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Side by Side Diff: content/renderer/media/rtc_peer_connection_handler.h

Issue 475263002: Stop all PeerConnections upon suspend from Blink side (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: rebase Created 6 years, 4 months ago
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1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. 1 // Copyright (c) 2012 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 #ifndef CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_ 5 #ifndef CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_
6 #define CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_ 6 #define CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_
7 7
8 #include <map> 8 #include <map>
9 #include <string> 9 #include <string>
10 10
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178 178
179 // Delegate functions to allow for mocking of WebKit interfaces. 179 // Delegate functions to allow for mocking of WebKit interfaces.
180 // getStats takes ownership of request parameter. 180 // getStats takes ownership of request parameter.
181 virtual void getStats(LocalRTCStatsRequest* request); 181 virtual void getStats(LocalRTCStatsRequest* request);
182 182
183 // Calls GetStats on |native_peer_connection_|. 183 // Calls GetStats on |native_peer_connection_|.
184 void GetStats(webrtc::StatsObserver* observer, 184 void GetStats(webrtc::StatsObserver* observer,
185 webrtc::MediaStreamTrackInterface* track, 185 webrtc::MediaStreamTrackInterface* track,
186 webrtc::PeerConnectionInterface::StatsOutputLevel level); 186 webrtc::PeerConnectionInterface::StatsOutputLevel level);
187 187
188 // Tells the |client_| to close RTCPeerConnection.
189 void CloseClientPeerConnection();
190
188 PeerConnectionTracker* peer_connection_tracker(); 191 PeerConnectionTracker* peer_connection_tracker();
189 192
190 protected: 193 protected:
191 webrtc::PeerConnectionInterface* native_peer_connection() { 194 webrtc::PeerConnectionInterface* native_peer_connection() {
192 return native_peer_connection_.get(); 195 return native_peer_connection_.get();
193 } 196 }
194 197
195 private: 198 private:
196 webrtc::SessionDescriptionInterface* CreateNativeSessionDescription( 199 webrtc::SessionDescriptionInterface* CreateNativeSessionDescription(
197 const blink::WebRTCSessionDescription& description, 200 const blink::WebRTCSessionDescription& description,
(...skipping 25 matching lines...) Expand all
223 RemoteStreamMap remote_streams_; 226 RemoteStreamMap remote_streams_;
224 scoped_refptr<webrtc::UMAObserver> uma_observer_; 227 scoped_refptr<webrtc::UMAObserver> uma_observer_;
225 base::TimeTicks ice_connection_checking_start_; 228 base::TimeTicks ice_connection_checking_start_;
226 229
227 DISALLOW_COPY_AND_ASSIGN(RTCPeerConnectionHandler); 230 DISALLOW_COPY_AND_ASSIGN(RTCPeerConnectionHandler);
228 }; 231 };
229 232
230 } // namespace content 233 } // namespace content
231 234
232 #endif // CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_ 235 #endif // CONTENT_RENDERER_MEDIA_RTC_PEER_CONNECTION_HANDLER_H_
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