modules/pacing/prioritized_packet_queue.cc - src.git - Git at Google

 /*
  *  Copyright (c) 2022 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/pacing/prioritized_packet_queue.h"

 #include <algorithm>
 #include <array>
 #include <utility>

 #include "absl/container/inlined_vector.h"
 #include "absl/types/optional.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"

 namespace webrtc {
 namespace {

 constexpr int kAudioPrioLevel = 0;

 int GetPriorityForType(
     RtpPacketMediaType type,
     absl::optional<RtpPacketToSend::OriginalType> original_type) {
   // Lower number takes priority over higher.
   switch (type) {
     case RtpPacketMediaType::kAudio:
       // Audio is always prioritized over other packet types.
       return kAudioPrioLevel;
     case RtpPacketMediaType::kRetransmission:
       // Send retransmissions before new media. If original_type is set, audio
       // retransmission is prioritized more than video retransmission.
       if (original_type == RtpPacketToSend::OriginalType::kVideo) {
         return kAudioPrioLevel + 2;
       }
       return kAudioPrioLevel + 1;
     case RtpPacketMediaType::kVideo:
     case RtpPacketMediaType::kForwardErrorCorrection:
       // Video has "normal" priority, in the old speak.
       // Send redundancy concurrently to video. If it is delayed it might have a
       // lower chance of being useful.
       return kAudioPrioLevel + 3;
     case RtpPacketMediaType::kPadding:
       // Packets that are in themselves likely useless, only sent to keep the
       // BWE high.
       return kAudioPrioLevel + 4;
   }
   RTC_CHECK_NOTREACHED();
 }

 }  // namespace

 absl::InlinedVector<TimeDelta, PrioritizedPacketQueue::kNumPriorityLevels>
 PrioritizedPacketQueue::ToTtlPerPrio(PacketQueueTTL packet_queue_ttl) {
   absl::InlinedVector<TimeDelta, PrioritizedPacketQueue::kNumPriorityLevels>
       ttl_per_prio(kNumPriorityLevels, TimeDelta::PlusInfinity());
   ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kRetransmission,
                                   RtpPacketToSend::OriginalType::kAudio)] =
       packet_queue_ttl.audio_retransmission;
   ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kRetransmission,
                                   RtpPacketToSend::OriginalType::kVideo)] =
       packet_queue_ttl.video_retransmission;
   ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kVideo, absl::nullopt)] =
       packet_queue_ttl.video;
   return ttl_per_prio;
 }

 DataSize PrioritizedPacketQueue::QueuedPacket::PacketSize() const {
   return DataSize::Bytes(packet->payload_size() + packet->padding_size());
 }

 PrioritizedPacketQueue::StreamQueue::StreamQueue(Timestamp creation_time)
     : last_enqueue_time_(creation_time), num_keyframe_packets_(0) {}

 bool PrioritizedPacketQueue::StreamQueue::EnqueuePacket(QueuedPacket packet,
                                                         int priority_level) {
   if (packet.packet->is_key_frame()) {
     ++num_keyframe_packets_;
   }
   bool first_packet_at_level = packets_[priority_level].empty();
   packets_[priority_level].push_back(std::move(packet));
   return first_packet_at_level;
 }

 PrioritizedPacketQueue::QueuedPacket
 PrioritizedPacketQueue::StreamQueue::DequeuePacket(int priority_level) {
   RTC_DCHECK(!packets_[priority_level].empty());
   QueuedPacket packet = std::move(packets_[priority_level].front());
   packets_[priority_level].pop_front();
   if (packet.packet->is_key_frame()) {
     RTC_DCHECK_GT(num_keyframe_packets_, 0);
     --num_keyframe_packets_;
   }
   return packet;
 }

 bool PrioritizedPacketQueue::StreamQueue::HasPacketsAtPrio(
     int priority_level) const {
   return !packets_[priority_level].empty();
 }

 bool PrioritizedPacketQueue::StreamQueue::IsEmpty() const {
   for (const std::deque<QueuedPacket>& queue : packets_) {
     if (!queue.empty()) {
       return false;
     }
   }
   return true;
 }

 Timestamp PrioritizedPacketQueue::StreamQueue::LeadingPacketEnqueueTime(
     int priority_level) const {
   RTC_DCHECK(!packets_[priority_level].empty());
   return packets_[priority_level].begin()->enqueue_time;
 }

