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/*
* Copyright (c) 2012 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_RTP_RTCP_DEFINES_H_
#define MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_
#include <stddef.h>
#include <list>
#include <vector>
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "absl/types/variant.h"
#include "api/array_view.h"
#include "api/audio_codecs/audio_format.h"
#include "api/rtp_headers.h"
#include "api/transport/network_types.h"
#include "modules/include/module_common_types.h"
#include "system_wrappers/include/clock.h"
#define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination
#define IP_PACKET_SIZE 1500 // we assume ethernet
namespace webrtc {
class RtpPacket;
namespace rtcp {
class TransportFeedback;
}
const int kVideoPayloadTypeFrequency = 90000;
// TODO(bugs.webrtc.org/6458): Remove this when all the depending projects are
// updated to correctly set rtp rate for RtcpSender.
const int kBogusRtpRateForAudioRtcp = 8000;
// Minimum RTP header size in bytes.
const uint8_t kRtpHeaderSize = 12;
enum StorageType { kDontRetransmit, kAllowRetransmission };
bool IsLegalMidName(absl::string_view name);
bool IsLegalRsidName(absl::string_view name);
// This enum must not have any gaps, i.e., all integers between
// kRtpExtensionNone and kRtpExtensionNumberOfExtensions must be valid enum
// entries.
enum RTPExtensionType : int {
kRtpExtensionNone,
kRtpExtensionTransmissionTimeOffset,
kRtpExtensionAudioLevel,
kRtpExtensionAbsoluteSendTime,
kRtpExtensionAbsoluteCaptureTime,
kRtpExtensionVideoRotation,
kRtpExtensionTransportSequenceNumber,
kRtpExtensionTransportSequenceNumber02,
kRtpExtensionPlayoutDelay,
kRtpExtensionVideoContentType,
kRtpExtensionVideoTiming,
kRtpExtensionFrameMarking,
kRtpExtensionRtpStreamId,
kRtpExtensionRepairedRtpStreamId,
kRtpExtensionMid,
kRtpExtensionGenericFrameDescriptor00,
kRtpExtensionGenericFrameDescriptor = kRtpExtensionGenericFrameDescriptor00,
kRtpExtensionGenericFrameDescriptor01,
kRtpExtensionGenericFrameDescriptor02,
kRtpExtensionColorSpace,
kRtpExtensionNumberOfExtensions // Must be the last entity in the enum.
};
enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 };
// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
enum RTCPPacketType : uint32_t {
kRtcpReport = 0x0001,
kRtcpSr = 0x0002,
kRtcpRr = 0x0004,
kRtcpSdes = 0x0008,
kRtcpBye = 0x0010,
kRtcpPli = 0x0020,
kRtcpNack = 0x0040,
kRtcpFir = 0x0080,
kRtcpTmmbr = 0x0100,
kRtcpTmmbn = 0x0200,
kRtcpSrReq = 0x0400,
kRtcpApp = 0x1000,
kRtcpLossNotification = 0x2000,
kRtcpRemb = 0x10000,
kRtcpTransmissionTimeOffset = 0x20000,
kRtcpXrReceiverReferenceTime = 0x40000,
kRtcpXrDlrrReportBlock = 0x80000,
kRtcpTransportFeedback = 0x100000,
kRtcpXrTargetBitrate = 0x200000
};
enum RtxMode {
kRtxOff = 0x0,
kRtxRetransmitted = 0x1, // Only send retransmissions over RTX.
kRtxRedundantPayloads = 0x2 // Preventively send redundant payloads
// instead of padding.
};
const size_t kRtxHeaderSize = 2;
struct RTCPReportBlock {
RTCPReportBlock()
: sender_ssrc(0),
source_ssrc(0),
fraction_lost(0),
packets_lost(0),
extended_highest_sequence_number(0),
jitter(0),
last_sender_report_timestamp(0),
delay_since_last_sender_report(0) {}
RTCPReportBlock(uint32_t sender_ssrc,
uint32_t source_ssrc,
uint8_t fraction_lost,
int32_t packets_lost,
uint32_t extended_highest_sequence_number,
uint32_t jitter,
uint32_t last_sender_report_timestamp,
uint32_t delay_since_last_sender_report)
: sender_ssrc(sender_ssrc),
source_ssrc(source_ssrc),
fraction_lost(fraction_lost),
packets_lost(packets_lost),
extended_highest_sequence_number(extended_highest_sequence_number),
jitter(jitter),
last_sender_report_timestamp(last_sender_report_timestamp),
delay_since_last_sender_report(delay_since_last_sender_report) {}
// Fields as described by RFC 3550 6.4.2.
