Stop using VoEVideoSync in Call/VideoReceiveStream.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2452163004
Cr-Commit-Position: refs/heads/master@{#16375}
diff --git a/webrtc/video/rtp_streams_synchronizer.h b/webrtc/video/rtp_streams_synchronizer.h
index bc24d6f..afa1501 100644
--- a/webrtc/video/rtp_streams_synchronizer.h
+++ b/webrtc/video/rtp_streams_synchronizer.h
@@ -19,14 +19,11 @@
 #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 VideoFrame;
-class VoEVideoSync;
+class Syncable;
 
 namespace vcm {
 class VideoReceiver;
@@ -34,11 +31,9 @@
 
 class RtpStreamsSynchronizer : public Module {
  public:
-  RtpStreamsSynchronizer(vcm::VideoReceiver* vcm,
-                         RtpStreamReceiver* rtp_stream_receiver);
+  explicit RtpStreamsSynchronizer(Syncable* syncable_video);
 
-  void ConfigureSync(int voe_channel_id,
-                     VoEVideoSync* voe_sync_interface);
+  void ConfigureSync(Syncable* syncable_audio);
 
   // Implements Module.
   int64_t TimeUntilNextProcess() override;
@@ -48,21 +43,16 @@
   // 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,
+  bool GetStreamSyncOffsetInMs(uint32_t timestamp,
+                               int64_t render_time_ms,
                                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_;
+  Syncable* syncable_video_;
 
   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_audio_ 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_);