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

Issue 2922953002: Reland of Protect new header extension by field trial experiment to allow hardcoding it in SDP (Closed)
Patch Set: Move field_trial check to VideoSendStreamImpl Created 3 years, 6 months 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 #include "webrtc/video/video_send_stream.h" 10 #include "webrtc/video/video_send_stream.h"
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821 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) 821 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
822 transport->packet_router()->AddSendRtpModule(rtp_rtcp); 822 transport->packet_router()->AddSendRtpModule(rtp_rtcp);
823 823
824 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) { 824 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) {
825 const std::string& extension = config_->rtp.extensions[i].uri; 825 const std::string& extension = config_->rtp.extensions[i].uri;
826 int id = config_->rtp.extensions[i].id; 826 int id = config_->rtp.extensions[i].id;
827 // One-byte-extension local identifiers are in the range 1-14 inclusive. 827 // One-byte-extension local identifiers are in the range 1-14 inclusive.
828 RTC_DCHECK_GE(id, 1); 828 RTC_DCHECK_GE(id, 1);
829 RTC_DCHECK_LE(id, 14); 829 RTC_DCHECK_LE(id, 14);
830 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); 830 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
831 if (StringToRtpExtensionType(extension) == kRtpExtensionVideoContentType &&
832 !field_trial::IsEnabled("WebRTC-VideoContentTypeExtension")) {
833 continue;
834 }
831 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { 835 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
832 RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension( 836 RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
833 StringToRtpExtensionType(extension), id)); 837 StringToRtpExtensionType(extension), id));
834 } 838 }
835 } 839 }
836 840
837 ConfigureProtection(); 841 ConfigureProtection();
838 ConfigureSsrcs(); 842 ConfigureSsrcs();
839 843
840 // TODO(pbos): Should we set CNAME on all RTP modules? 844 // TODO(pbos): Should we set CNAME on all RTP modules?
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1336 std::min(config_->rtp.max_packet_size, 1340 std::min(config_->rtp.max_packet_size,
1337 kPathMTU - transport_overhead_bytes_per_packet_); 1341 kPathMTU - transport_overhead_bytes_per_packet_);
1338 1342
1339 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { 1343 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
1340 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); 1344 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
1341 } 1345 }
1342 } 1346 }
1343 1347
1344 } // namespace internal 1348 } // namespace internal
1345 } // namespace webrtc 1349 } // namespace webrtc
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