June 2007 - chuck-users - lists.cs.princeton.edu

more miniAudicle things
by Atte André Jensen 10 Jun '07

10 Jun '07

Hi There are a few more things regarding miniAudicle that would be nice (I'm on linux BTW): 1) When opening a file the focus should be directed to the edit area. Now it's just at the window, which means you have to click in the text area before you can type. Almost similar: Accessing save as (or save for unsaved file) leaves focus on I-don't-know-what, whereas is should go to the file name field in the dialog. 2) The native linux cut'n'paste by select and middle-click would be appreciated. Also thinks like ctrl-k (kill line) and ctrl-y (yank from kill buffer) would be great. They might take on another form, but a copy/paste without touching the mouse or some sort would be nice. 3) I think pressing tab in the editor should always mean "indent current line to relevant intention". I looked at OnCharAdded in mAView.c, but it seems that tab presses are never showing up in event.GetKey(). I'm not really famliar with cpp/wx (I did however work a bit with wxPython) but maybe it's a instantiation flag for the window or something. Unfortunately I wasn't able to figure out how to get tab to show up in GetKey. Maybe someone here (Spencer ?) knows how? If so, I think getting tab to behave as I suggest would be possible even for me :-) -- peace, love & harmony Atte http://atte.dk | http://myspace.com/attejensen http://anagrammer.dk | http://atte.dk/compositions

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two gui wishes for miniAudicle
by Atte André Jensen 10 Jun '07

10 Jun '07

Hi (I'm back after too long) I have two minor requests for the mini: 1) Make the char position avaliable somewhere, so that when you get "line(2).char(12): syntax error" you know where to look. Even better (and not so minor) would be if were possible to click on the error in the Console Monitor and the cursor in the editor would jump to the correct location (or possibly jump to the first error automatically). 2) keyboard shortcuts for switching between tabs in the editor. -- peace, love & harmony Atte http://atte.dk | http://myspace.com/attejensen http://anagrammer.dk | http://atte.dk/compositions

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win32 woes
by James Hughes 07 Jun '07

07 Jun '07

I'm vainly trying to get chuck to work with my Via Epia CN10000EG in 5.1surround sound. It's running XP SP2. I'm able to play 6 channel audio files using mplayer. When I probe for audio devices I don't get any 6 channel capable devices: =========================================================== [chuck]: found 4 device(s) ... [chuck]: ------( chuck -- dac1 )--------------- [chuck]: device name = "Primary Sound Driver" [chuck]: probe [success] ... [chuck]: # output channels = 2 [chuck]: # input channels = 0 [chuck]: # duplex Channels = 0 [chuck]: default device = YES [chuck]: natively supported data formats: [chuck]: 8-bit int [chuck]: 16-bit int [chuck]: supported sample rates: [chuck]: 8000 Hz [chuck]: 96000 Hz [chuck]: [chuck]: ------( chuck -- dac2 )--------------- [chuck]: device name = "Vinyl AC'97 Audio (WAVE)" [chuck]: probe [success] ... [chuck]: # output channels = 2 [chuck]: # input channels = 2 [chuck]: # duplex Channels = 2 [chuck]: default device = NO [chuck]: natively supported data formats: [chuck]: 8-bit int [chuck]: 16-bit int [chuck]: supported sample rates: [chuck]: 11025 Hz [chuck]: 22050 Hz [chuck]: 44100 Hz [chuck]: [chuck]: ------( chuck -- dac3 )--------------- [chuck]: device name = "Primary Sound Capture Driver" [chuck]: probe [success] ... [chuck]: # output channels = 0 [chuck]: # input channels = 2 [chuck]: # duplex Channels = 0 [chuck]: default device = YES [chuck]: natively supported data formats: [chuck]: 8-bit int [chuck]: 16-bit int [chuck]: supported sample rates: [chuck]: 11025 Hz [chuck]: 22050 Hz [chuck]: 44100 Hz [chuck]: [chuck]: ------( chuck -- dac4 )--------------- [chuck]: device name = "Vinyl AC'97 Audio (WAVE)" [chuck]: probe [success] ... [chuck]: # output channels = 2 [chuck]: # input channels = 2 [chuck]: # duplex Channels = 2 [chuck]: default device = NO [chuck]: natively supported data formats: [chuck]: 8-bit int [chuck]: 16-bit int [chuck]: supported sample rates: [chuck]: 11025 Hz [chuck]: 22050 Hz [chuck]: 44100 Hz [chuck]: [chuck](via RtMidi): RtMidiIn::initialize: no MIDI input devices currently available. [chuck]: ------( chuck -- 0 MIDI inputs )------ [chuck]: [chuck]: ------( chuck -- 1 MIDI outputs )----- [chuck]: [0] : "Microsoft GS Wavetable SW Synth" [chuck]: =========================================================== Launching chuck doesn't fare much better. I tried using each different dac, in turn, but same results every time. =========================================================== $ chuck --dac0 -o6 chucksrc/inc.ck [chuck]: (via rtaudio): no devices found for given stream parameters: ... RtApiDs: requested channels (6) > than supported (2) by device (Primary Sound Driver). ... RtApiDs: requested channels (6) > than supported (2) by device (Vinyl AC'97 Audio (WAVE)). ... RtApiDs: requested channels (6) > than supported (0) by device (Primary Sound Capture Driver). ... RtApiDs: requested channels (6) > than supported (2) by device (Vinyl AC'97 Audio (WAVE)). [chuck]: cannot initialize audio device (try using --silent/-s) =========================================================== Does anyone have any insight? Thanks, James

