This is very informal documentation for the Smeck guitar processing patch described in the paper, "Patch for guitar" (available as HTML or PDF ) presented at the second Pd convention in Montreal, 2007.

The patch works for guitars having a separated (6-channel) pickup and audio interface so that the computer receives six channels of audio from the guitar. One such interface is the StringPort from Keith McMillen Instruments; this works fine with macintosh computer, and on Linux with a patch to the alsa driver code (write me for that). More do-it-yourself-ish people might want to build their own pre-amp, as I did before the StringPort came out. The patch has two audio outputs.


The patch runs in Pd and should work on linux, macintosh, or windows machines. You can download both Pd and Smeck from

When you download and unpack it, Smeck should appear as two directories, "smeck" and "lib". The "smeck" directory contains the actual patch, "smeck.pd" and supporting files; "lib" is a library of patches and externs used both by Smeck and by the Pd Repertory Project (PDRP), distributed separately. The main patch ("smeck.pd") expects that the "guitar" and "lib" directories should be in the same common directory as they are when first unpacked; if you move "smeck", move "lib" in parallel. (On my machine, the Pd repertory patches are aligned with these; "lib" is included in both distributions. If you have up-to-date copies of both, you can rearrange them that way on your machine too.)

To run the patch you will need to make sure it has the right channel assigments for the 6 guitar strings; this is specified in the file, "locale.txt". The one that's distributed should correspond to the StringPort. The variables, chnl-string1, etc, specify which audio channel corresponds to the six guitar strings. For example, "chnl-string1" specifies the ADC channel corresponding to string 1, and so on (Smeck numbers the strings as E=1 A=2 D=3 G=4 B=5 E=6.)

After seeing to that and plugging your guitar in, start "smeck.pd" and reach for the master volume. This is in dB; I usually set it to 80 or 90. You should then hear a "clean" mixdown of the 6 guitar strings. You can turn on the "tune" toggle to hear tuning tones that correspond to the 6 strings; while tuning up you can verify that the string assignments are right. (If they aren't the patch will still make sound but all the effects will be out of whack, since they're individualy tuned to the "natural" ranges of the strings!)

Then start recalling presets (the 70 buttons that dominate the patch). You can adjust individual parameters by opening "panel t" and mousing around, although you'll have to read the rest of this documentation to understand what they mean.

Various controls on the front panel control the global behavior of the patch. The "spread" control (0-100) controls stereo spread; if 0, both channels get the same signal; if 100, they are maximally different. The strings are each individually spatialized so that some frequency bands are panned left and others right.

The "octave" control (-200 to 100) puts the entire sound between two octaves down, and one up, from the original pitch of the guitar. Three buttons, "octsplit" through "octsplit3", arrange the octaves individually by string in useful ways, either spreading them out or compacting them together.

The "nutso" toggle fixes it so that, each time you pluck a string, its preset changes at random. This is surprisingly interesting to play through.

The "scatsinger" toggle works in tandem with preset 49, "voco". Voco uses a wavetable to make vocal sounds; if scatsinger is turned on this wavetable changes at random on each note.

That should be enough info so that you can play the patch in "music store" mode. What follows is an explanation of how the patch works, which you will need in order to develop your own sounds or extend the patch to do new stuff.


>P> NOTE: some details have changed from version 1 to version 2 of smeck that are not reflected in this documentation.

Following the principles described in the "patch for guitar" paper mentioned above, each string is processed separately according to the block diagram shown here:

The overall block diagram is at left, and the two boxes at right are details of the "MODULATION" and "FILTERS" stages. For each string, either the "phase vocoder" OR the rest of the patch is turned on (they take roughly equivalent amounts of computation time, and it's not clear anyone would want to use them in tandem anyway). The phase vocoder is used only in two presets, and uses only the four parameters at bottom right in the parameter panels (the "panel" abstractions), ignoring everything else.

