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/units/timestamp.h"
 #include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
 #include "rtc_base/logging.h"

 namespace webrtc {
 namespace {
 constexpr uint32_t kTimestampTicksPerMs = 90;
 constexpr TimeDelta kBitrateStatisticsWindow = TimeDelta::Seconds(1);
 constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
 constexpr TimeDelta kUpdateInterval = kBitrateStatisticsWindow;
 }  // namespace

 RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
     TaskQueueBase& worker_queue,
     RtpSenderEgress* sender,
     PacketSequencer* sequencer)
     : worker_queue_(worker_queue),
       transport_sequence_number_(0),
       sender_(sender),
       sequencer_(sequencer) {
   RTC_DCHECK(sequencer);
 }
 RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() {
   RTC_DCHECK_RUN_ON(&worker_queue_);
 }

 void RtpSenderEgress::NonPacedPacketSender::EnqueuePackets(
     std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
   if (!worker_queue_.IsCurrent()) {
     worker_queue_.PostTask(SafeTask(
         task_safety_.flag(), [this, packets = std::move(packets)]() mutable {
           EnqueuePackets(std::move(packets));
         }));
     return;
   }
   RTC_DCHECK_RUN_ON(&worker_queue_);
   for (auto& packet : packets) {
     PrepareForSend(packet.get());
     sender_->SendPacket(std::move(packet), PacedPacketInfo());
   }
   auto fec_packets = sender_->FetchFecPackets();
   if (!fec_packets.empty()) {
     EnqueuePackets(std::move(fec_packets));
   }
 }

 void RtpSenderEgress::NonPacedPacketSender::PrepareForSend(
     RtpPacketToSend* packet) {
   RTC_DCHECK_RUN_ON(&worker_queue_);
   // Assign sequence numbers, but not for flexfec which is already running on
   // an internally maintained sequence number series.
   if (packet->Ssrc() != sender_->FlexFecSsrc()) {
     sequencer_->Sequence(*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)
     : enable_send_packet_batching_(config.enable_send_packet_batching),
       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),
       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),
       fec_generator_(config.fec_generator),
       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),
       send_rates_(kNumMediaTypes, BitrateTracker(kBitrateStatisticsWindow)),
       rtp_sequence_number_map_(need_rtp_packet_infos_
                                    ? std::make_unique<RtpSequenceNumberMap>(
                                          kRtpSequenceNumberMapMaxEntries)
                                    : nullptr) {
   RTC_DCHECK(worker_queue_);
   RTC_DCHECK(config.transport_feedback_callback == nullptr)
       << "transport_feedback_callback is no longer used and will soon be "
          "deleted.";
   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(std::unique_ptr<RtpPacketToSend> packet,
                                  const PacedPacketInfo& pacing_info) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   RTC_DCHECK(packet);

   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 Timestamp now = clock_->CurrentTime();
 #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
   BweTestLoggingPlot(now, packet->Ssrc());
 #endif
   if (need_rtp_packet_infos_ &&
       packet->packet_type() == RtpPacketToSend::Type::kVideo) {
     // 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(
         packet->SequenceNumber(),
         RtpSequenceNumberMap::Info(
             timestamp, packet->is_first_packet_of_frame(), packet->Marker()));
   }

   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;
     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.
   if (packet->HasExtension<TransmissionOffset>() &&
       packet->capture_time() > Timestamp::Zero()) {
     TimeDelta diff = now - packet->capture_time();
     packet->SetExtension<TransmissionOffset>(kTimestampTicksPerMs * diff.ms());
   }
   if (packet->HasExtension<AbsoluteSendTime>()) {
     packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::To24Bits(now));
   }
   if (packet->HasExtension<TransportSequenceNumber>() &&
       packet->transport_sequence_number()) {
     packet->SetExtension<TransportSequenceNumber>(
         *packet->transport_sequence_number() & 0xFFFF);
   }

   if (packet->HasExtension<VideoTimingExtension>()) {
     if (populate_network2_timestamp_) {
       packet->set_network2_time(now);
     } else {
       packet->set_pacer_exit_time(now);
     }
   }

   auto compound_packet = Packet{std::move(packet), pacing_info, now};
   if (enable_send_packet_batching_ && !is_audio_) {
     packets_to_send_.push_back(std::move(compound_packet));
   } else {
     CompleteSendPacket(compound_packet, false);
   }
 }

 void RtpSenderEgress::OnBatchComplete() {
   RTC_DCHECK_RUN_ON(worker_queue_);
   for (auto& packet : packets_to_send_) {
     CompleteSendPacket(packet, &packet == &packets_to_send_.back());
   }
   packets_to_send_.clear();
 }

 void RtpSenderEgress::CompleteSendPacket(const Packet& compound_packet,
                                          bool last_in_batch) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   auto& [packet, pacing_info, now] = compound_packet;
   RTC_CHECK(packet);
   const bool is_media = packet->packet_type() == RtpPacketMediaType::kAudio ||
                         packet->packet_type() == RtpPacketMediaType::kVideo;

   PacketOptions options;
   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;

   // Set Packet id from transport sequence number header extension if it is
   // used. The source of the header extension is
   // RtpPacketToSend::transport_sequence_number(), but the extension is only 16
   // bit and will wrap. We should be able to use the 64bit value as id, but in
   // order to not change behaviour we use the 16bit extension value if it is
   // used.
   absl::optional<uint16_t> packet_id =
       packet->GetExtension<TransportSequenceNumber>();
   if (packet_id.has_value()) {
     options.packet_id = *packet_id;
     options.included_in_feedback = true;
     options.included_in_allocation = true;
   } else if (packet->transport_sequence_number()) {
     options.packet_id = *packet->transport_sequence_number();
   }

   if (packet->packet_type() != RtpPacketMediaType::kPadding &&
       packet->packet_type() != RtpPacketMediaType::kRetransmission &&
       send_packet_observer_ != nullptr && packet->capture_time().IsFinite()) {
     send_packet_observer_->OnSendPacket(packet_id, packet->capture_time(),
                                         packet->Ssrc());
   }
   options.batchable = enable_send_packet_batching_ && !is_audio_;
   options.last_packet_in_batch = last_in_batch;
   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);
   } 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);
     UpdateRtpStats(now, packet->Ssrc(), packet_type, std::move(counter),
                    packet->size());
   }
 }

 RtpSendRates RtpSenderEgress::GetSendRates(Timestamp now) const {
   RTC_DCHECK_RUN_ON(worker_queue_);
   RtpSendRates current_rates;
   for (size_t i = 0; i < kNumMediaTypes; ++i) {
     RtpPacketMediaType type = static_cast<RtpPacketMediaType>(i);
     current_rates[type] = send_rates_[i].Rate(now).value_or(DataRate::Zero());
   }
   return current_rates;
 }

 void RtpSenderEgress::GetDataCounters(StreamDataCounters* rtp_stats,
                                       StreamDataCounters* rtx_stats) const {
   RTC_DCHECK_RUN_ON(worker_queue_);
   *rtp_stats = rtp_stats_;
   *rtx_stats = rtx_rtp_stats_;
 }

 void RtpSenderEgress::ForceIncludeSendPacketsInAllocation(
     bool part_of_allocation) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   force_part_of_allocation_ = part_of_allocation;
 }

 bool RtpSenderEgress::MediaHasBeenSent() const {
   RTC_DCHECK_RUN_ON(worker_queue_);
   return media_has_been_sent_;
 }

 void RtpSenderEgress::SetMediaHasBeenSent(bool media_sent) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   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) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   pending_fec_params_.emplace(delta_params, key_params);
 }

