Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1302)

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

Issue 2060403002: Add task queue to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@move_getpadding
Patch Set: Fix audio thread check when adding audio to bitrateallocator. Created 4 years, 4 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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 84 matching lines...) Expand 10 before | Expand all | Expand 10 after
95 int received_packets_ = 0; 95 int received_packets_ = 0;
96 std::unique_ptr<webrtc::AudioSinkInterface> sink_; 96 std::unique_ptr<webrtc::AudioSinkInterface> sink_;
97 float gain_ = 1.0f; 97 float gain_ = 1.0f;
98 rtc::Buffer last_packet_; 98 rtc::Buffer last_packet_;
99 bool started_ = false; 99 bool started_ = false;
100 }; 100 };
101 101
102 class FakeVideoSendStream final : public webrtc::VideoSendStream, 102 class FakeVideoSendStream final : public webrtc::VideoSendStream,
103 public webrtc::VideoCaptureInput { 103 public webrtc::VideoCaptureInput {
104 public: 104 public:
105 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, 105 FakeVideoSendStream(webrtc::VideoSendStream::Config config,
106 const webrtc::VideoEncoderConfig& encoder_config); 106 webrtc::VideoEncoderConfig encoder_config);
107 webrtc::VideoSendStream::Config GetConfig() const; 107 const webrtc::VideoSendStream::Config& GetConfig() const;
108 webrtc::VideoEncoderConfig GetEncoderConfig() const; 108 const webrtc::VideoEncoderConfig& GetEncoderConfig() const;
109 std::vector<webrtc::VideoStream> GetVideoStreams(); 109 std::vector<webrtc::VideoStream> GetVideoStreams();
110 110
111 bool IsSending() const; 111 bool IsSending() const;
112 bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const; 112 bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const;
113 bool GetVp9Settings(webrtc::VideoCodecVP9* settings) const; 113 bool GetVp9Settings(webrtc::VideoCodecVP9* settings) const;
114 114
115 int GetNumberOfSwappedFrames() const; 115 int GetNumberOfSwappedFrames() const;
116 int GetLastWidth() const; 116 int GetLastWidth() const;
117 int GetLastHeight() const; 117 int GetLastHeight() const;
118 int64_t GetLastTimestamp() const; 118 int64_t GetLastTimestamp() const;
119 void SetStats(const webrtc::VideoSendStream::Stats& stats); 119 void SetStats(const webrtc::VideoSendStream::Stats& stats);
120 int num_encoder_reconfigurations() const { 120 int num_encoder_reconfigurations() const {
121 return num_encoder_reconfigurations_; 121 return num_encoder_reconfigurations_;
122 } 122 }
123 123
124 private: 124 private:
125 void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override; 125 void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override;
126 126
127 // webrtc::VideoSendStream implementation. 127 // webrtc::VideoSendStream implementation.
128 void Start() override; 128 void Start() override;
129 void Stop() override; 129 void Stop() override;
130 webrtc::VideoSendStream::Stats GetStats() override; 130 webrtc::VideoSendStream::Stats GetStats() override;
131 void ReconfigureVideoEncoder( 131 void ReconfigureVideoEncoder(webrtc::VideoEncoderConfig config) override;
132 const webrtc::VideoEncoderConfig& config) override;
133 webrtc::VideoCaptureInput* Input() override; 132 webrtc::VideoCaptureInput* Input() override;
134 133
135 bool sending_; 134 bool sending_;
136 webrtc::VideoSendStream::Config config_; 135 webrtc::VideoSendStream::Config config_;
137 webrtc::VideoEncoderConfig encoder_config_; 136 webrtc::VideoEncoderConfig encoder_config_;
138 bool codec_settings_set_; 137 bool codec_settings_set_;
139 union VpxSettings { 138 union VpxSettings {
140 webrtc::VideoCodecVP8 vp8; 139 webrtc::VideoCodecVP8 vp8;
141 webrtc::VideoCodecVP9 vp9; 140 webrtc::VideoCodecVP9 vp9;
142 } vpx_settings_; 141 } vpx_settings_;
(...skipping 58 matching lines...) Expand 10 before | Expand all | Expand 10 after
201 webrtc::AudioSendStream* CreateAudioSendStream( 200 webrtc::AudioSendStream* CreateAudioSendStream(
202 const webrtc::AudioSendStream::Config& config) override; 201 const webrtc::AudioSendStream::Config& config) override;
203 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override; 202 void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
204 203
205 webrtc::AudioReceiveStream* CreateAudioReceiveStream( 204 webrtc::AudioReceiveStream* CreateAudioReceiveStream(
206 const webrtc::AudioReceiveStream::Config& config) override; 205 const webrtc::AudioReceiveStream::Config& config) override;
207 void DestroyAudioReceiveStream( 206 void DestroyAudioReceiveStream(
208 webrtc::AudioReceiveStream* receive_stream) override; 207 webrtc::AudioReceiveStream* receive_stream) override;
209 208
210 webrtc::VideoSendStream* CreateVideoSendStream( 209 webrtc::VideoSendStream* CreateVideoSendStream(
211 const webrtc::VideoSendStream::Config& config, 210 webrtc::VideoSendStream::Config config,
212 const webrtc::VideoEncoderConfig& encoder_config) override; 211 webrtc::VideoEncoderConfig encoder_config) override;
213 void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override; 212 void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
214 213
215 webrtc::VideoReceiveStream* CreateVideoReceiveStream( 214 webrtc::VideoReceiveStream* CreateVideoReceiveStream(
216 webrtc::VideoReceiveStream::Config config) override; 215 webrtc::VideoReceiveStream::Config config) override;
217 void DestroyVideoReceiveStream( 216 void DestroyVideoReceiveStream(
218 webrtc::VideoReceiveStream* receive_stream) override; 217 webrtc::VideoReceiveStream* receive_stream) override;
219 webrtc::PacketReceiver* Receiver() override; 218 webrtc::PacketReceiver* Receiver() override;
220 219
221 DeliveryStatus DeliverPacket(webrtc::MediaType media_type, 220 DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
222 const uint8_t* packet, 221 const uint8_t* packet,
(...skipping 24 matching lines...) Expand all
247 std::vector<FakeAudioSendStream*> audio_send_streams_; 246 std::vector<FakeAudioSendStream*> audio_send_streams_;
248 std::vector<FakeVideoReceiveStream*> video_receive_streams_; 247 std::vector<FakeVideoReceiveStream*> video_receive_streams_;
249 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; 248 std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
250 249
251 int num_created_send_streams_; 250 int num_created_send_streams_;
252 int num_created_receive_streams_; 251 int num_created_receive_streams_;
253 }; 252 };
254 253
255 } // namespace cricket 254 } // namespace cricket
256 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ 255 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698