Metering in digital domain

My point was essentially that YES, digitally, signals are supposed to sound the same at any given level on paper but the reality of the matter is that at some point the signal will be either converted to digital or converted to analogue and will have to pass through an analogue stage. With recording this is doubly so because you will most likely have a front end like a mixer or multiple preamps before the inputs of the converters, which also incorporate their own analogue stage. This, essentially, is why it's important to be conscious of your levels and is why, in my opinion, converters and hardware with higher quality components and higher headroom sounds better at these levels and why budget gear falls apart. Digitally, yes, they're exactly the same but once an analogue stage is hit where voltage is concerned, levels become more critical.

My argument with ITB and plugins was that the reciprocal effect of slamming your levels within plugins (that may or may not be stable at these levels) and at the track level, while the software is technically able to handle these levels in a floating point environment, will cause overload and distortion once it his the reconstruction phase of the DA. As Massive pointed out, it's about voltage, and keeping conservative levels ensures proper voltage operation and that makes perfect sense to me. Further, to imply that we should all test each and every one of our plugins scientifically whether it's through FFT, null tests, etc, seems like something a scientist would do and audio engineers are not scientists. We are technical artists who use our ears to evaluate the technology. A processor should be selected based on it's functionality and overall sound and, I don't know about you, but I don't need a null test to tell me what sounds good. It either does, or it doesn't. Michael Schumacher does not need to know the science behind the internal combustion engine to be a good driver.

Preamps are the same story. In a digital system they are going to hand off the signal to a converter at some point and the converters' robustness to the signal will be heavily based upon it's analogue components. If these components go into distortion (which is not always immediately heard) at a few dB below clip point and it is being fed a hot signal from the preamp, the resulting sound will most certainly suffer. Of course, if this is your goal, then fine. Bob Katz talks about this at length in his Mastering Audio book. I suggest you all read it if you haven't already.

Finally, my views are wide and holistic. To me it's irrelevant to argue what will happen in a strictly digital environment because without the analogue input and output phases of the conversion process, we won't have audio. They all inter-relate. That's obvious, I know, but saying it's OK to have hot levels in a floating point environment because on paper the software can handle it without overs creates all sorts of variables that put a spanner in the workflow. Yes, the internal precision of the software can't handle it but can my plugins? Ok, let's null test them all. 10 hours later... Ok, so now what's happening at the DA? More tests.... Eventually you've wasted so much time trying to evaluate this the mix is no closer to being finished and you've wasted all your time on doing silly tests like a scientist would when you should just get on with being an audio ENGINEER.

Over and out.

Cheers :)
 
With many floating-point systems, to some extent, yes - you can lower the master fader. It's not what most would consider "good technique" but it'll avoid clipping the buss.

As far as the INPUT chain is concerned, it's completely unrelated -- It'd be like cooking a steak till it's completely burnt and then pouring ice cubes over it to make it more rare. It doesn't work like that.

:laughings:

That's a funny analogy.

I agree it's not good technique to rely on the digital master fader...but for the guys working ITB, they are often fooled by the floating-point systems, so they never hear the issues until it's coming out in analog again...and at that point, individually lowering track faders and plug-in levels becomes a mixing nightmare....so, just drop your master fader and finish your mix.

I still think the problems are in that ITB world because the DAW systems simply ignore/overcome level issues, and it is often transparent to the user until it's being mixdown and out to analog again.
At the front end, if people focus on preamp levels...the converter digital input levels kinda fall right into place where they should be....but...if people then try to adjust their digital tracks to compensate for monitoring levels...that's when the problems start, and the DAW forgives until it's too late.
 
My argument with ITB and plugins was that the reciprocal effect of slamming your levels within plugins (that may or may not be stable at these levels) and at the track level, while the software is technically able to handle these levels in a floating point environment, will cause overload and distortion once it his the reconstruction phase of the DA.

Where are you getting your information? Please define "unstable," and explain what that means in terms of how a plug-in responds to audio at high versus low signal levels. Be as specific as possible.

With the 32-bit FP math almost every DAW uses, there is no difference between having each track peak at 0 dB and reducing the master fader by 20 dB to compensate, versus lowering each track and leaving the master fader at zero. As long as you don't clip the D/A on the way out, everything will be fine. If you have evidence to the contrary, I'd love to see it.

