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webrtc/modules/rtp_rtcp/source/rtp_sender_egress.cc
Erik Språng bb90497eaa Add support for deferred sequence numbering. 
With this turned on, packets will be sequence number after the pacing
stage rather that during packetization.
This avoids a race where packets may be sent out of order, and paves
the way for the ability to cull packets from the pacer queue without
causing sequence number gaps.

For now, the feature is off by default. Follow-ups will enable it for
video and audio separately.

Bug: webrtc:11340, webrtc:12470
Change-Id: I6d411d8c85b9047e3e9b05ff4c2c3ed97c579aa1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208584
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34661}
2021-08-06 12:38:27 +00:00

662 lines
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/*
* 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/transport/field_trial_based_config.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "rtc_base/logging.h"
#include "rtc_base/task_utils/to_queued_task.h"
namespace webrtc {
namespace {
constexpr uint32_t kTimestampTicksPerMs = 90;
constexpr int kSendSideDelayWindowMs = 1000;
constexpr int kBitrateStatisticsWindowMs = 1000;
constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
constexpr TimeDelta kUpdateInterval =
TimeDelta::Millis(kBitrateStatisticsWindowMs);
bool IsTrialSetTo(const WebRtcKeyValueConfig* field_trials,
absl::string_view name,
absl::string_view value) {
FieldTrialBasedConfig default_trials;
auto& trials = field_trials ? *field_trials : default_trials;
return absl::StartsWith(trials.Lookup(name), value);
}
} // namespace
RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
RtpSenderEgress* sender,
SequenceNumberAssigner* sequence_number_assigner,
bool deferred_sequencing)
: deferred_sequencing_(deferred_sequencing),
transport_sequence_number_(0),
sender_(sender),
sequence_number_assigner_(sequence_number_assigner) {
RTC_DCHECK(sequence_number_assigner_);
}
RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() = default;
void RtpSenderEgress::NonPacedPacketSender::EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
for (auto& packet : packets) {
PrepareForSend(packet.get());
sender_->SendPacket(packet.get(), PacedPacketInfo());
}
auto fec_packets = sender_->FetchFecPackets();
if (!fec_packets.empty()) {
EnqueuePackets(std::move(fec_packets));
}
}
void RtpSenderEgress::NonPacedPacketSender::PrepareForSend(
RtpPacketToSend* packet) {
// Assign sequence numbers if deferred sequencing is used, but don't generate
// sequence numbers for flexfec, which is already running on an internally
// maintained sequence number series.
const bool is_flexfec = packet->Ssrc() == sender_->FlexFecSsrc();
if ((deferred_sequencing_ ||
packet->packet_type() == RtpPacketMediaType::kForwardErrorCorrection) &&
!is_flexfec) {
sequence_number_assigner_->AssignSequenceNumber(packet);
}
if (!packet->SetExtension<TransportSequenceNumber>(
++transport_sequence_number_)) {
--transport_sequence_number_;
}
packet->ReserveExtension<TransmissionOffset>();
packet->ReserveExtension<AbsoluteSendTime>();
}
RtpSenderEgress::RtpSenderEgress(const RtpRtcpInterface::Configuration& config,
RtpPacketHistory* packet_history)
: worker_queue_(TaskQueueBase::Current()),
ssrc_(config.local_media_ssrc),
rtx_ssrc_(config.rtx_send_ssrc),
flexfec_ssrc_(config.fec_generator ? config.fec_generator->FecSsrc()
: absl::nullopt),
populate_network2_timestamp_(config.populate_network2_timestamp),
send_side_bwe_with_overhead_(
!IsTrialSetTo(config.field_trials,
"WebRTC-SendSideBwe-WithOverhead",
"Disabled")),
deferred_sequencing_(config.use_deferred_sequencing),
clock_(config.clock),
packet_history_(packet_history),
transport_(config.outgoing_transport),
event_log_(config.event_log),
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
is_audio_(config.audio),
#endif
need_rtp_packet_infos_(config.need_rtp_packet_infos),
fec_generator_(config.fec_generator),
transport_feedback_observer_(config.transport_feedback_callback),
send_side_delay_observer_(config.send_side_delay_observer),
send_packet_observer_(config.send_packet_observer),
rtp_stats_callback_(config.rtp_stats_callback),
bitrate_callback_(config.send_bitrate_observer),
media_has_been_sent_(false),
force_part_of_allocation_(false),
timestamp_offset_(0),
max_delay_it_(send_delays_.end()),
sum_delays_ms_(0),
total_packet_send_delay_ms_(0),
send_rates_(kNumMediaTypes,
{kBitrateStatisticsWindowMs, RateStatistics::kBpsScale}),
rtp_sequence_number_map_(need_rtp_packet_infos_
? std::make_unique<RtpSequenceNumberMap>(
kRtpSequenceNumberMapMaxEntries)
: nullptr) {
RTC_DCHECK(worker_queue_);
pacer_checker_.Detach();
if (bitrate_callback_) {
update_task_ = RepeatingTaskHandle::DelayedStart(worker_queue_,
kUpdateInterval, [this]() {
PeriodicUpdate();
return kUpdateInterval;
});
}
}
RtpSenderEgress::~RtpSenderEgress() {
RTC_DCHECK_RUN_ON(worker_queue_);
update_task_.Stop();
}
void RtpSenderEgress::SendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK_RUN_ON(&pacer_checker_);
RTC_DCHECK(packet);
if (deferred_sequencing_) {
// Strict increasing of sequence numbers can only be guaranteed with
// deferred sequencing due to raciness with the pacer.
if (packet->Ssrc() == ssrc_ &&
packet->packet_type() != RtpPacketMediaType::kRetransmission) {
if (last_sent_seq_.has_value()) {
RTC_DCHECK_EQ(static_cast<uint16_t>(*last_sent_seq_ + 1),
packet->SequenceNumber());
}
last_sent_seq_ = packet->SequenceNumber();
} else if (packet->Ssrc() == rtx_ssrc_) {
if (last_sent_rtx_seq_.has_value()) {
RTC_DCHECK_EQ(static_cast<uint16_t>(*last_sent_rtx_seq_ + 1),
packet->SequenceNumber());
}
last_sent_rtx_seq_ = packet->SequenceNumber();
}
}
RTC_DCHECK(packet->packet_type().has_value());
RTC_DCHECK(HasCorrectSsrc(*packet));
if (packet->packet_type() == RtpPacketMediaType::kRetransmission) {
RTC_DCHECK(packet->retransmitted_sequence_number().has_value());
}
const uint32_t packet_ssrc = packet->Ssrc();
const int64_t now_ms = clock_->TimeInMilliseconds();
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
worker_queue_->PostTask(
ToQueuedTask(task_safety_, [this, now_ms, packet_ssrc]() {
BweTestLoggingPlot(now_ms, packet_ssrc);
}));
#endif
if (need_rtp_packet_infos_ &&
packet->packet_type() == RtpPacketToSend::Type::kVideo) {
worker_queue_->PostTask(ToQueuedTask(
task_safety_,
[this, packet_timestamp = packet->Timestamp(),
is_first_packet_of_frame = packet->is_first_packet_of_frame(),
is_last_packet_of_frame = packet->Marker(),
sequence_number = packet->SequenceNumber()]() {
RTC_DCHECK_RUN_ON(worker_queue_);
// Last packet of a frame, add it to sequence number info map.
const uint32_t timestamp = packet_timestamp - timestamp_offset_;
rtp_sequence_number_map_->InsertPacket(
sequence_number,
RtpSequenceNumberMap::Info(timestamp, is_first_packet_of_frame,
is_last_packet_of_frame));
}));
}
if (fec_generator_ && packet->fec_protect_packet()) {
// This packet should be protected by FEC, add it to packet generator.
