Clean up abs-send-time for audio.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 960 matching lines @@
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
   return recv_codecs_;
 }
 
 RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
   RtpCapabilities capabilities;
   capabilities.header_extensions.push_back(
       webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri,
                            webrtc::RtpExtension::kAudioLevelDefaultId));
-  capabilities.header_extensions.push_back(
-      webrtc::RtpExtension(webrtc::RtpExtension::kAbsSendTimeUri,
-                           webrtc::RtpExtension::kAbsSendTimeDefaultId));
   if (webrtc::field_trial::FindFullName("WebRTC-Audio-SendSideBwe") ==
       "Enabled") {
     capabilities.header_extensions.push_back(webrtc::RtpExtension(
         webrtc::RtpExtension::kTransportSequenceNumberUri,
         webrtc::RtpExtension::kTransportSequenceNumberDefaultId));
   }
   return capabilities;
 }
 
 int WebRtcVoiceEngine::GetLastEngineError() {
@@ skipping 1567 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE