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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_ |
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185 | 185 |
186 virtual std::unique_ptr<FakeAudioDevice::Capturer> CreateCapturer(); | 186 virtual std::unique_ptr<FakeAudioDevice::Capturer> CreateCapturer(); |
187 virtual std::unique_ptr<FakeAudioDevice::Renderer> CreateRenderer(); | 187 virtual std::unique_ptr<FakeAudioDevice::Renderer> CreateRenderer(); |
188 virtual void OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device, | 188 virtual void OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device, |
189 FakeAudioDevice* recv_audio_device); | 189 FakeAudioDevice* recv_audio_device); |
190 | 190 |
191 virtual Call::Config GetSenderCallConfig(); | 191 virtual Call::Config GetSenderCallConfig(); |
192 virtual Call::Config GetReceiverCallConfig(); | 192 virtual Call::Config GetReceiverCallConfig(); |
193 virtual void OnCallsCreated(Call* sender_call, Call* receiver_call); | 193 virtual void OnCallsCreated(Call* sender_call, Call* receiver_call); |
194 | 194 |
195 // The default implementation creates MediaType::VIDEO transports. | 195 // Returns VIDEO for video-only tests, AUDIO for audio-only tests, |
| 196 // and ANY for tests sending audio and video over the same |
| 197 // transport. |
| 198 virtual MediaType SelectMediaType(); |
196 virtual test::PacketTransport* CreateSendTransport(Call* sender_call); | 199 virtual test::PacketTransport* CreateSendTransport(Call* sender_call); |
197 virtual test::PacketTransport* CreateReceiveTransport(); | 200 virtual test::PacketTransport* CreateReceiveTransport(); |
198 | 201 |
199 virtual void ModifyVideoConfigs( | 202 virtual void ModifyVideoConfigs( |
200 VideoSendStream::Config* send_config, | 203 VideoSendStream::Config* send_config, |
201 std::vector<VideoReceiveStream::Config>* receive_configs, | 204 std::vector<VideoReceiveStream::Config>* receive_configs, |
202 VideoEncoderConfig* encoder_config); | 205 VideoEncoderConfig* encoder_config); |
203 virtual void ModifyVideoCaptureStartResolution(int* width, | 206 virtual void ModifyVideoCaptureStartResolution(int* width, |
204 int* heigt, | 207 int* heigt, |
205 int* frame_rate); | 208 int* frame_rate); |
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239 EndToEndTest(); | 242 EndToEndTest(); |
240 explicit EndToEndTest(unsigned int timeout_ms); | 243 explicit EndToEndTest(unsigned int timeout_ms); |
241 | 244 |
242 bool ShouldCreateReceivers() const override; | 245 bool ShouldCreateReceivers() const override; |
243 }; | 246 }; |
244 | 247 |
245 } // namespace test | 248 } // namespace test |
246 } // namespace webrtc | 249 } // namespace webrtc |
247 | 250 |
248 #endif // WEBRTC_TEST_CALL_TEST_H_ | 251 #endif // WEBRTC_TEST_CALL_TEST_H_ |
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