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

 /*
  *  Copyright (c) 2012 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/paced_sender.h"

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

 #include "absl/memory/memory.h"
 #include "logging/rtc_event_log/rtc_event_log.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 "system_wrappers/include/clock.h"

 namespace webrtc {
 namespace {
 // Time limit in milliseconds between packet bursts.
 const int64_t kDefaultMinPacketLimitMs = 5;
 const int64_t kCongestedPacketIntervalMs = 500;
 const int64_t kPausedProcessIntervalMs = kCongestedPacketIntervalMs;
 const int64_t kMaxElapsedTimeMs = 2000;

 // Upper cap on process interval, in case process has not been called in a long
 // time.
 const int64_t kMaxIntervalTimeMs = 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:
       // Redundancy is OK to drop, but the content is hopefully not useless.
       return 3;
     case RtpPacketToSend::Type::kPadding:
       // Packets that are in themselves likely useless, only sent to keep the
       // BWE high.
       return 4;
   }
 }

 }  // namespace
 const int64_t PacedSender::kMaxQueueLengthMs = 2000;
 const float PacedSender::kDefaultPaceMultiplier = 2.5f;

 PacedSender::PacedSender(Clock* clock,
                          PacketRouter* packet_router,
                          RtcEventLog* event_log,
                          const WebRtcKeyValueConfig* field_trials)
     : clock_(clock),
       packet_router_(packet_router),
       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_ms_("", kDefaultMinPacketLimitMs),
       last_timestamp_ms_(clock_->TimeInMilliseconds()),
       paused_(false),
       media_budget_(0),
       padding_budget_(0),
       prober_(*field_trials_),
       probing_send_failure_(false),
       pacing_bitrate_kbps_(0),
       time_last_process_us_(clock->TimeInMicroseconds()),
       last_send_time_us_(clock->TimeInMicroseconds()),
       first_sent_packet_ms_(-1),
       packets_(clock->TimeInMicroseconds()),
       packet_counter_(0),
       queue_time_limit(kMaxQueueLengthMs),
       account_for_audio_(false),
       legacy_packet_referencing_(
           !IsDisabled(*field_trials_, "WebRTC-Pacer-LegacyPacketReferencing")) {
   if (!drain_large_queues_) {
     RTC_LOG(LS_WARNING) << "Pacer queues will not be drained,"
                            "pushback experiment must be enabled.";
   }
   ParseFieldTrial({&min_packet_limit_ms_},
                   field_trials_->Lookup("WebRTC-Pacer-MinPacketLimitMs"));
   UpdateBudgetWithElapsedTime(min_packet_limit_ms_);
 }

 PacedSender::~PacedSender() {}

 void PacedSender::CreateProbeCluster(int bitrate_bps, int cluster_id) {
   rtc::CritScope cs(&critsect_);
   prober_.CreateProbeCluster(bitrate_bps, TimeMilliseconds(), cluster_id);
 }

 void PacedSender::Pause() {
   {
     rtc::CritScope cs(&critsect_);
     if (!paused_)
       RTC_LOG(LS_INFO) << "PacedSender paused.";
     paused_ = true;
     packets_.SetPauseState(true, TimeMilliseconds());
   }
   rtc::CritScope cs(&process_thread_lock_);
   // Tell the process thread to call our TimeUntilNextProcess() method to get
   // a new (longer) estimate for when to call Process().
   if (process_thread_)
     process_thread_->WakeUp(this);
 }

 void PacedSender::Resume() {
   {
     rtc::CritScope cs(&critsect_);
     if (paused_)
       RTC_LOG(LS_INFO) << "PacedSender resumed.";
     paused_ = false;
     packets_.SetPauseState(false, TimeMilliseconds());
   }
   rtc::CritScope cs(&process_thread_lock_);
   // Tell the process thread to call our TimeUntilNextProcess() method to
   // refresh the estimate for when to call Process().
   if (process_thread_)
     process_thread_->WakeUp(this);
 }

 void PacedSender::SetCongestionWindow(int64_t congestion_window_bytes) {
   rtc::CritScope cs(&critsect_);
   congestion_window_bytes_ = congestion_window_bytes;
 }

