Analog v Digital mixing is - 20dB on utility the answer ?

[3phase]

No.

Just another bad myth based on engineers not understanding what they are using or basic physics being spread around the internet as fact.

0dBFS = 0dBVU on a digital system and there's no correlation with an analogue set up.

thats of cause just bullshit.. because every digital signal has to hit the analog world at one point.


i dont know what you mean with dbvu

but because the digital scale counts from negativ values up to a max of 0dbFS that equals +22dbu

Not bullshit at all. 0dBFS does not equal +22dBU at all, it can be any number you want it to be, hence why you get different figures depending on what analogue equipment you are comparing to and what standard you are working to.

Obviously every channel hitting 0dBFS isn't going to sound good and you will have to turn down, but the amount you have to turn down by is enough to prevent all channels, including the master from ever hitting 0dBFS or -3, -6, -12 etc if you're sending it to be mastered and every mix is different.

Analogue & digital mixing are ultimately completely different and should not be directly compared as it is leading to bad advice and just because something is on Gearslutz, doesn't make it true or accurate.

You was talking bullshit when you say that ther is no relation between the digi and analog world because there most defenetly is one .. you dont have an usb port under your ear...


And its bullshit that 0ddFS equals 0 dbu... or that it can be everything you want.. there are international audiostandrds that dont seize to exsist because some ableton fanboys claim them to be unrelevant...

thats not how things work in the audio industry.

How that headroom is defined is of cause something soundengineers have decided at one point to have a standard to orientate...
And in the beginning ther was only national broadcast standrds..but they became international ..
even the ol nazi +6db is falling wright now in favor of the international +4db on zero vu

You are wright when you say that someone that is only in the box dont needs to care too much about. Or not as much as one working in the analog domain...

But that dont says that the advice form the gearslut guy is wrong...

The gearsluts articel is mainly reffering to mixing in the analog world or to insert analog processors into an in the box mix...

by keeping max quality be applying the analog gain structure to the digital world.. what was before programs like live actually essential in the digital world too.. All the first HD systems was build to match and replace analog devices in the pro studios..

So up to today you can overload busses easily in protools.. and many plugs are designed towards realworld demands and overload at 0dbfs...
An accurate model of an analog compressor defently should sound on 0dbFS as if you would drive the real unit this hot..
That applies for example to some UAD plugs..so..when you use them fanboy style..you dont run them at theier optimal sounding range...

other plugs that alow any internal overload give up on some possible max quality.
therfore you can internally overload wave plug ins easily, because they value quality over newby compatibility.

again.. the gearslut gives good advice to have some headroom also in the digital domain.. helps in interfacing with analog world..maximizes quailty in the digital one..

if you want to be radical you only record all your signals so low that you can stay unity gain thruout your digital chain..
before hitting your ssl console .. or ..also wehn mixing in the box.. when you record all signals that you dont need to change a level aferwards anymore..you have maxed the quality because you dont alter the source material anymore...

that guranties the purrest sound quality.. but is a bit unpractical approach..

And...
your soundcard most defently dont sounds wright when having it to send out on 0dbFS.. because ..do that on a pro interface and measure.. you will read 22dbu..
because the people building theese devices are reffering to the audio standards you say dont matter...

22dbu is a lot.. there are machines out there where you can destroy the inputs with such levels

so it matters even more to the cheapo equipment that dont cares for professional standards that you leave headroom.

equipment that is designd after the studio standard has to deliver 22dbu..and be able to surrive 22dbu...( mic pres excluded)
with a behringer style device you just dont know..

So its no bad advice to leave some headroom in any case...
ANd to argue again and claiming the advice comes from people that dont have a clue is a very big bullshit..

it just shows that you dont understand the gearslut article.

[Sage]

The article refers to in-the-box mixing, not integrating analogue hardware/outboard with DAWs.

I've not been talking about analogue and how that works. If you don't understand what I'm getting at, don't start claiming what is & isn't bullshit.

And ok, a few plugs clip at a lower level than 0dBFS and some are designed to emulate a device running at it's maximum voltage at 0dBFS doesn't have anything to do with professionalism or amateurism. A UAD emulation wouldn't be much of an emulation if you couldn't run far too hot into it and use it "wrong" as well as correctly.


I'm not saying don't leave headroom, just there's no specific amount of headroom necessary for strictly ITB mixing, just make sure nothing is clipping at any point. And isn't analogue metering done in RMS? and digital can do Peak...

[3phase]

Just another bad myth based on engineers not understanding what they are using or basic physics being spread around the internet as fact.

may i call yo dumbot ?

engineers not understanding what they are using? ähemm.. the guy that gave the advice on gearsluts knows more about the stuff than you will ever know.. and let me guess..
knew that allready before you was even born? maybe?

and basic physics spread around as fact.. whooo.. thats bad ... basic physics is a weard science.

so just keep on having all your levels up to the max.. more helps more...

[evon]

This is all about headroom, signal to noise ratio and gain staging (gain structure) and how to optimize all of this. Therefore, to maximize the headroom available metering will be very critical. Now that I mention it, I havent heard anyone mention the different metering standards around and how to convert between them.

No one can deny that the headroom above the noise floor (96dB in some instances of resolution) is far more dynamic than human ears will have use for. And that of the anolog is far less (20dB).

I think this topic can get very technical, and at the end of the day is not as crucial as we are making it out to be. Ears, remember...that is where the proof of the pudding is. To argue about this seems like just a next battle in the Loudness War, and afterall loudness just dont seem to cut it anymore. Music and more dynamics in it is the top of the top IMO.

Mixing to anywhere below -3dBFS is loud enough

[3phase]

This is all about headroom, signal to noise ratio and gain staging (gain structure) and how to optimize all of this. Therefore, to maximize the headroom available metering will be very critical. Now that I mention it, I havent heard anyone mention the different metering standards around and how to convert between them.

No one can deny that the headroom above the noise floor (96dB in some instances of resolution) is far more dynamic than human ears will have use for. And that of the anolog is far less (20dB).

I think this topic can get very technical, and at the end of the day is not as crucial as we are making it out to be. Ears, remember...that is where the proof of the pudding is. To argue about this seems like just a next battle in the Loudness War, and afterall loudness just dont seem to cut it anymore. Music and more dynamics in it is the top of the top IMO.

Mixing to anywhere below -3dBFS is loud enough

true. due the loudness war everthing is pretty much a chaos..

however ..you just mention another audio myth.. that analog has much less dynamic than digital..

nominal level -100db = -96dbu plus headroom +18 dbu gives the typical dynamical range for broadcast quality analog audio equipment...
gives 114 db dynamical range above noisefloor under clip level.. but because its analoge ther is something below the noise level and something above the clip level... so the actual usable dynamical range of propper analog audio equipment is rather something like 140 db...( defently can hear signals 20 db below the noisefloor in analog recordings its still there its just noisy ) and so rivals the digital world.. beside only the best AD converters do 140 db...

its not noise and dynamic that caused the swing towards digital technology...and most defenetly not soundquality..
its rather an economic factor.
the reason evrything has to hit the 0dbFS boarder comes actually from the quality drawwbacks of digital audio... within the 16 bit domain you ve to hit the 0dbFS to have a similar dynamic than we have in the analog domain.. ok.. you ve less noise than.. but actually.. noise is just the dither of nature.. it dont hurts really when its just in the reverb tails...

however.. how much dynamic ther is in modern productions? 3 db?

[evon]

This is all about headroom, signal to noise ratio and gain staging (gain structure) and how to optimize all of this. Therefore, to maximize the headroom available metering will be very critical. Now that I mention it, I havent heard anyone mention the different metering standards around and how to convert between them.

No one can deny that the headroom above the noise floor (96dB in some instances of resolution) is far more dynamic than human ears will have use for. And that of the anolog is far less (20dB).

