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;

[stefan-webrtc, 2016/05/28 05:01:44]

Initialize this to:
kPlayoutDelayGranularityMs * (1<<(kPlayoutDelayLength - 1))

[sprang, 2016/05/30 12:13:15]

Hm? kPlayoutDelayLength is the byte length of the whole extension (4 bytes),
right?

And even if kPlayoutDelayLength was 12, shouldn't that be
kPlayoutDelayGranularityMs * ((1 << kPlayoutDelayLength) - 1) ?

[Irfan, 2016/06/01 08:38:33]

This is kPlayoutDelayGranularityMs * ( (1 << ((kPlayoutDelayLength - 1)  * 8/
2))  - 1). Am going to leave this as 40950 with the comment since that seems
more clear to me :-)

 
 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);
     }