modules/pacing/pacing_controller.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/pacing/pacing_controller.h"

 #include <algorithm>
 #include <utility>
 #include <vector>

 #include "absl/memory/memory.h"
 #include "modules/pacing/bitrate_prober.h"
 #include "modules/pacing/interval_budget.h"
 #include "modules/utility/include/process_thread.h"
 #include "rtc_base/checks.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>();
 constexpr TimeDelta kMaxElapsedTime = TimeDelta::Seconds<2>();

 // Upper cap on process interval, in case process has not been called in a long
 // time.
 constexpr TimeDelta kMaxProcessingInterval = TimeDelta::Millis<30>();

 bool IsDisabled(const WebRtcKeyValueConfig& field_trials,
                 absl::string_view key) {
   return field_trials.Lookup(key).find("Disabled") == 0;
 }

 bool IsEnabled(const WebRtcKeyValueConfig& field_trials,
                absl::string_view key) {
   return field_trials.Lookup(key).find("Enabled") == 0;
 }

 int GetPriorityForType(RtpPacketToSend::Type type) {
   switch (type) {
     case RtpPacketToSend::Type::kAudio:
       // Audio is always prioritized over other packet types.
       return 0;
     case RtpPacketToSend::Type::kRetransmission:
       // Send retransmissions before new media.
       return 1;
     case RtpPacketToSend::Type::kVideo:
       // Video has "normal" priority, in the old speak.
       return 2;
     case RtpPacketToSend::Type::kForwardErrorCorrection:
       // Send redundancy concurrently to video. If it is delayed it might have a
       // lower chance of being useful.
       return 2;
     case RtpPacketToSend::Type::kPadding:
       // Packets that are in themselves likely useless, only sent to keep the
       // BWE high.
       return 3;
   }
 }

 }  // namespace

 const TimeDelta PacingController::kMaxExpectedQueueLength =
     TimeDelta::Millis<2000>();
 const float PacingController::kDefaultPaceMultiplier = 2.5f;
 const TimeDelta PacingController::kPausedProcessInterval =
     kCongestedPacketInterval;

 PacingController::PacingController(Clock* clock,
                                    PacketSender* packet_sender,
                                    RtcEventLog* event_log,
                                    const WebRtcKeyValueConfig* field_trials)
     : clock_(clock),
       packet_sender_(packet_sender),
       fallback_field_trials_(
           !field_trials ? absl::make_unique<FieldTrialBasedConfig>() : nullptr),
       field_trials_(field_trials ? field_trials : fallback_field_trials_.get()),
       drain_large_queues_(
           !IsDisabled(*field_trials_, "WebRTC-Pacer-DrainQueue")),
       send_padding_if_silent_(
           IsEnabled(*field_trials_, "WebRTC-Pacer-PadInSilence")),
       pace_audio_(!IsDisabled(*field_trials_, "WebRTC-Pacer-BlockAudio")),
       min_packet_limit_(kDefaultMinPacketLimit),
       last_timestamp_(clock_->CurrentTime()),
       paused_(false),
       media_budget_(0),
       padding_budget_(0),
       prober_(*field_trials_),
       probing_send_failure_(false),
       padding_failure_state_(false),
       pacing_bitrate_(DataRate::Zero()),
       time_last_process_(clock->CurrentTime()),
       last_send_time_(time_last_process_),
       packet_queue_(time_last_process_, field_trials),
       packet_counter_(0),
       congestion_window_size_(DataSize::PlusInfinity()),
       outstanding_data_(DataSize::Zero()),
       queue_time_limit(kMaxExpectedQueueLength),
       account_for_audio_(false),
       legacy_packet_referencing_(
           IsEnabled(*field_trials_, "WebRTC-Pacer-LegacyPacketReferencing")) {
   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::ms(min_packet_limit_ms.Get());
   UpdateBudgetWithElapsedTime(min_packet_limit_);
 }

 PacingController::~PacingController() = default;

 void PacingController::CreateProbeCluster(DataRate bitrate, int cluster_id) {
   prober_.CreateProbeCluster(bitrate.bps(), CurrentTime().ms(), cluster_id);
 }

