modules/rtp_rtcp/source/deprecated/deprecated_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/deprecated/deprecated_rtp_sender_egress.h"

 #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 "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "rtc_base/logging.h"

 namespace webrtc {
 namespace {
 constexpr uint32_t kTimestampTicksPerMs = 90;
 constexpr int kSendSideDelayWindowMs = 1000;
 constexpr int kBitrateStatisticsWindowMs = 1000;
 constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;

 bool IsDisabled(absl::string_view name,
                 const WebRtcKeyValueConfig* field_trials) {
   FieldTrialBasedConfig default_trials;
   auto& trials = field_trials ? *field_trials : default_trials;
   return absl::StartsWith(trials.Lookup(name), "Disabled");
 }
 }  // namespace

 DEPRECATED_RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
     DEPRECATED_RtpSenderEgress* sender,
     PacketSequencer* sequence_number_assigner)
     : transport_sequence_number_(0),
       sender_(sender),
       sequence_number_assigner_(sequence_number_assigner) {
   RTC_DCHECK(sequence_number_assigner_);
 }
 DEPRECATED_RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() =
     default;

 void DEPRECATED_RtpSenderEgress::NonPacedPacketSender::EnqueuePackets(
     std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
   for (auto& packet : packets) {
     // Assign sequence numbers, but not for flexfec which is already running on
     // an internally maintained sequence number series.
     if (packet->Ssrc() != sender_->FlexFecSsrc()) {
       sequence_number_assigner_->Sequence(*packet);
     }
     if (!packet->SetExtension<TransportSequenceNumber>(
             ++transport_sequence_number_)) {
       --transport_sequence_number_;
     }
     packet->ReserveExtension<TransmissionOffset>();
     packet->ReserveExtension<AbsoluteSendTime>();
     sender_->SendPacket(packet.get(), PacedPacketInfo());
   }
 }

 DEPRECATED_RtpSenderEgress::DEPRECATED_RtpSenderEgress(
     const RtpRtcpInterface::Configuration& config,
     RtpPacketHistory* packet_history)
     : 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_(
           !IsDisabled("WebRTC-SendSideBwe-WithOverhead", config.field_trials)),
       clock_(config.clock),
       packet_history_(packet_history),
       transport_(config.outgoing_transport),
       event_log_(config.event_log),
       is_audio_(config.audio),
       need_rtp_packet_infos_(config.need_rtp_packet_infos),
       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) {}

 void DEPRECATED_RtpSenderEgress::SendPacket(
     RtpPacketToSend* packet,
     const PacedPacketInfo& pacing_info) {
   RTC_DCHECK(packet);

   const uint32_t packet_ssrc = packet->Ssrc();
   RTC_DCHECK(packet->packet_type().has_value());
   RTC_DCHECK(HasCorrectSsrc(*packet));
   int64_t now_ms = clock_->TimeInMilliseconds();

   if (is_audio_) {
 #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
     BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
                                     GetSendRates().Sum().kbps(), packet_ssrc);
     BWE_TEST_LOGGING_PLOT_WITH_SSRC(
         1, "AudioNackBitrate_kbps", now_ms,
         GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
         packet_ssrc);
 #endif
   } else {
 #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
     BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
                                     GetSendRates().Sum().kbps(), packet_ssrc);
     BWE_TEST_LOGGING_PLOT_WITH_SSRC(
         1, "VideoNackBitrate_kbps", now_ms,
         GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
         packet_ssrc);
 #endif
   }

   PacketOptions options;
   {
     MutexLock lock(&lock_);
     options.included_in_allocation = force_part_of_allocation_;

     if (need_rtp_packet_infos_ &&
         packet->packet_type() == RtpPacketToSend::Type::kVideo) {
       RTC_DCHECK(rtp_sequence_number_map_);
       // Last packet of a frame, add it to sequence number info map.
       const uint32_t timestamp = packet->Timestamp() - timestamp_offset_;
       bool is_first_packet_of_frame = packet->is_first_packet_of_frame();
       bool is_last_packet_of_frame = packet->Marker();

       rtp_sequence_number_map_->InsertPacket(
           packet->SequenceNumber(),
           RtpSequenceNumberMap::Info(timestamp, is_first_packet_of_frame,
                                      is_last_packet_of_frame));
     }
   }

   // 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;

   // 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) {
     MutexLock lock(&lock_);
     UpdateRtpStats(*packet);
     media_has_been_sent_ = true;
   }
 }

 void DEPRECATED_RtpSenderEgress::ProcessBitrateAndNotifyObservers() {
   if (!bitrate_callback_)
     return;

   MutexLock lock(&lock_);
   RtpSendRates send_rates = GetSendRatesLocked();
   bitrate_callback_->Notify(
       send_rates.Sum().bps(),
       send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
 }

 RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRates() const {
   MutexLock lock(&lock_);
   return GetSendRatesLocked();
 }

 RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRatesLocked() const {
   const int64_t now_ms = clock_->TimeInMilliseconds();
   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 DEPRECATED_RtpSenderEgress::GetDataCounters(
     StreamDataCounters* rtp_stats,
     StreamDataCounters* rtx_stats) const {
   MutexLock lock(&lock_);
   *rtp_stats = rtp_stats_;
   *rtx_stats = rtx_rtp_stats_;
 }

 void DEPRECATED_RtpSenderEgress::ForceIncludeSendPacketsInAllocation(
     bool part_of_allocation) {
   MutexLock lock(&lock_);
   force_part_of_allocation_ = part_of_allocation;
 }

 bool DEPRECATED_RtpSenderEgress::MediaHasBeenSent() const {
   MutexLock lock(&lock_);
   return media_has_been_sent_;
 }

 void DEPRECATED_RtpSenderEgress::SetMediaHasBeenSent(bool media_sent) {
   MutexLock lock(&lock_);
   media_has_been_sent_ = media_sent;
 }

 void DEPRECATED_RtpSenderEgress::SetTimestampOffset(uint32_t timestamp) {
   MutexLock lock(&lock_);
   timestamp_offset_ = timestamp;
 }

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

   MutexLock lock(&lock_);
   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;
 }

 bool DEPRECATED_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 DEPRECATED_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;
     // TODO(bugs.webrtc.org/12713): Remove once downstream usage is gone.
     packet_info.ssrc = ssrc_;
     packet_info.media_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 DEPRECATED_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 DEPRECATED_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 DEPRECATED_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 DEPRECATED_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 DEPRECATED_RtpSenderEgress::UpdateRtpStats(const RtpPacketToSend& packet) {
   int64_t now_ms = clock_->TimeInMilliseconds();

   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.packet_type() == RtpPacketMediaType::kForwardErrorCorrection) {
     counters->fec.AddPacket(packet);
   }

   if (packet.packet_type() == RtpPacketMediaType::kRetransmission) {
     counters->retransmitted.AddPacket(packet);
   }
   counters->transmitted.AddPacket(packet);

   RTC_DCHECK(packet.packet_type().has_value());
   send_rates_[static_cast<size_t>(*packet.packet_type())].Update(packet.size(),
                                                                  now_ms);

   if (rtp_stats_callback_) {
     rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc());
   }
 }

 }  // 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/deprecated/deprecated_rtp_sender_egress.h"

	#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 "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
	#include "rtc_base/logging.h"

	namespace webrtc {
	namespace {
	constexpr uint32_t kTimestampTicksPerMs = 90;
	constexpr int kSendSideDelayWindowMs = 1000;
	constexpr int kBitrateStatisticsWindowMs = 1000;
	constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;

	bool IsDisabled(absl::string_view name,
	const WebRtcKeyValueConfig* field_trials) {
	FieldTrialBasedConfig default_trials;
	auto& trials = field_trials ? *field_trials : default_trials;
	return absl::StartsWith(trials.Lookup(name), "Disabled");
	}
	} // namespace

	DEPRECATED_RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
	DEPRECATED_RtpSenderEgress* sender,
	PacketSequencer* sequence_number_assigner)
	: transport_sequence_number_(0),
	sender_(sender),
	sequence_number_assigner_(sequence_number_assigner) {
	RTC_DCHECK(sequence_number_assigner_);
	}
	DEPRECATED_RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() =
	default;

	void DEPRECATED_RtpSenderEgress::NonPacedPacketSender::EnqueuePackets(
	std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
	for (auto& packet : packets) {
	// Assign sequence numbers, but not for flexfec which is already running on
	// an internally maintained sequence number series.
	if (packet->Ssrc() != sender_->FlexFecSsrc()) {
	sequence_number_assigner_->Sequence(*packet);
	}
	if (!packet->SetExtension<TransportSequenceNumber>(
	++transport_sequence_number_)) {
	--transport_sequence_number_;
	}
	packet->ReserveExtension<TransmissionOffset>();
	packet->ReserveExtension<AbsoluteSendTime>();
	sender_->SendPacket(packet.get(), PacedPacketInfo());
	}
	}

