1972ff8a6e
This will make a subsequent change I intend to do safer, where I'll change the return type of one of the base Module functions, by breaking the compile if I miss any overrides. This also highlighted a number of unused functions (in many cases apparently virtual "overrides" of no-longer-existent base functions). I've removed some of these. This also highlighted several cases where "virtual" was used unnecessarily to mark a function that was only defined in one class. Removed "virtual" in those cases. BUG=none TEST=none R=andrew@webrtc.org, henrik.lundin@webrtc.org, mallinath@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24419004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7146 4adac7df-926f-26a2-2b94-8c16560cd09d
68 lines
2.1 KiB
C++
68 lines
2.1 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// ViESyncModule is responsible for synchronization audio and video for a given
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// VoE and ViE channel couple.
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#ifndef WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
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#define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
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#include "webrtc/modules/interface/module.h"
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#include "webrtc/system_wrappers/interface/scoped_ptr.h"
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#include "webrtc/system_wrappers/interface/tick_util.h"
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#include "webrtc/video_engine/stream_synchronization.h"
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#include "webrtc/voice_engine/include/voe_video_sync.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class RtpRtcp;
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class VideoCodingModule;
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class ViEChannel;
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class VoEVideoSync;
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class ViESyncModule : public Module {
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public:
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ViESyncModule(VideoCodingModule* vcm,
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ViEChannel* vie_channel);
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~ViESyncModule();
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int ConfigureSync(int voe_channel_id,
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VoEVideoSync* voe_sync_interface,
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RtpRtcp* video_rtcp_module,
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RtpReceiver* video_receiver);
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int VoiceChannel();
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// Set target delay for buffering mode (0 = real-time mode).
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int SetTargetBufferingDelay(int target_delay_ms);
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// Implements Module.
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virtual int32_t TimeUntilNextProcess() OVERRIDE;
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virtual int32_t Process() OVERRIDE;
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private:
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scoped_ptr<CriticalSectionWrapper> data_cs_;
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VideoCodingModule* vcm_;
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ViEChannel* vie_channel_;
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RtpReceiver* video_receiver_;
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RtpRtcp* video_rtp_rtcp_;
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int voe_channel_id_;
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VoEVideoSync* voe_sync_interface_;
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TickTime last_sync_time_;
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scoped_ptr<StreamSynchronization> sync_;
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StreamSynchronization::Measurements audio_measurement_;
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StreamSynchronization::Measurements video_measurement_;
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};
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} // namespace webrtc
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#endif // WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
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