 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::SetCongestionWindow(DataSize congestion_window_size) {
   congestion_window_size_ = congestion_window_size;
 }

 void PacingController::UpdateOutstandingData(DataSize outstanding_data) {
   outstanding_data_ = outstanding_data;
 }

 bool PacingController::Congested() const {
   if (congestion_window_size_.IsFinite()) {
     return outstanding_data_ >= congestion_window_size_;
   }
   return false;
 }

 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_EQ(0, packet_counter_);
   prober_.SetEnabled(enabled);
 }

 void PacingController::SetPacingRates(DataRate pacing_rate,
                                       DataRate padding_rate) {
   RTC_DCHECK_GT(pacing_rate, DataRate::Zero());
   pacing_bitrate_ = pacing_rate;
   padding_budget_.set_target_rate_kbps(padding_rate.kbps());

   RTC_LOG(LS_VERBOSE) << "bwe:pacer_updated pacing_kbps="
                       << pacing_bitrate_.kbps()
                       << " padding_budget_kbps=" << padding_rate.kbps();
 }

 void PacingController::InsertPacket(RtpPacketSender::Priority priority,
                                     uint32_t ssrc,
                                     uint16_t sequence_number,
                                     int64_t capture_time_ms,
                                     size_t bytes,
                                     bool retransmission) {
   RTC_DCHECK(pacing_bitrate_ > DataRate::Zero())
       << "SetPacingRate must be called before InsertPacket.";

   Timestamp now = CurrentTime();
   prober_.OnIncomingPacket(bytes);

   if (capture_time_ms < 0)
     capture_time_ms = now.ms();

   RtpPacketToSend::Type type;
   switch (priority) {
     case RtpPacketSender::kHighPriority:
       type = RtpPacketToSend::Type::kAudio;
       break;
     case RtpPacketSender::kNormalPriority:
       type = RtpPacketToSend::Type::kRetransmission;
       break;
     default:
       type = RtpPacketToSend::Type::kVideo;
   }
   packet_queue_.Push(GetPriorityForType(type), type, ssrc, sequence_number,
                      capture_time_ms, now, DataSize::bytes(bytes),
                      retransmission, packet_counter_++);
 }

 void PacingController::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) {
   RTC_DCHECK(pacing_bitrate_ > DataRate::Zero())
       << "SetPacingRate must be called before InsertPacket.";

   Timestamp now = CurrentTime();
   prober_.OnIncomingPacket(packet->payload_size());

   if (packet->capture_time_ms() < 0) {
     packet->set_capture_time_ms(now.ms());
   }

   RTC_CHECK(packet->packet_type());
   int priority = GetPriorityForType(*packet->packet_type());
   packet_queue_.Push(priority, now, packet_counter_++, std::move(packet));
 }

 void PacingController::SetAccountForAudioPackets(bool account_for_audio) {
   account_for_audio_ = account_for_audio;
 }

 TimeDelta PacingController::ExpectedQueueTime() const {
   RTC_DCHECK_GT(pacing_bitrate_, DataRate::Zero());
   return TimeDelta::ms(
       (QueueSizeData().bytes() * 8 * rtc::kNumMillisecsPerSec) /
       pacing_bitrate_.bps());
 }

 size_t PacingController::QueueSizePackets() const {
   return packet_queue_.SizeInPackets();
 }

 DataSize PacingController::QueueSizeData() const {
   return packet_queue_.Size();
 }

 absl::optional<Timestamp> PacingController::FirstSentPacketTime() const {
   return first_sent_packet_time_;
 }

 TimeDelta PacingController::OldestPacketWaitTime() const {
   Timestamp oldest_packet = packet_queue_.OldestEnqueueTime();
   if (oldest_packet.IsInfinite()) {
     return TimeDelta::Zero();
   }

   return CurrentTime() - oldest_packet;
 }

 TimeDelta PacingController::UpdateTimeAndGetElapsed(Timestamp now) {
   TimeDelta elapsed_time = now - time_last_process_;
   time_last_process_ = 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()) {
     // We send a padding packet every 500 ms to ensure we won't get stuck in
     // congested state due to no feedback being received.
     TimeDelta elapsed_since_last_send = now - last_send_time_;
     if (elapsed_since_last_send >= kCongestedPacketInterval) {
       // We can not send padding unless a normal packet has first been sent. If
       // we do, timestamps get messed up.
       if (packet_counter_ > 0) {
         return true;
       }
     }
   }
   return false;
 }

 absl::optional<TimeDelta> PacingController::TimeUntilNextProbe() {
   if (!prober_.IsProbing()) {
     return absl::nullopt;
   }

