Stop using VoEVideoSync in Call/VideoReceiveStream.

webrtc/video/rtp_streams_synchronizer.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.
  */
 
 // RtpStreamsSynchronizer is responsible for synchronization audio and video for
 // a given voice engine channel and video receive stream.
 
 #ifndef WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
 #define WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
 
 #include <memory>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/video/rtp_stream_receiver.h"
 #include "webrtc/video/stream_synchronization.h"
 
 namespace webrtc {
 
 class Clock;
+class Syncable;
 class VideoFrame;
-class VoEVideoSync;
 
 namespace vcm {
 class VideoReceiver;
 }  // namespace vcm
 
 class RtpStreamsSynchronizer : public Module {
  public:
   RtpStreamsSynchronizer(vcm::VideoReceiver* vcm,
                          RtpStreamReceiver* rtp_stream_receiver);
 
-  void ConfigureSync(int voe_channel_id,
-                     VoEVideoSync* voe_sync_interface);
+  void ConfigureSync(Syncable* syncable);
 
   // Implements Module.
   int64_t TimeUntilNextProcess() override;
   void Process() override;
 
   // Gets the sync offset between the current played out audio frame and the
   // video |frame|. Returns true on success, false otherwise.
   // The estimated frequency is the frequency used in the RTP to NTP timestamp
   // conversion.
   bool GetStreamSyncOffsetInMs(const VideoFrame& frame,
                                int64_t* stream_offset_ms,
                                double* estimated_freq_khz) const;
 
  private:
   Clock* const clock_;
   vcm::VideoReceiver* const video_receiver_;
   RtpReceiver* const video_rtp_receiver_;
   RtpRtcp* const video_rtp_rtcp_;
 
   rtc::CriticalSection crit_;
-  int voe_channel_id_ GUARDED_BY(crit_);
-  VoEVideoSync* voe_sync_interface_ GUARDED_BY(crit_);
-  RtpReceiver* audio_rtp_receiver_ GUARDED_BY(crit_);
-  RtpRtcp* audio_rtp_rtcp_ GUARDED_BY(crit_);
+  Syncable* syncable_ GUARDED_BY(crit_);
   std::unique_ptr<StreamSynchronization> sync_ GUARDED_BY(crit_);
   StreamSynchronization::Measurements audio_measurement_ GUARDED_BY(crit_);
   StreamSynchronization::Measurements video_measurement_ GUARDED_BY(crit_);
 
   rtc::ThreadChecker process_thread_checker_;
   int64_t last_sync_time_ ACCESS_ON(&process_thread_checker_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_STREAMS_SYNCHRONIZER_H_