/* * 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 #include #include #include #include "absl/strings/match.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/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); // Upper cap on process interval, in case process has not been called in a long // time. Applies only to periodic mode. constexpr TimeDelta kMaxProcessingInterval = TimeDelta::Millis(30); // Allow probes to be processed slightly ahead of inteded send time. Currently // set to 1ms as this is intended to allow times be rounded down to the nearest // millisecond. constexpr TimeDelta kMaxEarlyProbeProcessing = TimeDelta::Millis(1); constexpr int kFirstPriority = 0; bool IsDisabled(const WebRtcKeyValueConfig& field_trials, absl::string_view key) { return absl::StartsWith(field_trials.Lookup(key), "Disabled"); } bool IsEnabled(const WebRtcKeyValueConfig& field_trials, absl::string_view key) { return absl::StartsWith(field_trials.Lookup(key), "Enabled"); } TimeDelta GetDynamicPaddingTarget(const WebRtcKeyValueConfig& field_trials) { FieldTrialParameter padding_target("timedelta", TimeDelta::Millis(5)); ParseFieldTrial({&padding_target}, field_trials.Lookup("WebRTC-Pacer-DynamicPaddingTarget")); return padding_target.Get(); } int GetPriorityForType(RtpPacketMediaType type) { // Lower number takes priority over higher. switch (type) { case RtpPacketMediaType::kAudio: // Audio is always prioritized over other packet types. return kFirstPriority + 1; case RtpPacketMediaType::kRetransmission: // Send retransmissions before new media. return kFirstPriority + 2; case RtpPacketMediaType::kVideo: case RtpPacketMediaType::kForwardErrorCorrection: // Video has "normal" priority, in the old speak. // Send redundancy concurrently to video. If it is delayed it might have a // lower chance of being useful. return kFirstPriority + 3; case RtpPacketMediaType::kPadding: // Packets that are in themselves likely useless, only sent to keep the // BWE high. return kFirstPriority + 4; } RTC_CHECK_NOTREACHED(); } } // 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); PacingController::PacingController(Clock* clock, PacketSender* packet_sender, RtcEventLog* event_log, const WebRtcKeyValueConfig* field_trials, ProcessMode mode) : mode_(mode), clock_(clock), packet_sender_(packet_sender), fallback_field_trials_( !field_trials ? std::make_unique() : 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_(IsEnabled(*field_trials_, "WebRTC-Pacer-BlockAudio")), ignore_transport_overhead_( IsEnabled(*field_trials_, "WebRTC-Pacer-IgnoreTransportOverhead")), padding_target_duration_(GetDynamicPaddingTarget(*field_trials_)), min_packet_limit_(kDefaultMinPacketLimit), transport_overhead_per_packet_(DataSize::Zero()), last_timestamp_(clock_->CurrentTime()), paused_(false), media_budget_(0), padding_budget_(0), media_debt_(DataSize::Zero()), padding_debt_(DataSize::Zero()), media_rate_(DataRate::Zero()), padding_rate_(DataRate::Zero()), prober_(*field_trials_), probing_send_failure_(false), pacing_bitrate_(DataRate::Zero()), last_process_time_(clock->CurrentTime()), last_send_time_(last_process_time_), packet_queue_(last_process_time_, field_trials_), packet_counter_(0), congestion_window_size_(DataSize::PlusInfinity()), outstanding_data_(DataSize::Zero()), 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 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(bitrate, CurrentTime(), 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) { const bool was_congested = Congested(); congestion_window_size_ = congestion_window_size; if (was_congested && !Congested()) { TimeDelta elapsed_time = UpdateTimeAndGetElapsed(CurrentTime()); UpdateBudgetWithElapsedTime(elapsed_time); } } void PacingController::UpdateOutstandingData(DataSize outstanding_data) { const bool was_congested = Congested(); outstanding_data_ = outstanding_data; if (was_congested && !Congested()) { TimeDelta elapsed_time = UpdateTimeAndGetElapsed(CurrentTime()); UpdateBudgetWithElapsedTime(elapsed_time); } } bool PacingController::Congested() const { if (congestion_window_size_.IsFinite()) { return outstanding_data_ >= congestion_window_size_; } return false; } 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_EQ(0, packet_counter_); prober_.SetEnabled(enabled); } void PacingController::SetPacingRates(DataRate pacing_rate, DataRate padding_rate) { RTC_DCHECK_GT(pacing_rate, DataRate::Zero()); media_rate_ = pacing_rate; padding_rate_ = padding_rate; 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::EnqueuePacket(std::unique_ptr packet) { RTC_DCHECK(pacing_bitrate_ > DataRate::Zero()) << "SetPacingRate must be called before InsertPacket."; RTC_CHECK(packet->packet_type()); // Get priority first and store in temporary, to avoid chance of object being // moved before GetPriorityForType() being called. const int priority = GetPriorityForType(*packet->packet_type()); EnqueuePacketInternal(std::move(packet), priority); } void PacingController::SetAccountForAudioPackets(bool account_for_audio) { account_for_audio_ = account_for_audio; } void PacingController::SetIncludeOverhead() { include_overhead_ = true; packet_queue_.SetIncludeOverhead(); } void PacingController::SetTransportOverhead(DataSize