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report on my first livecoding gig
by Kassen 05 Jun '07

05 Jun '07

Well, that went well. It went well because there were lots of nice people, the competitors were friendly amongst eachother as well. It went well because nobody had seen ChucK and the mini in action before and now people were interested. In the last place it went well because I won, that's still nice, even if competition is less fun then duets, I think. I was up against two MAX-MSP users who were very talented. They should be because they won previous rounds, this was sort of the finale of the comptition but due to the format that was about three people I was thrown in as a surprise challenger. It was a bit sad for my second oponent who was clearly handicaped by having had to borrow a laptop and was thrown off by the different cursor behaviour, that's probably a huge factor in a graphical system like MAX. I was using the Mini with a external keyboard. The audience was just used to MAX which had been used by all (I think) contenders in all previous rounds and at first they gigled at my batered old keyboard and non-pro typing speed but with chucK a few lines go a long way. One fun moment was showing off "if(maybe)....." which -as I expected went over very well as it realy is quite funny. Because of the 6 minute format and my typing speed I tried to go for eficiency so I didn't dare to spend time on something like "while(float rubber_ducky) {//loop goes here}" which would've been even more funny (try it, it works). For the first round I used the Shakers as something like "Std.rand2(0,22) => shakers.preset;" seemed like the most economical way of quickly generating a nice rithmic yet changing foundation. Then I added a algorithmic bassline based partially on modulating the amount of harmonics in Blit instead of a filter. At one point, due to a misplaced "greater then" which should've been smaller sign my algorithem spun out of controll towards the high end but I was able to remidy this in time and it turned into a pleasant surprise that sounded good This round was a very narrow win, I thought my oponent Luc performed a very strong set of paterned noise and the audience agreed, aplauding at length for both of us, creating a need for a tie-breaking hand-vote-count. I the second round I improvised based on sending Impulse into ResonZ and then treating the result by sampling the dac using LiSa. This didn't go completely perfect but I think the audience picked up that I was considdering all my parameters on the fly and apreceated this. Overall it was a great experience and a huge thrill. I'd definately recomend it to those on the fense. Clearly you'd need to practice but that's only good for your ChucKing skills. I look forward to "defending my title" next season already. Yours, Kas.

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Code that DOES modulate filter cutoff with ADSR
by Loscha 03 Jun '07

03 Jun '07

My answer is kind of a hack, but, it's working quite well. Use a Step waveform set to 1 at all times, have THAT 1 divided by the ADSR for the filter, then, my above code works like a charm! You could also, for example, spork off a different function that randomly triggered the filter's ADSR independently from the amplitude's ADSR, giving you all sorts of trance techno type patterns. Hope this helps you out, Josh, and other users. -Edward // Filter Wooby Wooby Woo BlitSaw bs; LPF lpf; ADSR amp; ADSR filt; Gain fg; amp.set( 1::ms, 350::ms, .2, 500::ms ); filt.set( 15::ms, 550::ms, 0, 400::ms ); Step s; s => filt; // 1 => s.value; 1 => s.next; bs => amp => lpf => dac; filt => fg => blackhole; 10 => fg.gain; 500 => lpf.freq; 2 => lpf.Q; spork ~ filteradsr (); for (int l; l < 32; l++) { <<<"L;",l,"">>>; spork ~ playanote ( Std.rand2(30,60) ); 800::ms => now; } fun void playanote (int nnum) { int mstodie; <<<"\n\n",nnum,"">>>; nnum => Std.mtof => bs.freq; <<<"On","">>>; amp.keyOn(); filt.keyOn(); Std.rand2(150,300) => mstodie; mstodie * 1::ms => now; <<<"Off","">>>; amp.keyOff(); filt.keyOff(); } fun void filteradsr () { while (true) { float currfilt; filt.last() * 800 + 250 => currfilt; currfilt => lpf.freq; //if (maybe) <<<filt.last(),currfilt>>>; 1::ms => now; } }

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modulate LPF with ADSR
by Josh Lawrence 03 Jun '07

03 Jun '07

well, the trek continues... I've been copying and pasting, playing with, and generally screwing up chuck code over the past week. with some help, I've come up with this simple synth patch: // declare and connect chain of ugens Gain g => dac; 1 => g.gain; SawOsc s => ADSR e =>LPF f => g; e.set( 1::ms, 5::ms, .8, 500::ms ); e.keyOn(); e.keyOff(); f.freq( 800 ); now what I would like to do with modulate the filter with adsr. I think this approach should work: SawOsc s => ADSR e =>LPF f => g; ADSR e2 => f => g; my thinking goes like this: I need to declare e2 to differientiate it from my first adsr, which is defined as e. I have chucked e2 to my existing low pass filter, f. yet I'm stumped as to where to define how the adsr opens up the filter and closes it back down. can someone help me understand where to go from here? -- Josh Lawrence http://www.hardbop200.com http://www.joshlawrencetrio.com

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