   TimeDelta time_delta =
       TimeDelta::ms(prober_.TimeUntilNextProbe(CurrentTime().ms()));
   if (time_delta > TimeDelta::Zero() ||
       (time_delta == TimeDelta::Zero() && !probing_send_failure_)) {
     return time_delta;
   }

   return absl::nullopt;
 }

 TimeDelta PacingController::TimeElapsedSinceLastProcess() const {
   return CurrentTime() - time_last_process_;
 }

 void PacingController::ProcessPackets() {
   Timestamp now = CurrentTime();
   TimeDelta elapsed_time = UpdateTimeAndGetElapsed(now);
   if (ShouldSendKeepalive(now)) {
     if (legacy_packet_referencing_) {
       OnPaddingSent(packet_sender_->TimeToSendPadding(DataSize::bytes(1),
                                                       PacedPacketInfo()));
     } else {
       DataSize keepalive_data_sent = DataSize::Zero();
       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_->SendRtpPacket(std::move(packet), PacedPacketInfo());
       }
       OnPaddingSent(keepalive_data_sent);
     }
   }

   if (paused_)
     return;

   if (elapsed_time > TimeDelta::Zero()) {
     DataRate target_rate = pacing_bitrate_;
     DataSize queue_size_data = packet_queue_.Size();
     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_.UpdateQueueTime(CurrentTime());
       if (drain_large_queues_) {
         TimeDelta avg_time_left =
             std::max(TimeDelta::ms(1),
                      queue_time_limit - packet_queue_.AverageQueueTime());
         DataRate min_rate_needed = queue_size_data / avg_time_left;
         if (min_rate_needed > target_rate) {
           target_rate = min_rate_needed;
           RTC_LOG(LS_VERBOSE) << "bwe:large_pacing_queue pacing_rate_kbps="
                               << target_rate.kbps();
         }
       }
     }

     media_budget_.set_target_rate_kbps(target_rate.kbps());
     UpdateBudgetWithElapsedTime(elapsed_time);
   }

   bool is_probing = prober_.IsProbing();
   PacedPacketInfo pacing_info;
   absl::optional<DataSize> recommended_probe_size;
   if (is_probing) {
     pacing_info = prober_.CurrentCluster();
     recommended_probe_size = DataSize::bytes(prober_.RecommendedMinProbeSize());
   }

   DataSize data_sent = DataSize::Zero();
   // The paused state is checked in the loop since it leaves the critical
   // section allowing the paused state to be changed from other code.
   while (!paused_) {
     auto* packet = GetPendingPacket(pacing_info);
     if (packet == nullptr) {
       // No packet available to send, check if we should send padding.
       if (!legacy_packet_referencing_) {
         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()) {
             // No padding packets were generated, quite send loop.
             break;
           }
           for (auto& packet : padding_packets) {
             EnqueuePacket(std::move(packet));
           }
           // Continue loop to send the padding that was just added.
           continue;
         }
       }

       // Can't fetch new packet and no padding to send, exit send loop.
       break;
     }

     std::unique_ptr<RtpPacketToSend> rtp_packet = packet->ReleasePacket();
     const bool owned_rtp_packet = rtp_packet != nullptr;
     RtpPacketSendResult success;

     if (rtp_packet != nullptr) {
       packet_sender_->SendRtpPacket(std::move(rtp_packet), pacing_info);
       success = RtpPacketSendResult::kSuccess;
     } else {
       success = packet_sender_->TimeToSendPacket(
           packet->ssrc(), packet->sequence_number(), packet->capture_time_ms(),
           packet->is_retransmission(), pacing_info);
     }