 std::vector<std::unique_ptr<RtpPacketToSend>>
 RtpSenderEgress::FetchFecPackets() {
   RTC_DCHECK_RUN_ON(worker_queue_);
   if (fec_generator_) {
     return fec_generator_->GetFecPackets();
   }
   return {};
 }

 void RtpSenderEgress::OnAbortedRetransmissions(
     rtc::ArrayView<const uint16_t> sequence_numbers) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   // Mark aborted retransmissions as sent, rather than leaving them in
   // a 'pending' state - otherwise they can not be requested again and
   // will not be cleared until the history has reached its max size.
   for (uint16_t seq_no : sequence_numbers) {
     packet_history_->MarkPacketAsSent(seq_no);
   }
 }

 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;
 }

 bool RtpSenderEgress::SendPacketToNetwork(const RtpPacketToSend& packet,
                                           const PacketOptions& options,
                                           const PacedPacketInfo& pacing_info) {
   RTC_DCHECK_RUN_ON(worker_queue_);
   int bytes_sent = -1;
   if (transport_) {
     bytes_sent = transport_->SendRtp(packet, 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(Timestamp now,
                                      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;

   StreamDataCounters* counters =
       packet_ssrc == rtx_ssrc_ ? &rtx_rtp_stats_ : &rtp_stats_;

   counters->MaybeSetFirstPacketTime(now);

   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);
     if (bitrate_callback_) {
     send_rates = GetSendRates(now);
     }

     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(clock_->CurrentTime());
   bitrate_callback_->Notify(
       send_rates.Sum().bps(),
       send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
 }

 #if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
 void RtpSenderEgress::BweTestLoggingPlot(Timestamp now, uint32_t packet_ssrc) {
   RTC_DCHECK_RUN_ON(worker_queue_);

   const auto rates = GetSendRates(now);
   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/units/timestamp.h"
	#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
	#include "rtc_base/logging.h"

	namespace webrtc {
	namespace {
	constexpr uint32_t kTimestampTicksPerMs = 90;
	constexpr TimeDelta kBitrateStatisticsWindow = TimeDelta::Seconds(1);
	constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
	constexpr TimeDelta kUpdateInterval = kBitrateStatisticsWindow;
	} // namespace

	RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
	TaskQueueBase& worker_queue,
	RtpSenderEgress* sender,
	PacketSequencer* sequencer)
	: worker_queue_(worker_queue),
	transport_sequence_number_(0),
	sender_(sender),
	sequencer_(sequencer) {
	RTC_DCHECK(sequencer);
	}
	RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() {
	RTC_DCHECK_RUN_ON(&worker_queue_);
	}

	void RtpSenderEgress::NonPacedPacketSender::EnqueuePackets(
	std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
	if (!worker_queue_.IsCurrent()) {
	worker_queue_.PostTask(SafeTask(
	task_safety_.flag(), [this, packets = std::move(packets)]() mutable {
	EnqueuePackets(std::move(packets));
	}));
	return;
	}
	RTC_DCHECK_RUN_ON(&worker_queue_);
	for (auto& packet : packets) {
	PrepareForSend(packet.get());
	sender_->SendPacket(std::move(packet), PacedPacketInfo());
	}
	auto fec_packets = sender_->FetchFecPackets();
	if (!fec_packets.empty()) {
	EnqueuePackets(std::move(fec_packets));
	}
	}

