Re: [PD] VJ tools using pure data on Mac

[chris clepper]

Anything with a Quicktime vdig component will work in GEM.  The range goes
from $10 webcams to $100k HD cams.

On Thu, Apr 29, 2010 at 10:49 PM, Adityo Pratomo wrote:

> wow cool. should try it then. haven't got the time to play, too busy at the
> office.
>
> Anyway, while we're at it, can i use a USB camcorder to get image and edit
> it directly using GEM / Pd? I understand that USB web cam and Firewire based
> camera are supported, but I haven't got any information whether USB
> camcorder is also possible. It's kinda hard to get a cheap camcorder with
> firewire port these days :)
>
> Any info guys?
>
> Thanks
>
>
> On Tue, Apr 27, 2010 at 12:01 AM, Pagano, Patrick <
> p...@digitalworlds.ufl.edu> wrote:
>
>>  Looks nice, congratulations.
>>
>>
>>
>> *From:* Husk 00 [mailto:hus...@gmail.com]
>> *Sent:* Monday, April 26, 2010 12:57 PM
>> *To:* Pagano, Patrick
>> *Cc:* Ben Baker-Smith; Adityo Pratomo; pd list
>>
>> *Subject:* Re: [PD] VJ tools using pure data on Mac
>>
>>
>>
>> On Mon, Apr 26, 2010 at 5:43 PM, Pagano, Patrick <
>> p...@digitalworlds.ufl.edu> wrote:
>>
>> This looks like a version of Qeve, but qeve depends on freeframe, so I
>> assume that pix_freeframe will need to be installed for it to work
>>
>>
>> Hi Patrick and list,
>> yes, gemQ is a new Qeve version that, as the name it self can suggest,
>> it's now based on Gem library. So it use pix_freeframe for freeframe part
>> and it works pretty well with pd-extended 0.42-5.
>> The website is allmost complete and next week I'll present gemQ in Berlin
>> during this festival:
>> http://www.cannibalcaniche.com/festival/fest_berlin.htm
>>
>> So, every one of you is invited!
>>
>> husk
>>
>
>
>
> --
> mataharipertama.wordpress.com
> kotakmakan.multiply.com
>
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Re: [PD] table abstractions

[Mike Moser-Booth]

Hey Matt,

I've been on vacation the past few days, so sorry for the late reply. 
But I just wanted to say that I very much like the idea of 
table-abstractions in the vein of Csound's GEN routines. I don't think I 
could be in charge of such a project either, but I would be happy to 
contribute wherever I can.


.mmb

Matt Barber wrote:

Hello list,

For a while there had been talk about starting a vanilla-only
table-abstraction collection in the spirit of list-abs and the iem_tab
objects.  This could be useful both for manipulation of or
calculations using table data and for populating tables with desired
data (in the manner of the csound GEN routines).

I'd love to start it, but I don't know if I have enough Pd cred yet to
be in charge of it.  There are a few problems I'd like input on:

1) Naming -- would it be worth it to have different prefixes for
objects which manipulated data, e.g. [table-reverse], and objects
which populated tables, e.g. [tabwrite-blackman]?

2) Input-Output -- for the ones which manipulate data, should that
generally happen in place unless a destination table is provided --
[table-qsort foo] sorts table values in table named foo and writes
results back to foo, while [table-qsort foo bar] would sort foo's
values and write them into bar?

3) In general I'm assuming that the objects which populate tables
should simply write to a table, not internally provide the table to be
written to.  For instance, I'm thinking a [tabwrite-hann foo] which
writes a hann window into foo would be more useful than a [table-hann
foo] which internally instantiated a table called foo AND wrote a hann
window to it, and responded to messages like "resize" with a larger or
smaller hann window (I think accessing the table from without might be
too buggy for this to be worth it).

4) Interpolation -- I'm also assuming that anything which would write
a table that would usually be read by [tabread4~] or [tabosc4~] should
write the guard points for proper interpolation by default.


I've attached an example [tabwrite-chebyshev], which writes a weighted
sum of chebyshev polynomials of the first kind to a table, probably
for use with waveshaping, as an example of what I'm thinking of, and
of a few options members of such a collection might have.
(Incidentally, it might be a good idea to abbreviate this particular
one to "cheb" or "cheby" due to the many transliterations, or even
something like "chebsum" to name it like the sinesum or cosinesum
array methods.)


My thought is that many of us have already been making and using
abstractions like these, and I think it would be worth it to collect
them in a vanilla library for general use.

Matt
  



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Re: [PD] VJ tools using pure data on Mac

[Adityo Pratomo]

wow cool. should try it then. haven't got the time to play, too busy at the
office.

Anyway, while we're at it, can i use a USB camcorder to get image and edit
it directly using GEM / Pd? I understand that USB web cam and Firewire based
camera are supported, but I haven't got any information whether USB
camcorder is also possible. It's kinda hard to get a cheap camcorder with
firewire port these days :)

Any info guys?

Thanks

On Tue, Apr 27, 2010 at 12:01 AM, Pagano, Patrick  wrote:

>  Looks nice, congratulations.
>
>
>
> *From:* Husk 00 [mailto:hus...@gmail.com]
> *Sent:* Monday, April 26, 2010 12:57 PM
> *To:* Pagano, Patrick
> *Cc:* Ben Baker-Smith; Adityo Pratomo; pd list
>
> *Subject:* Re: [PD] VJ tools using pure data on Mac
>
>
>
> On Mon, Apr 26, 2010 at 5:43 PM, Pagano, Patrick <
> p...@digitalworlds.ufl.edu> wrote:
>
> This looks like a version of Qeve, but qeve depends on freeframe, so I
> assume that pix_freeframe will need to be installed for it to work
>
>
> Hi Patrick and list,
> yes, gemQ is a new Qeve version that, as the name it self can suggest, it's
> now based on Gem library. So it use pix_freeframe for freeframe part and it
> works pretty well with pd-extended 0.42-5.
> The website is allmost complete and next week I'll present gemQ in Berlin
> during this festival:
> http://www.cannibalcaniche.com/festival/fest_berlin.htm
>
> So, every one of you is invited!
>
> husk
>



-- 
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kotakmakan.multiply.com
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Re: [PD] bit crusher abstraction

[Matt Barber]

>> Reduced bit depth (which is what I think 'bit-crushing' means)  can be
>> achieved by dividing the signal by x, pass it through something like
>> [int~], multiply it by x again. An [int~] can be implemented by using
>> [wrap~] and [-~], which are both vanilla.
>
> It is also worth considering adding a DC offset. Your [wrap~] solution
> implements a floor-function, that is, rounding downwards. You can make it
> round to closest, by adding x/2 before rounding downwards.
>
> [expr int($v1)] is rounding zerowards (thus output zero corresponds to a
> twice bigger input range as any other input). Thus it will behave in a
> weird unequal way.
>
> When the volume gets low, the _effective_ bits-per-sample gets very low,
> because the relative precision of integers is proportional to amplitude
> (which is not the case with floats). Therefore, even with 16-bit audio, if
> your amplitude is 0.0001 times the max, it will feel as if it were 3-bit
> audio. In such circumstances, the differences between the possible
> roundings will become quite audible.


Bit-depth transitions are really complicated things, especially when
going from normalized float to int (and among different utitlities
which do this, e.g. libsndfile or ecasound, I've found several
different standards).  The biggest problem is what to do with the
asymmetry about zero with int representations -- there are more
negative numbers than positive in a two's complement int system.

Here are three formulas I've used for noise-shaping dither in csound
(the first one is more compatible with the bit-depth transitions that
occur when csound writes a file).

let X = the input signal and N = target bits:

1)  Y=(floor([2^(N-1)]*X)+0.5)/[2^(N-1)]
2)  Y=floor([2^(N-1)]*X+0.5)/[2^(N-1)]
3)  Y=floor([2^(N-1)]*X)/[2^(N-1)]

The only difference is whether the 0.5 is added, and whether it's
added before or after the floor function.

So let's say you're moving from floats (-1 to +1) to 8-bits:

In the first, one multiplies the signal by 2^7, getting a signal which
varies from -128 to +128 (floats).  Flooring that will result in ints
in the same range (+128 after the floor will be a relative rarity in
real signals, though, so the effective range is -128 to 127).  Adding
0.5 to all of that gives a range of -127.5 to 127.5 (with the very
rarely encountered 128.5).  Then division by 128 gives a signal which
varies between two values equally spaced from 0, but a constant 0 in
will result in a DC-offset out (it's a mid-rise function with 256
possible values).

In the second, one multiplies the signal by 2^7, getting a signal
which varies from -128 to +128 (floats).  Adding 0.5 to this will
result in a signal that varies between -127.5 to +128.5, and flooring
this gives -128 to +128 (ints).  This time, +128 will not be as rare.
This also is really just a rounding function, with x.5 values rounded
up.  It's a mid-tread function, so zeros in will result in zeros out.
At first glance it may seem as though this has one more than 2^8
values (128 negative values, 128 positive values, and 0 = 257 total
possible values).  However, if one were to graph the rounding
function, one would find that anything that started out between -127.5
and +127.5 will be rounded to -127 to +127 (255 values), while
anything lower than -127.5 will round to -128, and anything higher
than 127.5 will round to +128 -- the extremes of the staircase
function have half-quantization-step mappings, so the total effect is
still 8-bit.  This one has the advantage that both zero in and maxamp
(positive or negative) give "correct" output values as well, but the
disadvantage that one of the quantized steps is divided between the
top and the bottom of the total range.  Also, its graph is a
reflection of that of the first function about a diagonal.

In the third, one starts with the same math as the first, up to the
floor function, which gives an effective range of -128 to 127.  Then
one simply divides by the original scalar, which just gives the
mid-tread version of the function from the first formula (the first
formula minus half the quantization step), with zeros-in-zeros out but
one extra negative value than positive.


Maybe sometime later I could post some pictures for two-bit audio.
I'm sure there are plenty on the web.  There's also the question of
whether, even for a bit-crushing effect, one would add dither.


Matt

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Re: [PD] problems using pd with 120 dpi

[Thomas Holzmann]

Am 2010-04-29 17:44, schrieb Hans-Christoph Steiner:


Could you guys try pd-gui-rewrite and see if it still has this 
problem?  The Tcl/Tk side has been entirely rewritten from scratch, so 
hopefully it'll "just work" now.  Otherwise, we can get the fixes in 
that branch.





I'll try it.
How can i use pd-gui-rewrite? Do I have to compile pd differently?

Thomas

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Re: [PD] ERROR building pd-0.41.4_EXTENDED from source on OSX snow leopard

[IOhannes m zmoelnig]

On 2010-04-29 21:29, t'es in t'es bat wrote:
> Hello Hans,
> Thanks for helping,  unfortunately
> i got  errors :
> cc -g -O2 -DPD  -Wall -W -Wstrict-prototypes -Wno-unused -Wno-parentheses
> -Wno-switch -DDL_OPEN -DMACOSX -DUNISTD -I/usr/X11R6/include
> -I../portaudio/include -I../portaudio/src/common
> -I../portaudio/src/os/mac_osx/ -I../portmidi/pm_common
> -I../portmidi/pm_mac -I../portmidi/porttime
> -DUSEAPI_PORTAUDIO -DPA19 -DPA_USE_COREAUDIO -DNEWBUFFER -DMACOSX3
> -DPA_BIG_ENDIAN -Wno-error -O2 -DUSEAPI_JACK -fast -fPIC -ftree-vectorize
> -ftree-vectorizer-verbose=2 -march=core2 -msse3 -mssse3 -mfpmath=sse
> -DINSTALL_PREFIX=\"\"  -c -o ../obj/s_audio_jack.o s_audio_jack.c
> s_audio_jack.c:11:23: error: jack/jack.h: No such file or directory

maybe that's the source of the problem?

try reading the error messages :-)
then try installing jack.

fgmasdr
IOhannes

> s_audio_jack.c:18: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
> before ‘jack_out_max’
> s_audio_jack.c:20: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
> before ‘jack_filled’
> s_audio_jack.c:26: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
> before ‘*’ token
> s_audio_jack.c:27: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
> before ‘*’ token
> s_audio_jack.c:29: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
> before ‘*’ token
> s_audio_jack.c:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
> before ‘jack_mutex’
> s_audio_jack.c:35: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
> before ‘jack_sem’
> s_audio_jack.c:39: error: expected ‘)’ before ‘nframes’
> s_audio_jack.c:94: error: expected ‘)’ before ‘srate’
> s_audio_jack.c: In function ‘jack_shutdown’:
> s_audio_jack.c:105: error: ‘jack_client’ undeclared (first use in this
> function)
> s_audio_jack.c:105: error: (Each undeclared identifier is reported only once
> s_audio_jack.c:105: error: for each function it appears in.)
> s_audio_jack.c: In function ‘jack_get_clients’:
> s_audio_jack.c:121: warning: implicit declaration of function
> ‘jack_get_ports’
> s_audio_jack.c:121: error: ‘jack_client’ undeclared (first use in this
> function)
> s_audio_jack.c:121: warning: assignment makes pointer from integer without a
> cast
> s_audio_jack.c: In function ‘jack_connect_ports’:
> s_audio_jack.c:196: error: ‘jack_client’ undeclared (first use in this
> function)
> s_audio_jack.c:197: error: ‘JackPortIsOutput’ undeclared (first use in this
> function)
> s_audio_jack.c:197: warning: assignment makes pointer from integer without a
> cast
> s_audio_jack.c:200: warning: implicit declaration of function ‘jack_connect’
> s_audio_jack.c:200: warning: implicit declaration of function
> ‘jack_port_name’
> s_audio_jack.c:200: error: ‘input_port’ undeclared (first use in this
> function)
> s_audio_jack.c:201: warning: format ‘%s’ expects type ‘char *’, but argument
> 3 has type ‘int’
> s_audio_jack.c:206: error: ‘JackPortIsInput’ undeclared (first use in this
> function)
> s_audio_jack.c:206: warning: assignment makes pointer from integer without a
> cast
> s_audio_jack.c:209: error: ‘output_port’ undeclared (first use in this
> function)
> s_audio_jack.c:210: warning: format ‘%s’ expects type ‘char *’, but argument
> 2 has type ‘int’
> s_audio_jack.c: In function ‘jack_open_audio’:
> s_audio_jack.c:234: error: ‘jack_status_t’ undeclared (first use in this
> function)
> s_audio_jack.c:234: error: expected ‘;’ before ‘status’
> s_audio_jack.c:236: error: ‘jack_client_new’ undeclared (first use in this
> function)
> s_audio_jack.c:257: error: ‘jack_client’ undeclared (first use in this
> function)
> s_audio_jack.c:262: warning: implicit declaration of function
> ‘jack_client_open’
> s_audio_jack.c:262: error: ‘JackNoStartServer’ undeclared (first use in this
> function)
> s_audio_jack.c:262: error: ‘status’ undeclared (first use in this function)
> s_audio_jack.c:263: error: ‘JackServerFailed’ undeclared (first use in this
> function)
> s_audio_jack.c:268: error: ‘JackNameNotUnique’ undeclared (first use in this
> function)
> s_audio_jack.c:271: error: ‘JackServerStarted’ undeclared (first use in this
> function)
> s_audio_jack.c:289: warning: implicit declaration of function
> ‘jack_set_process_callback’
> s_audio_jack.c:289: error: ‘process’ undeclared (first use in this function)
> s_audio_jack.c:291: warning: implicit declaration of function
> ‘jack_set_error_function’
> s_audio_jack.c:301: warning: implicit declaration of function
> ‘jack_set_sample_rate_callback’
> s_audio_jack.c:301: error: ‘jack_srate’ undeclared (first use in this
> function)
> s_audio_jack.c:309: warning: implicit declaration of function
> ‘jack_on_shutdown’
> s_audio_jack.c:312: error: ‘input_port’ undeclared (first use in this
> function)
> s_audio_jack.c:313: error: ‘output_port’ undeclared (first use in this
> function)
> s_audio_jack.c:324: warning: implicit declaration of function
> ‘jack_get_sample_rate’
> s_audio_jack.c:331: wa

Re: [PD] ERROR building pd-0.41.4_EXTENDED from source on OSX snow leopard

[t'es in t'es bat]

