chromium/third_party/webrtc/modules/audio_coding/neteq/packet_arrival_history.cc

/*
 *  Copyright (c) 2022 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/audio_coding/neteq/packet_arrival_history.h"

#include <algorithm>
#include <cstdint>

#include "api/neteq/tick_timer.h"
#include "rtc_base/checks.h"

namespace webrtc {

PacketArrivalHistory::PacketArrivalHistory(const TickTimer* tick_timer,
                                           int window_size_ms)
    : … { … }

bool PacketArrivalHistory::Insert(uint32_t rtp_timestamp,
                                  int packet_length_samples) { … }

void PacketArrivalHistory::Reset() { … }

int PacketArrivalHistory::GetDelayMs(uint32_t rtp_timestamp) const { … }

int PacketArrivalHistory::GetMaxDelayMs() const { … }

bool PacketArrivalHistory::IsNewestRtpTimestamp(uint32_t rtp_timestamp) const { … }

int PacketArrivalHistory::GetPacketArrivalDelayMs(
    const PacketArrival& packet_arrival) const { … }

bool PacketArrivalHistory::IsObsolete(
    const PacketArrival& packet_arrival) const { … }

bool PacketArrivalHistory::Contains(const PacketArrival& packet_arrival) const { … }

}  // namespace webrtc