overhead_per_packet) { if (ignore_transport_overhead_) return; transport_overhead_per_packet_ = overhead_per_packet; packet_queue_.SetTransportOverhead(overhead_per_packet); } TimeDelta PacingController::ExpectedQueueTime() const { RTC_DCHECK_GT(pacing_bitrate_, DataRate::Zero()); return TimeDelta::Millis( (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(); } DataSize PacingController::CurrentBufferLevel() const { return std::max(media_debt_, padding_debt_); } absl::optional 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; } void PacingController::EnqueuePacketInternal( std::unique_ptr packet, int priority) { prober_.OnIncomingPacket(DataSize::Bytes(packet->payload_size())); Timestamp now = CurrentTime(); if (mode_ == ProcessMode::kDynamic && packet_queue_.Empty() && NextSendTime() <= now) { TimeDelta elapsed_time = UpdateTimeAndGetElapsed(now); UpdateBudgetWithElapsedTime(elapsed_time); } packet_queue_.Push(priority, now, packet_counter_++, std::move(packet)); } 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(); } RTC_DCHECK_GE(now, last_process_time_); 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() || packet_counter_ == 0) { // 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) { return true; } } return false; } Timestamp PacingController::NextSendTime() const { const Timestamp now = CurrentTime(); if (paused_) { return last_send_time_ + kPausedProcessInterval; } // If probing is active, that always takes priority. if (prober_.is_probing()) { Timestamp probe_time = prober_.NextProbeTime(now); // |probe_time| == PlusInfinity indicates no probe scheduled. if (probe_time != Timestamp::PlusInfinity() && !probing_send_failure_) { return probe_time; } } if (mode_ == ProcessMode::kPeriodic) { // In periodic non-probing mode, we just have a fixed interval. return last_process_time_ + min_packet_limit_; } // In dynamic mode, figure out when the next packet should be sent, // given the current conditions. if (!pace_audio_) { // Not pacing audio, if leading packet is audio its target send // time is the time at which it was enqueued. absl::optional audio_enqueue_time = packet_queue_.LeadingAudioPacketEnqueueTime(); if (audio_enqueue_time.has_value()) { return *audio_enqueue_time; } } if (Congested() || packet_counter_ == 0) { // We need to at least send keep-alive packets with some interval. return last_send_time_ + kCongestedPacketInterval; } // Check how long until we can send the next media packet. if (media_rate_ > DataRate::Zero() && !packet_queue_.Empty()) { return std::min(last_send_time_ + kPausedProcessInterval, last_process_time_ + media_debt_ / media_rate_); } // 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. if (padding_rate_ > DataRate::Zero() && packet_queue_.Empty()) { TimeDelta drain_time = std::max(media_debt_ / media_rate_, padding_debt_ / padding_rate_); return std::min(last_send_time_ + kPausedProcessInterval, last_process_time_ + drain_time); } if (send_padding_if_silent_) { return last_send_time_ + kPausedProcessInterval; } return last_process_time_ + kPausedProcessInterval; } void PacingController::ProcessPackets() { Timestamp now = CurrentTime(); Timestamp target_send_time = now; if (mode_ == ProcessMode::kDynamic) { target_send_time = NextSendTime(); TimeDelta early_execute_margin = prober_.is_probing() ? kMaxEarlyProbeProcessing : TimeDelta::Zero(); if (target_send_time.IsMinusInfinity()) { target_send_time = now; } else if (now < target_send_time - early_execute_margin) { // We are too early, but if queue is empty still allow draining some debt. // Probing is allowed to be sent up to kMinSleepTime early. TimeDelta elapsed_time = UpdateTimeAndGetElapsed(now); UpdateBudgetWithElapsedTime(elapsed_time); return; } if (target_send_time < last_process_time_) { // After the last process call, at time X, the target send time // shifted to be earlier than X. This should normally not happen // but we want to make sure rounding errors or erratic behavior // of NextSendTime() does not cause issue. In particular, if the // buffer reduction of // rate * (target_send_time - previous_process_time) // in the main loop doesn't clean up the existing debt we may not // be able to send again. We don't want to check this reordering // there as it is the normal exit condtion when the buffer is // exhausted and there are packets in the queue. UpdateBudgetWithElapsedTime(last_process_time_ - target_send_time); target_send_time = last_process_time_; } } Timestamp previous_process_time = last_process_time_; TimeDelta elapsed_time = UpdateTimeAndGetElapsed(now); if (ShouldSendKeepalive(now)) { // We can not send padding unless a normal packet has first been sent. If // we do, timestamps get messed up. if (packet_counter_ == 0) { last_send_time_ = now; } else { DataSize keepalive_data_sent = DataSize::Zero(); std::vector> 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)); } } 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(now); if (drain_large_queues_) { 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 > target_rate) { target_rate = min_rate_needed; RTC_LOG(LS_VERBOSE) << "bwe:large_pacing_queue pacing_rate_kbps=" << target_rate.kbps(); } } } if (mode_ == ProcessMode::kPeriodic) { // In periodic processing mode, the IntevalBudget allows positive budget // up to (process interval duration) * (target rate), so we only need to // update it once before the packet sending loop. media_budget_.set_target_rate_kbps(target_rate.kbps()); UpdateBudgetWithElapsedTime(elapsed_time); } else { media_rate_ = target_rate; } } bool first_packet_in_probe = false; 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) { first_packet_in_probe = pacing_info.probe_cluster_bytes_sent == 0; 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(); // 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_) { if (first_packet_in_probe) { // If first packet in probe, insert a small padding packet so we have a // more reliable start window for the rate estimation. 