I'm using a pretty basic Pro Tools rig: new macbook, digi 003, API pres, etc. At most, I'm just tracking 3 instruments in stereo at one time (6 channels). No plugs. I'm running the outs on the 003 to a headphone amp to generate individual mixes for those tracking together.

I've just put this rig together and I am trying to figure out the best way to eliminate latency while recording, while still getting the best possible sound quality I can. I can't use 'low latency monitoring' and the headphone amp at the same time, and I've got the buffer size down but sometimes too low it seems (the system will crash and call for a larger buffer size). What is the best way to balance latency, buffer size, and anything else that i don't already know about??

Might it have anything to do with the fact that the macbook has only one firewire port?

Many thanks,
Chris

Hey Chris. I've got a very similar setup to yours. I've got a macbook and a Digi 002. When I'm tracking, I go under "Setup" and "Playback Engine" and I set "H/W Buffer Size" to "64 Samples." When I'm mixing I set it to "1024 Samples." You just can't run a bunch of plug-ins (especially big reverb plugs) when your tracking. If you just have to have reverb when your tracking, D-Verb doesn't use much juice.

quote:

What is the best way to balance latency, buffer size, and anything else that i don't already know about??

Sell the 003 and get a motu? Almost all non-PT interfaces have a zero-latency monitoring feature for inputs. They give you a software mixer, where you can set levels for monitoring each input channel. This software mixer runs seperately from the DAW. I'm surprised PT interfaces don't do this, since monitoring inputs with zero-latency is sort of a no-brainer in the recording world.

Hi, I just bought a digi 003 after working for years with a large tdm system (mix3). As I record a lot of real instruments and performances I was horryfied to hear the huge latency on the inputs. This is how I fixed it.
On the playback engine page (under setup)I changed the value of the DAE buffer size to the lowest possible. I think that is Level 0 (instead of the Level 2 which is the default one) and to the right of that there is another value, I think it's Cache, that one I set it to the maximum. That stopped the latency completelybut only if the input was directly to the audio channel I was recording. If I wanted to go through an auxilary first, with an eq and a compressor (so I can record the processed signal) there was still enough latency to put off the performer. The way I got rid of that is to set the H/W buffer to the minimum (32 samples). Now I have almost 0 latency and I am happy. Of course as the paople before me mentioned that is for recording, When you enter mixing mode you should return all this set ups to at least their default value.
I hope I was of some help.
All the best

There is a LOW LATENCY feature in LE. I forget the key command as I never need it and work in the HD realm. Go to the "options" menu I believe and select "Low Latency Monitoring" All sends are diabled (verbs, etc) but it does help ALOT while tracking. Also, the lower the latency you can get away with if NOT using the low latenct feature the better obviously. Most newer machines should be able to hit 64. 128 for sure. Also, if you are on a laptop you really need to turn off all the bells and whistles. Hit digi's site on the DUC forum to read about what to turn off. I have a dual core laptop that I take with me on airplanes to mix prep with and I have a dual boot setup specifically to allow my piece of shit Larger than life MBox 2 Dongle work. If Bluetooth and all that shit is enabled it will crap out on me. Sucks that you have to do this, but with high rez audio all that winblows shit adds up on the processing power.

Hope this helps?

Doc