Ok here is why we should record to digital as hot as we possibly can!!

Ok for some reason I can't post the scanned article but I have Micheal Stavrou's article which I will condense and hopfully I will be allowed to upload the scanned article latter.
 
Firstly a few of us have been having a disscussion about the merits of recording levels into Sonar.  I have sugested as hot as we can before we clip, this has been shot down many times.  Time to put up.
 
It also has nothing to do with 16 bit as it has also been suggested.
 
This information was taken from Audio Technology Magazine issue 13 pg68
 
This is the first two paragraphs:
 
I am by no means an expert on digital signal processing but would like to share my perspective on the differences of analouge and digital music recording.  If I am right, we really have to change the we hit the digital medium.
 
You know how I am always saying  "audio often behaves in the complete reverse of how you'd logically expect it?"  Digital is no exception.  For example in the analouge domain, reducing gain, always reduces distortion.  Meanwhile in the digital domain, reducing gain increase's distotion.
 
Ok he goes on to give the analogy of a skyscraper and how analouge captures the middle of the skyscraper best and digital captures the tip or the peak of the skyscraper best.
 
Ok we will drop into what Stav has to say next:
 
The skyscraper is analogous to the volume level of the sounds we record.  Analouge captures the most importent elements of the sound with pin-point accuracy (the middle of the skyscraper, (where the critical work is being carried out) while the top and the bottom where the activity is less critical (the peak and bottom of the skyscraper) is captured less shaprly.
 
We call the pin-sharp section or "operating level" or optium recording level because here it does not suffer from transient distortion or the noise and loss of detail from being under recorded.
 
Every medium has an optium recording level where it's specifications shine.  Everything about this we call headroom.
 
Parapharsing:
A digital veiw of the skyscraper is somewhat different we have to refocus our camera to the uppermost section of the building, only the tip of the Skyscraper is in focus now, while the rest of the building slowly loses focus as we move the camera down the building (I hope all are following the skyscraper analogy) Why?
 
Stavs actual words:
The pinicle of the building is captured by the bulk of the recording bits and thus enjoying a higher resolutution.
 
If this analogy holds true, then the sad thing is that we finish our mix onto a digital medium- with the cleanest elements being the tip of the skyscraper and send it to mastering.  What does the mastering engineer do? He or She surgicly removes te clearest most pin-sharp material to attain a higher overall level.  So unlike analouge, where the worst elements removed, here in the world of digital the best most focused elements are removed, leaving a grainer version of the lower level components.  I take this to mean this is why digital can sound so harsh. 
 
Parahrasing:
Stav then goes onto say most good analouge gear has at least 20 db of headroom, where as digital has zero, how do you work with device that has a zero amount of headroom? 
 
Stavs words:
What do I mean exactly??  Well when you look at digitals point of maxium clarity, minimum noise and minimum distortion, (0dbfs) there is 0db of headroom above this point -you simply can't go any louder without distortion cutting in.  Maxium level is digital's optimum level.  Therefore it is accurate to say there is no headroom in digital recoding devices.
 
Certinaly you can generate a false sense of headroom by targeting your recording level by aiming for say -18db (which some around here still advocate) but you must be aware you are degrading the signal to some degree by not recording it at the optimum level-namley you will have fewer bits describing the critical elements of your recording.
 
The beauty of multi-track recorders is they appear to have no noise- you can run 32 tracks up at unity and hear nothing unwanted.
 
When recording to digital multi-tracks don't be decived as I was by the ultra-quiet noise floor to think that from here on down is useable dynamic range.
This false sense of security tempts you to play all the track back at unity gain and hit the digital recorder with just the right level to mix itsself on playback with all the faders at zero. 
 
This is admittedly very convienent but one of the worse things you can do to the sound quality.
 
Paraphrasing:
And this is what I have been advocating: It is often too much trouble to optimise the recorded level and then re-adjust the output level to create balance.
 
You really need to massage the input signal until it reaches maxium level into the digital domain and then adjust the recorders channels output fader (trim, I prefer to use the input stage to readjust the balance then mix) to acheive the desired balance.
 
This folks iswhy I advocate recording into Sonar as hot as possible, the best part of digtal and the best operating level for digital is as close to 0dbfs.
 
Michael Stavrou is a world respected audio engineer who has worked for George Martin and Air Studios.
 
Next week I will explain my other crazy theory, optimum operating for your preamps and other outboard gear.
 
I will try and post the scanned (rather grainy) article for people to read it in more depth.
 
Peace Neb

 the best part of digtal and the best operating level for digital is as close to 0dbfs.

I'm pretty sure this is complete bull (I'm not going to say I know for certain cause I'm not an expert). The resolution you have doesn't change. It's not like at 0dB, each bit is in 0.01dB (for example) increments so it's nice and fine, but down at -18dB, each bit is only in 0.1dB increments and at -60dB it's in 1dB increments. I 'think' that's what he's attempting to say in his article. Correct me if I am wrong.




If you get any audible change from recording with 0dB peak vs -18dB peak once volume matched (assuming 24 bit), it's going to be from your non linearality of your preamp and the physical noisefloor of your particular equipment, not A/D converters doing a worse job.

