android-14.0.0_r21/s

/*
 *  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 <memory>
#include <vector>

#include "absl/algorithm/container.h"
#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 "api/units/time_delta.h"
#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.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;
class RtpPacketToSend;
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;

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,
  kRtpExtensionCsrcAudioLevel,
  kRtpExtensionInbandComfortNoise,
  kRtpExtensionAbsoluteSendTime,
  kRtpExtensionAbsoluteCaptureTime,
  kRtpExtensionVideoRotation,
  kRtpExtensionTransportSequenceNumber,
  kRtpExtensionTransportSequenceNumber02,
  kRtpExtensionPlayoutDelay,
  kRtpExtensionVideoContentType,
  kRtpExtensionVideoLayersAllocation,
  kRtpExtensionVideoTiming,
  kRtpExtensionRtpStreamId,
  kRtpExtensionRepairedRtpStreamId,
  kRtpExtensionMid,
  kRtpExtensionGenericFrameDescriptor00,
  kRtpExtensionGenericFrameDescriptor = kRtpExtensionGenericFrameDescriptor00,
  kRtpExtensionGenericFrameDescriptor02,
  kRtpExtensionColorSpace,
  kRtpExtensionVideoFrameTrackingId,
  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,
  kRtcpLossNotification = 0x2000,
  kRtcpRemb = 0x10000,
  kRtcpTransmissionTimeOffset = 0x20000,
  kRtcpXrReceiverReferenceTime = 0x40000,
  kRtcpXrDlrrReportBlock = 0x80000,
  kRtcpTransportFeedback = 0x100000,
  kRtcpXrTargetBitrate = 0x200000
};

enum class KeyFrameReqMethod : uint8_t {
  kNone,     // Don't request keyframes.
  kPliRtcp,  // Request keyframes through Picture Loss Indication.
  kFirRtcp   // Request keyframes through Full Intra-frame Request.
};

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),
        ssrc_has_acked(false) {}
  uint16_t sequence_number;
  uint32_t start_timestamp;
  uint32_t timestamp;
  int64_t capture_time_ms;
  int64_t last_timestamp_time_ms;
  bool ssrc_has_acked;
};

// 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() {}
};

// NOTE! `kNumMediaTypes` must be kept in sync with RtpPacketMediaType!
static constexpr size_t kNumMediaTypes = 5;
enum class RtpPacketMediaType : size_t {
  kAudio,                         // Audio media packets.
  kVideo,                         // Video media packets.
  kRetransmission,                // Retransmisions, sent as response to NACK.
  kForwardErrorCorrection,        // FEC packets.
  kPadding = kNumMediaTypes - 1,  // RTX or plain padding sent to maintain BWE.
  // Again, don't forget to udate `kNumMediaTypes` if you add another value!
};

struct RtpPacketSendInfo {
  uint16_t transport_sequence_number = 0;
  absl::optional<uint32_t> media_ssrc;
  uint16_t rtp_sequence_number = 0;  // Only valid if `media_ssrc` is set.
  uint32_t rtp_timestamp = 0;
  size_t length = 0;
  absl::optional<RtpPacketMediaType> packet_type;
  PacedPacketInfo pacing_info;
};

class NetworkStateEstimateObserver {
 public:
  virtual void OnRemoteNetworkEstimate(NetworkStateEstimate estimate) = 0;
  virtual ~NetworkStateEstimateObserver() = default;
};

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 void SendCombinedRtcpPacket(
      std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) = 0;
  virtual void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) = 0;
  virtual void UnsetRemb() = 0;
};

class StreamFeedbackObserver {
 public:
  struct StreamPacketInfo {
    bool received;

    // `rtp_sequence_number` and `is_retransmission` are only valid if `ssrc`
    // is populated.
    absl::optional<uint32_t> ssrc;
    uint16_t rtp_sequence_number;
    bool is_retransmission;
  };
  virtual ~StreamFeedbackObserver() = default;

  virtual void OnPacketFeedbackVector(
      std::vector<StreamPacketInfo> packet_feedback_vector) = 0;
};

class StreamFeedbackProvider {
 public:
  virtual void RegisterStreamFeedbackObserver(
      std::vector<uint32_t> ssrcs,
      StreamFeedbackObserver* observer) = 0;
  virtual void DeRegisterStreamFeedbackObserver(
      StreamFeedbackObserver* observer) = 0;
  virtual ~StreamFeedbackProvider() = default;
};

class RtcpRttStats {
 public:
  virtual void OnRttUpdate(int64_t rtt) = 0;

  virtual int64_t LastProcessedRtt() const = 0;

  virtual ~RtcpRttStats() {}
};

struct RtpPacketCounter {
  RtpPacketCounter()
      : header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}

  explicit RtpPacketCounter(const RtpPacket& packet);
  explicit RtpPacketCounter(const RtpPacketToSend& packet_to_send);

  void Add(const RtpPacketCounter& other) {
    header_bytes += other.header_bytes;
    payload_bytes += other.payload_bytes;
    padding_bytes += other.padding_bytes;
    packets += other.packets;
    total_packet_delay += other.total_packet_delay;
  }

  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;
    RTC_DCHECK_GE(total_packet_delay, other.total_packet_delay);
    total_packet_delay -= other.total_packet_delay;
  }

  bool operator==(const RtpPacketCounter& other) const {
    return header_bytes == other.header_bytes &&
           payload_bytes == other.payload_bytes &&
           padding_bytes == other.padding_bytes && packets == other.packets &&
           total_packet_delay == other.total_packet_delay;
  }

  // Not inlined, since use of RtpPacket would result in circular includes.
  void AddPacket(const RtpPacket& packet);
  void AddPacket(const RtpPacketToSend& packet_to_send);

  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.
  // The total delay of all `packets`. For RtpPacketToSend packets, this is
  // `time_in_send_queue()`. For receive packets, this is zero.
  webrtc::TimeDelta total_packet_delay = webrtc::TimeDelta::Zero();
};

// 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.
};

class RtpSendRates {
  template <std::size_t... Is>
  constexpr std::array<DataRate, sizeof...(Is)> make_zero_array(
      std::index_sequence<Is...>) {
    return {{(static_cast<void>(Is), DataRate::Zero())...}};
  }

 public:
  RtpSendRates()
      : send_rates_(
            make_zero_array(std::make_index_sequence<kNumMediaTypes>())) {}
  RtpSendRates(const RtpSendRates& rhs) = default;
  RtpSendRates& operator=(const RtpSendRates&) = default;

  DataRate& operator[](RtpPacketMediaType type) {
    return send_rates_[static_cast<size_t>(type)];
  }
  const DataRate& operator[](RtpPacketMediaType type) const {
    return send_rates_[static_cast<size_t>(type)];
  }
  DataRate Sum() const {
    return absl::c_accumulate(send_rates_, DataRate::Zero());
  }

 private:
  std::array<DataRate, kNumMediaTypes> send_rates_;
};

// Callback, called whenever byte/packet counts have been updated.
class StreamDataCountersCallback {
 public:
  virtual ~StreamDataCountersCallback() {}

  virtual void DataCountersUpdated(const StreamDataCounters& counters,
                                   uint32_t ssrc) = 0;
};

// Information exposed through the GetStats api.
struct RtpReceiveStats {
  // `packets_lost` and `jitter` are defined by RFC 3550, and exposed in the
  // RTCReceivedRtpStreamStats dictionary, see
  // https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*
  int32_t packets_lost = 0;
  // Interarrival jitter in samples.
  uint32_t jitter = 0;
  // Interarrival jitter in time.
  webrtc::TimeDelta interarrival_jitter = webrtc::TimeDelta::Zero();

  // Timestamp and counters exposed in RTCInboundRtpStreamStats, see
  // https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
  absl::optional<int64_t> last_packet_received_timestamp_ms;
  RtpPacketCounter packet_counter;
};

// 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,
                                    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;
};

}  // namespace webrtc
#endif  // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_