Fix crash when registering abs-send-time to AudioSend/ReceiveStream.

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 c325b9c9bc3fdb1f7b26bda8a42528e990e993a6..cc30939b92134af837ce91d76ec350d5ce570258 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -123,6 +123,9 @@ AudioReceiveStream::AudioReceiveStream(
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionTransportSequenceNumber, extension.id);
       RTC_DCHECK(registered);
+    } else if (extension.uri == RtpExtension::kAbsSendTimeUri) {
+      LOG(LS_WARNING) << RtpExtension::kAbsSendTimeUri
+                      << " is no longer supported for audio.";
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }