blob: 3c97163234f7c992113bc592ed7d0bf624153344 [file] [log] [blame]
/*
* Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/pacing/pacing_controller.h"
#include <algorithm>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "modules/pacing/bitrate_prober.h"
#include "modules/pacing/interval_budget.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
namespace {
// Time limit in milliseconds between packet bursts.
constexpr TimeDelta kDefaultMinPacketLimit = TimeDelta::Millis<5>();
constexpr TimeDelta kCongestedPacketInterval = TimeDelta::Millis<500>();
constexpr TimeDelta kMaxElapsedTime = TimeDelta::Seconds<2>();
// Upper cap on process interval, in case process has not been called in a long
// time.
constexpr TimeDelta kMaxProcessingInterval = TimeDelta::Millis<30>();
bool IsDisabled(const WebRtcKeyValueConfig& field_trials,
absl::string_view key) {
return field_trials.Lookup(key).find("Disabled") == 0;
}
bool IsEnabled(const WebRtcKeyValueConfig& field_trials,
absl::string_view key) {
return field_trials.Lookup(key).find("Enabled") == 0;
}
int GetPriorityForType(RtpPacketToSend::Type type) {
switch (type) {
case RtpPacketToSend::Type::kAudio:
// Audio is always prioritized over other packet types.
return 0;
case RtpPacketToSend::Type::kRetransmission:
// Send retransmissions before new media.
return 1;
case RtpPacketToSend::Type::kVideo:
// Video has "normal" priority, in the old speak.
return 2;
case RtpPacketToSend::Type::kForwardErrorCorrection:
// Send redundancy concurrently to video. If it is delayed it might have a
// lower chance of being useful.
return 2;
case RtpPacketToSend::Type::kPadding:
// Packets that are in themselves likely useless, only sent to keep the
// BWE high.
return 3;
}
}
} // namespace
const TimeDelta PacingController::kMaxExpectedQueueLength =
TimeDelta::Millis<2000>();
const float PacingController::kDefaultPaceMultiplier = 2.5f;
const TimeDelta PacingController::kPausedProcessInterval =
kCongestedPacketInterval;
PacingController::PacingController(Clock* clock,
PacketSender* packet_sender,
RtcEventLog* event_log,
const WebRtcKeyValueConfig* field_trials)
: clock_(clock),
packet_sender_(packet_sender),
fallback_field_trials_(
!field_trials ? absl::make_unique<FieldTrialBasedConfig>() : nullptr),
field_trials_(field_trials ? field_trials : fallback_field_trials_.get()),
drain_large_queues_(
!IsDisabled(*field_trials_, "WebRTC-Pacer-DrainQueue")),
send_padding_if_silent_(
IsEnabled(*field_trials_, "WebRTC-Pacer-PadInSilence")),
pace_audio_(!IsDisabled(*field_trials_, "WebRTC-Pacer-BlockAudio")),
min_packet_limit_(kDefaultMinPacketLimit),
last_timestamp_(clock_->CurrentTime()),
paused_(false),
media_budget_(0),
padding_budget_(0),
prober_(*field_trials_),
probing_send_failure_(false),
padding_failure_state_(false),
pacing_bitrate_(DataRate::Zero()),
time_last_process_(clock->CurrentTime()),
last_send_time_(time_last_process_),
packet_queue_(time_last_process_, field_trials),
packet_counter_(0),
congestion_window_size_(DataSize::PlusInfinity()),
outstanding_data_(DataSize::Zero()),
queue_time_limit(kMaxExpectedQueueLength),
account_for_audio_(false) {
if (!drain_large_queues_) {
RTC_LOG(LS_WARNING) << "Pacer queues will not be drained,"
"pushback experiment must be enabled.";
}
FieldTrialParameter<int> min_packet_limit_ms("", min_packet_limit_.ms());
ParseFieldTrial({&min_packet_limit_ms},
field_trials_->Lookup("WebRTC-Pacer-MinPacketLimitMs"));
min_packet_limit_ = TimeDelta::ms(min_packet_limit_ms.Get());
UpdateBudgetWithElapsedTime(min_packet_limit_);
}
PacingController::~PacingController() = default;
void PacingController::CreateProbeCluster(DataRate bitrate, int cluster_id) {
prober_.CreateProbeCluster(bitrate.bps(), CurrentTime().ms(), cluster_id);
