tor-browser

report_block_data.h (5809B)

---

/*
*  Copyright 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.
*/

#ifndef MODULES_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_
#define MODULES_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_

#include <cstddef>
#include <cstdint>

#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"

namespace webrtc {

// Represents fields and derived information received in RTCP report block
// attached to RTCP sender report or RTCP receiver report, as described in
// https://www.rfc-editor.org/rfc/rfc3550#section-6.4.1
class ReportBlockData {
public:
 ReportBlockData() = default;

 ReportBlockData(const ReportBlockData&) = default;
 ReportBlockData& operator=(const ReportBlockData&) = default;

 // The SSRC identifier for the originator of this report block,
 // i.e. remote receiver of the RTP stream.
 uint32_t sender_ssrc() const { return sender_ssrc_; }

 // The SSRC identifier of the source to which the information in this
 // reception report block pertains, i.e. local sender of the RTP stream.
 uint32_t source_ssrc() const { return source_ssrc_; }

 // The fraction of RTP data packets from 'source_ssrc()' lost since the
 // previous report block was sent.
 // Fraction loss in range [0.0, 1.0].
 float fraction_lost() const {
   return static_cast<float>(fraction_lost_raw()) / 256.0f;
 }

 // Fraction loss as was written in the raw packet: range is [0, 255] where 0
 // represents no loss, and 255 represents 99.6% loss (255/256 * 100%).
 uint8_t fraction_lost_raw() const { return fraction_lost_raw_; }

 // The total number of RTP data packets from 'source_ssrc()' that have been
 // lost since the beginning of reception.  This number is defined to be the
 // number of packets expected less the number of packets actually received,
 // where the number of packets received includes any which are late or
 // duplicates. Thus, packets that arrive late are not counted as lost, and the
 // loss may be negative if there are duplicates.
 int cumulative_lost() const { return cumulative_lost_; }

 // The low 16 bits contain the highest sequence number received in an RTP data
 // packet from 'source_ssrc()', and the most significant 16 bits extend that
 // sequence number with the corresponding count of sequence number cycles.
 uint32_t extended_highest_sequence_number() const {
   return extended_highest_sequence_number_;
 }

 // An estimate of the statistical variance of the RTP data packet interarrival
 // time, measured in RTP timestamp units. The interarrival jitter J is defined
 // to be the mean deviation (smoothed absolute value) of the difference D in
 // packet spacing at the receiver compared to the sender for a pair of
 // packets.
 uint32_t jitter() const { return jitter_; }

 // Jitter converted to common time units.
 TimeDelta jitter(int rtp_clock_rate_hz) const;

 // Time in utc epoch (Jan 1st, 1970) the report block was received.
 // TODO: bugs.webrtc.org/370535296 - Remove the utc timestamp when linked
 // issue is fixed.
 Timestamp report_block_timestamp_utc() const {
   return report_block_timestamp_utc_;
 }

 // Monotonic time when the report block was received.
 Timestamp report_block_timestamp() const { return report_block_timestamp_; }

 // Round Trip Time measurments for given (sender_ssrc, source_ssrc) pair.
 // Min, max, sum, number of measurements are since beginning of the call.
 TimeDelta last_rtt() const { return last_rtt_; }
 TimeDelta sum_rtts() const { return sum_rtt_; }
 size_t num_rtts() const { return num_rtts_; }
 bool has_rtt() const { return num_rtts_ != 0; }

 void set_sender_ssrc(uint32_t ssrc) { sender_ssrc_ = ssrc; }
 void set_source_ssrc(uint32_t ssrc) { source_ssrc_ = ssrc; }
 void set_fraction_lost_raw(uint8_t lost) { fraction_lost_raw_ = lost; }
 void set_cumulative_lost(int lost) { cumulative_lost_ = lost; }
 void set_extended_highest_sequence_number(uint32_t sn) {
   extended_highest_sequence_number_ = sn;
 }
 void set_jitter(uint32_t jitter) { jitter_ = jitter; }
 // TODO: bugs.webrtc.org/370535296 - Remove the utc timestamp when linked
 // issue is fixed.
 void set_report_block_timestamp_utc(Timestamp arrival_time) {
   report_block_timestamp_utc_ = arrival_time;
 }
 void set_report_block_timestamp(Timestamp arrival_time) {
   report_block_timestamp_ = arrival_time;
 }

 void SetReportBlock(uint32_t sender_ssrc,
                     const rtcp::ReportBlock& report_block,
                     Timestamp report_block_timestamp_utc,
                     Timestamp report_block_timestamp);
 void AddRoundTripTimeSample(TimeDelta rtt);

private:
 uint32_t sender_ssrc_ = 0;
 uint32_t source_ssrc_ = 0;
 uint8_t fraction_lost_raw_ = 0;
 int32_t cumulative_lost_ = 0;
 uint32_t extended_highest_sequence_number_ = 0;
 uint32_t jitter_ = 0;
 // TODO: bugs.webrtc.org/370535296 - Remove the utc timestamp when linked
 // issue is fixed.
 Timestamp report_block_timestamp_utc_ = Timestamp::Zero();
 Timestamp report_block_timestamp_ = Timestamp::Zero();
 TimeDelta last_rtt_ = TimeDelta::Zero();
 TimeDelta sum_rtt_ = TimeDelta::Zero();
 size_t num_rtts_ = 0;
};

class ReportBlockDataObserver {
public:
 virtual ~ReportBlockDataObserver() = default;

 virtual void OnReportBlockDataUpdated(ReportBlockData report_block_data) = 0;
};

}  // namespace webrtc

#endif  // MODULES_RTP_RTCP_INCLUDE_REPORT_BLOCK_DATA_H_