I think this topic can get very technical, and at the end of the day is not as crucial as we are making it out to be. Ears, remember...that is where the proof of the pudding is. To argue about this seems like just a next battle in the Loudness War, and afterall loudness just dont seem to cut it anymore. Music and more dynamics in it is the top of the top IMO.

Mixing to anywhere below -3dBFS is loud enough

its not noise and dynamic that caused the swing towards digital technology...and most defenetly not soundquality..
its rather an economic factor.
the reason evrything has to hit the 0dbFS boarder comes actually from the quality drawwbacks of digital audio... within the 16 bit domain you ve to hit the odbFS to have a similar dynamic than we have in the analog domain..

however.. how much dynamic ther is in modern productions? 3 db?

I never heard that before..I always thought that 0dBU was the lowest you could go in analog? And why does digital sound cleaner than analog? I have always thought that it was because of the lower noise floor.
Your statements seem rather radical. I am not saying you are wrong but I would need more facts to agree.

[3phase]

This is all about headroom, signal to noise ratio and gain staging (gain structure) and how to optimize all of this. Therefore, to maximize the headroom available metering will be very critical. Now that I mention it, I havent heard anyone mention the different metering standards around and how to convert between them.

No one can deny that the headroom above the noise floor (96dB in some instances of resolution) is far more dynamic than human ears will have use for. And that of the anolog is far less (20dB).

I think this topic can get very technical, and at the end of the day is not as crucial as we are making it out to be. Ears, remember...that is where the proof of the pudding is. To argue about this seems like just a next battle in the Loudness War, and afterall loudness just dont seem to cut it anymore. Music and more dynamics in it is the top of the top IMO.

Mixing to anywhere below -3dBFS is loud enough

its not noise and dynamic that caused the swing towards digital technology...and most defenetly not soundquality..
its rather an economic factor.
the reason evrything has to hit the 0dbFS boarder comes actually from the quality drawwbacks of digital audio... within the 16 bit domain you ve to hit the odbFS to have a similar dynamic than we have in the analog domain..

however.. how much dynamic ther is in modern productions? 3 db?

I never heard that before..I always thought that 0dBU was the lowest you could go in analog? And why does digital sound cleaner than analog? I have always thought that it was because of the lower noise floor.
Your statements seem rather radical. I am not saying you are wrong but I would need more facts to agree.

analog eqipment generates noise.. in the digital domain ther is no noise (at least theoreticaly)

so where you have the noisefloor in analog equipment.. typical at -100dbu its just a swamp of noise... within this noise you can well have still musical information.. like reverb tails.. or the conductors stick taping on the score sheet...

in the digital world the lowest value is zero.. thats silence.. ther is no noise below that..and also no reverb tails that could dive into it...

on the other side of the scale... in the digital world there is no more that max value..its an absolute border...

On an typical broadcast/studio console, that has to do +22db distortion free, you well can go a few db into the saturation before its starts to sound nasty.. depends on the desk..there are some that still sound good on +30dbu.. and have a noisefloor at -110 dbu..
so a real 140db dynamic, plus that what is going on inside the noise floor...

in digital the boundarys are like walls.. in analog they are rather liquid...


however.. within digital production ther is theoretical something like a noise free production... when you only use digital soundsources within the system and you dont program noise there is theoretical none.. but of cause.. ther are all kinds of artefacts during signal processing that qualify as noise aswell... but rather the eat your ear of kind of noise. crushed bits are crushed bits.. but on a lower level than in the usual 3 fx racks 1000 point patchbay analog production... noise builds up easily in an analog processing chain.. so you really deal with less noise in digital productions.. and also less distortions..therfore the cleaner sound..
but..we like the distortions and add tons of plugs to recreate them...as the actual plug in development clearly shows...

so.. you have more noise in a full analog production.. but usually more dynamic aswell.. the one dont excludes the other.

any Roland space echo generates more noise than all your plug ins together..but thats only on one channel and dont effects the total dynamic of the production..