Nominal Buffer size for Audiophile 2496

S

studiomaster

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I have been really trying out different buffer sizes for my audiophile 2496, but cant figure out which runs the best. currently, its set to 384 samples..but i get crackles in the audio when i playback audio in Cubase. anyone know what the problem seems to be? should i change the buffer size?
 
It ought to work well at the lowest setting of 64 samples!

Having a low buffer setting does mean the driver is using more CPU cycles as the card will be interupting more often to refill the buffer.
256 samples gives a good compromise between CPU usage and a low enough latency for playing software synth plugins. On a machine as powerful as yours, it should work with 64 samples easily, unless you run a really large number of tracks and plugins.

Are you sure Cubase is actually using the drivers ASIO? Steinberg call all the options "ASIO..." something or other, but the ones with "Multimedia" or "DirectSound" in the name are NOT really ASIO, so don't use any of those, they're for users whose interface doesn't have a real ASIO capable driver.

The clicks indicate something is wrong in your machine. It usually happens when some device is interupting too often and the soundcard misses out transfers of sample data over the pci bus. Graphics cards are the usual culprits, but also Network adapters, Disk Raid Array controllers and some SerialATA drive controllers can give problems.

Try reinstalling the card in another pci slot, to find one not sharing an Interupt (IRQ) line with something that will be busy like a disk controller of Graphics card. I've found it's ok if the sound card's sharing with a USB controller, but milage may vary!

Cubase ASIO (it's own soundcard interface protocol) apparently runs as a "background" service in Windows. Some think it helps to change Windows performance priority to Services rather than Programs, but I think this matters less on a well specified machine.

The info on optimising Windows XP for audio on the link below is worth a look...
www.musicxp.net

Good luck ;)
 
If Jim Y's excellent advice doesn't work,try an earlier version driver.

Back when I had a 2496,I had the same problems after updating my drivers so I rolled it back to an earlier functioning version. :eek:
 
Thanks for the wonderful reply Jim Y. thanks to acidrock also.

Yup.."M-audio Delta ASIO" is currently selected by Cubase...under the drop down menu of soundcards, there are 2 checkboxes: Release ASIO driver in background AND Direct Monitoring". Direct Monitoring is automatically checked for me. Should the other one be checked as well? Also..especially for Delta users, in the Delta Control panel, did you guys check the box "Disable audio app use of Monitor mixer and Patchbay/router"? mine is checked..im just curious to find out whether or not these are fine.

once again, thanks both of you. :)
 
I don't actually use Cubase.
The "Disable audio app use of Monitor mixer and Patchbay/router" thing should be checked, especially for Cubase as it apparently changes faders and pans for no good reason if it isn't checked.

Direct Monitoring is putting extra load on the system, and it makes no sense to use it with a high latency. Direct or Input Monitoring is in most DAW programs and echoes the incoming audio of any tracks armed for record so you can hear the effect of any FX plugins and eq in the tracks. There must be very low latency from the soundcard buffer for this to work, otherwise what you hear is hoplessly delayed by at least the twice the buffer latency time (the sound had to go in, thru the track fx and back out). If you don't need to monitor track fx while recording, turn Direct input monitoring off.

With the m-audio cards there is a "monitor mix" output option in the control panel patchbay which can allow you to hear the inputs and outputs over the soundcards main stereo output, according to the control panel mixers faders. You won't hear any Cubase track FX on inputs this way though.
It's important not to use the cards monitor mix as well as your softwares direct input monitoring, else you'll hear the audio with 2 different delays and it'll sound like all the upper mids and highs are missing from the sound!

Another fix that occurs to me is to turn down the graphics cards system time. The thing here is called IRQ (or less accurately) PCI Latency. It matters even if the Graphics is in an AGP slot but cannot be tweaked if it's a Pci-Express card. This affects the time the device hogs the computers internal paths (busses). N-videa and ATi graphics are very greedy in this respect, taking 248 or 255 clocks per access while the soundcard only asks for 32.

If you do a Google search for "pci latency tool download" you should find it, but for ATi cards, I found one just for those...
http://downloads.guru3d.com/download.php?det=733
...though I'd prefer the generic tool as it'll use less resources itself.
A latency setting of 128 or 64 might cure the performance problem for you.

I use a Matrox G550 AGP card in my DAW, these have a pci latency of just 64, much more reasonable and less prone to cause glitches in soundcards.
Absolute crap for 3d gaming though ;)
 
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