 void PacedSender::UpdateOutstandingData(int64_t outstanding_bytes) {
   rtc::CritScope cs(&critsect_);
   outstanding_bytes_ = outstanding_bytes;
 }

 bool PacedSender::Congested() const {
   if (congestion_window_bytes_ == kNoCongestionWindow)
     return false;
   return outstanding_bytes_ >= congestion_window_bytes_;
 }

 int64_t PacedSender::TimeMilliseconds() const {
   int64_t time_ms = clock_->TimeInMilliseconds();
   if (time_ms < last_timestamp_ms_) {
     RTC_LOG(LS_WARNING)
         << "Non-monotonic clock behavior observed. Previous timestamp: "
         << last_timestamp_ms_ << ", new timestamp: " << time_ms;
     RTC_DCHECK_GE(time_ms, last_timestamp_ms_);
     time_ms = last_timestamp_ms_;
   }
   last_timestamp_ms_ = time_ms;
   return time_ms;
 }

 void PacedSender::SetProbingEnabled(bool enabled) {
   rtc::CritScope cs(&critsect_);
   RTC_CHECK_EQ(0, packet_counter_);
   prober_.SetEnabled(enabled);
 }

 void PacedSender::SetPacingRates(uint32_t pacing_rate_bps,
                                  uint32_t padding_rate_bps) {
   rtc::CritScope cs(&critsect_);
   RTC_DCHECK(pacing_rate_bps > 0);
   pacing_bitrate_kbps_ = pacing_rate_bps / 1000;
   padding_budget_.set_target_rate_kbps(padding_rate_bps / 1000);

   RTC_LOG(LS_VERBOSE) << "bwe:pacer_updated pacing_kbps="
                       << pacing_bitrate_kbps_
                       << " padding_budget_kbps=" << padding_rate_bps / 1000;
 }

 void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
                                uint32_t ssrc,
                                uint16_t sequence_number,
                                int64_t capture_time_ms,
                                size_t bytes,
                                bool retransmission) {
   rtc::CritScope cs(&critsect_);
   RTC_DCHECK(pacing_bitrate_kbps_ > 0)
       << "SetPacingRate must be called before InsertPacket.";

   int64_t now_ms = TimeMilliseconds();
   prober_.OnIncomingPacket(bytes);

   if (capture_time_ms < 0)
     capture_time_ms = now_ms;

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

 void PacedSender::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) {
   rtc::CritScope cs(&critsect_);
   RTC_DCHECK(pacing_bitrate_kbps_ > 0)
       << "SetPacingRate must be called before InsertPacket.";

   int64_t now_ms = TimeMilliseconds();
   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());
   packets_.Push(priority, now_ms, packet_counter_++, std::move(packet));
 }

 void PacedSender::SetAccountForAudioPackets(bool account_for_audio) {
   rtc::CritScope cs(&critsect_);
   account_for_audio_ = account_for_audio;
 }

 int64_t PacedSender::ExpectedQueueTimeMs() const {
   rtc::CritScope cs(&critsect_);
   RTC_DCHECK_GT(pacing_bitrate_kbps_, 0);
   return static_cast<int64_t>(packets_.SizeInBytes() * 8 /
                               pacing_bitrate_kbps_);
 }

 size_t PacedSender::QueueSizePackets() const {
   rtc::CritScope cs(&critsect_);
   return packets_.SizeInPackets();
 }

 int64_t PacedSender::QueueSizeBytes() const {
   rtc::CritScope cs(&critsect_);
   return packets_.SizeInBytes();
 }

 int64_t PacedSender::FirstSentPacketTimeMs() const {
   rtc::CritScope cs(&critsect_);
   return first_sent_packet_ms_;
 }

 int64_t PacedSender::QueueInMs() const {
   rtc::CritScope cs(&critsect_);

   int64_t oldest_packet = packets_.OldestEnqueueTimeMs();
   if (oldest_packet == 0)
     return 0;

