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

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

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

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!

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

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.

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

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 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

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.

@ 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. 

"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

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?

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.