So Stav has said record hot, Jeff has come along and said that Stav now refutes this, I haven't seen this article so Jeff if you can give roughly what he said, issue AT Mag and pg number that would be nice.
 
Paul Frindle as Jeff or deering amps has pointed out started this idea, it is not wrong as deering amps has said, it just doesn't do anything because all we are doing is mixing as we would in analouge land.
 
But if you read through the whole thread that Frindle has started regarding Digital Myths which is here: http://www.gearslutz.com/board/music-computers/542885-paul-frindle-truth-myth.html (I haven't I've only got to page about 8, still reading) he contradicts himself, on about pg 3 or 4 he starts to talk about how digital converters can add pleasent digital harmonics and emulate analouge euipment in the that way if you drive a peice of analouge euipment these 2nd harmonics are added if it's tubes and 3rd harmonics if its transitor base.
 
I hope everyone here understands fundmental frequency's and when you drive a peice of equipment it multiplies the fundmental frequency by two for tubes and the third if its transistors.  My understanding is if your guitar has a fundmental frequency of say 250hz you drive a tube device we will end up with harmonic distortion at 500hz and 750hz for transistors.
 
Now this is where is get's tricky and sticky!!
 
We have established that our 24 bit audio interface's are not 24 bit, never have been some are 20bit, some are 18 bit and some may be 22bit.
 
And as Bitflipper has pointed out the last 4 bits of your converter's are useless, this is where all the noise, quantization and dithering goes on.
 
The question then remains, a) where does the harmonic distortion sit in terms of volume?, is it in those final 4 bits if we don't record a little hotter than we normally would?
 
And b) if digital converters can mimic analouge gear and add some form of digital harmonic distortion, recording slighty hotter has some benifits?
 
Hmm a lot to ponder!!
 
Neb   

Ben,

no.  24 bit interfaces are 24 bit - that is what they deliver to your computer.  They aren't perfect, of course, and have a maximum dB of 100+ to 120.  But neither are 16 bit interfaces delivering a full 96 dB, unless they are 24 bit.

So the full 144 dB can't be reached, and the signal at a very low level dissappears into noise - which is just about inaudible.  If you are recording a gnat's fart you won't hear it, but for any practical sound at less than an ear splitting level, you won't hear anything and your recorded signal probably disappears into the low level noise of your system - including amp noise.  For all practical puposes, a good ADDA interface specs out better than your ears.  Maybe not yours, but mine for sure.  What is the lowest dB sound we can hear -- 20 dB or so?  I used to know, but it has got lost in the noise of my fading memory.  The first 4 bits (the noise) is around 24 dB.  And the loudest, I think, is around 130 dB or so, where the ear starts shutting down to protect itself.

Digital converters don't mimic analog gear - they contain analog gear before the converter chips.  You can saturate those, tho most average units don't respond well.  Some transformer-based converters do, like burl, UA and JFC.  I know professionals that swear by them and find the analog part mimics tape.  But they also cost about $1000 a channel.  Most of us use "clean" (and cheaper) transistor ADDA units not driven so hard, since it don't sound so good.  Feel free to, tho, since it is art.  Digital harmonic distortion isn't harmonic but simply noise where the signal is greater than the ability of the numbers to represent it.  It clips, like transisters.  Again, you may like it and call it art.

You can get a rich, not abrupt saturation, by using a nice (read expensive) front end unit before your realitively cheap converter.  And again coming out to your analog transducer.  That is what I call art.  Or at least good sound.

@

Don't forget I'm just interpreting and evaluating this Paul Frindle charecter and what he is saying he is suppose to be the doyen of digital.

Have you read any of the thread, there is some other interestng stuff in what he is saying and all very controversial, esp the stuff about emulating analouge gear.

So I suggest everyone have a look through the aformentioned thread, I've still got to read some more of it myself, I mean it's 18 pages longs.

So if people put forth people like Paul Frindle as a doyen and still say what he is saying is wrong, how are to arrive at any sort of truth!!

Neb 

Awesome link thanks Ben.

That Paul Frindle guy coded some of the Sonnox plugins I believe.

That is an old thread I remember reading.

What he says (more succiently than me) is some of what I said.  A 16 bit system has a signal to noise ration of 93 dB.  96 dB maximum level and 3 dB of noise.  More than can be used - he makes the point that a typical orchestra  has a dB range of 60 - 80 dB.

And I think you have digital distortion confused w/ analog distortion.  Most people will say that digital distortion sounds bad - it sounds like clipping with a flat top on a graph.

Frindle does say that digital can replicate analog saturation/distorition - depending upon how good the coder is.  Which means some coding doesn't emulate it well.  Another thing to bear in mind is he is a coder.  Even dismissing the money part, a coder will have to believe he/she can write code that will mimic analog saturation well enough to use, or what is the point of their work.  It is like writing music that you think sux - why bother?

Theoretically you should be able to emulate anything w/ code - it is just math.  That is one of his points.  That doesn't mean anyone has nailed it, despite marketing hype (and not by Frindle as far as I know).  Digital has gotten better, no doubt.  IMHO, it is easier to use good analog to get the sound before it is turned into digital than adding it later.  You can, of course, and I do.  But just because someone designs a de-noiser bit of software doesn't mean I have to record everything overdriven and apply the effect later, hoping it gets rid of the "bad" part of the sound and not touching the "good," keeper part of the sound.  For the same reason I don't use a crappy little telephone mic to capture sound and hope that a convolution engine will make it sound like a vintage C12.  You can't restore what ain't there or fix all noise.

For what it is worth, I know the Stienberg/Yama RND comp and EQ are close to the hardware.  I've had the software for testing/review and have the hardware.  The Cake PC tools are good, very good, too.  The EQ sounds a about as good as the RND hardware.  Not the same but within reason.  And the comps are good.  And it is a hell of a lot easier to switch out comps in SONAR (even in the FX chain for the Sonitus comp which I still use a lot) than hardware 1176 or SSLs or an LA2A.  And cheaper.  If I have the hardware available I'll use it, but here at home I'm happy using the software for mixing.  But I still have some good hardware.

@

Ben the article you are interested in was in Issue 80 April 2011 Audio Technology. The front cover has a title 'The Low End'   Very good article as to why most PA's have too much bottom end. Very interesting article! and I totall agree. Most live sound engineers go way over with the low end. They are deaf to it.

The article you are interested in is called 'Too Low for Zero' by Jan Muths. This is not Stav refuting his original idea back in 2003 in his book. That was an error in your assumption and I never said or meant that so sorry. But it is obviously a much later article and it generally proves that recording lower is very good. And remember it is 8 years later than Stav's original idea.

If you cannot find it let me know and I can scan it and email it to you. It is only three pages.

Back around 2000 I was using an Emagic Audiowerk 8 sound card and although the converters are 18 bit, the specs say digitally the card can only go into 16 bit mode. I think many sound cards were like that at the time or the 24 bit ones were certainly not common.
Stav was suggesting we push the levels a little hotter then. I have got his email address, I could ask him what he thinks now about the situation.

But now of course 24 bit is standard for any sound card these days which is a good thing of course. It means we can go into that mode whenever we want to. All we need to do is create a session at that bit depth. If you have got a digital mixer you need to be able to put that into that res too and the Yamaha can of course. (Latest model handles all the sampling rates too which is even cooler!)

"Paul Frindle as Jeff or deering amps has pointed out started this idea, it is not wrong as deering amps has said, it just doesn't do anything because all we are doing is mixing as we would in analouge land"
Ben, I'm confused as to the point you are making here?

"b) if digital converters can mimic analouge gear and add some form of digital harmonic distortion, recording slighty hotter has some benifits?"
That is a pretty big IF there Ben.
Granted on page 9 (post #259) of Paul's thread he states:
"Then it transpired that some mastering engineers were actually doing this so they could use the ADCs as signal clippers - and increase the modulation levels (a secret weapon in the loudness wars)!
So what followed was scores of posts about the ways in which converters may (or may not) distort under these conditions - and then we find out that converter products are actually being selected depending on how they sound when deliberately over driven. At which point all bets are off - and any technical discussion becomes pointless :-("
But note the "unhappy" face at the end, Paul is NOT advocating "clipping" the DACs.
Certainly the distortion artifacts produced by the converters in my Tascam FW-1884 and my RME UFX are not, to my ears at least, "pleasant" sounding.
Not "tubey" by any stretch.
The most important concept I take from Paul's posts on GS, in the thread we are discussing, and the one I linked here is:
LOWER YOUR LEVELS!
Bring them in around -20, keep your "mix" peaks below -6 (-10 is better) and let the "Masters" MASTER; meaning get it to 0dBFS for CD (or mp3) replication.
There is no need to "PEG" the meters.
Its not "good" practice to "hear" the needle bouncing off the end pin on an analog desk.
Nor is it "good" practice for the meters to "glow" red in the DAW.
YMMV
Tom

Now this might BE controversial:
"I hope this helps - because the subject of float versus fixed has gone round and round for some while - because it's difficult to grasp.
The most 'controversial' conclusion for people from all of the above, is that for the most part if you are dealing with audio audio data of say 32bits wide (cos that's how wide your processing data buss is), you are significantly better off to do this in a fixed point representation than float."
That's Paul from page 10, post #297.
Have fun children...

T

Well Ben here is your "Smoking Gun"
pg 15 post #430
Q "So, to clarify, there is no difference, audio quality wise, between hitting your input converters soft or hard, e.g. with maximum peaks at -18dbfs and no gain change at the channel head, compared to -1dbfs with a 17db level cut at the channel head?
The reason I ask is that I think I've heard someone say that hitting you AD converter hard can degrade the audio quality a little bit."
A) "That depends entirely on the ADC design, quality and performance - and indeed whether it deliberately distorts to avoid clipping :-(.
Whilst it is true that hitting the ADC hard will increase signal to noise ratio, there's also the risk that it may produce more harmonic distortion - or even overloads sawing off your peaks (or other stuff which is more complex)."
I admit defeat! Ben will NOT want to hear the rest of the answer, but others might...
"If you can't test accurately what your ADC actually does (very difficult) then my advice is to aim for -6dB peak values out of your converter as a reasonable safety compromise (only losing 6dB SNR) - and then lose perhaps another 6dB at the head of your DAW channels - giving you total headroom of around 12dB for your mix and processes."

What was it Danny said in the other thread about "peak" values on his tracks?

But again, I admit DEFEAT...

T

DeeringAmps


Well Ben here is your "Smoking Gun"
pg 15 post #430
 The reason I ask is that I think I've heard someone say that hitting you AD converter hard can degrade the audio quality a little bit." 
A) "That depends entirely on the ADC design, quality and performance - and indeed whether it deliberately distorts to avoid clipping :-(.
 Whilst it is true that hitting the ADC hard will increase signal to noise ratio, there's also the risk that it may produce more harmonic distortion  

And to clarify further this specific piece - they're talking here about analog distortion produced on the analog side of the ADC before you reach digital clipping.

And as discussed before, increasing the S/N ratio isn't really relevant if the quantization noise from the ADC is more than a little below the noise already present in your signal - you just end up a raising the noise floor by a fraction of a dB.