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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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| 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( |
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| 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_ |
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