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

Issue 1984983002: Remove use of RtpHeaderExtension and clean up (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Addressed nit Created 4 years, 7 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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 <stdio.h> 10 #include <stdio.h>
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879 video_send_config_.encoder_settings.payload_name = params_.common.codec; 879 video_send_config_.encoder_settings.payload_name = params_.common.codec;
880 video_send_config_.encoder_settings.payload_type = payload_type; 880 video_send_config_.encoder_settings.payload_type = payload_type;
881 video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 881 video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
882 video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; 882 video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
883 for (size_t i = 0; i < num_streams; ++i) 883 for (size_t i = 0; i < num_streams; ++i)
884 video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]); 884 video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
885 885
886 video_send_config_.rtp.extensions.clear(); 886 video_send_config_.rtp.extensions.clear();
887 if (params_.common.send_side_bwe) { 887 if (params_.common.send_side_bwe) {
888 video_send_config_.rtp.extensions.push_back( 888 video_send_config_.rtp.extensions.push_back(
889 RtpExtension(RtpExtension::kTransportSequenceNumber, 889 RtpExtension(RtpExtension::kTransportSequenceNumberUri,
890 test::kTransportSequenceNumberExtensionId)); 890 test::kTransportSequenceNumberExtensionId));
891 } else { 891 } else {
892 video_send_config_.rtp.extensions.push_back(RtpExtension( 892 video_send_config_.rtp.extensions.push_back(RtpExtension(
893 RtpExtension::kAbsSendTime, test::kAbsSendTimeExtensionId)); 893 RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
894 } 894 }
895 895
896 video_encoder_config_.min_transmit_bitrate_bps = 896 video_encoder_config_.min_transmit_bitrate_bps =
897 params_.common.min_transmit_bps; 897 params_.common.min_transmit_bps;
898 video_encoder_config_.streams = params_.ss.streams; 898 video_encoder_config_.streams = params_.ss.streams;
899 video_encoder_config_.spatial_layers = params_.ss.spatial_layers; 899 video_encoder_config_.spatial_layers = params_.ss.spatial_layers;
900 900
901 CreateMatchingReceiveConfigs(recv_transport); 901 CreateMatchingReceiveConfigs(recv_transport);
902 902
903 for (size_t i = 0; i < num_streams; ++i) { 903 for (size_t i = 0; i < num_streams; ++i) {
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1155 video_send_stream_->Stop(); 1155 video_send_stream_->Stop();
1156 receive_stream->Stop(); 1156 receive_stream->Stop();
1157 1157
1158 call->DestroyVideoReceiveStream(receive_stream); 1158 call->DestroyVideoReceiveStream(receive_stream);
1159 call->DestroyVideoSendStream(video_send_stream_); 1159 call->DestroyVideoSendStream(video_send_stream_);
1160 1160
1161 transport.StopSending(); 1161 transport.StopSending();
1162 } 1162 }
1163 1163
1164 } // namespace webrtc 1164 } // namespace webrtc
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