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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_
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