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In Reply to: latency questions posted by MediaSeth on December 3, 2004 at 11:14:09:
> 1. Does latency from my computer -> m-audio audiophile -> analog out to NAD receiver have any effect on sound quality? <Only if the latency is too low, and starts to cause crackling or distortion. Increasing the latency through larger buffers prevents this.
> 2. What is an acceptable level of latency? I don't know if 3000ms-4000ms is considered low, medium, etc. <
Depends on what you use the card for exactly. If you are running pro-audio software and/or software synths etc, you'd like the latency to be as low as possible (before the limitations of your computer set in). If its purely for audio play back then having larger latency isnt a problem.
As for actual figures, for software synths somewhere in the range of 3ms to 15ms would be acceptable. Any larger and you would feel a laggy response which would become a discomfort.The other situation is if you use the card for dvd video playback also, then you need to worry about lip sync, but most modern software (nvdvd3.0/theatertek2.0 etc) can compensate for any constant audio latency.
> 3. What do I select in m-audio's control panel to compensate? m-audio seems to assume I know what this is.
4. I'm using kernal streaming at 24bit because I can. Is that..wise? <I dont own an m-audio card, so my help here will be limited. But a lot of stuff is similar between all pro-audio cards. Basically increasing the sample rate will lower the latency. Increasing the buffer size will increase latency. It is usually best to set the sample rate to "auto" if possible, that way no up/down sampling occurs when playing back music.. Unless of course you are of the mindset that upscaling is a good thing (of which Im not), then feel free to set it to the highest setting, be it 96khz or 192khz.
Not all cards will upsample in software/hardware on the fly, most ESI's and Hoontechs can, I dont know about M-audio.
With the buffer size, keep it large enough to not cause any crackling/distortion and you should be fine.
Typically a pro-audio card will allow you to set ASIO, WDM, GSIF, DirectSound etc compatibility... What you choose will depend almost entirely on the software you plan to run with the card. The ASIO api is normally the best choice when available, as it allows low latency multi-channel communcation to the card, including advanced control over monitoring and sample rate control. WDM being the normal windows interface, with DirectSound a subset of the DirectX api (quite a bit more advanced than WDM). GSIF is for gigastudio, chances are if you need it, you'd know.
Sorry about the generic reply, hopefully someone else can be more specific.
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Follow Ups:
Thanks! I was beginning to think no one would reply. Besides using Cool Edit Pro from time to time to edit stuff recorded elsewhere, I just use my PC for playback. I guess I don't have to worry, then. I switched from Kernal streaming to Direct Sound 2.0 and it seems smoother. I guess it's simple enough to play around with until I find a happy medium. Thanks again.
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