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!