Bob Katz: Hi, Mark. In general, from the point of view of the A/D converter, more is better, as long as you are not overloading any intermediate analog stage in line in order to do it. And as long as you have an accurate overload indicator. (See my article on levels: Part I)

BK: Making an initial recording peak near 0 dBFS is not the same as making the final mixed production as loud as possible. Making the track peak near 0 actually maximizes the dynamic range of the recording with respect to the signal to (dither) ratio of the recorded multitrack medium. Afterward, in post production, you can alter (lower) the mixed or post produced level of that previously recorded track to any point you wish, so, once again, the dynamic range of the final production is also not sacrificed. Unless I totally misunderstood the poster's motives, he is wrong.

BK: Correct, as I just said. But "Requires" is a dangerous word. For example, if I were recording a live performance direct to 2-track with a world-class properly dithered 16-bit A/D converter, and one peak in the entire hour hit 0 dBFS, but the second movement was all at -20 dBFS (the original range of the players), it would probably just be fine. The sound would be world-class. As long as you did not have to raise he gain of the second movement in post production or do any post production at all. The point being that in direct to 2-track of live material, if the original dynamic range is good-sounding, then you have made a proper recording. But if you have to do any post production on it, potentially needing to raise or lower some tracks, then it pays to record the 2 track (or in this case, more likely, multitrack) 24 bit with good ("24 bit") A/Ds, so that you have more room above the quantization noise floor when getting into post, where you might have to raise or lower gains. More on this in a moment.

BK: That's fine, not a problem. By recording "hot", you have made a good, clean original recording which you can then later lower in level inthe mix or post stage. That's called "having your cake and eating it, too." It's a good idea: Maximizing the record level of A/Ds, especially the cheap, shoddy Digidesign 16 bit A/Ds in the Avid system, is a very good thing. Another way of thinking of it is you are increasing the signal to garbage ratio of the original recording, where the garbage is at the bottom of the A/D converter. More on this, as we get into the "24 bit rules"....

BK: Correct. But the Avid is a losing proposition anyway, as its cheap, 16-bit digital mixing system adds considerable quantization noise all along the way.... Even if it's 24 bit internally, I understand that the Avid doesn't permit 24 bit input, 24 bit throughput, or 24 bit output. Correct me if I'm wrong.

BK: Yes and no. Yes---if you are significantly (say, 50-60 dB) above the noise floor of the A/D converter and recording very narrow range (say 10 dB range) music, the advantage of peaking near 0 dBFS is relatively insignificant for highly dynamically-compressed sources. But for multitracking and all the original tracking, I would still recommend (as a perfectionist) peaking the original A/D conversion near to 0 dBFS. Otherwise you are just adding the non-linear low level alias-type distortion of the crappy A/D to the already distorted original crappy source :-) So, maybe you won't hear that you're doing bad practice if your monitoring is so bad you can't hear it, your source material is so bad you can' hear it, and your Avid mixer is so bad it adds even more distortion.

BK: I still recommend recording (16 bit at least 0 "hot".... The decisions about volume, loudness, and all the rest of what to do should wait for the later mixing and final post production stage, and all those decisions are relevant to my article "How to Make Better Recordings in the 21st Century", which is my levels article Part II, originally published in the Sept. issue of the AES Journal. Basically, first, your friend needs a good, friendly dose of my article part I, and he's welcome to ask me any questions. I always like to revise my articles if they are not clear. And then, it wouldn't hurt for you both to to read a good dose of my article Part II, because the many issues involved in the dynamic range and loudness of the final product exceed simple questions of whether or not you are peaking at 0 dBFS. And someday the Avid will be replaced with something better, and bad habits die hard.

BK: Shall we clarify that to say: An Analog to digital conversion sounds best as you approach max. level..... On the D/A side and mixing side, it's a bit more complex, and it is possible to remain clean.... well, please read my article part II. Otherwise you could never mix productions with reasonably wide dynamic range that are designed to be reproduced well (gotta sweat the soft stuff....).

MB: "I try to record with pre-mixing in mind. I have the playback levels all set at zero on the console (10dB down from the top of the fader.) I set my record levels so that I have a pretty good mix coming back with all faders in this ruler-straight line. If a guitar peaking at full scale digital is too loud, I can either pull down the record level or the monitor level.

BK: If you are talking about the A/D conversion (tracking) situation, this is not a good thing in my opinion, at least unless you are running state of the art 24 Bit A/D converters. A 16 bit record system really does like to be maximized, especially with the cheap converters built into most recorders. (Even if the chips are not cheap, the topology, clock leakage, grounding, power supply, shielding and jitter make it sound like crap). So, regardless of the apparent convenience of having all the mixing faders lined up, better to do it later on in the mixing side. I repeat: Anything else is bad practice (but see below for acceptable tolerances on the 24 bit side). And if your friend likes to have all his faders at the same physical level, and he's mixing digitally, better get a decently-designed floating point mixer. (I say decently-designed, because a well-known manufacturer has recently made a $10,000 floating point boat anchor, I'm sorry to say). In the floating point case you could turn down or up a "prefader attenuator" in order to make the mix fader hold at a fixed level, without sacrificing dynamic range of the floating point mixer. That puts your gain staging in the right place. But if it's an Avid or Pro Tools (fixed point), that's a very bad thing, and the signal into the mix faders should never be pre attenuated (too many truncated multiplies yields grainy, harsh, cold sound).

BK: There certainly is, from the point of view of the sound quality of the original recorded track, and I am saying particularly in the case of 16 bit recording. I'm sorry, he's wrong there.

BK: Depends on what you do with that data and how you gain stage it in the final. If that -40 dB represents a pianissimo passage that is going to remain that way and not be turned up, then you haven't lost any resolution. If instead that -40 dB represents the peak level you are putting on the 24 bit recorder, then you have lost almost 8 bits of resolution! Think of it this way, you are "exercising" fewer bits in the original recording by recording at a lower level. A simple example would be on my bitscope page. You will see fewer bits being exercised for a lower level tone. But I think this may be a matter of semantics. Roger knows that fewer bits are exercised and there is more noise. He is thinking of "resolution" as a constant of the medium. I prefer to think of resolution as the ratio between the absolute highest RECORDED level and the noise or quantization noise floor of the medium (including the A/D). If you later turn the level back up 60dB you will have the same resolution, but more noise (still down 96dB.)

BK: My definition of resolution has EVERYTHING to do with record level. You can have a high resolution tape recorder but if the highest peak of the entire program is -24 dBFS, then you have lost 4 bits of resolution which you will never get back. As I explain somewhere else on this page, if you don't have to alter the gain in post, then those lost bits are probably not meaningful. This is true whether the original recorder is 16 or 24 bit.

BK: I disagree with his stating that resolution is the same when the signal is closer to the noise. It may be a matter of definition, but it's not my definition. I also disagree with "then it doesn't matter if you record at the lower level and mix with the master fader up, or record at the higher level on each track and pull the master fader down. The final noise floor is the same."

BK: Actually, the final noise floor is a complex question. Let's assume a noiseless musical source for the moment and consider the final noise floor of the mix from the point of view of only that single track's mix fader. The final noise floor of the mix is thus a sum of the original noise floor of the original A/D converter AS IT HAS BEEN PLACED gainwise in the final mix, that is, depending on the gain of the mix fader. It also doesn't matter if it is a floating or fixed point mixer. Just consider the source A/D noise/distortion as an analog voltage which is being gain manipulated by the position of the mixer's fader. Now, if the original A/D and recorder was 24 bit, there is considerable leeway in how far you pushed the original track, absolutely, but it is still true if the original track was recorded rather low, and for esthetic reasons you must raise the level of the mix fader----then the level of the original quantization noise and distortion of of the original A/D conversion gets raised above the 24 bit noise point in the final mixdown. As an extreme example, if you recorded the original track 48 dB too low (below 0 dBFS peak) on the original multitrack, and you had to raise the mix fader 48 dB (this much gain is not available, but you get my point), then the output noise floor of the final mixdown will now be at 16 bit levels, even if the mixer is a 24 bit mixer. Again, this doesn't matter whether the digital mixer is fixed or floating point, for this particular point of discussion.

BK: To get slightly deeper into fixed or floating point, in a floating point mixer, you don't have to worry about gain staging. Gain staging "errors" do not (in practicality) cause additional quantization artifacts in a floating point mixer. But they do in a fixed point mixer, and if in the above example you have a multitrack which is recorded very low, then in addition to deteriorating the signal to noise of the mixdown when you raise the mix fader, you will also create additional quantization artifacts from all the fixed point multiplies you have to accomplish to re-gain stage the mixdown. Another big HOWEVER: If the fixed point mixing system is double precision, with a right shift of, say 4 bits (24 dB) going in, and 4 bits going out, then the gain staging/quantization error almost never occurs, and you can then stagger your faders, compressors, and so on, without creating a quantization problem. So, shall we say, single precision fixed point is a problem, but double precision fixed point sounds as good as 32 bit float, maybe as good a 40 bit float.

BK: The Yamaha O2R is 32 bit fixed point. It does not sound as good when you have to apply too much gain in ANY gain stage. The Yamaha's multiplies just don't sound that good, to my ears. Ask George Massenburg.... if he mixes with the Yamaha, the most he'll do is move the faders, he'll stay away from any other processing. Too many bad multiplies....If your friend is referring to the new Sony "mini Oxford", it's a 32 bit (or if I understand, it may be 40 bit) floating point, and that's a whole different ball game. Reportedly that new Sony is one helluva fine mixer. The mixer's precision is one thing, but the resolution of the original A/D's (recorders) is far more important to the results that will be achieved, and I hope that my paragraphs above illustrate why that a 16 bit original recorder should definitely be maximized to make the final result sound good.

Hope this helps,

Bob