Oh, and I predict this thread is going to get LONG!

mattplaysguitar

 the best part of digtal and the best operating level for digital is as close to 0dbfs.

I'm pretty sure this is complete bull (I'm not going to say I know for certain cause I'm not an expert). The resolution you have doesn't change. It's not like at 0dB, each bit is in 0.01dB (for example) increments so it's nice and fine, but down at -18dB, each bit is only in 0.1dB increments and at -60dB it's in 1dB increments. I 'think' that's what he's attempting to say in his article. Correct me if I am wrong.




If you get any audible change from recording with 0dB peak vs -18dB peak once volume matched (assuming 24 bit), it's going to be from your non linearality of your preamp and the physical noisefloor of your particular equipment, not A/D converters doing a worse job.

Ah one of the Melbourne Music Mafia!!!, no he's not, he is saying that each recording medium has an optium level or operating level, digitals just happens to be as close to 0dbfs.
 
I will try and put the scaned article up some how.
 
You know that you have just called one of the most respected audio engineers ****!!
 
Neb

mattplaysguitar


Oh, and I predict this thread is going to get LONG!

I have put up the argument and the theory is pretty sound!!!
 
Bring it on!!!
 
Neb

Matt your not reading the article properly, he is saying you have to refocus the camera or the optimum level of the recording medium, also he is saying that as you turn down the digital signal it becomes more distorted the opposite of analouge!!

Read properly Matt!!

Neb 

Stavs actual words:  The pinicle of the building is captured by the bulk of the recording bits and thus enjoying a higher resolutution. 

Is what I said not the same?

The resolution you have doesn't change. It's not like at 0dB, each bit is in 0.01dB (for example) increments so it's nice and fine, but down at -18dB, each bit is only in 0.1dB increments and at -60dB it's in 1dB increments. I 'think' that's what he's attempting to say in his article. Correct me if I am wrong.

I believe the entirety of the building is captured evenly by all the bits, and at the noisefloor you have a bit of digital noise... At least that's how I understand it. I'll let people who have studied this in more depth reply!

BenMMusTech


Matt your not reading the article properly, he is saying you have to refocus the camera or the optimum level of the recording medium, also he is saying that as you turn down the digital signal it becomes more distorted the opposite of analouge!!

Read properly Matt!!

Neb 

Oh ok, I get what you're saying now. Or what he's saying.


So then now it comes down to the quality of your A/D converters?... So it's more a discussion over A/D converter quality than digital, as such? A/D converters read best at 0dB and degrade as the signal level decreases? Thus higher quality ones will convert more accurately with less distortion at lower input levels? That could make sense. But if that's the case, I'm calling the differences negligible and inaudible...

mattplaysguitar

BenMMusTech


Matt your not reading the article properly, he is saying you have to refocus the camera or the optimum level of the recording medium, also he is saying that as you turn down the digital signal it becomes more distorted the opposite of analouge!!

Read properly Matt!!

Neb 

Oh ok, I get what you're saying now. Or what he's saying.


So then now it comes down to the quality of your A/D converters?... So it's more a discussion over A/D converter quality than digital, as such? A/D converters read best at 0dB and degrade as the signal level decreases? Thus higher quality ones will convert more accurately with less distortion at lower input levels? That could make sense. But if that's the case, I'm calling the differences negligible and inaudible...
 
I think you have hit upon something but convertor quality is not an issue any more, RME and Motu and most of the audio interfaces that you and I use are built from the same company.
 
The theory still holds true though that is as you decrease the input signal distortion is added because you are still using less bits to describe the program audio.
 
Neb

Ok lets try this anaolgy:

Say if we are recording bass and our peak was -10db and our RMS was say -24db which would be well under 0dbfs on a digital recorder, so we arn't using enough of the bits of our ADDA converters and what Stav is saying by not using as much of the bits as possible we are introducing distortion because digital and I don't know why this is the case the lower volume level adds distortion.

I will have to research why distortion is added when you have lower volumes on the ADDA when recording.

Neb   

Yep, so he's saying that at higher levels, you're using more more bits to represent the data, and at lower resolutions, you're using less bits. This is what I thought he was saying in my original post. I think this is completely wrong and is not how digital works. I can't say I'm 100% sure, but I'm 95%. I'll gladly accept it if I'm wrong.

It's like people who say that 24 bit has more resolution that 16 bit. It doesn't. The resolution at a given level is the same, but the lower end is extended, so you can go even quieter before you hit the digital noisefloor. There is not a 'finer' resolution in 24 bit. It doesn't use more increments. The increments are the same. There are just more of them in total. These increments are the same amount over the entire recording level. They are not all squashed up at the 0dB region leaving only a few down lower.

Further more, once you introduce floating point, it doesn't matter where things are. You got a file recorded with a peak at 0dB and a noisefloor at say -120dB. If you aren't using floating point, save the file with a -60dB reduction and you now have a -60dB peak with a -120dB noisefloor. Bring it back to 0dB peak and your noisefloor moves up to -60dB. With floating point, that noisefloor moves with the peak down to -180dB and then back to -120dB. It's got some extra info in there simply providing an offset. At least that's how I THINK it works. Again, correct me if I'm completely wrong and I'll be wiser for the knowledge.



