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/*
* 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 "absl/memory/memory.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/congestion_controller/goog_cc/alr_detector.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"
#include "system_wrappers/include/field_trial.h"
namespace {
// Time limit in milliseconds between packet bursts.
const int64_t kMinPacketLimitMs = 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;
} // namespace
namespace webrtc {
const int64_t PacedSender::kMaxQueueLengthMs = 2000;
const float PacedSender::kDefaultPaceMultiplier = 2.5f;
PacedSender::PacedSender(const Clock* clock,
PacketSender* packet_sender,
RtcEventLog* event_log)
: clock_(clock),
packet_sender_(packet_sender),
alr_detector_(absl::make_unique<AlrDetector>(event_log)),
drain_large_queues_(!field_trial::IsDisabled("WebRTC-Pacer-DrainQueue")),
send_padding_if_silent_(
field_trial::IsEnabled("WebRTC-Pacer-PadInSilence")),
video_blocks_audio_(!field_trial::IsDisabled("WebRTC-Pacer-BlockAudio")),
last_timestamp_ms_(clock_->TimeInMilliseconds()),
paused_(false),
media_budget_(0),
padding_budget_(0),
prober_(event_log),
probing_send_failure_(false),
estimated_bitrate_bps_(0),
min_send_bitrate_kbps_(0u),
max_padding_bitrate_kbps_(0u),
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),
pacing_factor_(kDefaultPaceMultiplier),
queue_time_limit(kMaxQueueLengthMs),
account_for_audio_(false) {
if (!drain_large_queues_)
RTC_LOG(LS_WARNING) << "Pacer queues will not be drained,"
"pushback experiment must be enabled.";
UpdateBudgetWithElapsedTime(kMinPacketLimitMs);
}
PacedSender::~PacedSender() {}
void PacedSender::CreateProbeCluster(int bitrate_bps) {
rtc::CritScope cs(&critsect_);
prober_.CreateProbeCluster(bitrate_bps, TimeMilliseconds());
}
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::SetEstimatedBitrate(uint32_t bitrate_bps) {
if (bitrate_bps == 0)
RTC_LOG(LS_ERROR) << "PacedSender is not designed to handle 0 bitrate.";
rtc::CritScope cs(&critsect_);
estimated_bitrate_bps_ = bitrate_bps;
padding_budget_.set_target_rate_kbps(
std::min(estimated_bitrate_bps_ / 1000, max_padding_bitrate_kbps_));
pacing_bitrate_kbps_ =
std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
pacing_factor_;
alr_detector_->SetEstimatedBitrate(bitrate_bps);
}
void PacedSender::SetSendBitrateLimits(int min_send_bitrate_bps,
int padding_bitrate) {
rtc::CritScope cs(&critsect_);
min_send_bitrate_kbps_ = min_send_bitrate_bps / 1000;
pacing_bitrate_kbps_ =
std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000) *
pacing_factor_;
max_padding_bitrate_kbps_ = padding_bitrate / 1000;
padding_budget_.set_target_rate_kbps(
std::min(estimated_bitrate_bps_ / 1000, max_padding_bitrate_kbps_));
}
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);
}
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;
packets_.Push(RoundRobinPacketQueue::Packet(
priority, ssrc, sequence_number, capture_time_ms, now_ms, bytes,
retransmission, packet_counter_++));
}
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_);
}
absl::optional<int64_t> PacedSender::GetApplicationLimitedRegionStartTime()
const {
rtc::CritScope cs(&critsect_);
return alr_detector_->GetApplicationLimitedRegionStartTime();
}
size_t PacedSender::QueueSizePackets() const {
rtc::CritScope cs(&critsect_);
return packets_.SizeInPackets();
}
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>(kMinPacketLimitMs - elapsed_time_ms, 0);
}
void PacedSender::Process() {
int64_t now_us = clock_->TimeInMicroseconds();
rtc::CritScope cs(&critsect_);
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;
}
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) {
PacedPacketInfo pacing_info;
size_t bytes_sent = SendPadding(1, pacing_info);
alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
}
}
}
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;
}
}
media_budget_.set_target_rate_kbps(target_bitrate_kbps);
UpdateBudgetWithElapsedTime(elapsed_time_ms);
}
bool is_probing = prober_.IsProbing();
PacedPacketInfo pacing_info;
size_t bytes_sent = 0;
size_t recommended_probe_size = 0;
if (is_probing) {
pacing_info = prober_.CurrentCluster();
recommended_probe_size = prober_.RecommendedMinProbeSize();
}
// The paused state is checked in the loop since SendPacket leaves the
// critical section allowing the paused state to be changed from other code.
