Nabble - Supercollider - User

[ANN] Eisenkraut v0.70

2008-07-04T13:39:08Z

finally!! after 2 1/2 years the new version 0.70 is available as a
stable download. the number of changes and improvements are two
numerous to list here, so please check out the change log file. this
update is recommended for all users. new platform specific binary-
only downloads are provided and the downloads are plain archives now
instead of the need to run a particular installer.

https://sourceforge.net/project/platformdownload.php?group_id=132039

cheers, -sciss-

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Re: no audio input with macbook 10.4.11

2008-07-04T13:04:08Z

I would suggest creating a new user account and see if the issue persists. If not, then you need to trash the configuration files in your user library.

On Jul 4, 2008, at 11:43 AM, James Harkins wrote:

Sounds like a Mac problem, not a SC problem. Not necessarily hardware - could be something's gone wonky in the OS. I have no idea what it could be, though.

What do you see in Audio MIDI Setup? That's where you configure audio drivers - garage band audio settings aren't going to help you with this.

Maybe someone can help on the Apple discussion boards?
hjh

On Jul 3, 2008, at 1:32 AM, cam2424 wrote:

Hi everyone,
im currently looking for some help, i have an intel mac with os 10.4.11,
which was working perfectly fine till few days ago, where the audio stopped
working completely, none of the applications that require audio, work
(itunes, garageband, not even skype), and when i go to system preferences on
the audio settings layout it says "no built-in audio" , i've tried a lot a
of things but none of them have worked for me even reseting the PRAM, or
resetting the audio settings with garage band, but that only leads me to a
message that says "Core Audio: Selected Driver not found. (-10202).
I would be very grateful if anybody could come up with a solution for this
or if its definitely a hardware problem which im still crossing my fingers
expecting not to be.

: H. James Harkins
: jamshark70@...
: http://www.dewdrop-world.net
.::!:.:.......:.::........:..!.::.::...:..:...:.:.:.:..:

"Come said the Muse,
Sing me a song no poet has yet chanted,
Sing me the universal." -- Whitman

Re: making audio bit by bit

2008-07-04T12:08:56Z

Just to be absolutely sure... I opened "bits.aiff" in a hex editor and the audio data are

0x202020202020202020...

And 0x2020 == 2r0010000000100000, so dividing by 32768 is right. There should be no fp rounding error because 32768 is a power of two.

hjh

On Jul 4, 2008, at 11:01 AM, James Harkins wrote:

// or, write directly to a soundfile from the language
f = SoundFile("sounds/bits.aiff")

.numChannels_(1)

.headerFormat_("AIFF")

.sampleFormat_("int16");

if(f.openWrite) {
d = FloatArray.fill(44100, 2r0010000000100000 / 32768);

f.writeData(d);

};

f.close;

: H. James Harkins

: jamshark70@...

: http://www.dewdrop-world.net

.::!:.:.......:.::........:..!.::.::...:..:...:.:.:.:..:

"Come said the Muse,

Sing me a song no poet has yet chanted,

Sing me the universal." -- Whitman

Re: making audio bit by bit

2008-07-04T11:21:05Z

I always wanted to look into doing this in bash with dd or and stdin, havn't really gotten around to it though :)

On Fri, Jul 4, 2008 at 2:28 AM, simon blackmore <simon@...> wrote:

Sorry if this has been covered before but has anyone here explored making audio bit by bit

I am not really sure where to start. Say for example I wanted to make 1 second or 44100 samples
with the following 16 bits 0010000000100000 in each sample. Is this at all possible

It seems a bit crazy, but I am sure there is someone crazy enough to have explored it here.

