Files
platform-external-webrtc/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
danilchap a72e7349d5 [rtp_rtcp] cleanup in RTCPSender class internals.
PrepareReportBlock and AddReportBlock private functions merged:
  PrepareReportBlock moved report block from statistic to temporary structure
  AddReportBlock copied that temporary structure into temporary map right after.
  Thanks to rtcp packet classes that temporary structure is now unneccesary.

BUG=webrtc:5260
R=åsapersson

Review URL: https://codereview.webrtc.org/1538833002

Cr-Commit-Position: refs/heads/master@{#11112}
2015-12-22 16:07:48 +00:00

542 lines
18 KiB
C++

/*
* 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 "webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h"
#include <math.h>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/modules/rtp_rtcp/source/time_util.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
namespace webrtc {
const int64_t kStatisticsTimeoutMs = 8000;
const int64_t kStatisticsProcessIntervalMs = 1000;
StreamStatistician::~StreamStatistician() {}
StreamStatisticianImpl::StreamStatisticianImpl(
Clock* clock,
RtcpStatisticsCallback* rtcp_callback,
StreamDataCountersCallback* rtp_callback)
: clock_(clock),
stream_lock_(CriticalSectionWrapper::CreateCriticalSection()),
incoming_bitrate_(clock, NULL),
ssrc_(0),
max_reordering_threshold_(kDefaultMaxReorderingThreshold),
jitter_q4_(0),
cumulative_loss_(0),
jitter_q4_transmission_time_offset_(0),
last_receive_time_ms_(0),
last_received_timestamp_(0),
last_received_transmission_time_offset_(0),
received_seq_first_(0),
received_seq_max_(0),
received_seq_wraps_(0),
received_packet_overhead_(12),
last_report_inorder_packets_(0),
last_report_old_packets_(0),
last_report_seq_max_(0),
rtcp_callback_(rtcp_callback),
rtp_callback_(rtp_callback) {}
void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
size_t packet_length,
bool retransmitted) {
UpdateCounters(header, packet_length, retransmitted);
NotifyRtpCallback();
}
void StreamStatisticianImpl::UpdateCounters(const RTPHeader& header,
size_t packet_length,
bool retransmitted) {
CriticalSectionScoped cs(stream_lock_.get());
bool in_order = InOrderPacketInternal(header.sequenceNumber);
ssrc_ = header.ssrc;
incoming_bitrate_.Update(packet_length);
receive_counters_.transmitted.AddPacket(packet_length, header);
if (!in_order && retransmitted) {
receive_counters_.retransmitted.AddPacket(packet_length, header);
}
if (receive_counters_.transmitted.packets == 1) {
received_seq_first_ = header.sequenceNumber;
receive_counters_.first_packet_time_ms = clock_->TimeInMilliseconds();
}
// Count only the new packets received. That is, if packets 1, 2, 3, 5, 4, 6
// are received, 4 will be ignored.
if (in_order) {
// Current time in samples.
NtpTime receive_time(*clock_);
// Wrong if we use RetransmitOfOldPacket.
if (receive_counters_.transmitted.packets > 1 &&
received_seq_max_ > header.sequenceNumber) {
// Wrap around detected.
received_seq_wraps_++;
}
// New max.
received_seq_max_ = header.sequenceNumber;
// If new time stamp and more than one in-order packet received, calculate
// new jitter statistics.
if (header.timestamp != last_received_timestamp_ &&
(receive_counters_.transmitted.packets -
receive_counters_.retransmitted.packets) > 1) {
UpdateJitter(header, receive_time);
}
last_received_timestamp_ = header.timestamp;
last_receive_time_ntp_ = receive_time;
last_receive_time_ms_ = clock_->TimeInMilliseconds();
}
size_t packet_oh = header.headerLength + header.paddingLength;
// Our measured overhead. Filter from RFC 5104 4.2.1.2:
// avg_OH (new) = 15/16*avg_OH (old) + 1/16*pckt_OH,
received_packet_overhead_ = (15 * received_packet_overhead_ + packet_oh) >> 4;
}
void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
NtpTime receive_time) {
uint32_t receive_time_rtp =
NtpToRtp(receive_time, header.payload_type_frequency);
uint32_t last_receive_time_rtp =
NtpToRtp(last_receive_time_ntp_, header.payload_type_frequency);
int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
(header.timestamp - last_received_timestamp_);
time_diff_samples = 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);
}
// Extended jitter report, RFC 5450.
// Actual network jitter, excluding the source-introduced jitter.
int32_t time_diff_samples_ext =
(receive_time_rtp - last_receive_time_rtp) -
((header.timestamp +
header.extension.transmissionTimeOffset) -
(last_received_timestamp_ +
last_received_transmission_time_offset_));
time_diff_samples_ext = abs(time_diff_samples_ext);
if (time_diff_samples_ext < 450000) {
int32_t jitter_diffQ4TransmissionTimeOffset =
(time_diff_samples_ext << 4) - jitter_q4_transmission_time_offset_;
jitter_q4_transmission_time_offset_ +=
((jitter_diffQ4TransmissionTimeOffset + 8) >> 4);
}
}
void StreamStatisticianImpl::NotifyRtpCallback() {
StreamDataCounters data;
uint32_t ssrc;
{
CriticalSectionScoped cs(stream_lock_.get());
data = receive_counters_;
ssrc = ssrc_;
}
rtp_callback_->DataCountersUpdated(data, ssrc);
}
void StreamStatisticianImpl::NotifyRtcpCallback() {
RtcpStatistics data;
uint32_t ssrc;
{
CriticalSectionScoped cs(stream_lock_.get());
data = last_reported_statistics_;
ssrc = ssrc_;
}
rtcp_callback_->StatisticsUpdated(data, ssrc);
}
void StreamStatisticianImpl::FecPacketReceived(const RTPHeader& header,
size_t packet_length) {
{
CriticalSectionScoped cs(stream_lock_.get());
receive_counters_.fec.AddPacket(packet_length, header);
}
NotifyRtpCallback();
}
void StreamStatisticianImpl::SetMaxReorderingThreshold(
int max_reordering_threshold) {
CriticalSectionScoped cs(stream_lock_.get());
max_reordering_threshold_ = max_reordering_threshold;
}
bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics,
bool reset) {
{
CriticalSectionScoped cs(stream_lock_.get());
if (received_seq_first_ == 0 &&
receive_counters_.transmitted.payload_bytes == 0) {
// We have not received anything.
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();
}
NotifyRtcpCallback();
return true;
}
RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics() {
RtcpStatistics stats;
if (last_report_inorder_packets_ == 0) {
// First time we send a report.
last_report_seq_max_ = received_seq_first_ - 1;
}
// Calculate fraction lost.
uint16_t exp_since_last = (received_seq_max_ - last_report_seq_max_);
if (last_report_seq_max_ > received_seq_max_) {
// Can we assume that the seq_num can't go decrease over a full RTCP period?
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.cumulative_lost = cumulative_loss_;
stats.extended_max_sequence_number =
(received_seq_wraps_ << 16) + 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_;
return stats;
}
void StreamStatisticianImpl::GetDataCounters(
size_t* bytes_received, uint32_t* packets_received) const {
CriticalSectionScoped cs(stream_lock_.get());
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 {
CriticalSectionScoped cs(stream_lock_.get());
*data_counters = receive_counters_;
}
uint32_t StreamStatisticianImpl::BitrateReceived() const {
CriticalSectionScoped cs(stream_lock_.get());
return incoming_bitrate_.BitrateNow();
}
void StreamStatisticianImpl::ProcessBitrate() {
CriticalSectionScoped cs(stream_lock_.get());
incoming_bitrate_.Process();
}
void StreamStatisticianImpl::LastReceiveTimeNtp(uint32_t* secs,
uint32_t* frac) const {
CriticalSectionScoped cs(stream_lock_.get());
*secs = last_receive_time_ntp_.seconds();
*frac = last_receive_time_ntp_.fractions();
}
bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
const RTPHeader& header, int64_t min_rtt) const {
CriticalSectionScoped cs(stream_lock_.get());
if (InOrderPacketInternal(header.sequenceNumber)) {
return false;
}
uint32_t frequency_khz = header.payload_type_frequency / 1000;
assert(frequency_khz > 0);
int64_t time_diff_ms = clock_->TimeInMilliseconds() -
last_receive_time_ms_;
// Diff in time stamp since last received in order.
uint32_t timestamp_diff = header.timestamp - last_received_timestamp_;
uint32_t rtp_time_stamp_diff_ms = timestamp_diff / frequency_khz;
int64_t max_delay_ms = 0;
if (min_rtt == 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;
}
} else {
max_delay_ms = (min_rtt / 3) + 1;
}
return time_diff_ms > rtp_time_stamp_diff_ms + max_delay_ms;
}
bool StreamStatisticianImpl::IsPacketInOrder(uint16_t sequence_number) const {
CriticalSectionScoped cs(stream_lock_.get());
return InOrderPacketInternal(sequence_number);
}
bool StreamStatisticianImpl::InOrderPacketInternal(
uint16_t sequence_number) const {
// First packet is always in order.
if (last_receive_time_ms_ == 0)
return true;
if (IsNewerSequenceNumber(sequence_number, received_seq_max_)) {
return true;
} else {
// If we have a restart of the remote side this packet is still in order.
return !IsNewerSequenceNumber(sequence_number, received_seq_max_ -
max_reordering_threshold_);
}
}
ReceiveStatistics* ReceiveStatistics::Create(Clock* clock) {
return new ReceiveStatisticsImpl(clock);
}
ReceiveStatisticsImpl::ReceiveStatisticsImpl(Clock* clock)
: clock_(clock),
receive_statistics_lock_(CriticalSectionWrapper::CreateCriticalSection()),
last_rate_update_ms_(0),
rtcp_stats_callback_(NULL),
rtp_stats_callback_(NULL) {}
ReceiveStatisticsImpl::~ReceiveStatisticsImpl() {
while (!statisticians_.empty()) {
delete statisticians_.begin()->second;
statisticians_.erase(statisticians_.begin());
}
}
void ReceiveStatisticsImpl::IncomingPacket(const RTPHeader& header,
size_t packet_length,
bool retransmitted) {
StreamStatisticianImpl* impl;
{
CriticalSectionScoped cs(receive_statistics_lock_.get());
StatisticianImplMap::iterator it = statisticians_.find(header.ssrc);
if (it != statisticians_.end()) {
impl = it->second;
} else {
impl = new StreamStatisticianImpl(clock_, this, this);
statisticians_[header.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->IncomingPacket(header, packet_length, retransmitted);
}
void ReceiveStatisticsImpl::FecPacketReceived(const RTPHeader& header,
size_t packet_length) {
CriticalSectionScoped cs(receive_statistics_lock_.get());
StatisticianImplMap::iterator it = statisticians_.find(header.ssrc);
// Ignore FEC if it is the first packet.
if (it != statisticians_.end()) {
it->second->FecPacketReceived(header, packet_length);
}
}
StatisticianMap ReceiveStatisticsImpl::GetActiveStatisticians() const {
CriticalSectionScoped cs(receive_statistics_lock_.get());
StatisticianMap active_statisticians;
for (StatisticianImplMap::const_iterator it = statisticians_.begin();
it != statisticians_.end(); ++it) {
uint32_t secs;
uint32_t frac;
it->second->LastReceiveTimeNtp(&secs, &frac);
if (clock_->CurrentNtpInMilliseconds() -
Clock::NtpToMs(secs, frac) < kStatisticsTimeoutMs) {
active_statisticians[it->first] = it->second;
}
}
return active_statisticians;
}
StreamStatistician* ReceiveStatisticsImpl::GetStatistician(
uint32_t ssrc) const {
CriticalSectionScoped cs(receive_statistics_lock_.get());
StatisticianImplMap::const_iterator it = statisticians_.find(ssrc);
if (it == statisticians_.end())
return NULL;
return it->second;
}
void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
int max_reordering_threshold) {
CriticalSectionScoped cs(receive_statistics_lock_.get());
for (StatisticianImplMap::iterator it = statisticians_.begin();
it != statisticians_.end(); ++it) {
it->second->SetMaxReorderingThreshold(max_reordering_threshold);
}
}
int32_t ReceiveStatisticsImpl::Process() {
CriticalSectionScoped cs(receive_statistics_lock_.get());
for (StatisticianImplMap::iterator it = statisticians_.begin();
it != statisticians_.end(); ++it) {
it->second->ProcessBitrate();
}
last_rate_update_ms_ = clock_->TimeInMilliseconds();
return 0;
}
int64_t ReceiveStatisticsImpl::TimeUntilNextProcess() {
CriticalSectionScoped cs(receive_statistics_lock_.get());
int64_t time_since_last_update = clock_->TimeInMilliseconds() -
last_rate_update_ms_;
return std::max<int64_t>(
kStatisticsProcessIntervalMs - time_since_last_update, 0);
}
void ReceiveStatisticsImpl::RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) {
CriticalSectionScoped cs(receive_statistics_lock_.get());
if (callback != NULL)
assert(rtcp_stats_callback_ == NULL);
rtcp_stats_callback_ = callback;
}
void ReceiveStatisticsImpl::StatisticsUpdated(const RtcpStatistics& statistics,
uint32_t ssrc) {
CriticalSectionScoped cs(receive_statistics_lock_.get());
if (rtcp_stats_callback_)
rtcp_stats_callback_->StatisticsUpdated(statistics, ssrc);
}
void ReceiveStatisticsImpl::CNameChanged(const char* cname, uint32_t ssrc) {
CriticalSectionScoped cs(receive_statistics_lock_.get());
if (rtcp_stats_callback_)
rtcp_stats_callback_->CNameChanged(cname, ssrc);
}
void ReceiveStatisticsImpl::RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) {
CriticalSectionScoped cs(receive_statistics_lock_.get());
if (callback != NULL)
assert(rtp_stats_callback_ == NULL);
rtp_stats_callback_ = callback;
}
void ReceiveStatisticsImpl::DataCountersUpdated(const StreamDataCounters& stats,
uint32_t ssrc) {
CriticalSectionScoped cs(receive_statistics_lock_.get());
if (rtp_stats_callback_) {
rtp_stats_callback_->DataCountersUpdated(stats, ssrc);
}
}
void NullReceiveStatistics::IncomingPacket(const RTPHeader& rtp_header,
size_t packet_length,
bool retransmitted) {}
void NullReceiveStatistics::FecPacketReceived(const RTPHeader& header,
size_t packet_length) {}
StatisticianMap NullReceiveStatistics::GetActiveStatisticians() const {
return StatisticianMap();
}
StreamStatistician* NullReceiveStatistics::GetStatistician(
uint32_t ssrc) const {
return NULL;
}
void NullReceiveStatistics::SetMaxReorderingThreshold(
int max_reordering_threshold) {}
int64_t NullReceiveStatistics::TimeUntilNextProcess() { return 0; }
int32_t NullReceiveStatistics::Process() { return 0; }
void NullReceiveStatistics::RegisterRtcpStatisticsCallback(
RtcpStatisticsCallback* callback) {}
void NullReceiveStatistics::RegisterRtpStatisticsCallback(
StreamDataCountersCallback* callback) {}
} // namespace webrtc