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_