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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: Added handling for the case where the enum class value is outside of the valid range 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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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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