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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/types/variant.h"
#include "api/audio_codecs/audio_format.h"
#include "api/rtp_headers.h"
#include "api/transport/network_types.h"
#include "common_types.h" // NOLINT(build/include)
#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 {
namespace rtcp {
class TransportFeedback;
}
const int kVideoPayloadTypeFrequency = 90000;
// TODO(solenberg): RTP time stamp rate for RTCP is fixed at 8k, this is legacy
// and should be fixed.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6458
const int kBogusRtpRateForAudioRtcp = 8000;
// Minimum RTP header size in bytes.
const uint8_t kRtpHeaderSize = 12;
struct RtcpIntervalConfig final {
RtcpIntervalConfig() = default;
RtcpIntervalConfig(int64_t video_interval_ms, int64_t audio_interval_ms)
: video_interval_ms(video_interval_ms),
audio_interval_ms(audio_interval_ms) {}
int64_t video_interval_ms = 1000;
int64_t audio_interval_ms = 5000;
};
struct AudioPayload {
SdpAudioFormat format;
uint32_t rate;
};
struct VideoPayload {
VideoCodecType videoCodecType;
// The H264 profile only matters if videoCodecType == kVideoCodecH264.
H264::Profile h264_profile;
};
class PayloadUnion {
public:
explicit PayloadUnion(const AudioPayload& payload);
explicit PayloadUnion(const VideoPayload& payload);
PayloadUnion(const PayloadUnion&);
PayloadUnion(PayloadUnion&&);
~PayloadUnion();
PayloadUnion& operator=(const PayloadUnion&);
PayloadUnion& operator=(PayloadUnion&&);
bool is_audio() const {
return absl::holds_alternative<AudioPayload>(payload_);
}
bool is_video() const {
return absl::holds_alternative<VideoPayload>(payload_);
}
const AudioPayload& audio_payload() const {
return absl::get<AudioPayload>(payload_);
}
const VideoPayload& video_payload() const {
return absl::get<VideoPayload>(payload_);
}
AudioPayload& audio_payload() { return absl::get<AudioPayload>(payload_); }
VideoPayload& video_payload() { return absl::get<VideoPayload>(payload_); }
private:
absl::variant<AudioPayload, VideoPayload> payload_;
};
enum ProtectionType { kUnprotectedPacket, kProtectedPacket };
enum StorageType { kDontRetransmit, kAllowRetransmission };
// 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,
kRtpExtensionVideoRotation,
kRtpExtensionTransportSequenceNumber,
kRtpExtensionPlayoutDelay,
kRtpExtensionVideoContentType,
kRtpExtensionVideoTiming,
kRtpExtensionFrameMarking,
kRtpExtensionRtpStreamId,
kRtpExtensionRepairedRtpStreamId,
kRtpExtensionMid,
kRtpExtensionGenericFrameDescriptor,
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,
kRtcpRemb = 0x10000,
kRtcpTransmissionTimeOffset = 0x20000,
kRtcpXrReceiverReferenceTime = 0x40000,
kRtcpXrDlrrReportBlock = 0x80000,
kRtcpTransportFeedback = 0x100000,
kRtcpXrTargetBitrate = 0x200000
};
enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp };
enum RtpRtcpPacketType { kPacketRtp = 0, kPacketKeepAlive = 1 };
// kConditionallyRetransmitHigherLayers allows retransmission of video frames
// in higher layers if either the last frame in that layer was too far back in
// time, or if we estimate that a new frame will be available in a lower layer
// in a shorter time than it would take to request and receive a retransmission.
enum RetransmissionMode : uint8_t {
kRetransmitOff = 0x0,
kRetransmitFECPackets = 0x1,
kRetransmitBaseLayer = 0x2,
kRetransmitHigherLayers = 0x4,
kConditionallyRetransmitHigherLayers = 0x8,
kRetransmitAllPackets = 0xFF
};
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;
};
class RtpData {
public:
virtual ~RtpData() {}
virtual int32_t OnReceivedPayloadData(const uint8_t* payload_data,
size_t payload_size,
const WebRtcRTPHeader* rtp_header) = 0;
};
// 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;
};
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;
};
class PacketFeedbackComparator {
public:
inline bool operator()(const PacketFeedback& lhs, const PacketFeedback& rhs) {
if (lhs.arrival_time_ms != rhs.arrival_time_ms)
return lhs.arrival_time_ms < rhs.arrival_time_ms;
if (lhs.send_time_ms != rhs.send_time_ms)
return lhs.send_time_ms < rhs.send_time_ms;
return lhs.sequence_number < rhs.sequence_number;
}
};
class TransportFeedbackObserver {
public:
TransportFeedbackObserver() {}
virtual ~TransportFeedbackObserver() {}
// Note: Transport-wide sequence number as sequence number.
virtual void AddPacket(uint32_t ssrc,
uint16_t sequence_number,
size_t length,
const PacedPacketInfo& pacing_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() {}
};
// Statistics about packet loss for a single directional connection. All values
// are totals since the connection initiated.
struct RtpPacketLossStats {
// The number of packets lost in events where no adjacent packets were also
// lost.
uint64_t single_packet_loss_count;
// The number of events in which more than one adjacent packet was lost.
uint64_t multiple_packet_loss_event_count;
// The number of packets lost in events where more than one adjacent packet
// was lost.
uint64_t multiple_packet_loss_packet_count;
};
class RtpPacketSender {
public:
RtpPacketSender() {}
virtual ~RtpPacketSender() {}
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.
};
// Low priority packets are mixed with the normal priority packets
// while we are paused.
// Returns true if we send the packet now, else it will add 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;
}
void AddPacket(size_t packet_length, const RTPHeader& header) {
++packets;
header_bytes += header.headerLength;
padding_bytes += header.paddingLength;
payload_bytes +=
packet_length - (header.headerLength + header.paddingLength);
}
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.
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;
};
} // namespace webrtc
#endif // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_