no, I mean Pd's "global" sample rate, not an audio file's rate

by "global" I mean the one that is set in Pd, and not the up/down sampling that [samplerate~] reports when you have up/downsampling at subpatches with [block~]

I think there was an external that did it... anyway, I'm now at the point of doing myself an external for this

2017-10-14 17:27 GMT-03:00 IOhannes m zmölnig zmoelnig@iem.at:

On 10/14/2017 10:10 PM, Alexandre Torres Porres wrote:

...

Hi, I wanna get the actual global samplerate and not the patch's

samplerate

...

as given by samplerate~ - there was an external that did this I think. Anybody know it? thanks

[soundfiler]?

gfasdr IOhannes

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2017-10-14 18:27 GMT-03:00 Christof Ressi christof.ressi@gmx.at:

just curious: why/where do you need it?

for an abstraction

but now I also have an external I did so I can use in this abstraction

though I already did my own external, I'm still curious and wondering about this other external I still believe pretty much exists somewhere...

2017-10-14 19:29 GMT-03:00 Alexandre Torres Porres porres@gmail.com:

more precisely: a brickwall filter (8th order lowpass butterworth) whose cutoff is at 0.9 * Nyquist, so you can do upsampling + filter for anti-aliasing.

2017-10-14 19:19 GMT-03:00 Alexandre Torres Porres porres@gmail.com:

...

2017-10-14 18:27 GMT-03:00 Christof Ressi christof.ressi@gmx.at:

...

just curious: why/where do you need it?

for an abstraction

but now I also have an external I did so I can use in this abstraction

yeah, but I obviously didn't want to do that :)

2017-10-14 20:10 GMT-03:00 Christof Ressi christof.ressi@gmx.at:

you could also just pass the upsampling rate as a creation argument or message :-)

Gesendet: Sonntag, 15. Oktober 2017 um 00:29 Uhr Von: "Alexandre Torres Porres" porres@gmail.com An: "Christof Ressi" christof.ressi@gmx.at Cc: "pd-list@lists.iem.at" pd-list@lists.iem.at Betreff: Re: Re: [PD] how to get the actual sample rate?

more precisely: a brickwall filter (8th order lowpass butterworth) whose cutoff is at 0.9 * Nyquist, so you can do upsampling + filter for anti-aliasing.

2017-10-14 19:19 GMT-03:00 Alexandre Torres Porres <porres@gmail.com [mailto:porres@gmail.com]>:

2017-10-14 18:27 GMT-03:00 Christof Ressi <christof.ressi@gmx.at[mailto: christof.ressi@gmx.at]>:just curious: why/where do you need it?

for an abstraction

but now I also have an external I did so I can use in this abstraction

sorry to be dense..how´s that supposed to work? trying to recreate the ascii patch doesn´t yield aynthing here except 0

:-(

hans

Am 15.10.2017 um 19:59 schrieb IOhannes m zmölnig zmoelnig@iem.at:

On 10/14/2017 10:51 PM, Alexandre Torres Porres wrote:

...

no, I mean Pd's "global" sample rate, not an audio file's rate

well, my answer was specifically about the "global" sample rate, not about audio files.

if you are running Pd at 44.1kHz, then the following prints out "44100", even if run in an up- or downsampled subpatch:

[write -nframes 0 . foo(
|
[soundfiler]
          [t f]
          [print]

which i think is what you asked for.

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On 2017-10-16 01:29, hans w. koch wrote:

sorry to be dense..how´s that supposed to work?

when saving a file with [soundfiler], you can specify a nominal samplerate for the file; if you don't do that, it will use the system's samplerate (the "global" rate, alex wants to query).

since Pd-0.48, it will also send the meta information of the soundfile to the right outlet (along with some other info; anyhow, the first item in the list is the samplerate, hence the [t f] object to get rid of the rest). for whatever reasons, it will send that meta info to the right outlet, even if it failed to create a soundfile, e.g. because it had no write permissions on the requested path; or because ther's nt even a table holding the data.

trying to recreate the ascii patch doesn´t yield aynthing here except 0

iirc, it requires Pd>=0.48. and btw, the ascii diagram connects the 2nd outlet of [soundfiler].

dgamksr IOhannes