Remove VoEVideoSync interface.

webrtc/test/mock_voice_engine.h

 /*
  *  Copyright (c) 2015 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 282 matching lines @@
                    unsigned int& maxJitterMs,
                    unsigned int& discardedPackets));
   MOCK_METHOD2(GetRTCPStatistics, int(int channel, CallStatistics& stats));
   MOCK_METHOD2(GetRemoteRTCPReportBlocks,
                int(int channel, std::vector<ReportBlock>* receive_blocks));
   MOCK_METHOD3(SetREDStatus, int(int channel, bool enable, int redPayloadtype));
   MOCK_METHOD3(GetREDStatus,
                int(int channel, bool& enable, int& redPayloadtype));
   MOCK_METHOD3(SetNACKStatus, int(int channel, bool enable, int maxNoPackets));
 
-  // VoEVideoSync
-  MOCK_METHOD1(GetPlayoutBufferSize, int(int& buffer_ms));
-  MOCK_METHOD2(SetMinimumPlayoutDelay, int(int channel, int delay_ms));
-  MOCK_METHOD3(GetDelayEstimate,
-               int(int channel,
-                   int* jitter_buffer_delay_ms,
-                   int* playout_buffer_delay_ms));
-  MOCK_CONST_METHOD1(GetLeastRequiredDelayMs, int(int channel));
-  MOCK_METHOD2(SetInitTimestamp, int(int channel, unsigned int timestamp));
-  MOCK_METHOD2(SetInitSequenceNumber, int(int channel, short sequenceNumber));
-  MOCK_METHOD2(GetPlayoutTimestamp, int(int channel, unsigned int& timestamp));
-  MOCK_METHOD3(GetRtpRtcp,
-               int(int channel,
-                   RtpRtcp** rtpRtcpModule,
-                   RtpReceiver** rtp_receiver));
-
   // VoEVolumeControl
   MOCK_METHOD1(SetSpeakerVolume, int(unsigned int volume));
   MOCK_METHOD1(GetSpeakerVolume, int(unsigned int& volume));
   MOCK_METHOD1(SetMicVolume, int(unsigned int volume));
   MOCK_METHOD1(GetMicVolume, int(unsigned int& volume));
   MOCK_METHOD2(SetInputMute, int(int channel, bool enable));
   MOCK_METHOD2(GetInputMute, int(int channel, bool& enabled));
   MOCK_METHOD1(GetSpeechInputLevel, int(unsigned int& level));
   MOCK_METHOD2(GetSpeechOutputLevel, int(int channel, unsigned int& level));
   MOCK_METHOD1(GetSpeechInputLevelFullRange, int(unsigned int& level));
@@ skipping 14 matching lines @@
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 
   MockAudioDeviceModule mock_audio_device_;
   MockAudioProcessing mock_audio_processing_;
   MockAudioTransport mock_audio_transport_;
 };
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_MOCK_VOICE_ENGINE_H_