     if (success == RtpPacketSendResult::kSuccess ||
         success == RtpPacketSendResult::kPacketNotFound) {
       // Packet sent or invalid packet, remove it from queue.
       // TODO(webrtc:8052): Don't consume media budget on kInvalid.
       data_sent += packet->size();
       // Send succeeded, remove it from the queue.
       OnPacketSent(packet);
       if (recommended_probe_size && data_sent > *recommended_probe_size)
         break;
     } else if (owned_rtp_packet) {
       // Send failed, but we can't put it back in the queue, remove it without
       // consuming budget.
       packet_queue_.FinalizePop();
       break;
     } else {
       // Send failed, put it back into the queue.
       packet_queue_.CancelPop();
       break;
     }
   }

   if (legacy_packet_referencing_ && packet_queue_.Empty() && !Congested()) {
     // We can not send padding unless a normal packet has first been sent. If we
     // do, timestamps get messed up.
     if (packet_counter_ > 0) {
       DataSize padding_needed =
           (recommended_probe_size && *recommended_probe_size > data_sent)
               ? (*recommended_probe_size - data_sent)
               : DataSize::bytes(padding_budget_.bytes_remaining());
       if (padding_needed > DataSize::Zero()) {
         DataSize padding_sent = DataSize::Zero();
         padding_sent =
             packet_sender_->TimeToSendPadding(padding_needed, pacing_info);
         data_sent += padding_sent;
         OnPaddingSent(padding_sent);
       }
     }
   }

   if (is_probing) {
     probing_send_failure_ = data_sent == DataSize::Zero();
     if (!probing_send_failure_) {
       prober_.ProbeSent(CurrentTime().ms(), data_sent.bytes());
     }
   }
 }

 DataSize PacingController::PaddingToAdd(
     absl::optional<DataSize> recommended_probe_size,
     DataSize data_sent) {
   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 (packet_counter_ == 0) {
     // 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) {
     if (*recommended_probe_size > data_sent) {
       return *recommended_probe_size - data_sent;
     }
     return DataSize::Zero();
   }

   return DataSize::bytes(padding_budget_.bytes_remaining());
 }

 RoundRobinPacketQueue::QueuedPacket* PacingController::GetPendingPacket(
     const PacedPacketInfo& pacing_info) {
   if (packet_queue_.Empty()) {
     return nullptr;
   }

   // Since we need to release the lock in order to send, we first pop the
   // element from the priority queue but keep it in storage, so that we can
   // reinsert it if send fails.
   RoundRobinPacketQueue::QueuedPacket* packet = packet_queue_.BeginPop();
   bool audio_packet = packet->type() == RtpPacketToSend::Type::kAudio;
   bool apply_pacing = !audio_packet || pace_audio_;
   if (apply_pacing && (Congested() || (media_budget_.bytes_remaining() == 0 &&
                                        pacing_info.probe_cluster_id ==
                                            PacedPacketInfo::kNotAProbe))) {
     packet_queue_.CancelPop();
     return nullptr;
   }
   return packet;
 }

 void PacingController::OnPacketSent(
     RoundRobinPacketQueue::QueuedPacket* packet) {
   Timestamp now = CurrentTime();
   if (!first_sent_packet_time_) {
     first_sent_packet_time_ = now;
   }
   bool audio_packet = packet->type() == RtpPacketToSend::Type::kAudio;
   if (!audio_packet || account_for_audio_) {
     // Update media bytes sent.
     UpdateBudgetWithSentData(packet->size());
     last_send_time_ = now;
   }
   // Send succeeded, remove it from the queue.
   packet_queue_.FinalizePop();
   padding_failure_state_ = false;
 }

 void PacingController::OnPaddingSent(DataSize data_sent) {
   if (data_sent > DataSize::Zero()) {
     UpdateBudgetWithSentData(data_sent);
   } else {
     padding_failure_state_ = true;
   }
   last_send_time_ = CurrentTime();
 }

 void PacingController::UpdateBudgetWithElapsedTime(TimeDelta delta) {
   delta = std::min(kMaxProcessingInterval, delta);
   media_budget_.IncreaseBudget(delta.ms());
   padding_budget_.IncreaseBudget(delta.ms());
 }

 void PacingController::UpdateBudgetWithSentData(DataSize size) {
   outstanding_data_ += size;
   media_budget_.UseBudget(size.bytes());
   padding_budget_.UseBudget(size.bytes());
 }

 void PacingController::SetQueueTimeLimit(TimeDelta limit) {
   queue_time_limit = limit;
 }

