| /* |
| * 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 <array> |
| #include <cstddef> |
| #include <cstdint> |
| #include <memory> |
| #include <optional> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/cleanup/cleanup.h" |
| #include "absl/strings/match.h" |
| #include "absl/strings/string_view.h" |
| #include "api/array_view.h" |
| #include "api/field_trials_view.h" |
| #include "api/transport/network_types.h" |
| #include "api/units/data_rate.h" |
| #include "api/units/data_size.h" |
| #include "api/units/time_delta.h" |
| #include "api/units/timestamp.h" |
| #include "modules/pacing/bitrate_prober.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "system_wrappers/include/clock.h" |
| |
| namespace webrtc { |
| namespace { |
| 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); |
| |
| bool IsDisabled(const FieldTrialsView& field_trials, absl::string_view key) { |
| return absl::StartsWith(field_trials.Lookup(key), "Disabled"); |
| } |
| |
| bool IsEnabled(const FieldTrialsView& field_trials, absl::string_view key) { |
| return absl::StartsWith(field_trials.Lookup(key), "Enabled"); |
| } |
| |
| } // namespace |
| |
| const TimeDelta PacingController::kPausedProcessInterval = |
| kCongestedPacketInterval; |
| const TimeDelta PacingController::kMinSleepTime = TimeDelta::Millis(1); |
| const TimeDelta PacingController::kTargetPaddingDuration = TimeDelta::Millis(5); |
| const TimeDelta PacingController::kMaxPaddingReplayDuration = |
| TimeDelta::Millis(50); |
| const TimeDelta PacingController::kMaxEarlyProbeProcessing = |
| TimeDelta::Millis(1); |
| |
| PacingController::PacingController(Clock* clock, |
| PacketSender* packet_sender, |
| const FieldTrialsView& field_trials, |
| Configuration configuration) |
| : clock_(clock), |
| packet_sender_(packet_sender), |
| field_trials_(field_trials), |
| drain_large_queues_( |
| configuration.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")), |
| fast_retransmissions_( |
| IsEnabled(field_trials_, "WebRTC-Pacer-FastRetransmissions")), |
| keyframe_flushing_( |
| configuration.keyframe_flushing || |
| IsEnabled(field_trials_, "WebRTC-Pacer-KeyframeFlushing")), |
| transport_overhead_per_packet_(DataSize::Zero()), |
| send_burst_interval_(configuration.send_burst_interval), |
| last_timestamp_(clock_->CurrentTime()), |
| paused_(false), |
| media_debt_(DataSize::Zero()), |
| padding_debt_(DataSize::Zero()), |
| pacing_rate_(DataRate::Zero()), |
| adjusted_media_rate_(DataRate::Zero()), |
| padding_rate_(DataRate::Zero()), |
| prober_(field_trials_), |
| probing_send_failure_(false), |
| last_process_time_(clock->CurrentTime()), |
| last_send_time_(last_process_time_), |
| seen_first_packet_(false), |
| packet_queue_(/*creation_time=*/last_process_time_, |
| configuration.prioritize_audio_retransmission, |
| configuration.packet_queue_ttl), |
| congested_(false), |
| queue_time_limit_(configuration.queue_time_limit), |
| account_for_audio_(false), |
| include_overhead_(false), |
| circuit_breaker_threshold_(1 << 16) { |
| if (!drain_large_queues_) { |
| RTC_LOG(LS_WARNING) << "Pacer queues will not be drained," |
| "pushback experiment must be enabled."; |
| } |
| } |
| |
| PacingController::~PacingController() = default; |
| |
| void PacingController::CreateProbeClusters( |
| rtc::ArrayView<const ProbeClusterConfig> probe_cluster_configs) { |
| for (const ProbeClusterConfig probe_cluster_config : probe_cluster_configs) { |
| prober_.CreateProbeCluster(probe_cluster_config); |
| } |
| } |
| |
| void PacingController::Pause() { |
| if (!paused_) |
| RTC_LOG(LS_INFO) << "PacedSender paused."; |
| paused_ = true; |
| packet_queue_.SetPauseState(true, CurrentTime()); |
| } |
| |
