'Hot' Recording Levels vs. SOUND QUALITY Experiment

Didn't mean to thread hi-jack. Just wanted to know that info b/c I figured people from different genres would be attempting this experiment and want to know if it was aimed more at one particular genre than others.

I figured most mastering and mixing tips would be universal but wanted other opinions. Maybe I will start a new thread after all:)
 
i think what massive and neilweight say's is coloured by their genre, but is based on the universal masetering principes. Those are not ment to get as much loudness, but for making the music sound good and as one piece. therefore exciters are not good, they compres the sound way to much and take every detail out of it.

i used to love the l3 also, but now i hate it, and i only use soft compression to amateur master my music. Now i want to buy a revox real to real to put my master on to use the tape compression as the only compression in my masterstuff. I heared an other producer i know doing it, and that's the right kind of compression for a master, not the l3 and similar style squeezing. But real mastering happens in good studio's with pro's who studied it for years and who have equipment you normally never can buy or handle correct.
 
The points made in the original post aren't even directly related to mastering - Although if followed properly, it will certainly give the M.E. the greatest flexibility with the project.

Getting good levels and proper headroom throughout the recording process is universal. It has absolutely no "down-side" at all. It's how "the pros" do it, regardless of specific genres.

Those that don't are usually either inexperienced or misinformed (or both).

It's not even a "technique" per-se... It's simply common sense. It's recording 101. It's probably the very first thing taught on the very first day of "recording class" after "INs" and "OUTs" are covered.

I don't track too much anymore, but when I do, it can be anything from a small chamber orchestra to a gothic Power Metal band. The levels are going to pretty much be the same - "okay" - Not too low, and definitely not too high. ALWAYS err on the side of too low. In 24-bit digital, it's ridiculous to do anything else.
 
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the whole "headroom"-thing is very important to undestand, but is no longer a problem since floating-point file formats are already in use. they will surely become professional standart in some years. they have an infinite headroom, they cannot clip! the perfect thing for ME's i think. :)
 
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To a point - But that's *downward* headroom in practice. If you take a signal in FP that's overshooting -0dBFS and then render it to a 24 or 16-bit file, there's gonna be trouble somewhere.

Funny this should come up - I'm working on a project at this very moment where this is a huge issue - The mixes were done using 64-bit FP rendering to 32-bit FP. The oversampling meters here are reading over 8dB of clipping. So, the same thing - I have to turn it down considerably just to satisfy the converters. The theoretical unlimited headroom doesn't translate to reocnstructive D-A distortion. I don't think it ever can actually - As in, I don't think it's phyically possible.

But I'm not an expert on it -
 
Massive you are right, no converters can do that, but who knows, this could be a small revolution...

i think it is a good "project"-format especially before the mastering process. 32bit and you don't worry about processing resolution, floating point and you don't worry about clipping and headroom.

life can be easy ;)
 
As I read the remarks about 16 bit I'm wondering now how MM's sound advice translates to 20 bit.

In our humble pre-production studio, we're in a situation where we will still have to live a while with a 'vintage' 02R V1. (I went on an odyssee searching for a V2 eprom which enables 24 bit I/O. Impossible to find). The chip on the ADAT cards on our 02R only allows for 20bit max. There's no way around it although even the old 02R can handle 24bit internally. The digital I/O is routed via lightpipe into the ADAT cards from a RME 9632. We do record in 24 bit though, coming from a 24 bit outboard converter straight into Cubase. Anyway, the last 4 bits will be chopped off by the 02R ADAT cards when mixing.

What would be the most sensible approach in terms of headroom for this sort of setup? Any hint most appreciated [apart from "get a 02R96" ;)]

This is a brilliant thread btw.

easy
B#
 
Even back when DAT was 16-bit only, 0dBVU was *still* -14dBFS. That was the whole point of digital - Not having to turn it up - Not having to eat up the headroom to establish and audible gain in quality.

Although it's certainly more important to stay in a good range with 16-bit and even 20-bit - Hell, I try to stay in a "good" range in 24-bit. I let the front end decide. If you use (most) gear as it was designed, your levels will almost automatically be in this "ideal" range around 0dBVU (-14dBFS). That's how it's all designed to work. Always was.
 
just to go back to the 32bit floating point part, massive is right in that this is really reserve room. it mainly allows mistakes to be made and not matter so long as they are corrected before fixing the point.
that being said, some of the benefits of 24bit, summing accuracy, rounding error reduction must also hold for 32bit.

its been my experience though that 32bit files can be problematic to open between different apps and this is the real draw back at the moment to it.

with the headroom being left with 24bit working done correctly, the floating point shouldnt become an issue in practice as you should never be near 0dbfs
 
neilwight said:
its been my experience though that 32bit files can be problematic to open between different apps and this is the real draw back at the moment to it.
Testify, brutha! :D
 
mmmmmm

32 bit float isn't the same as 24 bits. The Floating point concept is similar to the way mathematicians and scientists denote very large numbers as a mantissa with an exponent. For example, instead of writing 1,234,567,890,000,000,000 you could write 1.23456789 x10^18 The decimal number part is the mantissa, and the '10 to the power 18' part is the exponent.

In Floating point digital audio, 32 bits are used, with 24 as the mantissa and the other eight to provide the exponent. Hence the resolution of the wanted audio is always maintained at 24 bits (probably where you got the idea that the two formats were the same...), but with a scaling factor so that extremely large or very tiny numbers can be denoted without losing that 24 bit resolution. The theoretical dynamic range of a 32 bit Floating point system is about 1500dB!

And yes, you do need to reapply dither when you convert from 32bits (float or fixed) to 24 bits -- but this usually happens automatically in most systems.

Best to clear up any misconceptions about floating points.

24 bit converters rarely achieve much more than 21 or 22 bits of true resolution, but even so they allow us to restore much of the headroom that we were forced to give up with 16 bit systems. Remember -- good professional analogue mixers provides a nominal headroom of 24dB or more above the zero level, with a noise floor some 90 to 100dB below that. 16 bits couldn't match it, but 24 bits can easily.

What that means is that we can use 24 bit recorders like we used to use analogue equipment -- with HEADROOM.

The biggest advantage is that you don't need to compress on the way in. You might want to for an effect that you are happy to commit to at the recording stage, but that's a choice. In the days of analogue recorders, you had to compress on the way in because they had such poor (in modern terms) signal-noise performance.

Finally, a quick word on the rounding errors thing. Assuming a properly dithered converter, there are no rounding errors. The conversion is entirely linear. The theoretical rounding errors due to quantisation are transformed into random errors by the dither, resulting in a noise floor (although you can still encode and recover signals at levels well below the average noise floor level).
Just like in analogue systems.

Just thought that this would clarify any debates about 16/24 bits and FPs.

Hope it didn't bore too many of you.
;)
 
MASSIVE Mastering said:
If you use (most) gear as it was designed, your levels will almost automatically be in this "ideal" range around 0dBVU (-14dBFS). That's how it's all designed to work. Always was. [/B]
Cheers mate, point taken but try to explain that to a vocalist ;)

I prefer using no compression at all on vox at tracking so I try keeping 4-8 db headroom below 0dbVU on the preamp, depending on the vocalist and the song. My math is rusty but it'll obviously translate to something way below -14dbfs. Coming from the Amek straight into Cubase 24 bit digital, the increased headroom shouldn't be too much of a problem (if I understood well your initial post). Now at mixdown through the 02R at 20bit, this 'cold' signal will probably end up somewhere around 14 bit which doesn't look quite appealing to me. I made a few tests this way and at first listen it seems like the vox and other miked instruments lose some depth and don't do justice to the Gefell mic and the Amek pre.

Now the question is where would I compromise more in the end - by running hotter levels staying as close to the max resolution as possible or by sticking to -14dbfs and compressing at tracking (which would be sometimes quite drastic with certain vocalists and compromise the sound up front)? I suppose a relatively hot track level at 24 bit can always be bounced down to the magic -14dbfs without risking anything...?

Or last possibility, I get even more confused with bit rate conversions, eventually dump the good old 02R along with all the 16bit gear catching dust in the racks, record everything in 32bit FP
and let others do the mixdowns... :D

l8rz
B#
 
dag nam it,
why dont i ever think to describe 32bit floating point in that way. i always end up with drawings of 1s and 0s in an attempt to illustrate. just saying it works with bits assigned similar to a "to the power of" in maths is much simpler and something people can follow.

what about rounding errors caused through processing and DAW operations, fading, panning etc etc?
it was my understanding that these will all appear, theres no internal dithering of each calculation performed and during summing to cover these.
if this is the case then while these will be addressed as a total sum and then through final dithering/truncation theres plenty of scope for them to compound during the mixing process itself.

with 24bit this is much less of an issue as its so far down the stream as to be tiny, with 16bit you are hitting right into CD range immediately.
 
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R2B - While you're in 24-bits, your *peaks* need to go *below -48dBFS* before you're even down to a 16-bit signal - So if you're a few dB below 0dB(VU), it really isn't anything to worry about. Even with the 20-bit final, you still have the ability to turn it up.

On top of that, the average Neve preamp has anywhere between 20 & 26 dB of headroom already (clean to over +20) so you really don't even have to leave that much headroom on the way into the preamp.

Not that it would be wrong if you did - I've just always found Neve & Amek preamps to get nice & "schveet" sounding a whisker hovering around the "0" mark...
 
MASSIVE Mastering said:
R2B - While you're in 24-bits, your *peaks* need to go *below -48dBFS* before you're even down to a 16-bit signal - So if you're a few dB below 0dB(VU), it really isn't anything to worry about. Even with the 20-bit final, you still have the ability to turn it up.

On top of that, the average Neve preamp has anywhere between 20 & 26 dB of headroom already (clean to over +20) so you really don't even have to leave that much headroom on the way into the preamp.

Not that it would be wrong if you did - I've just always found Neve & Amek preamps to get nice & "schveet" sounding a whisker hovering around the "0" mark...
Nice one mate.
I realise I'm probably stuck in my old ways tracking with the old 16 bit ADAT where I never worried about the 20bit in the 02R (or any bit rates)... always tried to keep the peaks around -10 dbFS or more.

Our only 'commercial' activity consists of tracking vocalists, some instrumentalists and some pre-prod MIDI stuff, so I will apply your 24bit advice and ignore the 02R issues for now. The little mixing and 'full production' we do is mainly for our own material so there's plenty of time to experiment.

cheers
B#

ps. oh and this thread was of great help indeed
 
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You can do 24 bits on the 02r. The drawback would be that you'd be limited to half your channels. The odd channel would do 16 bits and the even channel would do the upper 8 bits of data. The channel would still operate as 1 channel but would actually use 2.
 
sleepy said:
You can do 24 bits on the 02r. The drawback would be that you'd be limited to half your channels. The odd channel would do 16 bits and the even channel would do the upper 8 bits of data. The channel would still operate as 1 channel but would actually use 2.
Hi sleepy, this only works with version 2. We've got v.1. As I mentioned before, it's next to impossible to find. We had missed the boat back when it came out. But hey, if somebody on here can hook me up for the eproms... ;)
 
I'll usually leave around -4dbFS(dbFS is what I see on the stereo output on cubase SX right?) in my mix when I'm done. Is this good? or should I leave more?

At the mastering stage when I multiband compress, should I see like 1-2db of reduction or like a lot more?
 
if you are working with 24 bit then -4dbfs is fine, more is also ok too.

as for compression when you self master, 1-2db is fine if its 1-2db under compressed. there is no standard for what you should see, it depends on what the material requires and each time its different.

id try to avoid using multibands though IMHO.
 
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