At the end of the day, if what he is saying turns out to be true, I really don't imagine anyone is going to hear the difference, assuming your gear is quality and not introducing other issues to cloud your test. I'll do a few tests when I get a chance and see how it sounds. I don't actually expect to get the same sound result but I'm guessing it'll be due to my equipment cheapness, but we'll see! I'll do maybe -3dB peak, -18dB and -60dB. Something like that should be interesting.

Ben is basing his argument on the fact that Stavrou's book was written in 2003 and a lot has changed since then. He is not taking that into account. There was some truth in that then. Stavrou was merely saying that in order to get the best out of 16 bit digital at the time we should aim for a K reference of K-12 (rms) which is pretty loud and close to 0dB FS.

Notice Ben there is not even one mention of 24 bit recording in that chapter. Your assertion now is completely wrong and you should not be putting up incorrect facts and only creating confusion for new readers. Most of us I am sure agree that what you are saying is incorrect.


Here was my response to his argument back in the Analog VS digital camera thread:


Re:AnalogCameraVsDigitalCamera - March 25, 12 8:51 AM ( #17 )

I may be able to throw some light onto the subject. Ben is quoting a chapter from Mike Stavrou's book 'Mixing with your Mind' from the chapter 9 Digital VS Analogue. Mike uses the analogy of a large skyscraper. He says that because of its enormous height it is hard to capture the whole thing in focus. He is saying the best sound from analog is in the middle of the skyscraper and this is similar to level. Right at the top transients and things will get distorted and right at the bottom the tape hiss starts to mask low level information.

In Digital recording however Mike says the very top of the skyscraper is the cleanest and sharpest part of the picture and it gets progressively blurry as you go down to the bottom. Level wise we know that the maximum number of bits is used when our signal is high and as we go down lower in level we are using less bits to capture important information.

Mike is saying that setting your digital reference level way down at -18 or even -20 db FS as Bob Katz points out may not be the best idea because we are not using all the available bits to capture most of the music or where most of our music lies level wise. Mike also suggests that a better digital ref level might be the Katz -12 db FS instead where more bits are being used. But this requires you to lift your entire gain structure up there and also requires some effort in preventing clipping as we now only have 12 db of headroom. (I actually work at -14 a lot of the time)

But I must also remind Ben that Mike's book was written in 2003 and I really get the impression he is referring to 16 bit recording most of the time and if you are working at 16 bit level then what Mike is saying is probably true. But things have changed a lot since then and we are now in a different situation where 24 bit recording is common place. The skyscraper is different now and much more of it is in focus and the bits down the bottom are now -144 dB away from the very top of that sharp focus image.

So Ben a very recent article in the said Audio Technology magazine has completely debunked this concept of having to record at higher levels in the digital world and with 24 bit recording we are not under that pressure anymore so the -18 dB or Katz -20 dB ref level is perfectly acceptable. You are not loosing any detail now even at -20 because the noise floor is now still -124 dB below that. In the 16 Bit world if you are using a ref level of -20 db with only 96 dB of dynamic range (and we know we really only have -90 dB  in fact)  then the noise floor is only sitting 70 dB below that which could be considered dangerous in terms of very important low level harmonic material. Not so with 24 bit though.



We have just aquired a new mixer at the TAFE where I teach. It is an SSL AWS 948 console and close to $100,000. An SSL guy came all the way out from th UK to teach us how to use it. (I must say it is rather nice!) Anyway it has VU meters on the front. The reference levels that SSL recommend are switchable and are -24, -22, -20 and -18. Of course the -20 agrees perfectly with the K standard. I don't think SSL would recommend using a ref level that is even lower then the K -20 (eg -24) if it was SOOOO wrong.

If you are going to post things in the Techniques forum it is essential that at least the information is as correct as it can be.

The problem is the highest volume adda converters use is around 118db no adda converters use the full dynamic range of 144db.  In fact motu and rme use between 110db and 118db  Presonus have 118db of dynamic range. And lexicon us series are 101db. Pretty close to full range 96db. 

Once again you should not misslead readers. Nowhere in that article does stav mention 16 bit but it doesn't matter if adda converters still do not use the full 144db of theortical dynamic range of 24 bit.  

The way I interpret it is the theory is sound then and now.

I would like to see imperical evidence pg number and the exact quote of 16 bit. 

How can the theory be turned on its head due to bit depth ESP if 24 bit does not mean 24 bit. At the moment it's more like 20 bit. 

I have technical degrees you have musical degrees leave the technical stuff to me and start to worry about the relevance of your course as I hear tafe courses are about to be cut in Victoria and the relevance of Music Industry Technical Production is irrelevant and one of the to be first cut.  You can take that to bank!!

Ben

I accidentally recorded a whole song in 16 bit without realising once. I put lots of compression on all the tracks and mastered it loudish so brought that noisefloor right up. I never even noticed till one day I was looking at the original wav files and noticed they were 16 bit. I listened back to the song hard and only just managed to hear it. I could, but barely.

