Delete a chain of methods in ViE, VoE and ACM

webrtc/modules/audio_coding/main/include/audio_coding_module.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 687 matching lines @@
   //   -network_statistics : a structure that contains network statistics.
   //
   // Return value:
   //   -1 if failed to set the network statistics,
   //    0 if statistics are set successfully.
   //
   virtual int32_t GetNetworkStatistics(
       NetworkStatistics* network_statistics) = 0;
 
   //
-  // Set an initial delay for playout.
-  // An initial delay yields ACM playout silence until equivalent of |delay_ms|
-  // audio payload is accumulated in NetEq jitter. Thereafter, ACM pulls audio
-  // from NetEq in its regular fashion, and the given delay is maintained
-  // through out the call, unless channel conditions yield to a higher jitter
-  // buffer delay.
-  //
-  // Input:
-  //   -delay_ms           : delay in milliseconds.
-  //
-  // Return values:
-  //   -1 if failed to set the delay.
-  //    0 if delay is set successfully.
-  //
-  virtual int SetInitialPlayoutDelay(int delay_ms) = 0;
-
-  //
   // Enable NACK and set the maximum size of the NACK list. If NACK is already
   // enable then the maximum NACK list size is modified accordingly.
   //
   // If the sequence number of last received packet is N, the sequence numbers
   // of NACK list are in the range of [N - |max_nack_list_size|, N).
   //
   // |max_nack_list_size| should be positive (none zero) and less than or
   // equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1
   // is returned. 0 is returned at success.
   //
@@ skipping 14 matching lines @@
   virtual std::vector<uint16_t> GetNackList(
       int64_t round_trip_time_ms) const = 0;
 
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INCLUDE_AUDIO_CODING_MODULE_H_