Add new prioritized packet queue.

This queue is a more strict round robing queue, unlike the class
named RoundRobinPacketQueue. That is, we don't have the same logic to
prioritize lower-bitrate streams.

The queue time mechanism is essentially directly copied from the
previous implementation however.

Bug: webrtc:11340
Change-Id: Ie38ba8ce27c985f5f1e907cec068d6a365089bcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260562
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36737}

modules/pacing/BUILD.gn

@@ -23,6 +23,8 @@ rtc_library("pacing") {
     "pacing_controller.h",
     "packet_router.cc",
     "packet_router.h",
+    "prioritized_packet_queue.cc",
+    "prioritized_packet_queue.h",
     "round_robin_packet_queue.cc",
     "round_robin_packet_queue.h",
     "rtp_packet_pacer.h",
@@ -93,6 +95,7 @@ if (rtc_include_tests) {
       "paced_sender_unittest.cc",
       "pacing_controller_unittest.cc",
       "packet_router_unittest.cc",
+      "prioritized_packet_queue_unittest.cc",
       "task_queue_paced_sender_unittest.cc",
     ]
     deps = [

modules/pacing/pacing_controller.cc

@@ -18,6 +18,7 @@
 #include "absl/strings/match.h"
 #include "modules/pacing/bitrate_prober.h"
 #include "modules/pacing/interval_budget.h"
+#include "modules/pacing/prioritized_packet_queue.h"
 #include "modules/pacing/round_robin_packet_queue.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/experiments/field_trial_parser.h"
@@ -55,6 +56,15 @@ TimeDelta GetDynamicPaddingTarget(const FieldTrialsView& field_trials) {
   return padding_target.Get();
 }
 
+std::unique_ptr<PacingController::PacketQueue> CreatePacketQueue(
+    const FieldTrialsView& field_trials,
+    Timestamp creation_time) {
+  if (field_trials.IsEnabled("WebRTC-Pacer-UsePrioritizedPacketQueue")) {
+    return std::make_unique<PrioritizedPacketQueue>(creation_time);
+  }
+  return std::make_unique<RoundRobinPacketQueue>(creation_time);
+}
+
 }  // namespace
 
 const TimeDelta PacingController::kMaxExpectedQueueLength =
@@ -98,8 +108,7 @@ PacingController::PacingController(Clock* clock,
       last_process_time_(clock->CurrentTime()),
       last_send_time_(last_process_time_),
       seen_first_packet_(false),
-      packet_queue_(
-          std::make_unique<RoundRobinPacketQueue>(last_process_time_)),
+      packet_queue_(CreatePacketQueue(field_trials_, last_process_time_)),
       congested_(false),
       queue_time_limit_(kMaxExpectedQueueLength),
       account_for_audio_(false),
@@ -237,7 +246,7 @@ TimeDelta PacingController::ExpectedQueueTime() const {
 }
 
 size_t PacingController::QueueSizePackets() const {
-  return packet_queue_->SizeInPackets();
+  return rtc::checked_cast<size_t>(packet_queue_->SizeInPackets());
 }
 
 DataSize PacingController::QueueSizeData() const {

modules/pacing/pacing_controller.h

@@ -71,7 +71,7 @@ class PacingController {
                       std::unique_ptr<RtpPacketToSend> packet) = 0;
     virtual std::unique_ptr<RtpPacketToSend> Pop() = 0;
 
-    virtual size_t SizeInPackets() const = 0;
+    virtual int SizeInPackets() const = 0;
     bool Empty() const { return SizeInPackets() == 0; }
     virtual DataSize SizeInPayloadBytes() const = 0;
 
@@ -88,6 +88,7 @@ class PacingController {
     // Average queue time for the packets currently in the queue.
     // The queuing time is calculated from Push() to the last UpdateQueueTime()
     // call - with any time spent in a paused state subtracted.
+    // Returns TimeDelta::Zero() for an empty queue.
     virtual TimeDelta AverageQueueTime() const = 0;
 
     // Called during packet processing or when pause stats changes. Since the

modules/pacing/prioritized_packet_queue.cc

@@ -0,0 +1,253 @@
+/*
+ *  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 <utility>
+
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kAudioPrioLevel = 0;
+
+int GetPriorityForType(RtpPacketMediaType 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.
+      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 + 2;
+    case RtpPacketMediaType::kPadding:
+      // Packets that are in themselves likely useless, only sent to keep the
+      // BWE high.
+      return kAudioPrioLevel + 3;
+  }
+  RTC_CHECK_NOTREACHED();
+}
+
+}  // namespace
+
+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) {}
+
+bool PrioritizedPacketQueue::StreamQueue::EnqueuePacket(QueuedPacket packet,
+                                                        int priority_level) {
+  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::DequePacket(int priority_level) {
+  RTC_DCHECK(!packets_[priority_level].empty());
+  QueuedPacket packet = std::move(packets_[priority_level].front());
+  packets_[priority_level].pop_front();
+  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::LeadingAudioPacketEnqueueTime()
+    const {
+  RTC_DCHECK(!packets_[kAudioPrioLevel].empty());
+  return packets_[kAudioPrioLevel].begin()->enqueue_time;
+}
+
+Timestamp PrioritizedPacketQueue::StreamQueue::LastEnqueueTime() const {
+  return last_enqueue_time_;
+}
+
+PrioritizedPacketQueue::PrioritizedPacketQueue(Timestamp creation_time)
+    : queue_time_sum_(TimeDelta::Zero()),
+      pause_time_sum_(TimeDelta::Zero()),
+      size_packets_(0),
+      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);
+  int prio_level = GetPriorityForType(*packet->packet_type());
+  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_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.DequePacket(top_active_prio_level_);
+  --size_packets_;
+  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;
+
+  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);
+
+  // 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 if (streams_by_prio_[top_active_prio_level_].empty()) {
+    // No stream queues have packets at this prio level, find top priority
+    // that is not empty.
+    if (size_packets_ == 0) {
+      top_active_prio_level_ = -1;
+    } else {
+      for (int i = 0; i < kNumPriorityLevels; ++i) {
+        if (!streams_by_prio_[i].empty()) {
+          top_active_prio_level_ = i;
+          break;
+        }
+      }
+    }
+  }
+
+  return std::move(packet.packet);
+}
+
+int PrioritizedPacketQueue::SizeInPackets() const {
+  return size_packets_;
+}
+
+DataSize PrioritizedPacketQueue::SizeInPayloadBytes() const {
+  return size_payload_;
+}
+
+Timestamp PrioritizedPacketQueue::LeadingAudioPacketEnqueueTime() const {
+  if (streams_by_prio_[kAudioPrioLevel].empty()) {
+    return Timestamp::MinusInfinity();
+  }
+  return streams_by_prio_[kAudioPrioLevel]
+      .front()
+      ->LeadingAudioPacketEnqueueTime();
+}
+
+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;
+}
+
+}  // namespace webrtc

modules/pacing/prioritized_packet_queue.h

@@ -0,0 +1,124 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_PACING_PRIORITIZED_PACKET_QUEUE_H_
+#define MODULES_PACING_PRIORITIZED_PACKET_QUEUE_H_
+
+#include <stddef.h>
+
+#include <deque>
+#include <list>
+#include <memory>
+#include <unordered_map>
+
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/pacing/pacing_controller.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+namespace webrtc {
+
+class PrioritizedPacketQueue : public PacingController::PacketQueue {
+ public:
+  explicit PrioritizedPacketQueue(Timestamp creation_time);
+  PrioritizedPacketQueue(const PrioritizedPacketQueue&) = delete;
+  PrioritizedPacketQueue& operator=(const PrioritizedPacketQueue&) = delete;
+
+  void Push(Timestamp enqueue_time,
+            std::unique_ptr<RtpPacketToSend> packet) override;
+  std::unique_ptr<RtpPacketToSend> Pop() override;
+  int SizeInPackets() const override;
+  DataSize SizeInPayloadBytes() const override;
+  Timestamp LeadingAudioPacketEnqueueTime() const override;
+  Timestamp OldestEnqueueTime() const override;
+  TimeDelta AverageQueueTime() const override;
+  void UpdateAverageQueueTime(Timestamp now) override;
+  void SetPauseState(bool paused, Timestamp now) override;
+
+ private:
+  static constexpr int kNumPriorityLevels = 4;
+
+  class QueuedPacket {
+   public:
+    QueuedPacket(QueuedPacket&&) = default;
+    QueuedPacket& operator=(QueuedPacket&&) = default;
+
+    QueuedPacket(const QueuedPacket&) = delete;
+    QueuedPacket& operator=(const QueuedPacket&) = delete;
+
+    DataSize PacketSize() const;
+
+    std::unique_ptr<RtpPacketToSend> packet;
+    Timestamp enqueue_time;
+    std::list<Timestamp>::iterator enqueue_time_iterator;
+  };
+
+  // Class containing packets for an RTP stream.
+  // For each priority level, packets are simply stored in a fifo queue.
+  class StreamQueue {
+   public:
+    explicit StreamQueue(Timestamp creation_time);
+    StreamQueue(StreamQueue&&) = default;
+    StreamQueue& operator=(StreamQueue&&) = default;
+
+    StreamQueue(const StreamQueue&) = delete;
+    StreamQueue& operator=(const StreamQueue&) = delete;
+
+    // Enqueue packet at the given priority level. Returns true if the packet
+    // count for that priority level went from zero to non-zero.
+    bool EnqueuePacket(QueuedPacket packet, int priority_level);
+
+    QueuedPacket DequePacket(int priority_level);
+
+    bool HasPacketsAtPrio(int priority_level) const;
+    bool IsEmpty() const;
+    Timestamp LeadingAudioPacketEnqueueTime() const;
+    Timestamp LastEnqueueTime() const;
+
+   private:
+    std::deque<QueuedPacket> packets_[kNumPriorityLevels];
+    Timestamp last_enqueue_time_;
+  };
+
+  // Cumulative sum, over all packets, of time spent in the queue.
+  TimeDelta queue_time_sum_;
+  // Cumulative sum of time the queue has spent in a paused state.
+  TimeDelta pause_time_sum_;
+  // Total number of packets stored in this queue.
+  int size_packets_;
+  // Sum of payload sizes for all packts stored in this queue.
+  DataSize size_payload_;
+  // The last time queue/pause time sums were updated.
+  Timestamp last_update_time_;
+  bool paused_;
+
+  // Last time `streams_` was culled for inactive streams.
+  Timestamp last_culling_time_;
+
+  // Map from SSRC to packet queues for the associated RTP stream.
+  std::unordered_map<uint32_t, std::unique_ptr<StreamQueue>> streams_;
+
+  // For each priority level, a queue of StreamQueues which have at least one
+  // packet pending for that prio level.
+  std::deque<StreamQueue*> streams_by_prio_[kNumPriorityLevels];
+
+  // The first index into `stream_by_prio_` that is non-empty.
+  int top_active_prio_level_;
+
+  // Ordered list of enqueue times. Additions are always increasing and added to
+  // the end. QueuedPacket instances have a iterators into this list for fast
+  // removal.
+  std::list<Timestamp> enqueue_times_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_PACING_PRIORITIZED_PACKET_QUEUE_H_

