Old 07-27-2008, 03:22 PM   #1
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Default Global gain setting

Many of you may not be familiar with the dbVU vs. dbFS dilemma - briefly explained: 0"db" on my console equals -18"db" in my DAW resp. converter. In an GS forum article I just read about this:
"In consoles like the Sony OXF-R3, we therefore had a 'headroom global value' that was set on boot up options, which compensated for the gains in all signal paths - and passed 'real operating level' values to all internal processing, so that they 'knew' the intended output target level and behaved appropriately, whatever headroom was set up."
Wouldn't this be possible in a DAW? This would lead to the best sound ever (didn't the Ensoniq Paris sound magic have its source in this fact?) - BUT: all plugins would behave completely "wrong". Imagine a limiters threshhold at min. -18db. Most dynamic plugins wouldn't be working or at least setting them up would be really strange and illogical.
My suggestion for a solution: If I choose a "global gain setting" at -18dbFS, inserting a plugin would (either automatically or manually, depending on the kind of plugin) boost the signal 18db BEFORE and attenuate 18db AFTER processing!
Shouldn't this be the way for proper gain staging par excellence? It would be sooo perfect to have, during tracking, a meter flashing at zero - and indeed it's -18, no?
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Old 07-28-2008, 01:06 AM   #2
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Gain staging in a DAW is not an issue - the headroom is so huge you can more or less do what you like.
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Old 07-29-2008, 01:56 AM   #3
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OK, that's what I thought: you still didn't get it... Sorry.

Another attempt:

let's have a signal peaking at -18dbFS (which would be very good soundwise, enough clean headroom, no prob). I want to compress this signal with VST compressor plugin "X". "X" does not feature any input/output trim knobs, so it's designed to behave just like its analog counterparts. How do they behave? Ever tried to compress acoustic guitar at really low level with a Joe Meek? Even if you crank up its input to the max, almost nothing will happen.
And in the digital/analog hybrid world it's even worse: I want to use BOTH - VST and hardware compressors.
My idea, as I explained in the original first thread, would be to provide virtual input/output compensation, at a fixed global gain that I can choose. In my case I would choose -18db because of the analog desk, and all plugins would behave like 0db. No?

Last edited by beingmf; 07-29-2008 at 02:17 AM. Reason: forgot one topic
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Old 07-29-2008, 02:24 AM   #4
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No, I still don't get what you mean (heh, and nobody else seems to be wanting to try! )

You seem to be expecting the program to affect what happens before the digits from the converter reach it. Reaper just takes what the converter passes to it. You set the level at the converter. In Reaper the headroom is, I think, something over 1500dB so like I said before, it seems to me that gain staging then starts to lose all meaning. Just make sure it's brought back to a non-overload situation on the master channel, otherwise indeed you'll make the DA stage of your converter unhappy. But you do have to make sure that your analog to digital conversion, before Reaper, makes the best use of the available headroom.

Quote:
The closer to 0db I record/play back the worse sound quality gets, in terms of harshness, digital inaccuracies. This is more or less a physical fact, due to the design of a ADDA converter.
It sounds like your converter is poorly designed or defective if that's your experience.
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Old 07-29-2008, 07:06 AM   #5
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i also don't really get it.

i just can say: if you wanna boost the signal pre-fx, you have to use the volume (pre-fx) envelope on the track to do this.

(still hope that we can get a trim/gain-knob in the mixer in the future for this) ...
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Old 07-29-2008, 07:13 AM   #6
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Also probably not understanding what you're after, but if the underlying issue is that you want to apply some compensation to the signal coming from and going to your AD/DA, couldn't you just record at whatever levels make the AD happy, normalize the recorded items (that is, increase their volume to peak wherever you want inside Reaper with no loss in resolution), do whatever processing you want, then set the master fader down to whatever level makes the DA happy?
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Old 07-29-2008, 09:26 AM   #7
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Quote:
Originally Posted by schwa View Post
Also probably not understanding what you're after, but if the underlying issue is that you want to apply some compensation to the signal coming from and going to your AD/DA, couldn't you just record at whatever levels make the AD happy, normalize the recorded items (that is, increase their volume to peak wherever you want inside Reaper with no loss in resolution), do whatever processing you want, then set the master fader down to whatever level makes the DA happy?
Yes that would be the solution I guess.