 }  // 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/pacing/pacing_controller.h"

	#include <algorithm>
	#include <utility>
	#include <vector>

	#include "absl/memory/memory.h"
	#include "modules/pacing/bitrate_prober.h"
	#include "modules/pacing/interval_budget.h"
	#include "modules/utility/include/process_thread.h"
	#include "rtc_base/checks.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>();
	constexpr TimeDelta kMaxElapsedTime = TimeDelta::Seconds<2>();

	// Upper cap on process interval, in case process has not been called in a long
	// time.
	constexpr TimeDelta kMaxProcessingInterval = TimeDelta::Millis<30>();

	bool IsDisabled(const WebRtcKeyValueConfig& field_trials,
	absl::string_view key) {
	return field_trials.Lookup(key).find("Disabled") == 0;
	}

	bool IsEnabled(const WebRtcKeyValueConfig& field_trials,
	absl::string_view key) {
	return field_trials.Lookup(key).find("Enabled") == 0;
	}

	int GetPriorityForType(RtpPacketToSend::Type type) {
	switch (type) {
	case RtpPacketToSend::Type::kAudio:
	// Audio is always prioritized over other packet types.
	return 0;
	case RtpPacketToSend::Type::kRetransmission:
	// Send retransmissions before new media.
	return 1;
	case RtpPacketToSend::Type::kVideo:
	// Video has "normal" priority, in the old speak.
	return 2;
	case RtpPacketToSend::Type::kForwardErrorCorrection:
	// Send redundancy concurrently to video. If it is delayed it might have a
	// lower chance of being useful.
	return 2;
	case RtpPacketToSend::Type::kPadding:
	// Packets that are in themselves likely useless, only sent to keep the
	// BWE high.
	return 3;
	}
	}

	} // namespace

	const TimeDelta PacingController::kMaxExpectedQueueLength =
	TimeDelta::Millis<2000>();
	const float PacingController::kDefaultPaceMultiplier = 2.5f;
	const TimeDelta PacingController::kPausedProcessInterval =
	kCongestedPacketInterval;

	PacingController::PacingController(Clock* clock,
	PacketSender* packet_sender,
	RtcEventLog* event_log,
	const WebRtcKeyValueConfig* field_trials)
	: clock_(clock),
	packet_sender_(packet_sender),
	fallback_field_trials_(
	!field_trials ? absl::make_unique<FieldTrialBasedConfig>() : nullptr),
	field_trials_(field_trials ? field_trials : fallback_field_trials_.get()),
	drain_large_queues_(
	!IsDisabled(*field_trials_, "WebRTC-Pacer-DrainQueue")),
	send_padding_if_silent_(
	IsEnabled(*field_trials_, "WebRTC-Pacer-PadInSilence")),
	pace_audio_(!IsDisabled(*field_trials_, "WebRTC-Pacer-BlockAudio")),
	min_packet_limit_(kDefaultMinPacketLimit),
	last_timestamp_(clock_->CurrentTime()),
	paused_(false),
	media_budget_(0),
	padding_budget_(0),
	prober_(*field_trials_),
	probing_send_failure_(false),
	padding_failure_state_(false),
	pacing_bitrate_(DataRate::Zero()),
	time_last_process_(clock->CurrentTime()),
	last_send_time_(time_last_process_),
	packet_queue_(time_last_process_, field_trials),
	packet_counter_(0),
	congestion_window_size_(DataSize::PlusInfinity()),
	outstanding_data_(DataSize::Zero()),
	queue_time_limit(kMaxExpectedQueueLength),
	account_for_audio_(false),
	legacy_packet_referencing_(
	IsEnabled(*field_trials_, "WebRTC-Pacer-LegacyPacketReferencing")) {
	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::ms(min_packet_limit_ms.Get());
	UpdateBudgetWithElapsedTime(min_packet_limit_);
	}

	PacingController::~PacingController() = default;

	void PacingController::CreateProbeCluster(DataRate bitrate, int cluster_id) {
	prober_.CreateProbeCluster(bitrate.bps(), CurrentTime().ms(), cluster_id);
	}

