Namespaces
namespace	aliases_double

namespace	aliases_float

namespace	SpectralCentroid

Classes
class	AccurateSum
	Implements Kahan algorithm for floating point accumulations. More...

class	AdditiveNoise
	Mixes additive white noise to the signal. More...

struct	allFloatsEqualTo
	Returns `true` if all the values in the range are equal to a specific value within provided tolerance, `false` otherwise. More...

struct	allFloatsEqualToEachOther
	Returns `true` if all the values in the range are equal to each other within provided tolerance, `false` otherwise. More...

struct	allNaN
	Returns `true` if all elements in the range are `NaN` More...

class	AnalysisContext
	Contains audio-related artefacts useful for analysis by matchers. More...

struct	anyNaN
	Returns `true` if at least one element in the range is `NaN` More...

struct	argmax
	Returns the zero-based index of the largest element in the range. More...

struct	argmin
	Returns the zero-based index of the smallest element in the range. More...

class	AudioBuffer
	Container for audio data. More...

class	AudioBufferSignal
	Plays audio from a pre-rendered AudioBuffer. More...

class	AudioTestBuilder
	A DSP host used for building and running tests inside a test case. More...

class	ChannelFlags
	A set of boolean flags mapped to each audio channel. More...

class	ChannelLayoutError
	Thrown when a numbers of channels is mismatched. More...

struct	ChannelSubsets
	Helpers to generate common default channel subsets. More...

struct	CLIConfig
	Holds values set by the user via CLI interface. More...

struct	collect
	Forwards the entire range as an `std::vector`, preserving the original order. More...

class	Condition
	A class representing some condition. More...

class	ConfigurationError
	Thrown when the test runner is misconfigured. More...

class	DC
	Produces a DC signal. More...

class	DSP
	Base for DSP effects. More...

class	DSPBase
	Polymorphic base for all DSP. More...

class	DSPFunction
	A DSP processor defined by a user-provided function. More...

class	DSPSequence
	A DSP unit that renders audio through a linear sequence of DSP instances. More...

class	Envelope
	Represents an Envelope curve for DSP parameters. More...

class	EqualsTo
	Checks whether the audio is identical to some signal. More...

struct	first
	Returns the first element in the range. More...

class	FundamentalEquals
	Checks the fundamental frequency of the signal. More...

class	GainDb
	Applies gain in decibels to the signal. More...

class	GainLinear
	Applies linear gain (not decibels) to the signal. More...

class	HardClip
	Applies symmetrical hard clipping (no knee) to the signal. More...

class	HartAssertException
	Thrown when `hassert()` or `hassertfalse` are triggered. More...

class	Impulse
	Produces a {1, 0, 0, 0, ...} sequence. More...

class	IndexError
	Thrown when a container index is out of range. More...

class	IOError
	Thrown when some I/O operation fails. More...

struct	last
	Returns the last element in the range. More...

class	LatencyBelow
	Checks whether the output signal latency is below a specified amount. More...

class	Matcher
	Base for audio matchers. More...

class	MatcherBase
	Polymorphic base for all matchers. More...

struct	MatcherFailureDetails
	Details about matcher failure. More...

class	MatcherFunction
	Matcher defined by a user-provided function. More...

struct	max
	Returns the largest element in the range. More...

struct	mean
	Returns the arithmetic mean of all elements in the range. More...

class	MetricQuery
	Manages the metrics calculations. More...

struct	min
	Returns the smallest element in the range. More...

class	MixedSignal
	Produces a mix of multiple signals. More...

class	Mute
	Mutes selected channels in the signal. More...

class	NoDenormals
	Checks whether the audio has no denormal values. More...

struct	nth
	Returns the nth element in the range (zero-based) More...

class	NullPointerError
	Thrown when a nullptr could be handled gracefully. More...

class	NyquistSignal
	Produces a Nyquist signal. More...

class	PeaksAt
	Checks whether the audio peaks at specific level. More...

class	PeaksBelow
	Checks whether the audio peaks below specific level. More...

struct	percentile
	Returns the percentile value. More...

class	PolarityPreserved
	Checks whether the output signal preserves the polarity of the input signal. More...

struct	range
	Returns the difference between largest and smallest values in the range. More...

class	SampleRateError
	Thrown when sample rate is mismatched. More...

class	Sawtooth
	Produces a bandlimited sawtooth wave at fixed frequency. More...

class	SegmentedEnvelope
	A simple envelope constructed from semgents. More...

class	Signal
	Base class for signals. More...

class	SignalBase
	Polymorphic base for all signals. More...

class	SignalFunction
	Signal defined by a user-provided function. More...

class	Silence
	Produces silence (zeros) More...

class	SineSweep
	Produces a sine sweep. More...

class	SineWave
	Produces a sine wave at fixed frequency. More...

struct	size
	Returns the number of elements (values) in the range. More...

class	SizeError
	Thrown when an unexpected container size is encountered. More...

struct	Slice
	Represents a slice of analysis data. More...

class	Spectrum
	Frequency-domain representation of a multi-channel audio signal. More...

class	StateError
	Thrown when some unexpected state is encountered. More...

class	StereoToMidSide
	Converts regular stereo signal into mid-side signal. More...

struct	sum
	Returns the sum of all elements in the range. More...

class	TestAssertException
	Thrown by test asserts like `HART_ASSERT_TRUE()` and `AudioTestBuilder::assertFalse()` More...

class	TestRegistry
	Runs the test cases. More...

class	TimeIt
	Helper class for measuring duration of some block of code over multiple runs. More...

class	TimeShift
	Shifts an audio signal forward in time by a fixed amount. More...

class	TruePeaksBelow
	Checks whether the audio true-peaks below specific level. More...

class	UnitError
	Thrown when some metric is requested to return a value in an unsupported unit. More...

class	UnsupportedError
	Thrown when some parameter has an unsupported value. More...

class	ValueError
	Thrown when an inappropriate value is encountered. More...

class	WavFile
	Produces audio from a wav file. More...

class	WavWriter
	Helper class for writing audio buffers to wav files. More...

class	WhiteNoise
	Produces deterministic white noise. More...

Typedefs
using	EnvelopeBuffers = std::unordered_map< int, std::vector< double > >
	Hash table of automation envelope sequences mapped to param ids.

template<typename IteratorType >
using	IteratedValueType = typename std::iterator_traits< IteratorType >::value_type
	Value of the elements in a range that reducer is supposed to reduce.

Enumerations
enum class	Preparation { none , reset , prepare , resetAndPrepare }
	Describes whether to call reset() and/or prepare() before rendering through DSP or a Signal. More...

enum class	Save { always , whenFails , never }
	Determines when to save a file. More...

enum class	ResetSignal { no , yes }
	Determines whether to reset the Signal in a given context. More...

enum class	SilencePolicy { strict , relaxed }
	Defines how silence in various algorithms. More...

enum class	Unit { native , linear , dB , frames , seconds , Hz , ratio , none }
	Represents a physical unit. More...

enum	Channel { left = 0 , right = 1 }
	Helper values for channel indices. More...

enum	MidSideChannel { mid = 0 , side = 1 }
	Helper values for mid-side channel indices. More...

enum class	Loop { no , yes }
	Helper values for something that could loop, like a Signal. More...

enum	Oversampling { x4 = 4 , x8 = 8 , x16 = 16 }
	Oversampling ratio. More...

enum class	Interpolation { nearest , linear }
	Interpolation method. More...

enum class	WavFormat { pcm16 , pcm24 , pcm32 , float32 }
	Audio data storage format for the wav files. More...

enum	CorrelationSearchMode { bestSignedCorrelation , bestAbsoluteCorrelation }
	Describes how to look for best cross-correlation. More...

Functions
	HART_DSP_DECLARE_ALIASES_FOR (DSPSequence)

	HART_DSP_DECLARE_ALIASES_FOR (DSPSequenceBuilder)

std::ostream &	linPrecision (std::ostream &stream)
	Sets number of decimal places for linear (sample) values.

std::ostream &	dbPrecision (std::ostream &stream)
	Sets number of decimal places for values in decibels.

std::ostream &	secPrecision (std::ostream &stream)
	Sets number of decimal places for values in seconds.

std::ostream &	hzPrecision (std::ostream &stream)
	Sets number of decimal places for values in hertz.

std::ostream &	radPrecision (std::ostream &stream)
	Sets number of decimal places for values in radians.

std::ostream &	centsPrecision (std::ostream &stream)
	Sets number of decimal places for values in cents (frequency deviation)

static std::ostream &	correlationPrecision (std::ostream &stream)
	Sets number of decimal places for correlation values.

template<typename DSPType >
AudioTestBuilder< typename std::decay< DSPType >::type::SampleTypePublicAlias >	processAudioWith (DSPType &&dsp)
	Call this to start building your test using a DSP object.

template<typename DSPType >
AudioTestBuilder< typename DSPType::SampleTypePublicAlias >	processAudioWith (std::unique_ptr< DSPType > &&dsp)
	Call this to start building your test using a smart pointer to a DSP object.

std::ostream &	operator<< (std::ostream &os, Oversampling oversampling)

template<typename FloatType >
FloatType	nan ()
	Returns a quiet NaN value for the given floating-point type.

template<typename NumericType >
NumericType	clamp (const NumericType &value, const NumericType &low, const NumericType &high)
	`std::clamp()` replacement for C++11

template<typename SampleType >
static SampleType	decibelsToRatio (SampleType valueDb)
	Converts dB to linear value (ratio)

template<typename SampleType >
static SampleType	ratioToDecibels (SampleType valueLinear)
	Converts linear value (ratio) to dB.

template<typename SampleType >
static SampleType	decibelsToPower (SampleType valueDb)
	Converts dB to linear value (power)

template<typename SampleType >
static SampleType	powerToDecibels (SampleType valueLinear)
	Converts linear value (power) to dB.

template<typename SampleType >
static SampleType	floatsEqual (SampleType a, SampleType b, SampleType epsilon=(SampleType) 1e-8)
	Compares two floating point numbers within a given tolerance.

template<typename SampleType >
static SampleType	floatsNotEqual (SampleType a, SampleType b, SampleType epsilon=(SampleType) 1e-8)
	Compares two floating point numbers within a given tolerance.

template<typename SampleType >
static size_t	roundToSizeT (SampleType x)
	Rounds a floating point value to a `size_t` value.

double	addCents (double baseFrequencyHz, double cents)
	Anns an offset in cents to a frequency in Hz.

template<typename SampleType >
SampleType	wrapPhase (const SampleType phaseRadians)
	Keeps phase in 0..twoPi range.

static bool	isAbsolutePath (const std::string &path)
	Checks if the provided file path is absolute.

static std::string	toAbsolutePath (const std::string &path)
	Converts path to absolute, if it's relative.

template<typename KeyType , typename ValueType >
static bool	contains (const std::unordered_map< KeyType, ValueType > &map, const KeyType &key)
	`std::unordered_map::contains()` replacement for C++11

template<typename ObjectType , typename... Args>
std::unique_ptr< ObjectType >	make_unique (Args &&... args)
	`std::make_unique()` replacement for C++11

static bool	isExceptionUnwinding ()
	Returns `true` if an exception is currently being unwound.

template<typename SampleType >
MetricQuery< double >	channelCorrelation (const AudioBuffer< SampleType > &buffer)
	Calculates zero-lag normalized cross-correlation between two channels of an audio buffer.

template<typename SampleType >
MetricQuery< double >	crestFactor (const AudioBuffer< SampleType > &buffer)
	Calculates linear crest factor for a single channel of an audio buffer.

template<typename SampleType >
MetricQuery< double >	esr (const AudioBuffer< SampleType > &referenceBuffer, const AudioBuffer< SampleType > &estimatedBuffer)
	Calculates error-to-signal ratio (ESR)

MetricQuery< double >	interpolatedPeakFrequency (const Spectrum &spectrum)
	Returns the center frequency of the loudest FFT bin.

template<typename SampleType >
MetricQuery< double >	lagAtMaxCrossCorrelation (const AudioBuffer< SampleType > &bufferA, const AudioBuffer< SampleType > &bufferB, double maxLagSeconds, double minAbsBestCorrelation=0.5, CorrelationSearchMode searchMode=bestAbsoluteCorrelation)
	Calculates lag corresponding to maximum normalized cross-correlation between two audio buffers.

MetricQuery< double >	loudestBinFrequency (const Spectrum &spectrum)
	Returns the center frequency of the loudest FFT bin.

MetricQuery< double >	loudestBinMagnitude (const Spectrum &spectrum)
	Calculates the magnitude of the loudest FFT bin.

template<typename SampleType >
MetricQuery< double >	maxCrossCorrelation (const AudioBuffer< SampleType > &bufferA, const AudioBuffer< SampleType > &bufferB, double maxLagSeconds, CorrelationSearchMode searchMode=bestAbsoluteCorrelation)
	Calculates maximum normalized cross-correlation between two audio buffers.

MetricQuery< double >	quinns2 (const Spectrum &spectrum)
	Returns somewhat accurate loudest frequency in the spectrum.

template<typename SampleType >
MetricQuery< double >	rms (const AudioBuffer< SampleType > &buffer)
	Calculates root mean square (RMS) of a signal.

template<typename SampleType >
MetricQuery< double >	samplePeak (const AudioBuffer< SampleType > &audioBuffer)
	Calculates Sample Peak of an audio buffer.

MetricQuery< double >	spectralCentroid (const Spectrum &spectrum, SpectralCentroid::Weighting weighting=SpectralCentroid::Weighting::magnitude)
	Calculates spectral centroid.

MetricQuery< double >	spectralFlatness (const Spectrum &spectrum, double floorLinear=1e-16)
	Calculates spectral flatness, also known as Wiener entropy, or tonality coefficient.

template<typename SampleType >
MetricQuery< double >	truePeak (const AudioBuffer< SampleType > &audioBuffer, Oversampling oversamplingRatio=Oversampling::x4, typename TruePeak< SampleType >::FilterQuality filterQuality=TruePeak< SampleType >::FilterQuality::low)
	Estimates true peak (inter-sample peak) level.

template<typename SampleType >
MetricQuery< double >	zcr (const AudioBuffer< SampleType > &buffer)
	Calculates zero-crossing rate (ZCR) of a signal.

Variables
static const char *	hartAsciiArt = "88 \n88 ,d \n88 88 \n88,dPPYba, ,adPPYYba, 8b,dPPYba, MM88MMM\n88P' \"8a \"\" `Y8 88P' \"Y8 88 \n88 88 ,adPPPPP88 88 88 \n88 88 88, ,88 88 88, \n88 88 `\"8bbdP\"Y8 88 \"Y888\n"

static const char *	passAsciiArt = "8b,dPPYba, ,adPPYYba, ,adPPYba, ,adPPYba,\n88P' \"8a \"\" `Y8 I8[ \"\" I8[ \"\"\n88 d8 ,adPPPPP88 `\"Y8ba, `\"Y8ba, \n88b, ,a8\" 88, ,88 aa ]8I aa ]8I\n88`YbbdP\"' `\"8bbdP\"Y8 `\"YbbdP\"' `\"YbbdP\"'\n88 \n88\n"

static const char *	failAsciiArt = " ad88 88 88\n d8\" \"\" 88\n 88 88\nMM88MMM ,adPPYYba, 88 88\n 88 \"\" `Y8 88 88\n 88 ,adPPPPP88 88 88\n 88 88, ,88 88 88\n 88 `\"8bbdP\"Y8 88 88\n"

constexpr double	inf = std::numeric_limits<double>::infinity()
	Infinity.

constexpr double	oo = inf
	Infinity.

constexpr double	pi = 3.14159265358979323846
	pi

constexpr double	twoPi = 2.0 * pi
	2 * pi

constexpr double	halfPi = pi / 2.0
	pi / 2

Enumeration Type Documentation

◆ Preparation

enum class Preparation

strong

Describes whether to call reset() and/or prepare() before rendering through DSP or a Signal.

Enumerator
none
reset
prepare
resetAndPrepare

Definition at line 7 of file hart_preparation.hpp.

◆ Unit

enum class Unit

strong

Represents a physical unit.

You might want to use those without namespace, like someMetric (...).as (dB). If you're on C++17 or earlier, you can use the HART_DECLARE_ALIASES_FOR_UNITS macro. For C++20 an on, you can just use using enum Unit.

Note: New units may be introduced here, whenever necessary

Enumerator
native	Default (native) unit of whatever returns some value.
linear	Value of a sample (voltage) in a linear domain.
dB	Value of something in decibels. Can represent voltage, power, or a domain-specific unit like "LUFS" or "dB TP".
frames	Value of something in frames (samples)
seconds	Time stamps, intervals, durations.
Hz	Hertz.
ratio	Generic ratio.
none	Unitless value.

Definition at line 20 of file hart_units.hpp.

◆ WavFormat

enum class WavFormat

strong

Audio data storage format for the wav files.

Enumerator
pcm16
pcm24
pcm32
float32

Definition at line 7 of file hart_wavformat.hpp.

◆ CorrelationSearchMode

enum CorrelationSearchMode

Describes how to look for best cross-correlation.

Enumerator
bestSignedCorrelation
bestAbsoluteCorrelation

Definition at line 107 of file hart_metrics_common.hpp.

Function Documentation

◆ HART_DSP_DECLARE_ALIASES_FOR() [1/2]

HART_DSP_DECLARE_ALIASES_FOR ( DSPSequence )

◆ HART_DSP_DECLARE_ALIASES_FOR() [2/2]

HART_DSP_DECLARE_ALIASES_FOR ( DSPSequenceBuilder )

◆ esr()

template<typename SampleType >

MetricQuery< double > esr	(	const AudioBuffer< SampleType > &	referenceBuffer,
		const AudioBuffer< SampleType > &	estimatedBuffer
	)

Calculates error-to-signal ratio (ESR)

ESR is a useful way to express the degree of similarity between two signals or waveforms, calculated as:

\[ ESR=\frac{\sum_{k=0}^{N-1} (x - y) ^ 2}{\sum_{k=0}^{N-1} x^2}$$ \]

(sum ((x - y) ** 2) / sum (x ** 2))

Where x is a signal represented by referenceBuffer and y is represented by estimatedBuffer. It's a ratio, so appropriate units are Unit::native and Unit::ratio. ESR = 0 means two signals are identical.

Parameters

referenceBuffer	A buffer representing x in the formula above
estimatedBuffer	A buffer representing y in the formula above

Returns: Chainable MetricQuery, which calculates per-channel ESR values

Definition at line 34 of file hart_esr.hpp.

◆ rms()

template<typename SampleType >

MetricQuery< double > rms ( const AudioBuffer< SampleType > & buffer )

Calculates root mean square (RMS) of a signal.

RMS a metric that expresses the average magnitude, or effective energy level, of an audio signal over time. It is commonly used to estimate perceived loudness, and to measure overall signal level.

RMS is calculated this way:

\[ \mathrm{RMS} = \sqrt{\frac{1}{N} \sum_{n=0}^{N-1} x[n]^2} \]

(RMS = sqrt((1 / N) * sum(x[n] ** 2))),

where x[n] is audio sample value from one channel, and N is number of frames in the provided buffer.

Can be expressed as ratio or decibels, supports Unit:native, Unit::linear, Unit::dB units. Value in decibels is calculated as a ratio, not power.

Template Parameters

SampleType

Parameters

buffer Audio buffer to calculate RMS at

Returns: Chainable MetricQuery object, which calculates RMS as linear ratio or decibels

Definition at line 40 of file hart_rms.hpp.

◆ zcr()

template<typename SampleType >

MetricQuery< double > zcr ( const AudioBuffer< SampleType > & buffer )

Calculates zero-crossing rate (ZCR) of a signal.

Useful to estimate frequency of stationary monophonic signals. Supports Unit::native and Unit::Hz units, which both result in the same value.

Template Parameters

SampleType type of audio buffer data, typically float or double

Parameters

buffer Audio buffer to calculate ZCR at

Returns: Chainable MetricQuery object, which calculates RMS as linear ratio or decibels

Definition at line 22 of file hart_zcr.hpp.

Variable Documentation

◆ hartAsciiArt

const char* hartAsciiArt = "88 \n88 ,d \n88 88 \n88,dPPYba, ,adPPYYba, 8b,dPPYba, MM88MMM\n88P' \"8a \"\" `Y8 88P' \"Y8 88 \n88 88 ,adPPPPP88 88 88 \n88 88 88, ,88 88 88, \n88 88 `\"8bbdP\"Y8 88 \"Y888\n"

static

Definition at line 8 of file hart_ascii_art.hpp.

◆ passAsciiArt

const char* passAsciiArt = "8b,dPPYba, ,adPPYYba, ,adPPYba, ,adPPYba,\n88P' \"8a \"\" `Y8 I8[ \"\" I8[ \"\"\n88 d8 ,adPPPPP88 `\"Y8ba, `\"Y8ba, \n88b, ,a8\" 88, ,88 aa ]8I aa ]8I\n88`YbbdP\"' `\"8bbdP\"Y8 `\"YbbdP\"' `\"YbbdP\"'\n88 \n88\n"

static

Definition at line 9 of file hart_ascii_art.hpp.

◆ failAsciiArt

const char* failAsciiArt = " ad88 88 88\n d8\" \"\" 88\n 88 88\nMM88MMM ,adPPYYba, 88 88\n 88 \"\" `Y8 88 88\n 88 ,adPPPPP88 88 88\n 88 88, ,88 88 88\n 88 `\"8bbdP\"Y8 88 88\n"

static

Definition at line 10 of file hart_ascii_art.hpp.

Namespaces

Classes

Typedefs

Enumerations

Functions

Variables

Enumeration Type Documentation

◆ Preparation

◆ Unit

◆ WavFormat

◆ CorrelationSearchMode

Function Documentation

◆ HART_DSP_DECLARE_ALIASES_FOR() [1/2]

◆ HART_DSP_DECLARE_ALIASES_FOR() [2/2]

◆ esr()

◆ rms()

◆ zcr()

Variable Documentation

◆ hartAsciiArt

◆ passAsciiArt

◆ failAsciiArt