Stop using VoEVideoSync in Call/VideoReceiveStream.

webrtc/call/syncable.h

Index: webrtc/call/syncable.h
diff --git a/webrtc/call/syncable.h b/webrtc/call/syncable.h
new file mode 100644
index 0000000000000000000000000000000000000000..67df4dde646f38ac5135f3969acc89c4d39d3f11
--- /dev/null
+++ b/webrtc/call/syncable.h
@@ -0,0 +1,37 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+// Syncable is used by RtpStreamsSynchronizer in VideoReceiveStream, and
+// implemented by AudioReceiveStream.
+
+#ifndef WEBRTC_CALL_SYNCABLE_H_
+#define WEBRTC_CALL_SYNCABLE_H_
+
+#include <stdint.h>
+
+namespace webrtc {
+
+class RtpReceiver;
+class RtpRtcp;
+
+class Syncable {
+ public:
+  virtual ~Syncable();
+  virtual void GetRtpRtcp(RtpRtcp** rtp_rtcp,
+                          RtpReceiver** rtp_receiver) const = 0;
+  virtual void GetDelayEstimate(int* jitter_buffer_delay_ms,
+                                int* playout_buffer_delay_ms) const = 0;
+  virtual uint32_t GetPlayoutTimestamp() const = 0;
+  virtual void SetMinimumPlayoutDelay(int delay_ms) = 0;
+  virtual int id() const = 0;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_SYNCABLE_H_