On Wed, 2008-01-23 at 13:22 -0500, Darren Landrum wrote: There are a couple of scripting languages designed for audio processing and synthesis, for example Csound, SuperCollider and Chuck. For more high-level algorithms you could maybe use a modular synth (Ingen, AMS etc). --ll

On 1/23/08, Darren Landrum <darren.landrum(a)sbcglobal.net> wrote: I used to use matlab (or similar/equivalent scripting languages like octave or python + scipy + matplotlib +...) but now the CLAM framework[1] is my first option. You have the best of both worlds, visual prototyping and direct coding (and realtime capable), indeed you can make very nice visual prototypes... [1] http://clam.iua.upf.edu best, -- Hernán http://h.ordia.com.ar GnuPG: 0xEE8A3FE9

Darren Landrum wrote: As long as the DSP doesn't need any tight feedback loops, Pd [1] or some such patching environment is probably the easiest way to go. [1] http://puredata.info/ For more advanced stuff, or if you need to build your own custom DSP components, use Faust [2]. It's the most advanced DSP programming language available right now. Also, it's a real DSP language which allows you to program to the bare metal, instead of just patching together some ready-made components. I guess that's what you want. [2] http://faust.grame.fr/ Faust is a purely functional language (signals are streams of samples, DSPs are functions operating on those, which can easily combined in various ways using Faust's block diagram operators). It compiles to C++, output code is very good (comparable to carefully hand-crafted code), and it interfaces nicely to different environments like Jack, LADSPA, Pd, Max/MSP, VST, SC3, to name just a few. There's also a script to generate ready-to-use Pd patches from Faust programs which makes testing pretty easy. Also, you can compile your Faust programs online on the Faust website if you don't want to bother installing the Faust compiler (which is quite easy, though, it should readily compile on any Linux box). For more information, see Yann Orlarey et al's LAC/ICMC 2006 paper and my LAC 2007 paper (the latter is specifically about the Faust-Pd interface). You can find these and a lot more on the Faust website [3]. Julius Smith from CCRMA has a tutorial and various examples [4,5], and you can find some further Faust examples like guitar effects, various synthesis algorithms and even a KCS decoder on my Q website [6]. The Faust distribution also includes a lot of examples. (There's no book on Faust yet, so right now you'll have to find your way using the examples, the quick reference guide included in the Faust distribution, and the various tutorials available.) [3] http://faust.grame.fr/pubs.php [4] http://ccrma.stanford.edu/realsimple/faust/ [5] http://ccrma.stanford.edu/realsimple/faust_strings/faust_strings.html [6] http://q-lang.sourceforge.net/examples.html#Pd Faust has a learning curve, especially if you never used a functional programming language before. But it's definitely worth it, and it's addictive. ;-) Once mastered, you can program fairly complex DSPs in a few lines, and you don't have to worry about those nasty block wrapover issues which make programming non-trivial DSPs directly in C a pita. And since the output code is just plain C++ (the compiled DSP algorithm itself is actually C, being wrapped up in a C++ class for tidyness), you can easily integrate it into your own programs once you're done prototyping. To whet your appetite, here are a few Faust examples. A simple chorus effect: chorus(dtime,freq,depth,phase,x) = x+level*fdelay(1<<16, t, x) with { t = SR*dtime/2*(1+depth*tblosc(1<<16, sin, freq, phase)); }; Or how about a generic biquad filter: filter(b0,b1,b2,a0,a1,a2) = f : (+ ~ g) with { f(x) = (b0/a0)*x+(b1/a0)*x'+(b2/a0)*x''; g(y) = 0-(a1/a0)*y-(a2/a0)*y'; }; Note that ~ is Faust's feedback loop operator; the local f function is the feedforward, g the feedback part of the filter. x' means signal x delayed by one sample. Pretty easy. And here's how you use that definition to define a low shelf filter, straight from Robert Bristow-Johnson's Audio EQ Cookbook. f0 is the shelf midpoint frequency, g the desired gain in dB. S is the shelf slope parameter, we always set that to 1 here: low_shelf(f0,g) = filter(b0,b1,b2,a0,a1,a2) with { S = 1; A = pow(10,g/40); w0 = 2*PI*f0/SR; alpha = sin(w0)/2 * sqrt( (A + 1/A)*(1/S - 1) + 2 ); b0 = A*( (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha ); b1 = 2*A*( (A-1) - (A+1)*cos(w0) ); b2 = A*( (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha ); a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha; a1 = -2*( (A-1) + (A+1)*cos(w0) ); a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha; }; You can find these examples and a lot more in my Faust guitar effects collection (see ref. [6] above). Also make sure to take a look at Julius Smith's examples, he's Da Man. :) HTH, Albert -- Dr. Albert Gr"af Dept. of Music-Informatics, University of Mainz, Germany Email: Dr.Graef(a)t-online.de, ag(a)muwiinfa.geschichte.uni-mainz.de WWW: http://www.musikinformatik.uni-mainz.de/ag