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Side by Side Diff: webrtc/test/call_test.h

Issue 2794193003: Move SelectMediaType from RampUpTester to BaseTest. (Closed)
Patch Set: Created 3 years, 8 months ago
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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_
(...skipping 174 matching lines...) Expand 10 before | Expand all | Expand 10 after
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);
(...skipping 33 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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