Index: webrtc/config.cc |
diff --git a/webrtc/config.cc b/webrtc/config.cc |
index 19a9a96079dc3b90b09cb12ac43fa916a7ccaf4b..fe23a625b7251591d09ab75d4ee47f36815e983b 100644 |
--- a/webrtc/config.cc |
+++ b/webrtc/config.cc |
@@ -87,6 +87,13 @@ const int RtpExtension::kVideoTimingDefaultId = 8; |
const char RtpExtension::kEncryptHeaderExtensionsUri[] = |
"urn:ietf:params:rtp-hdrext:encrypt"; |
+// This extensions provides meta-information about the RTP streams outside the |
+// encrypted media payload, an RTP switch can do codec-agnostic |
+// selective forwarding without decrypting the payload |
+const char* RtpExtension::kFrameMarkingUri = |
+ "urn:ietf:params:rtp-hdrext:framemarking"; |
+const int RtpExtension::kFrameMarkingDefaultId = 9; |
+ |
const int RtpExtension::kMinId = 1; |
const int RtpExtension::kMaxId = 14; |
@@ -102,7 +109,8 @@ bool RtpExtension::IsSupportedForVideo(const std::string& uri) { |
uri == webrtc::RtpExtension::kTransportSequenceNumberUri || |
uri == webrtc::RtpExtension::kPlayoutDelayUri || |
uri == webrtc::RtpExtension::kVideoContentTypeUri || |
- uri == webrtc::RtpExtension::kVideoTimingUri; |
+ uri == webrtc::RtpExtension::kVideoTimingUri || |
+ uri == webrtc::RtpExtension::kFrameMarkingUri; |
} |
bool RtpExtension::IsEncryptionSupported(const std::string& uri) { |