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I recently switched my recording setup from Delta 1010 (8 tracks) and Delta 44 (4 tracks) to two 1010's. This means I have 16 tracks and can record people live without skimping on double mics on guitars/bass etc.

My next major step (aside from adding to the mic closet), is some more nice preamps. I posted a question about monitoring Syteks awhile ago, because they like many mic pres have only single outputs for each channel. This means that the most ideal situation is monitoring through the computer instead of the hardware, but the latency has always been off. Lo and behold I discover a latency adjuster on my software (Sony Vegas 7.0).

The auto-correct on the latency is default, and that obviously has been off. So I try manual correct and it has a wide array of settings 1,000 plus or minus milliseconds. I hooked up a vocal mic and some head phones and every time I thought I corrected it certain vowel sounds would reveal the phasing effects produced by latency problems. I also cross checked to a mixed signal between the computer output and a direct monitor from the preamp sans computer. No matter what I do I hear subtle phasing due to latency.

Is there a more accurate, or less trial and error method to zero in on the latency difference caused by the software processing? Is this always going to be a problem and I have to resign myself to only monitoring from my board and not the computers output?

Can you find your recording and playback buffer size setting in the software or in the soundcard's driver?

Latency should equal: buffer size (in samples) * 1/(sample rate)

Eg. for recording at a 48KHz sample rate at a buffer size of 256 samples. it should take 256 * 1/48000 = 5mS.

You need to consider both recording and playback latency possibly (I'm not sure exactly how the Sony software's compensation works). You can probably not worry about the latency due to the actual a/d and d/a conversion as it is usually no more than a few samples.

There are practical ways of doing this too - make a loop around from one output to an input, play tone and record, then use a wave editor to check the delay between source and newly recorded wave, but watch out for any compensation already in place.

damned informative. Thankyou. I will give it a shot.

It sounds like you are doing software monitoring, which can achieve low latency with a fast computer and good drivers. In my experience, however, when you get this low enough for it to be useful for real-time monitoring, you start getting glitches, etc. unless you are doing very minimal tracking. I'm pretty sure the 1010 can do hardware monitoring, which is latency-free. Instead of routing through the software, hardware monitoring routes signals at the input of the sound card and is zero latency. I'm not sure about Vegas/M-Audio, but we use Samplitude and with the ASIO driver it can directly control the hardware monitoring of the RME Fireface we are using. Otherwise, you'll just have to control the routing with the sound card's software.

Oh, and I just peeked at the 1010 manual. It's called the Monitor Mixer in the Delta Control Panel.

That's right. I used to use a pair of 1010's and using the driver software you can route any pair of inputs directly to any pair of outputs. No latency.

I'll have to get in there and try it. I've never really used the Delta mixer software. I suppose I will just mute the monitoring on the Vegas software (habitual anyways, because I always use my board for this reason) and see if I can get the two 1010's to monitor at proper levels and be able to adjust the monitor mix at the demand of my clients. Yeesh.

you can't record using ASIO?
i have an m-audio card and if i use ASIO for recording i have zero latency.

wait, pretend its the gorilla asking and explain what ASIO means.

ASIO is the type of drivers you're using, it's a windows only thing, and there're normally ASIO as WDM I believe (mac only here). So make sure you're using the best drivers available. And you settings are correct.
More importantly USE the zero latency monitoring mixer.