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fcf79cca7b
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp Partial implementation: currently only populated when a/v sync is enabled. Bug: webrtc:7065 Change-Id: I8595cc848d080d7c3bef152462a9becf0e5a2196 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155621 Commit-Queue: Åsa Persson <asapersson@webrtc.org> Reviewed-by: Oskar Sundbom <ossu@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29581}
65 lines
2.3 KiB
C++
65 lines
2.3 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/critical_section.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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class RtpStreamsSynchronizer : public Module {
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public:
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explicit RtpStreamsSynchronizer(Syncable* syncable_video);
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~RtpStreamsSynchronizer() override;
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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 estimated playout NTP timestamp for the video frame with
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// |rtp_timestamp| and the sync offset between the current played out audio
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// frame and the video frame. Returns true on success, false otherwise.
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// The |estimated_freq_khz| is the frequency used in the RTP to NTP timestamp
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// conversion.
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bool GetStreamSyncOffsetInMs(uint32_t rtp_timestamp,
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int64_t render_time_ms,
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int64_t* video_playout_ntp_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_GUARDED_BY(&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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