A brief summary like this might be useful although the manual isn't too bad
for this.

>Value Convertors:
>ftlen=function table length
>int=integer

int=Gives the integer portion of a real (floating point) number.

>frac=fraction

frac=Gives the fractional portion of a real number.

>dbamp=linear amplitude to decibels
>i=intialization

i=Converts from a k-rate or a-rate variable to an i-rate variable.

>abs=absolute

abs=Returns the absolute value of a number.

>exp=exponent

exp=Returns the Euler's constant e~=2.7183 raised to a power.

>log=logarithm

log=Returns the number of the power you would have to raise e to to get this
value.

>sqrt=square root
>sin=sine
>cos=cosine
>ampdb=decibels to linear amplitude
>
>
>Pitch Convertors:
>octpch=octave point decimal to octave pitch-class (8ve.pc)
>pchoct=octave pitch-class to octave point decimal
>cpspch=Cycles per second (Hz) to Octave pitch-class
>cpsoct=cycles per second to octave pitch-class
>sr= sample rate
>kr= control rate
>ksmps=control samples per second
>nchnls=no. of channels
>
>
>line=linear
>expon=exponential
>linseg=line segment
>expseg=exponential segments
>
>oscil=oscillator
>foscil=?

FM oscilator for computing frequency modulation synthesis.
asig = amp1*sin(frequency1*x + amp2*sin(frequency2*x))
Some people have suggested that this is more accurately called phase
modulation synthesis and that FM should be:
asig = amp1*sin(frequency1*x * amp2*sin(frequency2*x))


>oscili=oscilllator using interpolation
>loscil=?

Sample player with looping capability.

>
>
>buzz=?

buzz=Can be used to generate an alias free band limited impulse.  Can be
integrated to obtain alias-free sawtooth, square, pwm and triangle waves.
Any others?  I'm looking for Osc-Synch.

>gbuzz=?

I'm not sure what advantages this has over buzz.

>adsyn=additive synthesis
>pvoc=phase vocoder
>
>fof=fonctions d'ondes formantiques (French) Formant wave
>functions(English)
>
>pluck=pluck

pluck=digital waveguide synthesis pluck sound

>
>rand=random
>randh=random with holding
>randi=random with interpolation
>
>Signal Modifiers:
>
>linen=linear envelope
>linenr=?

Linear envelope with a release section like on most synthesizers.  Commonly
used with midi.

>envlpx=?
>
>port=portamento
>tone=tone,low-pass filter like knob on boombox
>reson=resonant filter

reson=a type of bandpass filter

>atone="anti-tone",the opposite of tone

high-pass filter

>areson= like above

areson=a type of band-reject/notch filter

>lpread=read linear predictive
>lpreson=?
>lpfreson=?
>
>downsamp=downsample

Used to convert from a-rate to k-rate.

>upsample=opp. of downsample

Used to convert from k-rate to arate similar to asig=ksig.

>integ=integration

something like:

asum init 0
asum =    asum + asig

>interp=interpolate
>diff=differentiation
>samphold=sample and hold