chromium/third_party/webrtc/modules/rtp_rtcp/source/absolute_capture_time_interpolator.h

/*
 *  Copyright (c) 2021 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 MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_
#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_

#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/clock.h"

namespace webrtc {

//
// Helper class for interpolating the `AbsoluteCaptureTime` header extension.
//
// Supports the "timestamp interpolation" optimization:
//   A receiver SHOULD memorize the capture system (i.e. CSRC/SSRC), capture
//   timestamp, and RTP timestamp of the most recently received abs-capture-time
//   packet on each received stream. It can then use that information, in
//   combination with RTP timestamps of packets without abs-capture-time, to
//   extrapolate missing capture timestamps.
//
// See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
//
class AbsoluteCaptureTimeInterpolator {};

}  // namespace webrtc

#endif  // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_INTERPOLATOR_H_