Add sender controlled playout delay limits

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index e391df471ed8c83ff228bacdc36b0db11da91a5f..1cd85ee0068b9decbbedb207c1fbfcc0387621ae 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -112,6 +112,10 @@ AudioReceiveStream::AudioReceiveStream(
       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.";
     }