | 19 #include "remoting/protocol/host_event_dispatcher.h" |
19 #include "remoting/protocol/host_stub.h" | 20 #include "remoting/protocol/host_stub.h" |
20 #include "remoting/protocol/input_stub.h" | 21 #include "remoting/protocol/input_stub.h" |
| 22 #include "remoting/protocol/transport_context.h" |
21 #include "remoting/protocol/webrtc_transport.h" | 23 #include "remoting/protocol/webrtc_transport.h" |
22 #include "remoting/protocol/webrtc_video_capturer_adapter.h" | 24 #include "remoting/protocol/webrtc_video_capturer_adapter.h" |
23 #include "remoting/protocol/webrtc_video_stream.h" | 25 #include "remoting/protocol/webrtc_video_stream.h" |
24 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" | 26 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" |
25 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h
" | 27 #include "third_party/libjingle/source/talk/app/webrtc/peerconnectioninterface.h
" |
26 #include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h" | 28 #include "third_party/libjingle/source/talk/app/webrtc/test/fakeconstraints.h" |
27 #include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" | 29 #include "third_party/libjingle/source/talk/app/webrtc/videosourceinterface.h" |
28 | 30 |
29 namespace remoting { | 31 namespace remoting { |
30 namespace protocol { | 32 namespace protocol { |
31 | 33 |
32 const char kStreamLabel[] = "screen_stream"; | 34 const char kStreamLabel[] = "screen_stream"; |
33 const char kVideoLabel[] = "screen_video"; | 35 const char kVideoLabel[] = "screen_video"; |
34 | 36 |
| 37 // Currently the network thread is also used as worker thread for webrtc. |
| 38 // |
| 39 // TODO(sergeyu): Figure out if we would benefit from using a separate |
| 40 // thread as a worker thread. |
35 WebrtcConnectionToClient::WebrtcConnectionToClient( | 41 WebrtcConnectionToClient::WebrtcConnectionToClient( |
36 scoped_ptr<protocol::Session> session) | 42 scoped_ptr<protocol::Session> session, |
37 : session_(std::move(session)), | 43 scoped_refptr<protocol::TransportContext> transport_context) |
| 44 : transport_(jingle_glue::JingleThreadWrapper::current(), |
| 45 transport_context, |
| 46 this), |
| 47 session_(std::move(session)), |
38 control_dispatcher_(new HostControlDispatcher()), | 48 control_dispatcher_(new HostControlDispatcher()), |
39 event_dispatcher_(new HostEventDispatcher()) { | 49 event_dispatcher_(new HostEventDispatcher()) { |
40 session_->SetEventHandler(this); | 50 session_->SetEventHandler(this); |
| 51 session_->SetTransport(&transport_); |
41 } | 52 } |
42 | 53 |
43 WebrtcConnectionToClient::~WebrtcConnectionToClient() {} | 54 WebrtcConnectionToClient::~WebrtcConnectionToClient() {} |
44 | 55 |
45 void WebrtcConnectionToClient::SetEventHandler( | 56 void WebrtcConnectionToClient::SetEventHandler( |
46 ConnectionToClient::EventHandler* event_handler) { | 57 ConnectionToClient::EventHandler* event_handler) { |
47 DCHECK(thread_checker_.CalledOnValidThread()); | 58 DCHECK(thread_checker_.CalledOnValidThread()); |
48 event_handler_ = event_handler; | 59 event_handler_ = event_handler; |
49 } | 60 } |
50 | 61 |
51 protocol::Session* WebrtcConnectionToClient::session() { | 62 protocol::Session* WebrtcConnectionToClient::session() { |
52 DCHECK(thread_checker_.CalledOnValidThread()); | 63 DCHECK(thread_checker_.CalledOnValidThread()); |
53 return session_.get(); | 64 return session_.get(); |
54 } | 65 } |
55 | 66 |
56 void WebrtcConnectionToClient::Disconnect(ErrorCode error) { | 67 void WebrtcConnectionToClient::Disconnect(ErrorCode error) { |
57 DCHECK(thread_checker_.CalledOnValidThread()); | 68 DCHECK(thread_checker_.CalledOnValidThread()); |
58 | 69 |
59 control_dispatcher_.reset(); | |
60 event_dispatcher_.reset(); | |
61 | |
62 // This should trigger OnConnectionClosed() event and this object | 70 // This should trigger OnConnectionClosed() event and this object |
