modules/rtp_rtcp/source/rtp_sender_egress.cc - src - Git at Google

 /*
  *  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)
     : 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()) {
     // Don't generate sequence numbers for flexfec, they are already running on
     // an internally maintained sequence.
     const bool generate_sequence_numbers = !sender_->FlexFecSsrc().has_value();

     for (auto& packet : fec_packets) {
       if (generate_sequence_numbers) {
         sequence_number_assigner_->AssignSequenceNumber(packet.get());
       }
       PrepareForSend(packet.get());
     }
     EnqueuePackets(std::move(fec_packets));
   }
 }

 void RtpSenderEgress::NonPacedPacketSender::PrepareForSend(
     RtpPacketToSend* 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")),
       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);

   RTC_DCHECK(packet->packet_type().has_value());
   RTC_DCHECK(HasCorrectSsrc(*packet));

   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.ssrc = ssrc_;
     packet_info.transport_sequence_number = packet_id;
     packet_info.rtp_sequence_number = packet.SequenceNumber();
     packet_info.length = packet_size;
     packet_info.pacing_info = pacing_info;
     packet_info.packet_type = packet.packet_type();
     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
	/*
	* 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)
	: 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()) {
	// Don't generate sequence numbers for flexfec, they are already running on
	// an internally maintained sequence.
	const bool generate_sequence_numbers = !sender_->FlexFecSsrc().has_value();

	for (auto& packet : fec_packets) {
	if (generate_sequence_numbers) {
	sequence_number_assigner_->AssignSequenceNumber(packet.get());
	}
	PrepareForSend(packet.get());
	}
	EnqueuePackets(std::move(fec_packets));
	}
	}

	void RtpSenderEgress::NonPacedPacketSender::PrepareForSend(
	RtpPacketToSend* 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")),
	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);

	RTC_DCHECK(packet->packet_type().has_value());
	RTC_DCHECK(HasCorrectSsrc(*packet));

	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.ssrc = ssrc_;
	packet_info.transport_sequence_number = packet_id;
	packet_info.rtp_sequence_number = packet.SequenceNumber();
	packet_info.length = packet_size;
	packet_info.pacing_info = pacing_info;
	packet_info.packet_type = packet.packet_type();
	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