Hello Hans,
Thanks for helping,  unfortunately
i got  errors :
cc -g -O2 -DPD  -Wall -W -Wstrict-prototypes -Wno-unused -Wno-parentheses
-Wno-switch -DDL_OPEN -DMACOSX -DUNISTD -I/usr/X11R6/include
-I../portaudio/include -I../portaudio/src/common
-I../portaudio/src/os/mac_osx/ -I../portmidi/pm_common
-I../portmidi/pm_mac -I../portmidi/porttime
-DUSEAPI_PORTAUDIO -DPA19 -DPA_USE_COREAUDIO -DNEWBUFFER -DMACOSX3
-DPA_BIG_ENDIAN -Wno-error -O2 -DUSEAPI_JACK -fast -fPIC -ftree-vectorize
-ftree-vectorizer-verbose=2 -march=core2 -msse3 -mssse3 -mfpmath=sse
-DINSTALL_PREFIX=\"\"  -c -o ../obj/s_audio_jack.o s_audio_jack.c
s_audio_jack.c:11:23: error: jack/jack.h: No such file or directory
s_audio_jack.c:18: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
before ‘jack_out_max’
s_audio_jack.c:20: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
before ‘jack_filled’
s_audio_jack.c:26: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
before ‘*’ token
s_audio_jack.c:27: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
before ‘*’ token
s_audio_jack.c:29: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
before ‘*’ token
s_audio_jack.c:34: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
before ‘jack_mutex’
s_audio_jack.c:35: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’
before ‘jack_sem’
s_audio_jack.c:39: error: expected ‘)’ before ‘nframes’
s_audio_jack.c:94: error: expected ‘)’ before ‘srate’
s_audio_jack.c: In function ‘jack_shutdown’:
s_audio_jack.c:105: error: ‘jack_client’ undeclared (first use in this
function)
s_audio_jack.c:105: error: (Each undeclared identifier is reported only once
s_audio_jack.c:105: error: for each function it appears in.)
s_audio_jack.c: In function ‘jack_get_clients’:
s_audio_jack.c:121: warning: implicit declaration of function
‘jack_get_ports’
s_audio_jack.c:121: error: ‘jack_client’ undeclared (first use in this
function)
s_audio_jack.c:121: warning: assignment makes pointer from integer without a
cast
s_audio_jack.c: In function ‘jack_connect_ports’:
s_audio_jack.c:196: error: ‘jack_client’ undeclared (first use in this
function)
s_audio_jack.c:197: error: ‘JackPortIsOutput’ undeclared (first use in this
function)
s_audio_jack.c:197: warning: assignment makes pointer from integer without a
cast
s_audio_jack.c:200: warning: implicit declaration of function ‘jack_connect’
s_audio_jack.c:200: warning: implicit declaration of function
‘jack_port_name’
s_audio_jack.c:200: error: ‘input_port’ undeclared (first use in this
function)
s_audio_jack.c:201: warning: format ‘%s’ expects type ‘char *’, but argument
3 has type ‘int’
s_audio_jack.c:206: error: ‘JackPortIsInput’ undeclared (first use in this
function)
s_audio_jack.c:206: warning: assignment makes pointer from integer without a
cast
s_audio_jack.c:209: error: ‘output_port’ undeclared (first use in this
function)
s_audio_jack.c:210: warning: format ‘%s’ expects type ‘char *’, but argument
2 has type ‘int’
s_audio_jack.c: In function ‘jack_open_audio’:
s_audio_jack.c:234: error: ‘jack_status_t’ undeclared (first use in this
function)
s_audio_jack.c:234: error: expected ‘;’ before ‘status’
s_audio_jack.c:236: error: ‘jack_client_new’ undeclared (first use in this
function)
s_audio_jack.c:257: error: ‘jack_client’ undeclared (first use in this
function)
s_audio_jack.c:262: warning: implicit declaration of function
‘jack_client_open’
s_audio_jack.c:262: error: ‘JackNoStartServer’ undeclared (first use in this
function)
s_audio_jack.c:262: error: ‘status’ undeclared (first use in this function)
s_audio_jack.c:263: error: ‘JackServerFailed’ undeclared (first use in this
function)
s_audio_jack.c:268: error: ‘JackNameNotUnique’ undeclared (first use in this
function)
s_audio_jack.c:271: error: ‘JackServerStarted’ undeclared (first use in this
function)
s_audio_jack.c:289: warning: implicit declaration of function
‘jack_set_process_callback’
s_audio_jack.c:289: error: ‘process’ undeclared (first use in this function)
s_audio_jack.c:291: warning: implicit declaration of function
‘jack_set_error_function’
s_audio_jack.c:301: warning: implicit declaration of function
‘jack_set_sample_rate_callback’
s_audio_jack.c:301: error: ‘jack_srate’ undeclared (first use in this
function)
s_audio_jack.c:309: warning: implicit declaration of function
‘jack_on_shutdown’
s_audio_jack.c:312: error: ‘input_port’ undeclared (first use in this
function)
s_audio_jack.c:313: error: ‘output_port’ undeclared (first use in this
function)
s_audio_jack.c:324: warning: implicit declaration of function
‘jack_get_sample_rate’
s_audio_jack.c:331: warning: implicit declaration of function
‘jack_port_register’
s_audio_jack.c:331: error: ‘JACK_DEFAULT_AUDIO_TYPE’ undeclared (first use
in this function)
s_audio_jack.c:331: error: ‘JackPortIsInput’ undeclared (first use in this
function)
s_audio_jack.c:341: error: ‘JackPortIsOutput’ undeclared (first use in this
function)
s_audio_jack.c:353: warning: implicit decl

[PD] Re : connecting a samphold and a phasor together

[Benoît Fortier]

Thank you, I just tried the loop~ object and it works very well for what I'm 
doing (a looping sampler which allows to change the offset and the size of the 
repeated segment without transposing it... rather simple stuff I suppose but 
fun to work on when learning!). One more question came up to my mind when 
trying the loop~ object : it is recommended under certain circonstances to add 
3 samples to an array to allow a "better playback" (or something like that) 
when building a sampler. Anyone can explain me why and when it is necessary to 
add those 3 samples?

I hope I'm not spaming this list with obvious, newbie questions! Thanks in 
advance.

Benoît





De : Miller Puckette 
À : Roman Haefeli 
Cc : Beno?t Fortier ; pd-list@iem.at
Envoyé le : Jeu 29 avril 2010, 11 h 43 min 23 s
Objet : Re: [PD] connecting a samphold and a phasor together

There's also the loop~ object (in pd/extra) which I had to write for exactly
this reason.  But I doubt it's general enough for every possible purpose :)


Miller

On Thu, Apr 29, 2010 at 04:26:13PM +0200, Roman Haefeli wrote:
> On Wed, 2010-04-28 at 08:03 -0700, Beno??t Fortier wrote:
> > First time on this mailing list, so hello everyone! I started using pd
> > a couple of month ago and I'm completely addicted.
> > 
> > Ok I have a simple problem for which I'm looking for the most elegant
> > solution.
> > 
> > I need a phasor (in a looping sampler) which frequency can be changed
> > only at the beginning of it's period (when it drops to zero). That's
> > usually the kind of problem a samphold can handle very well but when I
> > connect a phasor and a samphold directly together I get a "dsp loop
> > detected (some tilde objects not scheduled)" error. One solution could
> > be to "pipe" the new frequency to the phasor with a delay calculated
> > from the previous frequency but I have the feeling there must be a
> > more elegant solution to this problem. Anyone got a suggestion? 
> 
> In order to avoid recursiveness, you could do it by using [metro] and
> [vline~]. Then you have the feature 'update the frequency only
> loop-start' for free.
> 
> Roman
> 
> 
> 
> 
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Re: [PD] portaudio and asio windowz

[Marco Donnarumma]

Ok, now it's working everything fine, thanks for the help.
first Pd 0.42.5 -pa (asio) started recognizing all outputs, but to avoid
clicks I had to use a horrible latency i.e. 190ms... unusable.
thus I've got the "brilliant" idea of ASIO4ALL... (I know.. I could have
thought of it before).

Now I have 16ms delay in Pd and 64 samples ASIO buffer. Works like a
charm...
Asio4All fixed the issue, but I'm curious about the reason why Pd can't
handle properly ASIO by itself (well on my machine at least)?
thanks again,

M




On Thu, Apr 29, 2010 at 5:36 PM, Marco Donnarumma wrote:

> good, thanks. I'll give it a try right now.
>
> M
>
> On Thu, Apr 29, 2010 at 4:42 PM, Hans-Christoph Steiner wrote:
>
>>
>> I updated portaudio in Pd-extended 0.42.5, try a nightly build.
>>
>> .hc
>>
>> On Apr 29, 2010, at 8:21 AM, Pedro Lopes wrote:
>>
>> For example:
>>
>>
>> In Audition can you output with the ASIO driver with all channels? That's
>> a way to debug the issue... I have a MOTU Ultralite mk3 (which has no FFADO
>> support still for Linux) so I use it under Pd on Win XP, ASIO works fine and
>> I always use "MOTU ASIO" as the output then the dac object allows me to
>> select a mulitudee of channels like: [dac~ 1 2 3 4 5 6 7 8]
>>
>> Try "ASIO saffire" inside audition, create a multichannel with sine waves
>> and output each track to a different channel, check if it works that way.
>> Maybe its just a misconfigured ASIO (reinstall the saffire drivers could do
>> the trick).
>>
>> Best of luck,
>> Pedro
>>
>> On Wed, Apr 28, 2010 at 8:22 PM, Marco Donnarumma wrote:
>>
>>> hello,
>>> I just got a saffire pro 40. I'm testing it on windows atm 'cause FFADO
>>> driver works but it's still unstable on linux.
>>>
>>> In Pd output/input 1-2 work properly, but there is no way I can get the
>>> others working.
>>>
>>> I tried both MMIO (also with "multiple device"), and ASIO. No luck.
>>> Selecting ASIO I have two options:
>>>
>>> - (0)SaffireAudio: works fine only with 2 I/O selected. When I change the
>>> I/O to anything more than 2 sound stops working.
>>> - (1)ASIO Saffire: no sound at all.
>>>
>>> It looks like a problem with Portaudio/Pd..
>>> All I/O work properly with AudioMulch, Wavelab, and Audition.
>>>
>>> Any hint?
>>> thanks in advance..
>>> I have to prepare two installations for Mid May...
>>> :(
>>>
>>>
>>> p.s. Also tried to put the same number of I/O as suggested by Derek here
>>> http://www.mail-archive.com/pd-list@iem.at/msg20738.html
>>> and found this other thread
>>> http://www.mail-archive.com/pd-list@iem.at/msg27160.html but in that
>>> case the problem was not solved.
>>>
>>> --
>>> Marco Donnarumma aka TheSAD
>>> Independent New Media Arts Professional, Performer, Teacher - Edinburgh,
>>> UK
>>>
>>>
>>> PORTFOLIO: http://marcodonnarumma.com
>>> LAB: http://www.thesaddj.com | http://cntrl.sourceforge.net |
>>> http://www.flxer.net
>>> EVENT: http://www.liveperformersmeeting.net
>>>
>>> ___
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>>> UNSUBSCRIBE and account-management ->
>>> http://lists.puredata.info/listinfo/pd-list
>>>
>>>
>>
>>
>> --
>> Pedro Lopes
>> contacto: j...@radiozero.pt
>> website: http://web.ist.utl.pt/Pedro.Lopes
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>> http://lists.puredata.info/listinfo/pd-list
>>
>>
>>
>>
>>
>>
>>
>> 
>>
>> Programs should be written for people to read, and only incidentally for
>> machines to execute.
>>  - from Structure and Interpretation of Computer Programs
>>
>>
>
>
> --
> Marco Donnarumma aka TheSAD
> Independent New Media Arts Professional, Performer, Teacher - Edinburgh, UK
>
>
> PORTFOLIO: http://marcodonnarumma.com
> LAB: http://www.thesaddj.com | http://cntrl.sourceforge.net |
> http://www.flxer.net
> EVENT: http://www.liveperformersmeeting.net
>