63 // may be destroyed as the result. | 71 // may be destroyed as the result. |
64 session_->Close(error); | 72 session_->Close(error); |
65 } | 73 } |
66 | 74 |
67 void WebrtcConnectionToClient::OnInputEventReceived(int64_t timestamp) { | 75 void WebrtcConnectionToClient::OnInputEventReceived(int64_t timestamp) { |
68 DCHECK(thread_checker_.CalledOnValidThread()); | 76 DCHECK(thread_checker_.CalledOnValidThread()); |
69 event_handler_->OnInputEventReceived(this, timestamp); | 77 event_handler_->OnInputEventReceived(this, timestamp); |
70 } | 78 } |
71 | 79 |
72 scoped_ptr<VideoStream> WebrtcConnectionToClient::StartVideoStream( | 80 scoped_ptr<VideoStream> WebrtcConnectionToClient::StartVideoStream( |
73 scoped_ptr<webrtc::DesktopCapturer> desktop_capturer) { | 81 scoped_ptr<webrtc::DesktopCapturer> desktop_capturer) { |
74 // TODO(sergeyu): Reconsider Transport interface and how it's used here. | |
75 WebrtcTransport* transport = session_->GetTransport()->AsWebrtcTransport(); | |
76 CHECK(transport); | |
77 | |
78 scoped_ptr<WebrtcVideoCapturerAdapter> video_capturer_adapter( | 82 scoped_ptr<WebrtcVideoCapturerAdapter> video_capturer_adapter( |
79 new WebrtcVideoCapturerAdapter(std::move(desktop_capturer))); | 83 new WebrtcVideoCapturerAdapter(std::move(desktop_capturer))); |
80 | 84 |
81 // Set video stream constraints. | 85 // Set video stream constraints. |
82 webrtc::FakeConstraints video_constraints; | 86 webrtc::FakeConstraints video_constraints; |
83 video_constraints.AddMandatory( | 87 video_constraints.AddMandatory( |
84 webrtc::MediaConstraintsInterface::kMinFrameRate, 5); | 88 webrtc::MediaConstraintsInterface::kMinFrameRate, 5); |
85 | 89 |
86 rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track = | 90 rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track = |
87 transport->peer_connection_factory()->CreateVideoTrack( | 91 transport_.peer_connection_factory()->CreateVideoTrack( |
88 kVideoLabel, | 92 kVideoLabel, |
89 transport->peer_connection_factory()->CreateVideoSource( | 93 transport_.peer_connection_factory()->CreateVideoSource( |
90 video_capturer_adapter.release(), &video_constraints)); | 94 video_capturer_adapter.release(), &video_constraints)); |
91 | 95 |
92 rtc::scoped_refptr<webrtc::MediaStreamInterface> video_stream = | 96 rtc::scoped_refptr<webrtc::MediaStreamInterface> video_stream = |
93 transport->peer_connection_factory()->CreateLocalMediaStream( | 97 transport_.peer_connection_factory()->CreateLocalMediaStream( |
94 kStreamLabel); | 98 kStreamLabel); |
95 | 99 |
96 if (!video_stream->AddTrack(video_track) || | 100 if (!video_stream->AddTrack(video_track) || |
97 !transport->peer_connection()->AddStream(video_stream)) { | 101 !transport_.peer_connection()->AddStream(video_stream)) { |
98 return nullptr; | 102 return nullptr; |
99 } | 103 } |
100 | 104 |
101 return make_scoped_ptr( | 105 return make_scoped_ptr( |
102 new WebrtcVideoStream(transport->peer_connection(), video_stream)); | 106 new WebrtcVideoStream(transport_.peer_connection(), video_stream)); |
103 } | 107 } |
104 | 108 |
105 AudioStub* WebrtcConnectionToClient::audio_stub() { | 109 AudioStub* WebrtcConnectionToClient::audio_stub() { |
106 DCHECK(thread_checker_.CalledOnValidThread()); | 110 DCHECK(thread_checker_.CalledOnValidThread()); |
107 return nullptr; | 111 return nullptr; |
108 } | 112 } |
109 | 113 |
110 // Return pointer to ClientStub. | 114 // Return pointer to ClientStub. |
111 ClientStub* WebrtcConnectionToClient::client_stub() { | 115 ClientStub* WebrtcConnectionToClient::client_stub() { |
112 DCHECK(thread_checker_.CalledOnValidThread()); | 116 DCHECK(thread_checker_.CalledOnValidThread()); |
(...skipping 25 matching lines...) Expand all Loading... |
138 case Session::CONNECTING: | 142 case Session::CONNECTING: |
139 case Session::ACCEPTING: | 143 case Session::ACCEPTING: |
140 case Session::ACCEPTED: | 144 case Session::ACCEPTED: |
