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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
66 lines
2.1 KiB
C++
66 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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// RtpStreamsSynchronizer is responsible for synchronization audio and video for
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// a given voice engine channel and video receive stream.
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#ifndef VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
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#define VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
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#include <memory>
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#include "modules/include/module.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/thread_checker.h"
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#include "video/stream_synchronization.h"
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namespace webrtc {
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class Syncable;
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namespace vcm {
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class VideoReceiver;
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} // namespace vcm
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class RtpStreamsSynchronizer : public Module {
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public:
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explicit RtpStreamsSynchronizer(Syncable* syncable_video);
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void ConfigureSync(Syncable* syncable_audio);
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// Implements Module.
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int64_t TimeUntilNextProcess() override;
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void Process() override;
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// Gets the sync offset between the current played out audio frame and the
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// video |frame|. Returns true on success, false otherwise.
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// The estimated frequency is the frequency used in the RTP to NTP timestamp
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// conversion.
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bool GetStreamSyncOffsetInMs(uint32_t timestamp,
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int64_t render_time_ms,
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int64_t* stream_offset_ms,
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double* estimated_freq_khz) const;
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private:
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Syncable* syncable_video_;
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rtc::CriticalSection crit_;
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Syncable* syncable_audio_ RTC_GUARDED_BY(crit_);
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std::unique_ptr<StreamSynchronization> sync_ RTC_GUARDED_BY(crit_);
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StreamSynchronization::Measurements audio_measurement_ RTC_GUARDED_BY(crit_);
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StreamSynchronization::Measurements video_measurement_ RTC_GUARDED_BY(crit_);
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rtc::ThreadChecker process_thread_checker_;
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int64_t last_sync_time_ RTC_ACCESS_ON(&process_thread_checker_);
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};
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} // namespace webrtc
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#endif // VIDEO_RTP_STREAMS_SYNCHRONIZER_H_
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