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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/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h"
#include <limits>
#include <memory>
#include <utility>
#include "absl/strings/match.h"
#include "api/transport/field_trial_based_config.h"
#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace {
constexpr uint32_t kTimestampTicksPerMs = 90;
constexpr int kSendSideDelayWindowMs = 1000;
constexpr int kBitrateStatisticsWindowMs = 1000;
constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13;
bool IsDisabled(absl::string_view name,
const WebRtcKeyValueConfig* field_trials) {
FieldTrialBasedConfig default_trials;
auto& trials = field_trials ? *field_trials : default_trials;
return absl::StartsWith(trials.Lookup(name), "Disabled");
}
} // namespace
DEPRECATED_RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender(
DEPRECATED_RtpSenderEgress* sender)
: transport_sequence_number_(0), sender_(sender) {}
DEPRECATED_RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() =
default;
void DEPRECATED_RtpSenderEgress::NonPacedPacketSender::EnqueuePackets(
std::vector<std::unique_ptr<RtpPacketToSend>> packets) {
for (auto& packet : packets) {
if (!packet->SetExtension<TransportSequenceNumber>(
++transport_sequence_number_)) {
--transport_sequence_number_;
}
packet->ReserveExtension<TransmissionOffset>();
packet->ReserveExtension<AbsoluteSendTime>();
sender_->SendPacket(packet.get(), PacedPacketInfo());
}
}
DEPRECATED_RtpSenderEgress::DEPRECATED_RtpSenderEgress(
const RtpRtcpInterface::Configuration& config,
RtpPacketHistory* packet_history)
: ssrc_(config.local_media_ssrc),
rtx_ssrc_(config.rtx_send_ssrc),
flexfec_ssrc_(config.fec_generator ? config.fec_generator->FecSsrc()
: absl::nullopt),
populate_network2_timestamp_(config.populate_network2_timestamp),
send_side_bwe_with_overhead_(
!IsDisabled("WebRTC-SendSideBwe-WithOverhead", config.field_trials)),
clock_(config.clock),
packet_history_(packet_history),
transport_(config.outgoing_transport),
event_log_(config.event_log),
is_audio_(config.audio),
need_rtp_packet_infos_(config.need_rtp_packet_infos),
transport_feedback_observer_(config.transport_feedback_callback),
send_side_delay_observer_(config.send_side_delay_observer),
send_packet_observer_(config.send_packet_observer),
rtp_stats_callback_(config.rtp_stats_callback),
bitrate_callback_(config.send_bitrate_observer),
media_has_been_sent_(false),
force_part_of_allocation_(false),
timestamp_offset_(0),
max_delay_it_(send_delays_.end()),
sum_delays_ms_(0),
total_packet_send_delay_ms_(0),
send_rates_(kNumMediaTypes,
{kBitrateStatisticsWindowMs, RateStatistics::kBpsScale}),
rtp_sequence_number_map_(need_rtp_packet_infos_
? std::make_unique<RtpSequenceNumberMap>(
kRtpSequenceNumberMapMaxEntries)
: nullptr) {}
void DEPRECATED_RtpSenderEgress::SendPacket(
RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info) {
RTC_DCHECK(packet);
const uint32_t packet_ssrc = packet->Ssrc();
RTC_DCHECK(packet->packet_type().has_value());
RTC_DCHECK(HasCorrectSsrc(*packet));
int64_t now_ms = clock_->TimeInMilliseconds();
if (is_audio_) {
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms,
GetSendRates().Sum().kbps(), packet_ssrc);
BWE_TEST_LOGGING_PLOT_WITH_SSRC(
1, "AudioNackBitrate_kbps", now_ms,
GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
packet_ssrc);
#endif
} else {
#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms,
GetSendRates().Sum().kbps(), packet_ssrc);
BWE_TEST_LOGGING_PLOT_WITH_SSRC(
1, "VideoNackBitrate_kbps", now_ms,
GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(),
packet_ssrc);
#endif
}
PacketOptions options;
{
MutexLock lock(&lock_);
options.included_in_allocation = force_part_of_allocation_;
if (need_rtp_packet_infos_ &&
packet->packet_type() == RtpPacketToSend::Type::kVideo) {
RTC_DCHECK(rtp_sequence_number_map_);
// Last packet of a frame, add it to sequence number info map.
