Clean up abs-send-time for audio.

webrtc/voice_engine/voe_rtp_rtcp_impl.h

 /*
  *  Copyright (c) 2012 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 33 matching lines @@
   int GetRemoteSSRC(int channel, unsigned int& ssrc) override;
 
   // RTP Header Extension for Client-to-Mixer Audio Level Indication
   int SetSendAudioLevelIndicationStatus(int channel,
                                         bool enable,
                                         unsigned char id) override;
   int SetReceiveAudioLevelIndicationStatus(int channel,
                                            bool enable,
                                            unsigned char id) override;
 
-  // RTP Header Extension for Absolute Sender Time
-  int SetSendAbsoluteSenderTimeStatus(int channel,
-                                      bool enable,
-                                      unsigned char id) override;
-  int SetReceiveAbsoluteSenderTimeStatus(int channel,
-                                         bool enable,
-                                         unsigned char id) override;
-
   // Statistics
   int GetRTPStatistics(int channel,
                        unsigned int& averageJitterMs,
                        unsigned int& maxJitterMs,
                        unsigned int& discardedPackets) override;
 
   int GetRTCPStatistics(int channel, CallStatistics& stats) override;
 
   int GetRemoteRTCPReportBlocks(
       int channel,
       std::vector<ReportBlock>* report_blocks) override;
 
   // NACK
   int SetNACKStatus(int channel, bool enable, int maxNoPackets) override;
 
  protected:
   VoERTP_RTCPImpl(voe::SharedData* shared);
   ~VoERTP_RTCPImpl() override;
 
  private:
   voe::SharedData* _shared;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H