Index: webrtc/config.cc |
diff --git a/webrtc/config.cc b/webrtc/config.cc |
index ab2f394fbf771d03c4207914de63781ac4a86e08..e0c490d1ecd8038d546b7298fc9a885ab02718df 100644 |
--- a/webrtc/config.cc |
+++ b/webrtc/config.cc |
@@ -64,10 +64,6 @@ |
"http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01"; |
const int RtpExtension::kTransportSequenceNumberDefaultId = 5; |
-const char* RtpExtension::kVideoContentTypeUri = |
- "http://www.webrtc.org/experiments/rtp-hdrext/video-content-type"; |
-const int RtpExtension::kVideoContentTypeDefaultId = 6; |
- |
// This extension allows applications to adaptively limit the playout delay |
// on frames as per the current needs. For example, a gaming application |
// has very different needs on end-to-end delay compared to a video-conference |
@@ -89,8 +85,7 @@ |
uri == webrtc::RtpExtension::kAbsSendTimeUri || |
uri == webrtc::RtpExtension::kVideoRotationUri || |
uri == webrtc::RtpExtension::kTransportSequenceNumberUri || |
- uri == webrtc::RtpExtension::kPlayoutDelayUri || |
- uri == webrtc::RtpExtension::kVideoContentTypeUri; |
+ uri == webrtc::RtpExtension::kPlayoutDelayUri; |
} |
VideoStream::VideoStream() |