75 lines
2.7 KiB
C++
75 lines
2.7 KiB
C++
/*
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* Copyright (c) 2020 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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#ifndef VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_
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#define VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_
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#include <memory>
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#include "rtc_base/synchronization/sequence_checker.h"
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#include "rtc_base/task_queue.h"
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#include "rtc_base/task_utils/repeating_task.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 internal {
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// RtpStreamsSynchronizer is responsible for synchronizing audio and video for
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// a given audio receive stream and video receive stream.
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class RtpStreamsSynchronizer {
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public:
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RtpStreamsSynchronizer(TaskQueueBase* main_queue, Syncable* syncable_video);
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~RtpStreamsSynchronizer();
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void ConfigureSync(Syncable* syncable_audio);
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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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void UpdateDelay();
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TaskQueueBase* const task_queue_;
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// Used to check if we're running on the main thread/task queue.
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// The reason we currently don't use RTC_DCHECK_RUN_ON(task_queue_) is because
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// we might be running on an rtc::Thread implementation of TaskQueue, which
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// does not consistently set itself as the active TaskQueue.
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// Instead, we rely on a SequenceChecker for now.
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SequenceChecker main_checker_;
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Syncable* const syncable_video_;
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Syncable* syncable_audio_ RTC_GUARDED_BY(main_checker_) = nullptr;
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std::unique_ptr<StreamSynchronization> sync_ RTC_GUARDED_BY(main_checker_);
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StreamSynchronization::Measurements audio_measurement_
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RTC_GUARDED_BY(main_checker_);
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StreamSynchronization::Measurements video_measurement_
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RTC_GUARDED_BY(main_checker_);
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RepeatingTaskHandle repeating_task_ RTC_GUARDED_BY(main_checker_);
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int64_t last_stats_log_ms_ RTC_GUARDED_BY(&main_checker_);
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};
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} // namespace internal
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} // namespace webrtc
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#endif // VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_
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