blob: b696daacf680aeac0851ec58197bb01bda90c4b9 [file] [log] [blame]
/*
* Copyright (c) 2013 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/receive_statistics_impl.h"
#include <math.h>
#include <cstdlib>
#include <memory>
#include <vector>
#include "absl/memory/memory.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "modules/rtp_rtcp/source/time_util.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
const int64_t kStatisticsTimeoutMs = 8000;
const int64_t kStatisticsProcessIntervalMs = 1000;
StreamStatistician::~StreamStatistician() {}
StreamStatisticianImpl::StreamStatisticianImpl(
uint32_t ssrc,
Clock* clock,
bool enable_retransmit_detection,
int max_reordering_threshold,
RtcpStatisticsCallback* rtcp_callback,
StreamDataCountersCallback* rtp_callback)
: ssrc_(ssrc),
clock_(clock),
incoming_bitrate_(kStatisticsProcessIntervalMs,
RateStatistics::kBpsScale),
max_reordering_threshold_(max_reordering_threshold),
enable_retransmit_detection_(enable_retransmit_detection),
jitter_q4_(0),
cumulative_loss_(0),
last_receive_time_ms_(0),
last_received_timestamp_(0),
received_seq_first_(0),
received_seq_max_(-1),
last_report_inorder_packets_(0),
last_report_old_packets_(0),
last_report_seq_max_(-1),
rtcp_callback_(rtcp_callback),
rtp_callback_(rtp_callback) {}
StreamStatisticianImpl::~StreamStatisticianImpl() = default;
void StreamStatisticianImpl::OnRtpPacket(const RtpPacketReceived& packet) {
StreamDataCounters counters = UpdateCounters(packet);
if (rtp_callback_)
rtp_callback_->DataCountersUpdated(counters, ssrc_);
}
bool StreamStatisticianImpl::UpdateOutOfOrder(const RtpPacketReceived& packet,
int64_t sequence_number,
int64_t now_ms) {
RTC_DCHECK_EQ(sequence_number,
seq_unwrapper_.UnwrapWithoutUpdate(packet.SequenceNumber()));
// Check if |packet| is second packet of a stream restart.
if (received_seq_out_of_order_) {
uint16_t expected_sequence_number = *received_seq_out_of_order_ + 1;
received_seq_out_of_order_ = absl::nullopt;
if (packet.SequenceNumber() == expected_sequence_number) {
// Ignore sequence number gap caused by stream restart for next packet
// loss calculation.
last_report_seq_max_ = sequence_number;
last_report_inorder_packets_ = receive_counters_.transmitted.packets -
receive_counters_.retransmitted.packets;
// As final part of stream restart consider |packet| is not out of order.
return false;
}
}
if (std::abs(sequence_number - received_seq_max_) >
max_reordering_threshold_) {
// Sequence number gap looks too large, wait until next packet to check
// for a stream restart.
received_seq_out_of_order_ = packet.SequenceNumber();
return true;
}
if (sequence_number > received_seq_max_)
return false;
// Old out of order packet, may be retransmit.
if (enable_retransmit_detection_ && IsRetransmitOfOldPacket(packet, now_ms))
receive_counters_.retransmitted.AddPacket(packet);
return true;
}
StreamDataCounters StreamStatisticianImpl::UpdateCounters(
const RtpPacketReceived& packet) {
rtc::CritScope cs(&stream_lock_);
RTC_DCHECK_EQ(ssrc_, packet.Ssrc());
int64_t now_ms = clock_->TimeInMilliseconds();
incoming_bitrate_.Update(packet.size(), now_ms);
receive_counters_.last_packet_received_timestamp_ms = now_ms;
receive_counters_.transmitted.AddPacket(packet);
int64_t sequence_number =
seq_unwrapper_.UnwrapWithoutUpdate(packet.SequenceNumber());
if (!ReceivedRtpPacket()) {
received_seq_first_ = sequence_number;
last_report_seq_max_ = sequence_number - 1;
receive_counters_.first_packet_time_ms = now_ms;
} else if (UpdateOutOfOrder(packet, sequence_number, now_ms)) {
return receive_counters_;
}
// In order packet.