Cheers

Simon

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Re: making audio bit by bit

2008-07-04T09:58:01Z

Fredrik Olofsson made some Ugens that perform bitwise calculations,
which may be helpful in your explorations. You may also want to look
at his chip emulation Ugens, because they allow/encourage bitwise
changes to their parameters.

http://www.fredrikolofsson.com/pages/code-sc.html

~luke

>
> On Fri, Jul 4, 2008 at 11:28 AM, simon blackmore <simon@...>
> wrote:
>>
>> Sorry if this has been covered before but has anyone here explored making
>> audio bit by bit
>>
>> I am not really sure where to start. Say for example I wanted to make 1
>> second or 44100 samples
>> with the following 16 bits 0010000000100000 in each sample. Is this at all
>> possible
>>
>> It seems a bit crazy, but I am sure there is someone crazy enough to have
>> explored it here.
>>
>> Cheers
>>
>> Simon
>>
>>
>>
>
>

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Re: no audio input with macbook 10.4.11

2008-07-04T08:43:36Z

Sounds like a Mac problem, not a SC problem. Not necessarily hardware - could be something's gone wonky in the OS. I have no idea what it could be, though.

What do you see in Audio MIDI Setup? That's where you configure audio drivers - garage band audio settings aren't going to help you with this.

Maybe someone can help on the Apple discussion boards?

hjh

On Jul 3, 2008, at 1:32 AM, cam2424 wrote:

Hi everyone,
im currently looking for some help, i have an intel mac with os 10.4.11,
which was working perfectly fine till few days ago, where the audio stopped
working completely, none of the applications that require audio, work
(itunes, garageband, not even skype), and when i go to system preferences on
the audio settings layout it says "no built-in audio" , i've tried a lot a
of things but none of them have worked for me even reseting the PRAM, or
resetting the audio settings with garage band, but that only leads me to a
message that says "Core Audio: Selected Driver not found. (-10202).
I would be very grateful if anybody could come up with a solution for this
or if its definitely a hardware problem which im still crossing my fingers
expecting not to be.

: H. James Harkins

: jamshark70@...

: http://www.dewdrop-world.net

.::!:.:.......:.::........:..!.::.::...:..:...:.:.:.:..:

"Come said the Muse,

Sing me a song no poet has yet chanted,

Sing me the universal." -- Whitman

Re: making audio bit by bit

2008-07-04T08:05:46Z

g = Signal.fill(44100,{ arg i; your numbers here })

you can then save it to disk,

g.write(path)

load it to the server

b = Buffer.read(server,path)

and play it with a PlayBuf
of a SoundFile if its very large.

you can also open an audio file for writing and then write numbers directly to the file.

also aSignal.waveFill( function,

etc.

-cx

On Fri, Jul 4, 2008 at 11:28 AM, simon blackmore <simon@...> wrote:

Sorry if this has been covered before but has anyone here explored making audio bit by bit

I am not really sure where to start. Say for example I wanted to make 1 second or 44100 samples
with the following 16 bits 0010000000100000 in each sample. Is this at all possible

It seems a bit crazy, but I am sure there is someone crazy enough to have explored it here.

Cheers

Simon

Re: making audio bit by bit

2008-07-04T08:01:54Z

On Jul 4, 2008, at 5:28 AM, simon blackmore wrote:

Sorry if this has been covered before but has anyone here explored making audio bit by bit

I am not really sure where to start. Say for example I wanted to make 1 second or 44100 samples
with the following 16 bits 0010000000100000 in each sample. Is this at all possible

Not too hard...

The only tricky thing is that supercollider always processes sound data normalized to -1.0 .. 1.0, regardless of the bit depth of the hardware output or sound file. So if your input value is relative to 16 bits, you should divide by 32768 to get the corresponding floating-point value.

// as a server buffer

b = Buffer.alloc(s, 44100, 1, completionMessage: { |buf|

buf.fillMsg(0, 44100, 2r0010000000100000 / 32768)

});

// prove the values are OK

b.getn(0, 100, { |v| (v * 32768).postln });

// or, write directly to a soundfile from the language

f = SoundFile("sounds/bits.aiff")

.numChannels_(1)

.headerFormat_("AIFF")

.sampleFormat_("int16");

if(f.openWrite) {

d = FloatArray.fill(44100, 2r0010000000100000 / 32768);

f.writeData(d);

};

f.close;

// prove that it's good

f = SoundFile.openRead("sounds/bits.aiff");

if(f.isOpen) {

d = FloatArray.newClear(100);

f.readData(d);

(d * 32768).postln;

};

f.close;

hjh

: H. James Harkins

: jamshark70@...

