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    Searched refs:WebRtcSession (Results 1 - 8 of 8) sorted by null

  /external/webrtc/talk/app/webrtc/
webrtcsession.cc 28 #include "talk/app/webrtc/webrtcsession.h"
393 case webrtc::WebRtcSession::state: \
397 static std::string GetStateString(webrtc::WebRtcSession::State state) {
415 case webrtc::WebRtcSession::err: \
419 static std::string GetErrorCodeString(webrtc::WebRtcSession::Error err) {
542 WebRtcSession::WebRtcSession(webrtc::MediaControllerInterface* media_controller,
568 this, &WebRtcSession::OnTransportControllerConnectionState);
570 this, &WebRtcSession::OnTransportControllerReceiving);
572 this, &WebRtcSession::OnTransportControllerGatheringState)
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webrtcsessiondescriptionfactory.h 47 class WebRtcSession;
84 // the async DTLS identity generation for WebRtcSession.
94 WebRtcSession* session,
103 WebRtcSession* session,
112 WebRtcSession* session,
153 WebRtcSession* session,
185 WebRtcSession* const session_;
peerconnection.h 40 #include "talk/app/webrtc/webrtcsession.h"
66 // It uses WebRtcSession to implement the PeerConnection functionality.
86 virtual WebRtcSession* session() { return session_.get(); }
188 // Signals from WebRtcSession.
189 void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
315 // Notifications from WebRtcSession relating to BaseChannels.
388 rtc::scoped_ptr<WebRtcSession> session_;
webrtcsession.h 125 // A WebRtcSession manages general session state. This includes negotiation
132 class WebRtcSession : public AudioProviderInterface,
154 WebRtcSession(webrtc::MediaControllerInterface* media_controller,
158 virtual ~WebRtcSession();
183 sigslot::signal2<WebRtcSession*, State> SignalState;
513 // Declares the bundle policy for the WebRTCSession.
516 // Declares the RTCP mux policy for the WebRTCSession.
519 RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
webrtcsessiondescriptionfactory.cc 36 #include "talk/app/webrtc/webrtcsession.h"
138 WebRtcSession* session,
161 WebRtcSession* session,
177 WebRtcSession* session,
209 WebRtcSession* session,
247 // when WebRtcSession listens to the callback but it was the WebRtcSession
peerconnection.cc 578 // Need to detach RTP senders/receivers from WebRtcSession,
642 new WebRtcSession(media_controller_.get(), factory_->signaling_thread(),
646 // Initialize the WebRtcSession. It creates transport channels etc.
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webrtcsession_unittest.cc 43 #include "talk/app/webrtc/webrtcsession.h"
96 using webrtc::WebRtcSession;
237 class WebRtcSessionForTest : public webrtc::WebRtcSession {
244 : WebRtcSession(media_controller,
278 using webrtc::WebRtcSession::SetAudioPlayout;
279 using webrtc::WebRtcSession::SetAudioSend;
280 using webrtc::WebRtcSession::SetCaptureDevice;
281 using webrtc::WebRtcSession::SetVideoPlayout;
282 using webrtc::WebRtcSession::SetVideoSend;
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statscollector_unittest.cc 84 class MockWebRtcSession : public webrtc::WebRtcSession {
87 : WebRtcSession(media_controller,
116 MOCK_METHOD0(session, WebRtcSession*());
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