Delete a chain of methods in ViE, VoE and ACM

webrtc/voice_engine/include/voe_video_sync.h

 /*
  *  Copyright (c) 2011 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 46 matching lines @@
   // Gets the current sound card buffer size (playout delay).
   virtual int GetPlayoutBufferSize(int& buffer_ms) = 0;
 
   // Sets a minimum target delay for the jitter buffer. This delay is
   // maintained by the jitter buffer, unless channel condition (jitter in
   // inter-arrival times) dictates a higher required delay. The overall
   // jitter buffer delay is max of |delay_ms| and the latency that NetEq
   // computes based on inter-arrival times and its playout mode.
   virtual int SetMinimumPlayoutDelay(int channel, int delay_ms) = 0;
 
-  // Sets an initial delay for the playout jitter buffer. The playout of the
-  // audio is delayed by |delay_ms| in milliseconds. Thereafter, the delay is
-  // maintained, unless NetEq's internal mechanism requires a higher latency.
-  // Such a latency is computed based on inter-arrival times and NetEq's
-  // playout mode.
-  virtual int SetInitialPlayoutDelay(int channel, int delay_ms) = 0;
-
   // Gets the |jitter_buffer_delay_ms| (including the algorithmic delay), and
   // the |playout_buffer_delay_ms| for a specified |channel|.
   virtual int GetDelayEstimate(int channel,
                                int* jitter_buffer_delay_ms,
                                int* playout_buffer_delay_ms) = 0;
 
   // Returns the least required jitter buffer delay. This is computed by the
   // the jitter buffer based on the inter-arrival time of RTP packets and
   // playout mode. NetEq maintains this latency unless a higher value is
   // requested by calling SetMinimumPlayoutDelay().
@@ skipping 13 matching lines @@
                          RtpReceiver** rtp_receiver) = 0;
 
  protected:
   VoEVideoSync() {}
   virtual ~VoEVideoSync() {}
 };
 
 }  // namespace webrtc
 
 #endif  // #ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H