   return TimeMilliseconds() - oldest_packet;
 }

 int64_t PacedSender::TimeUntilNextProcess() {
   rtc::CritScope cs(&critsect_);
   int64_t elapsed_time_us =
       clock_->TimeInMicroseconds() - time_last_process_us_;
   int64_t elapsed_time_ms = (elapsed_time_us + 500) / 1000;
   // When paused we wake up every 500 ms to send a padding packet to ensure
   // we won't get stuck in the paused state due to no feedback being received.
   if (paused_)
     return std::max<int64_t>(kPausedProcessIntervalMs - elapsed_time_ms, 0);

   if (prober_.IsProbing()) {
     int64_t ret = prober_.TimeUntilNextProbe(TimeMilliseconds());
     if (ret > 0 || (ret == 0 && !probing_send_failure_))
       return ret;
   }
   return std::max<int64_t>(min_packet_limit_ms_ - elapsed_time_ms, 0);
 }

 int64_t PacedSender::UpdateTimeAndGetElapsedMs(int64_t now_us) {
   int64_t elapsed_time_ms = (now_us - time_last_process_us_ + 500) / 1000;
   time_last_process_us_ = now_us;
   if (elapsed_time_ms > kMaxElapsedTimeMs) {
     RTC_LOG(LS_WARNING) << "Elapsed time (" << elapsed_time_ms
                         << " ms) longer than expected, limiting to "
                         << kMaxElapsedTimeMs << " ms";
     elapsed_time_ms = kMaxElapsedTimeMs;
   }
   return elapsed_time_ms;
 }

 bool PacedSender::ShouldSendKeepalive(int64_t now_us) 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.
     int64_t elapsed_since_last_send_us = now_us - last_send_time_us_;
     if (elapsed_since_last_send_us >= kCongestedPacketIntervalMs * 1000) {
       // 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;
 }

 void PacedSender::Process() {
   rtc::CritScope cs(&critsect_);
   int64_t now_us = clock_->TimeInMicroseconds();
   int64_t elapsed_time_ms = UpdateTimeAndGetElapsedMs(now_us);
   if (ShouldSendKeepalive(now_us)) {
     if (legacy_packet_referencing_) {
       critsect_.Leave();
       size_t bytes_sent =
           packet_router_->TimeToSendPadding(1, PacedPacketInfo());
       critsect_.Enter();
       OnPaddingSent(bytes_sent);
     } else {
       critsect_.Leave();
       std::vector<std::unique_ptr<RtpPacketToSend>> keepalive_packets =
           packet_router_->GeneratePadding(1);
       critsect_.Enter();
       for (auto& packet : keepalive_packets) {
         EnqueuePacket(std::move(packet));
       }
     }
   }

   if (paused_)
     return;

   if (elapsed_time_ms > 0) {
     int target_bitrate_kbps = pacing_bitrate_kbps_;
     size_t queue_size_bytes = packets_.SizeInBytes();
     if (queue_size_bytes > 0) {
       // 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.
       packets_.UpdateQueueTime(TimeMilliseconds());
       if (drain_large_queues_) {
         int64_t avg_time_left_ms = std::max<int64_t>(
             1, queue_time_limit - packets_.AverageQueueTimeMs());
         int min_bitrate_needed_kbps =
             static_cast<int>(queue_size_bytes * 8 / avg_time_left_ms);
         if (min_bitrate_needed_kbps > target_bitrate_kbps) {
           target_bitrate_kbps = min_bitrate_needed_kbps;
           RTC_LOG(LS_VERBOSE) << "bwe:large_pacing_queue pacing_rate_kbps="
                               << target_bitrate_kbps;
         }
       }
     }

     media_budget_.set_target_rate_kbps(target_bitrate_kbps);
     UpdateBudgetWithElapsedTime(elapsed_time_ms);
   }

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

   size_t bytes_sent = 0;
   // 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_) {
         size_t padding_bytes_to_add =
             PaddingBytesToAdd(recommended_probe_size, bytes_sent);
         if (padding_bytes_to_add > 0) {
           critsect_.Leave();
           std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
               packet_router_->GeneratePadding(padding_bytes_to_add);
           critsect_.Enter();
           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) {
       critsect_.Leave();
       packet_router_->SendPacket(std::move(rtp_packet), pacing_info);
       critsect_.Enter();
       success = RtpPacketSendResult::kSuccess;
     } else {
       critsect_.Leave();
       success = packet_router_->TimeToSendPacket(
           packet->ssrc(), packet->sequence_number(), packet->capture_time_ms(),
           packet->is_retransmission(), pacing_info);
       critsect_.Enter();
     }

