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

Issue 1648813004: Remove candidates when doing continual gathering (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Address Alex's comments 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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123 const MediaConstraintsInterface* constraints) override; 123 const MediaConstraintsInterface* constraints) override;
124 void CreateAnswer(CreateSessionDescriptionObserver* observer, 124 void CreateAnswer(CreateSessionDescriptionObserver* observer,
125 const RTCOfferAnswerOptions& options) override; 125 const RTCOfferAnswerOptions& options) override;
126 void SetLocalDescription(SetSessionDescriptionObserver* observer, 126 void SetLocalDescription(SetSessionDescriptionObserver* observer,
127 SessionDescriptionInterface* desc) override; 127 SessionDescriptionInterface* desc) override;
128 void SetRemoteDescription(SetSessionDescriptionObserver* observer, 128 void SetRemoteDescription(SetSessionDescriptionObserver* observer,
129 SessionDescriptionInterface* desc) override; 129 SessionDescriptionInterface* desc) override;
130 bool SetConfiguration( 130 bool SetConfiguration(
131 const PeerConnectionInterface::RTCConfiguration& config) override; 131 const PeerConnectionInterface::RTCConfiguration& config) override;
132 bool AddIceCandidate(const IceCandidateInterface* candidate) override; 132 bool AddIceCandidate(const IceCandidateInterface* candidate) override;
133 bool RemoveIceCandidates(
134 const std::vector<cricket::Candidate>& candidates) override;
133 135
134 void RegisterUMAObserver(UMAObserver* observer) override; 136 void RegisterUMAObserver(UMAObserver* observer) override;
135 137
136 void Close() override; 138 void Close() override;
137 139
138 // Virtual for unit tests. 140 // Virtual for unit tests.
139 virtual const std::vector<rtc::scoped_refptr<DataChannel>>& 141 virtual const std::vector<rtc::scoped_refptr<DataChannel>>&
140 sctp_data_channels() const { 142 sctp_data_channels() const {
141 return sctp_data_channels_; 143 return sctp_data_channels_;
142 }; 144 };
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177 void DestroyAudioSender(MediaStreamInterface* stream, 179 void DestroyAudioSender(MediaStreamInterface* stream,
178 AudioTrackInterface* audio_track, 180 AudioTrackInterface* audio_track,
179 uint32_t ssrc); 181 uint32_t ssrc);
180 void DestroyVideoSender(MediaStreamInterface* stream, 182 void DestroyVideoSender(MediaStreamInterface* stream,
181 VideoTrackInterface* video_track); 183 VideoTrackInterface* video_track);
182 184
183 // Implements IceObserver 185 // Implements IceObserver
184 void OnIceConnectionChange(IceConnectionState new_state) override; 186 void OnIceConnectionChange(IceConnectionState new_state) override;
185 void OnIceGatheringChange(IceGatheringState new_state) override; 187 void OnIceGatheringChange(IceGatheringState new_state) override;
186 void OnIceCandidate(const IceCandidateInterface* candidate) override; 188 void OnIceCandidate(const IceCandidateInterface* candidate) override;
189 void OnIceCandidatesRemoved(
190 const std::vector<cricket::Candidate>& candidates) override;
187 void OnIceConnectionReceivingChange(bool receiving) override; 191 void OnIceConnectionReceivingChange(bool receiving) override;
188 192
189 // Signals from WebRtcSession. 193 // Signals from WebRtcSession.
190 void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state); 194 void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
191 void ChangeSignalingState(SignalingState signaling_state); 195 void ChangeSignalingState(SignalingState signaling_state);
192 196
193 // Signals from MediaStreamObserver. 197 // Signals from MediaStreamObserver.
194 void OnAudioTrackAdded(AudioTrackInterface* track, 198 void OnAudioTrackAdded(AudioTrackInterface* track,
195 MediaStreamInterface* stream); 199 MediaStreamInterface* stream);
196 void OnAudioTrackRemoved(AudioTrackInterface* track, 200 void OnAudioTrackRemoved(AudioTrackInterface* track,
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396 // because its destruction fires signals (such as VoiceChannelDestroyed) 400 // because its destruction fires signals (such as VoiceChannelDestroyed)
397 // which will trigger some final actions in PeerConnection... 401 // which will trigger some final actions in PeerConnection...
398 rtc::scoped_ptr<WebRtcSession> session_; 402 rtc::scoped_ptr<WebRtcSession> session_;
399 // ... But stats_ depends on session_ so it should be destroyed even earlier. 403 // ... But stats_ depends on session_ so it should be destroyed even earlier.
400 rtc::scoped_ptr<StatsCollector> stats_; 404 rtc::scoped_ptr<StatsCollector> stats_;
401 }; 405 };
402 406
403 } // namespace webrtc 407 } // namespace webrtc
404 408
405 #endif // WEBRTC_API_PEERCONNECTION_H_ 409 #endif // WEBRTC_API_PEERCONNECTION_H_
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