: http://www.dewdrop-world.net

.::!:.:.......:.::........:..!.::.::...:..:...:.:.:.:..:

"Come said the Muse,

Sing me a song no poet has yet chanted,

Sing me the universal." -- Whitman

Re: making audio bit by bit

2008-07-04T06:31:50Z

That would result in silence

On Fri, Jul 4, 2008 at 2:28 AM, simon blackmore <simon@...> wrote:

Sorry if this has been covered before but has anyone here explored making audio bit by bit

I am not really sure where to start. Say for example I wanted to make 1 second or 44100 samples
with the following 16 bits 0010000000100000 in each sample. Is this at all possible

It seems a bit crazy, but I am sure there is someone crazy enough to have explored it here.

Cheers

Simon

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Re: Patterns (help for Pindex requested)

2008-07-04T03:16:19Z

>Hello,
>
>I have the following code [1], when I try to use Pindex I do not
>hear anything anymore... what am I doing wrong?
>My intention is to create an index to cycle through the given data
>(widhData), and possibly reuse it later on (e.g. for another data
>part)

this runs for me:
(
Pbindef(\grain,
\instrument, \irisGrain,
\i, Prand(Array.series(200, 0, 1), inf),
\freq, Pindex(q.widthData + 1 * 500 , Pkey(\i), inf),
\dur, Pwhite(0.001,0.01, inf),
\sustain, 3/Pkey(\freq),
\pan, [-1, 1]
).stop.trace.play;
)

you were not using \instrument, \irisGrain,
and with something between [0 to 1, 0 to 2] * 500,
the freqs can get pretty low, which makes sustain quite long.

Pbindef sometimes gets confused with old streams it keeps around
that you are not overwriting, so clearing them when they becme
silent helps.

or using Pdef(\grain, Pbind(...))

hth, best, a

>thanx for any help :-)
>Till
>
>s.boot
>
>(
> // a gabor (approx. gaussian-shaped) grain
>SynthDef(\irisGrain, { |out, amp=0.1, freq=440, sustain=0.01, pan|
> var snd = FSinOsc.ar(freq);
> var env = EnvGen.ar(Env.sine(sustain, amp), doneAction: 2);
> OffsetOut.ar(out, Pan2.ar(snd * env, pan));
>}, \ir ! 5).store;
>)
>(
>Pbindef(\grain,
> \instrument, \irisGrain,
> \dur, 0.1,
> \freq, 1000,
> \sustain, 1/Pkey(\freq)
>).play;
>)
>
>Pbindef(\grain, \amp, 0.5);
>
>q.widthData = {[1.0.rand, 2.0.rand]}!200;
>
>Pbindef(\grain,
> \i, Prand(Array.series(q.dataInRows.size, 0, 1), inf),
> \freq, Pindex(q.widthData*500, Pkey(\i), inf),
> \dur, Pwhite(0.001,0.01, inf),
> \sustain, 3/Pkey(\freq),
> \pan, [-1, 1]
>);
>
>
>_______________________________________________
>sc-users mailing list
>
>
>info (subscribe and unsubscribe):
>http://swiki.hfbk-hamburg.de:8888/MusicTechnology/880
>archive: http://www.listarc.bham.ac.uk/marchives/sc-users/
>search: http://www.listarc.bham.ac.uk/lists/sc-users/search/

--
--
Alberto de Campo
Bergstrasse 59/33
A-8020 Graz, Austria
e-mail : adc@...
--

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making audio bit by bit

2008-07-04T02:28:43Z

Sorry if this has been covered before but has anyone here explored
making audio bit by bit

I am not really sure where to start. Say for example I wanted to make
1 second or 44100 samples
with the following 16 bits 0010000000100000 in each sample. Is this at
all possible

It seems a bit crazy, but I am sure there is someone crazy enough to
have explored it here.

