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

webrtc/video/vie_sync_module.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.
- */
-
-// ViESyncModule is responsible for synchronization audio and video for a given
-// VoE and ViE channel couple.
-
-#ifndef WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
-#define WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
-
-#include <memory>
-
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/modules/include/module.h"
-#include "webrtc/video/stream_synchronization.h"
-#include "webrtc/voice_engine/include/voe_video_sync.h"
-
-namespace webrtc {
-
-class Clock;
-class RtpRtcp;
-class VideoFrame;
-class ViEChannel;
-class VoEVideoSync;
-
-namespace vcm {
-class VideoReceiver;
-}  // namespace vcm
-
-class ViESyncModule : public Module {
- public:
-  explicit ViESyncModule(vcm::VideoReceiver* vcm);
-  ~ViESyncModule();
-
-  void ConfigureSync(int voe_channel_id,
-                     VoEVideoSync* voe_sync_interface,
-                     RtpRtcp* video_rtcp_module,
-                     RtpReceiver* rtp_receiver);
-
-  // 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:
-  rtc::CriticalSection data_cs_;
-  vcm::VideoReceiver* const video_receiver_;
-  Clock* const clock_;
-  RtpReceiver* rtp_receiver_;
-  RtpRtcp* video_rtp_rtcp_;
-  int voe_channel_id_;
-  VoEVideoSync* voe_sync_interface_;
-  int64_t last_sync_time_;
-  std::unique_ptr<StreamSynchronization> sync_;
-  StreamSynchronization::Measurements audio_measurement_;
-  StreamSynchronization::Measurements video_measurement_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_VIE_SYNC_MODULE_H_