Remove VoEVideoSync interface.

webrtc/modules/audio_coding/include/audio_coding_module.h

Index: webrtc/modules/audio_coding/include/audio_coding_module.h
diff --git a/webrtc/modules/audio_coding/include/audio_coding_module.h b/webrtc/modules/audio_coding/include/audio_coding_module.h
index cd99579e1d5d7365a14746261f61ee05d4965af7..7843fb876de4b38d716f66eebb214a5d55f20845 100644
--- a/webrtc/modules/audio_coding/include/audio_coding_module.h
+++ b/webrtc/modules/audio_coding/include/audio_coding_module.h
@@ -637,7 +637,8 @@ class AudioCodingModule {
   //
   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