Cheers

Simon

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sc-users mailing list

info (subscribe and unsubscribe): http://swiki.hfbk-hamburg.de:8888/MusicTechnology/880
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Patterns (help for Pindex requested)

2008-07-04T01:36:17Z

Hello,

I have the following code [1], when I try to use Pindex I do not hear
anything anymore... what am I doing wrong?
My intention is to create an index to cycle through the given data
(widhData), and possibly reuse it later on (e.g. for another data part)

thanx for any help :-)
Till

s.boot

(
// a gabor (approx. gaussian-shaped) grain
SynthDef(\irisGrain, { |out, amp=0.1, freq=440, sustain=0.01, pan|
var snd = FSinOsc.ar(freq);
var env = EnvGen.ar(Env.sine(sustain, amp), doneAction: 2);
OffsetOut.ar(out, Pan2.ar(snd * env, pan));
}, \ir ! 5).store;
)
(
Pbindef(\grain,
\instrument, \irisGrain,
\dur, 0.1,
\freq, 1000,
\sustain, 1/Pkey(\freq)
).play;
)

Pbindef(\grain, \amp, 0.5);

q.widthData = {[1.0.rand, 2.0.rand]}!200;

Pbindef(\grain,
\i, Prand(Array.series(q.dataInRows.size, 0, 1), inf),
\freq, Pindex(q.widthData*500, Pkey(\i), inf),
\dur, Pwhite(0.001,0.01, inf),
\sustain, 3/Pkey(\freq),
\pan, [-1, 1]
);

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Re: NRT mode and soundfile buffers

2008-07-03T23:40:11Z

Eric- do you want to use NRT simply to render things faster that you have already heard? Or does the speed aspect of it matter? You can always just render in real-time as well (removing the need to deal with different buffers or servers).

I'm working with hundreds of MBs of sound files loaded into buffers. It takes an appreciable amount of time to load the sound files to buffers. Once they're loaded I do numerous RT runs and when I'm satisfied with the result, I write it to disk. So that's the basic idea - load lots of data once, then repeatedly alternate between RT and NRT modes. The reason to use NRT at all is that many of the scores I'm writing cannot be reliably handled in RT. They need to be rendered in NRT to avoid discontinuities in the audio. Ctk does minimize the code transition between RT and NRT which is a good thing. But every switch between RT and NRT requires reloading lots of data, which I find sub-optimal. It's not a huge disruption, maybe about a minute or so, but it's enough to impede workflow. So what I'm suggesting is to spend a lot of time tuning SC's performance on what may turn out to be a fairly marginal use of the program. I realize it's a hard sell.

What can be easily improved is the documentation, as Stefan already mentioned.

Eric

If all you want to do is render things faster, then I really do suggest you look at using Ctk. You can preview things you want to use easily by simply telling the score to play. If you like it, you change one line of code, and it is rendered in NRT. If the NRT process is going to take awhile, you can even keep working in real-time while the rendering is happening! If you are on a multi-core system, you have even more of an advantage, since each server will run on its own core (so you shouldn't see a performance lost). Finally, the Ctk syntax look much more like the rest of the language. Parameters for SynthDef args can use the = or arg_(newValue) syntax, and the system is completely compatible for use in a non-score based system for live work. If you are familiar with the SC syntax in general, it is pretty quick to learn.

Take a look at the CtkScore helpfile, I think there is a good example there that shows what I mean. The example can actually play in real-time, render and save the score all at once for you. This sounds like what you want.

