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

Issue 1433393002: Add separate send-side UMA stats for screenshare and video. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Reset stats and update histograms on content type change Created 5 years, 1 month 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 10
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432 for (size_t i = video_codec.numberOfSimulcastStreams; 432 for (size_t i = video_codec.numberOfSimulcastStreams;
433 i < config_.rtp.ssrcs.size(); ++i) { 433 i < config_.rtp.ssrcs.size(); ++i) {
434 stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]); 434 stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]);
435 } 435 }
436 436
437 RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0); 437 RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
438 vie_encoder_->SetMinTransmitBitrate(config.min_transmit_bitrate_bps / 1000); 438 vie_encoder_->SetMinTransmitBitrate(config.min_transmit_bitrate_bps / 1000);
439 439
440 encoder_config_ = config; 440 encoder_config_ = config;
441 use_config_bitrate_ = false; 441 use_config_bitrate_ = false;
442 stats_proxy_.SetCodecMode(video_codec.mode);
442 return true; 443 return true;
443 } 444 }
444 445
445 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) { 446 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
446 return vie_channel_->ReceivedRTCPPacket(packet, length) == 0; 447 return vie_channel_->ReceivedRTCPPacket(packet, length) == 0;
447 } 448 }
448 449
449 VideoSendStream::Stats VideoSendStream::GetStats() { 450 VideoSendStream::Stats VideoSendStream::GetStats() {
450 return stats_proxy_.GetStats(); 451 return stats_proxy_.GetStats();
451 } 452 }
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566 vie_encoder_->SetSsrcs(used_ssrcs); 567 vie_encoder_->SetSsrcs(used_ssrcs);
567 568
568 // Restart the media flow 569 // Restart the media flow
569 vie_encoder_->Restart(); 570 vie_encoder_->Restart();
570 571
571 return true; 572 return true;
572 } 573 }
573 574
574 } // namespace internal 575 } // namespace internal
575 } // namespace webrtc 576 } // namespace webrtc
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