 Timestamp PrioritizedPacketQueue::StreamQueue::LastEnqueueTime() const {
   return last_enqueue_time_;
 }

 std::array<std::deque<PrioritizedPacketQueue::QueuedPacket>,
            PrioritizedPacketQueue::kNumPriorityLevels>
 PrioritizedPacketQueue::StreamQueue::DequeueAll() {
   std::array<std::deque<QueuedPacket>, kNumPriorityLevels> packets_by_prio;
   for (int i = 0; i < kNumPriorityLevels; ++i) {
     packets_by_prio[i].swap(packets_[i]);
   }
   num_keyframe_packets_ = 0;
   return packets_by_prio;
 }

 PrioritizedPacketQueue::PrioritizedPacketQueue(
     Timestamp creation_time,
     bool prioritize_audio_retransmission,
     PacketQueueTTL packet_queue_ttl)
     : prioritize_audio_retransmission_(prioritize_audio_retransmission),
       time_to_live_per_prio_(ToTtlPerPrio(packet_queue_ttl)),
       queue_time_sum_(TimeDelta::Zero()),
       pause_time_sum_(TimeDelta::Zero()),
       size_packets_(0),
       size_packets_per_media_type_({}),
       size_payload_(DataSize::Zero()),
       last_update_time_(creation_time),
       paused_(false),
       last_culling_time_(creation_time),
       top_active_prio_level_(-1) {}

 void PrioritizedPacketQueue::Push(Timestamp enqueue_time,
                                   std::unique_ptr<RtpPacketToSend> packet) {
   StreamQueue* stream_queue;
   auto [it, inserted] = streams_.emplace(packet->Ssrc(), nullptr);
   if (inserted) {
     it->second = std::make_unique<StreamQueue>(enqueue_time);
   }
   stream_queue = it->second.get();

   auto enqueue_time_iterator =
       enqueue_times_.insert(enqueue_times_.end(), enqueue_time);
   RTC_DCHECK(packet->packet_type().has_value());
   RtpPacketMediaType packet_type = packet->packet_type().value();
   int prio_level =
       GetPriorityForType(packet_type, prioritize_audio_retransmission_
                                           ? packet->original_packet_type()
                                           : absl::nullopt);
   PurgeOldPacketsAtPriorityLevel(prio_level, enqueue_time);
   RTC_DCHECK_GE(prio_level, 0);
   RTC_DCHECK_LT(prio_level, kNumPriorityLevels);
   QueuedPacket queued_packed = {.packet = std::move(packet),
                                 .enqueue_time = enqueue_time,
                                 .enqueue_time_iterator = enqueue_time_iterator};
   // In order to figure out how much time a packet has spent in the queue
   // while not in a paused state, we subtract the total amount of time the
   // queue has been paused so far, and when the packet is popped we subtract
   // the total amount of time the queue has been paused at that moment. This
   // way we subtract the total amount of time the packet has spent in the
   // queue while in a paused state.
   UpdateAverageQueueTime(enqueue_time);
   queued_packed.enqueue_time -= pause_time_sum_;
   ++size_packets_;
   ++size_packets_per_media_type_[static_cast<size_t>(packet_type)];
   size_payload_ += queued_packed.PacketSize();

   if (stream_queue->EnqueuePacket(std::move(queued_packed), prio_level)) {
     // Number packets at `prio_level` for this steam is now non-zero.
     streams_by_prio_[prio_level].push_back(stream_queue);
   }
   if (top_active_prio_level_ < 0 || prio_level < top_active_prio_level_) {
     top_active_prio_level_ = prio_level;
   }

   static constexpr TimeDelta kTimeout = TimeDelta::Millis(500);
   if (enqueue_time - last_culling_time_ > kTimeout) {
     for (auto it = streams_.begin(); it != streams_.end();) {
       if (it->second->IsEmpty() &&
           it->second->LastEnqueueTime() + kTimeout < enqueue_time) {
         streams_.erase(it++);
       } else {
         ++it;
       }
     }
     last_culling_time_ = enqueue_time;
   }
 }

 std::unique_ptr<RtpPacketToSend> PrioritizedPacketQueue::Pop() {
   if (size_packets_ == 0) {
     return nullptr;
   }

   RTC_DCHECK_GE(top_active_prio_level_, 0);
   StreamQueue& stream_queue = *streams_by_prio_[top_active_prio_level_].front();
   QueuedPacket packet = stream_queue.DequeuePacket(top_active_prio_level_);
   DequeuePacketInternal(packet);

   // Remove StreamQueue from head of fifo-queue for this prio level, and
   // and add it to the end if it still has packets.
   streams_by_prio_[top_active_prio_level_].pop_front();
   if (stream_queue.HasPacketsAtPrio(top_active_prio_level_)) {
     streams_by_prio_[top_active_prio_level_].push_back(&stream_queue);
   } else {
     MaybeUpdateTopPrioLevel();
   }

   return std::move(packet.packet);
 }

 int PrioritizedPacketQueue::SizeInPackets() const {
   return size_packets_;
 }

 DataSize PrioritizedPacketQueue::SizeInPayloadBytes() const {
   return size_payload_;
 }

 bool PrioritizedPacketQueue::Empty() const {
   return size_packets_ == 0;
 }

 const std::array<int, kNumMediaTypes>&
 PrioritizedPacketQueue::SizeInPacketsPerRtpPacketMediaType() const {
   return size_packets_per_media_type_;
 }

 Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTime(
     RtpPacketMediaType type) const {
   RTC_DCHECK(type != RtpPacketMediaType::kRetransmission);
   const int priority_level = GetPriorityForType(type, absl::nullopt);
   if (streams_by_prio_[priority_level].empty()) {
     return Timestamp::MinusInfinity();
   }
   return streams_by_prio_[priority_level].front()->LeadingPacketEnqueueTime(
       priority_level);
 }

 Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTimeForRetransmission()
     const {
   if (!prioritize_audio_retransmission_) {
     const int priority_level =
         GetPriorityForType(RtpPacketMediaType::kRetransmission, absl::nullopt);
     if (streams_by_prio_[priority_level].empty()) {
       return Timestamp::PlusInfinity();
     }
     return streams_by_prio_[priority_level].front()->LeadingPacketEnqueueTime(
         priority_level);
   }
   const int audio_priority_level =
       GetPriorityForType(RtpPacketMediaType::kRetransmission,
                          RtpPacketToSend::OriginalType::kAudio);
   const int video_priority_level =
       GetPriorityForType(RtpPacketMediaType::kRetransmission,
                          RtpPacketToSend::OriginalType::kVideo);

   Timestamp next_audio =
       streams_by_prio_[audio_priority_level].empty()
           ? Timestamp::PlusInfinity()
           : streams_by_prio_[audio_priority_level]
                 .front()
                 ->LeadingPacketEnqueueTime(audio_priority_level);
   Timestamp next_video =
       streams_by_prio_[video_priority_level].empty()
           ? Timestamp::PlusInfinity()
           : streams_by_prio_[video_priority_level]
                 .front()
                 ->LeadingPacketEnqueueTime(video_priority_level);
   return std::min(next_audio, next_video);
 }

 Timestamp PrioritizedPacketQueue::OldestEnqueueTime() const {
   return enqueue_times_.empty() ? Timestamp::MinusInfinity()
                                 : enqueue_times_.front();
 }

 TimeDelta PrioritizedPacketQueue::AverageQueueTime() const {
   if (size_packets_ == 0) {
     return TimeDelta::Zero();
   }
   return queue_time_sum_ / size_packets_;
 }

 void PrioritizedPacketQueue::UpdateAverageQueueTime(Timestamp now) {
   RTC_CHECK_GE(now, last_update_time_);
   if (now == last_update_time_) {
     return;
   }

   TimeDelta delta = now - last_update_time_;

   if (paused_) {
     pause_time_sum_ += delta;
   } else {
     queue_time_sum_ += delta * size_packets_;
   }

   last_update_time_ = now;
 }

 void PrioritizedPacketQueue::SetPauseState(bool paused, Timestamp now) {
   UpdateAverageQueueTime(now);
   paused_ = paused;
 }

 void PrioritizedPacketQueue::RemovePacketsForSsrc(uint32_t ssrc) {
   auto kv = streams_.find(ssrc);
   if (kv != streams_.end()) {
     // Dequeue all packets from the queue for this SSRC.
     StreamQueue& queue = *kv->second;
     std::array<std::deque<QueuedPacket>, kNumPriorityLevels> packets_by_prio =
         queue.DequeueAll();
     for (int i = 0; i < kNumPriorityLevels; ++i) {
       std::deque<QueuedPacket>& packet_queue = packets_by_prio[i];
       if (packet_queue.empty()) {
         continue;
       }

       // First erase all packets at this prio level.
       while (!packet_queue.empty()) {
         QueuedPacket packet = std::move(packet_queue.front());
         packet_queue.pop_front();
         DequeuePacketInternal(packet);
       }

       // Next, deregister this `StreamQueue` from the round-robin tables.
       RTC_DCHECK(!streams_by_prio_[i].empty());
       if (streams_by_prio_[i].size() == 1) {
         // This is the last and only queue that had packets for this prio level.
         // Update the global top prio level if neccessary.
         RTC_DCHECK(streams_by_prio_[i].front() == &queue);
         streams_by_prio_[i].pop_front();
       } else {
         // More than stream had packets at this prio level, filter this one out.
         std::deque<StreamQueue*> filtered_queue;
         for (StreamQueue* queue_ptr : streams_by_prio_[i]) {
           if (queue_ptr != &queue) {
             filtered_queue.push_back(queue_ptr);
           }
         }
         streams_by_prio_[i].swap(filtered_queue);
       }
     }
   }
   MaybeUpdateTopPrioLevel();
 }

 bool PrioritizedPacketQueue::HasKeyframePackets(uint32_t ssrc) const {
   auto it = streams_.find(ssrc);
   if (it != streams_.end()) {
     return it->second->has_keyframe_packets();
   }
   return false;
 }

 void PrioritizedPacketQueue::DequeuePacketInternal(QueuedPacket& packet) {
   --size_packets_;
   RTC_DCHECK(packet.packet->packet_type().has_value());
   RtpPacketMediaType packet_type = packet.packet->packet_type().value();
   --size_packets_per_media_type_[static_cast<size_t>(packet_type)];
   RTC_DCHECK_GE(size_packets_per_media_type_[static_cast<size_t>(packet_type)],
                 0);
   size_payload_ -= packet.PacketSize();

   // Calculate the total amount of time spent by this packet in the queue
   // while in a non-paused state. Note that the `pause_time_sum_ms_` was
   // subtracted from `packet.enqueue_time_ms` when the packet was pushed, and
   // by subtracting it now we effectively remove the time spent in in the
   // queue while in a paused state.
   TimeDelta time_in_non_paused_state =
       last_update_time_ - packet.enqueue_time - pause_time_sum_;
   queue_time_sum_ -= time_in_non_paused_state;

   // Set the time spent in the send queue, which is the per-packet equivalent of
   // totalPacketSendDelay. The notion of being paused is an implementation
   // detail that we do not want to expose, so it makes sense to report the
   // metric excluding the pause time. This also avoids spikes in the metric.
   // https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
   packet.packet->set_time_in_send_queue(time_in_non_paused_state);

   RTC_DCHECK(size_packets_ > 0 || queue_time_sum_ == TimeDelta::Zero());

   RTC_CHECK(packet.enqueue_time_iterator != enqueue_times_.end());
   enqueue_times_.erase(packet.enqueue_time_iterator);
 }

 void PrioritizedPacketQueue::MaybeUpdateTopPrioLevel() {
   if (top_active_prio_level_ != -1 &&
       !streams_by_prio_[top_active_prio_level_].empty()) {
     return;
   }
   // No stream queues have packets at top_active_prio_level_, find top priority
   // that is not empty.
   for (int i = 0; i < kNumPriorityLevels; ++i) {
     PurgeOldPacketsAtPriorityLevel(i, last_update_time_);
     if (!streams_by_prio_[i].empty()) {
       top_active_prio_level_ = i;
       break;
     }
   }
   if (size_packets_ == 0) {
     // There are no packets left to send. Last packet may have been purged. Prio
     // will change when a new packet is pushed.
     top_active_prio_level_ = -1;
   }
 }

 void PrioritizedPacketQueue::PurgeOldPacketsAtPriorityLevel(int prio_level,
                                                             Timestamp now) {
   RTC_DCHECK(prio_level >= 0 && prio_level < kNumPriorityLevels);
   TimeDelta time_to_live = time_to_live_per_prio_[prio_level];
   if (time_to_live.IsInfinite()) {
     return;
   }

   std::deque<StreamQueue*>& queues = streams_by_prio_[prio_level];
   auto iter = queues.begin();
   while (iter != queues.end()) {
     StreamQueue* queue_ptr = *iter;
     while (queue_ptr->HasPacketsAtPrio(prio_level) &&
            (now - queue_ptr->LeadingPacketEnqueueTime(prio_level)) >
                time_to_live) {
       QueuedPacket packet = queue_ptr->DequeuePacket(prio_level);
       RTC_LOG(LS_INFO) << "Dropping old packet on SSRC: "
                        << packet.packet->Ssrc()
                        << " seq:" << packet.packet->SequenceNumber()
                        << " time in queue:" << (now - packet.enqueue_time).ms()
                        << " ms";
       DequeuePacketInternal(packet);
     }
     if (!queue_ptr->HasPacketsAtPrio(prio_level)) {
       iter = queues.erase(iter);
     } else {
       ++iter;
     }
   }
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2022 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/pacing/prioritized_packet_queue.h"

	#include <algorithm>
	#include <array>
	#include <utility>

	#include "absl/container/inlined_vector.h"
	#include "absl/types/optional.h"
	#include "api/units/time_delta.h"
	#include "api/units/timestamp.h"
	#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
	#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
	#include "rtc_base/checks.h"
	#include "rtc_base/logging.h"

	namespace webrtc {
	namespace {

	constexpr int kAudioPrioLevel = 0;

	int GetPriorityForType(
	RtpPacketMediaType type,
	absl::optional<RtpPacketToSend::OriginalType> original_type) {
	// Lower number takes priority over higher.
	switch (type) {
	case RtpPacketMediaType::kAudio:
	// Audio is always prioritized over other packet types.
	return kAudioPrioLevel;
	case RtpPacketMediaType::kRetransmission:
	// Send retransmissions before new media. If original_type is set, audio
	// retransmission is prioritized more than video retransmission.
	if (original_type == RtpPacketToSend::OriginalType::kVideo) {
	return kAudioPrioLevel + 2;
	}
	return kAudioPrioLevel + 1;
	case RtpPacketMediaType::kVideo:
	case RtpPacketMediaType::kForwardErrorCorrection:
	// Video has "normal" priority, in the old speak.
	// Send redundancy concurrently to video. If it is delayed it might have a
	// lower chance of being useful.
	return kAudioPrioLevel + 3;
	case RtpPacketMediaType::kPadding:
	// Packets that are in themselves likely useless, only sent to keep the
	// BWE high.
	return kAudioPrioLevel + 4;
	}
	RTC_CHECK_NOTREACHED();
	}

	} // namespace

	absl::InlinedVector<TimeDelta, PrioritizedPacketQueue::kNumPriorityLevels>
	PrioritizedPacketQueue::ToTtlPerPrio(PacketQueueTTL packet_queue_ttl) {
	absl::InlinedVector<TimeDelta, PrioritizedPacketQueue::kNumPriorityLevels>
	ttl_per_prio(kNumPriorityLevels, TimeDelta::PlusInfinity());
	ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kRetransmission,
	RtpPacketToSend::OriginalType::kAudio)] =
	packet_queue_ttl.audio_retransmission;
	ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kRetransmission,
	RtpPacketToSend::OriginalType::kVideo)] =
	packet_queue_ttl.video_retransmission;
	ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kVideo, absl::nullopt)] =
	packet_queue_ttl.video;
	return ttl_per_prio;
	}

	DataSize PrioritizedPacketQueue::QueuedPacket::PacketSize() const {
	return DataSize::Bytes(packet->payload_size() + packet->padding_size());
	}

	PrioritizedPacketQueue::StreamQueue::StreamQueue(Timestamp creation_time)
	: last_enqueue_time_(creation_time), num_keyframe_packets_(0) {}

	bool PrioritizedPacketQueue::StreamQueue::EnqueuePacket(QueuedPacket packet,
	int priority_level) {
	if (packet.packet->is_key_frame()) {
	++num_keyframe_packets_;
	}
	bool first_packet_at_level = packets_[priority_level].empty();
	packets_[priority_level].push_back(std::move(packet));
	return first_packet_at_level;
	}

	PrioritizedPacketQueue::QueuedPacket
	PrioritizedPacketQueue::StreamQueue::DequeuePacket(int priority_level) {
	RTC_DCHECK(!packets_[priority_level].empty());
	QueuedPacket packet = std::move(packets_[priority_level].front());
	packets_[priority_level].pop_front();
	if (packet.packet->is_key_frame()) {
	RTC_DCHECK_GT(num_keyframe_packets_, 0);
	--num_keyframe_packets_;
	}
	return packet;
	}

	bool PrioritizedPacketQueue::StreamQueue::HasPacketsAtPrio(
	int priority_level) const {
	return !packets_[priority_level].empty();
	}

	bool PrioritizedPacketQueue::StreamQueue::IsEmpty() const {
	for (const std::deque<QueuedPacket>& queue : packets_) {
	if (!queue.empty()) {
	return false;
	}
	}
	return true;
	}

	Timestamp PrioritizedPacketQueue::StreamQueue::LeadingPacketEnqueueTime(
	int priority_level) const {
	RTC_DCHECK(!packets_[priority_level].empty());
	return packets_[priority_level].begin()->enqueue_time;
	}

	Timestamp PrioritizedPacketQueue::StreamQueue::LastEnqueueTime() const {
	return last_enqueue_time_;
	}

	std::array<std::deque<PrioritizedPacketQueue::QueuedPacket>,
	PrioritizedPacketQueue::kNumPriorityLevels>
	PrioritizedPacketQueue::StreamQueue::DequeueAll() {
	std::array<std::deque<QueuedPacket>, kNumPriorityLevels> packets_by_prio;
	for (int i = 0; i < kNumPriorityLevels; ++i) {
	packets_by_prio[i].swap(packets_[i]);
	}
	num_keyframe_packets_ = 0;
	return packets_by_prio;
	}

	PrioritizedPacketQueue::PrioritizedPacketQueue(
	Timestamp creation_time,
	bool prioritize_audio_retransmission,
	PacketQueueTTL packet_queue_ttl)
	: prioritize_audio_retransmission_(prioritize_audio_retransmission),
	time_to_live_per_prio_(ToTtlPerPrio(packet_queue_ttl)),
	queue_time_sum_(TimeDelta::Zero()),
	pause_time_sum_(TimeDelta::Zero()),
	size_packets_(0),
	size_packets_per_media_type_({}),
	size_payload_(DataSize::Zero()),
	last_update_time_(creation_time),
	paused_(false),
	last_culling_time_(creation_time),
	top_active_prio_level_(-1) {}

	void PrioritizedPacketQueue::Push(Timestamp enqueue_time,
	std::unique_ptr<RtpPacketToSend> packet) {
	StreamQueue* stream_queue;
	auto [it, inserted] = streams_.emplace(packet->Ssrc(), nullptr);
	if (inserted) {
	it->second = std::make_unique<StreamQueue>(enqueue_time);
	}
	stream_queue = it->second.get();

	auto enqueue_time_iterator =
	enqueue_times_.insert(enqueue_times_.end(), enqueue_time);
	RTC_DCHECK(packet->packet_type().has_value());
	RtpPacketMediaType packet_type = packet->packet_type().value();
	int prio_level =
	GetPriorityForType(packet_type, prioritize_audio_retransmission_
	? packet->original_packet_type()
	: absl::nullopt);
	PurgeOldPacketsAtPriorityLevel(prio_level, enqueue_time);
	RTC_DCHECK_GE(prio_level, 0);
	RTC_DCHECK_LT(prio_level, kNumPriorityLevels);
	QueuedPacket queued_packed = {.packet = std::move(packet),
	.enqueue_time = enqueue_time,
	.enqueue_time_iterator = enqueue_time_iterator};
	// In order to figure out how much time a packet has spent in the queue
	// while not in a paused state, we subtract the total amount of time the
	// queue has been paused so far, and when the packet is popped we subtract
	// the total amount of time the queue has been paused at that moment. This
	// way we subtract the total amount of time the packet has spent in the
	// queue while in a paused state.
	UpdateAverageQueueTime(enqueue_time);
	queued_packed.enqueue_time -= pause_time_sum_;
	++size_packets_;
	++size_packets_per_media_type_[static_cast<size_t>(packet_type)];
	size_payload_ += queued_packed.PacketSize();

	if (stream_queue->EnqueuePacket(std::move(queued_packed), prio_level)) {
	// Number packets at `prio_level` for this steam is now non-zero.
	streams_by_prio_[prio_level].push_back(stream_queue);
	}
	if (top_active_prio_level_ < 0 \|\| prio_level < top_active_prio_level_) {
	top_active_prio_level_ = prio_level;
	}

	static constexpr TimeDelta kTimeout = TimeDelta::Millis(500);
	if (enqueue_time - last_culling_time_ > kTimeout) {
	for (auto it = streams_.begin(); it != streams_.end();) {
	if (it->second->IsEmpty() &&
	it->second->LastEnqueueTime() + kTimeout < enqueue_time) {
	streams_.erase(it++);
	} else {
	++it;
	}
	}
	last_culling_time_ = enqueue_time;
	}
	}

	std::unique_ptr<RtpPacketToSend> PrioritizedPacketQueue::Pop() {
	if (size_packets_ == 0) {
	return nullptr;
	}

	RTC_DCHECK_GE(top_active_prio_level_, 0);
	StreamQueue& stream_queue = *streams_by_prio_[top_active_prio_level_].front();
	QueuedPacket packet = stream_queue.DequeuePacket(top_active_prio_level_);
	DequeuePacketInternal(packet);

