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

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+/*
+ *  Copyright (c) 2012 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_STREAM_SYNCHRONIZATION_H_
+#define VIDEO_STREAM_SYNCHRONIZATION_H_
+
+#include <stdint.h>
+
+#include "system_wrappers/include/rtp_to_ntp_estimator.h"
+
+namespace webrtc {
+
+class StreamSynchronization {
+ public:
+  struct Measurements {
+    Measurements() : latest_receive_time_ms(0), latest_timestamp(0) {}
+    RtpToNtpEstimator rtp_to_ntp;
+    int64_t latest_receive_time_ms;
+    uint32_t latest_timestamp;
+  };
+
+  StreamSynchronization(uint32_t video_stream_id, uint32_t audio_stream_id);
+
+  bool ComputeDelays(int relative_delay_ms,
+                     int current_audio_delay_ms,
+                     int* total_audio_delay_target_ms,
+                     int* total_video_delay_target_ms);
+
+  // On success `relative_delay_ms` contains the number of milliseconds later
+  // video is rendered relative audio. If audio is played back later than video
+  // `relative_delay_ms` will be negative.
+  static bool ComputeRelativeDelay(const Measurements& audio_measurement,
+                                   const Measurements& video_measurement,
+                                   int* relative_delay_ms);
+
+  // Set target buffering delay. Audio and video will be delayed by at least
+  // `target_delay_ms`.
+  void SetTargetBufferingDelay(int target_delay_ms);
+
+  // Lowers the audio delay by 10%. Can be used to recover from errors.
+  void ReduceAudioDelay();
+
+  // Lowers the video delay by 10%. Can be used to recover from errors.
+  void ReduceVideoDelay();
+
+  uint32_t audio_stream_id() const { return audio_stream_id_; }
+  uint32_t video_stream_id() const { return video_stream_id_; }
+
+ private:
+  struct SynchronizationDelays {
+    int extra_ms = 0;
+    int last_ms = 0;
+  };
+
+  const uint32_t video_stream_id_;
+  const uint32_t audio_stream_id_;
+  SynchronizationDelays audio_delay_;
+  SynchronizationDelays video_delay_;
+  int base_target_delay_ms_;
+  int avg_diff_ms_;
+};
+}  // namespace webrtc
+
+#endif  // VIDEO_STREAM_SYNCHRONIZATION_H_