Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/source/rtp_header_extension.h

Index: webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
index beaf989c895830eb3dacc6ecd6bd22ff70d48445..e5e6cf235960d58bc171769a968cbaf8a41d56ed 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
@@ -26,6 +26,14 @@ const size_t kAudioLevelLength = 2;
 const size_t kAbsoluteSendTimeLength = 4;
 const size_t kVideoRotationLength = 2;
 const size_t kTransportSequenceNumberLength = 3;
+const size_t kPlayoutDelayLength = 4;
+
+// Playout delay in milliseconds. A playout delay limit (min or max)
+// has 12 bits allocated. This allows a range of 0-4095 values which translates
+// to a range of 0-40950 in milliseconds.
+const int kPlayoutDelayGranularityMs = 10;
+// Maximum playout delay value in milliseconds.
+const int kPlayoutDelayMaxMs = 40950;
 
 struct HeaderExtension {
   explicit HeaderExtension(RTPExtensionType extension_type)
@@ -58,6 +66,9 @@ struct HeaderExtension {
       case kRtpExtensionTransportSequenceNumber:
         length = kTransportSequenceNumberLength;
         break;
+      case kRtpExtensionPlayoutDelay:
+        length = kPlayoutDelayLength;
+        break;
       default:
         assert(false);
     }