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Unified Diff: webrtc/config.cc

Issue 1363573002: Wire up transport sequence number / send time callbacks to webrtc via libjingle. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Cleanups. Created 5 years, 2 months ago
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« talk/session/media/channel.cc ('K') | « webrtc/call/call.cc ('k') | webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h » ('j') | no next file with comments »
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Index: webrtc/config.cc
diff --git a/webrtc/config.cc b/webrtc/config.cc
index ddff931e241fc1ea22f01b09dd149c754eb9686c..3a74e5238e7d831a145ae97d33d54f6d76b10c51 100644
--- a/webrtc/config.cc
+++ b/webrtc/config.cc
@@ -36,7 +36,7 @@ const char* RtpExtension::kVideoRotation = "urn:3gpp:video-orientation";
const char* RtpExtension::kAudioLevel =
"urn:ietf:params:rtp-hdrext:ssrc-audio-level";
const char* RtpExtension::kTransportSequenceNumber =
- "http://www.webrtc.org/experiments/rtp-hdrext/transport-sequence-number";
+ "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions";
bool RtpExtension::IsSupportedForAudio(const std::string& name) {
return name == webrtc::RtpExtension::kAbsSendTime ||
« talk/session/media/channel.cc ('K') | « webrtc/call/call.cc ('k') | webrtc/modules/remote_bitrate_estimator/transport_feedback_adapter.h » ('j') | no next file with comments »

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