modules/rtp_rtcp/source/receive_statistics_impl.cc - src - Git at Google

 /*
  *  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 <cmath>
 #include <cstdlib>
 #include <memory>
 #include <utility>
 #include <vector>

 #include "api/units/time_delta.h"
 #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/time_utils.h"
 #include "system_wrappers/include/clock.h"

 namespace webrtc {
 namespace {
 constexpr int64_t kStatisticsTimeoutMs = 8000;
 constexpr int64_t kStatisticsProcessIntervalMs = 1000;
 }  // namespace

 StreamStatistician::~StreamStatistician() {}

 StreamStatisticianImpl::StreamStatisticianImpl(uint32_t ssrc, Clock* clock,
                                                int max_reordering_threshold)
     : ssrc_(ssrc),
       clock_(clock),
       delta_internal_unix_epoch_ms_(clock_->CurrentNtpInMilliseconds() -
                                     clock_->TimeInMilliseconds() -
                                     rtc::kNtpJan1970Millisecs),
       incoming_bitrate_(kStatisticsProcessIntervalMs,
                         RateStatistics::kBpsScale),
       max_reordering_threshold_(max_reordering_threshold),
       enable_retransmit_detection_(false),
       cumulative_loss_is_capped_(false),
       jitter_q4_(0),
       cumulative_loss_(0),
       cumulative_loss_rtcp_offset_(0),
       last_receive_time_ms_(0),
       last_received_timestamp_(0),
       received_seq_first_(-1),
       received_seq_max_(-1),
       last_report_cumulative_loss_(0),
       last_report_seq_max_(-1),
       last_payload_type_frequency_(0) {}

 StreamStatisticianImpl::~StreamStatisticianImpl() = default;

 bool StreamStatisticianImpl::UpdateOutOfOrder(const RtpPacketReceived& packet,
                                               int64_t sequence_number,
                                               int64_t now_ms) {
   // Check if `packet` is second packet of a stream restart.
   if (received_seq_out_of_order_) {
     // Count the previous packet as a received; it was postponed below.
     --cumulative_loss_;

     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 packet loss
       // calculation, by setting received_seq_max_ to the sequence number just
       // before the out-of-order seqno. This gives a net zero change of
       // `cumulative_loss_`, for the two packets interpreted as a stream reset.
       //
       // Fraction loss for the next report may get a bit off, since we don't
       // update last_report_seq_max_ and last_report_cumulative_loss_ in a
       // consistent way.
       last_report_seq_max_ = sequence_number - 2;
       received_seq_max_ = sequence_number - 2;
       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();
     // Postpone counting this as a received packet until we know how to update
     // `received_seq_max_`, otherwise we temporarily decrement
     // `cumulative_loss_`. The
     // ReceiveStatisticsTest.StreamRestartDoesntCountAsLoss test expects
     // `cumulative_loss_` to be unchanged by the reception of the first packet
     // after stream reset.
     ++cumulative_loss_;
     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;
 }

 void StreamStatisticianImpl::UpdateCounters(const RtpPacketReceived& packet) {
   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);
   --cumulative_loss_;

   // Use PeekUnwrap and later update the state to avoid updating the state for
   // out of order packets.
   int64_t sequence_number = seq_unwrapper_.PeekUnwrap(packet.SequenceNumber());

   if (!ReceivedRtpPacket()) {
     received_seq_first_ = sequence_number;
     last_report_seq_max_ = sequence_number - 1;
     received_seq_max_ = sequence_number - 1;
     receive_counters_.first_packet_time_ms = now_ms;
   } else if (UpdateOutOfOrder(packet, sequence_number, now_ms)) {
     return;
   }
   // In order packet.
   cumulative_loss_ += sequence_number - received_seq_max_;
   received_seq_max_ = sequence_number;
   // Update the internal state of `seq_unwrapper_`.
   seq_unwrapper_.Unwrap(packet.SequenceNumber());

   // 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;
 }

 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);

   ReviseFrequencyAndJitter(packet.payload_type_frequency());

   // 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::ReviseFrequencyAndJitter(
     int payload_type_frequency) {
   if (payload_type_frequency == last_payload_type_frequency_) {
     return;
   }

   if (payload_type_frequency != 0) {
     if (last_payload_type_frequency_ != 0) {
       // Value in "jitter_q4_" variable is a number of samples.
       // I.e. jitter = timestamp (ms) * frequency (kHz).
       // Since the frequency has changed we have to update the number of samples
       // accordingly. The new value should rely on a new frequency.