	// Remove StreamQueue from head of fifo-queue for this prio level, and
	// and add it to the end if it still has packets.
	streams_by_prio_[top_active_prio_level_].pop_front();
	if (stream_queue.HasPacketsAtPrio(top_active_prio_level_)) {
	streams_by_prio_[top_active_prio_level_].push_back(&stream_queue);
	} else {
	MaybeUpdateTopPrioLevel();
	}

	return std::move(packet.packet);
	}

	int PrioritizedPacketQueue::SizeInPackets() const {
	return size_packets_;
	}

	DataSize PrioritizedPacketQueue::SizeInPayloadBytes() const {
	return size_payload_;
	}

	bool PrioritizedPacketQueue::Empty() const {
	return size_packets_ == 0;
	}

	const std::array<int, kNumMediaTypes>&
	PrioritizedPacketQueue::SizeInPacketsPerRtpPacketMediaType() const {
	return size_packets_per_media_type_;
	}

	Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTime(
	RtpPacketMediaType type) const {
	RTC_DCHECK(type != RtpPacketMediaType::kRetransmission);
	const int priority_level = GetPriorityForType(type, absl::nullopt);
	if (streams_by_prio_[priority_level].empty()) {
	return Timestamp::MinusInfinity();
	}
	return streams_by_prio_[priority_level].front()->LeadingPacketEnqueueTime(
	priority_level);
	}

	Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTimeForRetransmission()
	const {
	if (!prioritize_audio_retransmission_) {
	const int priority_level =
	GetPriorityForType(RtpPacketMediaType::kRetransmission, absl::nullopt);
	if (streams_by_prio_[priority_level].empty()) {
	return Timestamp::PlusInfinity();
	}
	return streams_by_prio_[priority_level].front()->LeadingPacketEnqueueTime(
	priority_level);
	}
	const int audio_priority_level =
	GetPriorityForType(RtpPacketMediaType::kRetransmission,
	RtpPacketToSend::OriginalType::kAudio);
	const int video_priority_level =
	GetPriorityForType(RtpPacketMediaType::kRetransmission,
	RtpPacketToSend::OriginalType::kVideo);

	Timestamp next_audio =
	streams_by_prio_[audio_priority_level].empty()
	? Timestamp::PlusInfinity()
	: streams_by_prio_[audio_priority_level]
	.front()
	->LeadingPacketEnqueueTime(audio_priority_level);
	Timestamp next_video =
	streams_by_prio_[video_priority_level].empty()
	? Timestamp::PlusInfinity()
	: streams_by_prio_[video_priority_level]
	.front()
	->LeadingPacketEnqueueTime(video_priority_level);
	return std::min(next_audio, next_video);
	}

	Timestamp PrioritizedPacketQueue::OldestEnqueueTime() const {
	return enqueue_times_.empty() ? Timestamp::MinusInfinity()
	: enqueue_times_.front();
	}

	TimeDelta PrioritizedPacketQueue::AverageQueueTime() const {
	if (size_packets_ == 0) {
	return TimeDelta::Zero();
	}
	return queue_time_sum_ / size_packets_;
	}

	void PrioritizedPacketQueue::UpdateAverageQueueTime(Timestamp now) {
	RTC_CHECK_GE(now, last_update_time_);
	if (now == last_update_time_) {
	return;
	}

	TimeDelta delta = now - last_update_time_;

	if (paused_) {
	pause_time_sum_ += delta;
	} else {
	queue_time_sum_ += delta * size_packets_;
	}

	last_update_time_ = now;
	}

	void PrioritizedPacketQueue::SetPauseState(bool paused, Timestamp now) {
	UpdateAverageQueueTime(now);
	paused_ = paused;
	}

	void PrioritizedPacketQueue::RemovePacketsForSsrc(uint32_t ssrc) {
	auto kv = streams_.find(ssrc);
	if (kv != streams_.end()) {
	// Dequeue all packets from the queue for this SSRC.
	StreamQueue& queue = *kv->second;
	std::array<std::deque<QueuedPacket>, kNumPriorityLevels> packets_by_prio =
	queue.DequeueAll();
	for (int i = 0; i < kNumPriorityLevels; ++i) {
	std::deque<QueuedPacket>& packet_queue = packets_by_prio[i];
	if (packet_queue.empty()) {
	continue;
	}

	// First erase all packets at this prio level.
	while (!packet_queue.empty()) {
	QueuedPacket packet = std::move(packet_queue.front());
	packet_queue.pop_front();
	DequeuePacketInternal(packet);
	}

	// Next, deregister this `StreamQueue` from the round-robin tables.
	RTC_DCHECK(!streams_by_prio_[i].empty());
	if (streams_by_prio_[i].size() == 1) {
	// This is the last and only queue that had packets for this prio level.
	// Update the global top prio level if neccessary.
	RTC_DCHECK(streams_by_prio_[i].front() == &queue);
	streams_by_prio_[i].pop_front();
	} else {
	// More than stream had packets at this prio level, filter this one out.
	std::deque<StreamQueue*> filtered_queue;
	for (StreamQueue* queue_ptr : streams_by_prio_[i]) {
	if (queue_ptr != &queue) {
	filtered_queue.push_back(queue_ptr);
	}
	}
	streams_by_prio_[i].swap(filtered_queue);
	}
	}
	}
	MaybeUpdateTopPrioLevel();
	}

	bool PrioritizedPacketQueue::HasKeyframePackets(uint32_t ssrc) const {
	auto it = streams_.find(ssrc);
	if (it != streams_.end()) {
	return it->second->has_keyframe_packets();
	}
	return false;
	}

	void PrioritizedPacketQueue::DequeuePacketInternal(QueuedPacket& packet) {
	--size_packets_;
	RTC_DCHECK(packet.packet->packet_type().has_value());
	RtpPacketMediaType packet_type = packet.packet->packet_type().value();
	--size_packets_per_media_type_[static_cast<size_t>(packet_type)];
	RTC_DCHECK_GE(size_packets_per_media_type_[static_cast<size_t>(packet_type)],
	0);
	size_payload_ -= packet.PacketSize();

	// Calculate the total amount of time spent by this packet in the queue
	// while in a non-paused state. Note that the `pause_time_sum_ms_` was
	// subtracted from `packet.enqueue_time_ms` when the packet was pushed, and
	// by subtracting it now we effectively remove the time spent in in the
	// queue while in a paused state.
	TimeDelta time_in_non_paused_state =
	last_update_time_ - packet.enqueue_time - pause_time_sum_;
	queue_time_sum_ -= time_in_non_paused_state;

	// Set the time spent in the send queue, which is the per-packet equivalent of
	// totalPacketSendDelay. The notion of being paused is an implementation
	// detail that we do not want to expose, so it makes sense to report the
	// metric excluding the pause time. This also avoids spikes in the metric.
	// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
	packet.packet->set_time_in_send_queue(time_in_non_paused_state);

	RTC_DCHECK(size_packets_ > 0 \|\| queue_time_sum_ == TimeDelta::Zero());

	RTC_CHECK(packet.enqueue_time_iterator != enqueue_times_.end());
	enqueue_times_.erase(packet.enqueue_time_iterator);
	}

	void PrioritizedPacketQueue::MaybeUpdateTopPrioLevel() {
	if (top_active_prio_level_ != -1 &&
	!streams_by_prio_[top_active_prio_level_].empty()) {
	return;
	}
	// No stream queues have packets at top_active_prio_level_, find top priority
	// that is not empty.
	for (int i = 0; i < kNumPriorityLevels; ++i) {
	PurgeOldPacketsAtPriorityLevel(i, last_update_time_);
	if (!streams_by_prio_[i].empty()) {
	top_active_prio_level_ = i;
	break;
	}
	}
	if (size_packets_ == 0) {
	// There are no packets left to send. Last packet may have been purged. Prio
	// will change when a new packet is pushed.
	top_active_prio_level_ = -1;
	}
	}

	void PrioritizedPacketQueue::PurgeOldPacketsAtPriorityLevel(int prio_level,
	Timestamp now) {
	RTC_DCHECK(prio_level >= 0 && prio_level < kNumPriorityLevels);
	TimeDelta time_to_live = time_to_live_per_prio_[prio_level];
	if (time_to_live.IsInfinite()) {
	return;
	}

	std::deque<StreamQueue*>& queues = streams_by_prio_[prio_level];
	auto iter = queues.begin();
	while (iter != queues.end()) {
	StreamQueue* queue_ptr = *iter;
	while (queue_ptr->HasPacketsAtPrio(prio_level) &&
	(now - queue_ptr->LeadingPacketEnqueueTime(prio_level)) >
	time_to_live) {
	QueuedPacket packet = queue_ptr->DequeuePacket(prio_level);
	RTC_LOG(LS_INFO) << "Dropping old packet on SSRC: "
	<< packet.packet->Ssrc()
	<< " seq:" << packet.packet->SequenceNumber()
	<< " time in queue:" << (now - packet.enqueue_time).ms()
	<< " ms";
	DequeuePacketInternal(packet);
	}
	if (!queue_ptr->HasPacketsAtPrio(prio_level)) {
	iter = queues.erase(iter);
	} else {
	++iter;
	}
	}
	}

	} // namespace webrtc