As Massive pointed out, it's about voltage, and keeping conservative levels ensures proper voltage operation and that makes perfect sense to me.

It may make perfect sense to you but it's incorrect! Have you ever tested what you are claiming? Earlier John said that preamps will sound worse when outputting 3 or 4 volts versus 1 volt. But that too is wrong, unless the preamp is really lame. I just looked at the specs for my 20-year old Mackie 1202 mixer, and its distortion is spec'd at 0.0007 percent at 14 dBu, which is 3.9 volts. It is so easy to prove these claims are myths, I honestly don't know why we're still discussing them.

Further, to imply that we should all test each and every one of our plugins scientifically whether it's through FFT, null tests, etc, seems like something a scientist would do and audio engineers are not scientists.

So you are arguing for ignorance? Do you not want to know at what levels your gear can and cannot provide good quality? Do you really think Michael Schumacher doesn't understand how his cars work at a highly technical level? Look, I understand that not everyone cares how their audio gear works. That's fine. But IMO it's a big problem when those who don't understand the science advise others and spread misinformation.

Again, my intent is to dispel the many audio myths that confuse people with misinformation. Digital hardware and software do not sound better at low levels versus 1 dB shy of clipping. Analog hardware does not sound better at 0 dBu than at 15 dBu. And it's equally a myth that tracks "stack" and accumulate losses in quality. In fact it's exactly the opposite.

You talked about wasting time and energy. The reason its important to dispel these myths is not only truth for truth's sake, but to avoid people wasting time and money chasing things they wrongly believe will improve the quality of their recordings. Once people realize that buying a $2,000 outboard summing box will not magically make their mixes sound more professional, they can focus on stuff that really does matter like mic placement and practicing their craft.

--Ethan
 
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I am seeing it this way. Since there is such a low noise floor and you can get everything sounding great with lower levels and more headroom what is the point of doing it any other way? Maybe like some have said with a hybrid system but if its all ITB it makes more sense to stay as far away from clipping as you can as long as you maintain a good sounding signal. I also think that it is good practice to stay away from the master fader and just keep everything in check down the line.
 
Where are you getting your information? Please define "unstable," and explain what that means in terms of how a plug-in responds to audio at high versus low signal levels. Be as specific as possible.

From a plugin developer that I know who shall remain nameless:

plug-ins internally represent the (audio-) sample values with floating-point values between -1 and +1. Those 2 values, when put on the output of a plug-in, correspond to 0db (+1 is the maximum positive phase, -1 is the maximum negative phase. Think of it as the maximum values of a sine). For ease of explanation I use +1 below, but for -1 it's the same.

Depending on what the plug-in does there's a lot of mathematics inside, additions, multiplications and whatnot. This processing results often in (internal) temporary values bigger than +1. That would 'normally' be not a problem as long as after all those internal processing steps everything's back to -1 ... +1, but...

...a lot, and with that I mean a LOT, DSP algorithms are designed to work ONLY in this range. Some work for all values, some more work for values slightly above +1, some work bad outside that range, some just crash or start to oscillate.

Unfortunately a LOT of plug-in programmers don't know shit about mathematics, so they just put some fancy DSP algorithms they've found in the internet/books inside their plug-in code and call it a day, without thinking twice what their code will do when used outside its designed range.

When you leave 10db headroom the chances that the plug-in behaves properly (and sounds best) are much higher. So, if in doubt, leave more headroom. It never hurts, and you can easily make up the gain to proper loudness levels at the end(!) of the processing chain.

With the 32-bit FP math almost every DAW uses, there is no difference between having each track peak at 0 dB and reducing the master fader by 20 dB to compensate, versus lowering each track and leaving the master fader at zero. As long as you don't clip the D/A on the way out, everything will be fine. If you have evidence to the contrary, I'd love to see it.

What I'd love to see is this purely digital world you're living in. How does one hear audio without the analogue stage?

It may make perfect sense to you but it's incorrect! Have you ever tested what you are claiming? Earlier John said that preamps will sound worse when outputting 3 or 4 volts versus 1 volt. But that too is wrong, unless the preamp is really lame. I just looked at the specs for my 20-year old Mackie 1202 mixer, and its distortion is spec'd at 0.0007 percent at 14 dBu, which is 3.9 volts. It is so easy to prove these claims are myths, I honestly don't know why we're still discussing them.