while (!packets_.Empty() && !paused_) {
// 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.
const RoundRobinPacketQueue::Packet& packet = packets_.BeginPop();
if (SendPacket(packet, pacing_info)) {
bytes_sent += packet.bytes;
// Send succeeded, remove it from the queue.
packets_.FinalizePop(packet);
if (is_probing && bytes_sent > recommended_probe_size)
break;
} else {
// Send failed, put it back into the queue.
packets_.CancelPop(packet);
break;
}
}
if (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>(is_probing ? (recommended_probe_size - bytes_sent)
: padding_budget_.bytes_remaining());
if (padding_needed > 0) {
bytes_sent += SendPadding(padding_needed, pacing_info);
}
}
}
if (is_probing) {
probing_send_failure_ = bytes_sent == 0;
if (!probing_send_failure_)
prober_.ProbeSent(TimeMilliseconds(), bytes_sent);
}
alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
}
void PacedSender::ProcessThreadAttached(ProcessThread* process_thread) {
RTC_LOG(LS_INFO) << "ProcessThreadAttached 0x" << process_thread;
rtc::CritScope cs(&process_thread_lock_);
process_thread_ = process_thread;
}
bool PacedSender::SendPacket(const RoundRobinPacketQueue::Packet& packet,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK(!paused_);
bool audio_packet = packet.priority == kHighPriority;
bool apply_pacing =
!audio_packet || account_for_audio_ || video_blocks_audio_;
if (apply_pacing && (Congested() || (media_budget_.bytes_remaining() == 0 &&
pacing_info.probe_cluster_id ==
PacedPacketInfo::kNotAProbe))) {
return false;
}
critsect_.Leave();
const bool success = packet_sender_->TimeToSendPacket(
packet.ssrc, packet.sequence_number, packet.capture_time_ms,
packet.retransmission, pacing_info);
critsect_.Enter();
if (success) {
if (first_sent_packet_ms_ == -1)
first_sent_packet_ms_ = TimeMilliseconds();
if (!audio_packet || account_for_audio_) {
// Update media bytes sent.
// TODO(eladalon): TimeToSendPacket() can also return |true| in some
// situations where nothing actually ended up being sent to the network,
// and we probably don't want to update the budget in such cases.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8052
UpdateBudgetWithBytesSent(packet.bytes);
last_send_time_us_ = clock_->TimeInMicroseconds();
}
}
return success;
}
size_t PacedSender::SendPadding(size_t padding_needed,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK_GT(packet_counter_, 0);
critsect_.Leave();
size_t bytes_sent =
packet_sender_->TimeToSendPadding(padding_needed, pacing_info);
critsect_.Enter();
if (bytes_sent > 0) {
UpdateBudgetWithBytesSent(bytes_sent);
}
last_send_time_us_ = clock_->TimeInMicroseconds();
return bytes_sent;
}
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::SetPacingFactor(float pacing_factor) {
rtc::CritScope cs(&critsect_);
pacing_factor_ = pacing_factor;
// Make sure new padding factor is applied immediately, otherwise we need to
// wait for the send bitrate estimate to be updated before this takes effect.
SetEstimatedBitrate(estimated_bitrate_bps_);
}
void PacedSender::SetQueueTimeLimit(int limit_ms) {
rtc::CritScope cs(&critsect_);
queue_time_limit = limit_ms;
}
} // namespace webrtc