I thought the converters were OK...but there were others that were better...I thought the "compressor" was a dripping piece of monkey fuck. I'd rather use a real compressor in front of a slightly better converter...but that may just be me.

I think the unit is best suited as the last item in the chain AFTER the 2-bus compressor, and as Fletcher said after a better converter (AES into the L2 from an Apogee, CraneSong, or Drawmer Masterflow A/D), but to be fair the A/D in on the Waves is way better than anything stock on a DAT machine or stand alone CDR machine, not just component wise, but for superior clock stability also sent via AES.

That said, this type of limiter/maximizer IMO doesnt handle the audio in the same way a traditional two bus handles audio, its not made for smoothing, its made for getting it into the last .02% of digital zero. If I were you I would get a high quality two bus compressor (the 1969 Drawmers are really nice) and a seperate A/D box like the Drawmer Masterflow and use the L2 in the computer or maybe the Sonic Timeworks Limiter for mastering if you are running Native (ie Paris, Samplitude). Following the past pattern of Waves, the L2 will probably be out for Native shortly. It all just depends on budget also. YMMV

Nathan Eldred
Atlas Pro Audio, Inc. http://www.atlasproaudio.com
Tampa, FL

Well, the limiter is digital (i.e., after the AD converter), so that completely discards any reason to use a limiter on the front end of digital. If you use a clean, supposed to be transparent limiter before you go digital, it's so that you don't clip the AD converter- that means good analog limiter. Turning down below clipping is cheaper, though, and probably will sound better more often. Similarly, turning up the volume *digitally* (with a limiter's gain make up, a digital fader, whatever) does nothing to increase resolution, and may harm the quality of the signal itself.

The 'purpose' of this box is to squeeze a little more volume out of a stereo mix transparently (generally in mastering)- other end of the chain from tracking. The only thing you're going to do if you hit it on the way in is run out of bits for processing headroom- this means that all the EQ, compression, and level changing you do on mixdown will sound slightly worse than they would have if you'd have just treated -15 or so as zero and left some room on the top of your tracks.

Same thing mixing through it, really- the difference between an un-L2'd mix and one where you squeeze it on the way out is that the mastering engineer has a more difficult job working with pre-limited material with no headroom, and your end product will suffer.

So, it's got pretty nice converters, from what I've heard, but it's spectacularly unsuited for tracking, and IMO, mixing through. For mastering, it's hard to beat.

Originally posted by Faeflora:
So, what's yourall's thoughts on this piece of equipment as the last stage before going digital? i've read that the converters are spiffy and that it works it's purpose well.

Ok, thanks guys :).

Originally posted by Jon Best:
Well, the limiter is digital (i.e., after the AD converter), so that completely discards any reason to use a limiter on the front end of digital.

Correct me if I am wrong, but I thought the maximum amount of bits were obtained by PRINTING to the digital storage medium at the maximum average volume without exceeding digital zero. Its not always practical to print to a 24 bit file either. I agree that slamming a digital limiter while printing a mix is probably a bad idea, but a slight amount would not hurt to help obtain the full 16 bits to CD.

Nathan Eldred
Atlas Pro Audio, Inc. http://www.atlasproaudio.com
Tampa, FL

Originally posted by atlasproaudio:
Correct me if I am wrong, but I thought the maximum amount of bits were obtained by PRINTING to the digital storage medium at the maximum average volume without exceeding digital zero.

Just because they're there doesn't mean you have to fill them all up! hehe. The low bits are important too! Use them, that's what they're there for.

Jon made the point that if you're handing the copy off to a mastering engineer, he won't have any room to work if you already went and squashed it up to -0.5dB average. So you ask, "well, what if I use just a little bit?" If the guy who's mastering it has the headroom he needs to do what he does, fine. If you don't fuck up and pump your compressor or some other kind of mistake that the mastering guy won't readily be able to fix, fine.

This is when the mixer has to ask himself a couple questions: What are the chances that my limiter will sound as good as the mastering studio's? What are the chances that my monitor system presents as detailed an image as the mastering studio's? What are the chances that I have the experience to make a decision about the mastering, even if I was in that room?

My answer to myself is usually Hell no, Skipper! I am perfectly content letting the converter attempt to capture the mix exactly as it sounds coming off the bus. And it shouldn't be that far from the intended average to begin with. If you need to add 40dB of gain to be in the same ballpark, you might want to approach the mix in different way.

Alternative situation... if you are certain the mix will NOT be going to a proper mastering studio, all bets are off. Do the best you can with whatever you can get your hands on, and say a prayer.

Well, kind of- actually, it's 'the maximum amount of bits are obtained WHEN you hit the converter.' Once you get into the digital realm, you have decided how many bits of resolution you are going to use- raising the volume after that just does some math with the information you already have. If you're going to leave the tracks in your digital recorder un-messed-with (i.e, no digital level changes or processing of any kind- think ADAT), and mix analog, then, yeah, getting as close as you can to zero without clipping may do you some good. Unless, of course, you're mixing a bunch of tracks that are close to clipping the line inputs of your analog mixer, in which case you're going to tax your mix bus, and it'll sound worse. On the other hand, if you're doing any math in the computer like processing and mixing, those things add information to your tracks. If you've used up all the available 'information slots (bits),' you end up with a longer-than-24 bit word, and something gets knocked off (kinda). I don't think this is as much of a big deal in an application/plugin that is 32 bit float or 48 fixed throughout. So, really, just pick a place like -18 or -15, call it zero, and average there, unless you really hear a difference (that can't be attributed to loudness) hitting it harder. Especially in 24 bit situations.

Of course, I could be completely wrong for any particular situation- listen!

Originally posted by atlasproaudio:

Nathan Eldred
Atlas Pro Audio, Inc. http://www.atlasproaudio.com
Tampa, FL

Originally posted by atlasproaudio:
Following the past pattern of Waves, the L2 will probably be out for Native shortly.

Isn't the L2 already just the hardware version of the native powered L1 ultramaximizer?

GigaBoy

No, more like an advanced version. I *believe* the main difference is that the L2 uses a 48 bit fixed internal data path, while the L1 is still 24 bit. I may be wrong, but either way, the L2 is better sounding, according to most every mastering engineer I've heard an opinion from.

Originally posted by GigaBoy:

GigaBoy