| void PacingController::Resume() { |
| if (paused_) |
| RTC_LOG(LS_INFO) << "PacedSender resumed."; |
| paused_ = false; |
| packet_queue_.SetPauseState(false, CurrentTime()); |
| } |
| |
| bool PacingController::IsPaused() const { |
| return paused_; |
| } |
| |
| void PacingController::SetCongested(bool congested) { |
| if (congested_ && !congested) { |
| UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(CurrentTime())); |
| } |
| congested_ = congested; |
| } |
| |
| void PacingController::SetCircuitBreakerThreshold(int num_iterations) { |
| circuit_breaker_threshold_ = num_iterations; |
| } |
| |
| void PacingController::RemovePacketsForSsrc(uint32_t ssrc) { |
| packet_queue_.RemovePacketsForSsrc(ssrc); |
| } |
| |
| bool PacingController::IsProbing() const { |
| return prober_.is_probing(); |
| } |
| |
| Timestamp PacingController::CurrentTime() const { |
| Timestamp time = clock_->CurrentTime(); |
| if (time < last_timestamp_) { |
| RTC_LOG(LS_WARNING) |
| << "Non-monotonic clock behavior observed. Previous timestamp: " |
| << last_timestamp_.ms() << ", new timestamp: " << time.ms(); |
| RTC_DCHECK_GE(time, last_timestamp_); |
| time = last_timestamp_; |
| } |
| last_timestamp_ = time; |
| return time; |
| } |
| |
| void PacingController::SetProbingEnabled(bool enabled) { |
| RTC_CHECK(!seen_first_packet_); |
| prober_.SetEnabled(enabled); |
| } |
| |
| void PacingController::SetPacingRates(DataRate pacing_rate, |
| DataRate padding_rate) { |
| RTC_CHECK_GT(pacing_rate, DataRate::Zero()); |
| RTC_CHECK_GE(padding_rate, DataRate::Zero()); |
| if (padding_rate > pacing_rate) { |
| RTC_LOG(LS_WARNING) << "Padding rate " << padding_rate.kbps() |
| << "kbps is higher than the pacing rate " |
| << pacing_rate.kbps() << "kbps, capping."; |
| padding_rate = pacing_rate; |
| } |
| |
| if (pacing_rate > max_rate || padding_rate > max_rate) { |
| RTC_LOG(LS_WARNING) << "Very high pacing rates ( > " << max_rate.kbps() |
| << " kbps) configured: pacing = " << pacing_rate.kbps() |
| << " kbps, padding = " << padding_rate.kbps() |
| << " kbps."; |
| max_rate = std::max(pacing_rate, padding_rate) * 1.1; |
| } |
| pacing_rate_ = pacing_rate; |
| padding_rate_ = padding_rate; |
| MaybeUpdateMediaRateDueToLongQueue(CurrentTime()); |
| |
| RTC_LOG(LS_VERBOSE) << "bwe:pacer_updated pacing_kbps=" << pacing_rate_.kbps() |
| << " padding_budget_kbps=" << padding_rate.kbps(); |
| } |
| |
| void PacingController::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) { |
| RTC_DCHECK(pacing_rate_ > DataRate::Zero()) |
| << "SetPacingRate must be called before InsertPacket."; |
| RTC_CHECK(packet->packet_type()); |
| |
| if (keyframe_flushing_ && |
| packet->packet_type() == RtpPacketMediaType::kVideo && |
| packet->is_key_frame() && packet->is_first_packet_of_frame() && |
| !packet_queue_.HasKeyframePackets(packet->Ssrc())) { |
| // First packet of a keyframe (and no keyframe packets currently in the |
| // queue). Flush any pending packets currently in the queue for that stream |
| // in order to get the new keyframe out as quickly as possible. |
| packet_queue_.RemovePacketsForSsrc(packet->Ssrc()); |
| std::optional<uint32_t> rtx_ssrc = |
| packet_sender_->GetRtxSsrcForMedia(packet->Ssrc()); |
| if (rtx_ssrc) { |
| packet_queue_.RemovePacketsForSsrc(*rtx_ssrc); |
| } |
| } |
| |
| prober_.OnIncomingPacket(DataSize::Bytes(packet->payload_size())); |
| |
| const Timestamp now = CurrentTime(); |
| if (packet_queue_.Empty()) { |
| // If queue is empty, we need to "fast-forward" the last process time, |
| // so that we don't use passed time as budget for sending the first new |
| // packet. |
| Timestamp target_process_time = now; |
| Timestamp next_send_time = NextSendTime(); |