Josh

On Jul 3, 2008, at 4:17 PM, Stefan Nussbaumer wrote:

Eric Lyon schrieb:

In the model I suggested, you wouldn't have to free the buffers, because NRT
would be done on the same server that generates your real-time tests. This
would be a non-trivial modification to SC, and it does not appear to have a
large constituency here, so most probably it will never happen. But if it
were implemented, it would make the work method you described more
convenient, if you want to go back and forth between real-time and NRT, and
would have no impact if you want to stay exclusively in the NRT world.

i certainly don't have the skills to judge how much your proposed change would interfere with the program-structure. i guess i can't judge your problem with the current structure either as don't know what you're working on. so far it seemed pretty convenient to me how things work now but please consider this just as my 2c - it's true, i wouldn't have found out without josh' post. the help-file on Score could be more explicit concerning the use of buffers in NRT.

stefan

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******************************************
/* Joshua D. Parmenter
http://www.realizedsound.net/josh/

"Every composer – at all times and in all cases – gives his own interpretation of how modern society is structured: whether actively or passively, consciously or unconsciously, he makes choices in this regard. He may be conservative or he may subject himself to continual renewal; or he may strive for a revolutionary, historical or social palingenesis." - Luigi Nono
*/

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Re: Using Amplitude in conditional IF

2008-07-03T21:44:47Z

Most UGens in SC output linear amplitude (between 0 and 1). This can be converted to dbs with .ampdb:

Amplitude.kr(src).ampdb;

However, 0 db represents a linear amplitude of 1.0, so your trigger values will most likely be negative db values.

Hope that helps.

Josh

On Jul 3, 2008, at 9:38 PM, Rael Bertarelli Gimenes Toffolo wrote:

hello..

I trying to make a code to a live-electronics work...

I need do something like that:

if the amplitude of Input sound became < than 40 db than change to effect 1 (like a Verb)
if the amplitude of input sound became > than 40 db than change to effect 2 (like a GrainBuf)

How Class I can use to check the Amplitude that return a value like dB?
I tried with Amplitude.kr(In.ar(.....).... but doesn't work!!

Sorry for the bad English
Thanks!!!

******************************************

/* Joshua D. Parmenter

http://www.realizedsound.net/josh/

“Every composer – at all times and in all cases – gives his own interpretation of how modern society is structured: whether actively or passively, consciously or unconsciously, he makes choices in this regard. He may be conservative or he may subject himself to continual renewal; or he may strive for a revolutionary, historical or social palingenesis." - Luigi Nono

Using Amplitude in conditional IF

2008-07-03T21:38:43Z

hello..

I trying to make a code to a live-electronics work...

I need do something like that:

if the amplitude of Input sound became < than 40 db than change to effect 1 (like a Verb)
if the amplitude of input sound became > than 40 db than change to effect 2 (like a GrainBuf)

How Class I can use to check the Amplitude that return a value like dB?
I tried with Amplitude.kr(In.ar(.....).... but doesn't work!!

Sorry for the bad English
Thanks!!!

Re: NRT mode and soundfile buffers

2008-07-03T16:33:40Z

I think one of the main difficulties in this would have to be how to
tell the server to work in real-time or NRT in a dynamic fashion. NRT
mode is based on the idea that it is passed a complete score to
render, basically allowing the NRT server to run things as quickly as
possible. But there is very little in the way of a mechanism set-up to
interrupt or pause this action. Also - would any new messages be a)
queued up b) lost? c) executed once the NRT process is done? I imagine
there would be even more problems with trying to get the same server
run in RT and NRT, and I would suggest taking a look at the server
source code itself. Right now, NRT is simply a state that is started
with a command line flag. If you really want this to change, I think
this is the first place to start (make that a parameter / state that
is controlled in RT, rather then only at startup).

Eric- do you want to use NRT simply to render things faster that you
have already heard? Or does the speed aspect of it matter? You can
always just render in real-time as well (removing the need to deal
with different buffers or servers).

If all you want to do is render things faster, then I really do
suggest you look at using Ctk. You can preview things you want to use
easily by simply telling the score to play. If you like it, you change
one line of code, and it is rendered in NRT. If the NRT process is
going to take awhile, you can even keep working in real-time while the
rendering is happening! If you are on a multi-core system, you have
even more of an advantage, since each server will run on its own core
(so you shouldn't see a performance lost). Finally, the Ctk syntax
look much more like the rest of the language. Parameters for SynthDef
args can use the = or arg_(newValue) syntax, and the system is
completely compatible for use in a non-score based system for live
work. If you are familiar with the SC syntax in general, it is pretty
quick to learn.