	void RtpSenderEgress::NonPacedPacketSender::PrepareForSend(
	RtpPacketToSend* packet) {
	RTC_DCHECK_RUN_ON(&worker_queue_);
	// Assign sequence numbers, but not for flexfec which is already running on
	// an internally maintained sequence number series.
	if (packet->Ssrc() != sender_->FlexFecSsrc()) {
	sequencer_->Sequence(*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)
	: enable_send_packet_batching_(config.enable_send_packet_batching),
	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),
	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),
	fec_generator_(config.fec_generator),
	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),
	send_rates_(kNumMediaTypes, BitrateTracker(kBitrateStatisticsWindow)),
	rtp_sequence_number_map_(need_rtp_packet_infos_
	? std::make_unique<RtpSequenceNumberMap>(
	kRtpSequenceNumberMapMaxEntries)
	: nullptr) {
	RTC_DCHECK(worker_queue_);
	RTC_DCHECK(config.transport_feedback_callback == nullptr)
	<< "transport_feedback_callback is no longer used and will soon be "
	"deleted.";
	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(std::unique_ptr<RtpPacketToSend> packet,
	const PacedPacketInfo& pacing_info) {
	RTC_DCHECK_RUN_ON(worker_queue_);
	RTC_DCHECK(packet);

	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 Timestamp now = clock_->CurrentTime();
	#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
	BweTestLoggingPlot(now, packet->Ssrc());
	#endif
	if (need_rtp_packet_infos_ &&
	packet->packet_type() == RtpPacketToSend::Type::kVideo) {
	// 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(
	packet->SequenceNumber(),
	RtpSequenceNumberMap::Info(
	timestamp, packet->is_first_packet_of_frame(), packet->Marker()));
	}

	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;
	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.
	if (packet->HasExtension<TransmissionOffset>() &&
	packet->capture_time() > Timestamp::Zero()) {
	TimeDelta diff = now - packet->capture_time();
	packet->SetExtension<TransmissionOffset>(kTimestampTicksPerMs * diff.ms());
	}
	if (packet->HasExtension<AbsoluteSendTime>()) {
	packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::To24Bits(now));
	}
	if (packet->HasExtension<TransportSequenceNumber>() &&
	packet->transport_sequence_number()) {
	packet->SetExtension<TransportSequenceNumber>(
	*packet->transport_sequence_number() & 0xFFFF);
	}

	if (packet->HasExtension<VideoTimingExtension>()) {
	if (populate_network2_timestamp_) {
	packet->set_network2_time(now);
	} else {
	packet->set_pacer_exit_time(now);
	}
	}

	auto compound_packet = Packet{std::move(packet), pacing_info, now};
	if (enable_send_packet_batching_ && !is_audio_) {
	packets_to_send_.push_back(std::move(compound_packet));
	} else {
	CompleteSendPacket(compound_packet, false);
	}
	}

	void RtpSenderEgress::OnBatchComplete() {
	RTC_DCHECK_RUN_ON(worker_queue_);
	for (auto& packet : packets_to_send_) {
	CompleteSendPacket(packet, &packet == &packets_to_send_.back());
	}
	packets_to_send_.clear();
	}

	void RtpSenderEgress::CompleteSendPacket(const Packet& compound_packet,
	bool last_in_batch) {
	RTC_DCHECK_RUN_ON(worker_queue_);
	auto& [packet, pacing_info, now] = compound_packet;
	RTC_CHECK(packet);
	const bool is_media = packet->packet_type() == RtpPacketMediaType::kAudio \|\|
	packet->packet_type() == RtpPacketMediaType::kVideo;

	PacketOptions options;
	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;

	// Set Packet id from transport sequence number header extension if it is
	// used. The source of the header extension is
	// RtpPacketToSend::transport_sequence_number(), but the extension is only 16
	// bit and will wrap. We should be able to use the 64bit value as id, but in
	// order to not change behaviour we use the 16bit extension value if it is
	// used.
	absl::optional<uint16_t> packet_id =
	packet->GetExtension<TransportSequenceNumber>();
	if (packet_id.has_value()) {
	options.packet_id = *packet_id;
	options.included_in_feedback = true;
	options.included_in_allocation = true;
	} else if (packet->transport_sequence_number()) {
	options.packet_id = *packet->transport_sequence_number();
	}

	if (packet->packet_type() != RtpPacketMediaType::kPadding &&
	packet->packet_type() != RtpPacketMediaType::kRetransmission &&
	send_packet_observer_ != nullptr && packet->capture_time().IsFinite()) {
	send_packet_observer_->OnSendPacket(packet_id, packet->capture_time(),
	packet->Ssrc());
	}
	options.batchable = enable_send_packet_batching_ && !is_audio_;
	options.last_packet_in_batch = last_in_batch;
	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);
	} 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);
	UpdateRtpStats(now, packet->Ssrc(), packet_type, std::move(counter),
	packet->size());
	}
	}

