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/*
* Copyright (c) 2022 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/prioritized_packet_queue.h"
#include <algorithm>
#include <array>
#include <utility>
#include "absl/container/inlined_vector.h"
#include "absl/types/optional.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace {
constexpr int kAudioPrioLevel = 0;
int GetPriorityForType(
RtpPacketMediaType type,
absl::optional<RtpPacketToSend::OriginalType> original_type) {
// Lower number takes priority over higher.
switch (type) {
case RtpPacketMediaType::kAudio:
// Audio is always prioritized over other packet types.
return kAudioPrioLevel;
case RtpPacketMediaType::kRetransmission:
// Send retransmissions before new media. If original_type is set, audio
// retransmission is prioritized more than video retransmission.
if (original_type == RtpPacketToSend::OriginalType::kVideo) {
return kAudioPrioLevel + 2;
}
return kAudioPrioLevel + 1;
case RtpPacketMediaType::kVideo:
case RtpPacketMediaType::kForwardErrorCorrection:
// Video has "normal" priority, in the old speak.
// Send redundancy concurrently to video. If it is delayed it might have a
// lower chance of being useful.
return kAudioPrioLevel + 3;
case RtpPacketMediaType::kPadding:
// Packets that are in themselves likely useless, only sent to keep the
// BWE high.
return kAudioPrioLevel + 4;
}
RTC_CHECK_NOTREACHED();
}
} // namespace
absl::InlinedVector<TimeDelta, PrioritizedPacketQueue::kNumPriorityLevels>
PrioritizedPacketQueue::ToTtlPerPrio(PacketQueueTTL packet_queue_ttl) {
absl::InlinedVector<TimeDelta, PrioritizedPacketQueue::kNumPriorityLevels>
ttl_per_prio(kNumPriorityLevels, TimeDelta::PlusInfinity());
ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kRetransmission,
RtpPacketToSend::OriginalType::kAudio)] =
packet_queue_ttl.audio_retransmission;
ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kRetransmission,
RtpPacketToSend::OriginalType::kVideo)] =
packet_queue_ttl.video_retransmission;
ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kVideo, absl::nullopt)] =
packet_queue_ttl.video;
return ttl_per_prio;
}
DataSize PrioritizedPacketQueue::QueuedPacket::PacketSize() const {
return DataSize::Bytes(packet->payload_size() + packet->padding_size());
}
PrioritizedPacketQueue::StreamQueue::StreamQueue(Timestamp creation_time)
: last_enqueue_time_(creation_time), num_keyframe_packets_(0) {}
bool PrioritizedPacketQueue::StreamQueue::EnqueuePacket(QueuedPacket packet,
int priority_level) {
if (packet.packet->is_key_frame()) {
++num_keyframe_packets_;
}
bool first_packet_at_level = packets_[priority_level].empty();
packets_[priority_level].push_back(std::move(packet));
return first_packet_at_level;
}
PrioritizedPacketQueue::QueuedPacket
PrioritizedPacketQueue::StreamQueue::DequeuePacket(int priority_level) {
RTC_DCHECK(!packets_[priority_level].empty());
QueuedPacket packet = std::move(packets_[priority_level].front());
packets_[priority_level].pop_front();
if (packet.packet->is_key_frame()) {
RTC_DCHECK_GT(num_keyframe_packets_, 0);
--num_keyframe_packets_;
}
return packet;
}
bool PrioritizedPacketQueue::StreamQueue::HasPacketsAtPrio(
int priority_level) const {
return !packets_[priority_level].empty();
}
bool PrioritizedPacketQueue::StreamQueue::IsEmpty() const {
for (const std::deque<QueuedPacket>& queue : packets_) {
if (!queue.empty()) {
return false;
}
}
return true;
}
Timestamp PrioritizedPacketQueue::StreamQueue::LeadingPacketEnqueueTime(
int priority_level) const {
RTC_DCHECK(!packets_[priority_level].empty());
return packets_[priority_level].begin()->enqueue_time;
}
