Remove Absolute Send Time from list of supported header extensions for audio streams.

webrtc/config.cc

Index: webrtc/config.cc
diff --git a/webrtc/config.cc b/webrtc/config.cc
index 281b89f46e5b5736d9da510b94596335c89e4ae7..e42725e0c852aabed9d6b4cc98b40c847b8c5e1c 100644
--- a/webrtc/config.cc
+++ b/webrtc/config.cc
@@ -86,8 +86,7 @@ const char* RtpExtension::kPlayoutDelayUri =
 const int RtpExtension::kPlayoutDelayDefaultId = 6;
 
 bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
-  return uri == webrtc::RtpExtension::kAbsSendTimeUri ||
-         uri == webrtc::RtpExtension::kAudioLevelUri ||
+  return uri == webrtc::RtpExtension::kAudioLevelUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
 }