	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::SetCongestionWindow(DataSize congestion_window_size) {
	congestion_window_size_ = congestion_window_size;
	}

	void PacingController::UpdateOutstandingData(DataSize outstanding_data) {
	outstanding_data_ = outstanding_data;
	}

	bool PacingController::Congested() const {
	if (congestion_window_size_.IsFinite()) {
	return outstanding_data_ >= congestion_window_size_;
	}
	return false;
	}

	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_EQ(0, packet_counter_);
	prober_.SetEnabled(enabled);
	}

	void PacingController::SetPacingRates(DataRate pacing_rate,
	DataRate padding_rate) {
	RTC_DCHECK_GT(pacing_rate, DataRate::Zero());
	pacing_bitrate_ = pacing_rate;
	padding_budget_.set_target_rate_kbps(padding_rate.kbps());

	RTC_LOG(LS_VERBOSE) << "bwe:pacer_updated pacing_kbps="
	<< pacing_bitrate_.kbps()
	<< " padding_budget_kbps=" << padding_rate.kbps();
	}

	void PacingController::InsertPacket(RtpPacketSender::Priority priority,
	uint32_t ssrc,
	uint16_t sequence_number,
	int64_t capture_time_ms,
	size_t bytes,
	bool retransmission) {
	RTC_DCHECK(pacing_bitrate_ > DataRate::Zero())
	<< "SetPacingRate must be called before InsertPacket.";

	Timestamp now = CurrentTime();
	prober_.OnIncomingPacket(bytes);

	if (capture_time_ms < 0)
	capture_time_ms = now.ms();

	RtpPacketToSend::Type type;
	switch (priority) {
	case RtpPacketSender::kHighPriority:
	type = RtpPacketToSend::Type::kAudio;
	break;
	case RtpPacketSender::kNormalPriority:
	type = RtpPacketToSend::Type::kRetransmission;
	break;
	default:
	type = RtpPacketToSend::Type::kVideo;
	}
	packet_queue_.Push(GetPriorityForType(type), type, ssrc, sequence_number,
	capture_time_ms, now, DataSize::bytes(bytes),
	retransmission, packet_counter_++);
	}

	void PacingController::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) {
	RTC_DCHECK(pacing_bitrate_ > DataRate::Zero())
	<< "SetPacingRate must be called before InsertPacket.";

	Timestamp now = CurrentTime();
	prober_.OnIncomingPacket(packet->payload_size());

	if (packet->capture_time_ms() < 0) {
	packet->set_capture_time_ms(now.ms());
	}

	RTC_CHECK(packet->packet_type());
	int priority = GetPriorityForType(*packet->packet_type());
	packet_queue_.Push(priority, now, packet_counter_++, std::move(packet));
	}

	void PacingController::SetAccountForAudioPackets(bool account_for_audio) {
	account_for_audio_ = account_for_audio;
	}

	TimeDelta PacingController::ExpectedQueueTime() const {
	RTC_DCHECK_GT(pacing_bitrate_, DataRate::Zero());
	return TimeDelta::ms(
	(QueueSizeData().bytes() * 8 * rtc::kNumMillisecsPerSec) /
	pacing_bitrate_.bps());
	}

	size_t PacingController::QueueSizePackets() const {
	return packet_queue_.SizeInPackets();
	}

	DataSize PacingController::QueueSizeData() const {
	return packet_queue_.Size();
	}

	absl::optional<Timestamp> PacingController::FirstSentPacketTime() const {
	return first_sent_packet_time_;
	}

	TimeDelta PacingController::OldestPacketWaitTime() const {
	Timestamp oldest_packet = packet_queue_.OldestEnqueueTime();
	if (oldest_packet.IsInfinite()) {
	return TimeDelta::Zero();
	}

	return CurrentTime() - oldest_packet;
	}

	TimeDelta PacingController::UpdateTimeAndGetElapsed(Timestamp now) {
	TimeDelta elapsed_time = now - time_last_process_;
	time_last_process_ = 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()) {
	// We send a padding packet every 500 ms to ensure we won't get stuck in
	// congested state due to no feedback being received.
	TimeDelta elapsed_since_last_send = now - last_send_time_;
	if (elapsed_since_last_send >= kCongestedPacketInterval) {
	// We can not send padding unless a normal packet has first been sent. If
	// we do, timestamps get messed up.
	if (packet_counter_ > 0) {
	return true;
	}
	}
	}
	return false;
	}

	absl::optional<TimeDelta> PacingController::TimeUntilNextProbe() {
	if (!prober_.IsProbing()) {
	return absl::nullopt;
	}

