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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_RTCP_COMMON_H_
#define CAST_STREAMING_RTCP_COMMON_H_
#include <stdint.h>
#include <tuple>
#include <vector>
#include "absl/types/optional.h"
#include "absl/types/span.h"
#include "cast/streaming/frame_id.h"
#include "cast/streaming/ntp_time.h"
#include "cast/streaming/rtp_defines.h"
#include "cast/streaming/rtp_time.h"
#include "cast/streaming/ssrc.h"
namespace openscreen {
namespace cast {
struct RtcpCommonHeader {
RtcpCommonHeader();
~RtcpCommonHeader();
RtcpPacketType packet_type = RtcpPacketType::kNull;
union {
// The number of report blocks if |packet_type| is kSenderReport or
// kReceiverReport.
int report_count;
// Indicates the type of an application-defined message if |packet_type| is
// kApplicationDefined or kPayloadSpecific.
RtcpSubtype subtype;
// Otherwise, not used.
} with{0};
// The size (in bytes) of the RTCP packet, not including the header.
int payload_size = 0;
// Serializes this header into the first |kRtcpCommonHeaderSize| bytes of the
// given |buffer| and adjusts |buffer| to point to the first byte after it.
void AppendFields(absl::Span<uint8_t>* buffer) const;
// Parse from the 4-byte wire format in |buffer|. Returns nullopt if the data
// is corrupt.
static absl::optional<RtcpCommonHeader> Parse(
absl::Span<const uint8_t> buffer);
};
// The middle 32-bits of the 64-bit NtpTimestamp field from the Sender Reports.
// This is used as an opaque identifier that the Receiver will use in its
// reports to refer to specific previous Sender Reports.
using StatusReportId = uint32_t;
constexpr StatusReportId ToStatusReportId(NtpTimestamp ntp_timestamp) {
return static_cast<uint32_t>(ntp_timestamp >> 16);
}
// One of these is optionally included with a Sender Report or a Receiver
// Report. See: https://tools.ietf.org/html/rfc3550#section-6.4.1
struct RtcpReportBlock {
RtcpReportBlock();
~RtcpReportBlock();
// The intended recipient of this report block.
Ssrc ssrc = 0;
// The fraction of RTP packets lost since the last report, specified as a
// variable numerator and fixed denominator. The numerator will always be in
// the range [0,255] since, semantically:
//
// a. Negative values are impossible.
// b. Values greater than 255 would indicate 100% packet loss, and so a
// report block would not be generated in the first place.
int packet_fraction_lost_numerator = 0;
static constexpr int kPacketFractionLostDenominator = 256;
// The total number of RTP packets lost since the start of the session. This
// value will always be in the range [0,2^24-1], as the wire format only
// provides 24 bits; so, wrap-around is possible.
int cumulative_packets_lost = 0;
// The highest sequence number received in any RTP packet. Wrap-around is
// possible.
uint32_t extended_high_sequence_number = 0;
// An estimate of the recent variance in RTP packet arrival times.
RtpTimeDelta jitter;
// The last Status Report received.
StatusReportId last_status_report_id{};
// The delay between when the peer received the most-recent Status Report and
// when this report was sent. The timebase is 65536 ticks per second and,
// because of the wire format, this value will always be in the range
// [0,65536) seconds.
using Delay = std::chrono::duration<int64_t, std::ratio<1, 65536>>;
Delay delay_since_last_report{};
// Convenience helper to compute/assign the |packet_fraction_lost_numerator|,
// based on the |num_apparently_sent| and |num_received| packet counts since
// the last report was sent.
void SetPacketFractionLostNumerator(int64_t num_apparently_sent,
int64_t num_received);
// Convenience helper to compute/assign the |cumulative_packets_lost|, based
// on the |num_apparently_sent| and |num_received| packet counts since the
// start of the entire session.
void SetCumulativePacketsLost(int64_t num_apparently_sent,
int64_t num_received);
// Convenience helper to convert the given |local_clock_delay| to the
// RtcpReportBlock::Delay timebase, then clamp and assign it to
// |delay_since_last_report|.
void SetDelaySinceLastReport(Clock::duration local_clock_delay);
// Serializes this report block in the first |kRtcpReportBlockSize| bytes of
// the given |buffer| and adjusts |buffer| to point to the first byte after
// it.
void AppendFields(absl::Span<uint8_t>* buffer) const;
// Scans the wire-format report blocks in |buffer|, searching for one with the
// matching |ssrc| and, if found, returns the parse result. Returns nullopt if
// the data is corrupt or no report block with the matching SSRC was found.
static absl::optional<RtcpReportBlock>
ParseOne(absl::Span<const uint8_t> buffer, int report_count, Ssrc ssrc);
};
struct RtcpSenderReport {
RtcpSenderReport();
~RtcpSenderReport();
// The point-in-time at which this report was sent, according to both: 1) the
// common reference clock shared by all RTP streams; 2) the RTP timestamp on
// the media capture/playout timeline. Together, these are used by a Receiver
// to achieve A/V synchronization across RTP streams for playout.
Clock::time_point reference_time{};
RtpTimeTicks rtp_timestamp;
// The total number of RTP packets transmitted since the start of the session
// (wrap-around is possible).
uint32_t send_packet_count = 0;
// The total number of payload bytes transmitted in RTP packets since the
// start of the session (wrap-around is possible).
uint32_t send_octet_count = 0;
// The report block, if present. While the RTCP spec allows for zero or
// multiple reports, Cast Streaming only uses zero or one.
absl::optional<RtcpReportBlock> report_block;
};
// A pair of IDs that refers to a specific missing packet within a frame. If
// |packet_id| is kAllPacketsLost, then it represents all the packets of a
// frame.
struct PacketNack {
FrameId frame_id;
FramePacketId packet_id;
constexpr bool operator==(const PacketNack& other) const {
return frame_id == other.frame_id && packet_id == other.packet_id;
}
constexpr bool operator!=(const PacketNack& other) const {
return frame_id != other.frame_id || packet_id != other.packet_id;
}
constexpr bool operator<(const PacketNack& other) const {
return (frame_id < other.frame_id) ||
(frame_id == other.frame_id && packet_id < other.packet_id);
}
};
} // namespace cast
} // namespace openscreen
#endif // CAST_STREAMING_RTCP_COMMON_H_
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