I have a 410 and I know how easy it is to hookup!
I was exploring some of the other good uses for it.
The 410 is pretty good, only issues I've had with it is sometimes it doesn't sit well with other firewire devices, may well be the fault of the other devices though.

But the whole point I'm trying to stress and Grazie to work on is this:

Lets start with just a plain old mic input. Plug mic into 410. I can monitor that in the 410, it just sends the signal thru its mixer back out, zero latency, no need for ASIO or anything. Vegas can record that as well. But what I'm hearing in that scenario may not be what Vegas is recording. To hear that I need to send the signal from Vegas back out to the 410. NOW latency becomes and issue and it hits at several points.

Lets consider midi, about which I know very little but I'll try. I assume I can plug a keyboard into a hardware synth and it had better play out the note when I hit it! I can I assume also route the midi commands to another app (Acid or whatever) to record the same midi command. Still no latency issues.

But if I send the midi commands to Acid and from Acid to a softsynth and the softsynth sends its output to the sound card then I can see latency issues at many places in that chain. Worse still if Acid is applying FXs and playing out other tracks that's more CPU overhead, same thing in Vegas with recording from a mic.

Hope I've explained it clearly, I'm an engineer and spent most of my life breaking problems down into little bits, the throw all this at it and it'll go away approach just doesn't sit well with my brain. I'm certain it all comes out the same in the end, but I like to understand the why and where, sometimes it can save a lot of money, thats why I still drive a 20 year old car, if pressed I could fix anything on it!

Bob.

Bob, your understanding of an initial "key" press on my M-Audio Oxygen 8 is very probably what happens. And I understand that at each stage when that input is "affected" by either ACID and/or an FX or virtual instrument can prolong the final audio being monitored at the speaker set.

"Hope I've explained it clearly, I'm an engineer and spent most of my life breaking problems down into little bits" yes, I'm with you on this one, " . . the throw all this at it and it'll go away approach just doesn't sit well with my brain." . .me neither, that's why I'm here working very hard to get a solution. And yes, you are correct, "I'm certain it all comes out the same in the end, but I like to understand the why and where, sometimes it can save a lot of money, that's why I still drive a 20 year old car, if pressed I could fix anything on it!" You paint a picture of yourself as being a truly interested and understanding in my dilemma of my knowing a little but enough not wanting to "screw-up" what I've got . . plus you've given some indication you would approach these issues and reason deductively the solution/s.

Ok . . My position is that of a chap wanting to have this MIDI to work with and within ACID. I make Community and Event videos for clients . .sometimes I'm wanting to "stamp" my creativity on a very short piece of video. Ok, that is why I'm asking hear, on the Vegas Forum, for solutions with ACID because I'm inextricably bound to the hardware overhead requirements that video=Vegas demands on my ASUS P4 3.2. Apart from that, I feel there are far more people who "know" me here and understand my level of IT Savvy - yeah? I know I can be infuriating .. I don't mean to do it - just raw talent! U-HUH! . . At the end of the day, my input in all this "stuff" is that it should be easier than I'm experiencing. Why not? Too much to ask? Am I being too naive?

As to the 410 . . ppphhhoargghh . . . this sounds interesting. And yes I don't want to throw more money at something that something I've already got might solve the "problem" - but by all accounts - thanks Chaps! - it would go a long way.

Ok .. Thursday . .another day . . another Dollar .. another Dime .. or Pound!

Grazie

The default samples value assigned by the audio driver to the buffer is probably high enough to assure that no dropouts or stuttering occurs during playback of all channels.
I don't know what settings-options gives you the sw for the on board audio chip, but the control panel of a decent card's driver interface should let you adjust latency by defining the buffers size.
Buffer size is calculated in audio samples and lower latency settings add processing time because of the need to refill the smaller data buffers more often (hence the default value is usually set high for safety).
This means that when you start adding real time effects such as a digital delay, reverb etc. or if you keep adding midi tracks with different soft synth instruments and playing all simultaneously there will be a time that low latency will make it difficult for the processor to feed the buffers in time and dropouts or stuttering will occur.
At some point you will have to mix down a couple of midi tracks to an audio track or increase latency.
With a decent audio card using WMD (Windows media drivers) you could easily setup a configuration where you would be able to playback flawlessly at least six to eight tracks (I am talking about midi tracks driving soft synths in stereo, not tracks with audio waveforms), some real time effects assigned to the audio output of each synth, a couple of stereo audio tracks for mixing (midi tracks to audio for not letting the number of midi tracks cross the cpu limit) and at the same time use your midi keyboard with latency of quite less than 10ms.
Using the above workflow (and with more audio tracks), a setting of 256 samples in the dma buffer gives me 5.8ms latency on a P4, 2.4 with an m-audio audiophile 24/96 soundcard.
Hope you find something useful in all this