}
void PacingController::Pause() {
if (!paused_)
RTC_LOG(LS_INFO) << "PacedSender paused.";
paused_ = true;
packet_queue_.SetPauseState(true, CurrentTime());
}
void PacingController::Resume() {
if (paused_)
RTC_LOG(LS_INFO) << "PacedSender resumed.";
paused_ = false;
packet_queue_.SetPauseState(false, CurrentTime());
}
bool PacingController::IsPaused() const {
return paused_;
}
void PacingController::SetCongestionWindow(DataSize congestion_window_size) {
congestion_window_size_ = congestion_window_size;
}
void PacingController::UpdateOutstandingData(DataSize outstanding_data) {
outstanding_data_ = outstanding_data;
}
bool PacingController::Congested() const {
if (congestion_window_size_.IsFinite()) {
return outstanding_data_ >= congestion_window_size_;
}
return false;
}
Timestamp PacingController::CurrentTime() const {
Timestamp time = clock_->CurrentTime();
if (time < last_timestamp_) {
RTC_LOG(LS_WARNING)
<< "Non-monotonic clock behavior observed. Previous timestamp: "
<< last_timestamp_.ms() << ", new timestamp: " << time.ms();
RTC_DCHECK_GE(time, last_timestamp_);
time = last_timestamp_;
}
last_timestamp_ = time;
return time;
}
void PacingController::SetProbingEnabled(bool enabled) {
RTC_CHECK_EQ(0, packet_counter_);
prober_.SetEnabled(enabled);
}
void PacingController::SetPacingRates(DataRate pacing_rate,
DataRate padding_rate) {
RTC_DCHECK_GT(pacing_rate, DataRate::Zero());
pacing_bitrate_ = pacing_rate;
padding_budget_.set_target_rate_kbps(padding_rate.kbps());
RTC_LOG(LS_VERBOSE) << "bwe:pacer_updated pacing_kbps="
<< pacing_bitrate_.kbps()
<< " padding_budget_kbps=" << padding_rate.kbps();
}
void PacingController::InsertPacket(RtpPacketSender::Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) {
RTC_NOTREACHED();
}
void PacingController::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) {
RTC_DCHECK(pacing_bitrate_ > DataRate::Zero())
<< "SetPacingRate must be called before InsertPacket.";
Timestamp now = CurrentTime();
prober_.OnIncomingPacket(packet->payload_size());
if (packet->capture_time_ms() < 0) {
packet->set_capture_time_ms(now.ms());
}
RTC_CHECK(packet->packet_type());
int priority = GetPriorityForType(*packet->packet_type());
packet_queue_.Push(priority, now, packet_counter_++, std::move(packet));
}
void PacingController::SetAccountForAudioPackets(bool account_for_audio) {
account_for_audio_ = account_for_audio;
}
TimeDelta PacingController::ExpectedQueueTime() const {
RTC_DCHECK_GT(pacing_bitrate_, DataRate::Zero());
return TimeDelta::ms(
(QueueSizeData().bytes() * 8 * rtc::kNumMillisecsPerSec) /
pacing_bitrate_.bps());
}
size_t PacingController::QueueSizePackets() const {
return packet_queue_.SizeInPackets();
}
DataSize PacingController::QueueSizeData() const {
return packet_queue_.Size();
}
absl::optional<Timestamp> PacingController::FirstSentPacketTime() const {
return first_sent_packet_time_;
}
TimeDelta PacingController::OldestPacketWaitTime() const {
Timestamp oldest_packet = packet_queue_.OldestEnqueueTime();
if (oldest_packet.IsInfinite()) {
return TimeDelta::Zero();
}
return CurrentTime() - oldest_packet;
}
TimeDelta PacingController::UpdateTimeAndGetElapsed(Timestamp now) {
TimeDelta elapsed_time = now - time_last_process_;
time_last_process_ = now;
if (elapsed_time > kMaxElapsedTime) {
RTC_LOG(LS_WARNING) << "Elapsed time (" << elapsed_time.ms()
<< " ms) longer than expected, limiting to "
<< kMaxElapsedTime.ms();
elapsed_time = kMaxElapsedTime;
}
return elapsed_time;
}
bool PacingController::ShouldSendKeepalive(Timestamp now) const {
if (send_padding_if_silent_ || paused_ || Congested()) {
// We send a padding packet every 500 ms to ensure we won't get stuck in
// congested state due to no feedback being received.
