Adding upstream version 110.0.1.upstream/110.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 681 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller.cc b/third_party/libwebrtc/modules/pacing/pacing_controller.cc
new file mode 100644
index 0000000000..cdd908c9f8
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/pacing_controller.cc
@@ -0,0 +1,681 @@
+/*
+ *  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/pacing/pacing_controller.h"
+
+#include <algorithm>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#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"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+namespace {
+// Time limit in milliseconds between packet bursts.
+constexpr TimeDelta kDefaultMinPacketLimit = TimeDelta::Millis(5);
+constexpr TimeDelta kCongestedPacketInterval = TimeDelta::Millis(500);
+// TODO(sprang): Consider dropping this limit.
+// The maximum debt level, in terms of time, capped when sending packets.
+constexpr TimeDelta kMaxDebtInTime = TimeDelta::Millis(500);
+constexpr TimeDelta kMaxElapsedTime = TimeDelta::Seconds(2);
+constexpr TimeDelta kTargetPaddingDuration = TimeDelta::Millis(5);
+
+bool IsDisabled(const FieldTrialsView& field_trials, absl::string_view key) {
+  return absl::StartsWith(field_trials.Lookup(key), "Disabled");
+}
+
+bool IsEnabled(const FieldTrialsView& field_trials, absl::string_view key) {
+  return absl::StartsWith(field_trials.Lookup(key), "Enabled");
+}
+
+std::unique_ptr<PacingController::PacketQueue> CreatePacketQueue(
+    const FieldTrialsView& field_trials,
+    Timestamp creation_time) {
+  if (field_trials.IsDisabled("WebRTC-Pacer-UsePrioritizedPacketQueue")) {
+    return std::make_unique<RoundRobinPacketQueue>(creation_time);
+  }
+  return std::make_unique<PrioritizedPacketQueue>(creation_time);
+}
+
+}  // namespace
+
+const TimeDelta PacingController::kMaxExpectedQueueLength =
+    TimeDelta::Millis(2000);
+const float PacingController::kDefaultPaceMultiplier = 2.5f;
+const TimeDelta PacingController::kPausedProcessInterval =
+    kCongestedPacketInterval;
+const TimeDelta PacingController::kMinSleepTime = TimeDelta::Millis(1);
+const TimeDelta PacingController::kMaxEarlyProbeProcessing =
+    TimeDelta::Millis(1);
+
+PacingController::PacingController(Clock* clock,
+                                   PacketSender* packet_sender,
+                                   const FieldTrialsView& field_trials)
+    : clock_(clock),
+      packet_sender_(packet_sender),
+      field_trials_(field_trials),
+      drain_large_queues_(
+          !IsDisabled(field_trials_, "WebRTC-Pacer-DrainQueue")),
+      send_padding_if_silent_(
+          IsEnabled(field_trials_, "WebRTC-Pacer-PadInSilence")),
+      pace_audio_(IsEnabled(field_trials_, "WebRTC-Pacer-BlockAudio")),
+      ignore_transport_overhead_(
+          IsEnabled(field_trials_, "WebRTC-Pacer-IgnoreTransportOverhead")),
+      min_packet_limit_(kDefaultMinPacketLimit),
+      transport_overhead_per_packet_(DataSize::Zero()),
+      send_burst_interval_(TimeDelta::Zero()),
+      last_timestamp_(clock_->CurrentTime()),
+      paused_(false),
+      media_debt_(DataSize::Zero()),
+      padding_debt_(DataSize::Zero()),
+      pacing_rate_(DataRate::Zero()),
+      adjusted_media_rate_(DataRate::Zero()),
+      padding_rate_(DataRate::Zero()),
+      prober_(field_trials_),
+      probing_send_failure_(false),
+      last_process_time_(clock->CurrentTime()),
+      last_send_time_(last_process_time_),
+      seen_first_packet_(false),
+      packet_queue_(CreatePacketQueue(field_trials_, last_process_time_)),
+      congested_(false),
+      queue_time_limit_(kMaxExpectedQueueLength),
+      account_for_audio_(false),
+      include_overhead_(false) {
+  if (!drain_large_queues_) {
+    RTC_LOG(LS_WARNING) << "Pacer queues will not be drained,"
+                           "pushback experiment must be enabled.";
+  }
+  FieldTrialParameter<int> min_packet_limit_ms("", min_packet_limit_.ms());
