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

Issue 2734753004: Revert of Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v… (Closed)
Patch Set: Created 3 years, 9 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 56 matching lines...) Expand 10 before | Expand all | Expand 10 after
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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