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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 WEBRTC_COMMON_TYPES_H_
-#define WEBRTC_COMMON_TYPES_H_
-
-#include <stddef.h>
-#include <string.h>
-
-#include <string>
-#include <vector>
-
-#include "webrtc/typedefs.h"
-
-#if defined(_MSC_VER)
-// Disable "new behavior: elements of array will be default initialized"
-// warning. Affects OverUseDetectorOptions.
-#pragma warning(disable:4351)
-#endif
-
-#ifdef WEBRTC_EXPORT
-#define WEBRTC_DLLEXPORT _declspec(dllexport)
-#elif WEBRTC_DLL
-#define WEBRTC_DLLEXPORT _declspec(dllimport)
-#else
-#define WEBRTC_DLLEXPORT
-#endif
-
-#ifndef NULL
-#define NULL 0
-#endif
-
-#define RTP_PAYLOAD_NAME_SIZE 32
-
-#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 Config;
-
-class InStream
-{
-public:
- // Reads |length| bytes from file to |buf|. Returns the number of bytes read
- // or -1 on error.
- virtual int Read(void *buf, size_t len) = 0;
- virtual int Rewind();
- virtual ~InStream() {}
-protected:
- InStream() {}
-};
-
-class OutStream
-{
-public:
- // Writes |length| 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;
- virtual int Rewind();
- virtual ~OutStream() {}
-protected:
- OutStream() {}
-};
-
-enum TraceModule
-{
- kTraceUndefined = 0,
- // not a module, triggered from the engine code
- kTraceVoice = 0x0001,
- // not a module, triggered from the engine code
- kTraceVideo = 0x0002,
- // not a module, triggered from the utility code
- kTraceUtility = 0x0003,
- kTraceRtpRtcp = 0x0004,
- kTraceTransport = 0x0005,
- kTraceSrtp = 0x0006,
- kTraceAudioCoding = 0x0007,
- kTraceAudioMixerServer = 0x0008,
- kTraceAudioMixerClient = 0x0009,
- kTraceFile = 0x000a,
- kTraceAudioProcessing = 0x000b,
- kTraceVideoCoding = 0x0010,
- kTraceVideoMixer = 0x0011,
- kTraceAudioDevice = 0x0012,
- kTraceVideoRenderer = 0x0014,
- kTraceVideoCapture = 0x0015,
- kTraceRemoteBitrateEstimator = 0x0017,
-};
-
-enum TraceLevel
-{
- kTraceNone = 0x0000, // no trace
- kTraceStateInfo = 0x0001,
- kTraceWarning = 0x0002,
- kTraceError = 0x0004,
- kTraceCritical = 0x0008,
- kTraceApiCall = 0x0010,
- kTraceDefault = 0x00ff,
-
- kTraceModuleCall = 0x0020,
- kTraceMemory = 0x0100, // memory info
- kTraceTimer = 0x0200, // timing info
- kTraceStream = 0x0400, // "continuous" stream of data
-
- // used for debug purposes
- kTraceDebug = 0x0800, // debug
- kTraceInfo = 0x1000, // debug info
-
- // Non-verbose level used by LS_INFO of logging.h. Do not use directly.
- kTraceTerseInfo = 0x2000,
-
- kTraceAll = 0xffff
-};
-
-// External Trace API
-class TraceCallback {
- public:
- virtual void Print(TraceLevel level, const char* message, int length) = 0;
-
- protected:
- virtual ~TraceCallback() {}
- TraceCallback() {}
-};
-
-enum FileFormats
-{
- kFileFormatWavFile = 1,
- kFileFormatCompressedFile = 2,
- kFileFormatPreencodedFile = 4,
- kFileFormatPcm16kHzFile = 7,
- kFileFormatPcm8kHzFile = 8,
- kFileFormatPcm32kHzFile = 9
-};
-
-enum ProcessingTypes
-{
- kPlaybackPerChannel = 0,
- kPlaybackAllChannelsMixed,
- kRecordingPerChannel,
- kRecordingAllChannelsMixed,
- kRecordingPreprocessing
-};
-
-enum FrameType {
- kEmptyFrame = 0,
- kAudioFrameSpeech = 1,
- kAudioFrameCN = 2,
- kVideoFrameKey = 3,
- kVideoFrameDelta = 4,
-};
-
-// Statistics for an RTCP channel
-struct RtcpStatistics {
- RtcpStatistics()
- : fraction_lost(0),
- cumulative_lost(0),
- extended_max_sequence_number(0),
- jitter(0) {}
-
- uint8_t fraction_lost;
- uint32_t cumulative_lost;
- 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;
- }
- }
-
- 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), timestamp_ms(0) {}
-
- uint32_t bitrate_bps; // Bitrate in bits per second.