RTC_DCHECK(fec_generator_);
RTC_DCHECK(packet->packet_type() == RtpPacketMediaType::kVideo);
absl::optional<std::pair<FecProtectionParams, FecProtectionParams>>
new_fec_params;
{
MutexLock lock(&lock_);
new_fec_params.swap(pending_fec_params_);
}
if (new_fec_params) {
fec_generator_->SetProtectionParameters(new_fec_params->first,
new_fec_params->second);
}
if (packet->is_red()) {
RtpPacketToSend unpacked_packet(*packet);
const rtc::CopyOnWriteBuffer buffer = packet->Buffer();
// Grab media payload type from RED header.
const size_t headers_size = packet->headers_size();
unpacked_packet.SetPayloadType(buffer[headers_size]);
// Copy the media payload into the unpacked buffer.
uint8_t* payload_buffer =
unpacked_packet.SetPayloadSize(packet->payload_size() - 1);
std::copy(&packet->payload()[0] + 1,
&packet->payload()[0] + packet->payload_size(), payload_buffer);
fec_generator_->AddPacketAndGenerateFec(unpacked_packet);
} else {
// If not RED encapsulated - we can just insert packet directly.
fec_generator_->AddPacketAndGenerateFec(*packet);
}
}
// Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
// the pacer, these modifications of the header below are happening after the
// FEC protection packets are calculated. This will corrupt recovered packets
// at the same place. It's not an issue for extensions, which are present in
// all the packets (their content just may be incorrect on recovered packets).
// In case of VideoTimingExtension, since it's present not in every packet,
// data after rtp header may be corrupted if these packets are protected by
// the FEC.
int64_t diff_ms = now_ms - packet->capture_time_ms();
if (packet->HasExtension<TransmissionOffset>()) {
packet->SetExtension<TransmissionOffset>(kTimestampTicksPerMs * diff_ms);
}
if (packet->HasExtension<AbsoluteSendTime>()) {
packet->SetExtension<AbsoluteSendTime>(
AbsoluteSendTime::MsTo24Bits(now_ms));
}
if (packet->HasExtension<VideoTimingExtension>()) {
if (populate_network2_timestamp_) {
packet->set_network2_time_ms(now_ms);
} else {
packet->set_pacer_exit_time_ms(now_ms);
}
}
const bool is_media = packet->packet_type() == RtpPacketMediaType::kAudio ||
packet->packet_type() == RtpPacketMediaType::kVideo;
PacketOptions options;
{
MutexLock lock(&lock_);
options.included_in_allocation = force_part_of_allocation_;
}
// Downstream code actually uses this flag to distinguish between media and
// everything else.
options.is_retransmit = !is_media;
if (auto packet_id = packet->GetExtension<TransportSequenceNumber>()) {
options.packet_id = *packet_id;
options.included_in_feedback = true;
options.included_in_allocation = true;
AddPacketToTransportFeedback(*packet_id, *packet, pacing_info);
}
options.additional_data = packet->additional_data();
if (packet->packet_type() != RtpPacketMediaType::kPadding &&
packet->packet_type() != RtpPacketMediaType::kRetransmission) {
UpdateDelayStatistics(packet->capture_time_ms(), now_ms, packet_ssrc);
UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
packet_ssrc);
}
const bool send_success = SendPacketToNetwork(*packet, options, pacing_info);
// Put packet in retransmission history or update pending status even if
// actual sending fails.
if (is_media && packet->allow_retransmission()) {
packet_history_->PutRtpPacket(std::make_unique<RtpPacketToSend>(*packet),
now_ms);
} else if (packet->retransmitted_sequence_number()) {
packet_history_->MarkPacketAsSent(*packet->retransmitted_sequence_number());
}
if (send_success) {
// |media_has_been_sent_| is used by RTPSender to figure out if it can send
// padding in the absence of transport-cc or abs-send-time.
// In those cases media must be sent first to set a reference timestamp.
media_has_been_sent_ = true;
// TODO(sprang): Add support for FEC protecting all header extensions, add
// media packet to generator here instead.
RTC_DCHECK(packet->packet_type().has_value());
RtpPacketMediaType packet_type = *packet->packet_type();
RtpPacketCounter counter(*packet);
size_t size = packet->size();
worker_queue_->PostTask(
ToQueuedTask(task_safety_, [this, now_ms, packet_ssrc, packet_type,
counter = std::move(counter), size]() {
RTC_DCHECK_RUN_ON(worker_queue_);
UpdateRtpStats(now_ms, packet_ssrc, packet_type, std::move(counter),
size);
}));
}
}
RtpSendRates RtpSenderEgress::GetSendRates() const {
MutexLock lock(&lock_);
const int64_t now_ms = clock_->TimeInMilliseconds();
return GetSendRatesLocked(now_ms);
}
RtpSendRates RtpSenderEgress::GetSendRatesLocked(int64_t now_ms) const {
RtpSendRates current_rates;
for (size_t i = 0; i < kNumMediaTypes; ++i) {
RtpPacketMediaType type = static_cast<RtpPacketMediaType>(i);
current_rates[type] =
DataRate::BitsPerSec(send_rates_[i].Rate(now_ms).value_or(0));
}
return current_rates;
}
void RtpSenderEgress::GetDataCounters(StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const {
// TODO(bugs.webrtc.org/11581): make sure rtx_rtp_stats_ and rtp_stats_ are
// only touched on the worker thread.