uint32_t sender_ssrc; // SSRC of sender of this report.
uint32_t source_ssrc; // SSRC of the RTP packet sender.
uint8_t fraction_lost;
int32_t packets_lost; // 24 bits valid.
uint32_t extended_highest_sequence_number;
uint32_t jitter;
uint32_t last_sender_report_timestamp;
uint32_t delay_since_last_sender_report;
};
typedef std::list<RTCPReportBlock> ReportBlockList;
struct RtpState {
RtpState()
: sequence_number(0),
start_timestamp(0),
timestamp(0),
capture_time_ms(-1),
last_timestamp_time_ms(-1),
media_has_been_sent(false) {}
uint16_t sequence_number;
uint32_t start_timestamp;
uint32_t timestamp;
int64_t capture_time_ms;
int64_t last_timestamp_time_ms;
bool media_has_been_sent;
};
// Callback interface for packets recovered by FlexFEC or ULPFEC. In
// the FlexFEC case, the implementation should be able to demultiplex
// the recovered RTP packets based on SSRC.
class RecoveredPacketReceiver {
public:
virtual void OnRecoveredPacket(const uint8_t* packet, size_t length) = 0;
protected:
virtual ~RecoveredPacketReceiver() = default;
};
class RtcpIntraFrameObserver {
public:
virtual ~RtcpIntraFrameObserver() {}
virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
};
// Observer for incoming LossNotification RTCP messages.
// See the documentation of LossNotification for details.
class RtcpLossNotificationObserver {
public:
virtual ~RtcpLossNotificationObserver() = default;
virtual void OnReceivedLossNotification(uint32_t ssrc,
uint16_t seq_num_of_last_decodable,
uint16_t seq_num_of_last_received,
bool decodability_flag) = 0;
};
class RtcpBandwidthObserver {
public:
// REMB or TMMBR
virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;
virtual void OnReceivedRtcpReceiverReport(
const ReportBlockList& report_blocks,
int64_t rtt,
int64_t now_ms) = 0;
virtual ~RtcpBandwidthObserver() {}
};
struct PacketFeedback {
PacketFeedback(int64_t arrival_time_ms, uint16_t sequence_number);
PacketFeedback(int64_t arrival_time_ms,
int64_t send_time_ms,
uint16_t sequence_number,
size_t payload_size,
const PacedPacketInfo& pacing_info);
PacketFeedback(int64_t creation_time_ms,
uint16_t sequence_number,
size_t payload_size,
uint16_t local_net_id,
uint16_t remote_net_id,
const PacedPacketInfo& pacing_info);
PacketFeedback(int64_t creation_time_ms,
int64_t arrival_time_ms,
int64_t send_time_ms,
uint16_t sequence_number,
size_t payload_size,
uint16_t local_net_id,
uint16_t remote_net_id,
const PacedPacketInfo& pacing_info);
PacketFeedback(const PacketFeedback&);
PacketFeedback& operator=(const PacketFeedback&);
~PacketFeedback();
static constexpr int kNotAProbe = -1;
static constexpr int64_t kNotReceived = -1;
static constexpr int64_t kNoSendTime = -1;
// NOTE! The variable |creation_time_ms| is not used when testing equality.
// This is due to |creation_time_ms| only being used by SendTimeHistory
// for book-keeping, and is of no interest outside that class.
// TODO(philipel): Remove |creation_time_ms| from PacketFeedback when cleaning
// up SendTimeHistory.
bool operator==(const PacketFeedback& rhs) const;
// Time corresponding to when this object was created.
int64_t creation_time_ms;
// Time corresponding to when the packet was received. Timestamped with the
// receiver's clock. For unreceived packet, the sentinel value kNotReceived
// is used.
int64_t arrival_time_ms;
// Time corresponding to when the packet was sent, timestamped with the
// sender's clock.
int64_t send_time_ms;
// Packet identifier, incremented with 1 for every packet generated by the
// sender.
uint16_t sequence_number;
// Session unique packet identifier, incremented with 1 for every packet
// generated by the sender.
int64_t long_sequence_number;
// Size of the packet excluding RTP headers.
size_t payload_size;
// Size of preceeding packets that are not part of feedback.
size_t unacknowledged_data;
// The network route ids that this packet is associated with.
uint16_t local_net_id;
uint16_t remote_net_id;
// Pacing information about this packet.
PacedPacketInfo pacing_info;
// The SSRC and RTP sequence number of the packet this feedback refers to.
absl::optional<uint32_t> ssrc;
uint16_t rtp_sequence_number;
};
struct RtpPacketSendInfo {
public:
RtpPacketSendInfo() = default;
uint16_t transport_sequence_number = 0;
uint32_t ssrc = 0;
uint16_t rtp_sequence_number = 0;
// Get rid of this flag when all code paths populate |rtp_sequence_number|.
bool has_rtp_sequence_number = false;
size_t length = 0;
PacedPacketInfo pacing_info;
};
class TransportFeedbackObserver {
public:
TransportFeedbackObserver() {}
virtual ~TransportFeedbackObserver() {}
virtual void OnAddPacket(const RtpPacketSendInfo& packet_info) = 0;
virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
};
// Interface for PacketRouter to send rtcp feedback on behalf of
// congestion controller.
// TODO(bugs.webrtc.org/8239): Remove and use RtcpTransceiver directly
// when RtcpTransceiver always present in rtp transport.
class RtcpFeedbackSenderInterface {
public:
virtual ~RtcpFeedbackSenderInterface() = default;
virtual uint32_t SSRC() const = 0;
virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& feedback) = 0;
virtual void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) = 0;
virtual void UnsetRemb() = 0;
};
class PacketFeedbackObserver {
public:
virtual ~PacketFeedbackObserver() = default;
virtual void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) = 0;
virtual void OnPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector) = 0;
};
class RtcpRttStats {
public:
virtual void OnRttUpdate(int64_t rtt) = 0;
virtual int64_t LastProcessedRtt() const = 0;
virtual ~RtcpRttStats() {}
};
// This class will be deprecated and replaced with RtpPacketPacer.
class RtpPacketSender {
public:
RtpPacketSender() {}
virtual ~RtpPacketSender() {}
// These are part of the legacy PacedSender interface and will be removed.
enum Priority {
kHighPriority = 0, // Pass through; will be sent immediately.
kNormalPriority = 2, // Put in back of the line.
kLowPriority = 3, // Put in back of the low priority line.
};
// Adds the packet information to the queue and call TimeToSendPacket when
// it's time to send.
virtual void InsertPacket(Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) = 0;
// Currently audio traffic is not accounted by pacer and passed through.
// With the introduction of audio BWE audio traffic will be accounted for
// the pacer budget calculation. The audio traffic still will be injected
// at high priority.
// TODO(alexnarest): Make it pure virtual after rtp_sender_unittest will be
// updated to support it.
virtual void SetAccountForAudioPackets(bool account_for_audio) {}
};
class TransportSequenceNumberAllocator {
public:
TransportSequenceNumberAllocator() {}
virtual ~TransportSequenceNumberAllocator() {}
virtual uint16_t AllocateSequenceNumber() = 0;
};
struct RtpPacketCounter {
RtpPacketCounter()
: header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}
void Add(const RtpPacketCounter& other) {
header_bytes += other.header_bytes;
payload_bytes += other.payload_bytes;
padding_bytes += other.padding_bytes;
packets += other.packets;
}
void Subtract(const RtpPacketCounter& other) {
RTC_DCHECK_GE(header_bytes, other.header_bytes);
header_bytes -= other.header_bytes;
RTC_DCHECK_GE(payload_bytes, other.payload_bytes);
payload_bytes -= other.payload_bytes;
RTC_DCHECK_GE(padding_bytes, other.padding_bytes);
padding_bytes -= other.padding_bytes;
RTC_DCHECK_GE(packets, other.packets);
packets -= other.packets;
}
// Not inlined, since use of RtpPacket would result in circular includes.
void AddPacket(const RtpPacket& packet);
size_t TotalBytes() const {
return header_bytes + payload_bytes + padding_bytes;
}
size_t header_bytes; // Number of bytes used by RTP headers.
size_t payload_bytes; // Payload bytes, excluding RTP headers and padding.
size_t padding_bytes; // Number of padding bytes.
uint32_t packets; // Number of packets.
