Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(2190)

Unified Diff: webrtc/video/audio_receive_stream.cc

Issue 1247293002: Add support for transport wide sequence numbers (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 5 years, 5 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 side-by-side diff with in-line comments
Download patch
Index: webrtc/video/audio_receive_stream.cc
diff --git a/webrtc/video/audio_receive_stream.cc b/webrtc/video/audio_receive_stream.cc
index 88fd431f8255ed032d72100b88f9398aa3c56ff4..a95c06a4ce9e604746d9b6495b25054ce1c7b75c 100644
--- a/webrtc/video/audio_receive_stream.cc
+++ b/webrtc/video/audio_receive_stream.cc
@@ -61,6 +61,9 @@ AudioReceiveStream::AudioReceiveStream(
} else if (ext.name == RtpExtension::kAbsSendTime) {
CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime, ext.id));
+ } else if (ext.name == RtpExtension::kTransportSequenceNumber) {
+ CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
+ kRtpExtensionTransportSequenceNumber, ext.id));
} else {
RTC_NOTREACHED() << "Unsupported RTP extension.";
}

Powered by Google App Engine
This is Rietveld 408576698