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

Issue 2111813002: Revert of Move RtcEventLog object from inside VoiceEngine to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 5 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
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224 webrtc::Call::Stats GetStats() const override; 224 webrtc::Call::Stats GetStats() const override;
225 225
226 void SetBitrateConfig( 226 void SetBitrateConfig(
227 const webrtc::Call::Config::BitrateConfig& bitrate_config) override; 227 const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
228 void OnNetworkRouteChanged(const std::string& transport_name, 228 void OnNetworkRouteChanged(const std::string& transport_name,
229 const rtc::NetworkRoute& network_route) override {} 229 const rtc::NetworkRoute& network_route) override {}
230 void SignalChannelNetworkState(webrtc::MediaType media, 230 void SignalChannelNetworkState(webrtc::MediaType media,
231 webrtc::NetworkState state) override; 231 webrtc::NetworkState state) override;
232 void OnSentPacket(const rtc::SentPacket& sent_packet) override; 232 void OnSentPacket(const rtc::SentPacket& sent_packet) override;
233 233
234 bool StartEventLog(rtc::PlatformFile log_file,
235 int64_t max_size_bytes) override;
236 void StopEventLog() override;
237
238 webrtc::Call::Config config_; 234 webrtc::Call::Config config_;
239 webrtc::NetworkState audio_network_state_; 235 webrtc::NetworkState audio_network_state_;
240 webrtc::NetworkState video_network_state_; 236 webrtc::NetworkState video_network_state_;
241 rtc::SentPacket last_sent_packet_; 237 rtc::SentPacket last_sent_packet_;
242 int last_sent_nonnegative_packet_id_ = -1; 238 int last_sent_nonnegative_packet_id_ = -1;
243 webrtc::Call::Stats stats_; 239 webrtc::Call::Stats stats_;
244 std::vector<FakeVideoSendStream*> video_send_streams_; 240 std::vector<FakeVideoSendStream*> video_send_streams_;
245 std::vector<FakeAudioSendStream*> audio_send_streams_; 241 std::vector<FakeAudioSendStream*> audio_send_streams_;
246 std::vector<FakeVideoReceiveStream*> video_receive_streams_; 242 std::vector<FakeVideoReceiveStream*> video_receive_streams_;
247 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; 243 std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
248 244
249 int num_created_send_streams_; 245 int num_created_send_streams_;
250 int num_created_receive_streams_; 246 int num_created_receive_streams_;
251 }; 247 };
252 248
253 } // namespace cricket 249 } // namespace cricket
254 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ 250 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_
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