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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 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 #ifndef WEBRTC_TEST_CALL_TEST_H_ | 10 #ifndef WEBRTC_TEST_CALL_TEST_H_ |
11 #define WEBRTC_TEST_CALL_TEST_H_ | 11 #define WEBRTC_TEST_CALL_TEST_H_ |
12 | 12 |
13 #include <memory> | 13 #include <memory> |
14 #include <vector> | 14 #include <vector> |
15 | 15 |
16 #include "webrtc/call/call.h" | 16 #include "webrtc/call/call.h" |
| 17 #include "webrtc/call/rtp_transport_controller_send.h" |
17 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" | 18 #include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
18 #include "webrtc/test/encoder_settings.h" | 19 #include "webrtc/test/encoder_settings.h" |
19 #include "webrtc/test/fake_audio_device.h" | 20 #include "webrtc/test/fake_audio_device.h" |
20 #include "webrtc/test/fake_decoder.h" | 21 #include "webrtc/test/fake_decoder.h" |
21 #include "webrtc/test/fake_encoder.h" | 22 #include "webrtc/test/fake_encoder.h" |
22 #include "webrtc/test/fake_videorenderer.h" | 23 #include "webrtc/test/fake_videorenderer.h" |
23 #include "webrtc/test/frame_generator_capturer.h" | 24 #include "webrtc/test/frame_generator_capturer.h" |
24 #include "webrtc/test/rtp_rtcp_observer.h" | 25 #include "webrtc/test/rtp_rtcp_observer.h" |
25 | 26 |
26 namespace webrtc { | 27 namespace webrtc { |
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101 | 102 |
102 void Start(); | 103 void Start(); |
103 void Stop(); | 104 void Stop(); |
104 void DestroyStreams(); | 105 void DestroyStreams(); |
105 void SetFakeVideoCaptureRotation(VideoRotation rotation); | 106 void SetFakeVideoCaptureRotation(VideoRotation rotation); |
106 | 107 |
107 Clock* const clock_; | 108 Clock* const clock_; |
108 | 109 |
109 std::unique_ptr<webrtc::RtcEventLog> event_log_; | 110 std::unique_ptr<webrtc::RtcEventLog> event_log_; |
110 std::unique_ptr<Call> sender_call_; | 111 std::unique_ptr<Call> sender_call_; |
| 112 RtpTransportControllerSend* sender_call_transport_controller_; |
111 std::unique_ptr<PacketTransport> send_transport_; | 113 std::unique_ptr<PacketTransport> send_transport_; |
112 VideoSendStream::Config video_send_config_; | 114 VideoSendStream::Config video_send_config_; |
113 VideoEncoderConfig video_encoder_config_; | 115 VideoEncoderConfig video_encoder_config_; |
114 VideoSendStream* video_send_stream_; | 116 VideoSendStream* video_send_stream_; |
115 AudioSendStream::Config audio_send_config_; | 117 AudioSendStream::Config audio_send_config_; |
116 AudioSendStream* audio_send_stream_; | 118 AudioSendStream* audio_send_stream_; |
117 | 119 |
118 std::unique_ptr<Call> receiver_call_; | 120 std::unique_ptr<Call> receiver_call_; |
119 std::unique_ptr<PacketTransport> receive_transport_; | 121 std::unique_ptr<PacketTransport> receive_transport_; |
120 std::vector<VideoReceiveStream::Config> video_receive_configs_; | 122 std::vector<VideoReceiveStream::Config> video_receive_configs_; |
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175 virtual size_t GetNumAudioStreams() const; | 177 virtual size_t GetNumAudioStreams() const; |
176 virtual size_t GetNumFlexfecStreams() const; | 178 virtual size_t GetNumFlexfecStreams() const; |
177 | 179 |
178 virtual std::unique_ptr<FakeAudioDevice::Capturer> CreateCapturer(); | 180 virtual std::unique_ptr<FakeAudioDevice::Capturer> CreateCapturer(); |
179 virtual std::unique_ptr<FakeAudioDevice::Renderer> CreateRenderer(); | 181 virtual std::unique_ptr<FakeAudioDevice::Renderer> CreateRenderer(); |
180 virtual void OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device, | 182 virtual void OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device, |
181 FakeAudioDevice* recv_audio_device); | 183 FakeAudioDevice* recv_audio_device); |
182 | 184 |
183 virtual Call::Config GetSenderCallConfig(); | 185 virtual Call::Config GetSenderCallConfig(); |
184 virtual Call::Config GetReceiverCallConfig(); | 186 virtual Call::Config GetReceiverCallConfig(); |
| 187 virtual void OnRtpTransportControllerSendCreated( |
| 188 RtpTransportControllerSend* controller); |
185 virtual void OnCallsCreated(Call* sender_call, Call* receiver_call); | 189 virtual void OnCallsCreated(Call* sender_call, Call* receiver_call); |
186 | 190 |
187 virtual test::PacketTransport* CreateSendTransport(Call* sender_call); | 191 virtual test::PacketTransport* CreateSendTransport(Call* sender_call); |
188 virtual test::PacketTransport* CreateReceiveTransport(); | 192 virtual test::PacketTransport* CreateReceiveTransport(); |
189 | 193 |
190 virtual void ModifyVideoConfigs( | 194 virtual void ModifyVideoConfigs( |
191 VideoSendStream::Config* send_config, | 195 VideoSendStream::Config* send_config, |
192 std::vector<VideoReceiveStream::Config>* receive_configs, | 196 std::vector<VideoReceiveStream::Config>* receive_configs, |
193 VideoEncoderConfig* encoder_config); | 197 VideoEncoderConfig* encoder_config); |
194 virtual void ModifyVideoCaptureStartResolution(int* width, | 198 virtual void ModifyVideoCaptureStartResolution(int* width, |
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230 EndToEndTest(); | 234 EndToEndTest(); |
231 explicit EndToEndTest(unsigned int timeout_ms); | 235 explicit EndToEndTest(unsigned int timeout_ms); |
232 | 236 |
233 bool ShouldCreateReceivers() const override; | 237 bool ShouldCreateReceivers() const override; |
234 }; | 238 }; |
235 | 239 |
236 } // namespace test | 240 } // namespace test |
237 } // namespace webrtc | 241 } // namespace webrtc |
238 | 242 |
239 #endif // WEBRTC_TEST_CALL_TEST_H_ | 243 #endif // WEBRTC_TEST_CALL_TEST_H_ |
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