Classes
class	audio_effect
class	modulation_effect
class	simple_phaser
class	chorus_base
class	simple_chorus
class	simple_flanger
class	reverb
class	filter_module_iface
class	biquad_filter_module
class	two_band_eq
class	simple_lfo
class	lookahead_limiter
	Lookahead Limiter by Markus Schmidt and Christian Holschuh. More...
class	transients
class	crossover
class	bitreduction
class	resampleN
class	samplereduction
class	tap_distortion
class	overlap_window
class	median_stat
struct	benchmark_globals
class	simple_benchmark
class	biquad_coeffs
struct	biquad_d1
struct	biquad_d2
struct	biquad_d1_lerp
class	filter_compose
	Compose two filters in series. More...
class	filter_sum
	Compose two filters in parallel. More...
struct	sample_traits
struct	sample_traits< stereo_sample< T > >
class	fixed_size_buffer
class	mem_fixed_size_buffer
class	auto_buffer
class	dynamic_buffer
struct	buffer_traits
struct	buffer_traits< fixed_size_buffer< N, T > >
	this class template defines some basic position operations for fixed_size_buffers More...
class	circular_buffer
	this is useless for now (and untested too) More...
class	mono_auto_buffer
	this is useless for now More...
class	stereo_auto_buffer
	this is useless for now More...
class	bypass
struct	simple_delay
class	adsr
struct	fadeout
	Simple linear fade out for note tails. More...
class	fft
class	fixed_point
class	linear_ramp
	Algorithm for a constant time linear ramp. More...
class	exponential_ramp
	Algorithm for a constant time linear ramp. More...
class	inertia
class	once_per_n
class	gain_smoothing
class	switcher
class	aweighter
class	riaacurve
struct	modulation_entry
	Single entry in modulation matrix. More...
class	sine_multi_lfo
class	multichorus
class	onepole
struct	organ_parameters
class	organ_voice_base
class	organ_vibrato
	A simple (and bad) simulation of scanner vibrato based on a series of modulated allpass filters. More...
class	scanner_vibrato
class	organ_voice
class	percussion_voice
	Not a true voice, just something with similar-ish interface. More...
struct	drawbar_organ
struct	simple_oscillator
struct	bandlimiter
struct	waveform_family
	Set of bandlimited wavetables. More...
struct	waveform_oscillator
struct	triangle_lfo
struct	stereo_sample
class	decay
class	sine_table
struct	note_desc
class	keystack
class	voice
class	block_allvoices_base
class	block_voice
struct	basic_pool
	A basic preallocated var-array with append and. More...
struct	basic_synth
struct	vumeter
	Peak meter class. More...
struct	dual_vumeter
class	wave_player

Typedefs
typedef fixed_point< unsigned long long int, 24 >	wpos
	wave position (unsigned 64-bit int including 24-bit fractional part)
typedef fixed_point< unsigned int, 20 >	chorus_phase

