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/*
* 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/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);
} // 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),
drain_large_queues_(configuration.drain_large_queues &&
!field_trials.IsDisabled("WebRTC-Pacer-DrainQueue")),
send_padding_if_silent_(
field_trials.IsEnabled("WebRTC-Pacer-PadInSilence")),
pace_audio_(field_trials.IsEnabled("WebRTC-Pacer-BlockAudio")),
ignore_transport_overhead_(
field_trials.IsEnabled("WebRTC-Pacer-IgnoreTransportOverhead")),
fast_retransmissions_(
field_trials.IsEnabled("WebRTC-Pacer-FastRetransmissions")),
keyframe_flushing_(
configuration.keyframe_flushing ||
field_trials.IsEnabled("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(
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->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 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& fec_packet : packet_sender_->FetchFec()) {
EnqueuePacket(std::move(fec_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
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