Clean up abs-send-time for audio.

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 657a7608ec935db8514eaa33da890fa97f54d58e..c325b9c9bc3fdb1f7b26bda8a42528e990e993a6 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -118,11 +118,6 @@ AudioReceiveStream::AudioReceiveStream(
       bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
           kRtpExtensionAudioLevel, extension.id);
       RTC_DCHECK(registered);
-    } 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(