MutexLock lock(&lock_);
*rtp_stats = rtp_stats_;
*rtx_stats = rtx_rtp_stats_;
}
void RtpSenderEgress::ForceIncludeSendPacketsInAllocation(
bool part_of_allocation) {
MutexLock lock(&lock_);
force_part_of_allocation_ = part_of_allocation;
}
bool RtpSenderEgress::MediaHasBeenSent() const {
RTC_DCHECK_RUN_ON(&pacer_checker_);
return media_has_been_sent_;
}
void RtpSenderEgress::SetMediaHasBeenSent(bool media_sent) {
RTC_DCHECK_RUN_ON(&pacer_checker_);
media_has_been_sent_ = media_sent;
}
void RtpSenderEgress::SetTimestampOffset(uint32_t timestamp) {
RTC_DCHECK_RUN_ON(worker_queue_);
timestamp_offset_ = timestamp;
}
std::vector<RtpSequenceNumberMap::Info> RtpSenderEgress::GetSentRtpPacketInfos(
rtc::ArrayView<const uint16_t> sequence_numbers) const {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_DCHECK(!sequence_numbers.empty());
if (!need_rtp_packet_infos_) {
return std::vector<RtpSequenceNumberMap::Info>();
}
std::vector<RtpSequenceNumberMap::Info> results;
results.reserve(sequence_numbers.size());
for (uint16_t sequence_number : sequence_numbers) {
const auto& info = rtp_sequence_number_map_->Get(sequence_number);
if (!info) {
// The empty vector will be returned. We can delay the clearing
// of the vector until after we exit the critical section.
return std::vector<RtpSequenceNumberMap::Info>();
}
results.push_back(*info);
}
return results;
}
void RtpSenderEgress::SetFecProtectionParameters(
const FecProtectionParams& delta_params,
const FecProtectionParams& key_params) {
// TODO(sprang): Post task to pacer queue instead, one pacer is fully
// migrated to a task queue.
MutexLock lock(&lock_);
pending_fec_params_.emplace(delta_params, key_params);
}
std::vector<std::unique_ptr<RtpPacketToSend>>
RtpSenderEgress::FetchFecPackets() {
RTC_DCHECK_RUN_ON(&pacer_checker_);
if (fec_generator_) {
return fec_generator_->GetFecPackets();
}
return {};
}
bool RtpSenderEgress::HasCorrectSsrc(const RtpPacketToSend& packet) const {
switch (*packet.packet_type()) {
case RtpPacketMediaType::kAudio:
case RtpPacketMediaType::kVideo:
return packet.Ssrc() == ssrc_;
case RtpPacketMediaType::kRetransmission:
case RtpPacketMediaType::kPadding:
// Both padding and retransmission must be on either the media or the
// RTX stream.
return packet.Ssrc() == rtx_ssrc_ || packet.Ssrc() == ssrc_;
case RtpPacketMediaType::kForwardErrorCorrection:
// FlexFEC is on separate SSRC, ULPFEC uses media SSRC.