Take a look at the CtkScore helpfile, I think there is a good example
there that shows what I mean. The example can actually play in real-
time, render and save the score all at once for you. This sounds like
what you want.

Josh

On Jul 3, 2008, at 4:17 PM, Stefan Nussbaumer wrote:

> Eric Lyon schrieb:
>> In the model I suggested, you wouldn't have to free the buffers,
>> because NRT
>> would be done on the same server that generates your real-time
>> tests. This
>> would be a non-trivial modification to SC, and it does not appear
>> to have a
>> large constituency here, so most probably it will never happen. But
>> if it
>> were implemented, it would make the work method you described more
>> convenient, if you want to go back and forth between real-time and
>> NRT, and
>> would have no impact if you want to stay exclusively in the NRT
>> world.
>
> i certainly don't have the skills to judge how much your proposed
> change would interfere with the program-structure. i guess i can't
> judge your problem with the current structure either as don't know
> what you're working on. so far it seemed pretty convenient to me how
> things work now but please consider this just as my 2c - it's true,
> i wouldn't have found out without josh' post. the help-file on Score
> could be more explicit concerning the use of buffers in NRT.
>
> stefan
>
>
>
> _______________________________________________
> sc-users mailing list
>
>
> info (subscribe and unsubscribe): http://swiki.hfbk-hamburg.de:8888/MusicTechnology/880
> archive: http://www.listarc.bham.ac.uk/marchives/sc-users/
> search: http://www.listarc.bham.ac.uk/lists/sc-users/search/

******************************************
/* Joshua D. Parmenter
http://www.realizedsound.net/josh/

“Every composer – at all times and in all cases – gives his own
interpretation of how modern society is structured: whether actively
or passively, consciously or unconsciously, he makes choices in this
regard. He may be conservative or he may subject himself to continual
renewal; or he may strive for a revolutionary, historical or social
palingenesis." - Luigi Nono
*/

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Re: NRT mode and soundfile buffers

2008-07-03T16:17:28Z

Eric Lyon schrieb:
> In the model I suggested, you wouldn't have to free the buffers, because NRT
> would be done on the same server that generates your real-time tests. This
> would be a non-trivial modification to SC, and it does not appear to have a
> large constituency here, so most probably it will never happen. But if it
> were implemented, it would make the work method you described more
> convenient, if you want to go back and forth between real-time and NRT, and
> would have no impact if you want to stay exclusively in the NRT world.

i certainly don't have the skills to judge how much your proposed change
would interfere with the program-structure. i guess i can't judge your
problem with the current structure either as don't know what you're
working on. so far it seemed pretty convenient to me how things work now
but please consider this just as my 2c - it's true, i wouldn't have
found out without josh' post. the help-file on Score could be more
explicit concerning the use of buffers in NRT.

stefan

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Re: NRT mode and soundfile buffers

2008-07-03T15:55:46Z

nescivi schrieb:
> ...
> Why weren't you able to free the buffers?
> that seems strange to me.
> If you keep track of what you still need and what you do not need anymore,
> then it is easy to ask the server to free a buffer...
>

i can't recall exactly what happened - i guess i didn't consider the
asynchronicity of buffer-generation well enough. i wrote a couple of
emails concerning my problem to the list (well, title is misleading as i
didn't understand what was going on):
http://www.listarc.bham.ac.uk/lists/sc-users/msg01141.html

... as i said, this used to be a problem with realtime generation, not
in nrt. might become an issue again in the future.

thanks, stefan

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Re: NRT mode and soundfile buffers

2008-07-03T15:31:24Z

In the model I suggested, you wouldn't have to free the buffers, because NRT would be done on the same server that generates your real-time tests. This would be a non-trivial modification to SC, and it does not appear to have a large constituency here, so most probably it will never happen. But if it were implemented, it would make the work method you described more convenient, if you want to go back and forth between real-time and NRT, and would have no impact if you want to stay exclusively in the NRT world.

