umm... last week i managed to install pd really quickly and easily, but couldn't get my soundcard to work.
eventually i reformatted my hard drive and started again, with ubuntu breezy.
now i have a workign soundcard, but i just checked out pd from cvs, and i can't run ./configure
i get this message:
mpi@bass:~/pd/src$ ./configure bash: ./configure: No such file or directory
what should i do???
Hallo, hard off hat gesagt: // hard off wrote:
YES!!!
pd makes sound!!! a distorted sine wave has never sounded so sweet!
Congratulations!
If you're running your Tascam still: USB soundcards don't like uneven samplerates. Try using 48 kHz instead of 44.1 kHz.
Ciao
Congratulations!
thank you. felt pretty good to get it up and running.
If you're running your Tascam still: USB soundcards don't like uneven
samplerates. Try using 48 kHz instead of 44.1 kHz.
actually, i tried an old USB soundcard, edirol UA-1A ...worked instantly with whatever drivers i had already installed.
the tascam still has some issues...i will figure them out soon hopefully.
why are you saying that about the sample rates? i never had any problems with that in os X... is it a linux specific thing? also, all my samples 44.1 and all my patches just assume 44.1
anyway, thanks for the help and encouragement frank.
when i get this up and running smoothly i will finish off that collection of synths and effects that i was making at the end of last year, too.
**** List of PLAYBACK Hardware Devices **** card 0: USX2Y [TASCAM US-X2Y], device 0: US-X2Y Audio [US-X2Y Audio #0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: UA1A [EDIROL UA-1A], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0
tascam now passes pd's "test audio and midi" test too. i still get some OSS error messages though.
what's OSS ??? do i even need it?
ooh! it's VERY LOUD!!! as in,,,,everything in the .doc file demo patches is just distorting like crazy unless i turn the volume down to about 50% or less.
Hallo, hard off hat gesagt: // hard off wrote:
Congratulations!
thank you. felt pretty good to get it up and running.
If you're running your Tascam still: USB soundcards don't like uneven
samplerates. Try using 48 kHz instead of 44.1 kHz.
actually, i tried an old USB soundcard, edirol UA-1A ...worked instantly with whatever drivers i had already installed.
the tascam still has some issues...i will figure them out soon hopefully.
why are you saying that about the sample rates? i never had any problems with that in os X... is it a linux specific thing?
It's an ALSA specific thing, however it seems, that this was sorted out in the meantime [1], so you can ignore my message. (It had to do with the fact, that USB audio packets are generally transfered in blocks of a constant duration. Quoting Clemens Ladisch again:
"One USB frame is 1 ms, but the number of samples per USB frame varies if the sample rate isn't an integer multiple of 1000. E.g., at 44100 Hz, each USB frame has 44 samples, with one out of ten having 45." [2]
also, all my samples 44.1 and all my patches just assume 44.1
This shouldn't matter for handmade players. E.g. if you load a 44.1 kHz file of 1 second into a table, you will have 44100 samples. If you play this with tabread driven by line~ like [0,44100 1000( it will be played at its original pitch, even if Pd is running at a global SR of 48000 Hz.
[readsf~] however would be out of tune. Is there a way to resample [readsf~]? I seldom use readsf~.
Actually we're becoming on-topic again and should move over to pd-list. ;)
[1] http://elists.resynthesize.com/linux-audio-users/2005/08/1518782/ [2] http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg11871.html
Ciao
hi all,
something im curious about and that i just cant believe:
Am Mittwoch, 22. Februar 2006 20:44 schrieb Frank Barknecht: [...snip...]
It's an ALSA specific thing, however it seems, that this was sorted out in the meantime [1], so you can ignore my message. (It had to do with the fact, that USB audio packets are generally transfered in blocks of a constant duration. Quoting Clemens Ladisch again:
"One USB frame is 1 ms, but the number of samples per USB frame varies if the sample rate isn't an integer multiple of 1000. E.g., at 44100 Hz, each USB frame has 44 samples, with one out of ten having 45." [2]
while it is sure that an usb frame is 1 msec (minimum, that is), i dont get it why "uneven" samplerates should cause a problem. first, you _never_ will have _exactly_ 44100 or 48000 for that matter. it has to do with the fact that crystals are always a little bit off. and since each soundcard has some crystal or oscillator somewhere, from which the actual samplerate is derieved, that samplerate would be almost never exactly be the given samplerate. its more like 44105 or 44093.
said that, same is true for each samplerate. the error gets lower if you use lower samplerates, but it will still exist.
_if_ soundcards, even of the same type and brand, _would_ have the exact samplerate, it would be super-easy to use multiple soundcards to get multichannel audio. but as we all know, that isnt possible easily.
if you have access to a frequency counter, you can check that yourself.
instead, to my understanding of the usb communication regarding sound, it is that the used "card" should handle that skew, even more, an intelligent device will request blocks as needed. that is, you have a constant blocksize of whatever you like, and while that block is processed, the next one arrives/is requested. so you can have fixed blocksizes for the usb frame, while just the timing is "skewed" a little to compensate, if at all.
also, 1 msec is the minimum time between each access to an endpoint (the device), but the device can stall the endpoint for a while when it has enough data left. that way, a device can use, for example, fixed blocks of 64 samples, "cache" a few blocks in advance, and only accept new data if there is a minimum amount left, so it would not get any underruns.
i may be that im wrong here with the above, but really, assuming that a samplerate of 44100 always gives you 44100 samples/seconds is plain wrong. same for any other samplerate. so in any case, an usb device has to handle samplerate variances anyway.....
on a different note about usb soundcards: there are single-chip "soundcards" like http://focus.ti.com/docs/prod/folders/print/pcm2702.html
if you read the datasheet, you will see that it has an "adaptive mode" to handle exactly the said problems .....
(and yes, only that chip and a few cheap other parts make a complete usb soundcard for at least audio out ;-)
greets,
chris
On Wed, 22 Feb 2006, Frank Barknecht wrote:
If you're running your Tascam still: USB soundcards don't like uneven samplerates.
Do you know why? Or is it just about them being snubbish?
Try using 48 kHz instead of 44.1 kHz.
44100 is super even. it's 2*2*3*3*5*5*7*7.
_ _ __ ___ _____ ________ _____________ _____________________ ... | Mathieu Bouchard - tél:+1.514.383.3801 - http://artengine.ca/matju | Freelance Digital Arts Engineer, Montréal QC Canada