My point here is there are so many other things that make SO much of a difference. We stress about 24 bit so much, but 99% of people would never even notice if you don't tell them. The difference in noisefloor from 16 bit and 24 bit is audible, but the 16 bit is still actually pretty darn low. You're only going to hear it in a song where only one instrument is playing, not when the whole thing kicks in. That noisefloor, even if it kicks up to say -60dB after mixing and mastering, it's going to be pretty well masked 90% of the time. How many of you actually have a background level and equipment noise level when recording that is LESS than -96dB? Not many.

I do find it interesting doing these tests and seeing what is better etc, but at the end of the day, that's all it is - interesting. You're talking a 0.2% increase in the sound quality of your song. Double your noise isolation when recording and you've already overshot the losses in the digital domain by far. Change mics and that distortion that you get from recording down low (if it is the case) is minuscule in comparison. I personally would rather just quickly set things safe so you don't get overs and get down to recording some great music. Stressing about getting things as high as possible then being really meticulous about your playing so you don't accidentally go over could ruin in the feel of your music. Just turn the gain down, stress less, put out a great performance, and work on the sounds and quality that really matters and makes a reasonable difference.

Very well put Matt I could not agree more. Even though I may go on about these things, they are insignificant in the scheme of things especially when compared to the music itself.

It's a boring argument this, do we record loud or softer and do we do it 16 bit or 24 bit. I have made great recordings at all levels and bit depths. It is good to get things right engineering wise for sure and you don't need any qualification to be able to do it either, just experience.

As Ben says we should leave all the technical details to him. I am happy to concentrate on the music instead. Once you get the engineering stuff right and it is easy in my opinion, you really need to concentrate on the music and the music only. That is the only thing that stays behind in the final recording, the music only. No one is really that interested in how you got there, it is more about what is there. Technical production can be sloppy or precise and also it is finite in some ways. There are limits on the ways we produce audio.

Our TAFE is doing very well. We are spending money and our course is getting bigger and better. Sure some TAFE sectors are going to be effected and I don't agree with the way our current Premier thinks about TAFE education. But even if things there change I can always increase the amount of music I compose at home and get paid for it. I am pushing that up and actually trying to move away from the other a bit.

But you have got to be reasonably good to create music and get paid well for it too. Something that requires a lot of creativity, art, performance and hard work too. The technical work. And maybe even a music degree or two doesn't go astray. You don't need those to make good music, but they do open doors into new territory. Music is unlimited and a constant learning experience and that is where the art is, not so much engineering art. That is boring art in some ways.

People get too worked up and too involved with minute details of engineering while the music is boring. Spend large amounts of time looking right into the music and all the ways you could make it better. Every aspect up to where the microphone is and don't worry about from that point on. The initial creative element could be done fast as some great works of art are created quickly or put it together slowly, improve it later and spend the least amount of time engineering. Good engineers do it fairly quickly. Everyone forgets that magic interaction between good music and good engineering. When the music is great all the engineering easily falls into place. If you are having a hard time engineering any part of your own or anyone's music it means the music is not good or good enough at that point. Re evalute the musical art at that point instead and the engineering will then sound amazing.

If you are spending more time on the software than the music, you have got the wrong program, if you are spending too much time ranting over insignificant engineering details you have got your priorities wrong. At the end of the day it is only your music that is going to effect the listener emotionally and if you want it to, sell you or not.

I think it something to do with age maybe. When I was younger I was really excited and fascinated in the engineering process and all the techncial stuff and I still enjoy it of course but at some point (when you get older or at a certain age) I think you have to just leave it all alone and only think about the music.  New music. Great music. We have to come up with fresh ideas and musical concepts or that is how I feel about it anyway.

I just want to perform my music well enough to get in the songs forum...no picking of nits here...

I personally think this will always be a subjective topic for everyone. My advice is to simply do a few small projects while logging your inputs and then determining at the end which is best for you and the soundcard/style you're dealing with.

I've tried every input setting I could mess with and for me personally, -6dB peak seems to be the best bet for me no matter what bit or sample rate I record at. Even an average of -6dB to where certain things may hit -4dB or even -3dB have been fine for me. But these days I've been really happy with no hotter than a -6dB peak. If a person gets good sound at the hottest point they can record digitally, I say go for it. If you prefer -10dB or less, go for it. Let your ears be the judge. :)

-Danny

First sentence of first paragraph (please correct me if I'm wrong)

I am by no means an expert on digital signal processing but

That says it all.

Ben, record all your tracks at 0dB, mix them together at 0dB, now try to get them out of the DAW WITHOUT turning them down (and, according to your expert creating distortion).
What you are presenting here is the "RESOLUTION" argument; it doesn't work.
It DID NOT work at 16 bits, when sig>noise MIGHT be an issue.

0vu = +4dBu (1.23v rms) = apprx -18dBfs (depends on the the DAW)
So to get your line signal at 0vu (this is analog now) into the DAC so it can go into the DAW at + 18dB.
YOU GET DISTORTION.

Anyone who suggests recording "as close to 0 as possible" is NO AUDIO EXPERT!
RUN don't walk away as quickly as possible.
read this thread at GS Gain Staging Mixing ITB (I took some liberty for the sake of brevity)
Absorb everything Skip Burrows and Paul Frindle said.

Tom

The error in this argument, like many errors involving audio, is that it assumes that humans have infinite resolution to hear quantization noise/distortion/error, and/or are recording through a perfectly noiseless analog chain and/or are recording in an environment with no (desirable) background noise.

In the real world, once something is of such low level that it will never be audible to humans and/or is significantly below the recorded noise floor, the word "better" no longer applies.

So you end up with people making heated arguments that are entirely hypothetical/theoretical and have nothing to do with the real world.

This is made much worse by the fact that humans have been proven to be both quite incapable of determining what they hear vs. what they imagine they hear, and routinely imagine they can hear things that aren't there.

Spot the 'real' approach.




I love that picture.  Thanks Ethan.

Now if you'll excuse me, I have no desire to feed the troll.

It's easy enough to figure out how to do a test on relative input levels, it doesn't even need another discussion.  How much better than I can't hear a difference do you need?

If you have failed to set up a proper test or cheated and find a difference then you have to determine which one is 'best'.

Good luck with that, I'm off to do something more worthwhile like clip my toenails.

@ Jon +1
But I DO hear a difference if I try to boost the signal enough to SLAM it to 0dbfs.
Its called DISTORTION.

T

DeeringAmps


@ Jon +1
But I DO hear a difference if I try to boost the signal enough to SLAM it to 0dbfs.
Its called DISTORTION.

T

There was a recent discussion of the definition of "bollocks". For those Yanks still confused about the term, read post #1. 

This is one of the reasons I am uncomfortable applying the term "engineer" to mixers. Though helpful, it is not necessary to understand the underlying technology to be a successful recordist, and a Grammy award is not an engineering degree. 

I prefer the title "Sound Engineer" to "The skirt on the desk"... 





(yes,  I overheard that one at a gig once...)

"There was a recent discussion of the definition of "bollocks". For those Yanks still confused about the term, read post #1."
+1 again!
"The skirt on the desk"...

Yeah, I get that sometimes too!

T

Jeff - I appreciate your insite.
Jonbouy - +1

Ben, my ears cannot decern any difference in final mix product whether I record low or hot so my ears tell me this argument won't make any difference in quality at all.

Rimshot

bitflipper


This is one of the reasons I am uncomfortable applying the term "engineer" to mixers. Though helpful, it is not necessary to understand the underlying technology to be a successful recordist, and a Grammy award is not an engineering degree. 

 
I am an engineer AND a mixer, although an structural engineer and a amateur mixer.
 
I have to say that although i agree that the answer to this question (if you ever find it) will probably be useless in the real world, as a relative newbie i find these discussion very interesting and helpful in my search for an understanding of the digital audio world. What i cant understand is that these discussions happen at all. Its like a discussion about religion. These recorders are man made things, surly someone knows how they work. The man who made them maybe?

Karyn


I prefer the title "Sound Engineer" to "The skirt on the desk"... 





(yes,  I overheard that one at a gig once...)

That's outrageous, I agree whoever she was she should have said 'mixing console'.

This Stavarou guy is wrong. I don't care who he's worked for or what he's done, those things do not change the fact that he is incorrect. 

Matt: you do have increased granularity in 24 bit, as it happens. But the increase is so vast that you also get the lowered digital noise floor effect you describe. 

"leave the technical stuff to me". Actually LOLed at that. 

Wow,
  I gotta say it doesn't seem to me that it would make any sense to push your input levels that close to 0db only to risk peaking and ruining the take.   The theory doesn't seem to hold up and even when faced with elaborate explanations Ben, your arrogance is a tough pill to swallow.

  In the end I would say for test it for yourself, do several recordings match the levels before exporting them, and have someone else rename the files and write down the original names.  give yourself a blind test and see how you do. 

He's a lost cause. He's more determined to Internet Win this argument - which he's started threads on multiple times now, with the same nonsensical arguments - than he is to be any good at audio engineering. If he's so bullishly determined to be incompetent, I say let him.

Ok firstly Stav say's as close to 0dbfs as possible, so what Danny has said is correct, I like to hit between -3db peak and -6db peak, not always possible.

Oh yea there is a difference in the bottom end, this is where is is most important stuff goes on for me and by getting that bass as loud as possible the more definition I can achieve and the more warmth, maybe it has something to do with the higher recording input and capturing the harmonics better.

Did no one understand the concept of optimum or optimum operational level and each recording medium has it!!

And so lord pontiff where is this pg number ascribing the concept to 16 bit only??

Finally I was giving The Ponitificater a kick when I said leave the technical stuff to me, it was tongue firmly in cheek!!!

Neb

John T


He's a lost cause. He's more determined to Internet Win this argument - which he's started threads on multiple times now, with the same nonsensical arguments - than he is to be any good at audio engineering. If he's so bullishly determined to be incompetent, I say let him.

Actually I put up the evidence this time for my theory, put up or shut up as they say JohnT, once again who are you???
 
Neb

BenMMusTech


Did no one understand the concept of optimum or optimum operational level and each recording medium has it!!

The "optimal" level for digital is anywhere you both aren't clipping and the quantization level is a bit below the noise floor of your signal. Quantizing this noise at a lower level doesn't add any benefit. And with 24bit audio you aren't going to hear the quantization noise in the real world, even if you amplify it.

drewfx1

BenMMusTech


Did no one understand the concept of optimum or optimum operational level and each recording medium has it!!

The "optimal" level for digital is anywhere you both aren't clipping and the quantization level is a bit below the noise floor of your signal. Quantizing this noise at a lower level doesn't add any benefit. And with 24bit audio you aren't going to hear the quantization noise in the real world, even if you amplify it.

But what if we are missing out importent information like harmonics because of recording too softly, as I say some convertors and it may come back to this question of convertor quality, don't use the full 144db of dynamic range, some as I have pointed out are barely using 101db, so just over 16 bit. 
  
And what about the idea of is we turn down the signal level in digital we are actually adding distortion, Drew I know that your the real tech head here explain these concepts please.
 
Neb

wow

BenMMusTech

drewfx1

BenMMusTech


Did no one understand the concept of optimum or optimum operational level and each recording medium has it!!

The "optimal" level for digital is anywhere you both aren't clipping and the quantization level is a bit below the noise floor of your signal. Quantizing this noise at a lower level doesn't add any benefit. And with 24bit audio you aren't going to hear the quantization noise in the real world, even if you amplify it.

But what if we are missing out importent information like harmonics because of recording too softly, as I say some convertors and it may come back to this question of convertor quality, don't use the full 144db of dynamic range, some as I have pointed out are barely using 101db, so just over 16 bit. 
  
      
Neb

It always comes down to what I said:

The "optimal" level for digital is anywhere you both aren't clipping and the quantization level is a bit below the noise floor of your signal.  

The mistake people make is they imagine the noise isn't there (which is quite easy to do). Or they change the rules midstream and create a moving target. Mind you, this is often done unconsciously and/or inadvertently.

And what about the idea of is we turn down the signal level in digital we are actually adding distortion,   

The "distortion" here is just the quantization error. But if the quantization level is below the noise floor, and the noise is more or less random, the noise acts as a form of "self-dither". So the quantization error becomes "noise", not "distortion" - just like when you add dither before reducing bit depth at the other end.

And, since this quantization "noise" is both very low in level and below the noise level of the signal, then even if it was amplified enough to be audible it would likely be masked by the signal + noise anyway.

Ok so what about, sorry Drew, I refer to you as the most technicaly minded of us all here.

So what about the notion of missing information because we are not using all the bit's avaliable to us eg: harmonics.

I hope that makes sense.

Neb

Ben,
IF you WANT a real education read the GS thread I liked to earlier.
Skip Burrows works WITH and ON SSL consoles EVERYDAY.
Paul Frindle was on the design team AT SSL.
They are ADAMANT that you keep your signal levels the same in the digital realm as you would in hardware.
Danny's input PEAKING at -6 is like pushing the VU all the way to the right. His AVERAGE signal will be right at Ovu.
At 0dBfs the needle will be PEGGED.
Nothing personal here Ben. You are free to record however "sounds" best to you.
You don't have to believe a single word I say, (and probably shouldn't, I'm just another internet "Know it all") BUT Skip and Paul have REAL credentials.

'nough said.

T

DeeringAmps


Ben,
IF you WANT a real education read the GS thread I liked to earlier.
Skip Burrows works WITH and ON SSL consoles EVERYDAY.
Paul Frindle was on the design team AT SSL.
They are ADAMANT that you keep your signal levels the same in the digital realm as you would in hardware.
Danny's input PEAKING at -6 is like pushing the VU all the way to the right. His AVERAGE signal will be right at Ovu.
At 0dBfs the needle will be PEGGED.
Nothing personal here Ben. You are free to record however "sounds" best to you.
You don't have to believe a single word I say, (and probably shouldn't, I'm just another internet "Know it all") BUT Skip and Paul have REAL credentials.

'nough said.

T

Hey Tom I'm not that disrespectful, people have me all wrong.
 
This console thing is a whole different argument though, I don't even know how companies like SSL can still keep afloat.
 
Look I know that Stavs theory works, maybe it has something to do with cheaper converters and cheaper convertors to get the best out of them requre this technique.
 
Do you know what the dynamic range of the converter's are on the SSL console because I am starting to think this is where the answer lies.
 
I know my peak is generally between -6db -3db that's what I aim for and as I say I believe to get the best clarity out of the bass instruments, this is where I like to sit.
 
I will have to look at RMS values next time I record.
 
Neb
 
Oh I know who Paul Frindle is, I have an interesting post of his on Gearsluts and I think it ties in with some of this stuff let me look it up, it was really cool!
 

BenMMusTech


Ok so what about, sorry Drew, I refer to you as the most technicaly minded of us all here.

So what about the notion of missing information because we are not using all the bit's avaliable to us eg: harmonics.

I hope that makes sense.

Neb

Remember what I said about imagining there's no noise or inadvertently moving the target?

We've already agreed that we want the quantization level to be a bit below the noise floor. The noise is causing the signal level that gets sampled to bounce around randomly at a magnitude greater than the quantization step size. How is measuring a randomly moving signal in smaller increments going to help?

With a dithered signal, do you understand that you can still hear stuff below the quantization level?

Ok I got that drew but if the harmonics are all in the area where the quantization is happening and or below and even though we can still hear them, because of dithering and the quantization errors is it not possible we are still getting the harmonics but at a reduced fidelity.

Does that make sense??

Neb

Just out of interest the SSL console I was referring to in one of my previous posts is a fully analog console only. All A to D and D to A in our case is handled by AVID HD 192 interfaces outside. (3 to be precise)

But you still need an rms ref level at some reference point and SSL were suggesting the ref levels I mentioned before eg -24 -22, -20, -18 etc. -20 is a good one because it matches the K level. The VU was calibrated at -20 on the particular console we have had installed at the factory.

Ben some of your ideas are not over wrong in that recording hot was a good thing to do in the past. (I spent many years working with EMU samplers from 8 bit into 12 bit and 14 bit so I know what you are talking about. You did have to sample hot back in those days otherwise it sounded just plain bad!) But now with both 24 bit recording and the fact that almost any converter you buy now is going to be pretty excellent in the scheme of things so now we can virtually relax levels wise now and just work with say a K ref level of -20 without any concerns that things are not being recorded well.

Even if we are at -20 for example and the music dropped down 30 db below that which is very very quiet compared to the original level, we are still only at -50 and at 24 bit there is still over 80 db down below -50 even before any of that distortion that you are rightly talking about would even come into effect.

Bass does not benefit from being recorded hot either. All frequencies will sound the same pretty well at any level.

Jeff Evans


Just out of interest the SSL console I was referring to in one of my previous posts is a fully analog console only. All A to D and D to A in our case is handled by AVID HD 192 interfaces outside. (3 to be precise)

But you still need an rms ref level at some reference point and SSL were suggesting the ref levels I mentioned before eg -24 -22, -20, -18 etc. -20 is a good one because it matches the K level. The VU was calibrated at -20 on the particular console we have had installed at the factory.

Ben some of your ideas are not over wrong in that recording hot was a good thing to do in the past. (I spent many years working with EMU samplers from 8 bit into 12 bit and 14 bit so I know what you are talking about. You did have to sample hot back in those days otherwise it sounded just plain bad!) But now with both 24 bit recording and the fact that almost any converter you buy now is going to be pretty excellent in the scheme of things so now we can virtually relax levels wise now and just work with say a K ref level of -20 without any concerns that things are not being recorded well.

Even if we are at -20 for example and the music dropped down 30 db below that which is very very quiet compared to the original level, we are still only at -50 and at 24 bit there is still over 80 db down below -50 even before any of that distortion that you are rightly talking about would even come into effect.

Bass does not benefit from being recorded hot either. All frequencies will sound the same pretty well at any level.

Ok I am going to play nice and not be rude, I put my gun away and you put your's away.
 
Do we agree that 24 bit is not 24 bit, the best converter's use a dynamic range of around 118db not the full spectrum of a 24bit 144db.  As I have mentioned the US Lexicon series use converters that use only 101db of 144db spectrum.  So 5 db above 16 bit.
 
Therefore the theory still has validity when it comes to lesser interfaces and converters.
 
Now even Presonus which have some of the best cheap interfaces on the market, I think if I remember rightly their converters use 114db of the 144db 24 bit spectrum, because they are so cheap and even though they claim to have low jitter specs, I still see the theory being of use because of the jitter problem and the distortion that jitter introduces.
 
I know I am vering off a bit here but do you get what I mean??
 
I am going to look at that Paul Frindle gearslutz post because he had posted about digital myths and this may help clarify what we are talking about.
 
Neb

BenMMusTech


Ok I got that drew but if the harmonics are all in the area where the quantization is happening and or below and even though we can still hear them, because of dithering and the quantization errors is it not possible we are still getting the harmonics but at a reduced fidelity.

Does that make sense??

Neb

What you're talking about is stuff that's already below the noise floor before your signal gets quantized. Since we use a quantization level even lower than the noise floor, we end up with quantization error (that sounds like noise) at a level below the existing noise floor. And keep in mind the dithering we are talking about here is not something we're adding - it's the "self-dither" from the background noise in the room, analog signal chain, etc. 

So you're talking about stuff that you can hear below the existing noise floor, but that will be masked by a little teensie, tiny bit of additional (quantization) noise? In the real world that's just not likely to happen, because it can only happen with a combination of all the factors where everything is just exactly so.

Ben what you are talking about now is the useable dynamic range that any converter is to actually going to give you. But that might be hard to quantify because while one converter might only use 101 dB out of the available 144 dB of range another might be better and in fact achieve say 120 dB out of 144 dB. How do we measure it? It is a good point and warrants some research into the subject to see how much actual dynamic range is available. I have heard that most converters use about 6 dB less dynamic range that is fully or theory wise avalaible. eg 16 bit in reality is 90 db of range and 24 is supposed to be 138 dB

But even if it varies etc I still feel that it is probably still a lot of dynamic range that is available. Even if a converter only used 101 dB of range and you had a signal way down at -50 then there is still 51 dB of range below that before the noise is going to even become an issue. Don't forget the -50 db signal is already 30 db down (from -20) and that represents a very quiet sound in itself.