When the phase vocoder is turned off, everything else is running. The "reset" values of the parameters set the OCTAVE, FILTER, and PITCHSHIFT modules to pass their inputs through unaltered, and bypasses the LOPASS and MODULATION stages using the bypass gain control, named "thru", that appears at top left in the parameter panels.

What follows is a block-by-block description of what the various stages of the patch are supposed to do.


In this stage, the string signals are low-pass filtered so that, in theory at least, the first harmonic contains most of the energy of the signal; this is so that the modulation step will act according to plan. The filter is fifth-order Butterworth, so there's a 30-dB-per-octave rolloff above the cutoff frequency. Because each string has a different range, the filters' cutoff frequencies are specified relative to the open string's fundamental, in half tones. So if the value is given as 0, each string gets a cutoff at its own open frequency; if set to 12, the cutoff is 12 half-tones higher, so each string's cutoff frequency is set to its own twelveth fret (an octave up). Usually this ranges from about 7 to 19. This parameter is named "bias".


The filtered string sound could theoretically be converted to any wavetable. Here, the four waveshapes are simply sinusoids at the first four harmonics. The first harmonic, in particular, passes the filtered string sound unchanged. The results aren't pure harmonics in reality because the incoming sould isn't sinusoidal, although the string can be filtered to get fairly close to pure harmonics if desired.


Here the phase of the filtered string is used to drive a phased pair of triangle wave generators. There are controls over the relative phase of the two waveforms (called the "duty cycle"), the relative strength of the second generator (between -100% and 100% of the first, called the "symmetry") and the slope of the rising segment of the triangles (called "partials"):


There are two formant generators. Each one makes a packet of harmonics, whose center frequency, amplitude, and bandwidth all are controlled by envelope generators (triggered by string attacks):

The spectrum of the formant (packet of harmonics) is as shown at bottom right in the diagram. The center frequency is controlled by a six-parameter envelope generator as shown top left. The output of this envelope is in half tones, like the LOPASS cutoff frequency, except that here the half tones are measured from a base frequency that depends on the "cfbias" control. If this is zero, each center frequency is indeed specified relative to the open string. If 20%, the low (fat) 'E' string's formant is specified relative to the 'A' string instead of itself, but the other 5 strings are as before. If 40%, the 'E', 'A', and 'D' are all specified relative to the 'D' string and the rest normally, and so on, until at 100% all six strings' formants are relative to the high (thin) 'E'.

The amplitude is specified in dB (100 being unit gain). The bandwidth is controlled linearly, relative to the string pitch. By the nature of this algorithm, the center frequency and bandwidth rise in parallel as the string is fretted higher up the neck of the guitar.


The fourth waveform modification technique works by lookup into wavetables that are generated using the phase bashing technique (the ICMC paper is reprinted as HTML or PDF .) A loose collection of pre-made wavetables is assigned numbers from 10 to 85, inclusive (this is the "vsamp" parameter). In particular, 30-85 is a set of 56 nonsense syllables with 7 leading consonants and 8 following vowels. The "vframes" parameter controls how far into the wavetable to precess over the life of a note, and "vspeed" controls the speed of precession. Finally, since as with formants the spectrum transposes ("chipmunks") up and down with the frequency of the string, a "vfret" parameter selects which fret should correspond to no transposition. If this is at 0, the open string is natural and the 12th fret chipmunks up 2-to-1; if it's 12, the 12th fret is normal but the open string gets a timbre that's transposed down by 2-to-1 instead.


All the above "modulation" (waveform modification) techniques' outputs are summed and the result is multiplies (amplitude modulated) by a pulse train. The pulse train is controlled by the phase of the string, but a multiplier ("pharm") allows selcting a harmonic above the fundamental frequency. The depth of puls modulation (the thinness of the pulse) is controlled statically by the "pmod" parameter, but further, is controlled by the instantaneous string amplitude multiplied by "penv"; if this parameter is nonzero the modulation depth increases with amplitude.