	TimeDelta time_delta =
	TimeDelta::ms(prober_.TimeUntilNextProbe(CurrentTime().ms()));
	if (time_delta > TimeDelta::Zero() \|\|
	(time_delta == TimeDelta::Zero() && !probing_send_failure_)) {
	return time_delta;
	}

	return absl::nullopt;
	}

	TimeDelta PacingController::TimeElapsedSinceLastProcess() const {
	return CurrentTime() - time_last_process_;
	}

	void PacingController::ProcessPackets() {
	Timestamp now = CurrentTime();
	TimeDelta elapsed_time = UpdateTimeAndGetElapsed(now);
	if (ShouldSendKeepalive(now)) {
	if (legacy_packet_referencing_) {
	OnPaddingSent(packet_sender_->TimeToSendPadding(DataSize::bytes(1),
	PacedPacketInfo()));
	} else {
	DataSize keepalive_data_sent = DataSize::Zero();
	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_->SendRtpPacket(std::move(packet), PacedPacketInfo());
	}
	OnPaddingSent(keepalive_data_sent);
	}
	}

	if (paused_)
	return;

	if (elapsed_time > TimeDelta::Zero()) {
	DataRate target_rate = pacing_bitrate_;
	DataSize queue_size_data = packet_queue_.Size();
	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_.UpdateQueueTime(CurrentTime());
	if (drain_large_queues_) {
	TimeDelta avg_time_left =
	std::max(TimeDelta::ms(1),
	queue_time_limit - packet_queue_.AverageQueueTime());
	DataRate min_rate_needed = queue_size_data / avg_time_left;
	if (min_rate_needed > target_rate) {
	target_rate = min_rate_needed;
	RTC_LOG(LS_VERBOSE) << "bwe:large_pacing_queue pacing_rate_kbps="
	<< target_rate.kbps();
	}
	}
	}

	media_budget_.set_target_rate_kbps(target_rate.kbps());
	UpdateBudgetWithElapsedTime(elapsed_time);
	}

	bool is_probing = prober_.IsProbing();
	PacedPacketInfo pacing_info;
	absl::optional<DataSize> recommended_probe_size;
	if (is_probing) {
	pacing_info = prober_.CurrentCluster();
	recommended_probe_size = DataSize::bytes(prober_.RecommendedMinProbeSize());
	}

	DataSize data_sent = DataSize::Zero();
	// The paused state is checked in the loop since it leaves the critical
	// section allowing the paused state to be changed from other code.
	while (!paused_) {
	auto* packet = GetPendingPacket(pacing_info);
	if (packet == nullptr) {
	// No packet available to send, check if we should send padding.
	if (!legacy_packet_referencing_) {
	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()) {
	// No padding packets were generated, quite send loop.
	break;
	}
	for (auto& packet : padding_packets) {
	EnqueuePacket(std::move(packet));
	}
	// Continue loop to send the padding that was just added.
	continue;
	}
	}

	// Can't fetch new packet and no padding to send, exit send loop.
	break;
	}

	std::unique_ptr<RtpPacketToSend> rtp_packet = packet->ReleasePacket();
	const bool owned_rtp_packet = rtp_packet != nullptr;
	RtpPacketSendResult success;

	if (rtp_packet != nullptr) {
	packet_sender_->SendRtpPacket(std::move(rtp_packet), pacing_info);
	success = RtpPacketSendResult::kSuccess;
	} else {
	success = packet_sender_->TimeToSendPacket(
	packet->ssrc(), packet->sequence_number(), packet->capture_time_ms(),
	packet->is_retransmission(), pacing_info);
	}