- uint32_t packet_rate; // Packet rate in packets per second.
- uint64_t timestamp_ms; // Ntp timestamp in ms at time of rate estimation.
-};
-
-// Callback, used to notify an observer whenever new rates have been estimated.
-class BitrateStatisticsObserver {
- public:
- virtual ~BitrateStatisticsObserver() {}
-
- virtual void Notify(const BitrateStatistics& total_stats,
- const BitrateStatistics& retransmit_stats,
- 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;
-};
-
-// ==================================================================
-// 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;
- int 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);
- }
-};
-
-// RTP
-enum {kRtpCsrcSize = 15}; // RFC 3550 page 13
-
-enum RTPDirections
-{
- kRtpIncoming = 0,
- kRtpOutgoing
-};
-
-enum PayloadFrequencies
-{
- kFreq8000Hz = 8000,
- kFreq16000Hz = 16000,
- kFreq32000Hz = 32000
-};
-
-enum VadModes // degree of bandwidth reduction
-{
- kVadConventional = 0, // lowest reduction
- kVadAggressiveLow,
- kVadAggressiveMid,
- kVadAggressiveHigh // highest reduction
-};
-
-struct NetworkStatistics // NETEQ statistics
-{
- // 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;
- // 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.
- 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;
- // 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) {}
-
- 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.
-};
-
-typedef struct
-{
- int min; // minumum
- int max; // maximum
- int average; // average
-} StatVal;
-
-typedef struct // All levels are reported in dBm0
-{
- StatVal speech_rx; // long-term speech levels on receiving side
- StatVal speech_tx; // long-term speech levels on transmitting side
- StatVal noise_rx; // long-term noise/silence levels on receiving side
- StatVal noise_tx; // long-term noise/silence levels on transmitting side
-} LevelStatistics;
-
-typedef struct // All levels are reported in dB
-{
- StatVal erl; // Echo Return Loss
- StatVal erle; // Echo Return Loss Enhancement
- StatVal rerl; // RERL = ERL + ERLE
- // Echo suppression inside EC at the point just before its NLP
- StatVal a_nlp;
-} EchoStatistics;
-
-enum NsModes // type of Noise Suppression
-{
- kNsUnchanged = 0, // previously set mode
- kNsDefault, // platform default
- kNsConference, // conferencing default
- kNsLowSuppression, // lowest suppression
- kNsModerateSuppression,
- kNsHighSuppression,
- kNsVeryHighSuppression, // highest suppression
-};
-
-enum AgcModes // type of Automatic Gain Control
-{
- kAgcUnchanged = 0, // previously set mode
- kAgcDefault, // platform default
- // adaptive mode for use when analog volume control exists (e.g. for
- // PC softphone)
- kAgcAdaptiveAnalog,
- // scaling takes place in the digital domain (e.g. for conference servers
- // and embedded devices)
- kAgcAdaptiveDigital,
- // can be used on embedded devices where the capture signal level
- // is predictable
- kAgcFixedDigital
-};
-
-// EC modes
-enum EcModes // type of Echo Control
-{
- kEcUnchanged = 0, // previously set mode
- kEcDefault, // platform default
- kEcConference, // conferencing default (aggressive AEC)
- kEcAec, // Acoustic Echo Cancellation
- kEcAecm, // AEC mobile
-};
-
-// AECM modes
-enum AecmModes // mode of AECM
-{
- kAecmQuietEarpieceOrHeadset = 0,
- // Quiet earpiece or headset use
- kAecmEarpiece, // most earpiece use
- kAecmLoudEarpiece, // Loud earpiece or quiet speakerphone use
- kAecmSpeakerphone, // most speakerphone use (default)
- kAecmLoudSpeakerphone // Loud speakerphone
-};
-
-// AGC configuration
-typedef struct
-{
- unsigned short targetLeveldBOv;
- unsigned short digitalCompressionGaindB;
- bool limiterEnable;
-} AgcConfig; // AGC configuration parameters
-
-enum StereoChannel
-{
- kStereoLeft = 0,
- kStereoRight,
- kStereoBoth
-};
-
-// Audio device layers
-enum AudioLayers
-{
- kAudioPlatformDefault = 0,
- kAudioWindowsWave = 1,
- kAudioWindowsCore = 2,
- kAudioLinuxAlsa = 3,
- kAudioLinuxPulse = 4
-};
-
-// TODO(henrika): to be removed.