141 // Don't care about these events. | 145 // Don't care about these events. |
142 break; | 146 break; |
143 case Session::AUTHENTICATING: | 147 case Session::AUTHENTICATING: |
144 event_handler_->OnConnectionAuthenticating(this); | 148 event_handler_->OnConnectionAuthenticating(this); |
145 break; | 149 break; |
146 case Session::AUTHENTICATED: { | 150 case Session::AUTHENTICATED: { |
147 // Initialize channels. | 151 // Initialize channels. |
148 control_dispatcher_->Init( | 152 control_dispatcher_->Init(transport_.GetStreamChannelFactory(), this); |
149 session_->GetTransport()->GetStreamChannelFactory(), | |
150 this); | |
151 | 153 |
152 event_dispatcher_->Init( | 154 event_dispatcher_->Init(transport_.GetStreamChannelFactory(), this); |
153 session_->GetTransport()->GetStreamChannelFactory(), this); | |
154 event_dispatcher_->set_on_input_event_callback(base::Bind( | 155 event_dispatcher_->set_on_input_event_callback(base::Bind( |
155 &ConnectionToClient::OnInputEventReceived, base::Unretained(this))); | 156 &ConnectionToClient::OnInputEventReceived, base::Unretained(this))); |
156 | 157 |
157 // Notify the handler after initializing the channels, so that | 158 // Notify the handler after initializing the channels, so that |
158 // ClientSession can get a client clipboard stub. | 159 // ClientSession can get a client clipboard stub. |
159 event_handler_->OnConnectionAuthenticated(this); | 160 event_handler_->OnConnectionAuthenticated(this); |
160 break; | 161 break; |
161 } | 162 } |
162 | 163 |
163 case Session::CONNECTED: | |
164 event_handler_->OnConnectionChannelsConnected(this); | |
165 break; | |
166 | |
167 case Session::CLOSED: | 164 case Session::CLOSED: |
168 case Session::FAILED: | 165 case Session::FAILED: |
169 control_dispatcher_.reset(); | 166 control_dispatcher_.reset(); |
170 event_dispatcher_.reset(); | 167 event_dispatcher_.reset(); |
171 event_handler_->OnConnectionClosed( | 168 event_handler_->OnConnectionClosed( |
172 this, state == Session::CLOSED ? OK : session_->error()); | 169 this, state == Session::CLOSED ? OK : session_->error()); |
173 break; | 170 break; |
174 } | 171 } |
175 } | 172 } |
176 | 173 |
177 void WebrtcConnectionToClient::OnSessionRouteChange( | 174 void WebrtcConnectionToClient::OnWebrtcTransportConnected() { |
178 const std::string& channel_name, | 175 event_handler_->OnConnectionChannelsConnected(this); |
179 const TransportRoute& route) { | 176 } |
180 event_handler_->OnRouteChange(this, channel_name, route); | 177 |
| 178 void WebrtcConnectionToClient::OnWebrtcTransportError(ErrorCode error) { |
| 179 DCHECK(thread_checker_.CalledOnValidThread()); |
| 180 Disconnect(error); |
181 } | 181 } |
182 | 182 |
183 void WebrtcConnectionToClient::OnChannelInitialized( | 183 void WebrtcConnectionToClient::OnChannelInitialized( |
184 ChannelDispatcherBase* channel_dispatcher) { | 184 ChannelDispatcherBase* channel_dispatcher) { |
185 DCHECK(thread_checker_.CalledOnValidThread()); | 185 DCHECK(thread_checker_.CalledOnValidThread()); |
186 } | 186 } |
187 | 187 |
188 void WebrtcConnectionToClient::OnChannelError( | 188 void WebrtcConnectionToClient::OnChannelError( |
189 ChannelDispatcherBase* channel_dispatcher, | 189 ChannelDispatcherBase* channel_dispatcher, |
190 ErrorCode error) { | 190 ErrorCode error) { |
191 DCHECK(thread_checker_.CalledOnValidThread()); | 191 DCHECK(thread_checker_.CalledOnValidThread()); |
192 | 192 |
193 LOG(ERROR) << "Failed to connect channel " | 193 LOG(ERROR) << "Failed to connect channel " |
194 << channel_dispatcher->channel_name(); | 194 << channel_dispatcher->channel_name(); |
195 session_->Close(CHANNEL_CONNECTION_ERROR); | 195 Disconnect(error); |
196 } | 196 } |
197 | 197 |
198 } // namespace protocol | 198 } // namespace protocol |
199 } // namespace remoting | 199 } // namespace remoting |
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