chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sequence_number_map.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_RTP_SEQUENCE_NUMBER_MAP_H_
#define MODULES_RTP_RTCP_SOURCE_RTP_SEQUENCE_NUMBER_MAP_H_

#include <cstddef>
#include <cstdint>
#include <deque>

#include "absl/types/optional.h"

namespace webrtc {

// Records the association of RTP sequence numbers to timestamps and to whether
// the packet was first and/or last in the frame.
//
// 1. Limits number of entries. Whenever `max_entries` is about to be exceeded,
//    the size is reduced by approximately 25%.
// 2. RTP sequence numbers wrap around relatively infrequently.
//    This class therefore only remembers at most the last 2^15 RTP packets,
//    so that the newest packet's sequence number is still AheadOf the oldest
//    packet's sequence number.
// 3. Media frames are sometimes split into several RTP packets.
//    In such a case, Insert() is expected to be called once for each packet.
//    The timestamp is not expected to change between those calls.
class RtpSequenceNumberMap final { … };

}  // namespace webrtc

#endif  // MODULES_RTP_RTCP_SOURCE_RTP_SEQUENCE_NUMBER_MAP_H_