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

/*
 *  Copyright (c) 2019 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_SENDER_H_
#define MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_

#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/ntp_time.h"

namespace webrtc {

//
// Helper class for sending the `AbsoluteCaptureTime` header extension.
//
// Supports the "timestamp interpolation" optimization:
//   A sender SHOULD save bandwidth by not sending abs-capture-time with every
//   RTP packet. It SHOULD still send them at regular intervals (e.g. every
//   second) to help mitigate the impact of clock drift and packet loss. Mixers
//   SHOULD always send abs-capture-time with the first RTP packet after
//   changing capture system.
//
//   Timestamp interpolation works fine as long as there’s reasonably low
//   NTP/RTP clock drift. This is not always true. Senders that detect “jumps”
//   between its NTP and RTP clock mappings SHOULD send abs-capture-time with
//   the first RTP packet after such a thing happening.
//
// See: https://webrtc.org/experiments/rtp-hdrext/abs-capture-time/
//
class AbsoluteCaptureTimeSender { … };

}  // namespace webrtc

#endif  // MODULES_RTP_RTCP_SOURCE_ABSOLUTE_CAPTURE_TIME_SENDER_H_