Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 2a9220d9f09255c85041fdfc4b4966850e117da9..75b89939465b68a5e9e44e83c67a14c77153b8d9 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -39,6 +39,8 @@ RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
     return kRtpExtensionVideoRotation;
   if (extension == RtpExtension::kTransportSequenceNumberUri)
     return kRtpExtensionTransportSequenceNumber;
+  if (extension == RtpExtension::kPlayoutDelayUri)
+    return kRtpExtensionPlayoutDelay;
   RTC_NOTREACHED() << "Looking up unsupported RTP extension.";
   return kRtpExtensionNone;
 }
@@ -922,6 +924,11 @@ void ModuleRtpRtcpImpl::OnReceivedNACK(
   rtp_sender_.OnReceivedNACK(nack_sequence_numbers, rtt);
 }
 
+void ModuleRtpRtcpImpl::OnReceivedRtcpReceiverReport(
+    const ReportBlockList& report_blocks) {
+  rtp_sender_.OnReceivedRtcpReceiverReport(report_blocks);
+}
+
 bool ModuleRtpRtcpImpl::LastReceivedNTP(
     uint32_t* rtcp_arrival_time_secs,  // When we got the last report.
     uint32_t* rtcp_arrival_time_frac,