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

/*
 *  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.
 */

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

#include <algorithm>
#include <cstdint>
#include <limits>
#include <memory>
#include <utility>

#include "modules/include/module_common_types_public.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/clock.h"

namespace webrtc {

namespace {

constexpr size_t kOldPayloadPaddingSizeHysteresis = …;
constexpr uint16_t kMaxOldPayloadPaddingSequenceNumber = …;

}  // namespace

RtpPacketHistory::StoredPacket::StoredPacket(
    std::unique_ptr<RtpPacketToSend> packet,
    Timestamp send_time,
    uint64_t insert_order)
    : … { … }

RtpPacketHistory::StoredPacket::StoredPacket(StoredPacket&&) = default;
RtpPacketHistory::StoredPacket& RtpPacketHistory::StoredPacket::operator=(
    RtpPacketHistory::StoredPacket&&) = default;
RtpPacketHistory::StoredPacket::~StoredPacket() = default;

void RtpPacketHistory::StoredPacket::IncrementTimesRetransmitted() { … }

RtpPacketHistory::RtpPacketHistory(Clock* clock, PaddingMode padding_mode)
    : … { … }

RtpPacketHistory::~RtpPacketHistory() { … }

void RtpPacketHistory::SetStorePacketsStatus(StorageMode mode,
                                             size_t number_to_store) { … }

RtpPacketHistory::StorageMode RtpPacketHistory::GetStorageMode() const { … }

void RtpPacketHistory::SetRtt(TimeDelta rtt) { … }

void RtpPacketHistory::PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
                                    Timestamp send_time) { … }

std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPacketAndMarkAsPending(
    uint16_t sequence_number) { … }

std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPacketAndMarkAsPending(
    uint16_t sequence_number,
    rtc::FunctionView<std::unique_ptr<RtpPacketToSend>(const RtpPacketToSend&)>
        encapsulate) { … }

void RtpPacketHistory::MarkPacketAsSent(uint16_t sequence_number) { … }

bool RtpPacketHistory::GetPacketState(uint16_t sequence_number) const { … }

bool RtpPacketHistory::VerifyRtt(
    const RtpPacketHistory::StoredPacket& packet) const { … }

std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPayloadPaddingPacket() { … }

std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPayloadPaddingPacket(
    rtc::FunctionView<std::unique_ptr<RtpPacketToSend>(const RtpPacketToSend&)>
        encapsulate) { … }

void RtpPacketHistory::CullAcknowledgedPackets(
    rtc::ArrayView<const uint16_t> sequence_numbers) { … }

void RtpPacketHistory::Clear() { … }

void RtpPacketHistory::Reset() { … }

void RtpPacketHistory::CullOldPackets() { … }

std::unique_ptr<RtpPacketToSend> RtpPacketHistory::RemovePacket(
    int packet_index) { … }

int RtpPacketHistory::GetPacketIndex(uint16_t sequence_number) const { … }

RtpPacketHistory::StoredPacket* RtpPacketHistory::GetStoredPacket(
    uint16_t sequence_number) { … }

}  // namespace webrtc