So I guess what the OP is saying is that ... okay I need to update everyone to get the basics about this topic I think... the db scale just expresses the magnitude of a signal to a reference level/voltage/power (or what ever unit is used as reference). So for dBFS this would be the reference to maximum representable value (known as full scale), for integer representation anyway or the as maximum specified value (the specified full scale if you want) as for instance in floating point DSP where the 1.0 is mapped to 0dBFS.

For hardware or anything other than digital you typically refer to a voltage (or a power) dBV (voltage relative to 1 volt rms across any impedance), then there is dBu (reference to 0.775 volt rms across any impedance), and many many more (such as dBW, etc...).
Funny thing, consumer gear runs at −10 dBV (that's why we got the -10dB switches) while pro gear ("line level") runs at +4 dBu that's why we got the +4dB switches (while +4dBu really is 1.23 volts rms = 1.8dBV ... but must stay on topic ....
Anyway, so what this 0 dBu = -18dBFS really means is that your ADC turns a 0 dBu signal into a -18dBFS signal.

Nothing more.

You can even find this value specified in your ADC's manual (it may even vary), it might be something like "0dBFS reference level = +18dBu" or "Cliping at +18dBu" or "Maximum Input 4dBu" ... whatnot, given in dBV,dBv,dBwhateverconfuses the customer.

So in what way knowing and taking advantage of that fact will improve your sound or not will only lead to a nasty flamewar, and since I lost my flamesuit, I won't comment on it.

But what I know and can say for certain is that this has NOTHING to do with how REAPER should handle or handles its DIGITAL signal. So do what Schwa said and if you feel like it is better for your ADC/DAC turn down the volume but in REAPER those rules of hardware don't apply! It's all prefect clean math in digital, no unwanted distortion, transmittion losses, and all that hardware flaws.

Okay I'm out, before it get's nasty ...
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Old 07-29-2008, 05:19 PM   #8
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Once your signal becomes electricity again, after the DAC, any compensation for operating formats should be kept analog in my opinion.

For instance, my DAC outputs +4dbu (pro line level)

My headphone amp inputs are -10dbV (consumer line level)

Connecting them right up, well firstly it sounds wrong/bad, there is no headroom, and the signal audibly distorts.

I own a simple little box, which will turn my +4dbu into -10dbV, i stick this in between and everything sounds just fine.

Now i could just turn the master fader down in the software.. this will probably help lower the distortion, but i end up with an inaccurate lower resolution signal this way.

Hope i didn't miss the point here!
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Old 07-30-2008, 02:07 AM   #9
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Thanks to you all!

Maybe I just lower the master fader (does it affect all 16 analog outputs?), therefore I'll have to record at much higher levels than I would like to. It's not a PROBLEM, it would be just convenient and extremely time saving not to adjust every single parameter. I just want to insert a dynamic plugin, adjust the sound, and that's it. I don't want (in certain "limiter" applications) to open a gain plugin before and after to compensate...
Why for god's sake would the Sony console offer such a feature???

[Why can't I "quote"?]
Art Evans, you're wrong when you say: "It sounds like your converter is poorly designed or defective if that's your experience."
EVERY 24bit converter will work best if you record at max -12db, drums peaking at maybe -6db. Just try it out! It's fantastic. And that's how it's SUPPOSED to be - that's the problem. Stupid enough, some guy invented the dbFS scale - probably because it was technically easier. His idea behind it probably was: in the analog world, within good gear, we have at least 18db HEADROOM until it clips audibly. But -18 would still equal 0. If everyone keeps on recording at the levels he is used to from analog, everything is mega safe.
Unfortunately, the meters were not labelled "0" at the -18dbFS point (except Ensoniq Paris, I don't know - it's said to have "lots of headroom" and sounding fantastic. This could be an explanation)...

Last edited by beingmf; 07-30-2008 at 02:09 AM. Reason: Sony!
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Old 07-30-2008, 03:50 AM   #10
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Default food for thoughts

There's a looong but interesting discussion that's related to this subject in the PSW forum:

http://recforums.prosoundweb.com/index.php/t/15038/0/

Cheers!
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Old 07-30-2008, 02:48 PM   #11
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Ciao Franz,
thanks a lot, that thread covers more or less what I was referring to. Great read (so far, I'm still on page 4 of 14 ).

Here's a very good quote of what I mean:
"Extreme Mixing
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Music sounds better when recorded at lower levels. I think that's mostly a function of the A/D converters and the preamps and compressors that we use on the front end. They are meant to work best around 0 VU. To get the level to digital zero, you may have to push your trusty neve up to +25 or 30; something it was never intended to do.

From a practical point of view, when mixing in the computer, it really helps to have your faders in a place where you can easily make sublte moves. Having the fader around unity is the best place. Try making a .5 db move there, then try the same .5 db move with the fader at -30.

My advice is to read the thread with Paul Frindle, and the others for a full explaination of what's going on behind the curtain. Digi also has a white paper on the DUC that is helpful. Do what you think is best, but don't fall into the "use all the bits for best resolution" trap. I think it's the same style of logic that President Bush is using for his Iraq strategy!

By that way, what ever happened to Nika? Did he go back to school or take a job somewhere? He was such a bright guy! It was always a great learning experience to have him jump into these discussions.

Steve"

Last edited by beingmf; 07-30-2008 at 02:53 PM. Reason: found something
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Old 07-30-2008, 10:58 PM   #12
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Default JUSTIN!



No, really: what do you think? Wouldn't it be absolutely super "pro" to have a "global gain setting" feature - just to be compatible with the analog world.
That would mean:
- plugin volume compensation (pre+/post-)
- minus 18 (or 20, or X, depending on the system you're working on) is the new zero. And there's "reds" in the meters that indicate your headroom...
- waveform graphics!!! Isn't it horrible to edit when you just can't see hardly anything? -> vertical zoom "+18".

Please think about it!

Last edited by beingmf; 07-30-2008 at 11:18 PM. Reason: waveforms!
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Old 07-31-2008, 12:33 AM   #13
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Quote:
Originally Posted by beingmf
- plugin volume compensation (pre+/post-)
use the volume (pre-fx) envelope unless we get a trim/gain knowb in the mixer


Quote:
Originally Posted by beingmf
- minus 18 (or 20, or X, depending on the system you're working on) is the new zero. And there's "reds" in the meters that indicate your headroom...
rightclick the master meter and set it up to your taste.


Quote:
Originally Posted by beingmf
- waveform graphics!!! Isn't it horrible to edit when you just can't see hardly anything? -> vertical zoom "+18".
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Old 07-31-2008, 01:05 AM   #14
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Quote:
Originally Posted by Dstruct View Post
use the volume (pre-fx) envelope unless we get a trim/gain knowb in the mixer
sag mal auf deutsch - den ersten teil versteh ich nicht. Stell dir vor, der Track ist auf -20 AUFGENOMMEN, ich zieh nicht den Fader runter, das bringt ja nix...

Setting up the master meter is clear, yes, but I meant to implement it automatically in the "Global Gain Setting feature".
Same applies to "waveform graphics". I know that I can zoom the waveform MANUALLY but FOR CONVENIENCE and SPEED I want my DAW to do this from the start. Am I so unclear???
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Old 08-04-2008, 09:54 AM   #15
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"bump" hasn't at least 10 characters

Last edited by beingmf; 08-04-2008 at 09:54 AM. Reason: misspelled :)
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Old 08-07-2008, 03:02 PM   #16
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b u m p de bump
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Old 03-06-2009, 08:36 AM   #17
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as supporting this enthusiastically. Others may not see it as important, but I reckon it's revolutionary.
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Old 03-06-2009, 08:43 AM   #18
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@kneel The headphone amp inputs and DAC outputs differ in more than level - they have different *impedance* - a transformer is necessary to connect them properly.
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Old 03-07-2009, 03:22 AM   #19
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it's nice to have such an old thread reanimated at least ONE guy that really digs it LOL thanks kerryg!
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Old 11-07-2010, 03:28 PM   #20
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There are multiple situations where gain-staging matters in digital, and they are a lot more common than most user manuals would have you believe. I suspect that ignorance of them is the cause of a great deal of shopping for ribbon mics, tube preamps, outboard processors, tape emulators, analog summing boxes, and other such stuff. If I were of a more paranoid mindset I might even suspect that it's deliberate...

First, where gain-staging DOESN'T matter is within a high-resolution floating-point audio engine like REAPER's. so you can turn your tracks way, way up within REAPER, and way, way down again, and the output will be exactly the same. This leads to the common but erroneous belief that all you need to worry about with digital is the clip LED.

There are several ways in which you can get overload problems without triggering the clip LED.

The first and most commonly-overlooked one is on the analog front-end of your AD converters. Before that signal is actually converted into ones and zeros, it has to go through one or more analog stages, including the (typically very steep) anti-aliasing filter. As the OP said, analog consoles and line levels are typically designed to see something like 1.2 volts steady-state, which is around -18dB on digital peak meter. Chances are very good that your AD converter/audio interface is designed that way as well-- all else being equal, more powerful circuitry is noisier, not to mention more expensive, and requiring bigger power supplies (IOW a $500 USB-powered audio interface probably does not have massively high-headroom overbuilt analog front-end).

So if you're following the instructions of most audio interfaces and conventional digital wisdom, and pushing your record levels as close to zero as possible, you might be pushing 8 times as much current through those circuits as they are really designed to handle. 0dBFS is like pinning the VU meter on a typical analog console, and doing that for more than a millisecond or two at a time is apt to sound pretty distorted. Moreover, if the AD or analog front-end has any filters (and we know it has at least one), and/or any kind of overload protection or pre-limiting (and the manual won't always tell you if it does), then you've got additional opportunities for analog or inter-sample clipping to occur EVEN IF THE CLIP LED NEVER LIGHTS UP.

Moreover, clip-detection is not precisely an exact science with audio interfaces, even expensive ones. There is no government agency testing audio interface meters for accuracy, and standards and implementation are pretty loosey-goosey. some of them are downright sloppy-- imprecise voltage-detection circuits feeding an LED, or whatever. So think of the "clip" LED as a courtesy provided by the manufacturer, not as a clinical test instrument. A lot of brittle, jaggy, "digital" ugliness can sneak in without tripping the LED.

The second place where gain-staging becomes an issue is within the digital processors used. "But wait", you're saying, "didn't we just establish that REAPER has basically infinite headroom thanks to high-resolution floating something or other???"

Yes, we did, and you can turn tracks up and down within reaper all day long to your heart's content without clipping. However, while you're turning things up and down, you might also decide to, say, add some effects. Now, maybe all your EQs and reverbs and technical compressors and noise gates and so on are ALSO high-res floating-point, or maybe they aren't. Do you know which ones are? I don't. And if they are NOT floating-point(including all ProTools TDM plugins), then they are CLIPPING if the signal goes above 0dBFS.

Moreover, what about stuff like, say, guitar effects, or "character" compressors, or tube emulators, or harmonic analog-ifiers of every sort that are so popular these days? Even supposing that all of those are floating-point, how do they know when to start "analog-ifiying"? If something is supposed to emulate the response of, say, a Neve preamp, or a Marshall amplifier, how does it know when to start, when the digital system is feeding it a 144dB+ dynamic range with no established zero reference?

Maybe the processor has some kind of internal smarts to figure this out, or maybe it relies on you to set the input trip just so, or maybe the designer never even thought about that, they just emulated the effect... hmm... that would mean that digital signal at, say, -6dBFS, is like an analog signal hitting the emulator at, um, PLUS 12 OR 14dBu! If we were doing that to a hardware processor through an analog console, we would be aware of POUNDING the poor device, because our VU meter would be pinned at the redline. And maybe that's what we want, or maybe it isn't, but an authentic emulator would just be doing its job by returning a very overloaded, distorted sound from this kind of treatment.

Once again, do you know how all your processors and plugins handle these kinds of things? I sure don't.

Last but not least, the output stage has the exact same problems as the input stage: a converter, possibility of inter-sample clipping, a steep filter, and an analog "line-level" stage on the back end that is just as susceptible to distortion as any other, along with a dubious "clip" LED and all the shortcomings of digital metering. Even if yours is perfect, do you expect every CD player to be? And how about the digital broadcast processors used by radio stations and nightclub PAs? Even car stereos these days come with digital processing-- are you counting on that to be 64-bit floating-point?
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Old 11-07-2010, 04:03 PM   #21
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So what does all this mean for the digital recordist?

It means keep your levels low. Target for -20dB average input and output level when tracking, mixing, processing, everything. If it's easier to think in peak terms, look for the peaks to bounce around -10 to -6 on your digital meters.

When you shove everything right up near the top of full-scale, you open up a whole bunch of opportunities for exactly the kinds of brief, subtle clipping that is hard to detect, even for experienced engineers, and that never triggers the "clip" LED. The "clip" LED will tell you when you've gone into nasty digital distortion, but you'll usually hear that anyway, if you're paying attention.