received_seq_max_ = sequence_number;
seq_unwrapper_.UpdateLast(sequence_number);
// If new time stamp and more than one in-order packet received, calculate
// new jitter statistics.
if (packet.Timestamp() != last_received_timestamp_ &&
(receive_counters_.transmitted.packets -
receive_counters_.retransmitted.packets) > 1) {
UpdateJitter(packet, now_ms);
}
last_received_timestamp_ = packet.Timestamp();
last_receive_time_ms_ = now_ms;
return receive_counters_;
}
void StreamStatisticianImpl::UpdateJitter(const RtpPacketReceived& packet,
int64_t receive_time_ms) {
int64_t receive_diff_ms = receive_time_ms - last_receive_time_ms_;
RTC_DCHECK_GE(receive_diff_ms, 0);
uint32_t receive_diff_rtp = static_cast<uint32_t>(
(receive_diff_ms * packet.payload_type_frequency()) / 1000);
int32_t time_diff_samples =
receive_diff_rtp - (packet.Timestamp() - last_received_timestamp_);
time_diff_samples = std::abs(time_diff_samples);
// lib_jingle sometimes deliver crazy jumps in TS for the same stream.
// If this happens, don't update jitter value. Use 5 secs video frequency
// as the threshold.
if (time_diff_samples < 450000) {
// Note we calculate in Q4 to avoid using float.
int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_;
jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
}
}
void StreamStatisticianImpl::FecPacketReceived(
const RtpPacketReceived& packet) {
StreamDataCounters counters;
{
rtc::CritScope cs(&stream_lock_);
receive_counters_.fec.AddPacket(packet);
counters = receive_counters_;
}
if (rtp_callback_)
rtp_callback_->DataCountersUpdated(counters, ssrc_);
}
void StreamStatisticianImpl::SetMaxReorderingThreshold(
int max_reordering_threshold) {
rtc::CritScope cs(&stream_lock_);
max_reordering_threshold_ = max_reordering_threshold;
}
void StreamStatisticianImpl::EnableRetransmitDetection(bool enable) {
rtc::CritScope cs(&stream_lock_);
enable_retransmit_detection_ = enable;
}
bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics,
bool reset) {
{
rtc::CritScope cs(&stream_lock_);
if (!ReceivedRtpPacket()) {
return false;
}
if (!reset) {
if (last_report_inorder_packets_ == 0) {
// No report.
return false;
}
// Just get last report.
*statistics = last_reported_statistics_;
return true;
}
*statistics = CalculateRtcpStatistics();
}
if (rtcp_callback_)
rtcp_callback_->StatisticsUpdated(*statistics, ssrc_);
return true;
}
bool StreamStatisticianImpl::GetActiveStatisticsAndReset(
RtcpStatistics* statistics) {
{
rtc::CritScope cs(&stream_lock_);
if (clock_->TimeInMilliseconds() - last_receive_time_ms_ >=
kStatisticsTimeoutMs) {
// Not active.
return false;
}
if (!ReceivedRtpPacket()) {
return false;
}
*statistics = CalculateRtcpStatistics();
}
if (rtcp_callback_)
rtcp_callback_->StatisticsUpdated(*statistics, ssrc_);
return true;
}
RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics() {
RtcpStatistics stats;
// Calculate fraction lost.
int64_t exp_since_last = received_seq_max_ - last_report_seq_max_;
RTC_DCHECK_GE(exp_since_last, 0);
// Number of received RTP packets since last report, counts all packets but
// not re-transmissions.
uint32_t rec_since_last = (receive_counters_.transmitted.packets -
receive_counters_.retransmitted.packets) -
last_report_inorder_packets_;
// With NACK we don't know the expected retransmissions during the last
// second. We know how many "old" packets we have received. We just count
// the number of old received to estimate the loss, but it still does not
// guarantee an exact number since we run this based on time triggered by
// sending of an RTP packet. This should have a minimum effect.
// With NACK we don't count old packets as received since they are
// re-transmitted. We use RTT to decide if a packet is re-ordered or
// re-transmitted.
uint32_t retransmitted_packets =
receive_counters_.retransmitted.packets - last_report_old_packets_;
rec_since_last += retransmitted_packets;
int32_t missing = 0;
if (exp_since_last > rec_since_last) {
missing = (exp_since_last - rec_since_last);
}
uint8_t local_fraction_lost = 0;
if (exp_since_last) {
// Scale 0 to 255, where 255 is 100% loss.
local_fraction_lost = static_cast<uint8_t>(255 * missing / exp_since_last);
}
stats.fraction_lost = local_fraction_lost;
// We need a counter for cumulative loss too.
