Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(82)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Issue 2812913002: Reland of Add content type information to encoded images and corresponding rtp extension header (Closed)
Patch Set: Fix indent Created 3 years, 8 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
OLDNEW
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
(...skipping 22 matching lines...) Expand all
33 if (extension == RtpExtension::kAudioLevelUri) 33 if (extension == RtpExtension::kAudioLevelUri)
34 return kRtpExtensionAudioLevel; 34 return kRtpExtensionAudioLevel;
35 if (extension == RtpExtension::kAbsSendTimeUri) 35 if (extension == RtpExtension::kAbsSendTimeUri)
36 return kRtpExtensionAbsoluteSendTime; 36 return kRtpExtensionAbsoluteSendTime;
37 if (extension == RtpExtension::kVideoRotationUri) 37 if (extension == RtpExtension::kVideoRotationUri)
38 return kRtpExtensionVideoRotation; 38 return kRtpExtensionVideoRotation;
39 if (extension == RtpExtension::kTransportSequenceNumberUri) 39 if (extension == RtpExtension::kTransportSequenceNumberUri)
40 return kRtpExtensionTransportSequenceNumber; 40 return kRtpExtensionTransportSequenceNumber;
41 if (extension == RtpExtension::kPlayoutDelayUri) 41 if (extension == RtpExtension::kPlayoutDelayUri)
42 return kRtpExtensionPlayoutDelay; 42 return kRtpExtensionPlayoutDelay;
43 if (extension == RtpExtension::kVideoContentTypeUri)
44 return kRtpExtensionVideoContentType;
43 RTC_NOTREACHED() << "Looking up unsupported RTP extension."; 45 RTC_NOTREACHED() << "Looking up unsupported RTP extension.";
44 return kRtpExtensionNone; 46 return kRtpExtensionNone;
45 } 47 }
46 48
47 RtpRtcp::Configuration::Configuration() 49 RtpRtcp::Configuration::Configuration()
48 : receive_statistics(NullObjectReceiveStatistics()) {} 50 : receive_statistics(NullObjectReceiveStatistics()) {}
49 51
50 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) { 52 RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
51 if (configuration.clock) { 53 if (configuration.clock) {
52 return new ModuleRtpRtcpImpl(configuration); 54 return new ModuleRtpRtcpImpl(configuration);
(...skipping 833 matching lines...) Expand 10 before | Expand all | Expand 10 after
886 StreamDataCountersCallback* 888 StreamDataCountersCallback*
887 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 889 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
888 return rtp_sender_->GetRtpStatisticsCallback(); 890 return rtp_sender_->GetRtpStatisticsCallback();
889 } 891 }
890 892
891 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation( 893 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
892 const BitrateAllocation& bitrate) { 894 const BitrateAllocation& bitrate) {
893 rtcp_sender_.SetVideoBitrateAllocation(bitrate); 895 rtcp_sender_.SetVideoBitrateAllocation(bitrate);
894 } 896 }
895 } // namespace webrtc 897 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698