Add sender controlled playout delay limits

webrtc/config.cc

Index: webrtc/config.cc
diff --git a/webrtc/config.cc b/webrtc/config.cc
index e9c56da32a24c97962a4cbce455884a21457d0aa..99146eba236f8dad677f83f982cb5db43ef6620e 100644
--- a/webrtc/config.cc
+++ b/webrtc/config.cc
@@ -49,6 +49,14 @@ const char* RtpExtension::kTransportSequenceNumberUri =
     "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
 const int RtpExtension::kTransportSequenceNumberDefaultId = 5;
 
+// 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
+// application.
+const char* RtpExtension::kPlayoutDelayUri =
+    "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
+const int RtpExtension::kPlayoutDelayDefaultId = 6;
+
 bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
   return uri == webrtc::RtpExtension::kAbsSendTimeUri ||
          uri == webrtc::RtpExtension::kAudioLevelUri ||
@@ -59,7 +67,8 @@ bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
   return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
          uri == webrtc::RtpExtension::kAbsSendTimeUri ||
          uri == webrtc::RtpExtension::kVideoRotationUri ||
-         uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
+         uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
+         uri == webrtc::RtpExtension::kPlayoutDelayUri;
 }
 
 VideoStream::VideoStream()