-- 
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Independent New Media Arts Professional, Performer, Teacher - Edinburgh, UK


PORTFOLIO: http://marcodonnarumma.com
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Re: [PD] portaudio and asio windowz

[Marco Donnarumma]

good, thanks. I'll give it a try right now.

M

On Thu, Apr 29, 2010 at 4:42 PM, Hans-Christoph Steiner wrote:

>
> I updated portaudio in Pd-extended 0.42.5, try a nightly build.
>
> .hc
>
> On Apr 29, 2010, at 8:21 AM, Pedro Lopes wrote:
>
> For example:
>
>
> In Audition can you output with the ASIO driver with all channels? That's a
> way to debug the issue... I have a MOTU Ultralite mk3 (which has no FFADO
> support still for Linux) so I use it under Pd on Win XP, ASIO works fine and
> I always use "MOTU ASIO" as the output then the dac object allows me to
> select a mulitudee of channels like: [dac~ 1 2 3 4 5 6 7 8]
>
> Try "ASIO saffire" inside audition, create a multichannel with sine waves
> and output each track to a different channel, check if it works that way.
> Maybe its just a misconfigured ASIO (reinstall the saffire drivers could do
> the trick).
>
> Best of luck,
> Pedro
>
> On Wed, Apr 28, 2010 at 8:22 PM, Marco Donnarumma wrote:
>
>> hello,
>> I just got a saffire pro 40. I'm testing it on windows atm 'cause FFADO
>> driver works but it's still unstable on linux.
>>
>> In Pd output/input 1-2 work properly, but there is no way I can get the
>> others working.
>>
>> I tried both MMIO (also with "multiple device"), and ASIO. No luck.
>> Selecting ASIO I have two options:
>>
>> - (0)SaffireAudio: works fine only with 2 I/O selected. When I change the
>> I/O to anything more than 2 sound stops working.
>> - (1)ASIO Saffire: no sound at all.
>>
>> It looks like a problem with Portaudio/Pd..
>> All I/O work properly with AudioMulch, Wavelab, and Audition.
>>
>> Any hint?
>> thanks in advance..
>> I have to prepare two installations for Mid May...
>> :(
>>
>>
>> p.s. Also tried to put the same number of I/O as suggested by Derek here
>> http://www.mail-archive.com/pd-list@iem.at/msg20738.html
>> and found this other thread
>> http://www.mail-archive.com/pd-list@iem.at/msg27160.html but in that case
>> the problem was not solved.
>>
>> --
>> Marco Donnarumma aka TheSAD
>> Independent New Media Arts Professional, Performer, Teacher - Edinburgh,
>> UK
>>
>>
>> PORTFOLIO: http://marcodonnarumma.com
>> LAB: http://www.thesaddj.com | http://cntrl.sourceforge.net |
>> http://www.flxer.net
>> EVENT: http://www.liveperformersmeeting.net
>>
>> ___
>> Pd-list@iem.at mailing list
>> UNSUBSCRIBE and account-management ->
>> http://lists.puredata.info/listinfo/pd-list
>>
>>
>
>
> --
> Pedro Lopes
> contacto: j...@radiozero.pt
> website: http://web.ist.utl.pt/Pedro.Lopes
> ___
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> http://lists.puredata.info/listinfo/pd-list
>
>
>
>
>
>
>
> 
>
> Programs should be written for people to read, and only incidentally for
> machines to execute.
>  - from Structure and Interpretation of Computer Programs
>
>


-- 
Marco Donnarumma aka TheSAD
Independent New Media Arts Professional, Performer, Teacher - Edinburgh, UK


PORTFOLIO: http://marcodonnarumma.com
LAB: http://www.thesaddj.com | http://cntrl.sourceforge.net |
http://www.flxer.net
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Re: [PD] problems using pd with 120 dpi

[Hans-Christoph Steiner]

On Apr 29, 2010, at 9:39 AM, Matt Barber wrote:

I'm using Pd version 0.42.5-extended-20100428 and Ubuntu Lucid  
10.04 :-)


Am 2010-04-28 20:07, schrieb Thomas Holzmann:

Hello,

I'm using the latest pd-extended and have display problems when  
using

pd on a screen which is set to 120 dpi. Then most of the texts don't
fit into the boxes. When I change the screen to 96 dpi everything
works fine.

I assume that's a problem with tk scaling!?

Does someone know a solution for this?



On Fedora/CCRMA I use the following stupid wrapper (it's a hack).



#!/bin/bash
shopt -s -o monitor
xrdb -merge /usr/local/lib/pd_fonthelper
/usr/bin/pd $@ &
sleep 1
xrdb -merge /etc/X11/Xresources
fg 1



pd_fonthelper is just:

Xft.dpi: 72


Obviously you would change this to fit whatever dpi was working for  
you before.


I also had to comment out some lines in pd.tk which test for 8.5 and
change font size accordingly; once I did, all my Pd font problems
melted away.

Matt


Could you guys try pd-gui-rewrite and see if it still has this  
problem?  The Tcl/Tk side has been entirely rewritten from scratch, so  
hopefully it'll "just work" now.  Otherwise, we can get the fixes in  
that branch.


.hc



If nature has made any one thing less susceptible than all others of  
exclusive property, it is the action of the thinking power called an  
idea, which an individual may exclusively possess as long as he keeps  
it to himself; but the moment it is divulged, it forces itself into  
the possession of everyone, and the receiver cannot dispossess himself  
of it.- Thomas Jefferson




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Re: [PD] connecting a samphold and a phasor together

[Miller Puckette]

There's also the loop~ object (in pd/extra) which I had to write for exactly
this reason.  But I doubt it's general enough for every possible purpose :)


Miller

On Thu, Apr 29, 2010 at 04:26:13PM +0200, Roman Haefeli wrote:
> On Wed, 2010-04-28 at 08:03 -0700, Beno??t Fortier wrote:
> > First time on this mailing list, so hello everyone! I started using pd
> > a couple of month ago and I'm completely addicted.
> > 
> > Ok I have a simple problem for which I'm looking for the most elegant
> > solution.
> > 
> > I need a phasor (in a looping sampler) which frequency can be changed
> > only at the beginning of it's period (when it drops to zero). That's
> > usually the kind of problem a samphold can handle very well but when I
> > connect a phasor and a samphold directly together I get a "dsp loop
> > detected (some tilde objects not scheduled)" error. One solution could
> > be to "pipe" the new frequency to the phasor with a delay calculated
> > from the previous frequency but I have the feeling there must be a
> > more elegant solution to this problem. Anyone got a suggestion? 
> 
> In order to avoid recursiveness, you could do it by using [metro] and
> [vline~]. Then you have the feature 'update the frequency only
> loop-start' for free.
> 
> Roman
> 
> 
> 
> 
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Re: [PD] portaudio and asio windowz

[Hans-Christoph Steiner]

I updated portaudio in Pd-extended 0.42.5, try a nightly build.

.hc

On Apr 29, 2010, at 8:21 AM, Pedro Lopes wrote:


For example:


In Audition can you output with the ASIO driver with all channels?  
That's a way to debug the issue... I have a MOTU Ultralite mk3  
(which has no FFADO support still for Linux) so I use it under Pd on  
Win XP, ASIO works fine and I always use "MOTU ASIO" as the output  
then the dac object allows me to select a mulitudee of channels  
like: [dac~ 1 2 3 4 5 6 7 8]


Try "ASIO saffire" inside audition, create a multichannel with sine  
waves and output each track to a different channel, check if it  
works that way. Maybe its just a misconfigured ASIO (reinstall the  
saffire drivers could do the trick).


