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/*
* 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 <memory>
#include "rtc_base/synchronization/sequence_checker.h"
#include "rtc_base/task_queue.h"
#include "rtc_base/task_utils/pending_task_safety_flag.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 QueueTimer();
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.
SequenceChecker main_checker_;
Syncable* const syncable_video_;
Syncable* syncable_audio_ RTC_GUARDED_BY(main_checker_) = nullptr;
std::unique_ptr<StreamSynchronization> sync_ RTC_GUARDED_BY(main_checker_);
StreamSynchronization::Measurements audio_measurement_
RTC_GUARDED_BY(main_checker_);
StreamSynchronization::Measurements video_measurement_
RTC_GUARDED_BY(main_checker_);
int64_t last_sync_time_ RTC_GUARDED_BY(&main_checker_);
int64_t last_stats_log_ms_ RTC_GUARDED_BY(&main_checker_);
bool timer_running_ RTC_GUARDED_BY(main_checker_) = false;
// Used to signal destruction to potentially pending tasks.
PendingTaskSafetyFlag::Pointer task_safety_flag_ =
PendingTaskSafetyFlag::Create();
};
} // namespace internal
} // namespace webrtc
#endif // VIDEO_RTP_STREAMS_SYNCHRONIZER2_H_
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