We're discussing them because you're as sure as I am about what you believe is right and have not considered that your empirical evidence may be correct on paper but incorrect in real life.

From Bob Katz:

Bob Katz said:
Cheap digital equipment is subject to edgy sounding distortion which can be caused by sharp filters, low sample rates, poor conversion technology, low resolution, (short wordlength), poor analog stages, jitter, improper dither, clock leakage in analog stages due to bad circuit design, and many others, such as placing sensitive A/D and D/A converters inside the same chassis with motors and spinning heads. It takes superior power supply and shielding design to make an integrated digital tape recorder that sounds good...

Bob Katz said:
Just use the level provided by your console manufacturer, right? Well, maybe not. +4 dBv (reference .775 volts) may be a bad choice of reference level. Let's examine some factors you may not have considered when deciding on an in-house standard analog (voltage) level. When was the last time you checked the clipping point of your console and outboard gear? Before the advent of inexpensive 8-buss consoles, most professional consoles' clipping points were +24 dBv or higher. A frequent compromise in low-priced console design is to use internal circuits that clip around +20 dBv (7.75 volts). This can be a big impediment to clean audio, especially when cascading stages (how many of those amplifiers are between your source and your multitrack?). In my opinion, to avoid the "solid state edginess" that plagues a lot of modern equipment, the minimum clip level of every amplifier in your system should be 6 dB above the potential peak level of the music. The reason: Many opamps and other solid state circuits exhibit an extreme distortion increase long before they reach the actual clipping point. This means at least +30 dBv (24.5 volts RMS) if 0 VU is+4 dBv.

Yeah, there are a lot of folks who are not in accordance with your views and, as Mixerman found out, you are stubborn as hell to relinquish them.

I suggest everyone reads the rest of the Bob Katz article here:

Level Practices (Part 1) | Bob Katz

So you are arguing for ignorance? Do you not want to know at what levels your gear can and cannot provide good quality? Do you really think Michael Schumacher doesn't understand how his cars work at a highly technical level? Look, I understand that not everyone cares how their audio gear works. That's fine. But IMO it's a big problem when those who don't understand the science advise others and spread misinformation.

I am arguing for listening and sensibility. You are arguing a case where science should somehow have precedence over listening. I could give a shit less about the science because not knowing it's intricacies has never stopped me from making a record or having a hit song.

Again, my intent is to dispel the many audio myths that confuse people with misinformation. Digital hardware and software do not sound better at low levels versus 1 dB shy of clipping. Analog hardware does not sound better at 0 dBu than at 15 dBu. And it's equally a myth that tracks "stack" and accumulate losses in quality. In fact it's exactly the opposite.

The opposite in that low level distortion combined in many tracks results in an INCREASE in quality?

You talked about wasting time and energy. The reason its important to dispel these myths is not only truth for truth's sake, but to avoid people wasting time and money chasing things they wrongly believe will improve the quality of their recordings. Once people realize that buying a $2,000 outboard summing box will not magically make their mixes sound more professional, they can focus on stuff that really does matter like mic placement and practicing their craft.

--Ethan

Well, I bought a $2000 summing box and I can hear a difference. I don't feel the need to prove it scientifically. And any professional worth his salt knows that being a "professional" is an attitude and skillset, not a platform. Let them eat cake!

The problem with overly-scientific minds is that they try to reduce everything to numbers when clearly reality is far more complex than that. I believe in science and I love technology, but I'm also a lover of the ART of music and the craft of audio production. As usual, this is a forest for the trees issue and full of extremist views. I'll leave it at that.

I'm tired of this now. Excuse me while I go walk the walk instead of talk the talk.

Cheers :)
 
:laughings:

I still think the problems are in that ITB world because the DAW systems simply ignore/overcome level issues, and it is often transparent to the user until it's being mixdown and out to analog again.
At the front end, if people focus on preamp levels...the converter digital input levels kinda fall right into place where they should be....but...if people then try to adjust their digital tracks to compensate for monitoring levels...that's when the problems start, and the DAW forgives until it's too late.

This is where i would go wrong when first mixing. I would leave my monitors at lower volume and try to compensate with driving the signals too hot. When I turned the monitors up and kept my signals in check everything started to sound better. I would also think that i was getting a great mix but as soon as I exported and burned to disc the mix fell apart on everything except on my mixing monitors. It seemed distorted and overblown to my ears. Now that I'm getting a firmer grasp on gain staging and understanding the difference between digital and analog the mixes are punchier, clearer and seem to have a more dynamic range.
 
