Add sender controlled playout delay limits

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 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 94 matching lines @@
     } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
       channel_proxy_->SetReceiveAbsoluteSenderTimeStatus(true, extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAbsoluteSendTime, extension.id);
       RTC_DCHECK(registered);
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, extension.id);
       RTC_DCHECK(registered);
+    } else if (extension.uri == RtpExtension::kPlayoutDelayUri) {
+      channel_proxy_->EnableReceivePlayoutDelayLimit(extension.id);
+      RTC_DCHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
+          kRtpExtensionPlayoutDelay, extension.id));
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
   // Configure bandwidth estimation.
   channel_proxy_->RegisterReceiverCongestionControlObjects(
       congestion_controller->packet_router());
   if (UseSendSideBwe(config)) {
     remote_bitrate_estimator_ =
         congestion_controller->GetRemoteBitrateEstimator(true);
@@ skipping 118 matching lines @@
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc