Limiter (hardware) for recording purposes

[cptgone]

@ JCJR: thank you so much, it' s appreciated.

They have a v3 version of this that is blue and has a settings knob for the "limiter" rather than a simple on/off button, but its a hundred bucks, this is fifty.

it's got a tube, and its not meant to be transparent, so if clinical cleanness or audiophile snobbery is what you're after, this ain't it.

I found the blue one, but it had bad reviews (customers complaining of hiss).
I think i'll instead try recording at higher bit depth and at conservative record levels.
Thanks though, it's great to read about gear like that, and how people use it.

[JCJR]

I think i'll instead try recording at higher bit depth and at conservative record levels.

If your interface will do 24 bit then I doubt it would stress your computer any more than 16 bit unless the computer and storage are remarkably slow for modern times. Long ago maybe it could have been a straw that breaks the camel's back.

Project sample rate could be stronger impact on computer performance but I personally think recording at the usual slowest rate of 44.1k is generally fine. Faster sample rate is ok as well but 44.1k generally gets the job done

If your daw allows adjustment of default track storage format, there could be advantages to setting it to always use 32bit float files. Those are somewhat bigger than 16 or 24 bit integer files but storage is cheap nowadays if you find yourself running short and most PC's come with fairly decent storage nowadays.

32 bit float file access speed might be a hair slower on slow hard drives, or maybe it would be such a wash that it would be hard to measure. Most daws will convert incoming 16 or 24 bit to float for internal processing so maybe (or maybe not) the disk is a little slower loading larger float files but in that case the daw doesn't have to convert to float because it is already float. Maybe a little more time loading and a little less time format converting, yadda yadda or whatever.

If your daw default track file format is float then regardless whether your interface is recording 16 or 24 bit the files get saved to float. Float files do not add fidelity to a raw incoming 16 or 24 bit stream but if you happen to process and re-save a track file then less information would be lost to math rounding errors.

Another advantage: If you make a mistake and save a processed float file with too much gain, so the signal exceeds 0 dB, then the float numbers will not clip. Unless some wrong headed daw does unnecessary disk write clipping then an accidentally too hot float file won't have clipped data. Just load the track back into a daw or editor program, turn it down so it doesn't clip, and play, mix or re-save. No gross harm done.

OTOH if you save 16 or 24 bit files with the gain too hot then the saved result is permanently clipped forever.

Float files can't undo clipping on your audio interface inputs of course. If you record a "live" track with your interface heavily overloaded and clipping then the float file will be a faithful copy of the input clipping. Float files advantages are mostly after audio has made the trip into the computer from the real world.

The conservative record levels work fine in most situations even at 16 bit 44.1k. Even back at the beginning of 16 bit digital recorders, DAT, ADAT, first generations of expensive 16 bit protools and such interfaces, the "audio pros" would typically use in the ballpark of -15 or -18 dB as the normal input operating level.

They would do something like: Send a test sine wave into the console and adjust gain so that the analog console output meter reads 0 dB. Then put the DAT machine into record mode and adjust so the test signal reads -15 or -18 dB on the DAT record meter.

When set that way, the console meter would have to go peaking up around +15 or +18 dB before the DAT would clip. It wasn't entirely cut and dried because different kinds of meters had different ballistics and peak reading accuracy. Was just a general procedure.

So a lot of very fine quality recordings were done in 16bit 44.1k where at least some stages of the recording were input at conservative level.

For mastering or other "final release" mixes the rules were different and the goal perhaps closer to recording to dat as hot possible without clipping. But those were later process steps where if you make a mistake then just adjust knobs and try again.

In the inirial tracking step the nightmare event is when the musician has played the same silly part all day. But the only time he played it right, the levels were set too high and his only good take of the day got badly distorted.

[cptgone]

Thanks JCJR, I'm not very well at the moment so it takes some time to process things, but wow what a treasure grove of useful info, I'm all set now