Originally posted by SoFine:
I guess what I really am asking is how I could use my mixer to better effect from a patching / recording / mixing point of view. I usually have to rewire the digi I/O a couple of times during each session when I have 20 empty channels of analog mixing desk staring me in the face just daring me to use it.
Ok, I see what you're getting at now. Sure, it might cut down your setup time to do things this way. You might even prefer the sound of the mix itself coming off the spirit as opposed to the internal mixing of PT. Good, and gooder!
I suppose that the functionality of the PT mixing environment is more powerful than my mixing desk so maybe it's not worth the effort.
If you prefer to keep everything in PT, a patchbay would help cut down your setup time and aggravation. Unfortunately, it can get expensive. But you might appreciate how much faster and easier routing becomes enough to justify the cost. You won't think twice about plugging a widget in because you have to deal with rack-spaghetti, you'll just plug it in at the bay and keep your creative flow going.
When you do a patchbay, if you leave anything out, you'll end up changing patches at the mixer, 001, or gear, thus defeating the purpose for having a patchbay in the first place. So you always want to have more points available than the gear you have. And if you're like most gear whores around the world, you
will be buying more gear in the future. Better to have too many open points than not enough. For all your mixer i/o's, 001, outboard stuff, etc., you might need a couple 96 pt bays, or several 48 pt bays if you want to go 1/4".
I had heard through various forums that sending your recording signal (e.g. guitar) through a mixing desk before it reaches the digi, and also monitoring from the mixing desk rather than the digi, will eliminate any digital recording-associated latency. Is this true? This has sort of confused me. I think I read this in the DUC.
I guess a quick explaination on latency is in order here.
When you put a track into record on any given recording system, the signal is routed one of two ways.
1. Direct signal, as soon as it goes into the recorder, is fed as straight as possible back to the outputs. This is sometimes referred to as "sync" monitoring.
2. Signal goes through the recording apparatus (and mixing algorithms in this case) so what you are hearing is exactly the same signal as when you playback. This is sometimes referred to as "record" or "confidence" monitoring.
Many DAW's do not give you the option of a sync monitor mode. Instead, they force you to listen to the signal after it has been converted from analog to digital, any processing you have inserted on the track, mixed internally, and finally converted from digital back to analog. This round trip could take a couple hundred milliseconds (or hours, depending on your frame of reference), and the person attempting to perform along with the song may find it difficult to approximate where the beat is supposed to be such a long time before it actually happens. (Hours, milliseconds, it doesn't matter if it prevents you from getting a decent performance. Right?) This delay is referred to as "latency".
The amount of latency will vary from system to system, depending on your a/d and d/a converters, processing speed, number of tracks, cheezy programming, etc. Systems which use their own boxes to do the processing generally fare much better than those that put all the weight on the CPU (host based). Some host based systems keep their latency down to a managable amount by limiting the functions of the program. Such as the 001, where you are limited to 24 tracks when the program could easily have been written to support 100 or more tracks.
One way to bypass latency altogether is to monitor off an external source, before it reaches the first a/d converter. This way of working comes with a couple caveats.
First, every performance you record will be late by the same number of milliseconds as the system's total latency, and the track will need to be shifted earlier by that amount before playback in order to match with the rest of the song. Since you are monitoring only the signal before it reaches the DAW and not the recorded track, your ability to punch in will be inconvenced if not completely impaired. Systems that have a sync monitor function built in usually compensate for this by shifting the start time of the recorded audio automatically after each take, making punching possible but sometimes with a delayed response to the in and out punching actions.
Second, if there is a problem with the sound between the output of the external device and the record track, you won't find out about it until you play back the take. The possible problems include but are not limited to input clipping, digital sync issues, incorrect patching, and incorrect routing inside the DAW. Imagine getting the perfect lead vocal take all the way down the track, and finding on playback that it is distorted, has pops, doesn't exist, or is combined with the snare drum. Don't you hate when that happens?
I hope that helps you understand latency a little better, and to decide which method will work best for you.
Cheers.