Add sender controlled playout delay limits

webrtc/audio/audio_send_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 67 matching lines @@
 
   channel_proxy_->RegisterExternalTransport(config.send_transport);
 
   for (const auto& extension : config.rtp.extensions) {
     if (extension.uri == RtpExtension::kAbsSendTimeUri) {
       channel_proxy_->SetSendAbsoluteSenderTimeStatus(true, extension.id);
     } else if (extension.uri == RtpExtension::kAudioLevelUri) {
       channel_proxy_->SetSendAudioLevelIndicationStatus(true, extension.id);
     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
       channel_proxy_->EnableSendTransportSequenceNumber(extension.id);
+    } else if (extension.uri == RtpExtension::kPlayoutDelayUri) {
+      channel_proxy_->EnableSendPlayoutDelayLimit(extension.id);
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
   channel_proxy_->DeRegisterExternalTransport();
@@ skipping 127 matching lines @@
 
 VoiceEngine* AudioSendStream::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