Move RTP for synchroninzation and rename classes, files and variables.

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 VideoFrame;
+class VoEVideoSync;
+
+namespace vcm {
+class VideoReceiver;
+}  // namespace vcm
+
+class RtpStreamsSynchronizer : public Module {

[mflodman, 2016/08/04 08:24:05]

This is not really a new file, rather vie_sync_module.h being renamed and with
name and signature changes.

+ public:
+  RtpStreamsSynchronizer(vcm::VideoReceiver* vcm,
+                         RtpStreamReceiver* rtp_stream_receiver);
+
+  void ConfigureSync(int voe_channel_id,
+                     VoEVideoSync* voe_sync_interface);
+
+  // 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.
+  bool GetStreamSyncOffsetInMs(const VideoFrame& frame,
+                               int64_t* stream_offset_ms) 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_);
+  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_