return packet.Ssrc() == ssrc_ || packet.Ssrc() == flexfec_ssrc_;
}
return false;
}
void RtpSenderEgress::AddPacketToTransportFeedback(
uint16_t packet_id,
const RtpPacketToSend& packet,
const PacedPacketInfo& pacing_info) {
if (transport_feedback_observer_) {
size_t packet_size = packet.payload_size() + packet.padding_size();
if (send_side_bwe_with_overhead_) {
packet_size = packet.size();
}
RtpPacketSendInfo packet_info;
packet_info.transport_sequence_number = packet_id;
packet_info.rtp_timestamp = packet.Timestamp();
packet_info.length = packet_size;
packet_info.pacing_info = pacing_info;
packet_info.packet_type = packet.packet_type();
switch (*packet_info.packet_type) {
case RtpPacketMediaType::kAudio:
case RtpPacketMediaType::kVideo:
packet_info.media_ssrc = ssrc_;
packet_info.rtp_sequence_number = packet.SequenceNumber();
break;
case RtpPacketMediaType::kRetransmission:
// For retransmissions, we're want to remove the original media packet
// if the rentrasmit arrives - so populate that in the packet info.
packet_info.media_ssrc = ssrc_;
packet_info.rtp_sequence_number =
*packet.retransmitted_sequence_number();
break;
case RtpPacketMediaType::kPadding:
case RtpPacketMediaType::kForwardErrorCorrection:
// We're not interested in feedback about these packets being received
// or lost.
break;
}
// TODO(bugs.webrtc.org/12713): Remove once downstream usage is gone.
packet_info.ssrc = packet_info.media_ssrc.value_or(0);
transport_feedback_observer_->OnAddPacket(packet_info);
}
}
void RtpSenderEgress::UpdateDelayStatistics(int64_t capture_time_ms,
int64_t now_ms,
uint32_t ssrc) {
if (!send_side_delay_observer_ || capture_time_ms <= 0)
return;
int avg_delay_ms = 0;
int max_delay_ms = 0;
uint64_t total_packet_send_delay_ms = 0;
{
MutexLock lock(&lock_);
// Compute the max and average of the recent capture-to-send delays.
// The time complexity of the current approach depends on the distribution
// of the delay values. This could be done more efficiently.
// Remove elements older than kSendSideDelayWindowMs.
auto lower_bound =
send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs);
for (auto it = send_delays_.begin(); it != lower_bound; ++it) {
if (max_delay_it_ == it) {
max_delay_it_ = send_delays_.end();
}
sum_delays_ms_ -= it->second;
}
send_delays_.erase(send_delays_.begin(), lower_bound);
if (max_delay_it_ == send_delays_.end()) {
// Removed the previous max. Need to recompute.
RecomputeMaxSendDelay();
}
// Add the new element.
RTC_DCHECK_GE(now_ms, 0);
RTC_DCHECK_LE(now_ms, std::numeric_limits<int64_t>::max() / 2);
RTC_DCHECK_GE(capture_time_ms, 0);
RTC_DCHECK_LE(capture_time_ms, std::numeric_limits<int64_t>::max() / 2);
int64_t diff_ms = now_ms - capture_time_ms;
RTC_DCHECK_GE(diff_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(diff_ms, std::numeric_limits<int>::max());
int new_send_delay = rtc::dchecked_cast<int>(now_ms - capture_time_ms);
SendDelayMap::iterator it;
bool inserted;
std::tie(it, inserted) =
send_delays_.insert(std::make_pair(now_ms, new_send_delay));
if (!inserted) {
// TODO(terelius): If we have multiple delay measurements during the same
// millisecond then we keep the most recent one. It is not clear that this
// is the right decision, but it preserves an earlier behavior.
