Repo created

TMessagesProj/jni/voip/webrtc/video/rtp_streams_synchronizer2.h

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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 "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<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_);
+  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_