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1 /* | 1 /* |
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 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 #include <functional> | 10 #include <functional> |
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166 is_sending_receiving_(false), | 166 is_sending_receiving_(false), |
167 send_stream_(nullptr), | 167 send_stream_(nullptr), |
168 audio_receive_stream_(nullptr), | 168 audio_receive_stream_(nullptr), |
169 video_receive_stream_(nullptr), | 169 video_receive_stream_(nullptr), |
170 frame_generator_capturer_(), | 170 frame_generator_capturer_(), |
171 fake_encoder_(Clock::GetRealTimeClock()), | 171 fake_encoder_(Clock::GetRealTimeClock()), |
172 fake_decoder_() { | 172 fake_decoder_() { |
173 test_->video_send_config_.rtp.ssrcs[0]++; | 173 test_->video_send_config_.rtp.ssrcs[0]++; |
174 test_->video_send_config_.encoder_settings.encoder = &fake_encoder_; | 174 test_->video_send_config_.encoder_settings.encoder = &fake_encoder_; |
175 send_stream_ = test_->sender_call_->CreateVideoSendStream( | 175 send_stream_ = test_->sender_call_->CreateVideoSendStream( |
176 test_->video_send_config_.Copy(), | 176 test_->video_send_config_, test_->video_encoder_config_); |
177 test_->video_encoder_config_.Copy()); | |
178 RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size()); | 177 RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size()); |
179 frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create( | 178 frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create( |
180 send_stream_->Input(), test_->video_encoder_config_.streams[0].width, | 179 send_stream_->Input(), test_->video_encoder_config_.streams[0].width, |
181 test_->video_encoder_config_.streams[0].height, 30, | 180 test_->video_encoder_config_.streams[0].height, 30, |
182 Clock::GetRealTimeClock())); | 181 Clock::GetRealTimeClock())); |
183 send_stream_->Start(); | 182 send_stream_->Start(); |
184 frame_generator_capturer_->Start(); | 183 frame_generator_capturer_->Start(); |
185 | 184 |
186 if (receive_audio) { | 185 if (receive_audio) { |
187 AudioReceiveStream::Config receive_config; | 186 AudioReceiveStream::Config receive_config; |
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326 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId); | 325 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId); |
327 receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); | 326 receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); |
328 receiver_log_.PushExpectedLogLine( | 327 receiver_log_.PushExpectedLogLine( |
329 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); | 328 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); |
330 streams_.push_back(new Stream(this, false)); | 329 streams_.push_back(new Stream(this, false)); |
331 streams_[0]->StopSending(); | 330 streams_[0]->StopSending(); |
332 streams_[1]->StopSending(); | 331 streams_[1]->StopSending(); |
333 EXPECT_TRUE(receiver_log_.Wait()); | 332 EXPECT_TRUE(receiver_log_.Wait()); |
334 } | 333 } |
335 } // namespace webrtc | 334 } // namespace webrtc |
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