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

Issue 1757683002: Make the audio channel communicate network state changes to the call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Updated with code review feedback Created 4 years, 9 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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191 webrtc::Call::Config GetConfig() const; 191 webrtc::Call::Config GetConfig() const;
192 const std::vector<FakeVideoSendStream*>& GetVideoSendStreams(); 192 const std::vector<FakeVideoSendStream*>& GetVideoSendStreams();
193 const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams(); 193 const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams();
194 194
195 const std::vector<FakeAudioSendStream*>& GetAudioSendStreams(); 195 const std::vector<FakeAudioSendStream*>& GetAudioSendStreams();
196 const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc); 196 const FakeAudioSendStream* GetAudioSendStream(uint32_t ssrc);
197 const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams(); 197 const std::vector<FakeAudioReceiveStream*>& GetAudioReceiveStreams();
198 const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc); 198 const FakeAudioReceiveStream* GetAudioReceiveStream(uint32_t ssrc);
199 199
200 rtc::SentPacket last_sent_packet() const { return last_sent_packet_; } 200 rtc::SentPacket last_sent_packet() const { return last_sent_packet_; }
201 webrtc::NetworkState GetNetworkState() const; 201 webrtc::NetworkState GetNetworkState(webrtc::MediaType media) const;
202 int GetNumCreatedSendStreams() const; 202 int GetNumCreatedSendStreams() const;
203 int GetNumCreatedReceiveStreams() const; 203 int GetNumCreatedReceiveStreams() const;
204 void SetStats(const webrtc::Call::Stats& stats); 204 void SetStats(const webrtc::Call::Stats& stats);
205 205
206 private: 206 private:
207 webrtc::AudioSendStream* CreateAudioSendStream( 207 webrtc::AudioSendStream* CreateAudioSendStream(
208 const webrtc::AudioSendStream::Config& config) override; 208 const webrtc::AudioSendStream::Config& config) override;
209 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; 209 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
210 210
211 webrtc::AudioReceiveStream* CreateAudioReceiveStream( 211 webrtc::AudioReceiveStream* CreateAudioReceiveStream(
(...skipping 14 matching lines...) Expand all
226 226
227 DeliveryStatus DeliverPacket(webrtc::MediaType media_type, 227 DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
228 const uint8_t* packet, 228 const uint8_t* packet,
229 size_t length, 229 size_t length,
230 const webrtc::PacketTime& packet_time) override; 230 const webrtc::PacketTime& packet_time) override;
231 231
232 webrtc::Call::Stats GetStats() const override; 232 webrtc::Call::Stats GetStats() const override;
233 233
234 void SetBitrateConfig( 234 void SetBitrateConfig(
235 const webrtc::Call::Config::BitrateConfig& bitrate_config) override; 235 const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
236 void SignalNetworkState(webrtc::NetworkState state) override; 236 void SignalChannelNetworkState(webrtc::MediaType media,
237 webrtc::NetworkState state) override;
237 void OnSentPacket(const rtc::SentPacket& sent_packet) override; 238 void OnSentPacket(const rtc::SentPacket& sent_packet) override;
238 239
239 webrtc::Call::Config config_; 240 webrtc::Call::Config config_;
240 webrtc::NetworkState network_state_; 241 webrtc::NetworkState audio_network_state_;
242 webrtc::NetworkState video_network_state_;
241 rtc::SentPacket last_sent_packet_; 243 rtc::SentPacket last_sent_packet_;
242 webrtc::Call::Stats stats_; 244 webrtc::Call::Stats stats_;
243 std::vector<FakeVideoSendStream*> video_send_streams_; 245 std::vector<FakeVideoSendStream*> video_send_streams_;
244 std::vector<FakeAudioSendStream*> audio_send_streams_; 246 std::vector<FakeAudioSendStream*> audio_send_streams_;
245 std::vector<FakeVideoReceiveStream*> video_receive_streams_; 247 std::vector<FakeVideoReceiveStream*> video_receive_streams_;
246 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; 248 std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
247 249
248 int num_created_send_streams_; 250 int num_created_send_streams_;
249 int num_created_receive_streams_; 251 int num_created_receive_streams_;
250 }; 252 };
251 253
252 } // namespace cricket 254 } // namespace cricket
253 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ 255 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_
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