Best of luck,
Pedro

On Wed, Apr 28, 2010 at 8:22 PM, Marco Donnarumma  
 wrote:

hello,
I just got a saffire pro 40. I'm testing it on windows atm 'cause  
FFADO driver works but it's still unstable on linux.


In Pd output/input 1-2 work properly, but there is no way I can get  
the others working.


I tried both MMIO (also with "multiple device"), and ASIO. No luck.
Selecting ASIO I have two options:

- (0)SaffireAudio: works fine only with 2 I/O selected. When I  
change the I/O to anything more than 2 sound stops working.

- (1)ASIO Saffire: no sound at all.

It looks like a problem with Portaudio/Pd..
All I/O work properly with AudioMulch, Wavelab, and Audition.

Any hint?
thanks in advance..
I have to prepare two installations for Mid May...
:(


p.s. Also tried to put the same number of I/O as suggested by Derek  
here http://www.mail-archive.com/pd-list@iem.at/msg20738.html
and found this other thread http://www.mail-archive.com/pd-list@iem.at/msg27160.html 
 but in that case the problem was not solved.


--
Marco Donnarumma aka TheSAD
Independent New Media Arts Professional, Performer, Teacher -  
Edinburgh, UK



PORTFOLIO: http://marcodonnarumma.com
LAB: http://www.thesaddj.com | http://cntrl.sourceforge.net | 
http://www.flxer.net
EVENT: http://www.liveperformersmeeting.net

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for machines to execute.

 - from Structure and Interpretation of Computer Programs

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Re: [PD] ERROR building pd-0.41.4_EXTENDED from source on OSX snow leopard

[Hans-Christoph Steiner]

I just noticed that you are trying to build 0.41.4.  I think you'll  
have better luck with 0.42.5, so try this:

http://puredata.info/docs/developer/GettingPdSource

Specifically, something like this in the Terminal:
rsync -av --delete rsync://128.238.56.50/distros/pd-extended/ pd- 
extended/

cd pd-extended/packages/darwin_app
make install
make package

.hc


On Apr 28, 2010, at 11:52 AM, t'es in t'es bat wrote:


hello thanks hans but where do i put this command ?

2010/4/28 Hans-Christoph Steiner 

Update your source code with rsync or "svn update".  That's been  
fixed.


.hc

On Apr 28, 2010, at 6:47 AM, t'es in t'es bat wrote:


Hello,
i'll try to build from source
i follow the howto from the pd.info website.

and i got this:

new-host-4:~ lepole$ cd /Users/lepole/Desktop/Pd-0.41.4-extended/ 
packages/darwin_app

new-host-4:darwin_app lepole$ make install && make package
cat Info.plist.in | \
sed 's/PD_APP_NAME/Pd-extended/' | \
sed 's/CFBUNDLESHORTVERSIONSTRING/0.41.4/' | \
sed 's/CFBUNDLEVERSION/2010.04.28/' > \
Info.plist
install -d "/Users/lepole/Desktop/Pd-0.41.4-extended/packages/ 
darwin_app/build/Pd-extended.app/Contents/MacOS"

install -m0755 -p \
"""/System/Library/Frameworks/Tk.framework/Resources/ 
Wish.app"/Contents"/MacOS/Wish" "/Users/lepole/Desktop/Pd-0.41.4- 
extended/packages/darwin_app/build/Pd-extended.app/Contents/MacOS"
install -d "/Users/lepole/Desktop/Pd-0.41.4-extended/packages/ 
darwin_app/build/Pd-extended.app/Contents/Resources"

install -m0644 -p \
"""/System/Library/Frameworks/Tk.framework/Resources/ 
Wish.app"/Contents"/Resources/Wish.rsrc" \
"/Users/lepole/Desktop/Pd-0.41.4-extended/packages/ 
darwin_app/build/Pd-extended.app/Contents/Resources"
install: /System/Library/Frameworks/Tk.framework/Resources/Wish.app/ 
Contents/Resources/Wish.rsrc: No such file or directory

make: *** [darwin_app_wrapper] Error 71
new-host-4:darwin_app lepole$

Thanks for helping

david
--
TNTB
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7 place Favier
13210 St Remy de Provence
T/: 04 90 26 95 09
P/: 06 86 86 12 19
|
http://www.tntb.net
LOVES
LePôle travaille sur le fil des accidents, des fractures, des bugs  
informatiques, vidéos.

-]>[-]<<[>+>+<<-]>>[<<+>>-]>>>
Lepôle cherche , il cause il triture les matières, les bits, les  
glitch, les formats. Il sort des cadres.

[-]<<<+<[>>>+<<[>+>[-]<<-]>[<+>-]>[<<++>>>+<
LePôle s'indiscipline dans de multiples tentatives fragile et  
poetiques.

-]<<-<-]+>[<->-]>>+>[<[-]<<+>>>-]>[-]+<<[>+>-<<-]<<<
LePôle improvise avec ses valises pleine de machines improbables et  
généreuse en surprises visuelles...

-]<<-<-]+>[<->-]>>+>[<[-]<<+>>>-]>[-]+<<[>+>-<<-]<<<
LePôle = numerikexperimentatatonnedebitengigaemotionel
//
00
 
\\
LePôle works on the edge of faults, errors and informatic bugs and  
videos

>+[<+++>-]<[>[
-]>[-]<<[>+>+<<-]>>[<<+>>-]>>>
LePôle looks for, causes, fiddles with bytes, glitches and formats.  
Something  out of the ordinary.

[-]<<<+<[>>>+<<[>+>[-]<<-]>[<+>-]>[<<++>>>+<
LePôle is undisciplined in many endeavours fragile and poetic
-]<<-<-]+>[<->-]>>+>[<[-]<<+>>>-]>[-]+<<[>+>-<<-]<<<
LePôle improvises with its suitcases full of unlikely machines and  
visual surprises...

-]<<-<-]+>[<->-]>>+>[<[-]<<+>>>-]>[-]+<<[>+>-<<-]<<<
LePôle =digitalexperimentationofemotionalgigabytes
//





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TNTB
t'es in t'es bat
7 place Favier
13210 St Remy de Provence
T/: 04 90 26 95 09
P/: 06 86 86 12 19
|
http://www.tntb.net
LOVES
LePôle travaille sur le fil des accidents, des fractures, des bugs  
informatiques, vidéos.

-]>[-]<<[>+>+<<-]>>[<<+>>-]>>>
Lepôle cherche , il cause il triture les matières, les bits, les  
glitch, les formats. Il sort des cadres.

[-]<<<+<[>>>+<<[>+>[-]<<-]>[<+>-]>[<<++>>>+<
LePôle s'indiscipline dans de multiples tentatives fragile et  
poetiques.

-]<<-<-]+>[<->-]>>+>[<[-]<<+>>>-]>[-]+<<[>+>-<<-]<<<
LePôle improvise avec ses valises pleine de machines improbables et  
généreuse en surprises visuelles...

-]<<-<-]+>[<->-]>>+>[<[-]<<+>>>-]>[-]+<<[>+>-<<-]<<<
LePôle = numerikexperimentatatonnedebitengigaemotionel
//
00
\\\

Re: [PD] bit crusher abstraction

[Mathieu Bouchard]

On Thu, 29 Apr 2010, Roman Haefeli wrote:

Reduced bit depth (which is what I think 'bit-crushing' means)  can be 
achieved by dividing the signal by x, pass it through something like 
[int~], multiply it by x again. An [int~] can be implemented by using 
[wrap~] and [-~], which are both vanilla.


It is also worth considering adding a DC offset. Your [wrap~] solution 
implements a floor-function, that is, rounding downwards. You can make it 
round to closest, by adding x/2 before rounding downwards.


[expr int($v1)] is rounding zerowards (thus output zero corresponds to a 
twice bigger input range as any other input). Thus it will behave in a 
weird unequal way.


When the volume gets low, the _effective_ bits-per-sample gets very low, 
because the relative precision of integers is proportional to amplitude 
(which is not the case with floats). Therefore, even with 16-bit audio, if 
your amplitude is 0.0001 times the max, it will feel as if it were 3-bit 
audio. In such circumstances, the differences between the possible 
roundings will become quite audible.


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Re: [PD] FLOSS translation into French

[Mathieu Bouchard]

On Thu, 29 Apr 2010, Pierre Massat wrote:

I personnaly read the FLOSS manual first. I think it should be 
translated first, because it's well referenced in google, and it's also 
concise but quite comprehensive for beginners. This is just my opinion. 
Should we vote?


There is no electoral list.
There is no voter authentication.
Then the vote has no authority.
And there can't be any enforcement.
All you have, is people who do what they want to do.

C'est juste fait de même, puis c'est tout.

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Re: [PD] bit crusher abstraction

[Roman Haefeli]

On Thu, 2010-04-29 at 16:47 +0200, Roman Haefeli wrote:
> On Thu, 2010-04-29 at 16:15 +0200, tim vets wrote:
> > there's a very simple way do do 'sample rate reduction' (which is i
> > think the sound you're looking for)
> > [readsf~]
> > |[phasor~ 100]
> > |/ 
> > [samphold~]
> > |
> > instant lo-fi gratification :)
> 
> This is a crude way to reduce sample rate, but not quite so to reduce
> bit depth.
> 
> I say crude, because if the frequency of the phasor~ is not a dividable
> whole number of the sampling rate, it will additionally add some jitter
> (which even enhances the lofi effect).
> 
> Reduced bit depth (which is what I think 'bit-crushing' means)  can be
> achieved by dividing the signal by x, pass it through something like
> [int~], multiply it by x again. An [int~] can be implemented by using
> [wrap~] and [-~], which are both vanilla.

As mentioned in Martin Brinkmann's previous mail and assuming x is a
number greater than 1, it would make more sense to multiply first and
divide after.

Roman 



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Re: [PD] how to log complete pd sessions

[Mathieu Bouchard]

On Thu, 29 Apr 2010, Gil More wrote:

i'd like to have log-files of complete pd sessions - ideally with a time 
stamp - so that i can kind of rerun the whole process of patch creation. 
i can get pd with the debugging option to output all actions, but i 
unfortunately do not know how to store (and timestamp) all that.


You can edit the pd.tk file for that. Just use NotePad. Find «proc pd» and 
duplicate the «puts» so that it also writes to «$::gilmore». Then outside 
of a proc, add something like


  set gilmore [open {c:\documents and settings\gil\whatever\log.txt} w]

or anything like it : basically, you do an open in mode «w» and you assign 
it to the variable that the «puts» will use.


the file will be closed automatically when you close pd. (but if you have 
to force-quit, then some kilobytes may be missing at the end)


i'm on a win xp machine and tried some dos piping (>) but apparently it 
did not do the job.


I swear it works on Linux and OSX... unless you are writing to an external 
drive and pull the cable while writing ;)


i also could not get cygwin to work for me. any smart solutions like 
sending to some other app or something?


pd.tk is a very good place to customise lots of different things.

sending to another app, if you needed to do that, is almost the same as 
writing to a file. If you can't get plain redirection to work (< >
) then piping isn't any easier (|), so, you'd have to use the [socket] 

command in Tcl, which is like using the [open] command in Tcl.

(Tcl is the programming language for pd.tk).

(in pd 43, pd.tk is not the same, it was replaced by many small files, 
and many procs were renamed, so, you'd have to do things in a slightly 
different way, but it's still Tcl).


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Re: [PD] bit crusher abstraction

[Dan Wilcox]

Try these:

rc-bitcrusher~-help.pd
Description: Binary data


rc-bitcrusher~.pd
Description: Binary data


rc-aliaser~_-help.pd
Description: Binary data


rc-aliaser~_.pd
Description: Binary data
rc-bitcrusher is a modified s-bitcrusher from s-abstractions and rc-alisaser~ is from an example posted to the list awhile ago.On Apr 29, 2010, at 2:54 PM, David Schaffer wrote:Hi everybody,    I'd like to know if there's a bit crusher/ lo-fi/ sonic destructor abs. somewhere out there. I'd like to take a look inside to see how it works, just out of curiosity.Thanks D.Shttp://www.flickr.com/photos/schafferdavid/http://audioblog.arteradio.com/David_Schaffer/Hotmail: Free, trusted and rich email service. Get it now.
Dan Wilcoxdanomatika.comrobotcowboy.com

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Re: [PD] bit crusher abstraction

[Roman Haefeli]

On Thu, 2010-04-29 at 16:15 +0200, tim vets wrote:
> there's a very simple way do do 'sample rate reduction' (which is i
> think the sound you're looking for)
> [readsf~]
> |[phasor~ 100]
> |/ 
> [samphold~]
> |
> instant lo-fi gratification :)

This is a crude way to reduce sample rate, but not quite so to reduce
bit depth.

I say crude, because if the frequency of the phasor~ is not a dividable
whole number of the sampling rate, it will additionally add some jitter
(which even enhances the lofi effect).

Reduced bit depth (which is what I think 'bit-crushing' means)  can be
achieved by dividing the signal by x, pass it through something like
[int~], multiply it by x again. An [int~] can be implemented by using
[wrap~] and [-~], which are both vanilla.

Roman


 
>  
> 2010/4/29 Pierre Massat 
> Same here! Only i want to use it, too. 
> 
> Pierre
> 
> 2010/4/29 David Schaffer 
> 
> Hi everybody, 
> 
>I'd like to know if there's a bit crusher/
> lo-fi/ sonic destructor abs. somewhere out there. I'd
> like to take a look inside to see how it works, just
> out of curiosity.
> 
> Thanks 
> 
> D.S
> 
> http://www.flickr.com/photos/schafferdavid/
> 
> http://audioblog.arteradio.com/David_Schaffer/
> 
> 
> 
> 
> 
> 
> __
> Hotmail: Free, trusted and rich email service. Get it
> now.
> 
> 
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Re: [PD] FLOSS translation into French

[Derek Holzer]

I'm happy that there is so much attention on the FLOSS Manual. Pure  
Data has been long in need of documentation for human beings. However,  
it needs a lot of writing and proofing before it will be ready for  
translation. I may try to spend a little time in July working on it.  
Get in touch with me directly if you're interested in helping and I  
can let you know the state of affairs.


Best!
Derek

On 4/29/10 5:52 AM, Pierre Massat wrote:
I personnaly read the FLOSS manual first. I think it should be  
translated first, because it's well referenced in google, and it's  
also concise but quite comprehensive for beginners.

This is just my opinion. Should we vote?


--
::: derek holzer ::: http://macumbista.net :::
---Oblique Strategy # 118:
"Make what's perfect more human"



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Re: [PD] bit crusher abstraction

[martin brinkmann]

David Schaffer wrote:
>I'd like to know if there's a bit crusher/ lo-fi/ sonic destructor abs.

a bit-reduction effect is quite easy to build:
multiply the signal by a (larger) number,
throw away the fractional part (by using expr~ int($v1), or building
something with wrap~)
and divide the signal by the same number.
the higher the number, the 'smoother' (less crushed) the result.
for 'authentic bit-reduction' you can use powers of 2,
but numbers inbetween are also fine (imho often better sounding).
there is something like that in my 'instrument collection'
(quantizer.pd) http://www.martin-brinkmann.de/mnb_instruments_wip.zip

bis denn!
martin

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Re: [PD] London Patching Circle - Tuesday 4th May

[Mathieu Bouchard]

On Thu, 29 Apr 2010, Ryan Jordan wrote:

not that i know of. i've only ever been to one meeting which was January 
2009. http://puredata.info/community/DeckSpace

if anyone knows of more, and/or wants to help organise, etc then please do.
i just thought i'd try and get something going.


Ah, sounds like the club was left alone by the original organisers and you 
try continuing it ? can't you get any info from the previous organisers 
about what has been going on ?


In any case, congrats for taking the initiative of continuing a club. I 
know from past experience, that it's rare that someone does so.


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Re: [PD] connecting a samphold and a phasor together

[Roman Haefeli]

On Wed, 2010-04-28 at 08:03 -0700, Benoît Fortier wrote:
> First time on this mailing list, so hello everyone! I started using pd
> a couple of month ago and I'm completely addicted.
> 
> Ok I have a simple problem for which I'm looking for the most elegant
> solution.
> 
> I need a phasor (in a looping sampler) which frequency can be changed
> only at the beginning of it's period (when it drops to zero). That's
> usually the kind of problem a samphold can handle very well but when I
> connect a phasor and a samphold directly together I get a "dsp loop
> detected (some tilde objects not scheduled)" error. One solution could
> be to "pipe" the new frequency to the phasor with a delay calculated
> from the previous frequency but I have the feeling there must be a
> more elegant solution to this problem. Anyone got a suggestion? 

In order to avoid recursiveness, you could do it by using [metro] and
[vline~]. Then you have the feature 'update the frequency only
loop-start' for free.