       // If we don't do such procedure we end up with the number of samples that
       // cannot be converted into milliseconds correctly
       // (i.e. jitter_ms = jitter_q4_ >> 4 / (payload_type_frequency / 1000)).
       // In such case, the number of samples has a "mix".

       // Doing so we pretend that everything prior and including the current
       // packet were computed on packet's frequency.
       jitter_q4_ = static_cast<int>(static_cast<uint64_t>(jitter_q4_) *
                                     payload_type_frequency /
                                     last_payload_type_frequency_);
     }
     // If last_payload_type_frequency_ is not present, the jitter_q4_
     // variable has its initial value.

     // Keep last_payload_type_frequency_ up to date and non-zero (set).
     last_payload_type_frequency_ = payload_type_frequency;
   }
 }

 void StreamStatisticianImpl::SetMaxReorderingThreshold(
     int max_reordering_threshold) {
   max_reordering_threshold_ = max_reordering_threshold;
 }

 void StreamStatisticianImpl::EnableRetransmitDetection(bool enable) {
   enable_retransmit_detection_ = enable;
 }

 RtpReceiveStats StreamStatisticianImpl::GetStats() const {
   RtpReceiveStats stats;
   stats.packets_lost = cumulative_loss_;
   // Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
   stats.jitter = jitter_q4_ >> 4;
   if (last_payload_type_frequency_ > 0) {
     // Divide value in fractional seconds by frequency to get jitter in
     // fractional seconds.
     stats.interarrival_jitter =
         webrtc::TimeDelta::Seconds(stats.jitter) / last_payload_type_frequency_;
   }
   if (receive_counters_.last_packet_received_timestamp_ms.has_value()) {
     stats.last_packet_received_timestamp_ms =
         *receive_counters_.last_packet_received_timestamp_ms +
         delta_internal_unix_epoch_ms_;
   }
   stats.packet_counter = receive_counters_.transmitted;
   return stats;
 }

 void StreamStatisticianImpl::MaybeAppendReportBlockAndReset(
     std::vector<rtcp::ReportBlock>& report_blocks) {
   int64_t now_ms = clock_->TimeInMilliseconds();
   if (now_ms - last_receive_time_ms_ >= kStatisticsTimeoutMs) {
     // Not active.
     return;
   }
   if (!ReceivedRtpPacket()) {
     return;
   }

   report_blocks.emplace_back();
   rtcp::ReportBlock& stats = report_blocks.back();
   stats.SetMediaSsrc(ssrc_);
   // Calculate fraction lost.
   int64_t exp_since_last = received_seq_max_ - last_report_seq_max_;
   RTC_DCHECK_GE(exp_since_last, 0);

   int32_t lost_since_last = cumulative_loss_ - last_report_cumulative_loss_;
   if (exp_since_last > 0 && lost_since_last > 0) {
     // Scale 0 to 255, where 255 is 100% loss.
     stats.SetFractionLost(255 * lost_since_last / exp_since_last);
   }

   int packets_lost = cumulative_loss_ + cumulative_loss_rtcp_offset_;
   if (packets_lost < 0) {
     // Clamp to zero. Work around to accomodate for senders that misbehave with
     // negative cumulative loss.
     packets_lost = 0;
     cumulative_loss_rtcp_offset_ = -cumulative_loss_;
   }
   if (packets_lost > 0x7fffff) {
     // Packets lost is a 24 bit signed field, and thus should be clamped, as
     // described in https://datatracker.ietf.org/doc/html/rfc3550#appendix-A.3
     if (!cumulative_loss_is_capped_) {
       cumulative_loss_is_capped_ = true;
       RTC_LOG(LS_WARNING) << "Cumulative loss reached maximum value for ssrc "
                           << ssrc_;
     }
     packets_lost = 0x7fffff;
   }
   stats.SetCumulativeLost(packets_lost);
   stats.SetExtHighestSeqNum(received_seq_max_);
   // Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
   stats.SetJitter(jitter_q4_ >> 4);

