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Side by Side Diff: webrtc/api/peerconnection.h

Issue 1788703003: Reland the CL to remove candidates when doing continual gathering (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Created 4 years, 9 months ago
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1 /* 1 /*
2 * Copyright 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
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125 const MediaConstraintsInterface* constraints) override; 125 const MediaConstraintsInterface* constraints) override;
126 void CreateAnswer(CreateSessionDescriptionObserver* observer, 126 void CreateAnswer(CreateSessionDescriptionObserver* observer,
127 const RTCOfferAnswerOptions& options) override; 127 const RTCOfferAnswerOptions& options) override;
128 void SetLocalDescription(SetSessionDescriptionObserver* observer, 128 void SetLocalDescription(SetSessionDescriptionObserver* observer,
129 SessionDescriptionInterface* desc) override; 129 SessionDescriptionInterface* desc) override;
130 void SetRemoteDescription(SetSessionDescriptionObserver* observer, 130 void SetRemoteDescription(SetSessionDescriptionObserver* observer,
131 SessionDescriptionInterface* desc) override; 131 SessionDescriptionInterface* desc) override;
132 bool SetConfiguration( 132 bool SetConfiguration(
133 const PeerConnectionInterface::RTCConfiguration& config) override; 133 const PeerConnectionInterface::RTCConfiguration& config) override;
134 bool AddIceCandidate(const IceCandidateInterface* candidate) override; 134 bool AddIceCandidate(const IceCandidateInterface* candidate) override;
135 bool RemoveIceCandidates(
136 const std::vector<cricket::Candidate>& candidates) override;
135 137
136 void RegisterUMAObserver(UMAObserver* observer) override; 138 void RegisterUMAObserver(UMAObserver* observer) override;
137 139
138 void Close() override; 140 void Close() override;
139 141
140 // Virtual for unit tests. 142 // Virtual for unit tests.
141 virtual const std::vector<rtc::scoped_refptr<DataChannel>>& 143 virtual const std::vector<rtc::scoped_refptr<DataChannel>>&
142 sctp_data_channels() const { 144 sctp_data_channels() const {
143 return sctp_data_channels_; 145 return sctp_data_channels_;
144 }; 146 };
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180 void DestroyAudioSender(MediaStreamInterface* stream, 182 void DestroyAudioSender(MediaStreamInterface* stream,
181 AudioTrackInterface* audio_track, 183 AudioTrackInterface* audio_track,
182 uint32_t ssrc); 184 uint32_t ssrc);
183 void DestroyVideoSender(MediaStreamInterface* stream, 185 void DestroyVideoSender(MediaStreamInterface* stream,
184 VideoTrackInterface* video_track); 186 VideoTrackInterface* video_track);
185 187
186 // Implements IceObserver 188 // Implements IceObserver
187 void OnIceConnectionChange(IceConnectionState new_state) override; 189 void OnIceConnectionChange(IceConnectionState new_state) override;
188 void OnIceGatheringChange(IceGatheringState new_state) override; 190 void OnIceGatheringChange(IceGatheringState new_state) override;
189 void OnIceCandidate(const IceCandidateInterface* candidate) override; 191 void OnIceCandidate(const IceCandidateInterface* candidate) override;
192 void OnIceCandidatesRemoved(
193 const std::vector<cricket::Candidate>& candidates) override;
190 void OnIceConnectionReceivingChange(bool receiving) override; 194 void OnIceConnectionReceivingChange(bool receiving) override;
191 195
192 // Signals from WebRtcSession. 196 // Signals from WebRtcSession.
193 void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state); 197 void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
194 void ChangeSignalingState(SignalingState signaling_state); 198 void ChangeSignalingState(SignalingState signaling_state);
195 199
196 // Signals from MediaStreamObserver. 200 // Signals from MediaStreamObserver.
197 void OnAudioTrackAdded(AudioTrackInterface* track, 201 void OnAudioTrackAdded(AudioTrackInterface* track,
198 MediaStreamInterface* stream); 202 MediaStreamInterface* stream);
199 void OnAudioTrackRemoved(AudioTrackInterface* track, 203 void OnAudioTrackRemoved(AudioTrackInterface* track,
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399 // because its destruction fires signals (such as VoiceChannelDestroyed) 403 // because its destruction fires signals (such as VoiceChannelDestroyed)
400 // which will trigger some final actions in PeerConnection... 404 // which will trigger some final actions in PeerConnection...
401 rtc::scoped_ptr<WebRtcSession> session_; 405 rtc::scoped_ptr<WebRtcSession> session_;
402 // ... But stats_ depends on session_ so it should be destroyed even earlier. 406 // ... But stats_ depends on session_ so it should be destroyed even earlier.
403 rtc::scoped_ptr<StatsCollector> stats_; 407 rtc::scoped_ptr<StatsCollector> stats_;
404 }; 408 };
405 409
406 } // namespace webrtc 410 } // namespace webrtc
407 411
408 #endif // WEBRTC_API_PEERCONNECTION_H_ 412 #endif // WEBRTC_API_PEERCONNECTION_H_
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