Eric

On Thu, Jul 3, 2008 at 10:48 PM, Stefan Nussbaumer <stefanus@...> wrote:

Eric Lyon schrieb:

... I suppose that the Server object could be enhanced to be able to do it's own NRT synthesis using whatever it already has loaded into memory. Then NRT operations would be faster, easier and more transparent to the user.

in a recent project of mine - a program that generates series of music-pieces - i originally worked with realtime-synthesis, which gave me a big headache as i wasn't able to free buffers like i needed it for the next generation (especially fft-buffers were really nasty, eventually crashing sc ...). switching to nrt solved those problems completely - as the the server-object, that creates my files, gets removed immediately after the job is done i had no conflicts anymore with buffers that didn't get freed properly.
sure, handling complex musical structures in nrt isn't as flexible as doing it e.g. in tasks, where i was able to let a synth react on other synths (using nodeWatchers) but finally i have been able to set up a structure that allows me to create 10, 20 or even 100 pieces in a row without crashing the program once.
i'd hope the concept stays as it is - anyway i would free a buffer i've allocated e.g. to give me a bufnum that i can use in a pattern-structure which i create my score from _before_ i render my final piece in nrt. josh' post ( http://www.nabble.com/Re%3A-RecordNRT-quesiton-p12726789.html ) demonstrates that technique quite well, i think. i would say that's pretty convenient.

stefan

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Re: NRT mode and soundfile buffers

2008-07-03T15:02:05Z

On Thursday 03 July 2008 17:48:40 Stefan Nussbaumer wrote:

> Eric Lyon schrieb:
> > ... I suppose that the Server object could be enhanced to be able to do
> > it's own NRT synthesis using whatever it already has loaded into memory.
> > Then NRT operations would be faster, easier and more transparent to the
> > user.
>
> in a recent project of mine - a program that generates series of
> music-pieces - i originally worked with realtime-synthesis, which gave
> me a big headache as i wasn't able to free buffers like i needed it for
> the next generation (especially fft-buffers were really nasty,
> eventually crashing sc ...).

Why weren't you able to free the buffers?
that seems strange to me.
If you keep track of what you still need and what you do not need anymore,
then it is easy to ask the server to free a buffer...

sincerely,
Marije

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Re: NRT mode and soundfile buffers

2008-07-03T14:48:40Z

Eric Lyon schrieb:
> ... I suppose that the Server object could be enhanced to be able to do it's own NRT synthesis using whatever it already has loaded into memory. Then NRT operations would be faster, easier and more transparent to the user.

in a recent project of mine - a program that generates series of
music-pieces - i originally worked with realtime-synthesis, which gave
me a big headache as i wasn't able to free buffers like i needed it for
the next generation (especially fft-buffers were really nasty,
eventually crashing sc ...). switching to nrt solved those problems
completely - as the the server-object, that creates my files, gets
removed immediately after the job is done i had no conflicts anymore
with buffers that didn't get freed properly.
sure, handling complex musical structures in nrt isn't as flexible as
doing it e.g. in tasks, where i was able to let a synth react on other
synths (using nodeWatchers) but finally i have been able to set up a
structure that allows me to create 10, 20 or even 100 pieces in a row
without crashing the program once.
i'd hope the concept stays as it is - anyway i would free a buffer i've
allocated e.g. to give me a bufnum that i can use in a pattern-structure
which i create my score from _before_ i render my final piece in nrt.
josh' post (
http://www.nabble.com/Re%3A-RecordNRT-quesiton-p12726789.html )
demonstrates that technique quite well, i think. i would say that's
pretty convenient.

stefan

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Re: Knob quark

2008-07-03T11:54:17Z

Ok thor,

you're right, it works perfectly and I don't get some other error messages I had as well!

It's thought a very strange behaviour.... Now everythings is tuned, thanks..

