Ok now to combine two threads condense and conclude!!

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.

DeeringAmps


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

It wasn't about winning mate, it was about answers and Danny and I both were of the same mind -6db peak is a good place to be although I am not adverse to peaks as high as -3db because I'm a raceing car driver and I like to skirt the edges a bit plus even if you do clip as long as it's just a click I know a trick or two to disguise it.
 
Ok so we get this straight, this had nothing to do with right or wrong, I had made some claims and I was told to shut up and go away and poop, so I decided to put up.
 
I was shot down again but it would seem even though the myth was debunkend, there is some truth as well, The Amazing World of Digital.
 
What has come of this is we have an optimum operating level for our digital recording and mediums.
 
We have an idea that we should think of our DAW's as analouge mixing desks and not be afraid to use the trim, this is something that I wanted to put out there as well, not understanding this concept of gain structure, even now in the audio courses I did and when I posted on another forum the idea of using trim, the answer was Huh and this was by the supposed "pros"
 
So I hope you all understood this was nothing to do with right or wrong but an exercise in working together to find some answers and we found answers, I know I am a bull in a china shop but we need bulls and we need pussy cats, if the world was only populated with bulls well you know how that would end and sometimes pussycats turn out to be tigers.
 
Neb

weather you guys are trying to revial the truth or find the ever so elusive invediable,this thread is really good and educating,thanks for the read guys.

seriously,

very intersting stuff.

Ben in case you don't get onto that article here is the jist of it in point form.

*     As I said there were not a lot of 24 bit interfaces around at the time of Stav's approach to recording higher for 16 bit. They are all 24 bit now, hence the reason we can use 24 bit so easily.

*     There is analog electronics even inside your A to D converters right before the converter part and the noise floor of those stages will be worse than the 24 bit digital noise floor.

*     Not all the bits of a 24 bit system are used. In the article it states more like 21 bits out of the 24 are being used.

*     Something that is not considered is that every time you add a track to a multitrack session the digital noise floor is in fact rising slightly. For one  or two tracks it is not a biggie but when you get into 32 tracks for example in 16 bit mode the noise floor comes up.

With 4 db of headroom 32 channels will create a noise floor of around -64 dB. (it gets 6 db worse as you double the tracks so for a 64 track mix the digital noise floor is -58dB) Good news is in 24 bit mode a 32 channel mix is creating a total digital noise floor of way down at -112 dB! MUCH better than in 16 bit mode. Where 24 bit excels is when you have large numbers of tracks.

*     Final conclusion is recording down at -20 dB rms is perfectly fine and very good. There is no improvement in sound quality or nothing to be gained sound wise by recording hotter.

The whole concept of thinking like the old analog days is simply right and excellent! Just create your rms levels down at K-20 for example and treat the whole digital system like an analog system. Hence the reason why the K system approach works so well. You still need some sort of VU meter to show you 0dB VU when you are down at -20 dB FS. Peak meters are also important for transient signals but VU's are required if you in fact want to operate a digital system in an analog way.  (that is what SSL tell us too with the AWS948)

Jeff Evans


*     Something that is not considered is that every time you add a track to a multitrack session the digital noise floor is in fact rising slightly. For one  or two tracks it is not a biggie but when you get into 32 tracks for example in 16 bit mode the noise floor comes up.

With 4 db of headroom 32 channels will create a noise floor of around -64 dB. (it gets 6 db worse as you double the tracks so for a 64 track mix the digital noise floor is -58dB) Good news is in 24 bit mode a 32 channel mix is creating a total digital noise floor of way down at -112 dB! MUCH better than in 16 bit mode. Where 24 bit excels is when you have large numbers of tracks.

 

This part sounds highly questionable (and that's being generous).

Jeff and Drew wanting to have the last word are muddying the water, what has come of this is two or three importent things:

1) Digital mediums and recording devices do have an optimum operating level, in this case both danny, myself and this paul frindle agree -6db, although I have said the odd peak of -3db is alright.  And I mean odd!!

2) The mixer in your DAW's should be treated like an analouge mixer, USE THE TRIM, it is your friend, here is a diagram of how much 32 tracks of summed audio creates http://www.prosoundweb.com/article/setting_sound_system_and_mixing_console_gain_staging/av/P3/ what this means is even before you start mixing and if you have all faders at unity you are already 15db over 0dbfs, so if you are going to mix 32 tracks turn the main buss trim down so you have headroom.  Because we all know digital has no headroom so, you have to create it.

3) Even though it still seems to be a point of conjecture, somewhere in the digital signal processing chain, harmonic distortion happens, so if this is the case listen for it when setting levels as it goes into the DAW.

As for the other stuff, it's all well and good but even I am struggling with it, I think the three above points are the most importent ideas that we have gained out of this disscusion.

And as I have pointed out before, it took them over 29 years to figure analouge out.

From the moment Bing and Les Paul, using stolen German technology figured out mulit track tape recordings and did the first home recordings in 1946, to the epitomy of multi-track tape recording or to put it plainly when tape could go no further.  Which in my opinion is Bohemian Rhapsody, this is 1975.

Funnily enough the first digital recorder appeared on the market in 1977, If I remember correctly, so 35 years to get to the point where we are now, hmm funny about the similar timeframe.

The problem with digital is the goal posts keep moving, from the 13 bit digital recorder of 1977, to 16 bit in the mid-80's finally the not so 24 bit of now.

But we now I believe have three constants that we can say yes to, and reiterate, that digital has an optimum operating level, that digital mixing should be treated the same as analouge mixing and that somwhere in our little boxes somewhere in the signal chain, some form of harmonic distortion is happening and if it's pleasent aim for it.

Neb 

BenMMusTech


Jeff and Drew wanting to have the last word are muddying the water, what has come of this is two or three importent things:

1) Digital mediums and recording devices do have an optimum operating level, in this case both danny, myself and this paul frindle agree -6db, although I have said the odd peak of -3db is alright.  And I mean odd!!

As I said before (repeatedly), anywhere the quantization level is reasonably far below the noise floor and you aren't clipping is "optimal".

We can go through it all again if you'd like.

One-man wolf pack 

drewfx1

BenMMusTech


Jeff and Drew wanting to have the last word are muddying the water, what has come of this is two or three importent things:

1) Digital mediums and recording devices do have an optimum operating level, in this case both danny, myself and this paul frindle agree -6db, although I have said the odd peak of -3db is alright.  And I mean odd!!

As I said before (repeatedly), anywhere the quantization level is reasonably far below the noise floor and you aren't clipping is "optimal".

We can go through it all again if you'd like.

Then why this:
 
"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."
 
And what about the prospect of the mere posibility of harmonic distrotion in you black box!!! 
 
I think you might have the science correct but I think the science is still pissing in the wind, 29 years to figure out tape and after 35 years we are still trying to figure out digital and the possibility's.
 
The three points I have made have nothing to do with the maths, screw the maths, it has to do with three people's opinion that there is an optimum operating level for digtal and this may tie in with this idea of harmonic distortion.
 
So even if the harmonic distortion is before the converter, at -6db we are going to hopefully pick some of that extra harmonic distortion up.
 
Sorry drew I am going with my ears and heart on this one, you may get the maths but something else is going on.  That ramdom factor that even maths most of the time can't explain.
 
Neb 

drewfx1 I am only quoting the article in question and why would 32 tracks not add some noise like they used to in the analog days with each one contributing some level of hiss. Except in the digital world the noise or quantisation error noise is much lower but why would it not add up. In 16 bit mode 64 tracks could be considered noisy. (assuming they are all up around unity and in the mix)

Analog does tend to have optimum levels so we need to be aware of them and set our levels there. Ben  is wrong however stating that digital has an optimum level, it does not.  As long as you are not down at some very low level eg -60 dB or so what I have said is simply this, you can record at quite a wide range of levels digitally and there will no change to the sound quality. My ears and my heart are telling me this too! (there will be variances in rms levels hence some tracks will sound louder than others though so it might be wise to concentrate on consistent rms instead of peak levels)

And when any of you say record up to -6db, record what at -6db! Are you referring to a peak. Because you certainly could not track at -6dB rms could you. So if a transient makes it all the way up to -6 db then the rms part of that signal could easily be down at -14 dB or even -20 dB rms.

As I have said before forget peaks and keep track of rms levels all the way through a production and keep them at a similar level. (in my case it is K -14 dB only because my Yamaha digital mixer is calibrated there) Worry about the rms levels and the peaks will all vary above any rms level and they tend to take care of themselves.

As long as you are a nice level eg -20 dB rms then you never really have to worry about peaks because rarely will a track have a peak that is greater than 20 dB above its rms level.

That is one of the problems of digital compared to analog. Analog does have optimum levels and reference levels and we have all worked with them but digital does not and it is all over the place level wise so we are trying to give them some sort of consistecy. K system does this well. Sound quality wise I say it again there is no real optimum level for digital, other than being so low we are down in the digital noise or so high we are smashing into 0dB FS. If you keep away from those extremes you will be fine.

Jeff Evans


drewfx1 I am only quoting the article in question and why would 32 tracks not add some noise like they used to in the analog days with each one contributing some level of hiss. Except in the digital world the noise or quantisation error noise is much lower but why would it not add up. In 16 bit mode 64 tracks could be considered noisy. (assuming they are all up aound unity and in the mix)

Analog does tend to have optimum levels so we need to be aware of them and set our levels there. Ben  is wrong however stating that digital has an optimum level, it does not.  As long as you are not down at some very low level eg -60 dB or so what I have said is simply this, you can record at quite a wide range of levels digitally and there will no change to the sound quality. (there will be variances in rms levels hence some tracks will sound louder than others though so it might be wise to concentrate on consistent rms instead of peak levels)

And when any of you say record up to -6db, record what at -6db! Are you referring to a peak. Because you certainly could not track at -6dB rms could you. So if a transient makes it all the way up to -6 db then the rms part of that signal could easily be down at -14 dB or even -20 dB rms.

As I have said before forget peaks and keep track of rms levels all the way through a production and keep them at a similar level. (in my case it is K -14 dB only because my Yamaha digital mixer is calibrated there) Worry about the rms levels and the peaks will all vary above any rms level and they tend to take care of themselves.

As long as you are a nice level eg -20 dB rms then you never really have to worry about peaks because rarely will a track have a peak that is greater than 20 dB above its rms level.

That is one of the problems of digital compared to analog. Analog does have optimum levels and reference levels and we have all worked with them but digital does not and it is all over the place level wise so we are trying to give them some sort of consistecy. K system does this well. Sound quality wise I say it again there is no real optimum level for digital, other than being so low we are down in the digital noise or so high we are smashing into 0dB FS. If you keep away from those extremes you will be fine.

Jeff of course we are talking about -6db peak not RMS do you think that I am that unskilled and mad!!!
 
-6db peak in Sonar equates to -20 to -16RMS, And how can you go against the tide of opinion, including Danny and Paul Frindle -6db peak seems to be the optimum operating level of digital, I of course push it a little harder, sometimes, also I use outboard gear and MOTU has compression and EQ built into every channel I use these devices to hit the optimum operating level.
 
Man stop muddying the water, Drew and Jeff, I think we have found three points, those of us who are less experienced can use.
 
Lets leave it at that!!
 
Neb

BenMMusTech

 
"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."
 

Don't see the word "optimal" there anywhere. But it cracks me up when people rail against math and theory and then use a purely theoretical (i.e. not real world) SNR improvement to justify something. 

And what about the prospect of the mere posibility of harmonic distrotion in you black box!!! 

This is assuming you want to use your converters as a distortion generator rather than providing the cleanest possible signal. Doesn't make any sense to me to use an ADC as a distortion box - because if you want a little more distortion you risk clipping, which you don't want. If you want to add distortion to your signal, why not just use a preamp or something designed to add it without having to risk clipping? Lots of analog gear out there that adds various forms of distortion with much more control than an ADC might.

Jeff Evans


And when any of you say record up to -6db, record what at -6db! Are you referring to a peak. Because you certainly could not track at -6dB rms could you. So if a transient makes it all the way up to -6 db then the rms part of that signal could easily be down at -14 dB or even -20 dB rms. 

Hi Jeff. I can't speak for everyone, but for me, I fire up my input levels and play the song I am recording. I don't play lead guitar riffs or riffs in different keys that aren't in this song I'm recording, I stick to the exact song and chords/progression I'll be playing.
 
I have my input signal set so that it peaks at -6dB. I used to have it dead on -6dB and of course other notes/chords or what have you, would go over that and hit anywhere from -4dB to -2dB peak. I never had any problems doing it that way, but I've switched things to a final of -6dB peak and never go any hotter.
 
Whether this is right or wrong, messes with the science or whatever....it's where I feel the most confident with my personal recordings as well as those I record here. I also always run a light outboard compressor on each track so that it helps me to stay at -6dB without over-compressing. The signal is just conditioned enough to stay where it needs to be.
 
As for the rest of this thread...and I mean no one any disrespect that has posted in it, but it blows me away that such a conversation could even take place over something as simple as a meter and the signal that is fed into it. I turn them on, get a good signal that sounds good to my ears and never look at the meter again unless I hear a clipping sound somewhere or when I export my audio. It baffles my mind that individuals that are 1000% more intelligent than me would even spend more than a minute looking into something like this.
 
It's like....(and this is why the science part of the audio world just doesn't do anything for me) let's say I'm working with a guitar tone. I play, I set my amp or my pre-amp to where this sound is as good as it can be, set the mic where it sounds the best or run a DI, run both...whatever works. I listen for things that stick out to me as possibly being problematic. Ok, in this song, I'm using a good amount of gain. I notice that when I ride or chug on an A chord, my meters ramp up.

I either compress a little more to keep that in check or I lower my signal so that the A doesn't ramp me up too much and I can compress it more later after it's been recorded. I make sure I'm not clipping and I make sure the sound doesn't sound too stressed or too weak in signal. When I like what I got, I press record. I'm not worrying about K-systems, crushing my signal near 0dB or having too weak of a signal. I just go with it and always wind up with something that is presentable. I just can't see how or why we'd even worry about the rest of this stuff...but that's just me.
 
-Danny

Jeff Evans


drewfx1 I am only quoting the article in question and why would 32 tracks not add some noise like they used to in the analog days with each one contributing some level of hiss. Except in the digital world the noise or quantisation error noise is much lower but why would it not add up. In 16 bit mode 64 tracks could be considered noisy. (assuming they are all up around unity and in the mix)

But if you add 2 tracks the overall level of your signal goes up so you have to turn down to compensate. So the only way the noise floor goes up is if the noise adds faster than your actual signal. Remember that you don't lose S/N ratio when you turn down under floating point (unlike analog).

And the +6dB number is peak, not RMS (which I think you'll agree is the more important number). White noise adds at +3dB RMS/+6dB peak. And unless your signal is largely out of phase, it will generally increase by +3dB RMS or more. 




The only other issue is calculation errors, but the numbers quoted are much higher than what you'd get with Sonar using even the 32bit bit single precision.



So there's either some context missing or something else wrong there.

I am interested in the statement that a peak of -6dB rms equates to 20 or -16 dB rms. Where did you get that info from? Yes, if the peak value happens to be 14 db say above its rms value as could be the case with a heavily slapped funk bass etc. But what if a sound comes along that only has a little peak say 3 db above its rms value.? What then. It means that the peak is still showing -6db but now the rms component is only down at -9dB a far cry from -20 or -16 as you put it Ben.

That is what is wrong with continuously monitoring peak values and nothing else. It is the rms values that need to be down around our ref level and keep them all the similar value and some peaks might be 3 db above that rms level and another sound might have a peak value 15 db above its rms level.

That is what is totally wrong with your (and many others too) approach Ben. Your theory only works if all sounds have the same rms to peak ratio and they certainly do not. My concept works best because it is about keeping rms levels constant and the success of it is not dependent on the sound itself, it works because you are doing something else (keeping rms levels constant) that is not related to the sounds rms to peak ratio.

You are muddying the waters Ben. I am basing my concepts on proven approaches which do work and I have used in many productions over the years. That is what is great about it, applying a sort of analog approach to digital. Choosing a ref level and working there! There are many as I have pointed out from our SSL training anything from -24 to -18 are common ones too. (Pro Tools HD interfaces are set at -18 dB FS as the ref level but they can be tweaked at the rear) At least the -20 dB rms is a standard that is used in the film industry so it is one level we can take from that and apply it to music production. Our new SSL mixer came calibrated to -20 dB which is interesting too.

Do you agree Ben that we need to be parking rms levels at some point anywhere between -18 to say -24 in the SSL case. Because if you do (and I get the impression you are starting to see the light) that is far cry now from the concept that you started out with saying we have to slam everything as high as we can!

drew good point about bringing tracks down to compensate, of course you are right. But what about a situation where all the tracks are actually at unity gain and no one on any track plays a sound at the same and each musician has a very quiet instrument eg a Vietnamese zither being plucked very quietly. (This is a soft sound I can assure you!) Imagine 32 tracks of this but no one is overlapping anyone else. Would you hear all that 16 bit noise building up, maybe! (BTW this article also mentions that any analog gear in this path is going to have a worse noise floor than digital at 24 bit so it is in fact the analog stages noise that you will hear building up) But yes under normal conditions it would not be an issue because this type of situation is pretty rare. But if you had to do it you would at least choose 24 bit then wouldn't you.

Danny I have just read your post. Yes I agree man! The reason what you do works is simply because digital does sound great and records really well over a very wide variety of levels and this is in direct contrast to what Ben is saying, he is saying there is only one special place for digital levels which is rubbish. Your work very well proves it in itself. No you don't have to get all crazy about K system and if you are a good mixer then you will still always get a good mix! But K metering is good and it does work and I believe even if you applied it Danny you would find your track levels would be very consistent and buss and final mix levels would also be the case. It does work and it does help. It makes mastering easier and more consistent too.

Ok Jeff, you have your Peak meter and your RMS meter going always don't you?.  So when you hit -6db peak it generally hits -18db RMS.  Simple!!, then to get the headroom right because in contrary to what you are saying, I was not saying that there is one magic spot, I'm with danny again and I can hit between -6db & -3db which would equate to an average of -16db RMS, I'm pulling these figures off the top off my head.

Now for mixing after I have tracked, I want an even mixing field and will make sure if I am going to process the audio signal, I will make sure I have -6db peak of headroom, equating on average according to the Sonar meter to minus 18db RMS.  Have you got it yet Jeff, I'm actually doing what you are doing except I am saying there is a sweet spot when tracking due to a combanation of factors, thats is the only difference to what you and I are saying.

Before I even start mixing, I tend to give myself up to 10db of headroom on the master buss.

Drew I am only railing against it because there is something random going on, something esoteric, something Jeff and you don't seem to understand, it may be unpleasant this harmonic distortion!! it may be pleasant but its there and its is random.

So whilst your maths is correct, your not taken into account some of the random elements going on, the majik if you will.

Danny thanks for the compliment, you were talking about me and being a 1000 times more intelligent
 
So lets get this straight, Jeff and I do the same thing we aim for a certain spot in terms of peak and RMS when tracking, we disagree as does Drew that there is a majik spot where some form of harmonic distortion happens!!
 
Danny and I do the same thing we are both happy to hit peak levels of -6db with a bit of room above!!!
 
Finally I like your signature Jeff, thanks for the compliment I'm talking about ideas, so by proxy and by way of your signature, I am a great mind

I don't agree that digital recording introduces harmonic distortion, that is bollocks! And it is not random either because it simply not there to start with. Look at the harmonic distortion figures of a typical digital recording setup and you will see it is almost immeasurably low. Like .001% maybe not sure. Wow and fluttewr is low too, noise is also low. These are things that are good about digital. So Ben as long we are well away from the extrems of digital then harmonic distortion is going to be very low or non existent. Sure if you bury the signal right down at -90 or something like that in 16 bit mode then you may hear something or if we are smashing into 0dB FS then it may be present then too but we are not there are we.

I want to mention that I am always talking about no processing on the way in to digital recording in terms of peak/rms ratios etc. Danny if I measured all the rms levels of one of your muktitrack sessions I would probably find they vary but the peaks would be very consistent but with me it is the other way around. You are compressing too at times on the way in which would be bringing rms and peak levels up to near each other. Heavily distorted guitar cabs are very compressed anyway so peak and rms levels could almost be the same! Which means if you are tracking and hitting -6db some of your guitar tracks could be pretty hot but you would pull your track faders down obviously to compensate. But with me I would still track those slammin cabs at say K-14. My track faders would be higher to achieve the same mix and same sound in the end.

Both ways work and many people here use the peak concept more consistent. That is a sort of digital thing. I like to keep rms levels constant and this is an analog thing being brought over from an era past into a modern digital approach and it also seems to work too.

You always want to have the last word, when I have tracked I too set the meters to around -18RMS, it as you say we do things diferently for insatance guitar cabinets are so last century LOL.  I'm not an engineer, so I don't use guitar cabs!!

Have you even read any of that Paul Frindle stuff I mean you are quoting most of his stuff and he def sugests the posibility of harmonic distortion,  who know's where it is happening??

Maybe it's a combo's of transitor preamps (audio interface) that are driven hitting the analouge side of the converters, this surely would add at least some kind of transistor based harmonics of the third order.

See another idea!!!

Neb

I hate to even contribute to this, but I think that what some folks are missing is that there are different issues involved. The AD/DA part of it is one thing, and just a reasonable level anywhere from -20 to -3 should probably be fine (peaks obviously), though -3 doesn't make much sense because you'd have to be so careful during recording and it wouldn't really make any sonic improvement to do so, so it's not something you'd want to do.

But the biggest issue in that huge Gearlutz thread is about *digital processing* on those tracks recorded at high levels, not about the actual in and out levels themselves. If you run your tracks (and buses) overly close to 0dBFS and you have plugins on those tracks, you can have issues in some cases, because they can raise the level in unexpected ways, and it can happen between multiple plugins on the same track so you might not see it on the actual track meters because a subsequent plugin could bring the levels back down.

In a floating point world, it might not make much difference since there's no clipping involved going over 0 (though there might be in the actual algorithms of any given plugin even still), and you can't assume necessarily that all of your plugins are processing in floating point format internally I guess. Though if you do go over 0 in the floating point world you can be needlessly throwing away resolution bits. But in the 64 bit floating point world it might not make that much difference even there.

Yeh, you can subsequently bring down the track levels via the trim, but again you aren't gaining anything sonically by tracking them higher, so what's the point? You are making your tracking more succeptible to overs and then just turning it down anyway before it gets into the DAW's tracks. So it's kind of silly.

On the commonly discussed issue of 'more resolution' when you record hotter, the thing that Paul pointed out to me is that this isn't true. More resolution comes from having finer gradations. 24 bit is 24 bit. The gradations are the same across the whole possible range. It's the same resolution at the bottom as at the top. So you aren't getting more resolution by using more bits.

The real issue with resolution comes later during the actual mixing. But if you are on a 64 bit floating point DAW, and that's the case for all of us here on SONAR, then there are vast number of values between every 24 bit captured value, and those are used during the actual processing, until it's time to spit it back out to the D/A.
 
The whole issue of harmonic distortion on A/D is just silly. Who would want to do that? If you want (good, harmonious harmonic) distortion, get it on the way in from your front end. This is a non-issue that it's not worth wasting air arguing about.
 

[accidentally quoted my own message]

Ben I am also saying that when digital systems are low in harmonic distortion I am assuming that any analog signal path is of the highest quality and is also introducing the least harmonic distortion possible. If I were to measure it I would be patching the oscillator direct to the sound card nothing else! And I would be putting the distortion analyser on the output of the sound card too!

But what you are saying is also correct in that harmonic distortion could be introduced in a variety of places prior to the A to D converters. And of course even if a pristine signal has got into a DAW now we have plugins that can introduce tons of harmonic distortion but that is on purpose.

Analog tape recorders are very high in distortion. Did you know of the order of several percent! Have you ever seen the square wave response of a reel to reel tape recorder, it is pretty bad overall. Only the finest machines can even achieve what looks like a decent square wave and even then it is not great. Digital can do this while walking in the park!

I try to avoid loud guitar cabs too I think they are too loud! I think we can leave that stuff to Danny. I love recording guitars direct and getting stuck into them inside the DAW. All be at a very low volume at 2am in the morning! I think we have really progressed in this area for sure. Plenty of distortion there but it's good distortion though!

BenMMusTech


You always want to have the last word

Oh the irony 

try to avoid loud guitar cabs too I think they are too loud! I think we can leave that stuff to Danny. I love recording guitars direct and getting stuck into them inside the DAW. All be at a very low volume at 2am in the morning! I think we have really progressed in this area for sure. Plenty of distortion there but it's good distortion though!

 
LOL Jeff! I got news for ya brother, it's rare for me to crank anything up loud unless I have to cab wise. When you get a guy looking for that Angus Young type tone, there's usually no other way other than loud. But for everything else, the newer high gain features in amps and pre-amps allows us to get a nice saturation at a lower volume. The only time I even record cabs is when we are looking for the coloration the actual cab has on the sound. They even have impulses for that now too that work a treat. RedWirez has some incredible cab impulses that are realistic. :)
 
-Danny

mattplaysguitar

BenMMusTech


You always want to have the last word

Oh the irony 

It was suppose to be!

Danny Danzi

try to avoid loud guitar cabs too I think they are too loud! I think we can leave that stuff to Danny. I love recording guitars direct and getting stuck into them inside the DAW. All be at a very low volume at 2am in the morning! I think we have really progressed in this area for sure. Plenty of distortion there but it's good distortion though!

 
LOL Jeff! I got news for ya brother, it's rare for me to crank anything up loud unless I have to cab wise. When you get a guy looking for that Angus Young type tone, there's usually no other way other than loud. But for everything else, the newer high gain features in amps and pre-amps allows us to get a nice saturation at a lower volume. The only time I even record cabs is when we are looking for the coloration the actual cab has on the sound. They even have impulses for that now too that work a treat. RedWirez has some incredible cab impulses that are realistic. :)
 
-Danny

I like being able to do this high gain stuff without blasting the ear drums out. 

For those interested in the twist introduced with the accumulation of noise across several channels...

Music signals sum additively (as you would expect) whereas noise signals do not - that is uncorrelated noise signals do not. In this sense uncorrelated means random and not in any way correlated to the music signal on the channel. This kind of noise is typically caused by thermal noise in resistive components and is usually the type of noise that comprises the noise floor of well designed circuits.

In a way uncorrelated noise is benign in that adding together two channels does not double its level but increases it by square root of 2 - about 1.4, and adding together 4 channels increases it by only 2 times, 16 channels - only 4 times. That is, its result is the square root of the sum of its squares. 

This is why some very low noise pre-amp circuits use parallel signal paths as you get a halving of noise to signal for each 4 parallel paths. It also explains why wide tape is quieter than narrow tape.

On the other hand correlated noise does sum normally. Correlated noise being noise related to the music in the channel eg intermodulation distortion, harmonic distortion, clipping, aliasing noise, over and undershoots, ringing etc.

On the main topic, imho the sensible approach is to be aware of the compromise necessary to make sure your signal is large enough to have a usable SNR, and small enough to guarantee no input clipping. That will depend on source material, won't it? 

Rock guitar has such a low dynamic range that it can be recorded at almost any digital level, any recording noise being swamped by amp and pick-up noise, and the loudest parts limited by the self compression and limiting of speakers and amp at full output. On the other hand, classical music demands the best from both recording equipment in terms of self noise, and in the choice of nominal recording level so that there is not a chance of input clipping. Also bear in mind that classical music is often released on CD at low levels in any case. We are talking 16 bit here and the music is quite capable of being achingly beautiful at levels far less than -20dbfs

The great thing about 24 bit recording is that you don't need to sweat too much to get as close to 0dbfs as possible. 

Can any of you say that pieces you have recorded at -20dbfs or less have suffered - except from when you got the gain-staging in the analogue world wrong? 

Jeff Evans


I don't agree that digital recording introduces harmonic distortion, that is bollocks!

Jeff I believe what it is is that some ADC makers set things up so the analog side starts to distort before they clip digitally - so that if you happen to clip, it might "soften" the clipping a bit and sound better than the hard digital clipping. But I think it's more designed to mitigate accidental clipping somewhat than it is meant to be a desirable coloration - not that it couldn't be used as such.

cliffsp8

In a way uncorrelated noise is benign in that adding together two channels does not double its level but increases it by square root of 2 - about 1.4, and adding together 4 channels increases it by only 2 times, 16 channels - only 4 times. That is, its result is the square root of the sum of its squares.  

 

And I want to point out that this ratio of 1.4 equals the +3dB RMS I stated earlier. Peak is still double (+6dB) though.

BenMMusTech


 

Have you even read any of that Paul Frindle stuff I mean you are quoting most of his stuff and he def sugests the posibility of harmonic distortion,  who know's where it is happening??

Pretty much everyone else in the thread knows. But as you usually do, you're assuming that you not understanding something means it's somehow ineffable. It isn't, it's just that YOU don't understand it. But being so aggressively determined not to listen to what people are telling you, I do wonder what the point of these threads is. After all this verbiage, you are still as clueless as you were before. When was that "digital warmth" one? Something like six months ago? That's a long time to have learned nothing.  

John T take it easy with him

There was a guy who used to post regularly over at GS by the name of OldAnalogeGuy and he used to come out with similar BS, I wonder if they're related

Hey, Cliff! You should post here more often, man. We here in the science contingency can always use another rational voice. Drew can't do it all himself, and the rest of us have day jobs.