deprecated_rtp_sender_egress.cc | Explore in Territory

/*
 *  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.
 */

#include "modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h"

#include <limits>
#include <memory>
#include <utility>

#include "absl/strings/match.h"
#include "api/units/timestamp.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "rtc_base/logging.h"

namespace webrtc {
namespace {
constexpr uint32_t kTimestampTicksPerMs = …;
constexpr TimeDelta kBitrateStatisticsWindow = …;
constexpr size_t kRtpSequenceNumberMapMaxEntries = …;

}  // namespace

DEPRECATED_RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
    DEPRECATED_RtpSenderEgress* sender,
    PacketSequencer* sequence_number_assigner)
    : … { … }
DEPRECATED_RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() =
    default;

void DEPRECATED_RtpSenderEgress::NonPacedPacketSender::EnqueuePackets(
    std::vector<std::unique_ptr<RtpPacketToSend>> packets) { … }

DEPRECATED_RtpSenderEgress::DEPRECATED_RtpSenderEgress(
    const RtpRtcpInterface::Configuration& config,
    RtpPacketHistory* packet_history)
    : … { … }

void DEPRECATED_RtpSenderEgress::SendPacket(
    RtpPacketToSend* packet,
    const PacedPacketInfo& pacing_info) { … }

void DEPRECATED_RtpSenderEgress::ProcessBitrateAndNotifyObservers() { … }

RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRates() const { … }

RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRatesLocked() const { … }

void DEPRECATED_RtpSenderEgress::GetDataCounters(
    StreamDataCounters* rtp_stats,
    StreamDataCounters* rtx_stats) const { … }

void DEPRECATED_RtpSenderEgress::ForceIncludeSendPacketsInAllocation(
    bool part_of_allocation) { … }

bool DEPRECATED_RtpSenderEgress::MediaHasBeenSent() const { … }

void DEPRECATED_RtpSenderEgress::SetMediaHasBeenSent(bool media_sent) { … }

void DEPRECATED_RtpSenderEgress::SetTimestampOffset(uint32_t timestamp) { … }

std::vector<RtpSequenceNumberMap::Info>
DEPRECATED_RtpSenderEgress::GetSentRtpPacketInfos(
    rtc::ArrayView<const uint16_t> sequence_numbers) const { … }

bool DEPRECATED_RtpSenderEgress::HasCorrectSsrc(
    const RtpPacketToSend& packet) const { … }

void DEPRECATED_RtpSenderEgress::AddPacketToTransportFeedback(
    uint16_t packet_id,
    const RtpPacketToSend& packet,
    const PacedPacketInfo& pacing_info) { … }

void DEPRECATED_RtpSenderEgress::UpdateOnSendPacket(int packet_id,
                                                    int64_t capture_time_ms,
                                                    uint32_t ssrc) { … }

bool DEPRECATED_RtpSenderEgress::SendPacketToNetwork(
    const RtpPacketToSend& packet,
    const PacketOptions& options,
    const PacedPacketInfo& pacing_info) { … }

void DEPRECATED_RtpSenderEgress::UpdateRtpStats(const RtpPacketToSend& packet) { … }

}  // namespace webrtc
chromium/third_party/webrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
