Started setting things up for our project studio and when running the card through its paces, we found that if we recorded at the 96k sample rate things sounded like they were clipping. All of the sources were not, and when we lowered it down to 24/44.1 everything returned to normal.
I am assuming this is some sort of latency problem, but I'm not sure how to go about resolving it. We are running Windows XP SP2, an Intel 2.4 gHz machine with 512Mb of Ram and a 7200 Rpm ATA100 hard drive. We are using Cubase SX 2 as a recording program.
Any ideas? I think we really want to record at this level (probably 88.2 actually) since we have heard that there is a noticable difference.
Read my first two posts in [[url=http://[/URL]="http://www.recording.org/modules.php?name=Forums&file=viewtopic&t=22424"]this[/]="http://www.recording.org/modules.php?name=Forums&file=viewtopic&t=22424"]this[/] thread. Try my buffer setting algorithm and see if that works. You will have to run the algorithm once for each sampling rate you plan to use. Store the results as presets.
Thanks David... I'll try it out and let you know how it goes!
what type of motherboard do you have?
I have a Abit BD7-RAID.
I had the same problem, and i found out that the onboard raid controller was the reason. i just had to flash my bios to fix it :)
That's another idea. If you don't have the latest updates for your hardware, get them.
We don't run that MB, and we do have all the latest updates for the hardware. Thanks for the tip tho! :)
Still havent had a chance to get back to the studio to try it yet, but will be doing it this Thursday. Will let you know.
David... 88.2 should be sufficient right, or should we go the full 96? We will be getting someone else to master it, so hopefully they should have the right gear for SRC.
The math involved in resampling from 96 kHz to 44.1 kHz is very complicated. Use 88.2 kHz since the math is much simpler and will theoretically lead to a better sound.
I tried your method but it didn't help. I'm not sure if I did it right, but I was running the simulation on the 1010 with the lowest possible buffer setting and it was still passing the test. I put the buffer up to 256 just in case, and I'm still getting distortion. If I go back to 44.1 everything is fine.
Any other ideas?
No Clue. I would call M-Audio at this point.
do you have enough memory?
you need at least 128MB for 96KHz recording.. (has anyone got less than that today :? )
You might want to try the WDM drivers if that is supported by Cubase. I've found, using Sonar 4 PE, that M-Audio's WDM drivers yield better performance than the ASIO drivers. I've recorded at 96/24 with very low latency (128 sample buffer) without out hitch using the Delta with WDM.
Just a thought,
Yes, we have 512 Mb of Ram... I would think that's enough.
As for the drivers, hmm... the one we are using is wdm, but there is another one on the site that says web. I'm not sure what the difference is, maybe I'll try that one out.
Still haven't heard back from m-audio support. :(
i would try a diffrent mobo. sounds alot like the motherboard is the problem.
I doubt it's the motherboard. If it was that, I would get pops, clicks and distortion whenever I recorded, not just at a certain sample rate.
It's likely either drivers, or the buffers being set incorrectly. Still haven't heard back from m-audio but it's only been a few days.
Hi, i use he 1010. Just wanted to clarify. This sound that you hear, is it actually clipping or is it a sort of crackling sound which seems to be accompanying the recording? I use Vegas Audio to record and get the same problem, not always, but about half the time. The pc i am using is not very fast. I have tried changing the buffer settings but have not been able to rectify the prob. Wrote to m-audio, they advised me to see if there is any IRQ conflict, did all that, still have the prob.
Don't have a solution, am hoping that things are fine when i get a new and much faster pc....
It may be worth taking a look at is your PCI bus timing for each device. If one device is hogging as much 256 cycles before your audio card gets its turn with the CPU, that could cause pops and clicks. You motherboards bios may let you adjust the latency timer settings for each device. Graphics cards are notorius for being PCI hogs. Ideally, you'd want your audio card to take priority over other devices.
Perhaps if you record a few more than twice as many tracks at 44.1 as you do at 96 you can test this theory. This would create the same bandwidth usage as recording a few less than half the tracks at 96. Even if the recording setup is redundant with respect to your inputs it will still work.
I'll try both of the suggestions and see how it goes.
I got an email back from M-Audio, basically they are giving me another driver to try... an older one. I'll try that out and see if it works or not.
David, if I'm recording fine at 44.1, is it even worth it to go through all this hassle in your opinion to record at 96 or 88.2? That will also significantly increase load on the cpu when processing 16 or more tracks at the same time during mixing correct?
For the gains, is it worth the "price"?
This is heavily debated, but my opinion is 'no'. Theoretically you do increase your plugins' processing accuracy when using a higher sample rate and this accuracy trickles down to the 44.1 end product, but I don't think it's justified at all for home studio recordings. Unless you have killer SRC and power to burn, I would stay with 44.1.