You should also check your monitor/room combination....make sure the room isn't contributing in a negative way to what the monitors are telling you.
 
You should also check your monitor/room combination....make sure the room isn't contributing in a negative way to what the monitors are telling you.

I am about to do a remodel of my studio and really work on treating every room the right way. The room that i mix in now is far from ideal.
 
The manual of my Firepod says to record as hot as you can without clipping. Internet "wisdom" always says "somewhere around -18 dbfs". Who is one supposed to believe? I dunno. I just believe my ears, do whatever I want, don't worry about it, and make music..
 
Icing a steak after you've finished cooking it WILL render it less cooked, (& therefor more rare), than just removing it from the heat & leaving it. Things "cook on" based on internal heat etc.
There's needs to be a RED BOOK or WHITE PAPER that all hard & software companies producing digital audio gizmos should comply with setting out just what 0 means as, currently we're runningw ith Freezing Farenhiet, freezing Celsius & have Absolute Kelvin running around dithering about it.
I rely on my lousy ears but I often pushed my tape recorder into the red because it sounded cool & keep my digital tracking lowish because clipping sounds awful.
 
Since there is such a low noise floor and you can get everything sounding great with lower levels and more headroom what is the point of doing it any other way?

There's nothing wrong with doing that. What is wrong is telling people that the sound quality is better 20 dB below digital zero because that's simply not true, as proven by the example files I posted. Did you play those files? Can you tell which was recorded near zero and which was recorded 20 dB softer?

if its all ITB it makes more sense to stay as far away from clipping as you can as long as you maintain a good sounding signal.

There is no clipping ITB. That's one huge feature of using 32-bit FP math. If there's some part of this you don't understand, let me know and I'll try to explain it again, even though it's demonstrated in detail in my AES Audio Myths video I've linked to several times now.

--Ethan
 
From a plugin developer that I know who shall remain nameless:

All I see there is that incompetent programmers can create incompetent code. Well duh.

We're discussing them because you're as sure as I am about what you believe is right and have not considered that your empirical evidence may be correct on paper but incorrect in real life.

Apparently you don't understand what empirical evidence means. It means I already proved that audio doesn't sound "better" when it's recorded 20 dB below zero, and preamps don't sound "worse" when outputting 4 volts compared to 1 volt. I have no idea why you continue trying to obfuscate the discussion away from these two core issues of this thread.

Yeah, there are a lot of folks who are not in accordance with your views and, as Mixerman found out, you are stubborn as hell to relinquish them.

I already proved my points beyond doubt with hard data and actual recorded files. That you continue to argue just for the sake of arguing shows that in fact it is you who is being stubborn. It also shows that you are the one doing the thread-jacking, by continuing to argue other topics because you're unable to discuss (or understand) the current topic.

I could give a shit less about the science

Again you argue in favor of ignorance. Amazing.

The opposite in that low level distortion combined in many tracks results in an INCREASE in quality?

Keeping multiple sources separate and passing each through its own "not good" preamp or converter (stacking) damages the sound less than mixing the same sources together cleanly and passing the mix through a single "not good" device.

--Ethan
 
Here's one
http://www.waves.com/Manuals/Plugins/kmt-white-papers.pdf
I also linked to several others a few pages ago

I read that entire PDF article because it was interesting. Alas, there wasn't one mention of plug-ins being "unstable" at levels above zero. Further, that's exactly the sort of plug-in I already mentioned is excluded from this discussion because its purpose is to add intentional coloration at higher levels. Normal 32-bit FP plug-ins do not do that, and they can accept a huge range of levels without sounding different.

I also looked through all of your previous posts to this thread and saw no other links.

--Ethan
 
There is no clipping ITB. That's one huge feature of using 32-bit FP math. If there's some part of this you don't understand, let me know and I'll try to explain it again, even though it's demonstrated in detail in my AES Audio Myths video I've linked to several times now.

--Ethan

I understand completely but at some point it will get converted to analog again and any clipping will be very apparent at that point in the process. If everyone were listening to the files through the software then maybe it wouldn't be much of an issue, although impossible for consumers who usually listen to cds and mp3 files.
 
