Files
platform-external-webrtc/webrtc/video_engine/vie_sync_module.h
pkasting@chromium.org 0b1534c52e Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.

This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".

BUG=chromium:81439
TEST=none
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-15 22:09:40 +00:00

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2.1 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/modules/interface/module.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.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:
ViESyncModule(VideoCodingModule* vcm,
ViEChannel* vie_channel);
~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.
virtual int64_t TimeUntilNextProcess() OVERRIDE;
virtual int32_t Process() OVERRIDE;
private:
scoped_ptr<CriticalSectionWrapper> data_cs_;
VideoCodingModule* vcm_;
ViEChannel* vie_channel_;
RtpReceiver* video_receiver_;
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_