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

webrtc/config.cc

 /*
  *  Copyright (c) 2014 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.
  */
 #include "webrtc/config.h"
@@ skipping 68 matching lines @@
 
 // This extension allows applications to adaptively limit the playout delay
 // on frames as per the current needs. For example, a gaming application
 // has very different needs on end-to-end delay compared to a video-conference
 // application.
 const char* RtpExtension::kPlayoutDelayUri =
     "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
 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;
 }
 
 bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
   return uri == webrtc::RtpExtension::kTimestampOffsetUri ||
          uri == webrtc::RtpExtension::kAbsSendTimeUri ||
          uri == webrtc::RtpExtension::kVideoRotationUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
          uri == webrtc::RtpExtension::kPlayoutDelayUri;
 }
@@ skipping 116 matching lines @@
 void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9(
     VideoCodecVP9* vp9_settings) const {
   *vp9_settings = specifics_;
 }
 
 DecoderSpecificSettings::DecoderSpecificSettings() = default;
 
 DecoderSpecificSettings::~DecoderSpecificSettings() = default;
 
 }  // namespace webrtc