     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.
       bytes_sent += packet->size_in_bytes();
       // Send succeeded, remove it from the queue.
       OnPacketSent(packet);
       if (recommended_probe_size && bytes_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.
       packets_.FinalizePop();
       break;
     } else {
       // Send failed, put it back into the queue.
       packets_.CancelPop();
       break;
     }
   }

   if (legacy_packet_referencing_ && packets_.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) {
       int padding_needed = static_cast<int>(
           recommended_probe_size ? (*recommended_probe_size - bytes_sent)
                                  : padding_budget_.bytes_remaining());
       if (padding_needed > 0) {
         size_t padding_sent = 0;
         critsect_.Leave();
         padding_sent =
             packet_router_->TimeToSendPadding(padding_needed, pacing_info);
         critsect_.Enter();
         bytes_sent += padding_sent;
         OnPaddingSent(padding_sent);
       }
     }
   }

   if (is_probing) {
     probing_send_failure_ = bytes_sent == 0;
     if (!probing_send_failure_)
       prober_.ProbeSent(TimeMilliseconds(), bytes_sent);
   }
 }

 void PacedSender::ProcessThreadAttached(ProcessThread* process_thread) {
   RTC_LOG(LS_INFO) << "ProcessThreadAttached 0x" << process_thread;
   rtc::CritScope cs(&process_thread_lock_);
   process_thread_ = process_thread;
 }

 size_t PacedSender::PaddingBytesToAdd(
     absl::optional<size_t> recommended_probe_size,
     size_t bytes_sent) {
   if (!packets_.Empty()) {
     // Actual payload available, no need to add padding.
     return 0;
   }

   if (Congested()) {
     // Don't add padding if congested, even if requested for probing.
     return 0;
   }

   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 0;
   }

   if (recommended_probe_size) {
     if (*recommended_probe_size > bytes_sent) {
       return *recommended_probe_size - bytes_sent;
     }
     return 0;
   }

   return padding_budget_.bytes_remaining();
 }

 RoundRobinPacketQueue::QueuedPacket* PacedSender::GetPendingPacket(
     const PacedPacketInfo& pacing_info) {
   if (packets_.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 = packets_.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))) {
     packets_.CancelPop();
     return nullptr;
   }
   return packet;
 }

 void PacedSender::OnPacketSent(RoundRobinPacketQueue::QueuedPacket* packet) {
   if (first_sent_packet_ms_ == -1)
     first_sent_packet_ms_ = TimeMilliseconds();
   bool audio_packet = packet->type() == RtpPacketToSend::Type::kAudio;
   if (!audio_packet || account_for_audio_) {
     // Update media bytes sent.
     UpdateBudgetWithBytesSent(packet->size_in_bytes());
     last_send_time_us_ = clock_->TimeInMicroseconds();
   }
   // Send succeeded, remove it from the queue.
   packets_.FinalizePop();
 }

 void PacedSender::OnPaddingSent(size_t bytes_sent) {
   if (bytes_sent > 0) {
     UpdateBudgetWithBytesSent(bytes_sent);
   }
   last_send_time_us_ = clock_->TimeInMicroseconds();
 }

 void PacedSender::UpdateBudgetWithElapsedTime(int64_t delta_time_ms) {
   delta_time_ms = std::min(kMaxIntervalTimeMs, delta_time_ms);
   media_budget_.IncreaseBudget(delta_time_ms);
   padding_budget_.IncreaseBudget(delta_time_ms);
 }

 void PacedSender::UpdateBudgetWithBytesSent(size_t bytes_sent) {
   outstanding_bytes_ += bytes_sent;
   media_budget_.UseBudget(bytes_sent);
   padding_budget_.UseBudget(bytes_sent);
 }

 void PacedSender::SetQueueTimeLimit(int limit_ms) {
   rtc::CritScope cs(&critsect_);
   queue_time_limit = limit_ms;
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2012 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/paced_sender.h"

