All SFPL Processors As Of 27 November 2012
This is constant work in progress - I am working my way down the list as I get chance - and updating as stuff gets added/updated. Please note that some times the descriptions says 'change the volume...' for example. This is just to make reading easier; SFPL has 'immutable' types so it is not possible to change the volume of a signal or trim white space off a string. What actually happens is that a new one is made and the old one left unchanged.
December 2013 - this is quite out of data, I will work on it over the next few days...
December 2013 - this is quite out of data, I will work on it over the next few days...
- MakeSawTooth
Convert a signal into a saw tooth using center crossing - MakeSquare
Convert a signal into a square wave using center crossing - SubOctave
Make a sin wave the period of which is half the rate of center crossing. - MakeTriangle
Make a signal into a triangle wave using centre crossing - Saturate
Distortion effect using 1-(1/x) law - ShapedThreshold
Convert a signal into a pulse wave using a threshold based on a shape - Threshold
Convert a signal into a pulse wave using a fixed threshold - WaveShaper
Distort a wave using a 6th order polynomial transfer function - ClipToSafe
Use DC removal and fadeout to attempt to get rid of clicks entering and leaving a audio signal - Concatenate
Join two or more audio signals end to end - Convolve
Convolve one signal with another. Convolution is forwards and backwards, i.e. the convolving signals centre is placed at the current sample on the main signal. - ConvolveDelay
Mix a signal with a delayed version of its self where the delayed version is actually mixed with the original via convolution via a second convolving signal. - ConvolveForward
Convolve one signal with another. Convolution is forwards only (see Convolve). - CrossFadeSplit
Create a bunch of subsections of a signal where the sub sections are overlapping and are pre-faded. - Cut
Take a subsection of a signal. - Divide
Sample my sample division of one signal by another. - Mix
Mix two or more signals together using direct sample addition. - MixAt
Mix two or more signals together using direct sample addition and with defined offsets for each. - Multiply
Sample by sample multiplication of one signal by another. - GaussianShape
Generate a signal which follows a Gaussian shape. - HammingShape
Generate a signal which follows a Hamming shape. - HannShape
Generate a signal which follows a Hann shape. - Note
Take a string representation of a note in equal temperament and forward the frequency in Hz based on A4 being 440Hz. It uses a post fix of b or # for flat and sharp. E.g. "C5#" "E2b". - NumericShape
Takes a bunch of time and magnitude values and creates a signal from the linear interpolation of the values. - PhasedSinWave
Create a sin wave of a given length and frequency and starting at a given phase. - ShapedSinWave
Create a sin wave the frequency of which is continuously set by a shape signal. This effectively is an infinitely variable sin wave generator and can create things like FM. - Silence
Generate a given length of silence. - SimpleShape
Generate a shape signal from a series of decible and time values. - SinWave
Generate a sin wave of a given length and frequency. - SincShape
Generate a Sinc shape signal with a given period and width. - Slide
Generate a sin wave which continuously in frequency varies based on a series of time and frequency values. - WhiteNoise
Generate pure white noise of a given length. - ToJSON
Convert a SFPL data type (signal, number, bunch etc) into a JSON string. - CreateBesselBandPass
Create a Bessel based band pass filter which can be passed to another processor. - CreateButterworthBandPass
Create a Butterworth based band pass filter which can be passed to another processor. - CreateButterworthHighPass
Create a Butterworth based high pass filter which can be passed to another processor. - CreateButterworthLowPass
Create a Butterworth based low pass filter which can be passed to another processor. - DirectRelength
Change the length of an audio signal by a fixed ratio via decimation and / or interpolation resulting in a new sample of different length and changing all the frequencies. - DirectResample
The same as DirectRelength except the resulting signal is cut or padded with silence to result in it being the same length as the original. - FrequencyModulate