Enumerations
enum	periodic_unit { UNIT_BPM , UNIT_MS , UNIT_HZ , UNIT_SYNC }

Functions
template<class T>
void	do_simple_benchmark (int runs=5, int repeats=50000)
template<int N>
int	wrap_around (int a)
	decrease by N if >= N (useful for circular buffers)
template<>
int	wrap_around< 2 > (int a)
template<>
int	wrap_around< 4 > (int a)
template<>
int	wrap_around< 8 > (int a)
template<>
int	wrap_around< 16 > (int a)
template<>
int	wrap_around< 32 > (int a)
template<>
int	wrap_around< 64 > (int a)
template<>
int	wrap_around< 128 > (int a)
template<>
int	wrap_around< 256 > (int a)
template<>
int	wrap_around< 512 > (int a)
template<>
int	wrap_around< 1024 > (int a)
template<>
int	wrap_around< 2048 > (int a)
template<>
int	wrap_around< 4096 > (int a)
template<>
int	wrap_around< 8192 > (int a)
template<>
int	wrap_around< 16384 > (int a)
template<>
int	wrap_around< 32768 > (int a)
template<>
int	wrap_around< 65536 > (int a)
template<class Buf, class T>
void	fill (Buf &buf, T value)
template<class T>
void	fill (T *data, int size, T value)
template<class T, class U>
void	copy (T dest, U src, int size, T scale=1, T add=0)
template<class T, class U>
void	copy_buf (T &dest_buf, const U &src_buf, T scale=1, T add=0)
uint32_t	shr (uint32_t v, int bits=1)
int32_t	shr (int32_t v, int bits=1)
uint64_t	shr (uint64_t v, int bits=1)
int64_t	shr (int64_t v, int bits=1)
float	shr (float v, int bits=1)
double	shr (double v, int bits=1)
template<class T, int FracBits>
T	shr (T v, int bits=1)
template<class T, int FractBits>
fixed_point< T, FractBits >	operator* (int v, fixed_point< T, FractBits > v2)
static void	normalize_waveform (float *table, unsigned int size)
	Simple stupid inline function to normalize a waveform (by removing DC offset and ensuring max absolute value of 1).
void	zero (float &v)
	Set a float to zero.
void	zero (double &v)
	Set a double to zero.
void	zero (uint64_t &v)
	Set 64-bit unsigned integer value to zero.
void	zero (uint32_t &v)
	Set 32-bit unsigned integer value to zero.
void	zero (uint16_t &v)
	Set 16-bit unsigned integer value to zero.
void	zero (uint8_t &v)
	Set 8-bit unsigned integer value to zero.
void	zero (int64_t &v)
	Set 64-bit signed integer value to zero.
void	zero (int32_t &v)
	Set 32-bit signed integer value to zero.
void	zero (int16_t &v)
	Set 16-bit signed integer value to zero.
void	zero (int8_t &v)
	Set 8-bit signed integer value to zero.
template<class T>
void	zero (T *data, unsigned int size)
	Set array (buffer or anything similar) to vector of zeroes.
template<class T>
void	fill (T *data, T value, unsigned int size)
	Set array (buffer or anything similar) to vector of values.
template<class T>
stereo_sample< T >	operator* (const T &value, const stereo_sample< T > &value2)
	Multiply constant by stereo_value.
template<class T>
stereo_sample< T >	operator+ (const T &value, const stereo_sample< T > &value2)
	Add constant to stereo_value.
template<class T>
stereo_sample< T >	operator- (const T &value, const stereo_sample< T > &value2)
	Subtract stereo_value from constant (yields stereo_value of course).
template<typename T>
stereo_sample< T >	shr (stereo_sample< T > v, int bits=1)
	Shift value right by 'bits' bits (multiply by 2^-bits).
template<typename T>
void	zero (stereo_sample< T > &v)
	Set a stereo_sample<T> value to zero.
template<typename T>
T	small_value ()
	'Small value' for integer and other types
template<>
float	small_value< float > ()
	'Small value' for floats (2^-24) - used for primitive underrun prevention. The value is pretty much arbitrary (allowing for 24-bit signals normalized to 1.0).
template<>
double	small_value< double > ()
	'Small value' for doubles (2^-24) - used for primitive underrun prevention. The value is pretty much arbitrary.
template<typename T>
float	mono (T v)
	Convert a single value to single value = do nothing :) (but it's a generic with specialisation for stereo_sample).
template<typename T>
T	mono (stereo_sample< T > v)
	Convert a stereo_sample to single value by averaging two channels.
template<typename T>
T	clip (T value, T min, T max)
	Clip a value to [min, max].
double	clip11 (double value)
	Clip a double to [-1.0, +1.0].
float	clip11 (float value)
	Clip a float to [-1.0f, +1.0f].
double	clip01 (double value)
	Clip a double to [0.0, +1.0].
float	clip01 (float value)
	Clip a float to [0.0f, +1.0f].
template<typename T, typename U>
T	lerp (T v1, T v2, U mix)
template<typename T>
stereo_sample< T >	lerp (stereo_sample< T > &v1, stereo_sample< T > &v2, float mix)
void	sanitize (float &value)
float	_sanitize (float value)
void	sanitize_denormal (float &value)
void	sanitize_denormal (double &value)
void	sanitize (double &value)
double	_sanitize (double value)
template<class T>
void	sanitize (stereo_sample< T > &value)
float	fract16 (unsigned int value)
int	fastf2i_drm (float f)
	fast float to int conversion using default rounding mode
float	note_to_hz (double note, double detune_cents=0.0)
	Convert MIDI note to frequency in Hz.
float	normalized_hermite (float t, float p0, float p1, float m0, float m1)
float	hermite_interpolation (float x, float x0, float x1, float p0, float p1, float m0, float m1)
float	amp2dB (float amp)
	convert amplitude value to dB
float	dB2amp (float db)
	convert dB to amplitude value
void	print_bits (size_t const size, void const *const ptr)
note_desc	hz_to_note (double hz, double tune)
double	convert_periodic (double val, periodic_unit unit_in, periodic_unit unit_out)
unsigned int	midi_note_to_phase (int note, double cents, int sr)
template<class T>
bool	load_wave (dsp::dynamic_buffer< T > &dest, const char *file_name)

