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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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| 67   void CreateCalls(const Call::Config& sender_config, | 67   void CreateCalls(const Call::Config& sender_config, | 
| 68                    const Call::Config& receiver_config); | 68                    const Call::Config& receiver_config); | 
| 69   void CreateSenderCall(const Call::Config& config); | 69   void CreateSenderCall(const Call::Config& config); | 
| 70   void CreateReceiverCall(const Call::Config& config); | 70   void CreateReceiverCall(const Call::Config& config); | 
| 71   void DestroyCalls(); | 71   void DestroyCalls(); | 
| 72 | 72 | 
| 73   void CreateSendConfig(size_t num_video_streams, | 73   void CreateSendConfig(size_t num_video_streams, | 
| 74                         size_t num_audio_streams, | 74                         size_t num_audio_streams, | 
| 75                         size_t num_flexfec_streams, | 75                         size_t num_flexfec_streams, | 
| 76                         Transport* send_transport); | 76                         Transport* send_transport); | 
| 77 |  | 
| 78   void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport); | 77   void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport); | 
| 79 | 78 | 
| 80   void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock, | 79   void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock, | 
| 81                                              float speed, | 80                                              float speed, | 
| 82                                              int framerate, | 81                                              int framerate, | 
| 83                                              int width, | 82                                              int width, | 
| 84                                              int height); | 83                                              int height); | 
| 85   void CreateFrameGeneratorCapturer(int framerate, int width, int height); | 84   void CreateFrameGeneratorCapturer(int framerate, int width, int height); | 
| 86   void CreateFakeAudioDevices(); | 85   void CreateFakeAudioDevices(); | 
| 87 | 86 | 
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| 207  public: | 206  public: | 
| 208   explicit EndToEndTest(unsigned int timeout_ms); | 207   explicit EndToEndTest(unsigned int timeout_ms); | 
| 209 | 208 | 
| 210   bool ShouldCreateReceivers() const override; | 209   bool ShouldCreateReceivers() const override; | 
| 211 }; | 210 }; | 
| 212 | 211 | 
| 213 }  // namespace test | 212 }  // namespace test | 
| 214 }  // namespace webrtc | 213 }  // namespace webrtc | 
| 215 | 214 | 
| 216 #endif  // WEBRTC_TEST_CALL_TEST_H_ | 215 #endif  // WEBRTC_TEST_CALL_TEST_H_ | 
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