I do agree about something though and that is the fact the final medium we are listening to is say 16 bit and I think it is good to make sure your final 16 bit master is reasonably loud and that still holds. Even a K-12 level is pretty loud and hot and still allows for plenty of transients.

Ok I will put it to bed but I will keep doing things my way!!

We will have to agree to disagree.

Peace Neb

Jeff Evans


Ben what you are talking about now is the useable dynamic range that any converter is to actually going to give you. But that might be hard to quantify because while one converter might only use 101 dB out of the available 144 dB of range another might be better and in fact achieve say 120 dB out of 144 dB. How do we measure it? It is a good point and warrants some research into the subject to see how much actual dynamic range is available. I have heard that most converters use about 6 dB less dynamic range that is fully or theory wise avalaible. eg 16 bit in reality is 90 db of range and 24 is supposed to be 138 dB

But even if it varies etc I still feel that it is probably still a lot of dynamic range that is available. Even if a converter only used 101 dB of range and you had a signal way down at -50 then there is still 51 dB of range below that before the noise is going to even become an issue. Don't forget the -50 db signal is already 30 db down (from -20) and that represents a very quiet sound in itself.

I do agree about something though and that is the fact the final medium we are listening to is say 16 bit and I think it is good to make sure your final 16 bit master is reasonably loud and that still holds. Even a K-12 level is pretty loud and hot and still allows for plenty of transients.

Check your specs Jeff on whatever interface you are using, I can't find mine, MOTU are very secretive but I believe mine are around 110db or that's what they use of the 144db of dynamic range, so in reality it's more like 20 bit not 24 bit. My interface that is.
 
Neb

BenMMusTech


Now even Presonus which have some of the best cheap interfaces on the market, I think if I remember rightly their converters use 114db of the 144db 24 bit spectrum, because they are so cheap and even though they claim to have low jitter specs, I still see the theory being of use because of the jitter problem and the distortion that jitter introduces.
   

Hmm. I wonder if I should point out that the level of jitter distortion is relative to the signal level, so if you have a hotter signal you get proportionally hotter jitter distortion too?

drewfx1

BenMMusTech


Now even Presonus which have some of the best cheap interfaces on the market, I think if I remember rightly their converters use 114db of the 144db 24 bit spectrum, because they are so cheap and even though they claim to have low jitter specs, I still see the theory being of use because of the jitter problem and the distortion that jitter introduces.
 

Hmm. I wonder if I should point out that the level of jitter distortion is relative to the signal level, so if you have a hotter signal you get proportionally hotter jitter distortion too?

See we learn so much because I am an ass!! He Haw He Haw
 
Neb
 
Actually I'm going to find that digital myths thread

Ben good idea about checking my specs. I thought of that too. I use a Yamaha digital mixer 01V as my main central device and it handles all A to D D to A etc.

In the manual they say the settings are 16 bit and 24 bit. They do say however the maximum signal to noise ratio is in digital in / out mode and it is quoted as 110 db which is obviously referring to 24 bit mode. In another part of the spec they also say the equivalent input noise is -128 dB. Not sure how that relates to the 110 dB spec. There is also the S/N ratio of the whole mixer input mic pre (analog) to stereeo out (analog) and that is 98 dB

I also found out that the first 12 inputs that are analog are only 20 bit. So the A to D there is at 20 bit only. I did not know this. But that is still prety decent and I am not complaining.

Looks like Yamaha don't mind giving some specs related to this.

Yes so the whole ADDA becomes so weird and this is possibly why we still do not have 32 bit audio interfaces, they still haven't figured out how to make propper 24 bit converters.

The plot thickens!!!

Neb

As I understand it, almost none (perhaps actually none) of the interfaces sold as 24 bit actually achieve 24 bits of meaningful recording, and 20 bit is about standard.

As I understand it, almost none (perhaps actually none) of the interfaces sold as 24 bit actually achieve 24 bits of meaningful recording, and 20 bit is about standard.

Correct. The voltage levels recorded in the least-significant bits are so tiny they fall into the range of thermal self-noise in a resistor. The bottom 4 bits are therefore pretty much random noise.

Do not read this thread, it will give you a headache. There is a difference between 0dbFS and 0dBu. There is no source input that will let you acheive more than the convertor can handle which is 0dBu not the peak of 0dBFS which is just oversampling for something that you can not record into. You have no source input that can acheive anything such recording any better and you must not understand chips and convertors or anything else. The limit is 129E.I.N. or the bandwidth continues to go smaller and smaller and less and less. You can not acheive what they can not build.

Better take your pill, spacealf. You're getting that wild look in your eye again...

If you want a real headache, make yourself a pot of strong coffee, set aside a few hours and read this

I did read that mostly and I agree. I was just thinking to myself that despite the analog (and not the age) I like '60's recordings better. Why? Because to me they sounded more alive in the end, and today music ends up sounding overboard, something, just wrong to me with a lot of it. Well, each to their own in the end, but I fail to see the selling point of hammering heads against the wall like some music is today.