Add sender controlled playout delay limits

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 21 matching lines @@
   if (extension == RtpExtension::kTimestampOffsetUri)
     return kRtpExtensionTransmissionTimeOffset;
   if (extension == RtpExtension::kAudioLevelUri)
     return kRtpExtensionAudioLevel;
   if (extension == RtpExtension::kAbsSendTimeUri)
     return kRtpExtensionAbsoluteSendTime;
   if (extension == RtpExtension::kVideoRotationUri)
     return kRtpExtensionVideoRotation;
   if (extension == RtpExtension::kTransportSequenceNumberUri)
     return kRtpExtensionTransportSequenceNumber;
+  if (extension == RtpExtension::kPlayoutDelayUri)
+    return kRtpExtensionPlayoutDelay;
   RTC_NOTREACHED() << "Looking up unsupported RTP extension.";
   return kRtpExtensionNone;
 }
 
 RtpRtcp::Configuration::Configuration()
     : audio(false),
       receiver_only(false),
       clock(nullptr),
       receive_statistics(NullObjectReceiveStatistics()),
       outgoing_transport(nullptr),
@@ skipping 863 matching lines @@
     return;
   }
   // Use RTT from RtcpRttStats class if provided.
   int64_t rtt = rtt_ms();
   if (rtt == 0) {
     rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
   }
   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,
     uint32_t* remote_sr) const {
   // Remote SR: NTP inside the last received (mid 16 bits from sec and frac).
   uint32_t ntp_secs = 0;
   uint32_t ntp_frac = 0;
 
   if (!rtcp_receiver_.NTP(&ntp_secs,
                           &ntp_frac,
@@ skipping 52 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc