/* * Copyright (c) 2020 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_ #define VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_ #include #include "api/sequence_checker.h" #include "api/task_queue/task_queue_base.h" #include "rtc_base/system/no_unique_address.h" #include "rtc_base/task_utils/repeating_task.h" #include "video/stream_synchronization.h" namespace webrtc { class Syncable; namespace internal { // RtpStreamsSynchronizer is responsible for synchronizing audio and video for // a given audio receive stream and video receive stream. class RtpStreamsSynchronizer { public: RtpStreamsSynchronizer(TaskQueueBase* main_queue, Syncable* syncable_video); ~RtpStreamsSynchronizer(); void ConfigureSync(Syncable* syncable_audio); // Gets the estimated playout NTP timestamp for the video frame with // `rtp_timestamp` and the sync offset between the current played out audio // frame and the video frame. Returns true on success, false otherwise. // The `estimated_freq_khz` is the frequency used in the RTP to NTP timestamp // conversion. bool GetStreamSyncOffsetInMs(uint32_t rtp_timestamp, int64_t render_time_ms, int64_t* video_playout_ntp_ms, int64_t* stream_offset_ms, double* estimated_freq_khz) const; private: void UpdateDelay(); TaskQueueBase* const task_queue_; // Used to check if we're running on the main thread/task queue. // The reason we currently don't use RTC_DCHECK_RUN_ON(task_queue_) is because // we might be running on an rtc::Thread implementation of TaskQueue, which // does not consistently set itself as the active TaskQueue. // Instead, we rely on a SequenceChecker for now. RTC_NO_UNIQUE_ADDRESS SequenceChecker main_checker_; Syncable* const syncable_video_; Syncable* syncable_audio_ RTC_GUARDED_BY(main_checker_) = nullptr; std::unique_ptr sync_ RTC_GUARDED_BY(main_checker_); StreamSynchronization::Measurements audio_measurement_ RTC_GUARDED_BY(main_checker_); StreamSynchronization::Measurements video_measurement_ RTC_GUARDED_BY(main_checker_); RepeatingTaskHandle repeating_task_ RTC_GUARDED_BY(main_checker_); int64_t last_stats_log_ms_ RTC_GUARDED_BY(&main_checker_); }; } // namespace internal } // namespace webrtc #endif // VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_