   // Only for report blocks in RTCP SR and RR.
   last_report_cumulative_loss_ = cumulative_loss_;
   last_report_seq_max_ = received_seq_max_;
   BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "cumulative_loss_pkts", now_ms,
                                   cumulative_loss_, ssrc_);
   BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "received_seq_max_pkts", now_ms,
                                   (received_seq_max_ - received_seq_first_),
                                   ssrc_);
 }

 absl::optional<int> StreamStatisticianImpl::GetFractionLostInPercent() const {
   if (!ReceivedRtpPacket()) {
     return absl::nullopt;
   }
   int64_t expected_packets = 1 + received_seq_max_ - received_seq_first_;
   if (expected_packets <= 0) {
     return absl::nullopt;
   }
   if (cumulative_loss_ <= 0) {
     return 0;
   }
   return 100 * static_cast<int64_t>(cumulative_loss_) / expected_packets;
 }

 StreamDataCounters StreamStatisticianImpl::GetReceiveStreamDataCounters()
     const {
   return receive_counters_;
 }

 uint32_t StreamStatisticianImpl::BitrateReceived() const {
   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 = 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) {
   return std::make_unique<ReceiveStatisticsLocked>(
       clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) {
         return std::make_unique<StreamStatisticianLocked>(
             ssrc, clock, max_reordering_threshold);
       });
 }

 std::unique_ptr<ReceiveStatistics> ReceiveStatistics::CreateThreadCompatible(
     Clock* clock) {
   return std::make_unique<ReceiveStatisticsImpl>(
       clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) {
         return std::make_unique<StreamStatisticianImpl>(
             ssrc, clock, max_reordering_threshold);
       });
 }

 ReceiveStatisticsImpl::ReceiveStatisticsImpl(
     Clock* clock,
     std::function<std::unique_ptr<StreamStatisticianImplInterface>(
         uint32_t ssrc,
         Clock* clock,
         int max_reordering_threshold)> stream_statistician_factory)
     : clock_(clock),
       stream_statistician_factory_(std::move(stream_statistician_factory)),
       last_returned_ssrc_idx_(0),
       max_reordering_threshold_(kDefaultMaxReorderingThreshold) {}

 void ReceiveStatisticsImpl::OnRtpPacket(const RtpPacketReceived& packet) {
   // 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).
   GetOrCreateStatistician(packet.Ssrc())->UpdateCounters(packet);
 }

 StreamStatistician* ReceiveStatisticsImpl::GetStatistician(
     uint32_t ssrc) const {
   const auto& it = statisticians_.find(ssrc);
   if (it == statisticians_.end())
     return nullptr;
   return it->second.get();
 }

 StreamStatisticianImplInterface* ReceiveStatisticsImpl::GetOrCreateStatistician(
     uint32_t ssrc) {
   std::unique_ptr<StreamStatisticianImplInterface>& impl = statisticians_[ssrc];
   if (impl == nullptr) {  // new element
     impl =
         stream_statistician_factory_(ssrc, clock_, max_reordering_threshold_);
     all_ssrcs_.push_back(ssrc);
   }
   return impl.get();
 }

 void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
     int max_reordering_threshold) {
   max_reordering_threshold_ = max_reordering_threshold;
   for (auto& statistician : statisticians_) {
     statistician.second->SetMaxReorderingThreshold(max_reordering_threshold);
   }
 }

 void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
     uint32_t ssrc,
     int max_reordering_threshold) {
   GetOrCreateStatistician(ssrc)->SetMaxReorderingThreshold(
       max_reordering_threshold);
 }

 void ReceiveStatisticsImpl::EnableRetransmitDetection(uint32_t ssrc,
                                                       bool enable) {
   GetOrCreateStatistician(ssrc)->EnableRetransmitDetection(enable);
 }

 std::vector<rtcp::ReportBlock> ReceiveStatisticsImpl::RtcpReportBlocks(
     size_t max_blocks) {
   std::vector<rtcp::ReportBlock> result;
   result.reserve(std::min(max_blocks, all_ssrcs_.size()));

   size_t ssrc_idx = 0;
   for (size_t i = 0; i < all_ssrcs_.size() && result.size() < max_blocks; ++i) {
     ssrc_idx = (last_returned_ssrc_idx_ + i + 1) % all_ssrcs_.size();
     const uint32_t media_ssrc = all_ssrcs_[ssrc_idx];
     auto statistician_it = statisticians_.find(media_ssrc);
     RTC_DCHECK(statistician_it != statisticians_.end());
     statistician_it->second->MaybeAppendReportBlockAndReset(result);
   }
   last_returned_ssrc_idx_ = ssrc_idx;
   return result;
 }

 }  // namespace webrtc
	/*
	* 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 <cmath>
	#include <cstdlib>
	#include <memory>
	#include <utility>
	#include <vector>