All the above "modulation" (waveform modification) techniques' outputs, and the bypass signal, are summed, and this sum goes through the octave-changer, a chain of filters, and a pitch shifter bank. Octave changing is done by cross-fading between four different octaves, from two down to one up. The downward octaves are made by ring modulation, and the upward one with a variable-delay comb filter. If the "octave" parameter is -200, the sound is two octaves down; if -150 the result os a 50-50 mix between two and one octaves down; if -100% it's the pure one-octave-down sound, if -50 a mix between that and the original; if 0 it's the pure original, if 50 a mix between that and the octave-up sound, and if 100 the pure octave-up sound.


The filter stage consists of three filters and a "VCA" (amplitude control unit) in series. The first filter is a two-pole resonant powpass filter whose center frequency, in halftones, is controlled by an envelope generator. As with the formant center frequency, the filter center frequency is given in half tones relative to each string, except that the "vcfbias" control lifts the nominal frequencies of the lower strings progressively toward the higher ones, until at 100% all frequencies are in half tones relative to the high (thin) 'E' string. If "q" is zero the filter has a flat frequency response; at 10-ish one gets a fairly neutral lowpass filter, and at higher values the resonant frequency becomes audible.

The next two are EQ-style "peaking" filters with frequency responses as shown:

Each filter gets a gain (in dB, positive or negative); a bandwidth, and a center frequency. The gain and center frequency are controlled by envelope generators. the center frequency is in half-tomes above the open string, subject to another bias control ("eqbias") which applies to both filters.


Pitch-shifting is done by cross-fading between variable delay lines. There are two pitch shifters with controllable gains ("amp1" and "amp2"), delays ("delay1", "delay2"), and transpositions ("trans1", "trans2"), and a bypass gain ("amp0"). In the parameter control panels, there are four buttons for calling up reasonable chorus-style settings.


There are about 100 parameters, each of which may be specified on a per-string basis. About 60 presets exist which may be called up for invididual strings or all 6 at once; the buttons on the front panel recall presets for all 6. There are 6 parameter control panels (one for each string) and on called "t" (for "tutti") which sets parameters fro all 6 strings smultaneously; this is what you'll want to use most of the time.

To save to a preset, specify the number of the preset to save to and then hit the "save" button; there's no undo or backup function provided.

To recall a preset to a specific string, send a message to "qlist-recall-string" with the string and preset numbers as arguments, e.g., "qlist-recall-string 2 40" sets the A string to preset 40.

The following is a list of all the parameters and their units:

------------------- FREQUENTLY USED UNITS -------------------

The following units have habitual ranges, unless otherwise noted in the
descriptions of the individual parameters:

DB100: decibels, with 100 as unity gain.  '0' is 'off', '100' is full blast.
Range is 0-100.

DB0: relative decibels, with 0 as unity gain.  Range is -30 - +30

HZ: frequency in Hertz.  Range is (-infinity) to infinity.  Don't try infinity

HALFSTEPS: a frequency expressed in half steps, either above the open string
(in the case of filter settings) or above the fretted string (for cf1, etc).

MSEC: time in milliseconds.  Range is 0-infinity.

MIDI: regular MIDI pitch.  Range is-127.

CENTS: musical intervals expressed in one-hundredths of a half-tone

OPENHALFTONES: half-tones above the open string.  For instance a value of
5 for the G string means middle C (since the G string is 5 halftones below
middle C.)  Range is 0-100.

FRETHALFTONES: half-tones above whatever pitch the string is played at.
Range is 0-100.

PERCENTAGE: any fraction expressed as a percentage.  Range is (-infinity) to

----------------------- PARAMETERS (UNITS) -------------------------

thru (DB100).  gain for the signal to bypass the waveshaping network.

filter bias (OPENHALFTONES): cutoff frequency of the low-pass filter that
precedes the waveshaping network.  (Doesn't affect the 'thru' signal).

