ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
63 lines
1.9 KiB
C++
63 lines
1.9 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/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_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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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_ENGINE_VIE_SYNC_MODULE_H_
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