	#include "api/units/time_delta.h"
	#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
	#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
	#include "modules/rtp_rtcp/source/rtp_packet_received.h"
	#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
	#include "rtc_base/logging.h"
	#include "rtc_base/time_utils.h"
	#include "system_wrappers/include/clock.h"

	namespace webrtc {
	namespace {
	constexpr int64_t kStatisticsTimeoutMs = 8000;
	constexpr int64_t kStatisticsProcessIntervalMs = 1000;
	} // namespace

	StreamStatistician::~StreamStatistician() {}

	StreamStatisticianImpl::StreamStatisticianImpl(uint32_t ssrc, Clock* clock,
	int max_reordering_threshold)
	: ssrc_(ssrc),
	clock_(clock),
	delta_internal_unix_epoch_ms_(clock_->CurrentNtpInMilliseconds() -
	clock_->TimeInMilliseconds() -
	rtc::kNtpJan1970Millisecs),
	incoming_bitrate_(kStatisticsProcessIntervalMs,
	RateStatistics::kBpsScale),
	max_reordering_threshold_(max_reordering_threshold),
	enable_retransmit_detection_(false),
	cumulative_loss_is_capped_(false),
	jitter_q4_(0),
	cumulative_loss_(0),
	cumulative_loss_rtcp_offset_(0),
	last_receive_time_ms_(0),
	last_received_timestamp_(0),
	received_seq_first_(-1),
	received_seq_max_(-1),
	last_report_cumulative_loss_(0),
	last_report_seq_max_(-1),
	last_payload_type_frequency_(0) {}

	StreamStatisticianImpl::~StreamStatisticianImpl() = default;

	bool StreamStatisticianImpl::UpdateOutOfOrder(const RtpPacketReceived& packet,
	int64_t sequence_number,
	int64_t now_ms) {
	// Check if `packet` is second packet of a stream restart.
	if (received_seq_out_of_order_) {
	// Count the previous packet as a received; it was postponed below.
	--cumulative_loss_;

	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 packet loss
	// calculation, by setting received_seq_max_ to the sequence number just
	// before the out-of-order seqno. This gives a net zero change of
	// `cumulative_loss_`, for the two packets interpreted as a stream reset.
	//
	// Fraction loss for the next report may get a bit off, since we don't
	// update last_report_seq_max_ and last_report_cumulative_loss_ in a
	// consistent way.
	last_report_seq_max_ = sequence_number - 2;
	received_seq_max_ = sequence_number - 2;
	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();
	// Postpone counting this as a received packet until we know how to update
	// `received_seq_max_`, otherwise we temporarily decrement
	// `cumulative_loss_`. The
	// ReceiveStatisticsTest.StreamRestartDoesntCountAsLoss test expects
	// `cumulative_loss_` to be unchanged by the reception of the first packet
	// after stream reset.
	++cumulative_loss_;
	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;
	}

	void StreamStatisticianImpl::UpdateCounters(const RtpPacketReceived& packet) {
	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);
	--cumulative_loss_;

	// Use PeekUnwrap and later update the state to avoid updating the state for
	// out of order packets.
	int64_t sequence_number = seq_unwrapper_.PeekUnwrap(packet.SequenceNumber());

	if (!ReceivedRtpPacket()) {
	received_seq_first_ = sequence_number;
	last_report_seq_max_ = sequence_number - 1;
	received_seq_max_ = sequence_number - 1;
	receive_counters_.first_packet_time_ms = now_ms;
	} else if (UpdateOutOfOrder(packet, sequence_number, now_ms)) {
	return;
	}
	// In order packet.
	cumulative_loss_ += sequence_number - received_seq_max_;
	received_seq_max_ = sequence_number;
	// Update the internal state of `seq_unwrapper_`.
	seq_unwrapper_.Unwrap(packet.SequenceNumber());

	// 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;
	}

	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);

	ReviseFrequencyAndJitter(packet.payload_type_frequency());

	// 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::ReviseFrequencyAndJitter(
	int payload_type_frequency) {
	if (payload_type_frequency == last_payload_type_frequency_) {
	return;
	}

	if (payload_type_frequency != 0) {
	if (last_payload_type_frequency_ != 0) {
	// Value in "jitter_q4_" variable is a number of samples.
	// I.e. jitter = timestamp (ms) * frequency (kHz).
	// Since the frequency has changed we have to update the number of samples
	// accordingly. The new value should rely on a new frequency.

