Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(58)

Side by Side Diff: talk/app/webrtc/peerconnection.h

Issue 1669993003: Remove deprecated PeerConnectionObserver::OnStateChange and OnIceComplete (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixed tests. Created 4 years, 10 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « no previous file | talk/app/webrtc/peerconnection.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2012 Google Inc. 3 * Copyright 2012 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 169 matching lines...) Expand 10 before | Expand all | Expand 10 after
180 void DestroyAudioSender(MediaStreamInterface* stream, 180 void DestroyAudioSender(MediaStreamInterface* stream,
181 AudioTrackInterface* audio_track, 181 AudioTrackInterface* audio_track,
182 uint32_t ssrc); 182 uint32_t ssrc);
183 void DestroyVideoSender(MediaStreamInterface* stream, 183 void DestroyVideoSender(MediaStreamInterface* stream,
184 VideoTrackInterface* video_track); 184 VideoTrackInterface* video_track);
185 185
186 // Implements IceObserver 186 // Implements IceObserver
187 void OnIceConnectionChange(IceConnectionState new_state) override; 187 void OnIceConnectionChange(IceConnectionState new_state) override;
188 void OnIceGatheringChange(IceGatheringState new_state) override; 188 void OnIceGatheringChange(IceGatheringState new_state) override;
189 void OnIceCandidate(const IceCandidateInterface* candidate) override; 189 void OnIceCandidate(const IceCandidateInterface* candidate) override;
190 void OnIceComplete() override;
191 void OnIceConnectionReceivingChange(bool receiving) override; 190 void OnIceConnectionReceivingChange(bool receiving) override;
192 191
193 // Signals from WebRtcSession. 192 // Signals from WebRtcSession.
194 void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state); 193 void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
195 void ChangeSignalingState(SignalingState signaling_state); 194 void ChangeSignalingState(SignalingState signaling_state);
196 195
197 // Signals from MediaStreamObserver. 196 // Signals from MediaStreamObserver.
198 void OnAudioTrackAdded(AudioTrackInterface* track, 197 void OnAudioTrackAdded(AudioTrackInterface* track,
199 MediaStreamInterface* stream); 198 MediaStreamInterface* stream);
200 void OnAudioTrackRemoved(AudioTrackInterface* track, 199 void OnAudioTrackRemoved(AudioTrackInterface* track,
(...skipping 190 matching lines...) Expand 10 before | Expand all | Expand 10 after
391 // because its destruction fires signals (such as VoiceChannelDestroyed) 390 // because its destruction fires signals (such as VoiceChannelDestroyed)
392 // which will trigger some final actions in PeerConnection... 391 // which will trigger some final actions in PeerConnection...
393 rtc::scoped_ptr<WebRtcSession> session_; 392 rtc::scoped_ptr<WebRtcSession> session_;
394 // ... But stats_ depends on session_ so it should be destroyed even earlier. 393 // ... But stats_ depends on session_ so it should be destroyed even earlier.
395 rtc::scoped_ptr<StatsCollector> stats_; 394 rtc::scoped_ptr<StatsCollector> stats_;
396 }; 395 };
397 396
398 } // namespace webrtc 397 } // namespace webrtc
399 398
400 #endif // TALK_APP_WEBRTC_PEERCONNECTION_H_ 399 #endif // TALK_APP_WEBRTC_PEERCONNECTION_H_
OLDNEW
« no previous file with comments | « no previous file | talk/app/webrtc/peerconnection.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698