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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: Created 5 years, 3 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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591 return; 591 return;
592 } 592 }
593 send_config_.encoder_settings.encoder = encoder_.get(); 593 send_config_.encoder_settings.encoder = encoder_.get();
594 send_config_.encoder_settings.payload_name = params.common.codec; 594 send_config_.encoder_settings.payload_name = params.common.codec;
595 send_config_.encoder_settings.payload_type = payload_type; 595 send_config_.encoder_settings.payload_type = payload_type;
596 596
597 send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; 597 send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
598 send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]); 598 send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
599 send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; 599 send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
600 600
601 encoder_config_.force_disable_wrapper_frame_dropper =
602 params.common.disable_wrapper_frame_dropper;
603
601 // Automatically fill out streams[0] with params. 604 // Automatically fill out streams[0] with params.
602 VideoStream* stream = &encoder_config_.streams[0]; 605 VideoStream* stream = &encoder_config_.streams[0];
603 stream->width = params.common.width; 606 stream->width = params.common.width;
604 stream->height = params.common.height; 607 stream->height = params.common.height;
605 stream->min_bitrate_bps = params.common.min_bitrate_bps; 608 stream->min_bitrate_bps = params.common.min_bitrate_bps;
606 stream->target_bitrate_bps = params.common.target_bitrate_bps; 609 stream->target_bitrate_bps = params.common.target_bitrate_bps;
607 stream->max_bitrate_bps = params.common.max_bitrate_bps; 610 stream->max_bitrate_bps = params.common.max_bitrate_bps;
608 stream->max_framerate = static_cast<int>(params.common.fps); 611 stream->max_framerate = static_cast<int>(params.common.fps);
609 612
610 stream->temporal_layer_thresholds_bps.clear(); 613 stream->temporal_layer_thresholds_bps.clear();
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807 send_stream_->Stop(); 810 send_stream_->Stop();
808 receive_stream->Stop(); 811 receive_stream->Stop();
809 812
810 call->DestroyVideoReceiveStream(receive_stream); 813 call->DestroyVideoReceiveStream(receive_stream);
811 call->DestroyVideoSendStream(send_stream_); 814 call->DestroyVideoSendStream(send_stream_);
812 815
813 transport.StopSending(); 816 transport.StopSending();
814 } 817 }
815 818
816 } // namespace webrtc 819 } // namespace webrtc
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