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/*
* 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 "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 fraction_lost_raw() / 256.0; }
// 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.
Timestamp report_block_timestamp_utc() const {
return report_block_timestamp_utc_;
}
// 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; }
void set_report_block_timestamp_utc(Timestamp arrival_time) {
report_block_timestamp_utc_ = arrival_time;
}
void SetReportBlock(uint32_t sender_ssrc,
const rtcp::ReportBlock& report_block,
Timestamp report_block_timestamp_utc);
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;
Timestamp report_block_timestamp_utc_ = 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_
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