webrtc_m130/webrtc/video_engine/vie_sync_module.h
mikhal@webrtc.org ef9f76a59d Adding a receive side API for buffering mode.
At the same time, renaming the send side API.

Review URL: https://webrtc-codereview.appspot.com/1104004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3525 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-15 23:22:18 +00:00

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1.9 KiB
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/*
* 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 "modules/interface/module.h"
#include "system_wrappers/interface/scoped_ptr.h"
#include "system_wrappers/interface/tick_util.h"
#include "video_engine/stream_synchronization.h"
#include "voice_engine/include/voe_video_sync.h"
namespace webrtc {
class CriticalSectionWrapper;
class RtpRtcp;
class VideoCodingModule;
class ViEChannel;
class VoEVideoSync;
class ViESyncModule : public Module {
public:
ViESyncModule(VideoCodingModule* vcm,
ViEChannel* vie_channel);
~ViESyncModule();
int ConfigureSync(int voe_channel_id,
VoEVideoSync* voe_sync_interface,
RtpRtcp* video_rtcp_module);
int VoiceChannel();
// Set target delay for buffering mode (0 = real-time mode).
void SetTargetBufferingDelay(int target_delay_ms);
// Implements Module.
virtual WebRtc_Word32 TimeUntilNextProcess();
virtual WebRtc_Word32 Process();
private:
scoped_ptr<CriticalSectionWrapper> data_cs_;
VideoCodingModule* vcm_;
ViEChannel* vie_channel_;
RtpRtcp* video_rtp_rtcp_;
int voe_channel_id_;
VoEVideoSync* voe_sync_interface_;
TickTime last_sync_time_;
scoped_ptr<StreamSynchronization> sync_;
StreamSynchronization::Measurements audio_measurement_;
StreamSynchronization::Measurements video_measurement_;
};
} // namespace webrtc
#endif // WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_