Clean up abs-send-time for audio.

webrtc/voice_engine/include/voe_rtp_rtcp.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 115 matching lines @@
 
   // Sets the status of receiving rtp-audio-level-indication on a specific
   // |channel|.
   virtual int SetReceiveAudioLevelIndicationStatus(int channel,
                                                    bool enable,
                                                    unsigned char id = 1) {
     // TODO(wu): Remove default implementation once talk is updated.
     return 0;
   }
 
-  // Sets the status of sending absolute sender time on a specific |channel|.
-  virtual int SetSendAbsoluteSenderTimeStatus(int channel,
-                                              bool enable,
-                                              unsigned char id) = 0;
-
-  // Sets status of receiving absolute sender time on a specific |channel|.
-  virtual int SetReceiveAbsoluteSenderTimeStatus(int channel,
-                                                 bool enable,
-                                                 unsigned char id) = 0;
-
   // Sets the RTCP status on a specific |channel|.
   virtual int SetRTCPStatus(int channel, bool enable) = 0;
 
   // Gets the RTCP status on a specific |channel|.
   virtual int GetRTCPStatus(int channel, bool& enabled) = 0;
 
   // Sets the canonical name (CNAME) parameter for RTCP reports on a
   // specific |channel|.
   virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0;
 
@@ skipping 38 matching lines @@
   virtual int SetNACKStatus(int channel, bool enable, int maxNoPackets) = 0;
 
  protected:
   VoERTP_RTCP() {}
   virtual ~VoERTP_RTCP() {}
 };
 
 }  // namespace webrtc
 
 #endif  // #ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H