/usr/include/webrtc_audio_processing/common_types.h is in libwebrtc-audio-processing-dev 0.1-3.
This file is owned by root:root, with mode 0o644.
The actual contents of the file can be viewed below.
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* Copyright (c) 2011 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 "typedefs.h"
#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
namespace webrtc {
class InStream
{
public:
virtual int Read(void *buf,int len) = 0;
virtual int Rewind() {return -1;}
virtual ~InStream() {}
protected:
InStream() {}
};
class OutStream
{
public:
virtual bool Write(const void *buf,int len) = 0;
virtual int Rewind() {return -1;}
virtual ~OutStream() {}
protected:
OutStream() {}
};
enum TraceModule
{
// 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,
kTraceVideoPreocessing = 0x0016
};
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
kTraceAll = 0xffff
};
// External Trace API
class TraceCallback
{
public:
virtual void Print(const TraceLevel level,
const char *traceString,
const int length) = 0;
protected:
virtual ~TraceCallback() {}
TraceCallback() {}
};
enum FileFormats
{
kFileFormatWavFile = 1,
kFileFormatCompressedFile = 2,
kFileFormatAviFile = 3,
kFileFormatPreencodedFile = 4,
kFileFormatPcm16kHzFile = 7,
kFileFormatPcm8kHzFile = 8,
kFileFormatPcm32kHzFile = 9
};
enum ProcessingTypes
{
kPlaybackPerChannel = 0,
kPlaybackAllChannelsMixed,
kRecordingPerChannel,
kRecordingAllChannelsMixed
};
// Encryption enums
enum CipherTypes
{
kCipherNull = 0,
kCipherAes128CounterMode = 1
};
enum AuthenticationTypes
{
kAuthNull = 0,
kAuthHmacSha1 = 3
};
enum SecurityLevels
{
kNoProtection = 0,
kEncryption = 1,
kAuthentication = 2,
kEncryptionAndAuthentication = 3
};
class Encryption
{
public:
virtual void encrypt(
int channel_no,
unsigned char* in_data,
unsigned char* out_data,
int bytes_in,
int* bytes_out) = 0;
virtual void decrypt(
int channel_no,
unsigned char* in_data,
unsigned char* out_data,
int bytes_in,
int* bytes_out) = 0;
virtual void encrypt_rtcp(
int channel_no,
unsigned char* in_data,
unsigned char* out_data,
int bytes_in,
int* bytes_out) = 0;
virtual void decrypt_rtcp(
int channel_no,
unsigned char* in_data,
unsigned char* out_data,
int bytes_in,
int* bytes_out) = 0;
protected:
virtual ~Encryption() {}
Encryption() {}
};
// External transport callback interface
class Transport
{
public:
virtual int SendPacket(int channel, const void *data, int len) = 0;
virtual int SendRTCPPacket(int channel, const void *data, int len) = 0;
protected:
virtual ~Transport() {}
Transport() {}
};
// ==================================================================
// Voice specific types
// ==================================================================
// Each codec supported can be described by this structure.
struct CodecInst
{
int pltype;
char plname[32];
int plfreq;
int pacsize;
int channels;
int rate;
};
enum FrameType
{
kFrameEmpty = 0,
kAudioFrameSpeech = 1,
kAudioFrameCN = 2,
kVideoFrameKey = 3, // independent frame
kVideoFrameDelta = 4, // depends on the previus frame
kVideoFrameGolden = 5, // depends on a old known previus frame
kVideoFrameAltRef = 6
};
// 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
WebRtc_UWord16 currentBufferSize;
// preferred (optimal) buffer size in ms
WebRtc_UWord16 preferredBufferSize;
// loss rate (network + late) in percent (in Q14)
WebRtc_UWord16 currentPacketLossRate;
// late loss rate in percent (in Q14)
WebRtc_UWord16 currentDiscardRate;
// fraction (of original stream) of synthesized speech inserted through
// expansion (in Q14)
WebRtc_UWord16 currentExpandRate;
// fraction of synthesized speech inserted through pre-emptive expansion
// (in Q14)
WebRtc_UWord16 currentPreemptiveRate;
// fraction of data removed through acceleration (in Q14)
WebRtc_UWord16 currentAccelerateRate;
};
struct JitterStatistics
{
// smallest Jitter Buffer size during call in ms
WebRtc_UWord32 jbMinSize;
// largest Jitter Buffer size during call in ms
WebRtc_UWord32 jbMaxSize;
// the average JB size, measured over time - ms
WebRtc_UWord32 jbAvgSize;
// number of times the Jitter Buffer changed (using Accelerate or
// Pre-emptive Expand)
WebRtc_UWord32 jbChangeCount;
// amount (in ms) of audio data received late
WebRtc_UWord32 lateLossMs;
// milliseconds removed to reduce jitter buffer size
WebRtc_UWord32 accelerateMs;
// milliseconds discarded through buffer flushing
WebRtc_UWord32 flushedMs;
// milliseconds of generated silence
WebRtc_UWord32 generatedSilentMs;
// milliseconds of synthetic audio data (non-background noise)
WebRtc_UWord32 interpolatedVoiceMs;
// milliseconds of synthetic audio data (background noise level)
WebRtc_UWord32 interpolatedSilentMs;
// count of tiny expansions in output audio
WebRtc_UWord32 countExpandMoreThan120ms;
// count of small expansions in output audio
WebRtc_UWord32 countExpandMoreThan250ms;
// count of medium expansions in output audio
WebRtc_UWord32 countExpandMoreThan500ms;
// count of long expansions in output audio
WebRtc_UWord32 countExpandMoreThan2000ms;
// duration of longest audio drop-out
WebRtc_UWord32 longestExpandDurationMs;
// count of times we got small network outage (inter-arrival time in
// [500, 1000) ms)
WebRtc_UWord32 countIAT500ms;
// count of times we got medium network outage (inter-arrival time in
// [1000, 2000) ms)
WebRtc_UWord32 countIAT1000ms;
// count of times we got large network outage (inter-arrival time >=
// 2000 ms)
WebRtc_UWord32 countIAT2000ms;
// longest packet inter-arrival time in ms
WebRtc_UWord32 longestIATms;
// min time incoming Packet "waited" to be played
WebRtc_UWord32 minPacketDelayMs;
// max time incoming Packet "waited" to be played
WebRtc_UWord32 maxPacketDelayMs;
// avg time incoming Packet "waited" to be played
WebRtc_UWord32 avgPacketDelayMs;
};
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 TelephoneEventDetectionMethods
{
kInBand = 0,
kOutOfBand = 1,
kInAndOutOfBand = 2
};
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 the capture signal is 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
};
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,
};
enum NetEqBgnModes // NetEQ Background Noise (BGN) configurations
{
// BGN is always on and will be generated when the incoming RTP stream
// stops (default).
kBgnOn = 0,
// The BGN is faded to zero (complete silence) after a few seconds.
kBgnFade = 1,
// BGN is not used at all. Silence is produced after speech extrapolation
// has faded.
kBgnOff = 2,
};
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.
};
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,
kVideoUnknown = 99
};
// Video codec
enum { kConfigParameterSize = 128};
enum { kPayloadNameSize = 32};
enum { kMaxSimulcastStreams = 4};
// H.263 specific
struct VideoCodecH263
{
char quality;
};
// H.264 specific
enum H264Packetization
{
kH264SingleMode = 0,
kH264NonInterleavedMode = 1
};
enum VideoCodecComplexity
{
kComplexityNormal = 0,
kComplexityHigh = 1,
kComplexityHigher = 2,
kComplexityMax = 3
};
enum VideoCodecProfile
{
kProfileBase = 0x00,
kProfileMain = 0x01
};
struct VideoCodecH264
{
H264Packetization packetization;
VideoCodecComplexity complexity;
VideoCodecProfile profile;
char level;
char quality;
bool useFMO;
unsigned char configParameters[kConfigParameterSize];
unsigned char configParametersSize;
};
// VP8 specific
struct VideoCodecVP8
{
bool pictureLossIndicationOn;
bool feedbackModeOn;
VideoCodecComplexity complexity;
unsigned char numberOfTemporalLayers;
};
// MPEG-4 specific
struct VideoCodecMPEG4
{
unsigned char configParameters[kConfigParameterSize];
unsigned char configParametersSize;
char level;
};
// Unknown specific
struct VideoCodecGeneric
{
};
// Video codec types
enum VideoCodecType
{
kVideoCodecH263,
kVideoCodecH264,
kVideoCodecVP8,
kVideoCodecMPEG4,
kVideoCodecI420,
kVideoCodecRED,
kVideoCodecULPFEC,
kVideoCodecUnknown
};
union VideoCodecUnion
{
VideoCodecH263 H263;
VideoCodecH264 H264;
VideoCodecVP8 VP8;
VideoCodecMPEG4 MPEG4;
VideoCodecGeneric Generic;
};
/*
* 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;
unsigned int qpMax; // minimum quality
};
// Common video codec properties
struct VideoCodec
{
VideoCodecType codecType;
char plName[kPayloadNameSize];
unsigned char plType;
unsigned short width;
unsigned short height;
unsigned int startBitrate;
unsigned int maxBitrate;
unsigned int minBitrate;
unsigned char maxFramerate;
VideoCodecUnion codecSpecific;
unsigned int qpMax;
unsigned char numberOfSimulcastStreams;
SimulcastStream simulcastStream[kMaxSimulcastStreams];
};
} // namespace webrtc
#endif // WEBRTC_COMMON_TYPES_H
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