Timestamp PrioritizedPacketQueue::StreamQueue::LastEnqueueTime() const {
return last_enqueue_time_;
}
std::array<std::deque<PrioritizedPacketQueue::QueuedPacket>,
PrioritizedPacketQueue::kNumPriorityLevels>
PrioritizedPacketQueue::StreamQueue::DequeueAll() {
std::array<std::deque<QueuedPacket>, kNumPriorityLevels> packets_by_prio;
for (int i = 0; i < kNumPriorityLevels; ++i) {
packets_by_prio[i].swap(packets_[i]);
}
num_keyframe_packets_ = 0;
return packets_by_prio;
}
PrioritizedPacketQueue::PrioritizedPacketQueue(
Timestamp creation_time,
bool prioritize_audio_retransmission,
PacketQueueTTL packet_queue_ttl)
: prioritize_audio_retransmission_(prioritize_audio_retransmission),
time_to_live_per_prio_(ToTtlPerPrio(packet_queue_ttl)),
queue_time_sum_(TimeDelta::Zero()),
pause_time_sum_(TimeDelta::Zero()),
size_packets_(0),
size_packets_per_media_type_({}),
size_payload_(DataSize::Zero()),
last_update_time_(creation_time),
paused_(false),
last_culling_time_(creation_time),
top_active_prio_level_(-1) {}
void PrioritizedPacketQueue::Push(Timestamp enqueue_time,
std::unique_ptr<RtpPacketToSend> packet) {
StreamQueue* stream_queue;
auto [it, inserted] = streams_.emplace(packet->Ssrc(), nullptr);
if (inserted) {
it->second = std::make_unique<StreamQueue>(enqueue_time);
}
stream_queue = it->second.get();
auto enqueue_time_iterator =
enqueue_times_.insert(enqueue_times_.end(), enqueue_time);
RTC_DCHECK(packet->packet_type().has_value());
RtpPacketMediaType packet_type = packet->packet_type().value();
int prio_level =
GetPriorityForType(packet_type, prioritize_audio_retransmission_
? packet->original_packet_type()
: absl::nullopt);
PurgeOldPacketsAtPriorityLevel(prio_level, enqueue_time);
RTC_DCHECK_GE(prio_level, 0);
RTC_DCHECK_LT(prio_level, kNumPriorityLevels);
QueuedPacket queued_packed = {.packet = std::move(packet),
.enqueue_time = enqueue_time,
.enqueue_time_iterator = enqueue_time_iterator};
// In order to figure out how much time a packet has spent in the queue
// while not in a paused state, we subtract the total amount of time the
// queue has been paused so far, and when the packet is popped we subtract
// the total amount of time the queue has been paused at that moment. This
// way we subtract the total amount of time the packet has spent in the
// queue while in a paused state.
UpdateAverageQueueTime(enqueue_time);
queued_packed.enqueue_time -= pause_time_sum_;
++size_packets_;
++size_packets_per_media_type_[static_cast<size_t>(packet_type)];
size_payload_ += queued_packed.PacketSize();
if (stream_queue->EnqueuePacket(std::move(queued_packed), prio_level)) {
// Number packets at `prio_level` for this steam is now non-zero.
streams_by_prio_[prio_level].push_back(stream_queue);
}
if (top_active_prio_level_ < 0 || prio_level < top_active_prio_level_) {
top_active_prio_level_ = prio_level;
}
static constexpr TimeDelta kTimeout = TimeDelta::Millis(500);
if (enqueue_time - last_culling_time_ > kTimeout) {
for (auto it = streams_.begin(); it != streams_.end();) {
if (it->second->IsEmpty() &&
it->second->LastEnqueueTime() + kTimeout < enqueue_time) {
streams_.erase(it++);
} else {
++it;
}
}
last_culling_time_ = enqueue_time;
}
}
std::unique_ptr<RtpPacketToSend> PrioritizedPacketQueue::Pop() {
if (size_packets_ == 0) {
return nullptr;
}
RTC_DCHECK_GE(top_active_prio_level_, 0);
StreamQueue& stream_queue = *streams_by_prio_[top_active_prio_level_].front();
QueuedPacket packet = stream_queue.DequeuePacket(top_active_prio_level_);
DequeuePacketInternal(packet);
// Remove StreamQueue from head of fifo-queue for this prio level, and
// and add it to the end if it still has packets.
