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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 <map>
#include <queue>
#include <set>
#include <vector>
#include "modules/include/module_common_types.h"
#include "modules/pacing/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 "rtc_base/ptr_util.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 kPausedPacketIntervalMs = 500;
// 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_(rtc::MakeUnique<AlrDetector>()),
paused_(false),
media_budget_(rtc::MakeUnique<IntervalBudget>(0)),
padding_budget_(rtc::MakeUnique<IntervalBudget>(0)),
prober_(rtc::MakeUnique<BitrateProber>(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_update_us_(clock->TimeInMicroseconds()),
first_sent_packet_ms_(-1),
packets_(webrtc::field_trial::IsEnabled("WebRTC-RoundRobinPacing")
? rtc::MakeUnique<PacketQueue2>(clock)
: rtc::MakeUnique<PacketQueue>(clock)),
packet_counter_(0),
pacing_factor_(kDefaultPaceMultiplier),
queue_time_limit(kMaxQueueLengthMs),
account_for_audio_(false) {
UpdateBudgetWithElapsedTime(kMinPacketLimitMs);
}
PacedSender::~PacedSender() {}
void PacedSender::CreateProbeCluster(int bitrate_bps) {
rtc::CritScope cs(&critsect_);
prober_->CreateProbeCluster(bitrate_bps, clock_->TimeInMilliseconds());
}
void PacedSender::Pause() {
{
rtc::CritScope cs(&critsect_);
if (!paused_)
LOG(LS_INFO) << "PacedSender paused.";
paused_ = true;
packets_->SetPauseState(true, clock_->TimeInMilliseconds());
}
// 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_)
LOG(LS_INFO) << "PacedSender resumed.";
paused_ = false;
packets_->SetPauseState(false, clock_->TimeInMilliseconds());
}
// 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::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)
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::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(estimated_bitrate_bps_ > 0)
<< "SetEstimatedBitrate must be called before InsertPacket.";
int64_t now_ms = clock_->TimeInMilliseconds();
prober_->OnIncomingPacket(bytes);
if (capture_time_ms < 0)
capture_time_ms = now_ms;
packets_->Push(PacketQueue::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_);
}
rtc::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 clock_->TimeInMilliseconds() - oldest_packet;
}
int64_t PacedSender::TimeUntilNextProcess() {
rtc::CritScope cs(&critsect_);
int64_t elapsed_time_us = clock_->TimeInMicroseconds() - time_last_update_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>(kPausedPacketIntervalMs - elapsed_time_ms, 0);
if (prober_->IsProbing()) {
int64_t ret = prober_->TimeUntilNextProbe(clock_->TimeInMilliseconds());
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 = std::min(
kMaxIntervalTimeMs, (now_us - time_last_update_us_ + 500) / 1000);
int target_bitrate_kbps = pacing_bitrate_kbps_;
if (paused_) {
PacedPacketInfo pacing_info;
time_last_update_us_ = now_us;
// 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;
size_t bytes_sent = SendPadding(1, pacing_info);
alr_detector_->OnBytesSent(bytes_sent, elapsed_time_ms);
return;
}
if (elapsed_time_ms > 0) {
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(clock_->TimeInMilliseconds());
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);
}
time_last_update_us_ = now_us;
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();
}
while (!packets_->Empty()) {
// 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 PacketQueue::Packet& packet = packets_->BeginPop();
if (SendPacket(packet, pacing_info)) {
// Send succeeded, remove it from the queue.
if (first_sent_packet_ms_ == -1)
first_sent_packet_ms_ = clock_->TimeInMilliseconds();
bytes_sent += packet.bytes;
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()) {
// 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(clock_->TimeInMilliseconds(), bytes_sent);
}
alr_detector_->OnBytesSent(bytes_sent, elapsed_time_ms);
}
void PacedSender::ProcessThreadAttached(ProcessThread* process_thread) {
LOG(LS_INFO) << "ProcessThreadAttached 0x" << std::hex << process_thread;
process_thread_ = process_thread;
}
bool PacedSender::SendPacket(const PacketQueue::Packet& packet,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK(!paused_);
if (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 (packet.priority != kHighPriority || 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);
}
}
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);
}
return bytes_sent;
}
void PacedSender::UpdateBudgetWithElapsedTime(int64_t delta_time_ms) {
media_budget_->IncreaseBudget(delta_time_ms);
padding_budget_->IncreaseBudget(delta_time_ms);
}
void PacedSender::UpdateBudgetWithBytesSent(size_t 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_);
}
float PacedSender::GetPacingFactor() const {
rtc::CritScope cs(&critsect_);
return pacing_factor_;
}
void PacedSender::SetQueueTimeLimit(int limit_ms) {
rtc::CritScope cs(&critsect_);
queue_time_limit = limit_ms;
}
} // namespace webrtc