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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 | 10 |
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292 int width, | 292 int width, |
293 int height) { | 293 int height) { |
294 frame_generator_capturer_.reset( | 294 frame_generator_capturer_.reset( |
295 test::FrameGeneratorCapturer::Create(width, height, framerate, clock_)); | 295 test::FrameGeneratorCapturer::Create(width, height, framerate, clock_)); |
296 video_send_stream_->SetSource( | 296 video_send_stream_->SetSource( |
297 frame_generator_capturer_.get(), | 297 frame_generator_capturer_.get(), |
298 VideoSendStream::DegradationPreference::kBalanced); | 298 VideoSendStream::DegradationPreference::kBalanced); |
299 } | 299 } |
300 | 300 |
301 void CallTest::CreateFakeAudioDevices() { | 301 void CallTest::CreateFakeAudioDevices() { |
302 fake_send_audio_device_.reset(new FakeAudioDevice(1.f, 48000, 256)); | 302 fake_send_audio_device_.reset(new FakeAudioDevice( |
303 fake_recv_audio_device_.reset(new FakeAudioDevice(1.f, 48000, 256)); | 303 FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000), nullptr, 1.f)); |
| 304 fake_recv_audio_device_.reset(new FakeAudioDevice( |
| 305 nullptr, FakeAudioDevice::CreateDiscardRenderer(48000), 1.f)); |
304 } | 306 } |
305 | 307 |
306 void CallTest::CreateVideoStreams() { | 308 void CallTest::CreateVideoStreams() { |
307 RTC_DCHECK(video_send_stream_ == nullptr); | 309 RTC_DCHECK(video_send_stream_ == nullptr); |
308 RTC_DCHECK(video_receive_streams_.empty()); | 310 RTC_DCHECK(video_receive_streams_.empty()); |
309 RTC_DCHECK(audio_send_stream_ == nullptr); | 311 RTC_DCHECK(audio_send_stream_ == nullptr); |
310 RTC_DCHECK(audio_receive_streams_.empty()); | 312 RTC_DCHECK(audio_receive_streams_.empty()); |
311 | 313 |
312 video_send_stream_ = sender_call_->CreateVideoSendStream( | 314 video_send_stream_ = sender_call_->CreateVideoSendStream( |
313 video_send_config_.Copy(), video_encoder_config_.Copy()); | 315 video_send_config_.Copy(), video_encoder_config_.Copy()); |
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495 | 497 |
496 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { | 498 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
497 } | 499 } |
498 | 500 |
499 bool EndToEndTest::ShouldCreateReceivers() const { | 501 bool EndToEndTest::ShouldCreateReceivers() const { |
500 return true; | 502 return true; |
501 } | 503 } |
502 | 504 |
503 } // namespace test | 505 } // namespace test |
504 } // namespace webrtc | 506 } // namespace webrtc |
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