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

Issue 1181653002: Base A/V synchronization on sync_labels. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 5 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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197 test_->encoder_config_.streams[0].width, 197 test_->encoder_config_.streams[0].width,
198 test_->encoder_config_.streams[0].height, 198 test_->encoder_config_.streams[0].height,
199 30, 199 30,
200 Clock::GetRealTimeClock())); 200 Clock::GetRealTimeClock()));
201 send_stream_->Start(); 201 send_stream_->Start();
202 frame_generator_capturer_->Start(); 202 frame_generator_capturer_->Start();
203 203
204 if (receive_audio) { 204 if (receive_audio) {
205 AudioReceiveStream::Config receive_config; 205 AudioReceiveStream::Config receive_config;
206 receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0]; 206 receive_config.rtp.remote_ssrc = test_->send_config_.rtp.ssrcs[0];
207 // Bogus non-default id to prevent hitting a DCHECK when creating the
208 // AudioReceiveStream. Every receive stream has to correspond to an
209 // underlying channel id.
210 receive_config.voe_channel_id = 0;
207 receive_config.rtp.extensions.push_back( 211 receive_config.rtp.extensions.push_back(
208 RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId)); 212 RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
209 audio_receive_stream_ = test_->receiver_call_->CreateAudioReceiveStream( 213 audio_receive_stream_ = test_->receiver_call_->CreateAudioReceiveStream(
210 receive_config); 214 receive_config);
211 } else { 215 } else {
212 VideoReceiveStream::Decoder decoder; 216 VideoReceiveStream::Decoder decoder;
213 decoder.decoder = &fake_decoder_; 217 decoder.decoder = &fake_decoder_;
214 decoder.payload_type = 218 decoder.payload_type =
215 test_->send_config_.encoder_settings.payload_type; 219 test_->send_config_.encoder_settings.payload_type;
216 decoder.payload_name = 220 decoder.payload_name =
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357 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId); 361 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
358 receiver_trace_.PushExpectedLogLine( 362 receiver_trace_.PushExpectedLogLine(
359 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); 363 "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
360 receiver_trace_.PushExpectedLogLine(kSingleStreamLog); 364 receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
361 streams_.push_back(new Stream(this, false)); 365 streams_.push_back(new Stream(this, false));
362 streams_[0]->StopSending(); 366 streams_[0]->StopSending();
363 streams_[1]->StopSending(); 367 streams_[1]->StopSending();
364 EXPECT_EQ(kEventSignaled, receiver_trace_.Wait()); 368 EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
365 } 369 }
366 } // namespace webrtc 370 } // namespace webrtc
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