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// Copyright 2019 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#ifndef CAST_STREAMING_PACKET_RECEIVE_STATS_TRACKER_H_
#define CAST_STREAMING_PACKET_RECEIVE_STATS_TRACKER_H_
#include <stdint.h>
#include "cast/streaming/expanded_value_base.h"
#include "cast/streaming/rtcp_common.h"
#include "cast/streaming/rtp_time.h"
#include "platform/api/time.h"
namespace openscreen {
namespace cast {
// Maintains statistics for RTP packet arrival timing, jitter, and loss rates;
// and then uses these to compute and set the related fields in a RTCP Receiver
// Report block.
class PacketReceiveStatsTracker {
public:
explicit PacketReceiveStatsTracker(int rtp_timebase);
~PacketReceiveStatsTracker();
// This should be called each time a RTP packet is successfully parsed,
// whether the packet is a duplicate or not. The |sequence_number| and
// |rtp_timestamp| arguments should be the values from the
// RtpPacketParser::ParseResult. |arrival_time| is when the packet was
// received (i.e., right-off the network socket, before any
// processing/parsing).
void OnReceivedValidRtpPacket(uint16_t sequence_number,
RtpTimeTicks rtp_timestamp,
Clock::time_point arrival_time);
// Populates *only* those fields in the given |report| that pertain to packet
// loss, jitter, and the latest-known RTP packet sequence number.
void PopulateNextReport(RtcpReportBlock* report);
private:
// Expands the 16-bit raw packet sequence counter values into full-form,
// initially constructed from a "first" value.
class PacketSequenceNumber
: public ExpandedValueBase<int64_t, PacketSequenceNumber> {
public:
constexpr PacketSequenceNumber()
: ExpandedValueBase(std::numeric_limits<int64_t>::min()) {}
constexpr explicit PacketSequenceNumber(uint16_t first_raw_counter_value)
: ExpandedValueBase(static_cast<int64_t>(first_raw_counter_value)) {}
constexpr bool is_null() const { return *this == PacketSequenceNumber(); }
constexpr PacketSequenceNumber previous() const {
return PacketSequenceNumber(value_ - 1);
}
// Distance operator.
constexpr int64_t operator-(PacketSequenceNumber rhs) const {
return value_ - rhs.value_;
}
private:
friend class ExpandedValueBase<int64_t, PacketSequenceNumber>;
constexpr explicit PacketSequenceNumber(int64_t value)
: ExpandedValueBase(value) {}
};
const int rtp_timebase_; // RTP timestamp ticks per second.
// Until |num_rtp_packets_received_| is greater than zero, the rest of these
// fields contain invalid values.
int64_t num_rtp_packets_received_ = 0;
int64_t num_rtp_packets_received_at_last_report_;
// The greatest packet sequence number seen in any RTP packet.
PacketSequenceNumber greatest_sequence_number_;
// One before the packet sequence number contained in the very first RTP
// packet seen. This is "one before" to simplify the packet count
// calculations.
PacketSequenceNumber base_sequence_number_;
// The value of |greatest_sequence_number_| when the last call to
// PopulateNextReport() was made. This is used in the computation of the
// packet loss rate between reports.
PacketSequenceNumber greatest_sequence_number_at_last_report_;
// The time the last RTP packet was received. This is used in the computation
// that updates |jitter_|.
Clock::time_point last_rtp_packet_arrival_time_;
// The RTP timestamp of the last RTP packet received. This is used in the
// computation that updates |jitter_|.
RtpTimeTicks last_rtp_packet_timestamp_;
// The interarrival jitter. See RFC 3550 spec, section 6.4.1. The Cast
// Streaming spec diverges from the algorithm in the RFC spec in that it uses
// different pieces of timing data to calculate this metric.
Clock::duration jitter_;
};
} // namespace cast
} // namespace openscreen
#endif // CAST_STREAMING_PACKET_RECEIVE_STATS_TRACKER_H_
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