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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 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;
};
// For the deprecated MediaFile module.
enum FileFormats {
kFileFormatWavFile = 1,
kFileFormatPcm16kHzFile = 7,
kFileFormatPcm8kHzFile = 8,
kFileFormatPcm32kHzFile = 9,
};
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,
kVideoCodecStereo,
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_