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

	#include "absl/memory/memory.h"
	#include "logging/rtc_event_log/rtc_event_log.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 "system_wrappers/include/clock.h"

	namespace webrtc {
	namespace {
	// Time limit in milliseconds between packet bursts.
	const int64_t kDefaultMinPacketLimitMs = 5;
	const int64_t kCongestedPacketIntervalMs = 500;
	const int64_t kPausedProcessIntervalMs = kCongestedPacketIntervalMs;
	const int64_t kMaxElapsedTimeMs = 2000;

	// Upper cap on process interval, in case process has not been called in a long
	// time.
	const int64_t kMaxIntervalTimeMs = 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:
	// Redundancy is OK to drop, but the content is hopefully not useless.
	return 3;
	case RtpPacketToSend::Type::kPadding:
	// Packets that are in themselves likely useless, only sent to keep the
	// BWE high.
	return 4;
	}
	}

	} // namespace
	const int64_t PacedSender::kMaxQueueLengthMs = 2000;
	const float PacedSender::kDefaultPaceMultiplier = 2.5f;

	PacedSender::PacedSender(Clock* clock,
	PacketRouter* packet_router,
	RtcEventLog* event_log,
	const WebRtcKeyValueConfig* field_trials)
	: clock_(clock),
	packet_router_(packet_router),
	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_ms_("", kDefaultMinPacketLimitMs),
	last_timestamp_ms_(clock_->TimeInMilliseconds()),
	paused_(false),
	media_budget_(0),
	padding_budget_(0),
	prober_(*field_trials_),
	probing_send_failure_(false),
	pacing_bitrate_kbps_(0),
	time_last_process_us_(clock->TimeInMicroseconds()),
	last_send_time_us_(clock->TimeInMicroseconds()),
	first_sent_packet_ms_(-1),
	packets_(clock->TimeInMicroseconds()),
	packet_counter_(0),
	queue_time_limit(kMaxQueueLengthMs),
	account_for_audio_(false),
	legacy_packet_referencing_(
	!IsDisabled(*field_trials_, "WebRTC-Pacer-LegacyPacketReferencing")) {
	if (!drain_large_queues_) {
	RTC_LOG(LS_WARNING) << "Pacer queues will not be drained,"
	"pushback experiment must be enabled.";
	}
	ParseFieldTrial({&min_packet_limit_ms_},
	field_trials_->Lookup("WebRTC-Pacer-MinPacketLimitMs"));
	UpdateBudgetWithElapsedTime(min_packet_limit_ms_);
	}

	PacedSender::~PacedSender() {}

	void PacedSender::CreateProbeCluster(int bitrate_bps, int cluster_id) {
	rtc::CritScope cs(&critsect_);
	prober_.CreateProbeCluster(bitrate_bps, TimeMilliseconds(), cluster_id);
	}

	void PacedSender::Pause() {
	{
	rtc::CritScope cs(&critsect_);
	if (!paused_)
	RTC_LOG(LS_INFO) << "PacedSender paused.";
	paused_ = true;
	packets_.SetPauseState(true, TimeMilliseconds());
	}
	rtc::CritScope cs(&process_thread_lock_);
	// Tell the process thread to call our TimeUntilNextProcess() method to get
	// a new (longer) estimate for when to call Process().
	if (process_thread_)
	process_thread_->WakeUp(this);
	}

	void PacedSender::Resume() {
	{
	rtc::CritScope cs(&critsect_);
	if (paused_)
	RTC_LOG(LS_INFO) << "PacedSender resumed.";
	paused_ = false;
	packets_.SetPauseState(false, TimeMilliseconds());
	}
	rtc::CritScope cs(&process_thread_lock_);
	// Tell the process thread to call our TimeUntilNextProcess() method to
	// refresh the estimate for when to call Process().
	if (process_thread_)
	process_thread_->WakeUp(this);
	}

	void PacedSender::SetCongestionWindow(int64_t congestion_window_bytes) {
	rtc::CritScope cs(&critsect_);
	congestion_window_bytes_ = congestion_window_bytes;
	}

