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

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 61 matching lines @@
   channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
                                 config_.rtp.nack.rtp_history_ms / 20);
 
   channel_proxy_->RegisterExternalTransport(config.send_transport);
 
   for (const auto& extension : config.rtp.extensions) {
     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::kAbsSendTimeUri) {
-      LOG(LS_WARNING) << RtpExtension::kAbsSendTimeUri
-                      << " is no longer supported for audio.";
     } else {
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
   if (!SetupSendCodec()) {
     LOG(LS_ERROR) << "Failed to set up send codec state.";
   }
 }
 
 AudioSendStream::~AudioSendStream() {
@@ skipping 282 matching lines @@
         LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError();
         return false;
       }
     }
   }
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc