common_types.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 COMMON_TYPES_H_
#define COMMON_TYPES_H_

#include <stddef.h>
#include <string.h>
#include <ostream>
#include <string>
#include <vector>

#include "api/array_view.h"
#include "api/optional.h"
#include "api/video/video_content_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_timing.h"
#include "rtc_base/checks.h"
#include "rtc_base/deprecation.h"
#include "typedefs.h"  // NOLINT(build/include)

#if defined(_MSC_VER)
// Disable "new behavior: elements of array will be default initialized"
// warning. Affects OverUseDetectorOptions.
#pragma warning(disable : 4351)
#endif

#if defined(WEBRTC_EXPORT)
#define WEBRTC_DLLEXPORT _declspec(dllexport)
#elif defined(WEBRTC_DLL)
#define WEBRTC_DLLEXPORT _declspec(dllimport)
#else
#define WEBRTC_DLLEXPORT
#endif

#ifndef NULL
#define NULL 0
#endif

#define RTP_PAYLOAD_NAME_SIZE 32u

#if defined(WEBRTC_WIN) || defined(WIN32)
// Compares two strings without regard to case.
#define STR_CASE_CMP(s1, s2) ::_stricmp(s1, s2)
// Compares characters of two strings without regard to case.
#define STR_NCASE_CMP(s1, s2, n) ::_strnicmp(s1, s2, n)
#else
#define STR_CASE_CMP(s1, s2) ::strcasecmp(s1, s2)
#define STR_NCASE_CMP(s1, s2, n) ::strncasecmp(s1, s2, n)
#endif

namespace webrtc {

class RewindableStream {
 public:
  virtual ~RewindableStream() {}
  virtual int Rewind() = 0;
};

class InStream : public RewindableStream {
 public:
  // Reads |len| bytes from file to |buf|. Returns the number of bytes read
  // or -1 on error.
  virtual int Read(void* buf, size_t len) = 0;
};

class OutStream : public RewindableStream {
 public:
  // Writes |len| bytes from |buf| to file. The actual writing may happen
  // some time later. Call Flush() to force a write.
  virtual bool Write(const void* buf, size_t len) = 0;
};

enum FileFormats {
  kFileFormatWavFile = 1,
  kFileFormatCompressedFile = 2,
  kFileFormatPreencodedFile = 4,
  kFileFormatPcm16kHzFile = 7,
  kFileFormatPcm8kHzFile = 8,
  kFileFormatPcm32kHzFile = 9,
  kFileFormatPcm48kHzFile = 10
};

enum FrameType {
  kEmptyFrame = 0,
  kAudioFrameSpeech = 1,
  kAudioFrameCN = 2,
  kVideoFrameKey = 3,
  kVideoFrameDelta = 4,
};

// Statistics for an RTCP channel
struct RtcpStatistics {
  RtcpStatistics()
      : fraction_lost(0),
        packets_lost(0),
        extended_highest_sequence_number(0),
        jitter(0) {}

  uint8_t fraction_lost;
  union {
    uint32_t packets_lost;
    RTC_DEPRECATED uint32_t cumulative_lost;
  };
  union {
    uint32_t extended_highest_sequence_number;
    RTC_DEPRECATED uint32_t extended_max_sequence_number;
  };
  uint32_t jitter;
};

class RtcpStatisticsCallback {
 public:
  virtual ~RtcpStatisticsCallback() {}

  virtual void StatisticsUpdated(const RtcpStatistics& statistics,
                                 uint32_t ssrc) = 0;
  virtual void CNameChanged(const char* cname, uint32_t ssrc) = 0;
};

// Statistics for RTCP packet types.
struct RtcpPacketTypeCounter {
  RtcpPacketTypeCounter()
      : first_packet_time_ms(-1),
        nack_packets(0),
        fir_packets(0),
        pli_packets(0),
        nack_requests(0),
        unique_nack_requests(0) {}

  void Add(const RtcpPacketTypeCounter& other) {
    nack_packets += other.nack_packets;
    fir_packets += other.fir_packets;
    pli_packets += other.pli_packets;
    nack_requests += other.nack_requests;
    unique_nack_requests += other.unique_nack_requests;
    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 RtcpPacketTypeCounter& other) {
    nack_packets -= other.nack_packets;
    fir_packets -= other.fir_packets;
    pli_packets -= other.pli_packets;
    nack_requests -= other.nack_requests;
    unique_nack_requests -= other.unique_nack_requests;
    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);
  }

  int UniqueNackRequestsInPercent() const {
    if (nack_requests == 0) {
      return 0;
    }
    return static_cast<int>((unique_nack_requests * 100.0f / nack_requests) +
                            0.5f);
  }

  int64_t first_packet_time_ms;   // Time when first packet is sent/received.
  uint32_t nack_packets;          // Number of RTCP NACK packets.
  uint32_t fir_packets;           // Number of RTCP FIR packets.
  uint32_t pli_packets;           // Number of RTCP PLI packets.
  uint32_t nack_requests;         // Number of NACKed RTP packets.
  uint32_t unique_nack_requests;  // Number of unique NACKed RTP packets.
};

class RtcpPacketTypeCounterObserver {
 public:
  virtual ~RtcpPacketTypeCounterObserver() {}
  virtual void RtcpPacketTypesCounterUpdated(
      uint32_t ssrc,
      const RtcpPacketTypeCounter& packet_counter) = 0;
};

// Rate statistics for a stream.
struct BitrateStatistics {
  BitrateStatistics() : bitrate_bps(0), packet_rate(0) {}

  uint32_t bitrate_bps;  // Bitrate in bits per second.
  uint32_t packet_rate;  // Packet rate in packets per second.
};

// 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;
};

struct FrameCounts {
  FrameCounts() : key_frames(0), delta_frames(0) {}
  int key_frames;
  int delta_frames;
};

// Callback, used to notify an observer whenever frame counts have been updated.
class FrameCountObserver {
 public:
  virtual ~FrameCountObserver() {}
  virtual void FrameCountUpdated(const FrameCounts& frame_counts,
                                 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;
};

// Callback, used to notify an observer when the overhead per packet
// has changed.
class OverheadObserver {
 public:
  virtual ~OverheadObserver() = default;
  virtual void OnOverheadChanged(size_t overhead_bytes_per_packet) = 0;
};

// ==================================================================
// Voice specific types
// ==================================================================

// Each codec supported can be described by this structure.
struct CodecInst {
  int pltype;
  char plname[RTP_PAYLOAD_NAME_SIZE];
  int plfreq;
  int pacsize;
  size_t channels;
  int rate;  // bits/sec unlike {start,min,max}Bitrate elsewhere in this file!

  bool operator==(const CodecInst& other) const {
    return pltype == other.pltype &&
           (STR_CASE_CMP(plname, other.plname) == 0) &&
           plfreq == other.plfreq && pacsize == other.pacsize &&
           channels == other.channels && rate == other.rate;
  }

  bool operator!=(const CodecInst& other) const { return !(*this == other); }

  friend std::ostream& operator<<(std::ostream& os, const CodecInst& ci) {
    os << "{pltype: " << ci.pltype;
    os << ", plname: " << ci.plname;
    os << ", plfreq: " << ci.plfreq;
    os << ", pacsize: " << ci.pacsize;
    os << ", channels: " << ci.channels;
    os << ", rate: " << ci.rate << "}";
    return os;
  }
};

// RTP
enum { kRtpCsrcSize = 15 };  // RFC 3550 page 13

enum PayloadFrequencies {
  kFreq8000Hz = 8000,
  kFreq16000Hz = 16000,
  kFreq32000Hz = 32000
};

// Degree of bandwidth reduction.
enum VadModes {
  kVadConventional = 0,  // lowest reduction
  kVadAggressiveLow,
  kVadAggressiveMid,
  kVadAggressiveHigh  // highest reduction
};

// NETEQ statistics.
struct NetworkStatistics {
  // current jitter buffer size in ms
  uint16_t currentBufferSize;
  // preferred (optimal) buffer size in ms
  uint16_t preferredBufferSize;
  // adding extra delay due to "peaky jitter"
  bool jitterPeaksFound;
  // Stats below correspond to similarly-named fields in the WebRTC stats spec.
  // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
  uint64_t totalSamplesReceived;
  uint64_t concealedSamples;
  uint64_t concealmentEvents;
  uint64_t jitterBufferDelayMs;
  // Stats below DO NOT correspond directly to anything in the WebRTC stats
  // Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
  uint16_t currentPacketLossRate;
  // Late loss rate; fraction between 0 and 1, scaled to Q14.
  union {
    RTC_DEPRECATED uint16_t currentDiscardRate;
  };
  // fraction (of original stream) of synthesized audio inserted through
  // expansion (in Q14)
  uint16_t currentExpandRate;
  // fraction (of original stream) of synthesized speech inserted through
  // expansion (in Q14)
  uint16_t currentSpeechExpandRate;
  // fraction of synthesized speech inserted through pre-emptive expansion
  // (in Q14)
  uint16_t currentPreemptiveRate;
  // fraction of data removed through acceleration (in Q14)
  uint16_t currentAccelerateRate;
  // fraction of data coming from secondary decoding (in Q14)
  uint16_t currentSecondaryDecodedRate;
  // Fraction of secondary data, including FEC and RED, that is discarded (in
  // Q14). Discarding of secondary data can be caused by the reception of the
  // primary data, obsoleting the secondary data. It can also be caused by early
  // or late arrival of secondary data.
  uint16_t currentSecondaryDiscardedRate;
  // clock-drift in parts-per-million (negative or positive)
  int32_t clockDriftPPM;
  // average packet waiting time in the jitter buffer (ms)
  int meanWaitingTimeMs;
  // median packet waiting time in the jitter buffer (ms)
  int medianWaitingTimeMs;
  // min packet waiting time in the jitter buffer (ms)
  int minWaitingTimeMs;
  // max packet waiting time in the jitter buffer (ms)
  int maxWaitingTimeMs;
  // added samples in off mode due to packet loss
  size_t addedSamples;
};

// Statistics for calls to AudioCodingModule::PlayoutData10Ms().
struct AudioDecodingCallStats {
  AudioDecodingCallStats()
      : calls_to_silence_generator(0),
        calls_to_neteq(0),
        decoded_normal(0),
        decoded_plc(0),
        decoded_cng(0),
        decoded_plc_cng(0),
        decoded_muted_output(0) {}

  int calls_to_silence_generator;  // Number of calls where silence generated,
                                   // and NetEq was disengaged from decoding.
  int calls_to_neteq;              // Number of calls to NetEq.
  int decoded_normal;  // Number of calls where audio RTP packet decoded.
  int decoded_plc;     // Number of calls resulted in PLC.
  int decoded_cng;  // Number of calls where comfort noise generated due to DTX.
  int decoded_plc_cng;  // Number of calls resulted where PLC faded to CNG.
  int decoded_muted_output;  // Number of calls returning a muted state output.
};

// ==================================================================
// Video specific types
// ==================================================================

// TODO(nisse): Delete, and switch to fourcc values everywhere?
// Supported video types.
enum class VideoType {
  kUnknown,
  kI420,
  kIYUV,
  kRGB24,
  kABGR,
  kARGB,
  kARGB4444,
  kRGB565,
  kARGB1555,
  kYUY2,
  kYV12,
  kUYVY,
  kMJPEG,
  kNV21,
  kNV12,
  kBGRA,
};

// Video codec
enum { kPayloadNameSize = 32 };
enum { kMaxSimulcastStreams = 4 };
enum { kMaxSpatialLayers = 5 };
enum { kMaxTemporalStreams = 4 };

enum VideoCodecComplexity {
  kComplexityNormal = 0,
  kComplexityHigh = 1,
  kComplexityHigher = 2,
  kComplexityMax = 3
};

enum VP8ResilienceMode {
  kResilienceOff,    // The stream produced by the encoder requires a
                     // recovery frame (typically a key frame) to be
                     // decodable after a packet loss.
  kResilientStream,  // A stream produced by the encoder is resilient to
                     // packet losses, but packets within a frame subsequent
                     // to a loss can't be decoded.
  kResilientFrames   // Same as kResilientStream but with added resilience
                     // within a frame.
};

class TemporalLayersFactory;
// VP8 specific
struct VideoCodecVP8 {
  // TODO(nisse): Unused, delete?
  bool pictureLossIndicationOn;
  VideoCodecComplexity complexity;
  VP8ResilienceMode resilience;
  unsigned char numberOfTemporalLayers;
  bool denoisingOn;
  bool errorConcealmentOn;
  bool automaticResizeOn;
  bool frameDroppingOn;
  int keyFrameInterval;
  TemporalLayersFactory* tl_factory;
};

// VP9 specific.
struct VideoCodecVP9 {
  VideoCodecComplexity complexity;
  bool resilienceOn;
  unsigned char numberOfTemporalLayers;
  bool denoisingOn;
  bool frameDroppingOn;
  int keyFrameInterval;
  bool adaptiveQpMode;
  bool automaticResizeOn;
  unsigned char numberOfSpatialLayers;
  bool flexibleMode;
};

// TODO(magjed): Move this and other H264 related classes out to their own file.
namespace H264 {

enum Profile {
  kProfileConstrainedBaseline,
  kProfileBaseline,
  kProfileMain,
  kProfileConstrainedHigh,
  kProfileHigh,
};

}  // namespace H264

// H264 specific.
struct VideoCodecH264 {
  bool frameDroppingOn;
  int keyFrameInterval;
  // These are NULL/0 if not externally negotiated.
  const uint8_t* spsData;
  size_t spsLen;
  const uint8_t* ppsData;
  size_t ppsLen;
  H264::Profile profile;
};

// Video codec types
enum VideoCodecType {
  kVideoCodecVP8,
  kVideoCodecVP9,
  kVideoCodecH264,
  kVideoCodecI420,
  kVideoCodecRED,
  kVideoCodecULPFEC,
  kVideoCodecFlexfec,
  kVideoCodecGeneric,
  kVideoCodecUnknown
};

// Translates from name of codec to codec type and vice versa.
const char* CodecTypeToPayloadString(VideoCodecType type);
VideoCodecType PayloadStringToCodecType(const std::string& name);

union VideoCodecUnion {
  VideoCodecVP8 VP8;
  VideoCodecVP9 VP9;
  VideoCodecH264 H264;
};

// Simulcast is when the same stream is encoded multiple times with different
// settings such as resolution.
struct SimulcastStream {
  unsigned short width;
  unsigned short height;
  unsigned char numberOfTemporalLayers;
  unsigned int maxBitrate;     // kilobits/sec.
  unsigned int targetBitrate;  // kilobits/sec.
  unsigned int minBitrate;     // kilobits/sec.
  unsigned int qpMax;          // minimum quality
};

struct SpatialLayer {
  int scaling_factor_num;
  int scaling_factor_den;
  int target_bitrate_bps;
  // TODO(ivica): Add max_quantizer and min_quantizer?
};

enum VideoCodecMode { kRealtimeVideo, kScreensharing };

// Common video codec properties
class VideoCodec {
 public:
  VideoCodec();

  // Public variables. TODO(hta): Make them private with accessors.
  VideoCodecType codecType;
  char plName[kPayloadNameSize];
  unsigned char plType;

  unsigned short width;
  unsigned short height;

  unsigned int startBitrate;   // kilobits/sec.
  unsigned int maxBitrate;     // kilobits/sec.
  unsigned int minBitrate;     // kilobits/sec.
  unsigned int targetBitrate;  // kilobits/sec.

  uint32_t maxFramerate;

  unsigned int qpMax;
  unsigned char numberOfSimulcastStreams;
  SimulcastStream simulcastStream[kMaxSimulcastStreams];
  SpatialLayer spatialLayers[kMaxSpatialLayers];

  VideoCodecMode mode;
  bool expect_encode_from_texture;

  // Timing frames configuration. There is delay of delay_ms between two
  // consequent timing frames, excluding outliers. Frame is always made a
  // timing frame if it's at least outlier_ratio in percent of "ideal" average
  // frame given bitrate and framerate, i.e. if it's bigger than
  // |outlier_ratio / 100.0 * bitrate_bps / fps| in bits. This way, timing
  // frames will not be sent too often usually. Yet large frames will always
  // have timing information for debug purposes because they are more likely to
  // cause extra delays.
  struct TimingFrameTriggerThresholds {
    int64_t delay_ms;
    uint16_t outlier_ratio_percent;
  } timing_frame_thresholds;

  bool operator==(const VideoCodec& other) const = delete;
  bool operator!=(const VideoCodec& other) const = delete;

  // Accessors for codec specific information.
  // There is a const version of each that returns a reference,
  // and a non-const version that returns a pointer, in order
  // to allow modification of the parameters.
  VideoCodecVP8* VP8();
  const VideoCodecVP8& VP8() const;
  VideoCodecVP9* VP9();
  const VideoCodecVP9& VP9() const;
  VideoCodecH264* H264();
  const VideoCodecH264& H264() const;

 private:
  // TODO(hta): Consider replacing the union with a pointer type.
  // This will allow removing the VideoCodec* types from this file.
  VideoCodecUnion codec_specific_;
};

class BitrateAllocation {
 public:
  static const uint32_t kMaxBitrateBps;
  BitrateAllocation();

  bool SetBitrate(size_t spatial_index,
                  size_t temporal_index,
                  uint32_t bitrate_bps);

  bool HasBitrate(size_t spatial_index, size_t temporal_index) const;

  uint32_t GetBitrate(size_t spatial_index, size_t temporal_index) const;