	void PacedSender::UpdateOutstandingData(int64_t outstanding_bytes) {
	rtc::CritScope cs(&critsect_);
	outstanding_bytes_ = outstanding_bytes;
	}

	bool PacedSender::Congested() const {
	if (congestion_window_bytes_ == kNoCongestionWindow)
	return false;
	return outstanding_bytes_ >= congestion_window_bytes_;
	}

	int64_t PacedSender::TimeMilliseconds() const {
	int64_t time_ms = clock_->TimeInMilliseconds();
	if (time_ms < last_timestamp_ms_) {
	RTC_LOG(LS_WARNING)
	<< "Non-monotonic clock behavior observed. Previous timestamp: "
	<< last_timestamp_ms_ << ", new timestamp: " << time_ms;
	RTC_DCHECK_GE(time_ms, last_timestamp_ms_);
	time_ms = last_timestamp_ms_;
	}
	last_timestamp_ms_ = time_ms;
	return time_ms;
	}

	void PacedSender::SetProbingEnabled(bool enabled) {
	rtc::CritScope cs(&critsect_);
	RTC_CHECK_EQ(0, packet_counter_);
	prober_.SetEnabled(enabled);
	}

	void PacedSender::SetPacingRates(uint32_t pacing_rate_bps,
	uint32_t padding_rate_bps) {
	rtc::CritScope cs(&critsect_);
	RTC_DCHECK(pacing_rate_bps > 0);
	pacing_bitrate_kbps_ = pacing_rate_bps / 1000;
	padding_budget_.set_target_rate_kbps(padding_rate_bps / 1000);

	RTC_LOG(LS_VERBOSE) << "bwe:pacer_updated pacing_kbps="
	<< pacing_bitrate_kbps_
	<< " padding_budget_kbps=" << padding_rate_bps / 1000;
	}

	void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
	uint32_t ssrc,
	uint16_t sequence_number,
	int64_t capture_time_ms,
	size_t bytes,
	bool retransmission) {
	rtc::CritScope cs(&critsect_);
	RTC_DCHECK(pacing_bitrate_kbps_ > 0)
	<< "SetPacingRate must be called before InsertPacket.";

	int64_t now_ms = TimeMilliseconds();
	prober_.OnIncomingPacket(bytes);

	if (capture_time_ms < 0)
	capture_time_ms = now_ms;

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

	void PacedSender::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) {
	rtc::CritScope cs(&critsect_);
	RTC_DCHECK(pacing_bitrate_kbps_ > 0)
	<< "SetPacingRate must be called before InsertPacket.";

	int64_t now_ms = TimeMilliseconds();
	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());
	packets_.Push(priority, now_ms, packet_counter_++, std::move(packet));
	}

	void PacedSender::SetAccountForAudioPackets(bool account_for_audio) {
	rtc::CritScope cs(&critsect_);
	account_for_audio_ = account_for_audio;
	}

	int64_t PacedSender::ExpectedQueueTimeMs() const {
	rtc::CritScope cs(&critsect_);
	RTC_DCHECK_GT(pacing_bitrate_kbps_, 0);
	return static_cast<int64_t>(packets_.SizeInBytes() * 8 /
	pacing_bitrate_kbps_);
	}

	size_t PacedSender::QueueSizePackets() const {
	rtc::CritScope cs(&critsect_);
	return packets_.SizeInPackets();
	}

	int64_t PacedSender::QueueSizeBytes() const {
	rtc::CritScope cs(&critsect_);
	return packets_.SizeInBytes();
	}

	int64_t PacedSender::FirstSentPacketTimeMs() const {
	rtc::CritScope cs(&critsect_);
	return first_sent_packet_ms_;
	}

	int64_t PacedSender::QueueInMs() const {
	rtc::CritScope cs(&critsect_);

	int64_t oldest_packet = packets_.OldestEnqueueTimeMs();
	if (oldest_packet == 0)
	return 0;