int previous_send_delay = it->second;
sum_delays_ms_ -= previous_send_delay;
it->second = new_send_delay;
if (max_delay_it_ == it && new_send_delay < previous_send_delay) {
RecomputeMaxSendDelay();
}
}
if (max_delay_it_ == send_delays_.end() ||
it->second >= max_delay_it_->second) {
max_delay_it_ = it;
}
sum_delays_ms_ += new_send_delay;
total_packet_send_delay_ms_ += new_send_delay;
total_packet_send_delay_ms = total_packet_send_delay_ms_;
size_t num_delays = send_delays_.size();
RTC_DCHECK(max_delay_it_ != send_delays_.end());
max_delay_ms = rtc::dchecked_cast<int>(max_delay_it_->second);
int64_t avg_ms = (sum_delays_ms_ + num_delays / 2) / num_delays;
RTC_DCHECK_GE(avg_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(avg_ms,
static_cast<int64_t>(std::numeric_limits<int>::max()));
avg_delay_ms =
rtc::dchecked_cast<int>((sum_delays_ms_ + num_delays / 2) / num_delays);
}
send_side_delay_observer_->SendSideDelayUpdated(
avg_delay_ms, max_delay_ms, total_packet_send_delay_ms, ssrc);
}
void RtpSenderEgress::RecomputeMaxSendDelay() {
max_delay_it_ = send_delays_.begin();
for (auto it = send_delays_.begin(); it != send_delays_.end(); ++it) {
if (it->second >= max_delay_it_->second) {
max_delay_it_ = it;
}
}
}
void RtpSenderEgress::UpdateOnSendPacket(int packet_id,
int64_t capture_time_ms,
uint32_t ssrc) {
if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1) {
return;
}
send_packet_observer_->OnSendPacket(packet_id, capture_time_ms, ssrc);
}
bool RtpSenderEgress::SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options,
const PacedPacketInfo& pacing_info) {
int bytes_sent = -1;
if (transport_) {
bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
? static_cast<int>(packet.size())
: -1;
if (event_log_ && bytes_sent > 0) {
event_log_->Log(std::make_unique<RtcEventRtpPacketOutgoing>(
packet, pacing_info.probe_cluster_id));
}
}
if (bytes_sent <= 0) {
RTC_LOG(LS_WARNING) << "Transport failed to send packet.";
return false;
}
return true;
}
void RtpSenderEgress::UpdateRtpStats(int64_t now_ms,
uint32_t packet_ssrc,
RtpPacketMediaType packet_type,
RtpPacketCounter counter,
size_t packet_size) {
RTC_DCHECK_RUN_ON(worker_queue_);
// TODO(bugs.webrtc.org/11581): send_rates_ should be touched only on the
// worker thread.
RtpSendRates send_rates;
{
MutexLock lock(&lock_);
// TODO(bugs.webrtc.org/11581): make sure rtx_rtp_stats_ and rtp_stats_ are
// only touched on the worker thread.
StreamDataCounters* counters =
packet_ssrc == rtx_ssrc_ ? &rtx_rtp_stats_ : &rtp_stats_;
if (counters->first_packet_time_ms == -1) {
counters->first_packet_time_ms = now_ms;
}
if (packet_type == RtpPacketMediaType::kForwardErrorCorrection) {
counters->fec.Add(counter);
} else if (packet_type == RtpPacketMediaType::kRetransmission) {
counters->retransmitted.Add(counter);
}
counters->transmitted.Add(counter);
send_rates_[static_cast<size_t>(packet_type)].Update(packet_size, now_ms);
if (bitrate_callback_) {
send_rates = GetSendRatesLocked(now_ms);
}
if (rtp_stats_callback_) {
rtp_stats_callback_->DataCountersUpdated(*counters, packet_ssrc);
}
}
// The bitrate_callback_ and rtp_stats_callback_ pointers in practice point
// to the same object, so these callbacks could be consolidated into one.
if (bitrate_callback_) {
bitrate_callback_->Notify(
send_rates.Sum().bps(),
send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
}
}
void RtpSenderEgress::PeriodicUpdate() {
RTC_DCHECK_RUN_ON(worker_queue_);
RTC_DCHECK(bitrate_callback_);
RtpSendRates send_rates = GetSendRates();
bitrate_callback_->Notify(
send_rates.Sum().bps(),
send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
}
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
void RtpSenderEgress::BweTestLoggingPlot(int64_t now_ms, uint32_t packet_ssrc) {
RTC_DCHECK_RUN_ON(worker_queue_);
const auto rates = GetSendRates();
if (is_audio_) {
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
rates.Sum().kbps(), packet_ssrc);
BWE_TEST_LOGGING_PLOT_WITH_SSRC(
1, "AudioNackBitrate_kbps", now_ms,
rates[RtpPacketMediaType::kRetransmission].kbps(), packet_ssrc);
} else {
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
rates.Sum().kbps(), packet_ssrc);
BWE_TEST_LOGGING_PLOT_WITH_SSRC(
1, "VideoNackBitrate_kbps", now_ms,
rates[RtpPacketMediaType::kRetransmission].kbps(), packet_ssrc);
}
}
#endif // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
} // namespace webrtc
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