Use an shape signal to continuously resample another signal so as to modulate its frequency. This is different from Resample in that the change in frequency is symmetric about zero for the modulating waveform. A positive value of the modulator will increase the frequency of the modulated and a negative modulator value decreases the frequency of the modulated. This method is centre frequency stable for stable modulation frequencies. The centre frequency of the modulate signal will shift if the frequency of the modulator signal shifts overall throughout the modulation period. For complex waveforms, things are more complex. - ShapedLadderFilter
A classic synthesiser low pass filter. This is loosely similar to something like the low pass on a minimoog but it is not as fruity. It acts as a low pass with variable cut off and variable resonance. - ButterwothBandPass
Use a Butterworth filter for band pass filtering up to 5 poles. - BesselBandPass
Use a Bessel filter for band pass filtering up to 5 poles. - ButterworthHighPass
Use a Butterworth filter for high pass filtering up to 11 poles. - BesselHighPass
Use a Bessel filter for high pass filterring up to 11 poles. - ButterworthLowPass
Use a Butterworth filter for low pass filtering up to 11 poles. - BesselLowPass
Use a Bessel filter for low pass filtering up to 11 poles. - RBJLowPass
Use a Robert Bristow-Johnson filter for 2 pole low pass variable Q filtering. - RBJHighPass
Use a Robert Bristow-Johnson filter for 2 pole high pass variable Q filtering. - RBJBandPass
Use a Robert Bristow-Johnson filter for 2 pole band pass filtering with variable Q. - RBJPeaking
Use a Robert Bristow-Johnson filter for 2 pole filter as a peaking resonator with variable Q and gain. - RBJBandReject
Use a Robert Bristow-Johnson filter for 2 pole notch filter with variable Q. - RBJLowShelf
Use a Robert Bristow-Johnson filter for 2 pole low shelf filter with variable Q. - RBJHighShelf
Use a Robert Bristow-Johnson filter for 2 pole high shelf filter with variable Q. - Resample
Resample a signal at a rate determined by a second signal. Unlike FrequencyModulate, the next sample rate is simply the modulation signal value. - ShapedButterworthBandPass
Use a Butterworth filter as a band pass filter up to 5 poles but make the low and high frequency shoulders controlled by two other signals allowing continuously variable filter width. - Mixer
Chooses a Java Audio mixer given a description. - LineWait
Waits for a Java Audio line to finish. - Mixers
Produces a bunch of description of the available Java Audio mixers. - Monitor
Plays a single audio signal on the default output device using Java Audio. - PlayFile
Plays the contents of a file on the default output device using Java Audio. - ReadFile
Reads a wav file into a bunch of signals. Forwards a bunch with one signal per audio channel in the file. - Semitone
Forwards a number representing the twelfth root of 2. - SpeedOfSound
Forwards a number giving the approximate speed of sound at sea level in meters per second. - StereoMonitor
Plays a stereo sound on the default audio device using Java Audio. Takes a bunch of two signals and plays them. - WriteFile16
Take a bunch of audio signals and write them out to a 16 bit wav file with one channel per element in the bunch. - WriteFile32
Take a bunch of audio signals and write them out to a 16 bit wav file with one channel per element in the bunch. - WriteFileString
Write a string to a file. - DirectResonate
- FilterResonate
- ResonantFilter
Filter via feedback resonance. Note that this is nothing at all like an IIR peaking resonator. - Resonate
- FilteredResonantFilter
Filter via feedback resonance with an IIR filter in the feedback loop (see ResonantFilter). - Reverse
Reverse the direction of an audio signal. - ShapedResonantFilter
Filter via feedback resonance where the feedback delay time is varied according to the value of a shaping signal. The shaping signal alters the delay in a simple multiplication way; e.g. a value of 2 will double the delay, a value of 0.5 will halve it (see ResonantFilter). - Frequency
Convert a period for a single cycle in milliseconds into a frequency in Hz. - Length
Forward the length of the passed audio signal in milliseconds. - Period
Convert a frequency in Hz into a period for a single cycle in milliseconds. - ValueAt