const uint32_t timestamp = packet->Timestamp() - timestamp_offset_;
bool is_first_packet_of_frame = packet->is_first_packet_of_frame();
bool is_last_packet_of_frame = packet->Marker();
rtp_sequence_number_map_->InsertPacket(
packet->SequenceNumber(),
RtpSequenceNumberMap::Info(timestamp, is_first_packet_of_frame,
is_last_packet_of_frame));
}
}
// Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after
// the pacer, these modifications of the header below are happening after the
// FEC protection packets are calculated. This will corrupt recovered packets
// at the same place. It's not an issue for extensions, which are present in
// all the packets (their content just may be incorrect on recovered packets).
// In case of VideoTimingExtension, since it's present not in every packet,
// data after rtp header may be corrupted if these packets are protected by
// the FEC.
int64_t diff_ms = now_ms - packet->capture_time_ms();
if (packet->HasExtension<TransmissionOffset>()) {
packet->SetExtension<TransmissionOffset>(kTimestampTicksPerMs * diff_ms);
}
if (packet->HasExtension<AbsoluteSendTime>()) {
packet->SetExtension<AbsoluteSendTime>(
AbsoluteSendTime::MsTo24Bits(now_ms));
}
if (packet->HasExtension<VideoTimingExtension>()) {
if (populate_network2_timestamp_) {
packet->set_network2_time_ms(now_ms);
} else {
packet->set_pacer_exit_time_ms(now_ms);
}
}
const bool is_media = packet->packet_type() == RtpPacketMediaType::kAudio ||
packet->packet_type() == RtpPacketMediaType::kVideo;
// Downstream code actually uses this flag to distinguish between media and
// everything else.
options.is_retransmit = !is_media;
if (auto packet_id = packet->GetExtension<TransportSequenceNumber>()) {
options.packet_id = *packet_id;
options.included_in_feedback = true;
options.included_in_allocation = true;
AddPacketToTransportFeedback(*packet_id, *packet, pacing_info);
}
options.additional_data = packet->additional_data();
if (packet->packet_type() != RtpPacketMediaType::kPadding &&
packet->packet_type() != RtpPacketMediaType::kRetransmission) {
UpdateDelayStatistics(packet->capture_time_ms(), now_ms, packet_ssrc);
UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
packet_ssrc);
}
const bool send_success = SendPacketToNetwork(*packet, options, pacing_info);
// Put packet in retransmission history or update pending status even if
// actual sending fails.
if (is_media && packet->allow_retransmission()) {
packet_history_->PutRtpPacket(std::make_unique<RtpPacketToSend>(*packet),
now_ms);
} else if (packet->retransmitted_sequence_number()) {
packet_history_->MarkPacketAsSent(*packet->retransmitted_sequence_number());
}
if (send_success) {
MutexLock lock(&lock_);
UpdateRtpStats(*packet);
media_has_been_sent_ = true;
}
}
void DEPRECATED_RtpSenderEgress::ProcessBitrateAndNotifyObservers() {
if (!bitrate_callback_)
return;
MutexLock lock(&lock_);
RtpSendRates send_rates = GetSendRatesLocked();
bitrate_callback_->Notify(
send_rates.Sum().bps(),
send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_);
}
RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRates() const {
MutexLock lock(&lock_);
return GetSendRatesLocked();
}
RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRatesLocked() const {
const int64_t now_ms = clock_->TimeInMilliseconds();
RtpSendRates current_rates;
for (size_t i = 0; i < kNumMediaTypes; ++i) {
RtpPacketMediaType type = static_cast<RtpPacketMediaType>(i);
current_rates[type] =
DataRate::BitsPerSec(send_rates_[i].Rate(now_ms).value_or(0));
}
return current_rates;
}
void DEPRECATED_RtpSenderEgress::GetDataCounters(
StreamDataCounters* rtp_stats,
StreamDataCounters* rtx_stats) const {
MutexLock lock(&lock_);
*rtp_stats = rtp_stats_;
*rtx_stats = rtx_rtp_stats_;
}
void DEPRECATED_RtpSenderEgress::ForceIncludeSendPacketsInAllocation(
bool part_of_allocation) {
MutexLock lock(&lock_);
force_part_of_allocation_ = part_of_allocation;
}
bool DEPRECATED_RtpSenderEgress::MediaHasBeenSent() const {
MutexLock lock(&lock_);
return media_has_been_sent_;
}
void DEPRECATED_RtpSenderEgress::SetMediaHasBeenSent(bool media_sent) {
MutexLock lock(&lock_);
media_has_been_sent_ = media_sent;
}
void DEPRECATED_RtpSenderEgress::SetTimestampOffset(uint32_t timestamp) {
MutexLock lock(&lock_);
timestamp_offset_ = timestamp;
}
std::vector<RtpSequenceNumberMap::Info>
DEPRECATED_RtpSenderEgress::GetSentRtpPacketInfos(
rtc::ArrayView<const uint16_t> sequence_numbers) const {
RTC_DCHECK(!sequence_numbers.empty());
if (!need_rtp_packet_infos_) {
return std::vector<RtpSequenceNumberMap::Info>();
}
std::vector<RtpSequenceNumberMap::Info> results;
results.reserve(sequence_numbers.size());
MutexLock lock(&lock_);
for (uint16_t sequence_number : sequence_numbers) {
const auto& info = rtp_sequence_number_map_->Get(sequence_number);
if (!info) {
// The empty vector will be returned. We can delay the clearing
// of the vector until after we exit the critical section.