TimeDelta elapsed_since_last_send = now - last_send_time_;
if (elapsed_since_last_send >= kCongestedPacketInterval) {
// We can not send padding unless a normal packet has first been sent. If
// we do, timestamps get messed up.
if (packet_counter_ > 0) {
return true;
}
}
}
return false;
}
absl::optional<TimeDelta> PacingController::TimeUntilNextProbe() {
if (!prober_.IsProbing()) {
return absl::nullopt;
}
TimeDelta time_delta =
TimeDelta::ms(prober_.TimeUntilNextProbe(CurrentTime().ms()));
if (time_delta > TimeDelta::Zero() ||
(time_delta == TimeDelta::Zero() && !probing_send_failure_)) {
return time_delta;
}
return absl::nullopt;
}
TimeDelta PacingController::TimeElapsedSinceLastProcess() const {
return CurrentTime() - time_last_process_;
}
void PacingController::ProcessPackets() {
Timestamp now = CurrentTime();
TimeDelta elapsed_time = UpdateTimeAndGetElapsed(now);
if (ShouldSendKeepalive(now)) {
DataSize keepalive_data_sent = DataSize::Zero();
std::vector<std::unique_ptr<RtpPacketToSend>> keepalive_packets =
packet_sender_->GeneratePadding(DataSize::bytes(1));
for (auto& packet : keepalive_packets) {
keepalive_data_sent +=
DataSize::bytes(packet->payload_size() + packet->padding_size());
packet_sender_->SendRtpPacket(std::move(packet), PacedPacketInfo());
}
OnPaddingSent(keepalive_data_sent);
}
if (paused_)
return;
if (elapsed_time > TimeDelta::Zero()) {
DataRate target_rate = pacing_bitrate_;
DataSize queue_size_data = packet_queue_.Size();
if (queue_size_data > DataSize::Zero()) {
// Assuming equal size packets and input/output rate, the average packet
// has avg_time_left_ms left to get queue_size_bytes out of the queue, if
// time constraint shall be met. Determine bitrate needed for that.
packet_queue_.UpdateQueueTime(CurrentTime());
if (drain_large_queues_) {
TimeDelta avg_time_left =
std::max(TimeDelta::ms(1),
queue_time_limit - packet_queue_.AverageQueueTime());
DataRate min_rate_needed = queue_size_data / avg_time_left;
if (min_rate_needed > target_rate) {
target_rate = min_rate_needed;
RTC_LOG(LS_VERBOSE) << "bwe:large_pacing_queue pacing_rate_kbps="
<< target_rate.kbps();
}
}
}
media_budget_.set_target_rate_kbps(target_rate.kbps());
UpdateBudgetWithElapsedTime(elapsed_time);
}
bool is_probing = prober_.IsProbing();
PacedPacketInfo pacing_info;
absl::optional<DataSize> recommended_probe_size;
if (is_probing) {
pacing_info = prober_.CurrentCluster();
recommended_probe_size = DataSize::bytes(prober_.RecommendedMinProbeSize());
}
DataSize data_sent = DataSize::Zero();
// The paused state is checked in the loop since it leaves the critical
// section allowing the paused state to be changed from other code.
while (!paused_) {
auto* packet = GetPendingPacket(pacing_info);
if (packet == nullptr) {
// No packet available to send, check if we should send padding.