| if (next_send_time.IsFinite()) { |
| // There was already a valid planned send time, such as a keep-alive. |
| // Use that as last process time only if it's prior to now. |
| target_process_time = std::min(now, next_send_time); |
| } |
| UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(target_process_time)); |
| } |
| packet_queue_.Push(now, std::move(packet)); |
| seen_first_packet_ = true; |
| |
| // Queue length has increased, check if we need to change the pacing rate. |
| MaybeUpdateMediaRateDueToLongQueue(now); |
| } |
| |
| void PacingController::SetAccountForAudioPackets(bool account_for_audio) { |
| account_for_audio_ = account_for_audio; |
| } |
| |
| void PacingController::SetIncludeOverhead() { |
| include_overhead_ = true; |
| } |
| |
| void PacingController::SetTransportOverhead(DataSize overhead_per_packet) { |
| if (ignore_transport_overhead_) |
| return; |
| transport_overhead_per_packet_ = overhead_per_packet; |
| } |
| |
| void PacingController::SetSendBurstInterval(TimeDelta burst_interval) { |
| send_burst_interval_ = burst_interval; |
| } |
| |
| void PacingController::SetAllowProbeWithoutMediaPacket(bool allow) { |
| prober_.SetAllowProbeWithoutMediaPacket(allow); |
| } |
| |
| TimeDelta PacingController::ExpectedQueueTime() const { |
| RTC_DCHECK_GT(adjusted_media_rate_, DataRate::Zero()); |
| return QueueSizeData() / adjusted_media_rate_; |
| } |
| |
| size_t PacingController::QueueSizePackets() const { |
| return rtc::checked_cast<size_t>(packet_queue_.SizeInPackets()); |
| } |
| |
| const std::array<int, kNumMediaTypes>& |
| PacingController::SizeInPacketsPerRtpPacketMediaType() const { |
| return packet_queue_.SizeInPacketsPerRtpPacketMediaType(); |
| } |
| |
| DataSize PacingController::QueueSizeData() const { |
| DataSize size = packet_queue_.SizeInPayloadBytes(); |
| if (include_overhead_) { |
| size += static_cast<int64_t>(packet_queue_.SizeInPackets()) * |
| transport_overhead_per_packet_; |
| } |
| return size; |
| } |
| |
| DataSize PacingController::CurrentBufferLevel() const { |
| return std::max(media_debt_, padding_debt_); |
| } |
| |
| std::optional<Timestamp> PacingController::FirstSentPacketTime() const { |
| return first_sent_packet_time_; |
| } |
| |
| Timestamp PacingController::OldestPacketEnqueueTime() const { |
| return packet_queue_.OldestEnqueueTime(); |
| } |
| |
| TimeDelta PacingController::UpdateTimeAndGetElapsed(Timestamp now) { |
| // If no previous processing, or last process was "in the future" because of |
| // early probe processing, then there is no elapsed time to add budget for. |
| if (last_process_time_.IsMinusInfinity() || now < last_process_time_) { |
| return TimeDelta::Zero(); |
| } |
| TimeDelta elapsed_time = now - last_process_time_; |
| last_process_time_ = now; |
| if (elapsed_time > kMaxElapsedTime) { |
| RTC_LOG(LS_WARNING) << "Elapsed time (" << elapsed_time |
| << ") longer than expected, limiting to " |
| << kMaxElapsedTime; |
| elapsed_time = kMaxElapsedTime; |
| } |
| return elapsed_time; |
| } |
| |
| bool PacingController::ShouldSendKeepalive(Timestamp now) const { |
| if (send_padding_if_silent_ || paused_ || congested_ || !seen_first_packet_) { |
| // We send a padding packet every 500 ms to ensure we won't get stuck in |
| // congested state due to no feedback being received. |
| if (now - last_send_time_ >= kCongestedPacketInterval) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| Timestamp PacingController::NextSendTime() const { |
| const Timestamp now = CurrentTime(); |
| Timestamp next_send_time = Timestamp::PlusInfinity(); |
| |
| if (paused_) { |
| return last_send_time_ + kPausedProcessInterval; |
| } |
| |
| // If probing is active, that always takes priority. |
