chromium/third_party/webrtc/modules/rtp_rtcp/source/packet_sequencer.cc

/*
 *  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.
 */

#include "modules/rtp_rtcp/source/packet_sequencer.h"

#include "rtc_base/checks.h"
#include "rtc_base/random.h"

namespace webrtc {

namespace {
// RED header is first byte of payload, if present.
constexpr size_t kRedForFecHeaderLength = …;

// Timestamps use a 90kHz clock.
constexpr uint32_t kTimestampTicksPerMs = …;
}  // namespace

PacketSequencer::PacketSequencer(uint32_t media_ssrc,
                                 absl::optional<uint32_t> rtx_ssrc,
                                 bool require_marker_before_media_padding,
                                 Clock* clock)
    : … { … }

void PacketSequencer::Sequence(RtpPacketToSend& packet) { … }

void PacketSequencer::SetRtpState(const RtpState& state) { … }

void PacketSequencer::PopulateRtpState(RtpState& state) const { … }

void PacketSequencer::UpdateLastPacketState(const RtpPacketToSend& packet) { … }

void PacketSequencer::PopulatePaddingFields(RtpPacketToSend& packet) { … }

bool PacketSequencer::CanSendPaddingOnMediaSsrc() const { … }

}  // namespace webrtc