Roman




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Re: [PD] Problems Following Floss "Generating Waveforms" Article

[martin brinkmann]

Alex Lucas wrote:

> there doesn't seem to be
> any visible clipping on the table. The audible clipping was present across
> the frequency range but changing the multiplication on the phasor~ to 2048
> has solved the problem.

it is not so much clipping, but a discontinuity in the waveform, which
produces the unwanted distortion. sinesum creates not exactly 1 cycle
of the waveform, but also extra samples for the tabread4-interpolation.
this is hardly visible (and audible) with big tables/low frequencys, but
easiely visible with a small table (sinesum 40 1 for example), you can
see that the waveform does not start and end at 0.

bis denn!
martin

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Re: [PD] bit crusher abstraction

[tim vets]

there's a very simple way do do 'sample rate reduction' (which is i think
the sound you're looking for)
[readsf~]
|[phasor~ 100]
|/
[samphold~]
|
instant lo-fi gratification :)
gr,
Tim


2010/4/29 Pierre Massat 

> Same here! Only i want to use it, too.
>
> Pierre
>
> 2010/4/29 David Schaffer 
>
>>   Hi everybody,
>>
>>I'd like to know if there's a bit crusher/ lo-fi/ sonic destructor
>> abs. somewhere out there. I'd like to take a look inside to see how it
>> works, just out of curiosity.
>>
>> Thanks
>>
>> D.S
>>
>>  http://www.flickr.com/photos/schafferdavid/
>>
>> http://audioblog.arteradio.com/David_Schaffer/
>>
>>
>>
>> --
>> Hotmail: Free, trusted and rich email service. Get it 
>> now.
>>
>> ___
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>
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Re: [PD] bit crusher abstraction

[Frank Barknecht]

Hi,

On Thu, Apr 29, 2010 at 02:54:18PM +0200, David Schaffer wrote:
> I'd like to know if there's a bit crusher/ lo-fi/ sonic destructor abs.
> somewhere out there. I'd like to take a look inside to see how it works, just
> out of curiosity.

The rj library has some, check out the "Effects" section in the OVERVIEW.pd
from here: http://osc.rjdj.me/~fbar/rj-100423.tgz There's e_bitchrusher,
e_alias and a nice exotic one: e_apdist.

Ciao
-- 
 Frank BarknechtDo You RjDj.me?  _ __footils.org__

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Re: [PD] Re : connecting a samphold and a phasor together

[Frank Barknecht]

On Wed, Apr 28, 2010 at 09:10:53AM -0700, Benoît Fortier wrote:
> Thank you, I didn't knew about the one block delay. Problem solved very 
> elegantly indeed!

However the problem is not really solved, only "delayed". :) Your phasor~'s
frequency will not change when it is at zero, but at least 1 sample or in Pd
even one block later (so usually 64 samples)  when the phasor~ is already at a
value different from 0. 

This affects certain algorithms more than others, but you have to be aware of
it. Here's Miller's notes from
http://crca.ucsd.edu/~msp/techniques/latest/book-html/node96.html about the
issue:

  Having the oscillator's phase control the updating of its own frequency is an
  example of feedback, which in general means using any process's output as one
  of its inputs. When processing digital audio signals at a fixed sample rate 
(as
  we're doing), it is never possible to have the process's current output as an
  input, since at the time we would need it we haven't yet calculated it. The
  best we can hope for is to use the previous sample of output--in effect, 
adding
  one sample of delay. In block environments (such as Max, Pd, and Csound) the
  situation becomes more complicated, but we will delay discussing that until 
the
  next chapter (and simply wish away the problem in the examples at the end of
  this one). 

A [phasor~] with a builtin [samphold~] on its own output could be a nice
external.

Ciao
-- 
 Frank BarknechtDo You RjDj.me?  _ __footils.org__

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Re: [PD] FLOSS translation into French

[Lorenzo]

Oli44 wrote:

Hello Pierre,

There is a FLOSS Manual about FLOSS Manuals that explains translation 
a bit:


http://en.flossmanuals.net/FLOSSManuals/XchangeIntro
Is there also a FLOSS manual about the FLOSS manual about FLOSS manuals, 
and in turn a FLOSS manual about the FLOSS manual about the FLOSS manual 
which, of course, is about FLOSS manuals ?


Sorry, couldn't resist :)
Lorenzo.


I think you will have to request the new manual to be created on the 
fm-discuss list:


http://lists.flossmanuals.net/listinfo.cgi/discuss-flossmanuals.net

However it would be good to wait until the content of this manual 
stabilizes and appears on the front page of the FLOSS Manuals site:


http://en.flossmanuals.net/

Best!
Derek




Pierre Massat wrote:

Hi all,

I sorry to bring this up again, there was a discution a while ago 
about translating the FLOSS manual in French. I m not sure how to go 
about doing this.
Can translate anything i want? Can i start with topic, or is there a 
priority rank?

And how do i submit the translated pages?
I know that i got a reply explaining all this last time, but i just 
can't find it...


Cheers!

Pierre


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Re: [PD] problems using pd with 120 dpi

[Matt Barber]

> I'm using Pd version 0.42.5-extended-20100428 and Ubuntu Lucid 10.04 :-)
>
> Am 2010-04-28 20:07, schrieb Thomas Holzmann:
>> Hello,
>>
>> I'm using the latest pd-extended and have display problems when using
>> pd on a screen which is set to 120 dpi. Then most of the texts don't
>> fit into the boxes. When I change the screen to 96 dpi everything
>> works fine.
>>
>> I assume that's a problem with tk scaling!?
>>
>> Does someone know a solution for this?
>>

On Fedora/CCRMA I use the following stupid wrapper (it's a hack).



#!/bin/bash
shopt -s -o monitor
xrdb -merge /usr/local/lib/pd_fonthelper
/usr/bin/pd $@ &
sleep 1
xrdb -merge /etc/X11/Xresources
fg 1



pd_fonthelper is just:

Xft.dpi: 72


Obviously you would change this to fit whatever dpi was working for you before.

I also had to comment out some lines in pd.tk which test for 8.5 and
change font size accordingly; once I did, all my Pd font problems
melted away.

Matt

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Re: [PD] bit crusher abstraction

[Pierre Massat]

Same here! Only i want to use it, too.

Pierre

2010/4/29 David Schaffer 

>  Hi everybody,
>
>I'd like to know if there's a bit crusher/ lo-fi/ sonic destructor
> abs. somewhere out there. I'd like to take a look inside to see how it
> works, just out of curiosity.
>
> Thanks
>
> D.S
>
> http://www.flickr.com/photos/schafferdavid/
>
> http://audioblog.arteradio.com/David_Schaffer/
>
>
>
> --
> Hotmail: Free, trusted and rich email service. Get it 
> now.
>
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Re: [PD] how to log complete pd sessions

[Vilson Vieira]

2010/4/29 Claude Heiland-Allen 

>
> Gil More wrote:
>
>> hi there,
>>
>> i'd like to have log-files of complete pd sessions - ideally with a time
>> stamp - so that i can kind of rerun the whole process of patch creation. i
>> can get pd with the debugging option to output all actions, but i
>> unfortunately do not know how to store (and timestamp) all that.
>>
>
> Pd doesn't support this directly, but maybe you could stick a proxy in
> between pd and pd-gui to capture/replay sessions.
>
> Something like peerdata (formerly known as multipd) might work:
>
> http://svn.umlaeute.mur.at/svnroot/zmoelnig/projects/peerdata/
>
>
Yes. You give me a great idea! I will save some "log" from "pd -verbose -d
2" on a text file and load it with topd [1], with a time break between every
command. So, topd will recreate the patch, command by command. I'll keep you
informed.

Cheers.

[1] http://github.com/automata/topd

-- 
Vilson Vieira

vil...@void.cc

((( http://automata.cc )))

((( http://musa.cc )))
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[PD] bit crusher abstraction

[David Schaffer]

Hi everybody, 

   I'd like to know if there's a bit crusher/ lo-fi/ sonic destructor abs. 
somewhere out there. I'd like to take a look inside to see how it works, just 
out of curiosity.

Thanks 

D.S


http://www.flickr.com/photos/schafferdavid/
http://audioblog.arteradio.com/David_Schaffer/

  
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Re: [PD] how to log complete pd sessions

[Claude Heiland-Allen]

Gil More wrote:

hi there,

i'd like to have log-files of complete pd sessions - ideally with a time stamp 
- so that i can kind of rerun the whole process of patch creation. i can get pd 
with the debugging option to output all actions, but i unfortunately do not 
know how to store (and timestamp) all that.


Pd doesn't support this directly, but maybe you could stick a proxy in 
between pd and pd-gui to capture/replay sessions.


Something like peerdata (formerly known as multipd) might work:

http://svn.umlaeute.mur.at/svnroot/zmoelnig/projects/peerdata/

(But I've just seen IOhannes' reply saying there will probably be 
problems even if you get logging working.)


Other than that, maybe use recordmydesktop to get a video file, not the 
same thing I know...



Claude


i'm on a win xp machine and tried some dos piping (>) but apparently it did not 
do the job. i also could not get cygwin to work for me. any smart solutions like 
sending to some other app or something?

thanks a bunch!
Xstal


--
http://claudiusmaximus.goto10.org

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Re: [PD] how to log complete pd sessions

[IOhannes m zmoelnig]

On 2010-04-29 14:32, Pedro Lopes wrote:
> On PD you can use the -verbose and - debug  options, for instance:
> 
> $ pd -verbose -d 2

"-d 1" means core->gui traffic
"-d 2" means gui->core traffic

"-d 3" means both

but this doesn't give you time stamps.
i experimented a bit with using tcpdump, but when replying, timing is
usually horrible (haven't found time to dig deeper into this)

the problem with all this is, that Pd uses pointer2string conversion for
naming windows, objects and whatelse on the gui side.
these names will eventually change with every re-run, so you cannot
really use whatever you log.

mfgasdr
IOhannes



smime.p7s
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Re: [PD] how to log complete pd sessions

[Pedro Lopes]

On PD you can use the -verbose and - debug  options, for instance:

$ pd -verbose -d 2
If you start pd like that you be nlown away, it outputs everything, from
loading, to each object creation and even mouse movement (scary), try o use
diferent debug levels.
If you use debug level = 1 you get different things, and so on... actually I
never found that info detailed anywhere - and I guess it makes a good
question to post here in the list..!

I'm running in Linux (ubuntu 9.10 rt), but It should be transparent, as far
as logging in Windows you can redirect the output of a program in cmd to a
file, using *>> file.txt.

It should work.
Best of luck,
Pedro
*

On Thu, Apr 29, 2010 at 12:33 PM, Gil More  wrote:

> hi there,
>
> i'd like to have log-files of complete pd sessions - ideally with a time
> stamp - so that i can kind of rerun the whole process of patch creation. i
> can get pd with the debugging option to output all actions, but i
> unfortunately do not know how to store (and timestamp) all that.
>
> i'm on a win xp machine and tried some dos piping (>) but apparently it did
> not do the job. i also could not get cygwin to work for me. any smart
> solutions like sending to some other app or something?
>
> thanks a bunch!
> Xstal
>
>
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-- 
Pedro Lopes
contacto: j...@radiozero.pt
website: http://web.ist.utl.pt/Pedro.Lopes
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Re: [PD] portaudio and asio windowz

[Pedro Lopes]

For example:


In Audition can you output with the ASIO driver with all channels? That's a
way to debug the issue... I have a MOTU Ultralite mk3 (which has no FFADO
support still for Linux) so I use it under Pd on Win XP, ASIO works fine and
I always use "MOTU ASIO" as the output then the dac object allows me to
select a mulitudee of channels like: [dac~ 1 2 3 4 5 6 7 8]

Try "ASIO saffire" inside audition, create a multichannel with sine waves
and output each track to a different channel, check if it works that way.
Maybe its just a misconfigured ASIO (reinstall the saffire drivers could do
the trick).

Best of luck,
Pedro

On Wed, Apr 28, 2010 at 8:22 PM, Marco Donnarumma wrote:

> hello,
> I just got a saffire pro 40. I'm testing it on windows atm 'cause FFADO
> driver works but it's still unstable on linux.
>
> In Pd output/input 1-2 work properly, but there is no way I can get the
> others working.
>
> I tried both MMIO (also with "multiple device"), and ASIO. No luck.
> Selecting ASIO I have two options:
>
> - (0)SaffireAudio: works fine only with 2 I/O selected. When I change the
> I/O to anything more than 2 sound stops working.
> - (1)ASIO Saffire: no sound at all.
>
> It looks like a problem with Portaudio/Pd..
> All I/O work properly with AudioMulch, Wavelab, and Audition.
>
> Any hint?
> thanks in advance..
> I have to prepare two installations for Mid May...
> :(
>
>
> p.s. Also tried to put the same number of I/O as suggested by Derek here
> http://www.mail-archive.com/pd-list@iem.at/msg20738.html
> and found this other thread
> http://www.mail-archive.com/pd-list@iem.at/msg27160.html but in that case
> the problem was not solved.
>
> --
> Marco Donnarumma aka TheSAD
> Independent New Media Arts Professional, Performer, Teacher - Edinburgh, UK
>
>
> PORTFOLIO: http://marcodonnarumma.com
> LAB: http://www.thesaddj.com | http://cntrl.sourceforge.net |
> http://www.flxer.net
> EVENT: http://www.liveperformersmeeting.net
>
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-- 
Pedro Lopes
contacto: j...@radiozero.pt
website: http://web.ist.utl.pt/Pedro.Lopes
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[PD] how to log complete pd sessions

[Gil More]

hi there,

i'd like to have log-files of complete pd sessions - ideally with a time stamp 
- so that i can kind of rerun the whole process of patch creation. i can get pd 
with the debugging option to output all actions, but i unfortunately do not 
know how to store (and timestamp) all that.

i'm on a win xp machine and tried some dos piping (>) but apparently it did not 
do the job. i also could not get cygwin to work for me. any smart solutions 
like sending to some other app or something?

thanks a bunch!
Xstal



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Re: [PD] London Patching Circle - Tuesday 4th May

[Ryan Jordan]

not that i know of.
i've only ever been to one meeting which was January 2009.
http://puredata.info/community/DeckSpace

if anyone knows of more, and/or wants to help organise, etc then please do.

i just thought i'd try and get something going.

r

http://ryanjordan.org/
http://a10lab.info/




> Date: Wed, 28 Apr 2010 18:30:05 -0400
> From: ma...@artengine.ca
> To: ryan-jor...@hotmail.co.uk
> CC: pd-list@iem.at
> Subject: Re: [PD] London Patching Circle - Tuesday 4th May
> 
> On Wed, 28 Apr 2010, Ryan Jordan wrote:
> 
> > http://puredata.info/community/LondonPatchingCircle
> > http://a10lab.info/
> 
> do you have summaries of past meetings somewhere ?
> 
>   _ _ __ ___ _  _ _ ...
> | Mathieu Bouchard, Montréal, Québec. téléphone: +1.514.383.3801
  
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Re: [PD] FLOSS translation into French

[Pierre Massat]

I personnaly read the FLOSS manual first. I think it should be translated
first, because it's well referenced in google, and it's also concise but
quite comprehensive for beginners.
This is just my opinion. Should we vote?

Pierre

2010/4/28 Mathieu Bouchard 

> On Wed, 28 Apr 2010, Pierre Massat wrote:
>
>  I sorry to bring this up again, there was a discution a while ago about
>> translating the FLOSS manual in French. I m not sure how to go about doing
>> this.
>>
>
> do you think that Pd's built-in manual should be translated after this, or
> before this ?... because, I don't know any translation of that book (the
> html pages of chapter 1 plus the patches of chapters 2 3 4).
>
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Re: [PD] metro default arg

[IOhannes m zmoelnig]

On 2010-04-29 10:01, colet.patr...@free.fr wrote:
> 
> - "IOhannes m zmoelnig"  a écrit :
> 
>> "safe" value. "1" is not very safe (it triggers messages quite
>> often),
>> so i'd say "something higher" (20, 100, 1000?) or (as you seem to
>> suggest) nothing at all.
> 
> after reflexion, 1000 or even 500 (120 BPM) sounds like good default value 
> for that kind of object.
> 

actually, i don't think the default value should be readily appliable.
else all music made with Pd will end up being 120bpm techno, which would
be a sad moment...

fg
asdmr
IOhannes



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Re: [PD] metro default arg

[colet . patrice]

- "IOhannes m zmoelnig"  a écrit :

> "safe" value. "1" is not very safe (it triggers messages quite
> often),
> so i'd say "something higher" (20, 100, 1000?) or (as you seem to
> suggest) nothing at all.

after reflexion, 1000 or even 500 (120 BPM) sounds like good default value for 
that kind of object.

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Re: [PD] gem pipe? (fwd)

[IOhannes m zmoelnig]

On 2010-04-29 01:30, Mathieu Bouchard wrote:
> 
>> My issue is that I've wound up spending quite some time putting [pipe]
>> objects between different parameters of gem chains in order for one to
>> change a set number of milliseconds after another. I basically want a
>> whole chain parameter pipe,

do you know the infamous [trigger] object?
it can do wonders!

mfgasdr#
IOhannes


PS: for dynamic manipulation of render order see: [spigot], [gemhead](!)
and others (e.g. zexy's [demultiplex])



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Re: [PD] metro default arg

[IOhannes m zmoelnig]

On 2010-04-28 22:14, Mathieu Bouchard wrote:
> 
>> i'm not opposed to raising the "default", but it should only affect
>> the arguments.
> 
> If I say [metro 0.5], I'm not using a default value.
> Furthermore, if I send 0.5 to [metro], right inlet, I'm not using a
> default value.
> 
> [metro] doesn't respect the principles that you want Patrice to respect.
>

i don't understand what you want to tell me.
your examples with 0.5 are both true, but i can't see how they are
related to what i understand to be patrice's proposal.

that metro cannot go down below 1 is - imho - a separate issue than the
default time for a [metro] without arguments.

we have briefly touched the issue of "default argument" vs "explicit
arguments" (even if they backup the default) on IRC the other day.
in the case of metro i think: [metro] (without args) should default to a
"safe" value. "1" is not very safe (it triggers messages quite often),
so i'd say "something higher" (20, 100, 1000?) or (as you seem to
suggest) nothing at all.

as for the range of valid values for metro (not the default argument),
i'd vote for going right down to "0".
i would then make a difference between [metro] and [metro 0]

fgmasdr
IOhannes










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