streams_by_prio_[top_active_prio_level_].pop_front();
if (stream_queue.HasPacketsAtPrio(top_active_prio_level_)) {
streams_by_prio_[top_active_prio_level_].push_back(&stream_queue);
} else {
MaybeUpdateTopPrioLevel();
}
return std::move(packet.packet);
}
int PrioritizedPacketQueue::SizeInPackets() const {
return size_packets_;
}
DataSize PrioritizedPacketQueue::SizeInPayloadBytes() const {
return size_payload_;
}
bool PrioritizedPacketQueue::Empty() const {
return size_packets_ == 0;
}
const std::array<int, kNumMediaTypes>&
PrioritizedPacketQueue::SizeInPacketsPerRtpPacketMediaType() const {
return size_packets_per_media_type_;
}
Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTime(
RtpPacketMediaType type) const {
RTC_DCHECK(type != RtpPacketMediaType::kRetransmission);
const int priority_level = GetPriorityForType(type, absl::nullopt);
if (streams_by_prio_[priority_level].empty()) {
return Timestamp::MinusInfinity();
}
return streams_by_prio_[priority_level].front()->LeadingPacketEnqueueTime(
priority_level);
}
Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTimeForRetransmission()
const {
if (!prioritize_audio_retransmission_) {
const int priority_level =
GetPriorityForType(RtpPacketMediaType::kRetransmission, absl::nullopt);
if (streams_by_prio_[priority_level].empty()) {
return Timestamp::PlusInfinity();
}
return streams_by_prio_[priority_level].front()->LeadingPacketEnqueueTime(
priority_level);
}
const int audio_priority_level =
GetPriorityForType(RtpPacketMediaType::kRetransmission,
RtpPacketToSend::OriginalType::kAudio);
const int video_priority_level =
GetPriorityForType(RtpPacketMediaType::kRetransmission,
RtpPacketToSend::OriginalType::kVideo);
Timestamp next_audio =
streams_by_prio_[audio_priority_level].empty()
? Timestamp::PlusInfinity()
: streams_by_prio_[audio_priority_level]
.front()
->LeadingPacketEnqueueTime(audio_priority_level);
Timestamp next_video =
streams_by_prio_[video_priority_level].empty()
? Timestamp::PlusInfinity()
: streams_by_prio_[video_priority_level]
.front()
->LeadingPacketEnqueueTime(video_priority_level);
return std::min(next_audio, next_video);
}
Timestamp PrioritizedPacketQueue::OldestEnqueueTime() const {
return enqueue_times_.empty() ? Timestamp::MinusInfinity()
: enqueue_times_.front();
}
TimeDelta PrioritizedPacketQueue::AverageQueueTime() const {
if (size_packets_ == 0) {
return TimeDelta::Zero();
}
return queue_time_sum_ / size_packets_;
}
void PrioritizedPacketQueue::UpdateAverageQueueTime(Timestamp now) {
RTC_CHECK_GE(now, last_update_time_);
if (now == last_update_time_) {
return;
}
TimeDelta delta = now - last_update_time_;
if (paused_) {
pause_time_sum_ += delta;
} else {
queue_time_sum_ += delta * size_packets_;
}
last_update_time_ = now;
}
void PrioritizedPacketQueue::SetPauseState(bool paused, Timestamp now) {
UpdateAverageQueueTime(now);
paused_ = paused;
}
void PrioritizedPacketQueue::RemovePacketsForSsrc(uint32_t ssrc) {
auto kv = streams_.find(ssrc);
if (kv != streams_.end()) {
// Dequeue all packets from the queue for this SSRC.
StreamQueue& queue = *kv->second;
std::array<std::deque<QueuedPacket>, kNumPriorityLevels> packets_by_prio =
queue.DequeueAll();
for (int i = 0; i < kNumPriorityLevels; ++i) {
std::deque<QueuedPacket>& packet_queue = packets_by_prio[i];
if (packet_queue.empty()) {
continue;
}
// First erase all packets at this prio level.
while (!packet_queue.empty()) {
QueuedPacket packet = std::move(packet_queue.front());
packet_queue.pop_front();
DequeuePacketInternal(packet);
}
// Next, deregister this `StreamQueue` from the round-robin tables.
RTC_DCHECK(!streams_by_prio_[i].empty());
if (streams_by_prio_[i].size() == 1) {
// This is the last and only queue that had packets for this prio level.
