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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.h')
-rw-r--r-- | third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.h | 123 |
1 files changed, 123 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.h b/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.h new file mode 100644 index 0000000000..9987fc46b9 --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/include/report_block_data.h @@ -0,0 +1,123 @@ +/* + * 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_ |