+  ParseFieldTrial({&min_packet_limit_ms},
+                  field_trials_.Lookup("WebRTC-Pacer-MinPacketLimitMs"));
+  min_packet_limit_ = TimeDelta::Millis(min_packet_limit_ms.Get());
+  UpdateBudgetWithElapsedTime(min_packet_limit_);
+}
+
+PacingController::~PacingController() = default;
+
+void PacingController::CreateProbeCluster(DataRate bitrate, int cluster_id) {
+  prober_.CreateProbeCluster({.at_time = CurrentTime(),
+                              .target_data_rate = bitrate,
+                              .target_duration = TimeDelta::Millis(15),
+                              .target_probe_count = 5,
+                              .id = cluster_id});
+}
+
+void PacingController::CreateProbeClusters(
+    rtc::ArrayView<const ProbeClusterConfig> probe_cluster_configs) {
+  for (const ProbeClusterConfig probe_cluster_config : probe_cluster_configs) {
+    prober_.CreateProbeCluster(probe_cluster_config);
+  }
+}
+
+void PacingController::Pause() {
+  if (!paused_)
+    RTC_LOG(LS_INFO) << "PacedSender paused.";
+  paused_ = true;
+  packet_queue_->SetPauseState(true, CurrentTime());
+}
+
+void PacingController::Resume() {
+  if (paused_)
+    RTC_LOG(LS_INFO) << "PacedSender resumed.";
+  paused_ = false;
+  packet_queue_->SetPauseState(false, CurrentTime());
+}
+
+bool PacingController::IsPaused() const {
+  return paused_;
+}
+
+void PacingController::SetCongested(bool congested) {
+  if (congested_ && !congested) {
+    UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(CurrentTime()));
+  }
+  congested_ = congested;
+}
+
+bool PacingController::IsProbing() const {
+  return prober_.is_probing();
+}
+
+Timestamp PacingController::CurrentTime() const {
+  Timestamp time = clock_->CurrentTime();
+  if (time < last_timestamp_) {
+    RTC_LOG(LS_WARNING)
+        << "Non-monotonic clock behavior observed. Previous timestamp: "
+        << last_timestamp_.ms() << ", new timestamp: " << time.ms();
+    RTC_DCHECK_GE(time, last_timestamp_);
+    time = last_timestamp_;
+  }
+  last_timestamp_ = time;
+  return time;
+}
+
+void PacingController::SetProbingEnabled(bool enabled) {
+  RTC_CHECK(!seen_first_packet_);
+  prober_.SetEnabled(enabled);
+}
+
+void PacingController::SetPacingRates(DataRate pacing_rate,
+                                      DataRate padding_rate) {
+  static constexpr DataRate kMaxRate = DataRate::KilobitsPerSec(100'000);
+  RTC_CHECK_GT(pacing_rate, DataRate::Zero());
+  RTC_CHECK_GE(padding_rate, DataRate::Zero());
+  if (padding_rate > pacing_rate) {
+    RTC_LOG(LS_WARNING) << "Padding rate " << padding_rate.kbps()
+                        << "kbps is higher than the pacing rate "
+                        << pacing_rate.kbps() << "kbps, capping.";
+    padding_rate = pacing_rate;
+  }
+
+  if (pacing_rate > kMaxRate || padding_rate > kMaxRate) {
+    RTC_LOG(LS_WARNING) << "Very high pacing rates ( > " << kMaxRate.kbps()
+                        << " kbps) configured: pacing = " << pacing_rate.kbps()
+                        << " kbps, padding = " << padding_rate.kbps()
+                        << " kbps.";
+  }
+  pacing_rate_ = pacing_rate;
+  padding_rate_ = padding_rate;
+  MaybeUpdateMediaRateDueToLongQueue(CurrentTime());
+
+  RTC_LOG(LS_VERBOSE) << "bwe:pacer_updated pacing_kbps=" << pacing_rate_.kbps()
+                      << " padding_budget_kbps=" << padding_rate.kbps();
+}
+
+void PacingController::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) {
+  RTC_DCHECK(pacing_rate_ > DataRate::Zero())
+      << "SetPacingRate must be called before InsertPacket.";
+  RTC_CHECK(packet->packet_type());
+
+  prober_.OnIncomingPacket(DataSize::Bytes(packet->payload_size()));
+
+  const Timestamp now = CurrentTime();
+  if (packet_queue_->Empty()) {
+    // If queue is empty, we need to "fast-forward" the last process time,
+    // so that we don't use passed time as budget for sending the first new
+    // packet.
+    Timestamp target_process_time = now;
+    Timestamp next_send_time = NextSendTime();
+    if (next_send_time.IsFinite()) {
+      // There was already a valid planned send time, such as a keep-alive.
+      // Use that as last process time only if it's prior to now.
+      target_process_time = std::min(now, next_send_time);
+    }
+    UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(target_process_time));
+  }
+  packet_queue_->Push(now, std::move(packet));
+  seen_first_packet_ = true;
+
+  // Queue length has increased, check if we need to change the pacing rate.
+  MaybeUpdateMediaRateDueToLongQueue(now);
+}
+
+void PacingController::SetAccountForAudioPackets(bool account_for_audio) {
+  account_for_audio_ = account_for_audio;
+}
+
+void PacingController::SetIncludeOverhead() {
+  include_overhead_ = true;
+}
+
+void PacingController::SetTransportOverhead(DataSize overhead_per_packet) {
+  if (ignore_transport_overhead_)
+    return;
+  transport_overhead_per_packet_ = overhead_per_packet;
+}
+
+void PacingController::SetSendBurstInterval(TimeDelta burst_interval) {
+  send_burst_interval_ = burst_interval;
+}
+
+TimeDelta PacingController::ExpectedQueueTime() const {
+  RTC_DCHECK_GT(adjusted_media_rate_, DataRate::Zero());
+  return QueueSizeData() / adjusted_media_rate_;
+}
+
+size_t PacingController::QueueSizePackets() const {
+  return rtc::checked_cast<size_t>(packet_queue_->SizeInPackets());
+}
+
+const std::array<int, kNumMediaTypes>&
+PacingController::SizeInPacketsPerRtpPacketMediaType() const {
+  return packet_queue_->SizeInPacketsPerRtpPacketMediaType();
+}
+
+DataSize PacingController::QueueSizeData() const {
+  DataSize size = packet_queue_->SizeInPayloadBytes();
+  if (include_overhead_) {
+    size += static_cast<int64_t>(packet_queue_->SizeInPackets()) *
+            transport_overhead_per_packet_;
+  }
+  return size;
+}
+
+DataSize PacingController::CurrentBufferLevel() const {
+  return std::max(media_debt_, padding_debt_);
+}
+
+absl::optional<Timestamp> PacingController::FirstSentPacketTime() const {
+  return first_sent_packet_time_;
+}
+
+Timestamp PacingController::OldestPacketEnqueueTime() const {
+  return packet_queue_->OldestEnqueueTime();
+}
+
+TimeDelta PacingController::UpdateTimeAndGetElapsed(Timestamp now) {
+  // If no previous processing, or last process was "in the future" because of
+  // early probe processing, then there is no elapsed time to add budget for.