| if (prober_.is_probing() && !probing_send_failure_) { |
| Timestamp probe_time = prober_.NextProbeTime(now); |
| if (!probe_time.IsPlusInfinity()) { |
| return probe_time.IsMinusInfinity() ? now : probe_time; |
| } |
| } |
| |
| // If queue contains a packet which should not be paced, its target send time |
| // is the time at which it was enqueued. |
| Timestamp unpaced_send_time = NextUnpacedSendTime(); |
| if (unpaced_send_time.IsFinite()) { |
| return unpaced_send_time; |
| } |
| |
| if (congested_ || !seen_first_packet_) { |
| // We need to at least send keep-alive packets with some interval. |
| return last_send_time_ + kCongestedPacketInterval; |
| } |
| |
| if (adjusted_media_rate_ > DataRate::Zero() && !packet_queue_.Empty()) { |
| // If packets are allowed to be sent in a burst, the |
| // debt is allowed to grow up to one packet more than what can be sent |
| // during 'send_burst_period_'. |
| TimeDelta drain_time = media_debt_ / adjusted_media_rate_; |
| // Ensure that a burst of sent packet is not larger than kMaxBurstSize in |
| // order to not risk overfilling socket buffers at high bitrate. |
| TimeDelta send_burst_interval = |
| std::min(send_burst_interval_, kMaxBurstSize / adjusted_media_rate_); |
| next_send_time = |
| last_process_time_ + |
| ((send_burst_interval > drain_time) ? TimeDelta::Zero() : drain_time); |
| } else if (padding_rate_ > DataRate::Zero() && packet_queue_.Empty()) { |
| // If we _don't_ have pending packets, check how long until we have |
| // bandwidth for padding packets. Both media and padding debts must |
| // have been drained to do this. |
| RTC_DCHECK_GT(adjusted_media_rate_, DataRate::Zero()); |
| TimeDelta drain_time = std::max(media_debt_ / adjusted_media_rate_, |
| padding_debt_ / padding_rate_); |
| |
| if (drain_time.IsZero() && |
| (!media_debt_.IsZero() || !padding_debt_.IsZero())) { |
| // We have a non-zero debt, but drain time is smaller than tick size of |
| // TimeDelta, round it up to the smallest possible non-zero delta. |
| drain_time = TimeDelta::Micros(1); |
| } |
| next_send_time = last_process_time_ + drain_time; |
| } else { |
| // Nothing to do. |
| next_send_time = last_process_time_ + kPausedProcessInterval; |
| } |
| |
| if (send_padding_if_silent_) { |
| next_send_time = |
| std::min(next_send_time, last_send_time_ + kPausedProcessInterval); |
| } |
| |
| return next_send_time; |
| } |
| |
| void PacingController::ProcessPackets() { |
| absl::Cleanup cleanup = [packet_sender = packet_sender_] { |
| packet_sender->OnBatchComplete(); |
| }; |
| const Timestamp now = CurrentTime(); |
| Timestamp target_send_time = now; |
| |
| if (ShouldSendKeepalive(now)) { |
| DataSize keepalive_data_sent = DataSize::Zero(); |
| // We can not send padding unless a normal packet has first been sent. If |
| // we do, timestamps get messed up. |
| if (seen_first_packet_) { |
| std::vector<std::unique_ptr<RtpPacketToSend>> keepalive_packets = |
| packet_sender_->GeneratePadding(DataSize::Bytes(1)); |
| for (auto& packet : keepalive_packets) { |
| keepalive_data_sent += |
| DataSize::Bytes(packet->payload_size() + packet->padding_size()); |
| packet_sender_->SendPacket(std::move(packet), PacedPacketInfo()); |
| for (auto& packet : packet_sender_->FetchFec()) { |
| EnqueuePacket(std::move(packet)); |
| } |
| } |
| } |
| OnPacketSent(RtpPacketMediaType::kPadding, keepalive_data_sent, now); |
| } |
| |
| if (paused_) { |
| return; |
| } |
| |
| TimeDelta early_execute_margin = |
| prober_.is_probing() ? kMaxEarlyProbeProcessing : TimeDelta::Zero(); |
| |
| target_send_time = NextSendTime(); |
| if (now + early_execute_margin < target_send_time) { |
