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/rtp_sender_egress.h"

#include <algorithm>
#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 = …;
constexpr TimeDelta kUpdateInterval = …;

bool GetUseNtpTimeForAbsoluteSendTime(const FieldTrialsView* field_trials) { … }

}  // namespace

RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
    TaskQueueBase& worker_queue,
    RtpSenderEgress* sender,
    PacketSequencer* sequencer)
    : … { … }
RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() { … }

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

void RtpSenderEgress::NonPacedPacketSender::PrepareForSend(
    RtpPacketToSend* packet) { … }

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

RtpSenderEgress::~RtpSenderEgress() { … }

void RtpSenderEgress::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
                                 const PacedPacketInfo& pacing_info) { … }

void RtpSenderEgress::OnBatchComplete() { … }

void RtpSenderEgress::CompleteSendPacket(const Packet& compound_packet,
                                         bool last_in_batch) { … }

RtpSendRates RtpSenderEgress::GetSendRates(Timestamp now) const { … }

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

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

bool RtpSenderEgress::MediaHasBeenSent() const { … }

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

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

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

void RtpSenderEgress::SetFecProtectionParameters(
    const FecProtectionParams& delta_params,
    const FecProtectionParams& key_params) { … }

std::vector<std::unique_ptr<RtpPacketToSend>>
RtpSenderEgress::FetchFecPackets() { … }

void RtpSenderEgress::OnAbortedRetransmissions(
    rtc::ArrayView<const uint16_t> sequence_numbers) { … }

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

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

void RtpSenderEgress::UpdateRtpStats(Timestamp now,
                                     uint32_t packet_ssrc,
                                     RtpPacketMediaType packet_type,
                                     RtpPacketCounter counter,
                                     size_t packet_size) { … }

void RtpSenderEgress::PeriodicUpdate() { … }
}  // namespace webrtc
chromium/third_party/webrtc/modules/rtp_rtcp/source/rtp_sender_egress.cc