	// If we don't do such procedure we end up with the number of samples that
	// cannot be converted into milliseconds correctly
	// (i.e. jitter_ms = jitter_q4_ >> 4 / (payload_type_frequency / 1000)).
	// In such case, the number of samples has a "mix".

	// Doing so we pretend that everything prior and including the current
	// packet were computed on packet's frequency.
	jitter_q4_ = static_cast<int>(static_cast<uint64_t>(jitter_q4_) *
	payload_type_frequency /
	last_payload_type_frequency_);
	}
	// If last_payload_type_frequency_ is not present, the jitter_q4_
	// variable has its initial value.

	// Keep last_payload_type_frequency_ up to date and non-zero (set).
	last_payload_type_frequency_ = payload_type_frequency;
	}
	}

	void StreamStatisticianImpl::SetMaxReorderingThreshold(
	int max_reordering_threshold) {
	max_reordering_threshold_ = max_reordering_threshold;
	}

	void StreamStatisticianImpl::EnableRetransmitDetection(bool enable) {
	enable_retransmit_detection_ = enable;
	}

	RtpReceiveStats StreamStatisticianImpl::GetStats() const {
	RtpReceiveStats stats;
	stats.packets_lost = cumulative_loss_;
	// Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
	stats.jitter = jitter_q4_ >> 4;
	if (last_payload_type_frequency_ > 0) {
	// Divide value in fractional seconds by frequency to get jitter in
	// fractional seconds.
	stats.interarrival_jitter =
	webrtc::TimeDelta::Seconds(stats.jitter) / last_payload_type_frequency_;
	}
	if (receive_counters_.last_packet_received_timestamp_ms.has_value()) {
	stats.last_packet_received_timestamp_ms =
	*receive_counters_.last_packet_received_timestamp_ms +
	delta_internal_unix_epoch_ms_;
	}
	stats.packet_counter = receive_counters_.transmitted;
	return stats;
	}

	void StreamStatisticianImpl::MaybeAppendReportBlockAndReset(
	std::vector<rtcp::ReportBlock>& report_blocks) {
	int64_t now_ms = clock_->TimeInMilliseconds();
	if (now_ms - last_receive_time_ms_ >= kStatisticsTimeoutMs) {
	// Not active.
	return;
	}
	if (!ReceivedRtpPacket()) {
	return;
	}

	report_blocks.emplace_back();
	rtcp::ReportBlock& stats = report_blocks.back();
	stats.SetMediaSsrc(ssrc_);
	// Calculate fraction lost.
	int64_t exp_since_last = received_seq_max_ - last_report_seq_max_;
	RTC_DCHECK_GE(exp_since_last, 0);

	int32_t lost_since_last = cumulative_loss_ - last_report_cumulative_loss_;
	if (exp_since_last > 0 && lost_since_last > 0) {
	// Scale 0 to 255, where 255 is 100% loss.
	stats.SetFractionLost(255 * lost_since_last / exp_since_last);
	}

	int packets_lost = cumulative_loss_ + cumulative_loss_rtcp_offset_;
	if (packets_lost < 0) {
	// Clamp to zero. Work around to accomodate for senders that misbehave with
	// negative cumulative loss.
	packets_lost = 0;
	cumulative_loss_rtcp_offset_ = -cumulative_loss_;
	}
	if (packets_lost > 0x7fffff) {
	// Packets lost is a 24 bit signed field, and thus should be clamped, as
	// described in https://datatracker.ietf.org/doc/html/rfc3550#appendix-A.3
	if (!cumulative_loss_is_capped_) {
	cumulative_loss_is_capped_ = true;
	RTC_LOG(LS_WARNING) << "Cumulative loss reached maximum value for ssrc "
	<< ssrc_;
	}
	packets_lost = 0x7fffff;
	}
	stats.SetCumulativeLost(packets_lost);
	stats.SetExtHighestSeqNum(received_seq_max_);
	// Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
	stats.SetJitter(jitter_q4_ >> 4);

