modules/rtp_rtcp/include/rtp_rtcp_defines.h

/*
 *  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(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;

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_