On 03 Ιουλ 2008, at 9:41 ΜΜ, thor wrote:

Hi Marinos

I'm not sure what's the problem, especially since SC is not complaining when
you compile without the XiiQuarks.new line in your startup (but ixi still installed)

So a question: what happens if you remove XiiQuarks.new from your startup
file but then run it after you have compiled?
(create an empty document, type this line there and hit Enter)

This has happened to me and I therefore have the XiiQuarks.new line at the end
in my Main:startup method.

good luck
thor

On 3 Jul 2008, at 14:03, Marinos Koutsomichalis wrote:

Well, it works if I don't use the ixi quarks.... I don't know why, I had this line in my startup

XiiQuarks.new;

with that I get this error

ERROR: Message 'default' not understood.
RECEIVER:
nil
ARGS:
CALL STACK:
DoesNotUnderstandError:reportError 170A4210
arg this = <instance of DoesNotUnderstandError>
Nil:handleError 170A4060
arg this = nil
arg error = <instance of DoesNotUnderstandError>
Thread:handleError 170A3880
arg this = <instance of Thread>
arg error = <instance of DoesNotUnderstandError>
Object:throw 1709FDB0
arg this = <instance of DoesNotUnderstandError>
Function:protect 170A0470
arg this = <instance of Function>
arg handler = <instance of Function>
var result = <instance of DoesNotUnderstandError>
Interpreter:executeFile 174CC570
arg this = <instance of Interpreter>
arg pathName = "/Users/marinosk/Library/Appl..."
arg args = [*0]
var result = nil
var saveExecutingPath = nil
SCButton:doAction 1709FC00
arg this = <instance of SCButton>
arg modifiers = 256

without that (without the line the quark is still installed) I have no problems at all....

funny, so I guess I must wait mr Thor to fix sth, and by the way there is a class there that has the same name with one of LJP lib, the scale class...

thx again

On 03 Ιουλ 2008, at 4:50 ΜΜ, Marinos Koutsomichalis wrote:

Ok, with the latst version it does work as long as I don't use a certain library, that's strange.. I need to check it..

thx anyway

On 03 Ιουλ 2008, at 4:41 ΜΜ, Andrea Valle wrote:

Same here, I've installed right now the latest dmg 3.2.something and everything is fine. With 3.2 I had problems.

Best

-a-

On 3 Jul 2008, at 15:11, Dan Stowell wrote:

Hi -

Are you using the official SC 3.2, or a more recent 'snapshot' version?

This difficulty with Knob is a "known issue" with 3.2. If you're using
3.2 or 3.2.1, you might prefer to use one of the more recent
'snapshot' builds: http://tinyurl.com/3qn7na

I *think* that the Knob issue goes away with a more recent version.
But of course, let us know if not, maybe someone can suggest something
else.

Dan

2008/7/3, Marinos Koutsomichalis <marinos@...>:

I tried in the past several times to work with the Knob quark. I remind to
everybody my problem. I use an older Knob quark with absolute no problem,
the latests quarks though fail to work because "the message default is not
understood"

I guess the problem is in that lines

GUI.skins.put('knob', (
default: (
scale: Color.black.alpha_(0.3),
center: Color.blue(0.7, 0.5),
level: Color.green(0.8, 0.8),
dial: Color.black.alpha_(0.7),
defaultMode: 'round'
)
));

Anyway I tried a lot to read the code on both quarks but the older one is
different in a lot of ways and I can't quite trace where the error is.

I hope that we will reach a solution at a time, the latest knob quark is
needed by Conductor, so I can't work with conductor as well...

Does it work for the rest on the latest release??

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--
http://www.mcld.co.uk

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--------------------------------------------------
Andrea Valle
--------------------------------------------------
CIRMA - DAMS
Università degli Studi di Torino
--> http://www.cirma.unito.it/andrea/
--> http://www.myspace.com/andreavalle
--> andrea.valle@...
--------------------------------------------------

"
Think of it as seasoning
. noise [salt] is boring
. F(blah) [food without salt] can be boring
. F(noise, blah) can be really tasty
"
(Ken Perlin on noise)