| // We are too early, but if queue is empty still allow draining some debt. |
| // Probing is allowed to be sent up to kMinSleepTime early. |
| UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(now)); |
| return; |
| } |
| |
| TimeDelta elapsed_time = UpdateTimeAndGetElapsed(target_send_time); |
| |
| if (elapsed_time > TimeDelta::Zero()) { |
| UpdateBudgetWithElapsedTime(elapsed_time); |
| } |
| |
| PacedPacketInfo pacing_info; |
| DataSize recommended_probe_size = DataSize::Zero(); |
| bool is_probing = prober_.is_probing(); |
| if (is_probing) { |
| // Probe timing is sensitive, and handled explicitly by BitrateProber, so |
| // use actual send time rather than target. |
| pacing_info = prober_.CurrentCluster(now).value_or(PacedPacketInfo()); |
| if (pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe) { |
| recommended_probe_size = prober_.RecommendedMinProbeSize(); |
| RTC_DCHECK_GT(recommended_probe_size, DataSize::Zero()); |
| } else { |
| // No valid probe cluster returned, probe might have timed out. |
| is_probing = false; |
| } |
| } |
| |
| DataSize data_sent = DataSize::Zero(); |
| int iteration = 0; |
| int packets_sent = 0; |
| int padding_packets_generated = 0; |
| for (; iteration < circuit_breaker_threshold_; ++iteration) { |
| // Fetch packet, so long as queue is not empty or budget is not |
| // exhausted. |
| std::unique_ptr<RtpPacketToSend> rtp_packet = |
| GetPendingPacket(pacing_info, target_send_time, now); |
| if (rtp_packet == nullptr) { |
| // No packet available to send, check if we should send padding. |
| if (now - target_send_time > kMaxPaddingReplayDuration) { |
| // The target send time is more than `kMaxPaddingReplayDuration` behind |
| // the real-time clock. This can happen if the clock is adjusted forward |
| // without `ProcessPackets()` having been called at the expected times. |
| target_send_time = now - kMaxPaddingReplayDuration; |
| last_process_time_ = std::max(last_process_time_, target_send_time); |
| } |
| |
| DataSize padding_to_add = PaddingToAdd(recommended_probe_size, data_sent); |
| if (padding_to_add > DataSize::Zero()) { |
| std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets = |
| packet_sender_->GeneratePadding(padding_to_add); |
| if (!padding_packets.empty()) { |
| padding_packets_generated += padding_packets.size(); |
| for (auto& packet : padding_packets) { |
| EnqueuePacket(std::move(packet)); |
| } |
| // Continue loop to send the padding that was just added. |
| continue; |
| } else { |
| // Can't generate padding, still update padding budget for next send |
| // time. |
| UpdatePaddingBudgetWithSentData(padding_to_add); |
| } |
| } |
| // Can't fetch new packet and no padding to send, exit send loop. |
| break; |
| } else { |
| RTC_DCHECK(rtp_packet); |
| RTC_DCHECK(rtp_packet->packet_type().has_value()); |
| const RtpPacketMediaType packet_type = *rtp_packet->packet_type(); |
| DataSize packet_size = DataSize::Bytes(rtp_packet->payload_size() + |
| rtp_packet->padding_size()); |
| |
| if (include_overhead_) { |
| packet_size += DataSize::Bytes(rtp_packet->headers_size()) + |
| transport_overhead_per_packet_; |
| } |
| |
| packet_sender_->SendPacket(std::move(rtp_packet), pacing_info); |
| for (auto& packet : packet_sender_->FetchFec()) { |
| EnqueuePacket(std::move(packet)); |
| } |
| data_sent += packet_size; |
| ++packets_sent; |
| |
| // Send done, update send time. |
| OnPacketSent(packet_type, packet_size, now); |
| |
| if (is_probing) { |
| pacing_info.probe_cluster_bytes_sent += packet_size.bytes(); |
| // If we are currently probing, we need to stop the send loop when we |
| // have reached the send target. |
| if (data_sent >= recommended_probe_size) { |
| break; |
| } |
| } |
| |
