Remove VoEVideoSync interface.

webrtc/modules/audio_coding/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 619 matching lines @@
   //
   // Input:
   //   -time_ms            : maximum delay in milliseconds.
   //
   // Return value:
   //   -1 if failed to set the delay,
   //    0 if the maximum delay is set.
   //
   virtual int SetMaximumPlayoutDelay(int time_ms) = 0;
 
-  //
+  // TODO(kwiberg): Consider if this is needed anymore, now that voe::Channel
+  //                doesn't use it.
   // The shortest latency, in milliseconds, required by jitter buffer. This
   // is computed based on inter-arrival times and playout mode of NetEq. The
   // actual delay is the maximum of least-required-delay and the minimum-delay
   // specified by SetMinumumPlayoutDelay() API.
   //
   virtual int LeastRequiredDelayMs() const = 0;
 
   // int32_t PlayoutTimestamp()
   // The send timestamp of an RTP packet is associated with the decoded
   // audio of the packet in question. This function returns the timestamp of
@@ skipping 164 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_INCLUDE_AUDIO_CODING_MODULE_H_