	return TimeMilliseconds() - oldest_packet;
	}

	int64_t PacedSender::TimeUntilNextProcess() {
	rtc::CritScope cs(&critsect_);
	int64_t elapsed_time_us =
	clock_->TimeInMicroseconds() - time_last_process_us_;
	int64_t elapsed_time_ms = (elapsed_time_us + 500) / 1000;
	// When paused we wake up every 500 ms to send a padding packet to ensure
	// we won't get stuck in the paused state due to no feedback being received.
	if (paused_)
	return std::max<int64_t>(kPausedProcessIntervalMs - elapsed_time_ms, 0);

	if (prober_.IsProbing()) {
	int64_t ret = prober_.TimeUntilNextProbe(TimeMilliseconds());
	if (ret > 0 \|\| (ret == 0 && !probing_send_failure_))
	return ret;
	}
	return std::max<int64_t>(min_packet_limit_ms_ - elapsed_time_ms, 0);
	}

	int64_t PacedSender::UpdateTimeAndGetElapsedMs(int64_t now_us) {
	int64_t elapsed_time_ms = (now_us - time_last_process_us_ + 500) / 1000;
	time_last_process_us_ = now_us;
	if (elapsed_time_ms > kMaxElapsedTimeMs) {
	RTC_LOG(LS_WARNING) << "Elapsed time (" << elapsed_time_ms
	<< " ms) longer than expected, limiting to "
	<< kMaxElapsedTimeMs << " ms";
	elapsed_time_ms = kMaxElapsedTimeMs;
	}
	return elapsed_time_ms;
	}

	bool PacedSender::ShouldSendKeepalive(int64_t now_us) 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.
	int64_t elapsed_since_last_send_us = now_us - last_send_time_us_;
	if (elapsed_since_last_send_us >= kCongestedPacketIntervalMs * 1000) {
	// 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;
	}

	void PacedSender::Process() {
	rtc::CritScope cs(&critsect_);
	int64_t now_us = clock_->TimeInMicroseconds();
	int64_t elapsed_time_ms = UpdateTimeAndGetElapsedMs(now_us);
	if (ShouldSendKeepalive(now_us)) {
	if (legacy_packet_referencing_) {
	critsect_.Leave();
	size_t bytes_sent =
	packet_router_->TimeToSendPadding(1, PacedPacketInfo());
	critsect_.Enter();
	OnPaddingSent(bytes_sent);
	} else {
	critsect_.Leave();
	std::vector<std::unique_ptr<RtpPacketToSend>> keepalive_packets =
	packet_router_->GeneratePadding(1);
	critsect_.Enter();
	for (auto& packet : keepalive_packets) {
	EnqueuePacket(std::move(packet));
	}
	}
	}

	if (paused_)
	return;

	if (elapsed_time_ms > 0) {
	int target_bitrate_kbps = pacing_bitrate_kbps_;
	size_t queue_size_bytes = packets_.SizeInBytes();
	if (queue_size_bytes > 0) {
	// 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.
	packets_.UpdateQueueTime(TimeMilliseconds());
	if (drain_large_queues_) {
	int64_t avg_time_left_ms = std::max<int64_t>(
	1, queue_time_limit - packets_.AverageQueueTimeMs());
	int min_bitrate_needed_kbps =
	static_cast<int>(queue_size_bytes * 8 / avg_time_left_ms);
	if (min_bitrate_needed_kbps > target_bitrate_kbps) {
	target_bitrate_kbps = min_bitrate_needed_kbps;
	RTC_LOG(LS_VERBOSE) << "bwe:large_pacing_queue pacing_rate_kbps="
	<< target_bitrate_kbps;
	}
	}
	}

	media_budget_.set_target_rate_kbps(target_bitrate_kbps);
	UpdateBudgetWithElapsedTime(elapsed_time_ms);
	}

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

	size_t bytes_sent = 0;
	// 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_) {
	size_t padding_bytes_to_add =
	PaddingBytesToAdd(recommended_probe_size, bytes_sent);
	if (padding_bytes_to_add > 0) {
	critsect_.Leave();
	std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
	packet_router_->GeneratePadding(padding_bytes_to_add);
	critsect_.Enter();
	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) {
	critsect_.Leave();
	packet_router_->SendPacket(std::move(rtp_packet), pacing_info);
	critsect_.Enter();
	success = RtpPacketSendResult::kSuccess;
	} else {
	critsect_.Leave();
	success = packet_router_->TimeToSendPacket(
	packet->ssrc(), packet->sequence_number(), packet->capture_time_ms(),
	packet->is_retransmission(), pacing_info);
	critsect_.Enter();
	}