// TODO(danilchap): Ensure cumulative loss is below maximum value of 2^24.
cumulative_loss_ += missing;
stats.packets_lost = cumulative_loss_;
stats.extended_highest_sequence_number =
static_cast<uint32_t>(received_seq_max_);
// Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
stats.jitter = jitter_q4_ >> 4;
// Store this report.
last_reported_statistics_ = stats;
// Only for report blocks in RTCP SR and RR.
last_report_inorder_packets_ = receive_counters_.transmitted.packets -
receive_counters_.retransmitted.packets;
last_report_old_packets_ = receive_counters_.retransmitted.packets;
last_report_seq_max_ = received_seq_max_;
BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "cumulative_loss_pkts",
clock_->TimeInMilliseconds(),
cumulative_loss_, ssrc_);
BWE_TEST_LOGGING_PLOT_WITH_SSRC(
1, "received_seq_max_pkts", clock_->TimeInMilliseconds(),
(received_seq_max_ - received_seq_first_), ssrc_);
return stats;
}
void StreamStatisticianImpl::GetDataCounters(size_t* bytes_received,
uint32_t* packets_received) const {
rtc::CritScope cs(&stream_lock_);
if (bytes_received) {
*bytes_received = receive_counters_.transmitted.payload_bytes +
receive_counters_.transmitted.header_bytes +
receive_counters_.transmitted.padding_bytes;
}
if (packets_received) {
*packets_received = receive_counters_.transmitted.packets;
}
}
void StreamStatisticianImpl::GetReceiveStreamDataCounters(
StreamDataCounters* data_counters) const {
rtc::CritScope cs(&stream_lock_);
*data_counters = receive_counters_;
}
uint32_t StreamStatisticianImpl::BitrateReceived() const {
rtc::CritScope cs(&stream_lock_);
return incoming_bitrate_.Rate(clock_->TimeInMilliseconds()).value_or(0);
}
bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
const RtpPacketReceived& packet,
int64_t now_ms) const {
uint32_t frequency_khz = packet.payload_type_frequency() / 1000;
RTC_DCHECK_GT(frequency_khz, 0);
int64_t time_diff_ms = now_ms - last_receive_time_ms_;
// Diff in time stamp since last received in order.
uint32_t timestamp_diff = packet.Timestamp() - last_received_timestamp_;
uint32_t rtp_time_stamp_diff_ms = timestamp_diff / frequency_khz;
int64_t max_delay_ms = 0;
// Jitter standard deviation in samples.
float jitter_std = sqrt(static_cast<float>(jitter_q4_ >> 4));
// 2 times the standard deviation => 95% confidence.
// And transform to milliseconds by dividing by the frequency in kHz.
max_delay_ms = static_cast<int64_t>((2 * jitter_std) / frequency_khz);
// Min max_delay_ms is 1.
if (max_delay_ms == 0) {
max_delay_ms = 1;
}
return time_diff_ms > rtp_time_stamp_diff_ms + max_delay_ms;
}
std::unique_ptr<ReceiveStatistics> ReceiveStatistics::Create(
Clock* clock,
RtcpStatisticsCallback* rtcp_callback,
StreamDataCountersCallback* rtp_callback) {
return absl::make_unique<ReceiveStatisticsImpl>(clock, rtcp_callback,
rtp_callback);
}
ReceiveStatisticsImpl::ReceiveStatisticsImpl(
Clock* clock,
RtcpStatisticsCallback* rtcp_callback,
StreamDataCountersCallback* rtp_callback)
: clock_(clock),
last_returned_ssrc_(0),
max_reordering_threshold_(kDefaultMaxReorderingThreshold),
rtcp_stats_callback_(rtcp_callback),
rtp_stats_callback_(rtp_callback) {}
ReceiveStatisticsImpl::~ReceiveStatisticsImpl() {
while (!statisticians_.empty()) {
delete statisticians_.begin()->second;
statisticians_.erase(statisticians_.begin());
}
}
void ReceiveStatisticsImpl::OnRtpPacket(const RtpPacketReceived& packet) {
StreamStatisticianImpl* impl;
{
rtc::CritScope cs(&receive_statistics_lock_);
auto it = statisticians_.find(packet.Ssrc());
if (it != statisticians_.end()) {
impl = it->second;
} else {
impl = new StreamStatisticianImpl(
packet.Ssrc(), clock_, /* enable_retransmit_detection = */ false,
max_reordering_threshold_, rtcp_stats_callback_, rtp_stats_callback_);
statisticians_[packet.Ssrc()] = impl;
}
}
// StreamStatisticianImpl instance is created once and only destroyed when
// this whole ReceiveStatisticsImpl is destroyed. StreamStatisticianImpl has
// it's own locking so don't hold receive_statistics_lock_ (potential
// deadlock).