DataSize padding_to_add = PaddingToAdd(recommended_probe_size, data_sent);
if (padding_to_add > DataSize::Zero()) {
std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
packet_sender_->GeneratePadding(padding_to_add);
if (padding_packets.empty()) {
// No padding packets were generated, quite send loop.
break;
}
for (auto& packet : padding_packets) {
EnqueuePacket(std::move(packet));
}
// Continue loop to send the padding that was just added.
continue;
}
// Can't fetch new packet and no padding to send, exit send loop.
break;
}
std::unique_ptr<RtpPacketToSend> rtp_packet = packet->ReleasePacket();
RTC_DCHECK(rtp_packet);
packet_sender_->SendRtpPacket(std::move(rtp_packet), pacing_info);
data_sent += packet->size();
// Send succeeded, remove it from the queue.
OnPacketSent(packet);
if (recommended_probe_size && data_sent > *recommended_probe_size)
break;
}
if (is_probing) {
probing_send_failure_ = data_sent == DataSize::Zero();
if (!probing_send_failure_) {
prober_.ProbeSent(CurrentTime().ms(), data_sent.bytes());
}
}
}
DataSize PacingController::PaddingToAdd(
absl::optional<DataSize> recommended_probe_size,
DataSize data_sent) {
if (!packet_queue_.Empty()) {
// Actual payload available, no need to add padding.
return DataSize::Zero();
}
if (Congested()) {
// Don't add padding if congested, even if requested for probing.
return DataSize::Zero();
}
if (packet_counter_ == 0) {
// We can not send padding unless a normal packet has first been sent. If we
// do, timestamps get messed up.
return DataSize::Zero();
}
if (recommended_probe_size) {
if (*recommended_probe_size > data_sent) {
return *recommended_probe_size - data_sent;
}
return DataSize::Zero();
}
return DataSize::bytes(padding_budget_.bytes_remaining());
}
RoundRobinPacketQueue::QueuedPacket* PacingController::GetPendingPacket(
const PacedPacketInfo& pacing_info) {
if (packet_queue_.Empty()) {
return nullptr;
}
// Since we need to release the lock in order to send, we first pop the
// element from the priority queue but keep it in storage, so that we can
// reinsert it if send fails.
RoundRobinPacketQueue::QueuedPacket* packet = packet_queue_.BeginPop();
bool audio_packet = packet->type() == RtpPacketToSend::Type::kAudio;
bool apply_pacing = !audio_packet || pace_audio_;
if (apply_pacing && (Congested() || (media_budget_.bytes_remaining() == 0 &&
pacing_info.probe_cluster_id ==
PacedPacketInfo::kNotAProbe))) {
packet_queue_.CancelPop();
return nullptr;
}
return packet;
}
void PacingController::OnPacketSent(
RoundRobinPacketQueue::QueuedPacket* packet) {
Timestamp now = CurrentTime();
if (!first_sent_packet_time_) {
first_sent_packet_time_ = now;
}
bool audio_packet = packet->type() == RtpPacketToSend::Type::kAudio;
if (!audio_packet || account_for_audio_) {
// Update media bytes sent.
UpdateBudgetWithSentData(packet->size());
last_send_time_ = now;
}
// Send succeeded, remove it from the queue.
packet_queue_.FinalizePop();
padding_failure_state_ = false;
}
void PacingController::OnPaddingSent(DataSize data_sent) {
if (data_sent > DataSize::Zero()) {
UpdateBudgetWithSentData(data_sent);
} else {
padding_failure_state_ = true;
}
last_send_time_ = CurrentTime();
}
void PacingController::UpdateBudgetWithElapsedTime(TimeDelta delta) {
delta = std::min(kMaxProcessingInterval, delta);
media_budget_.IncreaseBudget(delta.ms());
padding_budget_.IncreaseBudget(delta.ms());
}
void PacingController::UpdateBudgetWithSentData(DataSize size) {
outstanding_data_ += size;
media_budget_.UseBudget(size.bytes());
padding_budget_.UseBudget(size.bytes());
}
void PacingController::SetQueueTimeLimit(TimeDelta limit) {
queue_time_limit = limit;
}
} // namespace webrtc