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Side by Side Diff: webrtc/call/bitrate_estimator_tests.cc

Issue 2250123002: Reland of Add task queue to Call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 4 years, 3 months ago
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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_, test_->video_encoder_config_); 176 test_->video_send_config_.Copy(),
177 test_->video_encoder_config_.Copy());
177 RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size()); 178 RTC_DCHECK_EQ(1u, test_->video_encoder_config_.streams.size());
178 frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create( 179 frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
179 send_stream_->Input(), test_->video_encoder_config_.streams[0].width, 180 send_stream_->Input(), test_->video_encoder_config_.streams[0].width,
180 test_->video_encoder_config_.streams[0].height, 30, 181 test_->video_encoder_config_.streams[0].height, 30,
181 Clock::GetRealTimeClock())); 182 Clock::GetRealTimeClock()));
182 send_stream_->Start(); 183 send_stream_->Start();
183 frame_generator_capturer_->Start(); 184 frame_generator_capturer_->Start();
184 185
185 if (receive_audio) { 186 if (receive_audio) {
186 AudioReceiveStream::Config receive_config; 187 AudioReceiveStream::Config receive_config;
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322 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId); 323 RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
323 receiver_log_.PushExpectedLogLine(kAbsSendTimeLog); 324 receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
324 receiver_log_.PushExpectedLogLine( 325 receiver_log_.PushExpectedLogLine(
325 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); 326 "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
326 streams_.push_back(new Stream(this, false)); 327 streams_.push_back(new Stream(this, false));
327 streams_[0]->StopSending(); 328 streams_[0]->StopSending();
328 streams_[1]->StopSending(); 329 streams_[1]->StopSending();
329 EXPECT_TRUE(receiver_log_.Wait()); 330 EXPECT_TRUE(receiver_log_.Wait());
330 } 331 }
331 } // namespace webrtc 332 } // namespace webrtc
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