+  if (last_process_time_.IsMinusInfinity() || now < last_process_time_) {
+    return TimeDelta::Zero();
+  }
+  TimeDelta elapsed_time = now - last_process_time_;
+  last_process_time_ = now;
+  if (elapsed_time > kMaxElapsedTime) {
+    RTC_LOG(LS_WARNING) << "Elapsed time (" << elapsed_time.ms()
+                        << " ms) longer than expected, limiting to "
+                        << kMaxElapsedTime.ms();
+    elapsed_time = kMaxElapsedTime;
+  }
+  return elapsed_time;
+}
+
+bool PacingController::ShouldSendKeepalive(Timestamp now) const {
+  if (send_padding_if_silent_ || paused_ || congested_ || !seen_first_packet_) {
+    // We send a padding packet every 500 ms to ensure we won't get stuck in
+    // congested state due to no feedback being received.
+    if (now - last_send_time_ >= kCongestedPacketInterval) {
+      return true;
+    }
+  }
+  return false;
+}
+
+Timestamp PacingController::NextSendTime() const {
+  const Timestamp now = CurrentTime();
+  Timestamp next_send_time = Timestamp::PlusInfinity();
+
+  if (paused_) {
+    return last_send_time_ + kPausedProcessInterval;
+  }
+
+  // If probing is active, that always takes priority.
+  if (prober_.is_probing() && !probing_send_failure_) {
+    Timestamp probe_time = prober_.NextProbeTime(now);
+    if (!probe_time.IsPlusInfinity()) {
+      return probe_time.IsMinusInfinity() ? now : probe_time;
+    }
+  }
+
+  // Not pacing audio, if leading packet is audio its target send
+  // time is the time at which it was enqueued.
+  Timestamp unpaced_audio_time =
+      pace_audio_ ? Timestamp::PlusInfinity()
+                  : packet_queue_->LeadingAudioPacketEnqueueTime();
+  if (unpaced_audio_time.IsFinite()) {
+    return unpaced_audio_time;
+  }
+
+  if (congested_ || !seen_first_packet_) {
+    // We need to at least send keep-alive packets with some interval.
+    return last_send_time_ + kCongestedPacketInterval;
+  }
+
+  if (adjusted_media_rate_ > DataRate::Zero() && !packet_queue_->Empty()) {
+    // If packets are allowed to be sent in a burst, the
+    // debt is allowed to grow up to one packet more than what can be sent
+    // during 'send_burst_period_'.
+    TimeDelta drain_time = media_debt_ / adjusted_media_rate_;
+    next_send_time =
+        last_process_time_ +
+        ((send_burst_interval_ > drain_time) ? TimeDelta::Zero() : drain_time);
+  } else if (padding_rate_ > DataRate::Zero() && packet_queue_->Empty()) {
+    // If we _don't_ have pending packets, check how long until we have
+    // bandwidth for padding packets. Both media and padding debts must
+    // have been drained to do this.
+    RTC_DCHECK_GT(adjusted_media_rate_, DataRate::Zero());
+    TimeDelta drain_time = std::max(media_debt_ / adjusted_media_rate_,
+                                    padding_debt_ / padding_rate_);
+
+    if (drain_time.IsZero() &&
+        (!media_debt_.IsZero() || !padding_debt_.IsZero())) {
+      // We have a non-zero debt, but drain time is smaller than tick size of
+      // TimeDelta, round it up to the smallest possible non-zero delta.
+      drain_time = TimeDelta::Micros(1);
+    }
+    next_send_time = last_process_time_ + drain_time;
+  } else {
+    // Nothing to do.
+    next_send_time = last_process_time_ + kPausedProcessInterval;
+  }
+
+  if (send_padding_if_silent_) {
+    next_send_time =
+        std::min(next_send_time, last_send_time_ + kPausedProcessInterval);
+  }
+
+  return next_send_time;
+}
+
+void PacingController::ProcessPackets() {
+  const Timestamp now = CurrentTime();
+  Timestamp target_send_time = now;
+
+  if (ShouldSendKeepalive(now)) {
+    DataSize keepalive_data_sent = DataSize::Zero();
+    // We can not send padding unless a normal packet has first been sent. If
+    // we do, timestamps get messed up.