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()) { // Insert with high priority so larger media packets don't preempt it. EnqueuePacketInternal(std::move(padding[0]), kFirstPriority); // We should never get more than one padding packets with a requested // size of 1 byte. RTC_DCHECK_EQ(padding.size(), 1u); } first_packet_in_probe = false; } if (mode_ == ProcessMode::kDynamic && previous_process_time < target_send_time) { // Reduce buffer levels with amount corresponding to time between last // process and target send time for the next packet. // If the process call is late, that may be the time between the optimal // send times for two packets we should already have sent. UpdateBudgetWithElapsedTime(target_send_time - previous_process_time); previous_process_time = target_send_time; } // Fetch the next packet, so long as queue is not empty or budget is not // exhausted. std::unique_ptr 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> 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; } 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; // Send done, update send/process time to the target send time. OnPacketSent(packet_type, packet_size, target_send_time); // If we are currently probing, we need to stop the send loop when we have // reached the send target. if (is_probing && data_sent >= recommended_probe_size) { break; } if (mode_ == ProcessMode::kDynamic) { // Update target send time in case that are more packets that we are late // in processing. Timestamp next_send_time = NextSendTime(); if (next_send_time.IsMinusInfinity()) { target_send_time = now; } else { target_send_time = std::min(now, next_send_time); } } } last_process_time_ = std::max(last_process_time_, previous_process_time); if (is_probing) { probing_send_failure_ = data_sent == DataSize::Zero(); if (!probing_send_failure_) { prober_.ProbeSent(CurrentTime(), data_sent); } } } 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 (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.IsZero()) { if (recommended_probe_size > data_sent) { return recommended_probe_size - data_sent; } return DataSize::Zero(); } if (mode_ == ProcessMode::kPeriodic) { return DataSize::Bytes(padding_budget_.bytes_remaining()); } else if (padding_rate_ > DataRate::Zero() && padding_debt_ == DataSize::Zero()) { return padding_target_duration_ * padding_rate_; } return DataSize::Zero(); } std::unique_ptr PacingController::GetPendingPacket( const PacedPacketInfo& pacing_info, Timestamp target_send_time, Timestamp now) { 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().has_value(); bool is_probe = pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe; if (!unpaced_audio_packet && !is_probe) { if (Congested()) { // Don't send anything if congested. return nullptr; } if (mode_ == ProcessMode::kPeriodic) { if (media_budget_.bytes_remaining() <= 0) { // Not enough budget. return nullptr; } } else { // Dynamic processing mode. if (now <= target_send_time) { // 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. TimeDelta flush_time = media_debt_ / 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_) { first_sent_packet_time_ = send_time; } bool audio_packet = packet_type == RtpPacketMediaType::kAudio; if (!audio_packet || account_for_audio_) { // Update media bytes sent. UpdateBudgetWithSentData(packet_size); } last_send_time_ = send_time; last_process_time_ = send_time; } void PacingController::OnPaddingSent(DataSize data_sent) { if (data_sent > DataSize::Zero()) { UpdateBudgetWithSentData(data_sent); } Timestamp now = CurrentTime(); last_send_time_ = now; last_process_time_ = now; } void PacingController::UpdateBudgetWithElapsedTime(TimeDelta delta) { if (mode_ == ProcessMode::kPeriodic) { delta = std::min(kMaxProcessingInterval, delta); media_budget_.IncreaseBudget(delta.ms()); padding_budget_.IncreaseBudget(delta.ms()); } else { media_debt_ -= std::min(media_debt_, media_rate_ * delta); padding_debt_ -= std::min(padding_debt_, padding_rate_ * delta); } } void PacingController::UpdateBudgetWithSentData(DataSize size) { outstanding_data_ += size; if (mode_ == ProcessMode::kPeriodic) { media_budget_.UseBudget(size.bytes()); padding_budget_.UseBudget(size.bytes()); } else { media_debt_ += size; media_debt_ = std::min(media_debt_, media_rate_ * kMaxDebtInTime); padding_debt_ += size; padding_debt_ = std::min(padding_debt_, padding_rate_ * kMaxDebtInTime); } } void PacingController::SetQueueTimeLimit(TimeDelta limit) { queue_time_limit = limit; } } // namespace webrtc