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

Issue 1452883002: Require negotiation to send transport cc feedback over RTCP. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: . Created 5 years, 1 month 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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821 encoder_config_.streams = params_.ss.streams; 821 encoder_config_.streams = params_.ss.streams;
822 encoder_config_.spatial_layers = params_.ss.spatial_layers; 822 encoder_config_.spatial_layers = params_.ss.spatial_layers;
823 823
824 CreateMatchingReceiveConfigs(recv_transport); 824 CreateMatchingReceiveConfigs(recv_transport);
825 825
826 for (size_t i = 0; i < num_streams; ++i) { 826 for (size_t i = 0; i < num_streams; ++i) {
827 receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 827 receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
828 receive_configs_[i].rtp.rtx[kSendRtxPayloadType].ssrc = kSendRtxSsrcs[i]; 828 receive_configs_[i].rtp.rtx[kSendRtxPayloadType].ssrc = kSendRtxSsrcs[i];
829 receive_configs_[i].rtp.rtx[kSendRtxPayloadType].payload_type = 829 receive_configs_[i].rtp.rtx[kSendRtxPayloadType].payload_type =
830 kSendRtxPayloadType; 830 kSendRtxPayloadType;
831 receive_configs_[i].rtp.transport_cc_feedback =
832 params_.common.send_side_bwe;
831 } 833 }
832 } 834 }
833 835
834 void VideoQualityTest::SetupScreenshare() { 836 void VideoQualityTest::SetupScreenshare() {
835 RTC_CHECK(params_.screenshare.enabled); 837 RTC_CHECK(params_.screenshare.enabled);
836 838
837 // Fill out codec settings. 839 // Fill out codec settings.
838 encoder_config_.content_type = VideoEncoderConfig::ContentType::kScreen; 840 encoder_config_.content_type = VideoEncoderConfig::ContentType::kScreen;
839 if (params_.common.codec == "VP8") { 841 if (params_.common.codec == "VP8") {
840 codec_settings_.VP8 = VideoEncoder::GetDefaultVp8Settings(); 842 codec_settings_.VP8 = VideoEncoder::GetDefaultVp8Settings();
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1063 send_stream_->Stop(); 1065 send_stream_->Stop();
1064 receive_stream->Stop(); 1066 receive_stream->Stop();
1065 1067
1066 call->DestroyVideoReceiveStream(receive_stream); 1068 call->DestroyVideoReceiveStream(receive_stream);
1067 call->DestroyVideoSendStream(send_stream_); 1069 call->DestroyVideoSendStream(send_stream_);
1068 1070
1069 transport.StopSending(); 1071 transport.StopSending();
1070 } 1072 }
1071 1073
1072 } // namespace webrtc 1074 } // namespace webrtc
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