Variables
template<class T, int N, int Multiplier>
bool	sine_table< T, N, Multiplier >::initialized = false
template<class T, int N, int Multiplier>
T	sine_table< T, N, Multiplier >::data [N+1]

Typedef Documentation

◆ chorus_phase

typedef fixed_point<unsigned int, 20> dsp::chorus_phase

◆ wpos

typedef fixed_point<unsigned long long int, 24> dsp::wpos

wave position (unsigned 64-bit int including 24-bit fractional part)

Enumeration Type Documentation

◆ periodic_unit

enum dsp::periodic_unit

Enumerator
UNIT_BPM
UNIT_MS
UNIT_HZ
UNIT_SYNC

Function Documentation

◆ _sanitize() [1/2]

double dsp::_sanitize ( double value )

inline

◆ _sanitize() [2/2]

float dsp::_sanitize ( float value )

inline

◆ amp2dB()

float dsp::amp2dB ( float amp )

inline

convert amplitude value to dB

◆ clip()

template<typename T>

T dsp::clip	(	T	value,
		T	min,
		T	max )

inline

Clip a value to [min, max].

◆ clip01() [1/2]

double dsp::clip01 ( double value )

inline

Clip a double to [0.0, +1.0].

◆ clip01() [2/2]

float dsp::clip01 ( float value )

inline

Clip a float to [0.0f, +1.0f].

◆ clip11() [1/2]

double dsp::clip11 ( double value )

inline

Clip a double to [-1.0, +1.0].

◆ clip11() [2/2]

float dsp::clip11 ( float value )

inline

Clip a float to [-1.0f, +1.0f].

◆ convert_periodic()

double dsp::convert_periodic	(	double	val,
		periodic_unit	unit_in,
		periodic_unit	unit_out )

inline

◆ copy()

template<class T, class U>

void dsp::copy	(	T *	dest,
		U *	src,
		int	size,
		T	scale = 1,
		T	add = 0 )

◆ copy_buf()

template<class T, class U>

void dsp::copy_buf	(	T &	dest_buf,
		const U &	src_buf,
		T	scale = 1,
		T	add = 0 )

◆ dB2amp()

float dsp::dB2amp ( float db )

inline

convert dB to amplitude value

◆ do_simple_benchmark()

template<class T>

void dsp::do_simple_benchmark	(	int	runs = 5,
		int	repeats = 50000 )

◆ fastf2i_drm()

int dsp::fastf2i_drm ( float f )

inline

fast float to int conversion using default rounding mode

◆ fill() [1/3]

template<class Buf, class T>

void dsp::fill	(	Buf &	buf,
		T	value )

◆ fill() [2/3]

template<class T>

void dsp::fill	(	T *	data,
		int	size,
		T	value )

◆ fill() [3/3]

template<class T>

void dsp::fill	(	T *	data,
		T	value,
		unsigned int	size )

Set array (buffer or anything similar) to vector of values.

◆ fract16()

float dsp::fract16 ( unsigned int value )

inline

◆ hermite_interpolation()

float dsp::hermite_interpolation	(	float	x,
		float	x0,
		float	x1,
		float	p0,
		float	p1,
		float	m0,
		float	m1 )

inline

Hermite interpolation between two points and slopes

x point within interval (x0 <= x <= x1)
x0 interval start
x1 interval end
p0 value at x0
p1 value at x1
m0 slope (steepness, tangent) at x0
m1 slope at x1

◆ hz_to_note()

note_desc dsp::hz_to_note	(	double	hz,
		double	tune )

inline

◆ lerp() [1/2]

template<typename T>

stereo_sample< T > dsp::lerp	(	stereo_sample< T > &	v1,
		stereo_sample< T > &	v2,
		float	mix )

inline

◆ lerp() [2/2]

template<typename T, typename U>

T dsp::lerp	(	T	v1,
		T	v2,
		U	mix )

inline

◆ load_wave()

template<class T>

bool dsp::load_wave	(	dsp::dynamic_buffer< T > &	dest,
		const char *	file_name )

◆ midi_note_to_phase()

unsigned int dsp::midi_note_to_phase	(	int	note,
		double	cents,
		int	sr )

inline

Convert MIDI note number to normalized UINT phase (where 1<<32 is full cycle).

Parameters

MIDI	note number
cents	detune in cents (1/100 of a semitone)
sr	sample rate

◆ mono() [1/2]

template<typename T>

T dsp::mono ( stereo_sample< T > v )

inline

Convert a stereo_sample to single value by averaging two channels.

◆ mono() [2/2]

template<typename T>

float dsp::mono ( T v )

inline

Convert a single value to single value = do nothing :) (but it's a generic with specialisation for stereo_sample).

◆ normalize_waveform()

void dsp::normalize_waveform	(	float *	table,
		unsigned int	size )

inlinestatic

Simple stupid inline function to normalize a waveform (by removing DC offset and ensuring max absolute value of 1).

◆ normalized_hermite()

float dsp::normalized_hermite	(	float	t,
		float	p0,
		float	p1,
		float	m0,
		float	m1 )

inline

Hermite interpolation between two points and slopes in normalized range (written after Wikipedia article)

t normalized x coordinate (0-1 over the interval in question)
p0 first point
p1 second point
m0 first slope (multiply by interval width when using over non-1-wide interval)
m1 second slope (multiply by interval width when using over non-1-wide interval)

◆ note_to_hz()

float dsp::note_to_hz	(	double	note,
		double	detune_cents = 0.0 )

inline

Convert MIDI note to frequency in Hz.

◆ operator*() [1/2]

template<class T>

stereo_sample< T > dsp::operator*	(	const T &	value,
		const stereo_sample< T > &	value2 )

inline

Multiply constant by stereo_value.

◆ operator*() [2/2]

template<class T, int FractBits>

fixed_point< T, FractBits > dsp::operator*	(	int	v,
		fixed_point< T, FractBits >	v2 )

inline

◆ operator+()

template<class T>

stereo_sample< T > dsp::operator+	(	const T &	value,
		const stereo_sample< T > &	value2 )

inline

Add constant to stereo_value.

◆ operator-()

template<class T>

stereo_sample< T > dsp::operator-	(	const T &	value,
		const stereo_sample< T > &	value2 )

inline

Subtract stereo_value from constant (yields stereo_value of course).

◆ print_bits()

void dsp::print_bits	(	size_t const	size,
		void const *const	ptr )

inline

print binary of any data type assumes little endian

◆ sanitize() [1/3]

void dsp::sanitize ( double & value )

inline

Force "small enough" double value to zero

◆ sanitize() [2/3]

void dsp::sanitize ( float & value )

inline

Force "small enough" float value to zero

◆ sanitize() [3/3]

template<class T>

void dsp::sanitize ( stereo_sample< T > & value )

inline

Force "small enough" stereo value to zero

◆ sanitize_denormal() [1/2]

void dsp::sanitize_denormal ( double & value )

inline

Force already-denormal float value to zero

◆ sanitize_denormal() [2/2]

void dsp::sanitize_denormal ( float & value )

inline

Force already-denormal float value to zero

◆ shr() [1/8]

double dsp::shr	(	double	v,
		int	bits = 1 )

inline

◆ shr() [2/8]

float dsp::shr	(	float	v,
		int	bits = 1 )

inline

◆ shr() [3/8]

int32_t dsp::shr	(	int32_t	v,
		int	bits = 1 )

inline

◆ shr() [4/8]

int64_t dsp::shr	(	int64_t	v,
		int	bits = 1 )

inline

◆ shr() [5/8]

template<typename T>

stereo_sample< T > dsp::shr	(	stereo_sample< T >	v,
		int	bits = 1 )

inline

Shift value right by 'bits' bits (multiply by 2^-bits).

◆ shr() [6/8]

template<class T, int FracBits>

T dsp::shr	(	T	v,
		int	bits = 1 )

inline

◆ shr() [7/8]

uint32_t dsp::shr	(	uint32_t	v,
		int	bits = 1 )

inline

◆ shr() [8/8]

uint64_t dsp::shr	(	uint64_t	v,
		int	bits = 1 )

inline

◆ small_value()

template<typename T>

T dsp::small_value ( )

inline

'Small value' for integer and other types

◆ small_value< double >()

template<>

double dsp::small_value< double > ( )

inline

'Small value' for doubles (2^-24) - used for primitive underrun prevention. The value is pretty much arbitrary.

◆ small_value< float >()

template<>

float dsp::small_value< float > ( )

inline

'Small value' for floats (2^-24) - used for primitive underrun prevention. The value is pretty much arbitrary (allowing for 24-bit signals normalized to 1.0).

◆ wrap_around()

◆ zero() [8/12]

template<class T>

void dsp::zero	(	T *	data,
		unsigned int	size )

Classes

Typedefs

Enumerations

Functions

Variables

Typedef Documentation

◆ chorus_phase

◆ wpos

Enumeration Type Documentation

◆ periodic_unit

Function Documentation

◆ _sanitize() [1/2]

◆ _sanitize() [2/2]

◆ amp2dB()

◆ clip()

◆ clip01() [1/2]

◆ clip01() [2/2]

◆ clip11() [1/2]

◆ clip11() [2/2]

◆ convert_periodic()

◆ copy()

◆ copy_buf()

◆ dB2amp()

◆ do_simple_benchmark()

◆ fastf2i_drm()

◆ fill() [1/3]

◆ fill() [2/3]

◆ fill() [3/3]

◆ fract16()

◆ hermite_interpolation()

◆ hz_to_note()

◆ lerp() [1/2]

◆ lerp() [2/2]

◆ load_wave()

◆ midi_note_to_phase()

◆ mono() [1/2]

◆ mono() [2/2]

◆ normalize_waveform()

◆ normalized_hermite()

◆ note_to_hz()

◆ operator*() [1/2]

◆ operator*() [2/2]

◆ operator+()

◆ operator-()

◆ print_bits()

◆ sanitize() [1/3]

◆ sanitize() [2/3]

◆ sanitize() [3/3]

◆ sanitize_denormal() [1/2]

◆ sanitize_denormal() [2/2]

◆ shr() [1/8]

◆ shr() [2/8]

◆ shr() [3/8]

◆ shr() [4/8]

◆ shr() [5/8]

◆ shr() [6/8]

◆ shr() [7/8]

◆ shr() [8/8]

◆ small_value()

◆ small_value< double >()

◆ small_value< float >()

◆ wrap_around()

◆ wrap_around< 1024 >()

◆ wrap_around< 128 >()

◆ wrap_around< 16 >()

◆ wrap_around< 16384 >()

◆ wrap_around< 2 >()

◆ wrap_around< 2048 >()

◆ wrap_around< 256 >()

◆ wrap_around< 32 >()

◆ wrap_around< 32768 >()

◆ wrap_around< 4 >()

◆ wrap_around< 4096 >()

◆ wrap_around< 512 >()

◆ wrap_around< 64 >()

◆ wrap_around< 65536 >()

◆ wrap_around< 8 >()

◆ wrap_around< 8192 >()

◆ zero() [1/12]

◆ zero() [2/12]