};
// Data usage statistics for a (rtp) stream.
struct StreamDataCounters {
StreamDataCounters();
void Add(const StreamDataCounters& other) {
transmitted.Add(other.transmitted);
retransmitted.Add(other.retransmitted);
fec.Add(other.fec);
if (other.first_packet_time_ms != -1 &&
(other.first_packet_time_ms < first_packet_time_ms ||
first_packet_time_ms == -1)) {
// Use oldest time.
first_packet_time_ms = other.first_packet_time_ms;
}
}
void Subtract(const StreamDataCounters& other) {
transmitted.Subtract(other.transmitted);
retransmitted.Subtract(other.retransmitted);
fec.Subtract(other.fec);
if (other.first_packet_time_ms != -1 &&
(other.first_packet_time_ms > first_packet_time_ms ||
first_packet_time_ms == -1)) {
// Use youngest time.
first_packet_time_ms = other.first_packet_time_ms;
}
}
int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
}
// Returns the number of bytes corresponding to the actual media payload (i.e.
// RTP headers, padding, retransmissions and fec packets are excluded).
// Note this function does not have meaning for an RTX stream.
size_t MediaPayloadBytes() const {
return transmitted.payload_bytes - retransmitted.payload_bytes -
fec.payload_bytes;
}
int64_t first_packet_time_ms; // Time when first packet is sent/received.
// The timestamp at which the last packet was received, i.e. the time of the
// local clock when it was received - not the RTP timestamp of that packet.
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-lastpacketreceivedtimestamp
absl::optional<int64_t> last_packet_received_timestamp_ms;
RtpPacketCounter transmitted; // Number of transmitted packets/bytes.
RtpPacketCounter retransmitted; // Number of retransmitted packets/bytes.
RtpPacketCounter fec; // Number of redundancy packets/bytes.
};
// Callback, called whenever byte/packet counts have been updated.
class StreamDataCountersCallback {
public:
virtual ~StreamDataCountersCallback() {}
virtual void DataCountersUpdated(const StreamDataCounters& counters,
uint32_t ssrc) = 0;
};
class RtcpAckObserver {
public:
// This method is called on received report blocks matching the sender ssrc.
// TODO(nisse): Use of "extended" sequence number is a bit brittle, since the
// observer for this callback typically has its own sequence number unwrapper,
// and there's no guarantee that they are in sync. Change to pass raw sequence
// number, possibly augmented with timestamp (if available) to aid
// disambiguation.
virtual void OnReceivedAck(int64_t extended_highest_sequence_number) = 0;
virtual ~RtcpAckObserver() = default;
};
// Callback, used to notify an observer whenever new rates have been estimated.
class BitrateStatisticsObserver {
public:
virtual ~BitrateStatisticsObserver() {}
virtual void Notify(uint32_t total_bitrate_bps,
uint32_t retransmit_bitrate_bps,
uint32_t ssrc) = 0;
};
// Callback, used to notify an observer whenever the send-side delay is updated.
class SendSideDelayObserver {
public:
virtual ~SendSideDelayObserver() {}
virtual void SendSideDelayUpdated(int avg_delay_ms,
int max_delay_ms,
uint64_t total_delay_ms,
uint32_t ssrc) = 0;
};
// Callback, used to notify an observer whenever a packet is sent to the
// transport.
// TODO(asapersson): This class will remove the need for SendSideDelayObserver.
// Remove SendSideDelayObserver once possible.
class SendPacketObserver {
public:
virtual ~SendPacketObserver() {}
virtual void OnSendPacket(uint16_t packet_id,
int64_t capture_time_ms,
uint32_t ssrc) = 0;
};
// Status returned from TimeToSendPacket() family of callbacks.
enum class RtpPacketSendResult {
kSuccess, // Packet sent OK.
kTransportUnavailable, // Network unavailable, try again later.
kPacketNotFound // SSRC/sequence number does not map to an available packet.
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_