Phase Modulation:

fshz (HZ) Hertz by which the sound is aliased downward or upward.

fs percent (PERCENTAGE): additional frequency by which the sound is aliased
up or down, expressed as a percentage of the measured frequency of the string.

inter1, inter2 (DB100) -- gain of imtermodulation distortion in the waveshaping
chain from the next string down and up, respectively.


h1 (DB100).  Gain of the first-harmonic waveshaping output.  If the input is a
pure sinusoid, this is the gain at which the sinusoid is output.

h2, h3, h4 (DB100).  Gains of the 2-4 harmonics of waveshaping.  If the input is
a pure sinusoid, this will be heard as gains of the 2-4 harmonics.  More
complicated signal inputs result in progressively brighter sounds.

Classic synthesizer waveforms:

tri (DB100). triangle waveshaping gain.  The next 3 parameters apply (see fig.

partials (1-100). The bandwidth, in partials, of the triangle wave, implemented
as the rising slope of the wave.

duty (PERCENTAGE, 0-100): the relative phase of a second triangle wave that is mixed in
with the primary one.

symmetry (PERCENTAGE, -100 to 100): the multiple of the second triangle wave that
is added to the first.

triangle wave: partials=1, duty=anything, symmetry=0
sawtooth: partials=100, duty=anything, symmetry=0
square wave: partials=100, duty=50, symmetry=-100

Formant generators:

A formant is a spectral bump that has a specifiable gain, center frequency, and
bandwidth (fig. 2).  The bandwitch is shared between the two formant generators
but the gains and center frequencies are specified separately.

First formant generator:

a1 (DB100) gain of the first formant when envelope generator is at rest.

a1p (DB100) peak gain of the formant

a1a (MSEC) attack time of the formant

a1d (MSEC) decay time of the formant

cf1 (HALFSTEPS) center frequency of the formant, as a multiple of fundamental
frequency, when envelope generator is at rest.

cf1p (HALFSTEPS) peak value for center frequency

cf1q (HALFSTEPS) second target value for center frequency

cf1a (MSEC) attack time for envelope generator to reach value of cf1p

cf1b (MSEC) second attack time for envelope generator to reach value of cf1q
after having reached cf1p

cf1d (MSEC) decay time to return to value cf1 after having reached cf1q.

cf1bias (PERCENT): if nonzero, raises center frequencies for lower strings so
that, instead of being measured in halfsteps above the played string, they
match the corresponding fret on a higher string.  This increases the brilliance
of the lower strings without changing that of the higer ones.  For example, if
set at 60%, the low E, A, and D strings formants are moved up to match the G
string; if 80%, they and the G all move up to match the B string.

bw (PARTIAL) bandwidth, as a multiple of fundamental
frequency, when envelope generator is at rest.

bwp (PARTIAL) peak value for bandwidth

bwa (MSEC) attack time for envelope generator to reach value of bwp

bwd (MSEC) decay time to return to value bw after having reached bwp.

a2, a2p, a2a, a2d, cf2, cf2p, cf2q, cf2a, cf2b, cf2d: corresponding values
for second formant generator; "bw" parameters are shared between the two.

clip (PERCENT, 0-100):  Acts only on the first of the two formant generators.
percentage by which the formant signal is clipped.  100 percent clipping removes
the formant entirely; intermediate values cause distortion.

clip-even (PERCENT, 0-100):  Acts only on the first of the two formant
generators.Alters clipping to act progressively less on the negative half of the
waveform and more on the positive half.  If zero, clipping tends to generate odd
harmonics; increaseing this value increases the even haronics in the mix.