-enum NetEqModes // NetEQ playout configurations
-{
- // Optimized trade-off between low delay and jitter robustness for two-way
- // communication.
- kNetEqDefault = 0,
- // Improved jitter robustness at the cost of increased delay. Can be
- // used in one-way communication.
- kNetEqStreaming = 1,
- // Optimzed for decodability of fax signals rather than for perceived audio
- // quality.
- kNetEqFax = 2,
- // Minimal buffer management. Inserts zeros for lost packets and during
- // buffer increases.
- kNetEqOff = 3,
-};
-
-// TODO(henrika): to be removed.
-enum OnHoldModes // On Hold direction
-{
- kHoldSendAndPlay = 0, // Put both sending and playing in on-hold state.
- kHoldSendOnly, // Put only sending in on-hold state.
- kHoldPlayOnly // Put only playing in on-hold state.
-};
-
-// TODO(henrika): to be removed.
-enum AmrMode
-{
- kRfc3267BwEfficient = 0,
- kRfc3267OctetAligned = 1,
- kRfc3267FileStorage = 2,
-};
-
-// ==================================================================
-// Video specific types
-// ==================================================================
-
-// Raw video types
-enum RawVideoType
-{
- kVideoI420 = 0,
- kVideoYV12 = 1,
- kVideoYUY2 = 2,
- kVideoUYVY = 3,
- kVideoIYUV = 4,
- kVideoARGB = 5,
- kVideoRGB24 = 6,
- kVideoRGB565 = 7,
- kVideoARGB4444 = 8,
- kVideoARGB1555 = 9,
- kVideoMJPEG = 10,
- kVideoNV12 = 11,
- kVideoNV21 = 12,
- kVideoBGRA = 13,
- kVideoUnknown = 99
-};
-
-// Video codec
-enum { kConfigParameterSize = 128};
-enum { kPayloadNameSize = 32};
-enum { kMaxSimulcastStreams = 4};
-enum { kMaxSpatialLayers = 5 };
-enum { kMaxTemporalStreams = 4};
-
-enum VideoCodecComplexity
-{
- kComplexityNormal = 0,
- kComplexityHigh = 1,
- kComplexityHigher = 2,
- kComplexityMax = 3
-};
-
-enum VideoCodecProfile
-{
- kProfileBase = 0x00,
- kProfileMain = 0x01
-};
-
-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.
-};
-
-// VP8 specific
-struct VideoCodecVP8 {
- bool pictureLossIndicationOn;
- bool feedbackModeOn;
- VideoCodecComplexity complexity;
- VP8ResilienceMode resilience;
- unsigned char numberOfTemporalLayers;
- bool denoisingOn;
- bool errorConcealmentOn;
- bool automaticResizeOn;
- bool frameDroppingOn;
- int keyFrameInterval;
-
- bool operator==(const VideoCodecVP8& other) const {
- return pictureLossIndicationOn == other.pictureLossIndicationOn &&
- feedbackModeOn == other.feedbackModeOn &&
- complexity == other.complexity &&
- resilience == other.resilience &&
- numberOfTemporalLayers == other.numberOfTemporalLayers &&
- denoisingOn == other.denoisingOn &&
- errorConcealmentOn == other.errorConcealmentOn &&
- automaticResizeOn == other.automaticResizeOn &&
- frameDroppingOn == other.frameDroppingOn &&
- keyFrameInterval == other.keyFrameInterval;
- }
-
- bool operator!=(const VideoCodecVP8& other) const {
- return !(*this == other);
- }
-};
-
-// VP9 specific.
-struct VideoCodecVP9 {
- VideoCodecComplexity complexity;
- int resilience;
- unsigned char numberOfTemporalLayers;
- bool denoisingOn;
- bool frameDroppingOn;
- int keyFrameInterval;
- bool adaptiveQpMode;
- bool automaticResizeOn;
- unsigned char numberOfSpatialLayers;
- bool flexibleMode;
-};
-
-// H264 specific.
-struct VideoCodecH264 {
- VideoCodecProfile profile;
- 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;
-};
-
-// Video codec types
-enum VideoCodecType {
- kVideoCodecVP8,
- kVideoCodecVP9,
- kVideoCodecH264,
- kVideoCodecI420,
- kVideoCodecRED,
- kVideoCodecULPFEC,
- kVideoCodecGeneric,
- kVideoCodecUnknown
-};
-
-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
-
- bool operator==(const SimulcastStream& other) const {
- return width == other.width &&
- height == other.height &&
- numberOfTemporalLayers == other.numberOfTemporalLayers &&
- maxBitrate == other.maxBitrate &&
- targetBitrate == other.targetBitrate &&
- minBitrate == other.minBitrate &&
- qpMax == other.qpMax;
- }
-
- bool operator!=(const SimulcastStream& other) const {
- return !(*this == other);
- }
-};
-
-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
-struct VideoCodec {
- 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.
-
- unsigned char maxFramerate;
-
- VideoCodecUnion codecSpecific;
-
- unsigned int qpMax;
- unsigned char numberOfSimulcastStreams;
- SimulcastStream simulcastStream[kMaxSimulcastStreams];
- SpatialLayer spatialLayers[kMaxSpatialLayers];
-
- VideoCodecMode mode;
-
- // When using an external encoder/decoder this allows to pass
- // extra options without requiring webrtc to be aware of them.
- Config* extra_options;
-
- bool operator==(const VideoCodec& other) const {
- bool ret = codecType == other.codecType &&
- (STR_CASE_CMP(plName, other.plName) == 0) &&
- plType == other.plType &&
- width == other.width &&
- height == other.height &&
- startBitrate == other.startBitrate &&
- maxBitrate == other.maxBitrate &&
- minBitrate == other.minBitrate &&
- targetBitrate == other.targetBitrate &&
- maxFramerate == other.maxFramerate &&
- qpMax == other.qpMax &&
- numberOfSimulcastStreams == other.numberOfSimulcastStreams &&
- mode == other.mode;
- if (ret && codecType == kVideoCodecVP8) {
- ret &= (codecSpecific.VP8 == other.codecSpecific.VP8);
- }
-
- for (unsigned char i = 0; i < other.numberOfSimulcastStreams && ret; ++i) {
- ret &= (simulcastStream[i] == other.simulcastStream[i]);
- }
- return ret;
- }
-
- bool operator!=(const VideoCodec& other) const {
- return !(*this == other);
- }
-};
-
-// Bandwidth over-use detector options. These are used to drive
-// experimentation with bandwidth estimation parameters.
-// See modules/remote_bitrate_estimator/overuse_detector.h
-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-2;
- }
- 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.
-};
-
-struct RTPHeaderExtension {
- RTPHeaderExtension();
-
- 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;
- uint8_t videoRotation;
-};
-
-struct RTPHeader {
- RTPHeader();
-
- 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 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;
- }
- }
-
- 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 };
-
-} // namespace webrtc
-
-#endif // WEBRTC_COMMON_TYPES_H_
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