Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(18)

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

Issue 3003673002: Remove WebRTC-videocontenttypeextension field trial completely (Closed)
Patch Set: Created 3 years, 4 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
« no previous file with comments | « webrtc/video/end_to_end_tests.cc ('k') | webrtc/video/video_send_stream_tests.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
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"
(...skipping 724 matching lines...) Expand 10 before | Expand all | Expand 10 after
735 transport->packet_router()->AddSendRtpModule(rtp_rtcp, remb_candidate); 735 transport->packet_router()->AddSendRtpModule(rtp_rtcp, remb_candidate);
736 } 736 }
737 737
738 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) { 738 for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) {
739 const std::string& extension = config_->rtp.extensions[i].uri; 739 const std::string& extension = config_->rtp.extensions[i].uri;
740 int id = config_->rtp.extensions[i].id; 740 int id = config_->rtp.extensions[i].id;
741 // One-byte-extension local identifiers are in the range 1-14 inclusive. 741 // One-byte-extension local identifiers are in the range 1-14 inclusive.
742 RTC_DCHECK_GE(id, 1); 742 RTC_DCHECK_GE(id, 1);
743 RTC_DCHECK_LE(id, 14); 743 RTC_DCHECK_LE(id, 14);
744 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension)); 744 RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
745 if (StringToRtpExtensionType(extension) == kRtpExtensionVideoContentType &&
746 !field_trial::IsEnabled("WebRTC-VideoContentTypeExtension")) {
747 continue;
748 }
749 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { 745 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
750 RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension( 746 RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
751 StringToRtpExtensionType(extension), id)); 747 StringToRtpExtensionType(extension), id));
752 } 748 }
753 } 749 }
754 750
755 ConfigureProtection(); 751 ConfigureProtection();
756 ConfigureSsrcs(); 752 ConfigureSsrcs();
757 753
758 // TODO(pbos): Should we set CNAME on all RTP modules? 754 // TODO(pbos): Should we set CNAME on all RTP modules?
(...skipping 497 matching lines...) Expand 10 before | Expand all | Expand 10 after
1256 std::min(config_->rtp.max_packet_size, 1252 std::min(config_->rtp.max_packet_size,
1257 kPathMTU - transport_overhead_bytes_per_packet_); 1253 kPathMTU - transport_overhead_bytes_per_packet_);
1258 1254
1259 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { 1255 for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
1260 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size); 1256 rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
1261 } 1257 }
1262 } 1258 }
1263 1259
1264 } // namespace internal 1260 } // namespace internal
1265 } // namespace webrtc 1261 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/video/end_to_end_tests.cc ('k') | webrtc/video/video_send_stream_tests.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698