  // Whether the specific spatial layers has the bitrate set in any of its
  // temporal layers.
  bool IsSpatialLayerUsed(size_t spatial_index) const;

  // Get the sum of all the temporal layer for a specific spatial layer.
  uint32_t GetSpatialLayerSum(size_t spatial_index) const;

  uint32_t get_sum_bps() const { return sum_; }  // Sum of all bitrates.
  uint32_t get_sum_kbps() const { return (sum_ + 500) / 1000; }

  inline bool operator==(const BitrateAllocation& other) const {
    return memcmp(bitrates_, other.bitrates_, sizeof(bitrates_)) == 0;
  }
  inline bool operator!=(const BitrateAllocation& other) const {
    return !(*this == other);
  }

  // Expensive, please use only in tests.
  std::string ToString() const;
  std::ostream& operator<<(std::ostream& os) const;

 private:
  uint32_t sum_;
  uint32_t bitrates_[kMaxSpatialLayers][kMaxTemporalStreams];
  bool has_bitrate_[kMaxSpatialLayers][kMaxTemporalStreams];
};

// Bandwidth over-use detector options.  These are used to drive
// experimentation with bandwidth estimation parameters.
// See modules/remote_bitrate_estimator/overuse_detector.h
// TODO(terelius): This is only used in overuse_estimator.cc, and only in the
// default constructed state. Can we move the relevant variables into that
// class and delete this? See also disabled warning at line 27
struct OverUseDetectorOptions {
  OverUseDetectorOptions()
      : initial_slope(8.0 / 512.0),
        initial_offset(0),
        initial_e(),
        initial_process_noise(),
        initial_avg_noise(0.0),
        initial_var_noise(50) {
    initial_e[0][0] = 100;
    initial_e[1][1] = 1e-1;
    initial_e[0][1] = initial_e[1][0] = 0;
    initial_process_noise[0] = 1e-13;
    initial_process_noise[1] = 1e-3;
  }
  double initial_slope;
  double initial_offset;
  double initial_e[2][2];
  double initial_process_noise[2];
  double initial_avg_noise;
  double initial_var_noise;
};

// This structure will have the information about when packet is actually
// received by socket.
struct PacketTime {
  PacketTime() : timestamp(-1), not_before(-1) {}
  PacketTime(int64_t timestamp, int64_t not_before)
      : timestamp(timestamp), not_before(not_before) {}

  int64_t timestamp;   // Receive time after socket delivers the data.
  int64_t not_before;  // Earliest possible time the data could have arrived,
                       // indicating the potential error in the |timestamp|
                       // value,in case the system is busy.
                       // For example, the time of the last select() call.
                       // If unknown, this value will be set to zero.
};

// Minimum and maximum playout delay values from capture to render.
// These are best effort values.
//
// A value < 0 indicates no change from previous valid value.
//
// min = max = 0 indicates that the receiver should try and render
// frame as soon as possible.
//
// min = x, max = y indicates that the receiver is free to adapt
// in the range (x, y) based on network jitter.
//
// Note: Given that this gets embedded in a union, it is up-to the owner to
// initialize these values.
struct PlayoutDelay {
  int min_ms;
  int max_ms;
};

// Class to represent the value of RTP header extensions that are
// variable-length strings (e.g., RtpStreamId and RtpMid).
// Unlike std::string, it can be copied with memcpy and cleared with memset.
//
// Empty value represents unset header extension (use empty() to query).
class StringRtpHeaderExtension {
 public:
  // String RTP header extensions are limited to 16 bytes because it is the
  // maximum length that can be encoded with one-byte header extensions.
  static constexpr size_t kMaxSize = 16;

  static bool IsLegalName(rtc::ArrayView<const char> name);

  StringRtpHeaderExtension() { value_[0] = 0; }
  explicit StringRtpHeaderExtension(rtc::ArrayView<const char> value) {
    Set(value.data(), value.size());
  }
  StringRtpHeaderExtension(const StringRtpHeaderExtension&) = default;
  StringRtpHeaderExtension& operator=(const StringRtpHeaderExtension&) =
      default;

  bool empty() const { return value_[0] == 0; }
  const char* data() const { return value_; }
  size_t size() const { return strnlen(value_, kMaxSize); }

  void Set(rtc::ArrayView<const uint8_t> value) {
    Set(reinterpret_cast<const char*>(value.data()), value.size());
  }
  void Set(const char* data, size_t size);

  friend bool operator==(const StringRtpHeaderExtension& lhs,
                         const StringRtpHeaderExtension& rhs) {
    return strncmp(lhs.value_, rhs.value_, kMaxSize) == 0;
  }
  friend bool operator!=(const StringRtpHeaderExtension& lhs,
                         const StringRtpHeaderExtension& rhs) {
    return !(lhs == rhs);
  }

 private:
  char value_[kMaxSize];
};

// StreamId represents RtpStreamId which is a string.
typedef StringRtpHeaderExtension StreamId;

// Mid represents RtpMid which is a string.
typedef StringRtpHeaderExtension Mid;

struct RTPHeaderExtension {
  RTPHeaderExtension();
  RTPHeaderExtension(const RTPHeaderExtension& other);
  RTPHeaderExtension& operator=(const RTPHeaderExtension& other);

  bool hasTransmissionTimeOffset;
  int32_t transmissionTimeOffset;
  bool hasAbsoluteSendTime;
  uint32_t absoluteSendTime;
  bool hasTransportSequenceNumber;
  uint16_t transportSequenceNumber;

  // Audio Level includes both level in dBov and voiced/unvoiced bit. See:
  // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
  bool hasAudioLevel;
  bool voiceActivity;
  uint8_t audioLevel;

  // For Coordination of Video Orientation. See
  // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
  // ts_126114v120700p.pdf
  bool hasVideoRotation;
  VideoRotation videoRotation;

  // TODO(ilnik): Refactor this and one above to be rtc::Optional() and remove
  // a corresponding bool flag.
  bool hasVideoContentType;
  VideoContentType videoContentType;

  bool has_video_timing;
  VideoSendTiming video_timing;

  PlayoutDelay playout_delay = {-1, -1};

  // For identification of a stream when ssrc is not signaled. See
  // https://tools.ietf.org/html/draft-ietf-avtext-rid-09
  // TODO(danilchap): Update url from draft to release version.
  StreamId stream_id;
  StreamId repaired_stream_id;

  // For identifying the media section used to interpret this RTP packet. See
  // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-38
  Mid mid;
};

struct RTPHeader {
  RTPHeader();
  RTPHeader(const RTPHeader& other);
  RTPHeader& operator=(const RTPHeader& other);

  bool markerBit;
  uint8_t payloadType;
  uint16_t sequenceNumber;
  uint32_t timestamp;
  uint32_t ssrc;
  uint8_t numCSRCs;
  uint32_t arrOfCSRCs[kRtpCsrcSize];
  size_t paddingLength;
  size_t headerLength;
  int payload_type_frequency;
  RTPHeaderExtension extension;
};

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;
};

// RTCP mode to use. Compound mode is described by RFC 4585 and reduced-size
// RTCP mode is described by RFC 5506.
enum class RtcpMode { kOff, kCompound, kReducedSize };

enum NetworkState {
  kNetworkUp,
  kNetworkDown,
};

struct RtpKeepAliveConfig final {
  // If no packet has been sent for |timeout_interval_ms|, send a keep-alive
  // packet. The keep-alive packet is an empty (no payload) RTP packet with a
  // payload type of 20 as long as the other end has not negotiated the use of
  // this value. If this value has already been negotiated, then some other
  // unused static payload type from table 5 of RFC 3551 shall be used and set
  // in |payload_type|.
  int64_t timeout_interval_ms = -1;
  uint8_t payload_type = 20;

  bool operator==(const RtpKeepAliveConfig& o) const {
    return timeout_interval_ms == o.timeout_interval_ms &&
           payload_type == o.payload_type;
  }
  bool operator!=(const RtpKeepAliveConfig& o) const { return !(*this == o); }
};

// Currently only VP8/VP9 specific.
struct RtpPayloadState {
  int16_t picture_id = -1;
};

}  // namespace webrtc

#endif  // COMMON_TYPES_H_