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Side by Side Diff: webrtc/video_engine/vie_channel.cc

Issue 1392513002: Disable pacer disabling. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 2 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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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695 } 695 }
696 696
697 int ViEChannel::SetReceiveTransportSequenceNumber(bool enable, int id) { 697 int ViEChannel::SetReceiveTransportSequenceNumber(bool enable, int id) {
698 return vie_receiver_.SetReceiveTransportSequenceNumber(enable, id) ? 0 : -1; 698 return vie_receiver_.SetReceiveTransportSequenceNumber(enable, id) ? 0 : -1;
699 } 699 }
700 700
701 void ViEChannel::SetRtcpXrRrtrStatus(bool enable) { 701 void ViEChannel::SetRtcpXrRrtrStatus(bool enable) {
702 rtp_rtcp_modules_[0]->SetRtcpXrRrtrStatus(enable); 702 rtp_rtcp_modules_[0]->SetRtcpXrRrtrStatus(enable);
703 } 703 }
704 704
705 void ViEChannel::SetTransmissionSmoothingStatus(bool enable) {
706 RTC_DCHECK(paced_sender_ && "No paced sender registered.");
707 paced_sender_->SetStatus(enable);
708 }
709
710 void ViEChannel::EnableTMMBR(bool enable) { 705 void ViEChannel::EnableTMMBR(bool enable) {
711 rtp_rtcp_modules_[0]->SetTMMBRStatus(enable); 706 rtp_rtcp_modules_[0]->SetTMMBRStatus(enable);
712 } 707 }
713 708
714 int32_t ViEChannel::SetSSRC(const uint32_t SSRC, 709 int32_t ViEChannel::SetSSRC(const uint32_t SSRC,
715 const StreamType usage, 710 const StreamType usage,
716 const uint8_t simulcast_idx) { 711 const uint8_t simulcast_idx) {
717 RtpRtcp* rtp_rtcp = rtp_rtcp_modules_[simulcast_idx]; 712 RtpRtcp* rtp_rtcp = rtp_rtcp_modules_[simulcast_idx];
718 if (usage == kViEStreamTypeRtx) { 713 if (usage == kViEStreamTypeRtx) {
719 rtp_rtcp->SetRtxSsrc(SSRC); 714 rtp_rtcp->SetRtxSsrc(SSRC);
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1255 CriticalSectionScoped cs(crit_.get()); 1250 CriticalSectionScoped cs(crit_.get());
1256 receive_stats_callback_ = receive_statistics_proxy; 1251 receive_stats_callback_ = receive_statistics_proxy;
1257 } 1252 }
1258 1253
1259 void ViEChannel::SetIncomingVideoStream( 1254 void ViEChannel::SetIncomingVideoStream(
1260 IncomingVideoStream* incoming_video_stream) { 1255 IncomingVideoStream* incoming_video_stream) {
1261 CriticalSectionScoped cs(crit_.get()); 1256 CriticalSectionScoped cs(crit_.get());
1262 incoming_video_stream_ = incoming_video_stream; 1257 incoming_video_stream_ = incoming_video_stream;
1263 } 1258 }
1264 } // namespace webrtc 1259 } // namespace webrtc
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