return std::vector<RtpSequenceNumberMap::Info>();
}
results.push_back(*info);
}
return results;
}
bool DEPRECATED_RtpSenderEgress::HasCorrectSsrc(
const RtpPacketToSend& packet) const {
switch (*packet.packet_type()) {
case RtpPacketMediaType::kAudio:
case RtpPacketMediaType::kVideo:
return packet.Ssrc() == ssrc_;
case RtpPacketMediaType::kRetransmission:
case RtpPacketMediaType::kPadding:
// Both padding and retransmission must be on either the media or the
// RTX stream.
return packet.Ssrc() == rtx_ssrc_ || packet.Ssrc() == ssrc_;
case RtpPacketMediaType::kForwardErrorCorrection:
// FlexFEC is on separate SSRC, ULPFEC uses media SSRC.
return packet.Ssrc() == ssrc_ || packet.Ssrc() == flexfec_ssrc_;
}
return false;
}
void DEPRECATED_RtpSenderEgress::AddPacketToTransportFeedback(
uint16_t packet_id,
const RtpPacketToSend& packet,
const PacedPacketInfo& pacing_info) {
if (transport_feedback_observer_) {
size_t packet_size = packet.payload_size() + packet.padding_size();
if (send_side_bwe_with_overhead_) {
packet_size = packet.size();
}
RtpPacketSendInfo packet_info;
// TODO(bugs.webrtc.org/12713): Remove once downstream usage is gone.
packet_info.ssrc = ssrc_;
packet_info.media_ssrc = ssrc_;
packet_info.transport_sequence_number = packet_id;
packet_info.rtp_sequence_number = packet.SequenceNumber();
packet_info.length = packet_size;
packet_info.pacing_info = pacing_info;
packet_info.packet_type = packet.packet_type();
transport_feedback_observer_->OnAddPacket(packet_info);
}
}
void DEPRECATED_RtpSenderEgress::UpdateDelayStatistics(int64_t capture_time_ms,
int64_t now_ms,
uint32_t ssrc) {
if (!send_side_delay_observer_ || capture_time_ms <= 0)
return;
int avg_delay_ms = 0;
int max_delay_ms = 0;
uint64_t total_packet_send_delay_ms = 0;
{
MutexLock lock(&lock_);
// Compute the max and average of the recent capture-to-send delays.
// The time complexity of the current approach depends on the distribution
// of the delay values. This could be done more efficiently.
// Remove elements older than kSendSideDelayWindowMs.
auto lower_bound =
send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs);
for (auto it = send_delays_.begin(); it != lower_bound; ++it) {
if (max_delay_it_ == it) {
max_delay_it_ = send_delays_.end();
}
sum_delays_ms_ -= it->second;
}
send_delays_.erase(send_delays_.begin(), lower_bound);
if (max_delay_it_ == send_delays_.end()) {
// Removed the previous max. Need to recompute.
RecomputeMaxSendDelay();
}
// Add the new element.