modules/pacing/prioritized_packet_queue_unittest.cc

@@ -0,0 +1,233 @@
+/*
+ *  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 <utility>
+
+#include "api/units/time_delta.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "rtc_base/checks.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+constexpr uint32_t kDefaultSsrc = 123;
+constexpr int kDefaultPayloadSize = 789;
+
+std::unique_ptr<RtpPacketToSend> CreatePacket(RtpPacketMediaType type,
+                                              uint16_t sequence_number,
+                                              uint32_t ssrc = kDefaultSsrc) {
+  auto packet = std::make_unique<RtpPacketToSend>(/*extensions=*/nullptr);
+  packet->set_packet_type(type);
+  packet->SetSsrc(ssrc);
+  packet->SetSequenceNumber(sequence_number);
+  packet->SetPayloadSize(kDefaultPayloadSize);
+  return packet;
+}
+
+}  // namespace
+
+TEST(PrioritizedPacketQueue, ReturnsPacketsInPrioritizedOrder) {
+  Timestamp now = Timestamp::Zero();
+  PrioritizedPacketQueue queue(now);
+
+  // Add packets in low to high packet order.
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/1));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kForwardErrorCorrection,
+                               /*seq=*/3));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, /*seq=*/4));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/5));
+
+  // Packets should be returned in high to low order.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 5);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 4);
+  // Video and FEC prioritized equally - but video was enqueued first.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 2);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 3);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 1);
+}
+
+TEST(PrioritizedPacketQueue, ReturnsEqualPrioPacketsInRoundRobinOrder) {
+  Timestamp now = Timestamp::Zero();
+  PrioritizedPacketQueue queue(now);
+
+  // Insert video packets (prioritized equally), simulating a simulcast-type use
+  // case.
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/1, /*ssrc=*/100));
+
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2, /*ssrc=*/101));
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/3, /*ssrc=*/101));
+
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/4, /*ssrc=*/102));
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/5, /*ssrc=*/102));
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/6, /*ssrc=*/102));
+  queue.Push(now,
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/7, /*ssrc=*/102));
+
+  // First packet from each SSRC.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 1);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 2);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 4);
+
+  // Second packets from streams that have packets left.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 3);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 5);
+
+  // Only packets from last stream remaining.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 6);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 7);
+}
+
+TEST(PrioritizedPacketQueue, ReportsSizeInPackets) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.SizeInPackets(), 0);
+
+  queue.Push(/*enqueue_time=*/Timestamp::Zero(),
+             CreatePacket(RtpPacketMediaType::kVideo,
+                          /*seq_no=*/1));
+  EXPECT_EQ(queue.SizeInPackets(), 1);
+
+  queue.Pop();
+  EXPECT_EQ(queue.SizeInPackets(), 0);
+}
+
+TEST(PrioritizedPacketQueue, ReportsPayloadSize) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.SizeInPayloadBytes(), DataSize::Zero());
+
+  queue.Push(/*enqueue_time=*/Timestamp::Zero(),
+             CreatePacket(RtpPacketMediaType::kVideo,
+                          /*seq_no=*/1));
+  EXPECT_EQ(queue.SizeInPayloadBytes(), DataSize::Bytes(kDefaultPayloadSize));
+
+  queue.Pop();
+  EXPECT_EQ(queue.SizeInPayloadBytes(), DataSize::Zero());
+}
+
+TEST(PrioritizedPacketQueue, ReportsPaddingSize) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.SizeInPayloadBytes(), DataSize::Zero());
+  static constexpr DataSize kPaddingSize = DataSize::Bytes(190);
+
+  auto packet = std::make_unique<RtpPacketToSend>(/*extensions=*/nullptr);
+  packet->set_packet_type(RtpPacketMediaType::kPadding);
+  packet->SetSsrc(kDefaultSsrc);
+  packet->SetSequenceNumber(/*seq=*/1);
+  packet->SetPadding(kPaddingSize.bytes());
+  queue.Push(/*enqueue_time=*/Timestamp::Zero(), std::move(packet));
+  EXPECT_EQ(queue.SizeInPayloadBytes(), kPaddingSize);
+
+  queue.Pop();
+  EXPECT_EQ(queue.SizeInPayloadBytes(), DataSize::Zero());
+}
+
+TEST(PrioritizedPacketQueue, ReportsOldestEnqueueTime) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.OldestEnqueueTime(), Timestamp::MinusInfinity());
+
+  // Add three packets, with the middle packet having higher prio.
+  queue.Push(Timestamp::Millis(10),
+             CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/1));
+  queue.Push(Timestamp::Millis(20),
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2));
+  queue.Push(Timestamp::Millis(30),
+             CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/3));
+  EXPECT_EQ(queue.OldestEnqueueTime(), Timestamp::Millis(10));
+
+  queue.Pop();  // Pop packet with enqueue time 20.
+  EXPECT_EQ(queue.OldestEnqueueTime(), Timestamp::Millis(10));
+
+  queue.Pop();  // Pop packet with enqueue time 10.
+  EXPECT_EQ(queue.OldestEnqueueTime(), Timestamp::Millis(30));
+
+  queue.Pop();  // Pop packet with enqueue time 30, queue empty again.
+  EXPECT_EQ(queue.OldestEnqueueTime(), Timestamp::MinusInfinity());
+}
+
+TEST(PrioritizedPacketQueue, ReportsAverageQueueTime) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Zero());
+
+  // Add three packets, with the middle packet having higher prio.
+  queue.Push(Timestamp::Millis(10),
+             CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/1));
+  queue.Push(Timestamp::Millis(20),
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2));
+  queue.Push(Timestamp::Millis(30),
+             CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/3));
+
+  queue.UpdateAverageQueueTime(Timestamp::Millis(40));
+  // Packets have waited 30, 20, 10 ms -> average = 20ms.
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(20));
+
+  queue.Pop();  // Pop packet with enqueue time 20.
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(20));
+
+  queue.Pop();  // Pop packet with enqueue time 10.
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(10));
+
+  queue.Pop();  // Pop packet with enqueue time 30, queue empty again.
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Zero());
+}
+
+TEST(PrioritizedPacketQueue, SubtractsPusedTimeFromAverageQueueTime) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Zero());
+
+  // Add a packet and then enable paused state.
+  queue.Push(Timestamp::Millis(100),
+             CreatePacket(RtpPacketMediaType::kPadding, /*seq=*/1));
+  queue.SetPauseState(true, Timestamp::Millis(600));
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(500));
+
+  // Enqueue a packet 500ms into the paused state. Queue time of
+  // original packet is still seen as 500ms and new one has 0ms giving
+  // an average of 250ms.
+  queue.Push(Timestamp::Millis(1100),
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2));
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(250));
+
+  // Unpause some time later, queue time still unchanged.
+  queue.SetPauseState(false, Timestamp::Millis(1600));
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(250));
+
+  // Update queue time 500ms after pause state ended.
+  queue.UpdateAverageQueueTime(Timestamp::Millis(2100));
+  EXPECT_EQ(queue.AverageQueueTime(), TimeDelta::Millis(750));
+}
+
+TEST(PrioritizedPacketQueue, ReportsLeadingAudioEnqueueTime) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero());
+  EXPECT_EQ(queue.LeadingAudioPacketEnqueueTime(), Timestamp::MinusInfinity());
+
+  queue.Push(Timestamp::Millis(10),
+             CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/1));
+  EXPECT_EQ(queue.LeadingAudioPacketEnqueueTime(), Timestamp::MinusInfinity());
+
+  queue.Push(Timestamp::Millis(20),
+             CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/2));
+
+  EXPECT_EQ(queue.LeadingAudioPacketEnqueueTime(), Timestamp::Millis(20));
+
+  queue.Pop();  // Pop audio packet.
+  EXPECT_EQ(queue.LeadingAudioPacketEnqueueTime(), Timestamp::MinusInfinity());
+}
+
+}  // namespace webrtc

modules/pacing/round_robin_packet_queue.cc

@@ -182,7 +182,7 @@ std::unique_ptr<RtpPacketToSend> RoundRobinPacketQueue::Pop() {
     return rtp_packet;
   }
 
-  RTC_DCHECK_GT(size_packets_, 0u);
+  RTC_DCHECK_GT(size_packets_, 0);
   Stream* stream = GetHighestPriorityStream();
   const QueuedPacket& queued_packet = stream->packet_queue.top();
 
@@ -231,7 +231,7 @@ std::unique_ptr<RtpPacketToSend> RoundRobinPacketQueue::Pop() {
   return rtp_packet;
 }
 
-size_t RoundRobinPacketQueue::SizeInPackets() const {
+int RoundRobinPacketQueue::SizeInPackets() const {
   return size_packets_;
 }
 
@@ -280,7 +280,7 @@ void RoundRobinPacketQueue::UpdateAverageQueueTime(Timestamp now) {
   if (paused_) {
     pause_time_sum_ += delta;
   } else {
-    queue_time_sum_ += TimeDelta::Micros(delta.us() * size_packets_);
+    queue_time_sum_ += delta * size_packets_;
   }
 
   time_last_updated_ = now;

modules/pacing/round_robin_packet_queue.h

@@ -28,7 +28,6 @@
 #include "modules/pacing/pacing_controller.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
-#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 
@@ -41,7 +40,7 @@ class RoundRobinPacketQueue : public PacingController::PacketQueue {
             std::unique_ptr<RtpPacketToSend> packet) override;
   std::unique_ptr<RtpPacketToSend> Pop() override;
 
-  size_t SizeInPackets() const override;
+  int SizeInPackets() const override;
   DataSize SizeInPayloadBytes() const override;
   Timestamp LeadingAudioPacketEnqueueTime() const override;
   Timestamp OldestEnqueueTime() const override;
@@ -142,7 +141,7 @@ class RoundRobinPacketQueue : public PacingController::PacketQueue {
   int64_t enqueue_count_;
 
   bool paused_;
-  size_t size_packets_;
+  int size_packets_;
   DataSize size_;
   DataSize max_size_;
   TimeDelta queue_time_sum_;