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Issue 1484443003: Remove duplicate decoders in BitrateEstimatorTest. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years 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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203 receive_config.combined_audio_video_bwe = true; 203 receive_config.combined_audio_video_bwe = true;
204 audio_receive_stream_ = 204 audio_receive_stream_ =
205 test_->receiver_call_->CreateAudioReceiveStream(receive_config); 205 test_->receiver_call_->CreateAudioReceiveStream(receive_config);
206 } else { 206 } else {
207 VideoReceiveStream::Decoder decoder; 207 VideoReceiveStream::Decoder decoder;
208 decoder.decoder = &fake_decoder_; 208 decoder.decoder = &fake_decoder_;
209 decoder.payload_type = 209 decoder.payload_type =
210 test_->send_config_.encoder_settings.payload_type; 210 test_->send_config_.encoder_settings.payload_type;
211 decoder.payload_name = 211 decoder.payload_name =
212 test_->send_config_.encoder_settings.payload_name; 212 test_->send_config_.encoder_settings.payload_name;
213 test_->receive_config_.decoders.clear();
213 test_->receive_config_.decoders.push_back(decoder); 214 test_->receive_config_.decoders.push_back(decoder);
214 test_->receive_config_.rtp.remote_ssrc = 215 test_->receive_config_.rtp.remote_ssrc =
215 test_->send_config_.rtp.ssrcs[0]; 216 test_->send_config_.rtp.ssrcs[0];
216 test_->receive_config_.rtp.local_ssrc++; 217 test_->receive_config_.rtp.local_ssrc++;
217 video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream( 218 video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
218 test_->receive_config_); 219 test_->receive_config_);
219 video_receive_stream_->Start(); 220 video_receive_stream_->Start();
220 } 221 }
221 is_sending_receiving_ = true; 222 is_sending_receiving_ = true;
222 } 223 }
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353 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId); 354 RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
354 receiver_trace_.PushExpectedLogLine( 355 receiver_trace_.PushExpectedLogLine(
355 "WrappingBitrateEstimator: Switching to transmission time offset RBE."); 356 "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
356 receiver_trace_.PushExpectedLogLine(kSingleStreamLog); 357 receiver_trace_.PushExpectedLogLine(kSingleStreamLog);
357 streams_.push_back(new Stream(this, false)); 358 streams_.push_back(new Stream(this, false));
358 streams_[0]->StopSending(); 359 streams_[0]->StopSending();
359 streams_[1]->StopSending(); 360 streams_[1]->StopSending();
360 EXPECT_EQ(kEventSignaled, receiver_trace_.Wait()); 361 EXPECT_EQ(kEventSignaled, receiver_trace_.Wait());
361 } 362 }
362 } // namespace webrtc 363 } // namespace webrtc
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