Files
platform-external-webrtc/webrtc/video_engine/vie_sync_module.h
kjellander 8237abf563 Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
Reason for revert:
Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.

Original issue's description:
> Merge webrtc/video_engine/ into webrtc/video/
>
> BUG=webrtc:1695
> R=mflodman@webrtc.org
>
> Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> Cr-Commit-Position: refs/heads/master@{#10926}

TBR=mflodman@webrtc.org,pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:1695

Review URL: https://codereview.webrtc.org/1507903005

Cr-Commit-Position: refs/heads/master@{#10937}
2015-12-08 15:12:11 +00:00

63 lines
1.9 KiB
C++

/*
* 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_ENGINE_VIE_SYNC_MODULE_H_
#define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/system_wrappers/include/tick_util.h"
#include "webrtc/video_engine/stream_synchronization.h"
#include "webrtc/voice_engine/include/voe_video_sync.h"
namespace webrtc {
class CriticalSectionWrapper;
class RtpRtcp;
class VideoCodingModule;
class ViEChannel;
class VoEVideoSync;
class ViESyncModule : public Module {
public:
explicit ViESyncModule(VideoCodingModule* vcm);
~ViESyncModule();
int ConfigureSync(int voe_channel_id,
VoEVideoSync* voe_sync_interface,
RtpRtcp* video_rtcp_module,
RtpReceiver* video_receiver);
int VoiceChannel();
// Implements Module.
int64_t TimeUntilNextProcess() override;
int32_t Process() override;
private:
rtc::scoped_ptr<CriticalSectionWrapper> data_cs_;
VideoCodingModule* const vcm_;
RtpReceiver* video_receiver_;
RtpRtcp* video_rtp_rtcp_;
int voe_channel_id_;
VoEVideoSync* voe_sync_interface_;
TickTime last_sync_time_;
rtc::scoped_ptr<StreamSynchronization> sync_;
StreamSynchronization::Measurements audio_measurement_;
StreamSynchronization::Measurements video_measurement_;
};
} // namespace webrtc
#endif // WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_