| // Update target send time in case that are more packets that we are late |
| // in processing. |
| target_send_time = NextSendTime(); |
| if (target_send_time > now) { |
| // Exit loop if not probing. |
| if (!is_probing) { |
| break; |
| } |
| target_send_time = now; |
| } |
| UpdateBudgetWithElapsedTime(UpdateTimeAndGetElapsed(target_send_time)); |
| } |
| } |
| |
| if (iteration >= circuit_breaker_threshold_) { |
| // Circuit break activated. Log warning, adjust send time and return. |
| // TODO(sprang): Consider completely clearing state. |
| RTC_LOG(LS_ERROR) |
| << "PacingController exceeded max iterations in " |
| "send-loop. Debug info: " |
| << " packets sent = " << packets_sent |
| << ", padding packets generated = " << padding_packets_generated |
| << ", bytes sent = " << data_sent.bytes() |
| << ", probing = " << (is_probing ? "true" : "false") |
| << ", recommended_probe_size = " << recommended_probe_size.bytes() |
| << ", now = " << now.us() |
| << ", target_send_time = " << target_send_time.us() |
| << ", last_process_time = " << last_process_time_.us() |
| << ", last_send_time = " << last_send_time_.us() |
| << ", paused = " << (paused_ ? "true" : "false") |
| << ", media_debt = " << media_debt_.bytes() |
| << ", padding_debt = " << padding_debt_.bytes() |
| << ", pacing_rate = " << pacing_rate_.bps() |
| << ", adjusted_media_rate = " << adjusted_media_rate_.bps() |
| << ", padding_rate = " << padding_rate_.bps() |
| << ", queue size (packets) = " << packet_queue_.SizeInPackets() |
| << ", queue size (payload bytes) = " |
| << packet_queue_.SizeInPayloadBytes(); |
| last_send_time_ = now; |
| last_process_time_ = now; |
| return; |
| } |
| |
| if (is_probing) { |
| probing_send_failure_ = data_sent == DataSize::Zero(); |
| if (!probing_send_failure_) { |
| prober_.ProbeSent(CurrentTime(), data_sent); |
| } |
| } |
| |
| // Queue length has probably decreased, check if pacing rate needs to updated. |
| // Poll the time again, since we might have enqueued new fec/padding packets |
| // with a later timestamp than `now`. |
| MaybeUpdateMediaRateDueToLongQueue(CurrentTime()); |
| } |
| |
| DataSize PacingController::PaddingToAdd(DataSize recommended_probe_size, |
| DataSize data_sent) const { |
| if (!packet_queue_.Empty()) { |
| // Actual payload available, no need to add padding. |
| return DataSize::Zero(); |
| } |
| |
| if (congested_) { |
| // Don't add padding if congested, even if requested for probing. |
| return DataSize::Zero(); |
| } |
| |
| if (!recommended_probe_size.IsZero()) { |
| if (recommended_probe_size > data_sent) { |
| return recommended_probe_size - data_sent; |
| } |
| return DataSize::Zero(); |
| } |
| |
| if (padding_rate_ > DataRate::Zero() && padding_debt_ == DataSize::Zero()) { |
| return kTargetPaddingDuration * padding_rate_; |
| } |
| return DataSize::Zero(); |
| } |
| |
| std::unique_ptr<RtpPacketToSend> PacingController::GetPendingPacket( |
| const PacedPacketInfo& pacing_info, |
| Timestamp target_send_time, |
| Timestamp now) { |
| const bool is_probe = |
| pacing_info.probe_cluster_id != PacedPacketInfo::kNotAProbe; |
| // If first packet in probe, insert a small padding packet so we have a |
| // more reliable start window for the rate estimation. |
| if (is_probe && pacing_info.probe_cluster_bytes_sent == 0) { |
| auto padding = packet_sender_->GeneratePadding(DataSize::Bytes(1)); |
| // If no RTP modules sending media are registered, we may not get a |
| // padding packet back. |
| if (!padding.empty()) { |
| // We should never get more than one padding packets with a requested |
| // size of 1 byte. |
| RTC_DCHECK_EQ(padding.size(), 1u); |
| return std::move(padding[0]); |
| } |
| } |
| |
| if (packet_queue_.Empty()) { |