	// Only for report blocks in RTCP SR and RR.
	last_report_cumulative_loss_ = cumulative_loss_;
	last_report_seq_max_ = received_seq_max_;
	BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "cumulative_loss_pkts", now_ms,
	cumulative_loss_, ssrc_);
	BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "received_seq_max_pkts", now_ms,
	(received_seq_max_ - received_seq_first_),
	ssrc_);
	}

	absl::optional<int> StreamStatisticianImpl::GetFractionLostInPercent() const {
	if (!ReceivedRtpPacket()) {
	return absl::nullopt;
	}
	int64_t expected_packets = 1 + received_seq_max_ - received_seq_first_;
	if (expected_packets <= 0) {
	return absl::nullopt;
	}
	if (cumulative_loss_ <= 0) {
	return 0;
	}
	return 100 * static_cast<int64_t>(cumulative_loss_) / expected_packets;
	}

	StreamDataCounters StreamStatisticianImpl::GetReceiveStreamDataCounters()
	const {
	return receive_counters_;
	}

	uint32_t StreamStatisticianImpl::BitrateReceived() const {
	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 = 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) {
	return std::make_unique<ReceiveStatisticsLocked>(
	clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) {
	return std::make_unique<StreamStatisticianLocked>(
	ssrc, clock, max_reordering_threshold);
	});
	}

	std::unique_ptr<ReceiveStatistics> ReceiveStatistics::CreateThreadCompatible(
	Clock* clock) {
	return std::make_unique<ReceiveStatisticsImpl>(
	clock, [](uint32_t ssrc, Clock* clock, int max_reordering_threshold) {
	return std::make_unique<StreamStatisticianImpl>(
	ssrc, clock, max_reordering_threshold);
	});
	}

	ReceiveStatisticsImpl::ReceiveStatisticsImpl(
	Clock* clock,
	std::function<std::unique_ptr<StreamStatisticianImplInterface>(
	uint32_t ssrc,
	Clock* clock,
	int max_reordering_threshold)> stream_statistician_factory)
	: clock_(clock),
	stream_statistician_factory_(std::move(stream_statistician_factory)),
	last_returned_ssrc_idx_(0),
	max_reordering_threshold_(kDefaultMaxReorderingThreshold) {}

	void ReceiveStatisticsImpl::OnRtpPacket(const RtpPacketReceived& packet) {
	// 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).
	GetOrCreateStatistician(packet.Ssrc())->UpdateCounters(packet);
	}

	StreamStatistician* ReceiveStatisticsImpl::GetStatistician(
	uint32_t ssrc) const {
	const auto& it = statisticians_.find(ssrc);
	if (it == statisticians_.end())
	return nullptr;
	return it->second.get();
	}

	StreamStatisticianImplInterface* ReceiveStatisticsImpl::GetOrCreateStatistician(
	uint32_t ssrc) {
	std::unique_ptr<StreamStatisticianImplInterface>& impl = statisticians_[ssrc];
	if (impl == nullptr) { // new element
	impl =
	stream_statistician_factory_(ssrc, clock_, max_reordering_threshold_);
	all_ssrcs_.push_back(ssrc);
	}
	return impl.get();
	}

	void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
	int max_reordering_threshold) {
	max_reordering_threshold_ = max_reordering_threshold;
	for (auto& statistician : statisticians_) {
	statistician.second->SetMaxReorderingThreshold(max_reordering_threshold);
	}
	}

	void ReceiveStatisticsImpl::SetMaxReorderingThreshold(
	uint32_t ssrc,
	int max_reordering_threshold) {
	GetOrCreateStatistician(ssrc)->SetMaxReorderingThreshold(
	max_reordering_threshold);
	}

	void ReceiveStatisticsImpl::EnableRetransmitDetection(uint32_t ssrc,
	bool enable) {
	GetOrCreateStatistician(ssrc)->EnableRetransmitDetection(enable);
	}

	std::vector<rtcp::ReportBlock> ReceiveStatisticsImpl::RtcpReportBlocks(
	size_t max_blocks) {
	std::vector<rtcp::ReportBlock> result;
	result.reserve(std::min(max_blocks, all_ssrcs_.size()));

	size_t ssrc_idx = 0;
	for (size_t i = 0; i < all_ssrcs_.size() && result.size() < max_blocks; ++i) {
	ssrc_idx = (last_returned_ssrc_idx_ + i + 1) % all_ssrcs_.size();
	const uint32_t media_ssrc = all_ssrcs_[ssrc_idx];
	auto statistician_it = statisticians_.find(media_ssrc);
	RTC_DCHECK(statistician_it != statisticians_.end());
	statistician_it->second->MaybeAppendReportBlockAndReset(result);
	}
	last_returned_ssrc_idx_ = ssrc_idx;
	return result;
	}

	} // namespace webrtc