	if (success == RtpPacketSendResult::kSuccess \|\|
	success == RtpPacketSendResult::kPacketNotFound) {
	// Packet sent or invalid packet, remove it from queue.
	// TODO(webrtc:8052): Don't consume media budget on kInvalid.
	data_sent += packet->size();
	// Send succeeded, remove it from the queue.
	OnPacketSent(packet);
	if (recommended_probe_size && data_sent > *recommended_probe_size)
	break;
	} else if (owned_rtp_packet) {
	// Send failed, but we can't put it back in the queue, remove it without
	// consuming budget.
	packet_queue_.FinalizePop();
	break;
	} else {
	// Send failed, put it back into the queue.
	packet_queue_.CancelPop();
	break;
	}
	}

	if (legacy_packet_referencing_ && packet_queue_.Empty() && !Congested()) {
	// We can not send padding unless a normal packet has first been sent. If we
	// do, timestamps get messed up.
	if (packet_counter_ > 0) {
	DataSize padding_needed =
	(recommended_probe_size && *recommended_probe_size > data_sent)
	? (*recommended_probe_size - data_sent)
	: DataSize::bytes(padding_budget_.bytes_remaining());
	if (padding_needed > DataSize::Zero()) {
	DataSize padding_sent = DataSize::Zero();
	padding_sent =
	packet_sender_->TimeToSendPadding(padding_needed, pacing_info);
	data_sent += padding_sent;
	OnPaddingSent(padding_sent);
	}
	}
	}

	if (is_probing) {
	probing_send_failure_ = data_sent == DataSize::Zero();
	if (!probing_send_failure_) {
	prober_.ProbeSent(CurrentTime().ms(), data_sent.bytes());
	}
	}
	}

	DataSize PacingController::PaddingToAdd(
	absl::optional<DataSize> recommended_probe_size,
	DataSize data_sent) {
	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 (packet_counter_ == 0) {
	// 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) {
	if (*recommended_probe_size > data_sent) {
	return *recommended_probe_size - data_sent;
	}
	return DataSize::Zero();
	}

	return DataSize::bytes(padding_budget_.bytes_remaining());
	}

	RoundRobinPacketQueue::QueuedPacket* PacingController::GetPendingPacket(
	const PacedPacketInfo& pacing_info) {
	if (packet_queue_.Empty()) {
	return nullptr;
	}

	// Since we need to release the lock in order to send, we first pop the
	// element from the priority queue but keep it in storage, so that we can
	// reinsert it if send fails.
	RoundRobinPacketQueue::QueuedPacket* packet = packet_queue_.BeginPop();
	bool audio_packet = packet->type() == RtpPacketToSend::Type::kAudio;
	bool apply_pacing = !audio_packet \|\| pace_audio_;
	if (apply_pacing && (Congested() \|\| (media_budget_.bytes_remaining() == 0 &&
	pacing_info.probe_cluster_id ==
	PacedPacketInfo::kNotAProbe))) {
	packet_queue_.CancelPop();
	return nullptr;
	}
	return packet;
	}

	void PacingController::OnPacketSent(
	RoundRobinPacketQueue::QueuedPacket* packet) {
	Timestamp now = CurrentTime();
	if (!first_sent_packet_time_) {
	first_sent_packet_time_ = now;
	}
	bool audio_packet = packet->type() == RtpPacketToSend::Type::kAudio;
	if (!audio_packet \|\| account_for_audio_) {
	// Update media bytes sent.
	UpdateBudgetWithSentData(packet->size());
	last_send_time_ = now;
	}
	// Send succeeded, remove it from the queue.
	packet_queue_.FinalizePop();
	padding_failure_state_ = false;
	}

	void PacingController::OnPaddingSent(DataSize data_sent) {
	if (data_sent > DataSize::Zero()) {
	UpdateBudgetWithSentData(data_sent);
	} else {
	padding_failure_state_ = true;
	}
	last_send_time_ = CurrentTime();
	}

	void PacingController::UpdateBudgetWithElapsedTime(TimeDelta delta) {
	delta = std::min(kMaxProcessingInterval, delta);
	media_budget_.IncreaseBudget(delta.ms());
	padding_budget_.IncreaseBudget(delta.ms());
	}

	void PacingController::UpdateBudgetWithSentData(DataSize size) {
	outstanding_data_ += size;
	media_budget_.UseBudget(size.bytes());
	padding_budget_.UseBudget(size.bytes());
	}

	void PacingController::SetQueueTimeLimit(TimeDelta limit) {
	queue_time_limit = limit;
	}

	} // namespace webrtc