| return nullptr; |
| } |
| |
| // First, check if there is any reason _not_ to send the next queued packet. |
| // Unpaced packets and probes are exempted from send checks. |
| if (NextUnpacedSendTime().IsInfinite() && !is_probe) { |
| if (congested_) { |
| // Don't send anything if congested. |
| return nullptr; |
| } |
| |
| if (now <= target_send_time && send_burst_interval_.IsZero()) { |
| // We allow sending slightly early if we think that we would actually |
| // had been able to, had we been right on time - i.e. the current debt |
| // is not more than would be reduced to zero at the target sent time. |
| // If we allow packets to be sent in a burst, packet are allowed to be |
| // sent early. |
| TimeDelta flush_time = media_debt_ / adjusted_media_rate_; |
| if (now + flush_time > target_send_time) { |
| return nullptr; |
| } |
| } |
| } |
| |
| return packet_queue_.Pop(); |
| } |
| |
| void PacingController::OnPacketSent(RtpPacketMediaType packet_type, |
| DataSize packet_size, |
| Timestamp send_time) { |
| if (!first_sent_packet_time_ && packet_type != RtpPacketMediaType::kPadding) { |
| first_sent_packet_time_ = send_time; |
| } |
| |
| bool audio_packet = packet_type == RtpPacketMediaType::kAudio; |
| if ((!audio_packet || account_for_audio_) && packet_size > DataSize::Zero()) { |
| UpdateBudgetWithSentData(packet_size); |
| } |
| |
| last_send_time_ = send_time; |
| } |
| |
| void PacingController::UpdateBudgetWithElapsedTime(TimeDelta delta) { |
| media_debt_ -= std::min(media_debt_, adjusted_media_rate_ * delta); |
| padding_debt_ -= std::min(padding_debt_, padding_rate_ * delta); |
| } |
| |
| void PacingController::UpdateBudgetWithSentData(DataSize size) { |
| media_debt_ += size; |
| media_debt_ = std::min(media_debt_, adjusted_media_rate_ * kMaxDebtInTime); |
| UpdatePaddingBudgetWithSentData(size); |
| } |
| |
| void PacingController::UpdatePaddingBudgetWithSentData(DataSize size) { |
| padding_debt_ += size; |
| padding_debt_ = std::min(padding_debt_, padding_rate_ * kMaxDebtInTime); |
| } |
| |
| void PacingController::SetQueueTimeLimit(TimeDelta limit) { |
| queue_time_limit_ = limit; |
| } |
| |
| void PacingController::MaybeUpdateMediaRateDueToLongQueue(Timestamp now) { |
| adjusted_media_rate_ = pacing_rate_; |
| if (!drain_large_queues_) { |
| return; |
| } |
| |
| DataSize queue_size_data = QueueSizeData(); |
| if (queue_size_data > DataSize::Zero()) { |
| // Assuming equal size packets and input/output rate, the average packet |
| // has avg_time_left_ms left to get queue_size_bytes out of the queue, if |
| // time constraint shall be met. Determine bitrate needed for that. |
| packet_queue_.UpdateAverageQueueTime(now); |
| TimeDelta avg_time_left = |
| std::max(TimeDelta::Millis(1), |
| queue_time_limit_ - packet_queue_.AverageQueueTime()); |
| DataRate min_rate_needed = queue_size_data / avg_time_left; |
| if (min_rate_needed > pacing_rate_) { |
| adjusted_media_rate_ = min_rate_needed; |
| RTC_LOG(LS_VERBOSE) << "bwe:large_pacing_queue pacing_rate_kbps=" |
| << pacing_rate_.kbps(); |
| } |
| } |
| } |
| |
| Timestamp PacingController::NextUnpacedSendTime() const { |
| if (!pace_audio_) { |
| Timestamp leading_audio_send_time = |
| packet_queue_.LeadingPacketEnqueueTime(RtpPacketMediaType::kAudio); |
| if (leading_audio_send_time.IsFinite()) { |
| return leading_audio_send_time; |
| } |
| } |
| if (fast_retransmissions_) { |
| Timestamp leading_retransmission_send_time = |
| packet_queue_.LeadingPacketEnqueueTimeForRetransmission(); |
| if (leading_retransmission_send_time.IsFinite()) { |
| return leading_retransmission_send_time; |
| } |
| } |
| return Timestamp::MinusInfinity(); |
| } |
| |
| } // namespace webrtc |