// Update the global top prio level if neccessary.
RTC_DCHECK(streams_by_prio_[i].front() == &queue);
streams_by_prio_[i].pop_front();
} else {
// More than stream had packets at this prio level, filter this one out.
std::deque<StreamQueue*> filtered_queue;
for (StreamQueue* queue_ptr : streams_by_prio_[i]) {
if (queue_ptr != &queue) {
filtered_queue.push_back(queue_ptr);
}
}
streams_by_prio_[i].swap(filtered_queue);
}
}
}
MaybeUpdateTopPrioLevel();
}
bool PrioritizedPacketQueue::HasKeyframePackets(uint32_t ssrc) const {
auto it = streams_.find(ssrc);
if (it != streams_.end()) {
return it->second->has_keyframe_packets();
}
return false;
}
void PrioritizedPacketQueue::DequeuePacketInternal(QueuedPacket& packet) {
--size_packets_;
RTC_DCHECK(packet.packet->packet_type().has_value());
RtpPacketMediaType packet_type = packet.packet->packet_type().value();
--size_packets_per_media_type_[static_cast<size_t>(packet_type)];
RTC_DCHECK_GE(size_packets_per_media_type_[static_cast<size_t>(packet_type)],
0);
size_payload_ -= packet.PacketSize();
// Calculate the total amount of time spent by this packet in the queue
// while in a non-paused state. Note that the `pause_time_sum_ms_` was
// subtracted from `packet.enqueue_time_ms` when the packet was pushed, and
// by subtracting it now we effectively remove the time spent in in the
// queue while in a paused state.
TimeDelta time_in_non_paused_state =
last_update_time_ - packet.enqueue_time - pause_time_sum_;
queue_time_sum_ -= time_in_non_paused_state;
// Set the time spent in the send queue, which is the per-packet equivalent of
// totalPacketSendDelay. The notion of being paused is an implementation
// detail that we do not want to expose, so it makes sense to report the
// metric excluding the pause time. This also avoids spikes in the metric.
// https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
packet.packet->set_time_in_send_queue(time_in_non_paused_state);
RTC_DCHECK(size_packets_ > 0 || queue_time_sum_ == TimeDelta::Zero());
RTC_CHECK(packet.enqueue_time_iterator != enqueue_times_.end());
enqueue_times_.erase(packet.enqueue_time_iterator);
}
void PrioritizedPacketQueue::MaybeUpdateTopPrioLevel() {
if (top_active_prio_level_ == -1 ||
streams_by_prio_[top_active_prio_level_].empty()) {
// No stream queues have packets at this prio level, find top priority
// that is not empty.
if (size_packets_ == 0) {
top_active_prio_level_ = -1;
} else {
for (int i = 0; i < kNumPriorityLevels; ++i) {
PurgeOldPacketsAtPriorityLevel(i, last_update_time_);
if (!streams_by_prio_[i].empty()) {
top_active_prio_level_ = i;
break;
}
}
}
}
}
void PrioritizedPacketQueue::PurgeOldPacketsAtPriorityLevel(int prio_level,
Timestamp now) {
RTC_DCHECK(prio_level >= 0 && prio_level < kNumPriorityLevels);
TimeDelta time_to_live = time_to_live_per_prio_[prio_level];
if (time_to_live.IsInfinite()) {
return;
}
std::deque<StreamQueue*>& queues = streams_by_prio_[prio_level];
auto iter = queues.begin();
while (iter != queues.end()) {
StreamQueue* queue_ptr = *iter;
while (queue_ptr->HasPacketsAtPrio(prio_level) &&
(now - queue_ptr->LeadingPacketEnqueueTime(prio_level)) >
time_to_live) {
QueuedPacket packet = queue_ptr->DequeuePacket(prio_level);
RTC_LOG(LS_INFO) << "Dropping old packet on SSRC: "
<< packet.packet->Ssrc()
<< " seq:" << packet.packet->SequenceNumber()
<< " time in queue:" << (now - packet.enqueue_time).ms()
<< " ms";
DequeuePacketInternal(packet);
}
if (!queue_ptr->HasPacketsAtPrio(prio_level)) {
iter = queues.erase(iter);
} else {
++iter;
}
}
}
} // namespace webrtc