+    if (seen_first_packet_) {
+      std::vector<std::unique_ptr<RtpPacketToSend>> keepalive_packets =
+          packet_sender_->GeneratePadding(DataSize::Bytes(1));
+      for (auto& packet : keepalive_packets) {
+        keepalive_data_sent +=
+            DataSize::Bytes(packet->payload_size() + packet->padding_size());
+        packet_sender_->SendPacket(std::move(packet), PacedPacketInfo());
+        for (auto& packet : packet_sender_->FetchFec()) {
+          EnqueuePacket(std::move(packet));
+        }
+      }
+    }
+    OnPacketSent(RtpPacketMediaType::kPadding, keepalive_data_sent, now);
+  }
+
+  if (paused_) {
+    return;
+  }
+
+  TimeDelta early_execute_margin =
+      prober_.is_probing() ? kMaxEarlyProbeProcessing : TimeDelta::Zero();
+
+  target_send_time = NextSendTime();
+  if (now + early_execute_margin < target_send_time) {
+    // We are too early, but if queue is empty still allow draining some debt.
+    // Probing is allowed to be sent up to kMinSleepTime early.
+    UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(now));
+    return;
+  }
+
+  TimeDelta elapsed_time = UpdateTimeAndGetElapsed(target_send_time);
+
+  if (elapsed_time > TimeDelta::Zero()) {
+    UpdateBudgetWithElapsedTime(elapsed_time);
+  }
+
+  PacedPacketInfo pacing_info;
+  DataSize recommended_probe_size = DataSize::Zero();
+  bool is_probing = prober_.is_probing();
+  if (is_probing) {
+    // Probe timing is sensitive, and handled explicitly by BitrateProber, so
+    // use actual send time rather than target.
+    pacing_info = prober_.CurrentCluster(now).value_or(PacedPacketInfo());
+    if (pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe) {
+      recommended_probe_size = prober_.RecommendedMinProbeSize();
+      RTC_DCHECK_GT(recommended_probe_size, DataSize::Zero());
+    } else {
+      // No valid probe cluster returned, probe might have timed out.
+      is_probing = false;
+    }
+  }
+
+  DataSize data_sent = DataSize::Zero();
+  // Circuit breaker, making sure main loop isn't forever.
+  static constexpr int kMaxIterations = 1 << 16;
+  int iteration = 0;
+  int packets_sent = 0;
+  int padding_packets_generated = 0;
+  for (; iteration < kMaxIterations; ++iteration) {
+    // Fetch packet, so long as queue is not empty or budget is not
+    // exhausted.
+    std::unique_ptr<RtpPacketToSend> rtp_packet =
+        GetPendingPacket(pacing_info, target_send_time, now);
+    if (rtp_packet == nullptr) {
+      // No packet available to send, check if we should send padding.
+      DataSize padding_to_add = PaddingToAdd(recommended_probe_size, data_sent);
+      if (padding_to_add > DataSize::Zero()) {
+        std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
+            packet_sender_->GeneratePadding(padding_to_add);
+        if (!padding_packets.empty()) {
+          padding_packets_generated += padding_packets.size();
+          for (auto& packet : padding_packets) {
+            EnqueuePacket(std::move(packet));
+          }
+          // Continue loop to send the padding that was just added.
+          continue;
+        } else {
+          // Can't generate padding, still update padding budget for next send
+          // time.
+          UpdatePaddingBudgetWithSentData(padding_to_add);
+        }
+      }
+      // Can't fetch new packet and no padding to send, exit send loop.
+      break;
+    } else {
+      RTC_DCHECK(rtp_packet);
+      RTC_DCHECK(rtp_packet->packet_type().has_value());
+      const RtpPacketMediaType packet_type = *rtp_packet->packet_type();
+      DataSize packet_size = DataSize::Bytes(rtp_packet->payload_size() +
+                                             rtp_packet->padding_size());
+
+      if (include_overhead_) {
+        packet_size += DataSize::Bytes(rtp_packet->headers_size()) +
+                       transport_overhead_per_packet_;
+      }
+
+      packet_sender_->SendPacket(std::move(rtp_packet), pacing_info);
+      for (auto& packet : packet_sender_->FetchFec()) {
+        EnqueuePacket(std::move(packet));
+      }
+      data_sent += packet_size;
+      ++packets_sent;
+
+      // Send done, update send time.
+      OnPacketSent(packet_type, packet_size, now);
+
+      if (is_probing) {
+        pacing_info.probe_cluster_bytes_sent += packet_size.bytes();
+        // If we are currently probing, we need to stop the send loop when we
+        // have reached the send target.
+        if (data_sent >= recommended_probe_size) {
+          break;
+        }
+      }
+
+      // Update target send time in case that are more packets that we are late
+      // in processing.
+      target_send_time = NextSendTime();
+      if (target_send_time > now) {
+        // Exit loop if not probing.
+        if (!is_probing) {
+          break;
+        }
+        target_send_time = now;
+      }
+      UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(target_send_time));
+    }
+  }
+
+  if (iteration >= kMaxIterations) {
+    // Circuit break activated. Log warning, adjust send time and return.
+    // TODO(sprang): Consider completely clearing state.
+    RTC_LOG(LS_ERROR) << "PacingController exceeded max iterations in "
+                         "send-loop: packets sent = "
+                      << packets_sent << ", padding packets generated = "
+                      << padding_packets_generated
+                      << ", bytes sent = " << data_sent.bytes();
+    last_send_time_ = now;
+    last_process_time_ = now;
+    return;
+  }
+
+  if (is_probing) {
+    probing_send_failure_ = data_sent == DataSize::Zero();
+    if (!probing_send_failure_) {
+      prober_.ProbeSent(CurrentTime(), data_sent);
+    }
+  }
+
+  // Queue length has probably decreased, check if pacing rate needs to updated.
+  // Poll the time again, since we might have enqueued new fec/padding packets
+  // with a later timestamp than `now`.
+  MaybeUpdateMediaRateDueToLongQueue(CurrentTime());
+}
+
+DataSize PacingController::PaddingToAdd(DataSize recommended_probe_size,
+                                        DataSize data_sent) const {
+  if (!packet_queue_->Empty()) {
+    // Actual payload available, no need to add padding.
+    return DataSize::Zero();
+  }
+
+  if (congested_) {
+    // Don't add padding if congested, even if requested for probing.
+    return DataSize::Zero();
+  }
+
+  if (!seen_first_packet_) {
+    // We can not send padding unless a normal packet has first been sent. If
+    // we do, timestamps get messed up.
+    return DataSize::Zero();
+  }
+
+  if (!recommended_probe_size.IsZero()) {
+    if (recommended_probe_size > data_sent) {
+      return recommended_probe_size - data_sent;
+    }
+    return DataSize::Zero();
+  }
+
+  if (padding_rate_ > DataRate::Zero() && padding_debt_ == DataSize::Zero()) {
+    return kTargetPaddingDuration * padding_rate_;
+  }
+  return DataSize::Zero();
+}
+
+std::unique_ptr<RtpPacketToSend> PacingController::GetPendingPacket(
+    const PacedPacketInfo& pacing_info,
+    Timestamp target_send_time,
+    Timestamp now) {
+  const bool is_probe =
+      pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe;
+  // If first packet in probe, insert a small padding packet so we have a
+  // more reliable start window for the rate estimation.
+  if (is_probe && pacing_info.probe_cluster_bytes_sent == 0) {
+    auto padding = packet_sender_->GeneratePadding(DataSize::Bytes(1));
+    // If no RTP modules sending media are registered, we may not get a
+    // padding packet back.
+    if (!padding.empty()) {
+      // We should never get more than one padding packets with a requested
+      // size of 1 byte.
+      RTC_DCHECK_EQ(padding.size(), 1u);
+      return std::move(padding[0]);
+    }
+  }
+
+  if (packet_queue_->Empty()) {
+    return nullptr;
+  }
+
+  // First, check if there is any reason _not_ to send the next queued packet.
+
+  // Unpaced audio packets and probes are exempted from send checks.
+  bool unpaced_audio_packet =
+      !pace_audio_ && packet_queue_->LeadingAudioPacketEnqueueTime().IsFinite();
+  if (!unpaced_audio_packet && !is_probe) {
+    if (congested_) {
+      // Don't send anything if congested.
+      return nullptr;
+    }
+
+    if (now <= target_send_time && send_burst_interval_.IsZero()) {
+      // We allow sending slightly early if we think that we would actually
+      // had been able to, had we been right on time - i.e. the current debt
+      // is not more than would be reduced to zero at the target sent time.
+      // If we allow packets to be sent in a burst, packet are allowed to be
+      // sent early.
+      TimeDelta flush_time = media_debt_ / adjusted_media_rate_;
+      if (now + flush_time > target_send_time) {
+        return nullptr;
+      }
+    }
+  }
+
+  return packet_queue_->Pop();
+}
+
+void PacingController::OnPacketSent(RtpPacketMediaType packet_type,
+                                    DataSize packet_size,
+                                    Timestamp send_time) {
+  if (!first_sent_packet_time_ && packet_type != RtpPacketMediaType::kPadding) {
+    first_sent_packet_time_ = send_time;
+  }
+
+  bool audio_packet = packet_type == RtpPacketMediaType::kAudio;
+  if ((!audio_packet || account_for_audio_) && packet_size > DataSize::Zero()) {
+    UpdateBudgetWithSentData(packet_size);
+  }
+
+  last_send_time_ = send_time;
+}
+
+void PacingController::UpdateBudgetWithElapsedTime(TimeDelta delta) {
+  media_debt_ -= std::min(media_debt_, adjusted_media_rate_ * delta);
+  padding_debt_ -= std::min(padding_debt_, padding_rate_ * delta);
+}
+
+void PacingController::UpdateBudgetWithSentData(DataSize size) {
+  media_debt_ += size;
+  media_debt_ = std::min(media_debt_, adjusted_media_rate_ * kMaxDebtInTime);
+  UpdatePaddingBudgetWithSentData(size);
+}
+
+void PacingController::UpdatePaddingBudgetWithSentData(DataSize size) {
+  padding_debt_ += size;
+  padding_debt_ = std::min(padding_debt_, padding_rate_ * kMaxDebtInTime);
+}
+
+void PacingController::SetQueueTimeLimit(TimeDelta limit) {
+  queue_time_limit_ = limit;
+}
+
+void PacingController::MaybeUpdateMediaRateDueToLongQueue(Timestamp now) {
+  adjusted_media_rate_ = pacing_rate_;
+  if (!drain_large_queues_) {
+    return;
+  }
+
+  DataSize queue_size_data = QueueSizeData();
+  if (queue_size_data > DataSize::Zero()) {
+    // Assuming equal size packets and input/output rate, the average packet
+    // has avg_time_left_ms left to get queue_size_bytes out of the queue, if
+    // time constraint shall be met. Determine bitrate needed for that.
+    packet_queue_->UpdateAverageQueueTime(now);
+    TimeDelta avg_time_left =
+        std::max(TimeDelta::Millis(1),
+                 queue_time_limit_ - packet_queue_->AverageQueueTime());
+    DataRate min_rate_needed = queue_size_data / avg_time_left;
+    if (min_rate_needed > pacing_rate_) {
+      adjusted_media_rate_ = min_rate_needed;
+      RTC_LOG(LS_VERBOSE) << "bwe:large_pacing_queue pacing_rate_kbps="
+                          << pacing_rate_.kbps();
+    }
+  }
+}
+
+}  // namespace webrtc