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

Issue 1394243006: Move ownership of send ViEChannels and ViEEncoder to VideoSendStream. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Removed header leftovers. 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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718 vie_receiver_.SetRtxSsrc(SSRC); 718 vie_receiver_.SetRtxSsrc(SSRC);
719 return 0; 719 return 0;
720 } 720 }
721 721
722 int32_t ViEChannel::GetLocalSSRC(uint8_t idx, unsigned int* ssrc) { 722 int32_t ViEChannel::GetLocalSSRC(uint8_t idx, unsigned int* ssrc) {
723 RTC_DCHECK_LE(idx, rtp_rtcp_modules_.size()); 723 RTC_DCHECK_LE(idx, rtp_rtcp_modules_.size());
724 *ssrc = rtp_rtcp_modules_[idx]->SSRC(); 724 *ssrc = rtp_rtcp_modules_[idx]->SSRC();
725 return 0; 725 return 0;
726 } 726 }
727 727
728 int32_t ViEChannel::GetRemoteSSRC(uint32_t* ssrc) { 728 uint32_t ViEChannel::GetRemoteSSRC() {
729 *ssrc = vie_receiver_.GetRemoteSsrc(); 729 return vie_receiver_.GetRemoteSsrc();
730 return 0;
731 } 730 }
732 731
733 int ViEChannel::SetRtxSendPayloadType(int payload_type, 732 int ViEChannel::SetRtxSendPayloadType(int payload_type,
734 int associated_payload_type) { 733 int associated_payload_type) {
735 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) 734 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
736 rtp_rtcp->SetRtxSendPayloadType(payload_type, associated_payload_type); 735 rtp_rtcp->SetRtxSendPayloadType(payload_type, associated_payload_type);
737 SetRtxSendStatus(true); 736 SetRtxSendStatus(true);
738 return 0; 737 return 0;
739 } 738 }
740 739
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1245 CriticalSectionScoped cs(crit_.get()); 1244 CriticalSectionScoped cs(crit_.get());
1246 receive_stats_callback_ = receive_statistics_proxy; 1245 receive_stats_callback_ = receive_statistics_proxy;
1247 } 1246 }
1248 1247
1249 void ViEChannel::SetIncomingVideoStream( 1248 void ViEChannel::SetIncomingVideoStream(
1250 IncomingVideoStream* incoming_video_stream) { 1249 IncomingVideoStream* incoming_video_stream) {
1251 CriticalSectionScoped cs(crit_.get()); 1250 CriticalSectionScoped cs(crit_.get());
1252 incoming_video_stream_ = incoming_video_stream; 1251 incoming_video_stream_ = incoming_video_stream;
1253 } 1252 }
1254 } // namespace webrtc 1253 } // namespace webrtc
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