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

Issue 1351693005: Read the number of TLs for VP9 too + cleanup (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase master Created 5 years, 2 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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604 604
605 send_config_.rtp.extensions.clear(); 605 send_config_.rtp.extensions.clear();
606 if (params.common.send_side_bwe) { 606 if (params.common.send_side_bwe) {
607 send_config_.rtp.extensions.push_back(RtpExtension( 607 send_config_.rtp.extensions.push_back(RtpExtension(
608 RtpExtension::kTransportSequenceNumber, kTransportSeqExtensionId)); 608 RtpExtension::kTransportSequenceNumber, kTransportSeqExtensionId));
609 } else { 609 } else {
610 send_config_.rtp.extensions.push_back( 610 send_config_.rtp.extensions.push_back(
611 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId)); 611 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
612 } 612 }
613 613
614 encoder_config_.disable_generic_bitrate_frame_dropper =
615 params.common.disable_generic_bitrate_frame_dropper;
616
614 // Automatically fill out streams[0] with params. 617 // Automatically fill out streams[0] with params.
615 VideoStream* stream = &encoder_config_.streams[0]; 618 VideoStream* stream = &encoder_config_.streams[0];
616 stream->width = params.common.width; 619 stream->width = params.common.width;
617 stream->height = params.common.height; 620 stream->height = params.common.height;
618 stream->min_bitrate_bps = params.common.min_bitrate_bps; 621 stream->min_bitrate_bps = params.common.min_bitrate_bps;
619 stream->target_bitrate_bps = params.common.target_bitrate_bps; 622 stream->target_bitrate_bps = params.common.target_bitrate_bps;
620 stream->max_bitrate_bps = params.common.max_bitrate_bps; 623 stream->max_bitrate_bps = params.common.max_bitrate_bps;
621 stream->max_framerate = static_cast<int>(params.common.fps); 624 stream->max_framerate = static_cast<int>(params.common.fps);
622 625
623 stream->temporal_layer_thresholds_bps.clear(); 626 stream->temporal_layer_thresholds_bps.clear();
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832 send_stream_->Stop(); 835 send_stream_->Stop();
833 receive_stream->Stop(); 836 receive_stream->Stop();
834 837
835 call->DestroyVideoReceiveStream(receive_stream); 838 call->DestroyVideoReceiveStream(receive_stream);
836 call->DestroyVideoSendStream(send_stream_); 839 call->DestroyVideoSendStream(send_stream_);
837 840
838 transport.StopSending(); 841 transport.StopSending();
839 } 842 }
840 843
841 } // namespace webrtc 844 } // namespace webrtc
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