Veli-Pekka Tätilä wrote:
Hi list, I'm new here. I'm a 20 something sight-impaired programmer and musician from Finland. I've got experience using analogs both virtual and real and know how to build Reaktor patches as well as program in Java, Perl and C. Now I've picked up the basics of ChucK and read through the PDF manual that comes with the program. Looks good and certainly a great deal more accessible than recent versions of Reaktor if you're using magnification and a screen reader program. I've got a flurry of questions I'll ask here.
ChucK says it is: chuck version: 1.2.0.7b (dracula) exe target: microsoft win32
1. What's the range of the audio float datatype and what would be the best way to detect zero crossings? My aim is to write a simple app that counts the samples in the low and high phases of a pulse wave and prints out the pulse width whenever it changes. If I can also get MIDi input into the app it would be quite easy to determine how pulse width in percentages changes as the function of the pulse width knob in my virtual and real analogs, whose exact values are not documented.
Here's some prototypical code (this is my first real chucK script):
Code: 100::ms => dur blockSize; // Processing resolution. until(adc.last() < 0) // Measure low-phase first. blockSize => now; // SAmple counters: 0 => int positive; 0 => int negative; while(true) { adc.last() => float sample; if(sample > 0) ++positive; else if(sample < 0) { ++negative; if(positive > 0) // Measured at least one cycle. <<< "Width: ", 100 * negative / (negative + positive) >>>; 0 => positive => negative; // Reset counters. } // else if // Ignore the pure 0 value. blockSize => now; } // while End code.
However, when I run this, the app doesn't ever seem to get past the until loop. I'm assuming here that samples are floats or doubles from -1 to 1 as in VST, as I didn't find the range in the manual. Is this correct? IF not, it's no wonder the code won't work, <smile>. Of course the rather grainy test processing rate, ten times a sec, affects matters greatly but I don't ever seem to get negative sample values even in arbitrary audio. The adc and dac modules do work. IF I patch them together and record from the wave input I get a delayed copy of the input in the output I guess this is the audio equivalent of "cat".
i didn't look too closely to your code but what you seem to have no statement like: adc a => blackhole; or adc a => dac d; at the beginning. so in fact adc isn't working at all because it's not in the ugen chain. best joerg -- http://joerg.piringer.net http://www.transacoustic-research.com http://www.iftaf.org http://www.vegetableorchestra.org/