	DEPRECATED_RtpSenderEgress::DEPRECATED_RtpSenderEgress(
	const RtpRtcpInterface::Configuration& config,
	RtpPacketHistory* packet_history)
	: 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_(
	!IsDisabled("WebRTC-SendSideBwe-WithOverhead", config.field_trials)),
	clock_(config.clock),
	packet_history_(packet_history),
	transport_(config.outgoing_transport),
	event_log_(config.event_log),
	is_audio_(config.audio),
	need_rtp_packet_infos_(config.need_rtp_packet_infos),
	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) {}

	void DEPRECATED_RtpSenderEgress::SendPacket(
	RtpPacketToSend* packet,
	const PacedPacketInfo& pacing_info) {
	RTC_DCHECK(packet);

	const uint32_t packet_ssrc = packet->Ssrc();
	RTC_DCHECK(packet->packet_type().has_value());
	RTC_DCHECK(HasCorrectSsrc(*packet));
	int64_t now_ms = clock_->TimeInMilliseconds();

	if (is_audio_) {
	#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
	BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
	GetSendRates().Sum().kbps(), packet_ssrc);
	BWE_TEST_LOGGING_PLOT_WITH_SSRC(
	1, "AudioNackBitrate_kbps", now_ms,
	GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
	packet_ssrc);
	#endif
	} else {
	#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
	BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
	GetSendRates().Sum().kbps(), packet_ssrc);
	BWE_TEST_LOGGING_PLOT_WITH_SSRC(
	1, "VideoNackBitrate_kbps", now_ms,
	GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
	packet_ssrc);
	#endif
	}

	PacketOptions options;
	{
	MutexLock lock(&lock_);
	options.included_in_allocation = force_part_of_allocation_;

	if (need_rtp_packet_infos_ &&
	packet->packet_type() == RtpPacketToSend::Type::kVideo) {
	RTC_DCHECK(rtp_sequence_number_map_);
	// Last packet of a frame, add it to sequence number info map.
	const uint32_t timestamp = packet->Timestamp() - timestamp_offset_;
	bool is_first_packet_of_frame = packet->is_first_packet_of_frame();
	bool is_last_packet_of_frame = packet->Marker();

	rtp_sequence_number_map_->InsertPacket(
	packet->SequenceNumber(),
	RtpSequenceNumberMap::Info(timestamp, is_first_packet_of_frame,
	is_last_packet_of_frame));
	}
	}

	// 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;

	// 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) {
	MutexLock lock(&lock_);
	UpdateRtpStats(*packet);
	media_has_been_sent_ = true;
	}
	}

	void DEPRECATED_RtpSenderEgress::ProcessBitrateAndNotifyObservers() {
	if (!bitrate_callback_)
	return;

	MutexLock lock(&lock_);
	RtpSendRates send_rates = GetSendRatesLocked();
	bitrate_callback_->Notify(
	send_rates.Sum().bps(),
	send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
	}

	RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRates() const {
	MutexLock lock(&lock_);
	return GetSendRatesLocked();
	}

	RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRatesLocked() const {
	const int64_t now_ms = clock_->TimeInMilliseconds();
	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 DEPRECATED_RtpSenderEgress::GetDataCounters(
	StreamDataCounters* rtp_stats,
	StreamDataCounters* rtx_stats) const {
	MutexLock lock(&lock_);
	*rtp_stats = rtp_stats_;
	*rtx_stats = rtx_rtp_stats_;
	}

	void DEPRECATED_RtpSenderEgress::ForceIncludeSendPacketsInAllocation(
	bool part_of_allocation) {
	MutexLock lock(&lock_);
	force_part_of_allocation_ = part_of_allocation;
	}

	bool DEPRECATED_RtpSenderEgress::MediaHasBeenSent() const {
	MutexLock lock(&lock_);
	return media_has_been_sent_;
	}

	void DEPRECATED_RtpSenderEgress::SetMediaHasBeenSent(bool media_sent) {
	MutexLock lock(&lock_);
	media_has_been_sent_ = media_sent;
	}

	void DEPRECATED_RtpSenderEgress::SetTimestampOffset(uint32_t timestamp) {
	MutexLock lock(&lock_);
	timestamp_offset_ = timestamp;
	}

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

	MutexLock lock(&lock_);
	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;
	}

	bool DEPRECATED_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 DEPRECATED_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;
	// TODO(bugs.webrtc.org/12713): Remove once downstream usage is gone.
	packet_info.ssrc = ssrc_;
	packet_info.media_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 DEPRECATED_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 DEPRECATED_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 DEPRECATED_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 DEPRECATED_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 DEPRECATED_RtpSenderEgress::UpdateRtpStats(const RtpPacketToSend& packet) {
	int64_t now_ms = clock_->TimeInMilliseconds();

	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.packet_type() == RtpPacketMediaType::kForwardErrorCorrection) {
	counters->fec.AddPacket(packet);
	}

	if (packet.packet_type() == RtpPacketMediaType::kRetransmission) {
	counters->retransmitted.AddPacket(packet);
	}
	counters->transmitted.AddPacket(packet);

	RTC_DCHECK(packet.packet_type().has_value());
	send_rates_[static_cast<size_t>(*packet.packet_type())].Update(packet.size(),
	now_ms);

	if (rtp_stats_callback_) {
	rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc());
	}
	}

	} // namespace webrtc