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Side by Side Diff: webrtc/media/engine/fakewebrtccall.h

Issue 2353033005: Refactoring: move ownership of RtcEventLog from Call to PeerConnection (Closed)
Patch Set: Moved DEPS entry to subdirectory Created 4 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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
(...skipping 219 matching lines...) Expand 10 before | Expand all | Expand 10 after
230 webrtc::Call::Stats GetStats() const override; 230 webrtc::Call::Stats GetStats() const override;
231 231
232 void SetBitrateConfig( 232 void SetBitrateConfig(
233 const webrtc::Call::Config::BitrateConfig& bitrate_config) override; 233 const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
234 void OnNetworkRouteChanged(const std::string& transport_name, 234 void OnNetworkRouteChanged(const std::string& transport_name,
235 const rtc::NetworkRoute& network_route) override {} 235 const rtc::NetworkRoute& network_route) override {}
236 void SignalChannelNetworkState(webrtc::MediaType media, 236 void SignalChannelNetworkState(webrtc::MediaType media,
237 webrtc::NetworkState state) override; 237 webrtc::NetworkState state) override;
238 void OnSentPacket(const rtc::SentPacket& sent_packet) override; 238 void OnSentPacket(const rtc::SentPacket& sent_packet) override;
239 239
240 bool StartEventLog(rtc::PlatformFile log_file,
241 int64_t max_size_bytes) override;
242 void StopEventLog() override;
243
244 webrtc::Call::Config config_; 240 webrtc::Call::Config config_;
245 webrtc::NetworkState audio_network_state_; 241 webrtc::NetworkState audio_network_state_;
246 webrtc::NetworkState video_network_state_; 242 webrtc::NetworkState video_network_state_;
247 rtc::SentPacket last_sent_packet_; 243 rtc::SentPacket last_sent_packet_;
248 int last_sent_nonnegative_packet_id_ = -1; 244 int last_sent_nonnegative_packet_id_ = -1;
249 webrtc::Call::Stats stats_; 245 webrtc::Call::Stats stats_;
250 std::vector<FakeVideoSendStream*> video_send_streams_; 246 std::vector<FakeVideoSendStream*> video_send_streams_;
251 std::vector<FakeAudioSendStream*> audio_send_streams_; 247 std::vector<FakeAudioSendStream*> audio_send_streams_;
252 std::vector<FakeVideoReceiveStream*> video_receive_streams_; 248 std::vector<FakeVideoReceiveStream*> video_receive_streams_;
253 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; 249 std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
254 250
255 int num_created_send_streams_; 251 int num_created_send_streams_;
256 int num_created_receive_streams_; 252 int num_created_receive_streams_;
257 }; 253 };
258 254
259 } // namespace cricket 255 } // namespace cricket
260 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ 256 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_
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