impl->OnRtpPacket(packet);
}
void ReceiveStatisticsImpl::FecPacketReceived(const RtpPacketReceived& packet) {
StreamStatisticianImpl* impl;
{
rtc::CritScope cs(&receive_statistics_lock_);
auto it = statisticians_.find(packet.Ssrc());
// Ignore FEC if it is the first packet.
if (it == statisticians_.end())
return;
impl = it->second;
}
impl->FecPacketReceived(packet);
}
StreamStatistician* ReceiveStatisticsImpl::GetStatistician(
uint32_t ssrc) const {
rtc::CritScope cs(&receive_statistics_lock_);
auto it = statisticians_.find(ssrc);
if (it == statisticians_.end())
return NULL;
return it->second;
}
void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
int max_reordering_threshold) {
std::map<uint32_t, StreamStatisticianImpl*> statisticians;
{
rtc::CritScope cs(&receive_statistics_lock_);
max_reordering_threshold_ = max_reordering_threshold;
statisticians = statisticians_;
}
for (auto& statistician : statisticians) {
statistician.second->SetMaxReorderingThreshold(max_reordering_threshold);
}
}
void ReceiveStatisticsImpl::EnableRetransmitDetection(uint32_t ssrc,
bool enable) {
StreamStatisticianImpl* impl;
{
rtc::CritScope cs(&receive_statistics_lock_);
StreamStatisticianImpl*& impl_ref = statisticians_[ssrc];
if (impl_ref == nullptr) { // new element
impl_ref = new StreamStatisticianImpl(
ssrc, clock_, enable, max_reordering_threshold_, rtcp_stats_callback_,
rtp_stats_callback_);
return;
}
impl = impl_ref;
}
impl->EnableRetransmitDetection(enable);
}
std::vector<rtcp::ReportBlock> ReceiveStatisticsImpl::RtcpReportBlocks(
size_t max_blocks) {
std::map<uint32_t, StreamStatisticianImpl*> statisticians;
{
rtc::CritScope cs(&receive_statistics_lock_);
statisticians = statisticians_;
}
std::vector<rtcp::ReportBlock> result;
result.reserve(std::min(max_blocks, statisticians.size()));
auto add_report_block = [&result](uint32_t media_ssrc,
StreamStatisticianImpl* statistician) {
// Do we have receive statistics to send?
RtcpStatistics stats;
if (!statistician->GetActiveStatisticsAndReset(&stats))
return;
result.emplace_back();
rtcp::ReportBlock& block = result.back();
block.SetMediaSsrc(media_ssrc);
block.SetFractionLost(stats.fraction_lost);
if (!block.SetCumulativeLost(stats.packets_lost)) {
RTC_LOG(LS_WARNING) << "Cumulative lost is oversized.";
result.pop_back();
return;
}
block.SetExtHighestSeqNum(stats.extended_highest_sequence_number);
block.SetJitter(stats.jitter);
};
const auto start_it = statisticians.upper_bound(last_returned_ssrc_);
for (auto it = start_it;
result.size() < max_blocks && it != statisticians.end(); ++it)
add_report_block(it->first, it->second);
for (auto it = statisticians.begin();
result.size() < max_blocks && it != start_it; ++it)
add_report_block(it->first, it->second);
if (!result.empty())
last_returned_ssrc_ = result.back().source_ssrc();
return result;
}
} // namespace webrtc