Waveform grabber:

Using a set of short, pre-recorded soundfiles, this replaces the guitar waveform
with waveforms grabbed from a soundfile.

voz (DB100): gain of the waveform grabber.

vsamp (0-99): the number of the soundfile to use.  The available soundfiles are:
(10-29) twenty miscellaneous syllables of speech;  (30-85) a series of
56 nonsense syllables as follows: (bah, dah, gah, nah, wah, lah, yah), followed
by the same starting consonants but vowels replaced by "aa", "eh", "oo", "ow",
"uh", "ee", and "ing".  Usful for artificial scat singing.

vfret (FRET LOCATION): the fret location at which the timbre comes out "natural";
playing on higher frets gives chipmunk effect.

vframes (0-99): The number of 11-millisecond frames in the soundfile to sweep
over during a guitar note.  (Each note starts a new sweep from the beginning of
the soundfile.)

vspeed (1-300): relative time over which sweep takes place.  25 means original
speed; 50 means half speed; 1 means 25x normal speed.

Pulse modulation:

pmod (0-100) - depth of pulse modulation.  Units are artificial; 100 is LOTS
of modulation.

penv (0-100) - envelope sensitivity of pulse modulation depth to the string's
momentary amplitude.  Units are artificial.

pharm (1-100) - harmonic multiplier for pulse modulation.

Octave changer:

octave (PERCENT, -200 to +100) Octave dividing or raising.  -200 means two
octaves down, 0 means original pitch, +100 means one octave up.


This is a resonant, lowpass filter, as in a wahwah pedal.  The center (resonant)
frequency is controlled by an envelope generator, whose parameters are vcf,
vcfp, vcfq, vcfa, vcfb, vcfd, and vcfbias:

vcf (HALFSTEPS): value for resonant frequency, in halfsteps above the open
string, when envelope generator is at rest.

vcfp (HALFSTEPS): peak value for resonant frequency.

vcfq (HALFSTEPS) second target value for resonant frequency

vcfa (MSEC):  attack time for envelope generator to reach value of vcfa

vcfa (MSEC):  time for envelope generator to reach value of vcfb

vcfd (MSEC):  decay time to return to value vcf after having reached vcfq
after having reached vcfp

vcfbias (PERCENT):  as for cf1bias, this raises the resonant frequencies for the
lower strings to values calculated for upper ones. 

q;  the resonance of the filter.  0 is a flat response (as if there were no
filter), 10 is a gentle lowpass, 30 is a distinct wahwah, and 100 s an outright

equalizing (peaking) filters :  each one gets a boost in decibels (positive
or negative; 0 means flat response) at its own center frequency and bandwidth.
The boost gets a 2-segment emvelope generator and the center frequncy gets a
3-segment one.

eq1 (DB, -30 - +30) decibel gain (positive or negative) at the center frequency.
Gain is unity at both DC and 20K.

eq1p (DB, -30 - +30)  - peak value for decibel gain.

eq1a (MSEC) - attack time for envelope generator to reach value eq1p.

eq1d  (MSEC) - decay time for envelope generator to return to value eq1.

ef1 (MIDI) decibel gain (positive or negative) at the center frequency.
Gain is unity at both DC and 20K.

ef1p (DB, -30 - +30)  - peak value for decibel gain.

ef1q (DB, -30 - +30)  - second peak value for decibel gain.

efq1a (MSEC) - attack time for envelope generator to reach value eq1p.

ef1b (MSEC) - second attack time for envelope generator to reach value eq1q
after eq1p.

ef1d  (MSEC) - decay time for envelope generator to return to value eq1.

bw1 (MIDI) - bandwidth of the filter, in MIDI units


Next there's an envelope generator (VCA) to control gain.

amp (DB100) - gain at rest

ampp (DB100) - peak gain

ampa (MSEC) - atack time to reach peak gain

ampd (MSEC) - release time to reach rest gain


There are two pitch shifters with parameters as follows:

amp0 (DB100) bpass gain

delay0 (MSEC) delay of bypass signal

window (MSEC) - window size for the pitch shifters

amp1 (DB100) gain of the first pitch shifter

delay1 (MSEC) first delay

transp1 - (CENTS) - transposition of first pitch shifter

amp2, delay2, transp2, corresponding parameters for second pitch shifter