Add sender controlled playout delay limits

webrtc/voice_engine/channel.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 2518 matching lines @@
   rtp_header_parser_->DeregisterRtpHeaderExtension(
       kRtpExtensionAbsoluteSendTime);
   if (enable &&
       !rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAbsoluteSendTime, id)) {
     return -1;
   }
   return 0;
 }
 
+void Channel::EnableSendPlayoutDelayLimit(int id) {
+  int ret = _rtpRtcpModule->RegisterSendRtpHeaderExtension(
+      kRtpExtensionPlayoutDelay, id);
+  RTC_DCHECK_EQ(0, ret);
+}
+
+void Channel::EnableReceivePlayoutDelayLimit(int id) {
+  bool ret = rtp_header_parser_->RegisterRtpHeaderExtension(
+      kRtpExtensionPlayoutDelay, id);
+  RTC_DCHECK(ret);
+}
+
 void Channel::EnableSendTransportSequenceNumber(int id) {
   int ret =
       SetSendRtpHeaderExtension(true, kRtpExtensionTransportSequenceNumber, id);
   RTC_DCHECK_EQ(0, ret);
 }
 
 void Channel::EnableReceiveTransportSequenceNumber(int id) {
   rtp_header_parser_->DeregisterRtpHeaderExtension(
       kRtpExtensionTransportSequenceNumber);
   bool ret = rtp_header_parser_->RegisterRtpHeaderExtension(
@@ skipping 1010 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc