modules/rtp_rtcp/source/receive_statistics_impl.cc

/*
 *  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 <cstdlib>
#include <vector>

#include "webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
#include "webrtc/modules/rtp_rtcp/source/time_util.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/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,
    RtcpStatisticsCallback* rtcp_callback,
    StreamDataCountersCallback* rtp_callback)
    : ssrc_(ssrc),
      clock_(clock),
      incoming_bitrate_(kStatisticsProcessIntervalMs,
                        RateStatistics::kBpsScale),
      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) {
  auto counters = UpdateCounters(header, packet_length, retransmitted);
  rtp_callback_->DataCountersUpdated(counters, ssrc_);
}

StreamDataCounters StreamStatisticianImpl::UpdateCounters(
    const RTPHeader& header,
    size_t packet_length,
    bool retransmitted) {
  rtc::CritScope cs(&stream_lock_);
  bool in_order = InOrderPacketInternal(header.sequenceNumber);
  RTC_DCHECK_EQ(ssrc_, header.ssrc);
  incoming_bitrate_.Update(packet_length, clock_->TimeInMilliseconds());
  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_->CurrentNtpTime();

    // 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;
  return receive_counters_;
}

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 = 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);
  }

  // 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 = std::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::FecPacketReceived(const RTPHeader& header,
                                               size_t packet_length) {
  StreamDataCounters counters;
  {
    rtc::CritScope cs(&stream_lock_);
    receive_counters_.fec.AddPacket(packet_length, header);
    counters = receive_counters_;
  }
  rtp_callback_->DataCountersUpdated(counters, ssrc_);
}

void StreamStatisticianImpl::SetMaxReorderingThreshold(
    int max_reordering_threshold) {
  rtc::CritScope cs(&stream_lock_);
  max_reordering_threshold_ = max_reordering_threshold;
}

bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics,
                                           bool reset) {
  {
    rtc::CritScope cs(&stream_lock_);
    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();
  }

  rtcp_callback_->StatisticsUpdated(*statistics, ssrc_);
  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.packets_lost = cumulative_loss_;
  stats.extended_highest_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_;
  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);
}

void StreamStatisticianImpl::LastReceiveTimeNtp(uint32_t* secs,
                                                uint32_t* frac) const {
  rtc::CritScope cs(&stream_lock_);
  *secs = last_receive_time_ntp_.seconds();
  *frac = last_receive_time_ntp_.fractions();
}

bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
    const RTPHeader& header, int64_t min_rtt) const {
  rtc::CritScope cs(&stream_lock_);
  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 {
  rtc::CritScope cs(&stream_lock_);
  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),
      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;
  {
    rtc::CritScope cs(&receive_statistics_lock_);
    StatisticianImplMap::iterator it = statisticians_.find(header.ssrc);
    if (it != statisticians_.end()) {
      impl = it->second;
    } else {
      impl = new StreamStatisticianImpl(header.ssrc, 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) {
  StreamStatisticianImpl* impl;
  {
    rtc::CritScope cs(&receive_statistics_lock_);
    StatisticianImplMap::iterator it = statisticians_.find(header.ssrc);
    // Ignore FEC if it is the first packet.
    if (it == statisticians_.end())
      return;
    impl = it->second;
  }
  impl->FecPacketReceived(header, packet_length);
}

StatisticianMap ReceiveStatisticsImpl::GetActiveStatisticians() const {
  StatisticianMap active_statisticians;
  rtc::CritScope cs(&receive_statistics_lock_);
  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 {
  rtc::CritScope cs(&receive_statistics_lock_);
  StatisticianImplMap::const_iterator it = statisticians_.find(ssrc);
  if (it == statisticians_.end())
    return NULL;
  return it->second;
}

void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
    int max_reordering_threshold) {
  rtc::CritScope cs(&receive_statistics_lock_);
  for (StatisticianImplMap::iterator it = statisticians_.begin();
       it != statisticians_.end(); ++it) {
    it->second->SetMaxReorderingThreshold(max_reordering_threshold);
  }
}

void ReceiveStatisticsImpl::RegisterRtcpStatisticsCallback(
    RtcpStatisticsCallback* callback) {
  rtc::CritScope cs(&receive_statistics_lock_);
  if (callback != NULL)
    assert(rtcp_stats_callback_ == NULL);
  rtcp_stats_callback_ = callback;
}

void ReceiveStatisticsImpl::StatisticsUpdated(const RtcpStatistics& statistics,
                                              uint32_t ssrc) {
  rtc::CritScope cs(&receive_statistics_lock_);
  if (rtcp_stats_callback_)
    rtcp_stats_callback_->StatisticsUpdated(statistics, ssrc);
}

void ReceiveStatisticsImpl::CNameChanged(const char* cname, uint32_t ssrc) {
  rtc::CritScope cs(&receive_statistics_lock_);
  if (rtcp_stats_callback_)
    rtcp_stats_callback_->CNameChanged(cname, ssrc);
}

void ReceiveStatisticsImpl::RegisterRtpStatisticsCallback(
    StreamDataCountersCallback* callback) {
  rtc::CritScope cs(&receive_statistics_lock_);
  if (callback != NULL)
    assert(rtp_stats_callback_ == NULL);
  rtp_stats_callback_ = callback;
}

void ReceiveStatisticsImpl::DataCountersUpdated(const StreamDataCounters& stats,
                                                uint32_t ssrc) {
  rtc::CritScope cs(&receive_statistics_lock_);
  if (rtp_stats_callback_) {
    rtp_stats_callback_->DataCountersUpdated(stats, ssrc);
  }
}

std::vector<rtcp::ReportBlock> ReceiveStatisticsImpl::RtcpReportBlocks(
    size_t max_blocks) {
  StatisticianMap statisticians = GetActiveStatisticians();
  std::vector<rtcp::ReportBlock> result;
  result.reserve(std::min(max_blocks, statisticians.size()));
  for (auto& statistician : statisticians) {
    // TODO(danilchap): Select statistician subset across multiple calls using
    // round-robin, as described in rfc3550 section 6.4 when single
    // rtcp_module/receive_statistics will be used for more rtp streams.
    if (result.size() == max_blocks)
      break;

    // Do we have receive statistics to send?
    RtcpStatistics stats;
    if (!statistician.second->GetStatistics(&stats, true))
      continue;
    result.emplace_back();
    rtcp::ReportBlock& block = result.back();
    block.SetMediaSsrc(statistician.first);
    block.SetFractionLost(stats.fraction_lost);
    if (!block.SetCumulativeLost(stats.packets_lost)) {
      LOG(LS_WARNING) << "Cumulative lost is oversized.";
      result.pop_back();
      continue;
    }
    block.SetExtHighestSeqNum(stats.extended_highest_sequence_number);
    block.SetJitter(stats.jitter);
  }
  return result;
}

}  // namespace webrtc