	RtpSendRates RtpSenderEgress::GetSendRates(Timestamp now) const {
	RTC_DCHECK_RUN_ON(worker_queue_);
	RtpSendRates current_rates;
	for (size_t i = 0; i < kNumMediaTypes; ++i) {
	RtpPacketMediaType type = static_cast<RtpPacketMediaType>(i);
	current_rates[type] = send_rates_[i].Rate(now).value_or(DataRate::Zero());
	}
	return current_rates;
	}

	void RtpSenderEgress::GetDataCounters(StreamDataCounters* rtp_stats,
	StreamDataCounters* rtx_stats) const {
	RTC_DCHECK_RUN_ON(worker_queue_);
	*rtp_stats = rtp_stats_;
	*rtx_stats = rtx_rtp_stats_;
	}

	void RtpSenderEgress::ForceIncludeSendPacketsInAllocation(
	bool part_of_allocation) {
	RTC_DCHECK_RUN_ON(worker_queue_);
	force_part_of_allocation_ = part_of_allocation;
	}

	bool RtpSenderEgress::MediaHasBeenSent() const {
	RTC_DCHECK_RUN_ON(worker_queue_);
	return media_has_been_sent_;
	}

	void RtpSenderEgress::SetMediaHasBeenSent(bool media_sent) {
	RTC_DCHECK_RUN_ON(worker_queue_);
	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) {
	RTC_DCHECK_RUN_ON(worker_queue_);
	pending_fec_params_.emplace(delta_params, key_params);
	}

	std::vector<std::unique_ptr<RtpPacketToSend>>
	RtpSenderEgress::FetchFecPackets() {
	RTC_DCHECK_RUN_ON(worker_queue_);
	if (fec_generator_) {
	return fec_generator_->GetFecPackets();
	}
	return {};
	}

	void RtpSenderEgress::OnAbortedRetransmissions(
	rtc::ArrayView<const uint16_t> sequence_numbers) {
	RTC_DCHECK_RUN_ON(worker_queue_);
	// Mark aborted retransmissions as sent, rather than leaving them in
	// a 'pending' state - otherwise they can not be requested again and
	// will not be cleared until the history has reached its max size.
	for (uint16_t seq_no : sequence_numbers) {
	packet_history_->MarkPacketAsSent(seq_no);
	}
	}

	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;
	}

	bool RtpSenderEgress::SendPacketToNetwork(const RtpPacketToSend& packet,
	const PacketOptions& options,
	const PacedPacketInfo& pacing_info) {
	RTC_DCHECK_RUN_ON(worker_queue_);
	int bytes_sent = -1;
	if (transport_) {
	bytes_sent = transport_->SendRtp(packet, 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(Timestamp now,
	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;

	StreamDataCounters* counters =
	packet_ssrc == rtx_ssrc_ ? &rtx_rtp_stats_ : &rtp_stats_;

	counters->MaybeSetFirstPacketTime(now);

	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);
	if (bitrate_callback_) {
	send_rates = GetSendRates(now);
	}

	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(clock_->CurrentTime());
	bitrate_callback_->Notify(
	send_rates.Sum().bps(),
	send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
	}

	#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
	void RtpSenderEgress::BweTestLoggingPlot(Timestamp now, uint32_t packet_ssrc) {
	RTC_DCHECK_RUN_ON(worker_queue_);

	const auto rates = GetSendRates(now);
	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