Remove VoEVideoSync interface.

webrtc/voice_engine/channel.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 255 matching lines @@
   bool InputMute() const;
   int SetOutputVolumePan(float left, float right);
   int GetOutputVolumePan(float& left, float& right) const;
   int SetChannelOutputVolumeScaling(float scaling);
   int GetChannelOutputVolumeScaling(float& scaling) const;
 
   // VoENetEqStats
   int GetNetworkStatistics(NetworkStatistics& stats);
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
 
-  // VoEVideoSync
-  bool GetDelayEstimate(int* jitter_buffer_delay_ms,
-                        int* playout_buffer_delay_ms) const;
+  // Audio+Video Sync
   uint32_t GetDelayEstimate() const;
-  int LeastRequiredDelayMs() const;
   int SetMinimumPlayoutDelay(int delayMs);
   int GetPlayoutTimestamp(unsigned int& timestamp);
-  int SetInitTimestamp(unsigned int timestamp);
-  int SetInitSequenceNumber(short sequenceNumber);
-
-  // VoEVideoSyncExtended
   int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
 
   // DTMF
   int SendTelephoneEventOutband(int event, int duration_ms);
   int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
 
   // VoEAudioProcessingImpl
   int VoiceActivityIndicator(int& activity);
 
   // VoERTP_RTCP
@@ skipping 257 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_