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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, |
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| 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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