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I read that entire PDF article because it was interesting. Alas, there wasn't one mention of plug-ins being "unstable" at levels above zero. Further, that's exactly the sort of plug-in I already mentioned is excluded from this discussion because its purpose is to add intentional coloration at higher levels. Normal 32-bit FP plug-ins do not do that, and they can accept a huge range of levels without sounding different.

I also looked through all of your previous posts to this thread and saw no other links.

--Ethan

I think that the point he was making with that link was that the operation of the plugin was that the signal being fed into it was meant to replicate the original hardware and react to the input. This is one of the plugins that will react differently when driving the signal hotter so it will react as if you were pushing the signal like it was actually tape. It even tells you the results of the effect at certain input levels. My terminology might not be right but I am pretty sure im getting the right idea behind the way it operates. So im guessing that running the signal hotter doesnt make it unstable it makes it work the way it was designed work.

Operating Level. For example, a tape that was designed to record signals at 250 nWb/m was said to be recording at +3 dB (over the Standard Operating Level of 185 nWb/m). As a point of reference, since the Flux Control on Kramer Master Tape is calibrated in nWb/m, here is a quick reference guide for comparison (Source Quantegy):

-2 dB = 150 nWb/m
0 dB = 185 nWb/m (Standard Operating Level)
+3 dB = 250 nWb/m
+5 dB = 320 nWb/m
+6 dB = 370 nWb/m
+9 dB = 520 nWb/m
 
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I understand completely but at some point it will get converted to analog again and any clipping will be very apparent at that point in the process. If everyone were listening to the files through the software then maybe it wouldn't be much of an issue, although impossible for consumers who usually listen to cds and mp3 files.

You're still missing something very important. Yes, obviously if a 32-bit FP digital signal exceeds full scale, it must be turned down at some point before it goes out to the sound card or is rendered to a Wave file. But that's also true when "listening to the files through the software" as you put it. This is why it's perfectly acceptable to let a mix "grow" to well past zero, as long as the volume is reduced at the master output bus to avoid distortion.

So im guessing that running the signal hotter doesnt make it unstable it makes it work the way it was designed work.

Exactly. Which is why that article isn't relevant to this discussion.

--Ethan
 
Exactly. Which is why that article isn't relevant to this discussion.

--Ethan

Well I thought that i read somewhere that you said the input signal through a plugin chain will not effect the way a plugin responds. And that is obviously not true with the Kramer tape and certain other plugins. I also get that you can turn down the master fader but why do that if you can just keep every channel at a level that will not stack up to clip the master? But since you have dedicated so much time in making this thread your forum to turn things so scientific I feel that im loosing my inspiration so I will do my own tests and see what works for myself. Thanks for your extremely analytical views but im getting back to the music. If anyone else can add their thoughts I would like to read their views but out of everything ive been reading from very successful and acclaimed audio engineers it seems that you take the art away from the art. I am not trying to be offensive but why try to disprove things that are working very well for others? It is pointless when the proof is in the pudding and these guys are making some great sounding stuff.
 
I thought that i read somewhere that you said the input signal through a plugin chain will not effect the way a plugin responds. And that is obviously not true with the Kramer tape and certain other plugins.

From my second post to this thread:

No, what I'm saying is most plug-ins sound exactly the same no matter what level you throw at them. Now, this is not true for all plug-ins, but it is for most of them.

From my fourth post to this thread:

some plug-ins add intentional analog "flavor" that increases at higher levels. Other 32-bit plug-ins are not affected.

So it's not like I didn't explain this already, twice.

I also get that you can turn down the master fader but why do that if you can just keep every channel at a level that will not stack up to clip the master?

Sure, that's fine. But sometimes as you work on a mix you turn up the bass, then realize the rhythm guitar is too soft, and next thing you know all of the tracks together exceed digital zero. You could lower every track by 4 dB or whatever, but that's a nuisance when there are 50 tracks, especially if you have detailed volume automation that must also be adjusted. My point in this thread is not to tell people how to work. Only to explain that you don't have to worry about this stuff, even as others wrongly claim that you do have to worry.

But since you have dedicated so much time in making this thread your forum to turn things so scientific I feel that im loosing my inspiration so I will do my own tests and see what works for myself. Thanks for your extremely analytical views but im getting back to the music.