RTC_DCHECK_GE(now_ms, 0);
RTC_DCHECK_LE(now_ms, std::numeric_limits<int64_t>::max() / 2);
RTC_DCHECK_GE(capture_time_ms, 0);
RTC_DCHECK_LE(capture_time_ms, std::numeric_limits<int64_t>::max() / 2);
int64_t diff_ms = now_ms - capture_time_ms;
RTC_DCHECK_GE(diff_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(diff_ms, std::numeric_limits<int>::max());
int new_send_delay = rtc::dchecked_cast<int>(now_ms - capture_time_ms);
SendDelayMap::iterator it;
bool inserted;
std::tie(it, inserted) =
send_delays_.insert(std::make_pair(now_ms, new_send_delay));
if (!inserted) {
// TODO(terelius): If we have multiple delay measurements during the same
// millisecond then we keep the most recent one. It is not clear that this
// is the right decision, but it preserves an earlier behavior.
int previous_send_delay = it->second;
sum_delays_ms_ -= previous_send_delay;
it->second = new_send_delay;
if (max_delay_it_ == it && new_send_delay < previous_send_delay) {
RecomputeMaxSendDelay();
}
}
if (max_delay_it_ == send_delays_.end() ||
it->second >= max_delay_it_->second) {
max_delay_it_ = it;
}
sum_delays_ms_ += new_send_delay;
total_packet_send_delay_ms_ += new_send_delay;
total_packet_send_delay_ms = total_packet_send_delay_ms_;
size_t num_delays = send_delays_.size();
RTC_DCHECK(max_delay_it_ != send_delays_.end());
max_delay_ms = rtc::dchecked_cast<int>(max_delay_it_->second);
int64_t avg_ms = (sum_delays_ms_ + num_delays / 2) / num_delays;
RTC_DCHECK_GE(avg_ms, static_cast<int64_t>(0));
RTC_DCHECK_LE(avg_ms,
static_cast<int64_t>(std::numeric_limits<int>::max()));
avg_delay_ms =
rtc::dchecked_cast<int>((sum_delays_ms_ + num_delays / 2) / num_delays);
}
send_side_delay_observer_->SendSideDelayUpdated(
avg_delay_ms, max_delay_ms, total_packet_send_delay_ms, ssrc);
}
void DEPRECATED_RtpSenderEgress::RecomputeMaxSendDelay() {
max_delay_it_ = send_delays_.begin();
for (auto it = send_delays_.begin(); it != send_delays_.end(); ++it) {
if (it->second >= max_delay_it_->second) {
max_delay_it_ = it;
}
}
}
void DEPRECATED_RtpSenderEgress::UpdateOnSendPacket(int packet_id,
int64_t capture_time_ms,
uint32_t ssrc) {
if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1) {
return;
}
send_packet_observer_->OnSendPacket(packet_id, capture_time_ms, ssrc);
}
bool DEPRECATED_RtpSenderEgress::SendPacketToNetwork(
const RtpPacketToSend& packet,
const PacketOptions& options,
const PacedPacketInfo& pacing_info) {
int bytes_sent = -1;
if (transport_) {
bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
? static_cast<int>(packet.size())
: -1;
if (event_log_ && bytes_sent > 0) {
event_log_->Log(std::make_unique<RtcEventRtpPacketOutgoing>(
packet, pacing_info.probe_cluster_id));
}
}
if (bytes_sent <= 0) {
RTC_LOG(LS_WARNING) << "Transport failed to send packet.";
return false;
}
return true;
}
void DEPRECATED_RtpSenderEgress::UpdateRtpStats(const RtpPacketToSend& packet) {
int64_t now_ms = clock_->TimeInMilliseconds();
StreamDataCounters* counters =
packet.Ssrc() == rtx_ssrc_ ? &rtx_rtp_stats_ : &rtp_stats_;
if (counters->first_packet_time_ms == -1) {
counters->first_packet_time_ms = now_ms;
}
if (packet.packet_type() == RtpPacketMediaType::kForwardErrorCorrection) {
counters->fec.AddPacket(packet);
}
if (packet.packet_type() == RtpPacketMediaType::kRetransmission) {
counters->retransmitted.AddPacket(packet);
}
counters->transmitted.AddPacket(packet);
RTC_DCHECK(packet.packet_type().has_value());
send_rates_[static_cast<size_t>(*packet.packet_type())].Update(packet.size(),
now_ms);
if (rtp_stats_callback_) {
rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc());
}
}
} // namespace webrtc