Forward the numeric value of a signal at a point in time in milliseconds. - Normalise
Remove DC from an audio signal and make set the magnitude so the maximum excursion from zero is 1. - NormaliseArea
Scale an audio signal so that the area between the signal and zero sums to 1. - RemoveDC
Remove an DC component from an audio signal. - ShapedPower
Raise all the values of a signal to the power of an equivalent length shaping signal. E.g. if the shaping signal as a value of 2 at 10 milliseconds, then the value of the forwarded audio signal at 10 milliseconds will be the square of the input signal at that point. - DoneAll
Perform Done on a bunch of Do Tasks in order and forward a bunch holding the results. - DoAll
Do a bunch of code blocks, then accumulate the Done result in a bunch in order of the code blocks in to input bunch, Forward the resulting in a bunch of the same order. - Max
Forward the greatest (closest to positive infinity) numerical value from a bunch of numbers. - Min
Forward the lowest (closest to negative infinity) numeric value from a bunch of numbers. - Prime
Forward the closest prim number which is larger than the passed number. - Random
Forward a random number between 0 and 1. - Truncate
Forward a number with any decimal part removed. E.g. 1.57 will forward to 1. - Clip
Set any value in an audio signal which is above 1 to 1 and any below -1 to -1. - DirectMix
Add a constant value to every value in an audio signal and forward the result. - Invert
Invert an audio signal about zero. - Power
Raise every value in an audio signal to a index. - Rectify
Convert all negative values in an audio signal to positive values of the same magnitude. - Volume
Scale the volume of an audio signal by a value given in db. - NumericVolume
Scale the volume of an audio signal by a value given as a linear number (2 will double etc). - FromDBs
Convert a number passed in dbs to a linear number and forward that number. - ToDBs
The oposite of FromDBs. - MaxValue
Find the largest (closest to positive infinity) value of an audio signal. - PrintLn
Send a string to standard out and append a new line. - |
Forward the modulus of two numbers. - -
Forward the different of two numbers. - /
Forward one number divided by another. - GT
Forward TRUE if the first passed number is greater than the second, otherwise forward false. - LT
Forward TRUE if the first passed number is less than the second, otherwise forward false. - EQ
Forward TRUE if the first passed number is equal to the second, otherwise forward false. - NOT
If passed TRUE forward FALSE, if passed FALSE forward TRUE. - OR
Forward the Boolean OR of the passed bunch of Booleans. - AND
Forward the Boolean AND of the passed bunch of Booleans. - XOR
Forward the Boolean XOR of the bassed bunch of Booleans. - StrCat
Concatenate all the members of the passed bunch of string and forward the result. - StrIndex
Forward the index (zero based) of a string in another string. -1 if not found. - String
Convert any operand into a string representation of its self. - StrLen
Forward the length of the passed string. - StrToUpper
Forward the a string based on the passed string where all the characters are turned to upper case using the current locale. - StrToLower
Forward the a string based on the passed string where all the characters are turned to upper case using the current locale. - StrTrim
Forward a string with all the white space at the start and end of the passed string removed. - Integer
Round a number to the nearest integer using the Java default rounding rules. - Follow
Create an envelop signal from the magnitude of the passed audio signal with settable attack and release rates. - Console
Cause patch execution to pause the move over to interactive console working. - Granulate
Split an audio signal into tapered (fade in and fade out) grains of a given length with an optionally given random variation up from that length. - AutoCorrelate
Create a signal which is the direct auto-correlation of the passed signal. - DBs-100 DBs-99 DBs-98 ... DBs+98 DBs+99 DBs-100
Change the volume of the passed audio signal by an amount in dbs and forward the result. - Pcnt-100 Pcnt-99 Pcnt-98 ... Pcnt+98 Pcnt+99 Pcnt+100
Change the volume of the passed audio signal by an amount in linear numbers as a percent and forward the result.
No comments:
Post a Comment