OLD | NEW |
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 182 matching lines...) Expand 10 before | Expand all | Expand 10 after Loading... |
193 webrtc::Call::Config GetConfig() const; | 193 webrtc::Call::Config GetConfig() const; |
194 const std::vector<FakeVideoSendStream*>& GetVideoSendStreams(); | 194 const std::vector<FakeVideoSendStream*>& GetVideoSendStreams(); |
195 const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams(); | 195 const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams(); |
196 | 196 |
197 const std::vector<FakeAudioSendStream*>& GetAudioSendStreams(); | 197 const std::vector<FakeAudioSendStream*>& GetAudioSendStreams(); |
198 const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc); | 198 const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc); |
199 const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams(); | 199 const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams(); |
200 const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc); | 200 const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc); |
201 | 201 |
202 rtc::SentPacket last_sent_packet() const { return last_sent_packet_; } | 202 rtc::SentPacket last_sent_packet() const { return last_sent_packet_; } |
203 webrtc::NetworkState GetNetworkState() const; | 203 webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const; |
204 int GetNumCreatedSendStreams() const; | 204 int GetNumCreatedSendStreams() const; |
205 int GetNumCreatedReceiveStreams() const; | 205 int GetNumCreatedReceiveStreams() const; |
206 void SetStats(const webrtc::Call::Stats& stats); | 206 void SetStats(const webrtc::Call::Stats& stats); |
207 | 207 |
208 private: | 208 private: |
209 webrtc::AudioSendStream* CreateAudioSendStream( | 209 webrtc::AudioSendStream* CreateAudioSendStream( |
210 const webrtc::AudioSendStream::Config& config) override; | 210 const webrtc::AudioSendStream::Config& config) override; |
211 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; | 211 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; |
212 | 212 |
213 webrtc::AudioReceiveStream* CreateAudioReceiveStream( | 213 webrtc::AudioReceiveStream* CreateAudioReceiveStream( |
(...skipping 14 matching lines...) Expand all Loading... |
228 | 228 |
229 DeliveryStatus DeliverPacket(webrtc::MediaType media_type, | 229 DeliveryStatus DeliverPacket(webrtc::MediaType media_type, |
230 const uint8_t* packet, | 230 const uint8_t* packet, |
231 size_t length, | 231 size_t length, |
232 const webrtc::PacketTime& packet_time) override; | 232 const webrtc::PacketTime& packet_time) override; |
233 | 233 |
234 webrtc::Call::Stats GetStats() const override; | 234 webrtc::Call::Stats GetStats() const override; |
235 | 235 |
236 void SetBitrateConfig( | 236 void SetBitrateConfig( |
237 const webrtc::Call::Config::BitrateConfig& bitrate_config) override; | 237 const webrtc::Call::Config::BitrateConfig& bitrate_config) override; |
238 void SignalNetworkState(webrtc::NetworkState state) override; | 238 void SignalChannelNetworkState(webrtc::MediaType media, |
| 239 webrtc::NetworkState state) override; |
239 void OnSentPacket(const rtc::SentPacket& sent_packet) override; | 240 void OnSentPacket(const rtc::SentPacket& sent_packet) override; |
240 | 241 |
241 webrtc::Call::Config config_; | 242 webrtc::Call::Config config_; |
242 webrtc::NetworkState network_state_; | 243 webrtc::NetworkState audio_network_state_; |
| 244 webrtc::NetworkState video_network_state_; |
243 rtc::SentPacket last_sent_packet_; | 245 rtc::SentPacket last_sent_packet_; |
244 webrtc::Call::Stats stats_; | 246 webrtc::Call::Stats stats_; |
245 std::vector<FakeVideoSendStream*> video_send_streams_; | 247 std::vector<FakeVideoSendStream*> video_send_streams_; |
246 std::vector<FakeAudioSendStream*> audio_send_streams_; | 248 std::vector<FakeAudioSendStream*> audio_send_streams_; |
247 std::vector<FakeVideoReceiveStream*> video_receive_streams_; | 249 std::vector<FakeVideoReceiveStream*> video_receive_streams_; |
248 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; | 250 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; |
249 | 251 |
250 int num_created_send_streams_; | 252 int num_created_send_streams_; |
251 int num_created_receive_streams_; | 253 int num_created_receive_streams_; |
252 }; | 254 }; |
253 | 255 |
254 } // namespace cricket | 256 } // namespace cricket |
255 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ | 257 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ |
OLD | NEW |