I honestly don't understand your hostility. :confused:

I'm trying to help you! However, I absolutely support your desire to experiment for yourself. If only others would do that we'd have a lot less contention and less misinformation!

It is pointless when the proof is in the pudding and these guys are making some great sounding stuff.

Actually, I'm still waiting for Mister Facta to list the hit records he's produced, or even post some of his music for us to listen to. I'm also waiting for him or anyone else to say which of those two mix files I posted were made from tracks recorded hot or not.

--Ethan
 
Thanks for your extremely analytical views but im getting back to the music. If anyone else can add their thoughts I would like to read their views but out of everything ive been reading from very successful and acclaimed audio engineers it seems that you take the art away from the art.

I'll just say that Ethan means well here...and he certainly is an artist/musician himself, so don't think he's just crunching numbers. :D

I've had a few debates with Ethan about some things in the past, and I don't think he's ever had any ulterior motives...he just tries to disprove certain audio myths that have been around for a long time. He may not always convert everyone over...but he does alway make strong points and usually backs them with solid science.
That said...I got no problem with audio myths either...as long as they help get you where you want to go.

I still track at 24/88.2....yet I know that Ethan thinks that's just a waste of good hard drive space. In the end, I don't much worry about the HD space....so I keep recording at 88.2. :)

Anyway...I think in this thread there are two main points that are related, yet seem to be fighting with each other.

The tracking levels, yes, you can record them lower rather than higher, and it will work perfectly...and I do think Ethan agrees with that. The point he is making is that if the levels are on the higher end (and we don't need to always go from -18 dBFS all the way to 0 dBFS to call it the higher end)...stuff in the -8, -6 range...
...it will be just as fine as the -18.
The counterargument is not about those digital levels...but what was happening at the analog front end to get them.
Yes, some preamps could get more "fuzzy" when pushed harder to get you that -6 dBFS digital signal, and that's why some folks say, keep it lower, don't strain your analog front end just so you can get higher digital levels.
I said earlier that I pay NO attention to my digital input levels...only my analog front end levels.
I've got a few decent pres, and I can guarantee that a couple of them could easily push out enough analog level without distortion to bring the digital levels up into the red zone.
Sometimes THAT is where I want my analog level...not always though.
I have other preamps that certainly DO get "fuzzy" when pushed hard, and they too can kick out a pretty hot analog signal that will make the digital level high...and sometimes that's the sound I like.

So really it's all about the analog front end levels and sound you are going for...and IMHO...NOT about your digital level.
Yes, with most analog gear, if you are seeing -18 to -14 dBFS on your digital meters, it means the analog front end is probably running a healthy level that will work fine in most cases....but I certainly would not set my front end just by looking at my digital level and trying to keep it in the -18 to -14 dBFS range.
It certainly will fall there often...but I still think the best thing is to use the sound and level of the analog gear as you guide, and let the digital levels fall where they will.
Granted...lots of combination preamp/converter interfaces only give you the digital input level to view...so that is what it is.

You can play it safe, and most times it will be fine...but sometimes the best sounds come from pushing analog gear into the hot zone, and some pres certainly can kick out a high level, and your digital meters might be closer to -10...-8, especially on dynamic peaks.
Don't sweat that as long as your pre is working as it should and you like what you are hearing.

And that is mainly what Ethan is saying with his analytical views, and instead focus on the music.....as I read it.


The other main point is that once your audio is all digital...it's just numbers after that with most DAWs these days.
So as you edit/comp/process/mix...even if your original/raw signal was conservative, it's easy for it to creep up...but there's no need to keep pulling individual faders back if you have a nice mix going...just pull down the Master Fader.
It's the same thing....just number crunching.
AFA some plugs that directly work off the track levels...well, just adjust them as needed. IOW...you don't really have to keep your initial levels low just 'cuz you might at some point add a plug that needs less level.
Again...once it is all digital...just move the level where it needs to go...it's all the same.
When you get that final mix, and it's quite hot (but not clipping, thanks to 32 bit float-point math)...again, just drop your Master fader.
In this regard too, Ethan is right...he IS saying that you don't need to be all analytical about the level...just put it where it needs to go, and your audio signal will not suffer for the number chrunching...so you can just focus on the art. :)

Oh...you want to get Ethan going some more...talk about analog tape recording! ;)
I still track to tape...Ethan use to, but hasn't in many, many years.
It's all good....focus on the art.
 
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