7623ce4aeb
Reason for revert: Bot breakage caused by TickTime::UseFakeClock has been removed. Original issue's description: > 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 > > Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518 > Cr-Commit-Position: refs/heads/master@{#10937} BUG=webrtc:1695 TBR=mflodman@webrtc.org,kjellander@webrtc.org NOPRESUBMIT=true Review URL: https://codereview.webrtc.org/1510183002 . Cr-Commit-Position: refs/heads/master@{#10948}
63 lines
1.8 KiB
C++
63 lines
1.8 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_VIE_SYNC_MODULE_H_
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#define WEBRTC_VIDEO_VIE_SYNC_MODULE_H_
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/include/module.h"
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#include "webrtc/system_wrappers/include/tick_util.h"
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#include "webrtc/video/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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explicit ViESyncModule(VideoCodingModule* vcm);
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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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// Implements Module.
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int64_t TimeUntilNextProcess() override;
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int32_t Process() override;
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private:
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rtc::scoped_ptr<CriticalSectionWrapper> data_cs_;
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VideoCodingModule* const vcm_;
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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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rtc::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_VIE_SYNC_MODULE_H_
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