	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.
	bytes_sent += packet->size_in_bytes();
	// Send succeeded, remove it from the queue.
	OnPacketSent(packet);
	if (recommended_probe_size && bytes_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.
	packets_.FinalizePop();
	break;
	} else {
	// Send failed, put it back into the queue.
	packets_.CancelPop();
	break;
	}
	}

	if (legacy_packet_referencing_ && packets_.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) {
	int padding_needed = static_cast<int>(
	recommended_probe_size ? (*recommended_probe_size - bytes_sent)
	: padding_budget_.bytes_remaining());
	if (padding_needed > 0) {
	size_t padding_sent = 0;
	critsect_.Leave();
	padding_sent =
	packet_router_->TimeToSendPadding(padding_needed, pacing_info);
	critsect_.Enter();
	bytes_sent += padding_sent;
	OnPaddingSent(padding_sent);
	}
	}
	}

	if (is_probing) {
	probing_send_failure_ = bytes_sent == 0;
	if (!probing_send_failure_)
	prober_.ProbeSent(TimeMilliseconds(), bytes_sent);
	}
	}

	void PacedSender::ProcessThreadAttached(ProcessThread* process_thread) {
	RTC_LOG(LS_INFO) << "ProcessThreadAttached 0x" << process_thread;
	rtc::CritScope cs(&process_thread_lock_);
	process_thread_ = process_thread;
	}

	size_t PacedSender::PaddingBytesToAdd(
	absl::optional<size_t> recommended_probe_size,
	size_t bytes_sent) {
	if (!packets_.Empty()) {
	// Actual payload available, no need to add padding.
	return 0;
	}

	if (Congested()) {
	// Don't add padding if congested, even if requested for probing.
	return 0;
	}

	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 0;
	}

	if (recommended_probe_size) {
	if (*recommended_probe_size > bytes_sent) {
	return *recommended_probe_size - bytes_sent;
	}
	return 0;
	}

	return padding_budget_.bytes_remaining();
	}

	RoundRobinPacketQueue::QueuedPacket* PacedSender::GetPendingPacket(
	const PacedPacketInfo& pacing_info) {
	if (packets_.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 = packets_.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))) {
	packets_.CancelPop();
	return nullptr;
	}
	return packet;
	}

	void PacedSender::OnPacketSent(RoundRobinPacketQueue::QueuedPacket* packet) {
	if (first_sent_packet_ms_ == -1)
	first_sent_packet_ms_ = TimeMilliseconds();
	bool audio_packet = packet->type() == RtpPacketToSend::Type::kAudio;
	if (!audio_packet \|\| account_for_audio_) {
	// Update media bytes sent.
	UpdateBudgetWithBytesSent(packet->size_in_bytes());
	last_send_time_us_ = clock_->TimeInMicroseconds();
	}
	// Send succeeded, remove it from the queue.
	packets_.FinalizePop();
	}

	void PacedSender::OnPaddingSent(size_t bytes_sent) {
	if (bytes_sent > 0) {
	UpdateBudgetWithBytesSent(bytes_sent);
	}
	last_send_time_us_ = clock_->TimeInMicroseconds();
	}

	void PacedSender::UpdateBudgetWithElapsedTime(int64_t delta_time_ms) {
	delta_time_ms = std::min(kMaxIntervalTimeMs, delta_time_ms);
	media_budget_.IncreaseBudget(delta_time_ms);
	padding_budget_.IncreaseBudget(delta_time_ms);
	}

	void PacedSender::UpdateBudgetWithBytesSent(size_t bytes_sent) {
	outstanding_bytes_ += bytes_sent;
	media_budget_.UseBudget(bytes_sent);
	padding_budget_.UseBudget(bytes_sent);
	}

	void PacedSender::SetQueueTimeLimit(int limit_ms) {
	rtc::CritScope cs(&critsect_);
	queue_time_limit = limit_ms;
	}

	} // namespace webrtc