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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 |
| 234 webrtc::Call::Config config_; | 238 webrtc::Call::Config config_; |
| 235 webrtc::NetworkState audio_network_state_; | 239 webrtc::NetworkState audio_network_state_; |
| 236 webrtc::NetworkState video_network_state_; | 240 webrtc::NetworkState video_network_state_; |
| 237 rtc::SentPacket last_sent_packet_; | 241 rtc::SentPacket last_sent_packet_; |
| 238 int last_sent_nonnegative_packet_id_ = -1; | 242 int last_sent_nonnegative_packet_id_ = -1; |
| 239 webrtc::Call::Stats stats_; | 243 webrtc::Call::Stats stats_; |
| 240 std::vector<FakeVideoSendStream*> video_send_streams_; | 244 std::vector<FakeVideoSendStream*> video_send_streams_; |
| 241 std::vector<FakeAudioSendStream*> audio_send_streams_; | 245 std::vector<FakeAudioSendStream*> audio_send_streams_; |
| 242 std::vector<FakeVideoReceiveStream*> video_receive_streams_; | 246 std::vector<FakeVideoReceiveStream*> video_receive_streams_; |
| 243 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; | 247 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; |
| 244 | 248 |
| 245 int num_created_send_streams_; | 249 int num_created_send_streams_; |
| 246 int num_created_receive_streams_; | 250 int num_created_receive_streams_; |
| 247 }; | 251 }; |
| 248 | 252 |
| 249 } // namespace cricket | 253 } // namespace cricket |
| 250 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ | 254 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ |
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