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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 "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/interface/module.h"
#include "webrtc/system_wrappers/interface/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();
// Set target delay for buffering mode (0 = real-time mode).
int SetTargetBufferingDelay(int target_delay_ms);
// 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_