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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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