hardware = emu0404 usb2 soundcard/advent IntelCore2Duo 1.6, 1GB ram
software = latest emu asio drivers/puredata/midiyoke

problem is audio latency (not yet concerned about midi lag), in a 'full duplex' setup; i.e. realtime audio I/O

I have a very respectable 5-10ms full duplex latency in all other software environments (Ableton, Cubase, Audiomulch). The audio hardware entries I have in pd (via -listdev) are:

(0)microsoft mapper
(0)emu usb 0404
(1)asio emu usb 0404

selecting the first emu entry gives clear audio, but with>37ms delay; selecting the asio entry creates garbled audio (buffers overflowing?) at any setting.
Could any knowledgable soul suggest suitable startup flags to address this problem? I'd like to be able to use midiyoke at the same time, but it's not vital; I have a sneaking suspicion that midiyoke is at fault
thanks
Brendan

Hi Brendan, and welcome.

I can't help you with your issue,... but I do have a question in return.

How on earth do you accurately calculate latency in a Win-XP (~:o( situation?

Keep well ~ Shankar

It depends on what exactly you mean by latency: are you talking about midi-latency (between pressing a midi trigger and hearing the result), or processing digital audio (realtime and/or offline)? It also depends on what your software/hardware environment is;
let me know what your full specs are and i'll try to help; still no luck with pd latency though, do you run pd?

Hello Brendan,

i'm afraid i'm struggling with the same problem you're talking about. I've already posted this a couple of times, but without a satisfying answer. I also wanted to know if i could take my presonus firebox under linux to make things better, but no one seemed to know. I'm under the impression that latency isn't an issue for most pd users because :

• for art/installation/experimental stuff latency doesn't seem important
• for music stuff, most pd users seem to work with preloaded samples and don't take external audio input

I for myself i've learned to live with it. I play guitar and i've adopted my style of playing to the latency. Although i've already considered switching to max/msp, because they have a respectable asio implementation.

If you find more information please let us know.

Cheers,

Domien

damn!
is there really no solution? I seem to remember running pd on my desktop pc, using an emu pci soundcard, and audio-in latency wasn't a problem.
Oh well, i'll just stick with Audiomulch and glovePIE then
Brendan

i think pd defaults to 50ms latency and you can set it shorter in the preferences dialogue.

solved the problem:
locate the folder called "/pd"
right-click and select "delete"
download max/msp

@domien said:

If you find more information please let us know.

@jjcomet said:

solved the problem:
locate the folder called "/pd"
right-click and select "delete"
download max/msp

Mmm... That was not really the information i was looking for. Oh well, if the latency issue is that problematic, it's the way to go i suppose (although 500$ is a bit much for me). Good luck and keep us informed about your max/msp findings.

Cheers,

Domien

Hi JJ,

I do suggest sticking with Pd and getting it set up properly for your machine. In Linux we have
Jack which allows a pretty easy setup of soundcard options to match Pd. I don't know much about
Windows, but maybe messing with your soundcard buffers is worth a try. You can launch Pd
with a number of options too, if you do it from a command prompt you can see verbose output
telling when x-runs happen (when buffers mess up),how big the buffers are...latency is really
a function of the buffer size, smaller == faster.

Does the input latency issue go away with Max? What are the settings in Max, try applying those
to Pd.

best

Andy

Andy,

i'm sorry, i don't wan to play advocate of the devil, but i've tried every possible setting for pd so far (under winxp). All other audio applications run smooth with low latency asio settings, but pd doesn't like it at all. I can run at a theoretical 3ms, but in pd, when going lower than 40-30, things start to break up. I really wished the devs could do something about the portaudio implementation, because apparently that's the same technology used by audiomulch, and i can run that one also at 3ms.

Domien

That must be really frustrating. I guess this is now a known issue yes? I wish I knew something about portaudio on Windows so I could look into it more, but I 'm clueless about it I'm afraid. If you are you able to get any kind of debug outpout from Pd that helps then file "shit audio latency" as a bug report Remedy here is patience and looking for any clues to help with dev. I guess what you say above is true, wouldn't actually notice it myself even if I was on Windows because the way I work it doesn't require low latency much.

Andy

I've figured it out. What worked for me was changing the Windows Audio Settings to my USB soundcard (Tascam UC122L), once that was set, I went back into PD and the ASIO drivers work. I haven't tried sending audio in, but there was no more clicks in the audio and when I created a basic synth, I could play with my midi keyboard via midi yoke without any noticeable latency. Hope this works for you!

No guys doesn't work with me.
only know I want to play my guitar, without changing soundcard RME fireface800 neither winzoz xp.
I really like pd, I couldn't imagine it was so nice when I threw away max/msp last summer.
I only ask for help cause I don't want to leave from pd.

thanks

Hi all, after a long work I think this is my final hardware setting, that allows to play my guitar without headaches...

from fireface settings I use 64 samples as buffer size (48 buffer size for cheap projects)

inside Pd I choose the standard MMIO, then fireface MME drivers (+ channels). Asio seems not to work.

then set the delay (milliseconds) an equal or higher value than the resulting latency caused by hardware settings. Around 3 - 6 milliseconds is ok for me.

cheers

Enrico

Hey,
RME stuff is well supported by ALSA and you'll get the lowest possible latency with linux. Now that jack is sort of working for 'doze, that's probably the best way to go. Jack's favorite samplerate is 48000 (or 96k), So start off with that if you can.
Also, if you aren't willing to go to great lengths for PD's sake, like changing OS'es, than you probably should switch to MAX/MSP. It is a commercial product, for those who support commercial products like windows, software patents, and ridiculously high hard/software prices..

Thanks Nestor, I worked on linux for 6 months and I'm not the right user for that.
Further I cannot use linux because: my sound card is the only one not supported by alsa drivers among RME stuff, and, I'm a really convinced Cubase user, cannot work on a emulator.
But in respect to Max/msp I prefer PD, it is radical, deeper and stronger than Max (to me), open source, with a really nice community, I like to be involved in.
People says that PD works better on Mac OS, maybe I will move on that system for the next laptop I buy.
Anyway until that day if I want go on with my hybrid computer music I have to struggle to get the best from what I have.

bye

If moving on to OS X, definitely use Jack OSX natively for best audio performance/latency.

Portaudio is a bit wrong i think.

Unfortunately there is still a bug with the multiprocessor version of jack (later than 0.74), Leopard and Pd 0.40.3, but 0.74 works fine.

S

so the only solution with the latencyproblem in windows is to change os?

is there no way to change the buffersize in windows-pd to get lower latency?

you can try to install Asio4all, I obtain a nominal latency of 15 ms with my built in audio card realtek. Inside pd choose ASIO (portaudio) and lower latency from there.
Instead using my "good" sound card I cannot choose the asio drivers (that is misterious)

this did not work for me.

its sad that a such a good program should have so poor latency. =(

there is also Jack for Windows too. Give that a try
http://www.grame.fr/~letz/jackdmp.html
s