The "hidden" clipping is more likely to sound like brittle, cheap, "digital" sound, especially with a bunch of tracks all combined that each have slightly clipped waveforms. This is not always obvious or easy to hear when you're in the thick of recording and mixing. If the track immediately went into complete digital overload, and sounded like that kind of white-noise modem sound of deep-red digital clipping, you'd probably notice it (moreover, the "clip" LED would light up). Instead, it sounds like a subtly unpleasant cheapening and harshness (especially at the extreme lows and highs, where cutoff filters tend to live).

A lot of "analog magic" (possibly most of it, maybe even all of it) is that analog has good metering, digital has bad metering.

Analog "clipping" is a gradual, sometimes even pleasant thing. Analog meters are calibrated somewhat subjectively, to set the "zero" point as a target level for clean sound. That might be anywhere from 14 to 24 decibels below the point where the designer hears or measures "clipping", and how the designer handles the tradeoffs between headroom, noise, and pleasantness of intermediate saturation is a often huge part of what gives different analog components their "sound".

Take the exact same analog console or preamp, and calibrate the meters so that 0dBu on the meter is, say, 6dB closer to saturation point, and anyone reviewing it would call it a more "aggressive", "colored", maybe "fatter" device than the exact same thing, but with the meters calibrated differently. Do the same with a digital device (set the clip indicator 6dB below "actual clipping), and the reviewer will find it to sound "harsh", "cheap", "brittle", and "digital"-sounding compared to the original.

The grand irony would come about when an excruciatingly technical reviewer opened up the boxes and announced that they were identical, and that the supposed differences were all just placebo effect or marketing woowoo. then a years-long flame war would erupt over brands of capacitors and ICs and testing methodology and accusations of bad hearing and so on. (sound familiar?)
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Old 11-07-2010, 04:41 PM   #22
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Bringing it back to the OP...

I would personally love to see VU meters on digital equipment. And I cannot think of any good reason why it wouldn't be pretty easy to do, although it would necessarily require some kind of agreement on a "standard" (I'd probably vote for something like "zero" at -22dB full-scale, with a "clip" indicator at -2dB).

The hard part, as far as I understand, is that DAW has to rely on the interface or soundcard to report the "zero point". IOW, REAPER does not and cannot know what the signal voltage was prior to conversion to digital. It only knows what the converter reports as "full scale" (0dB on the digital meter).

This might seem like a non-issue if you do everything "in the box" (i.e. entirely within REAPER), but it potentially becomes a problem when, for example, sending digital audio out to a legacy 14-bit fixed-point reverb box (and there are some very good-sounding ones out there, believe it or not).

Since digital signal is necessarily referenced to full-scale (regardless of how the metering works), the digital reverb box is receiving a -24dB average signal. At 14bit processing depth, it really only has something like 84dB dynamic range, and in practice more like 78dB. Subtract 24dB from that, and you have ~60dB from average level to silence. And especially with something like reverb, that might very well be more like a 30dB "range" where you can actually hear it "working". That's a measly FIVE BITS of real resolution for processing reverb tails, which can sound pretty ugly.

In a case like the above, the reverb box would almost certainly perform a lot better if the digitial output were normalled to something like, say, -0.05dBFS before export, especially with "flat" or "smooth" content such as strings, vocals, and electric guitar.

For the record, that's why digital metering started out as referenced to peak, instead of referenced to average (as analog does). Well, that plus the fact that there is no meaningful "average" target for digital-- in a technical sense, digital is either clipped or it's not, whereas analog is constantly threading the needle between noise and distortion.

Perhaps more to the point, we would run into MASSIVE problems trying to compare record levels with modern CDs, MP3s, and the rest of it. Forget about what SHOULD BE for the moment, and consider what IS: modern commercial releases would show up with an average signal level of something like +14dB on my above -22 scale: the meter would pinned constantly. That's a serious problem for widespread acceptance and usability.

Having said all that, REAPER might be just the place to start with VU-style metering as an option. (I would personally love it if it were mandatory). Somebody's got to be the first.
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Old 11-07-2010, 04:58 PM   #23
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I want this option(!) as well. Where do i vote?
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Old 11-08-2010, 03:20 AM   #24
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No FR to vote for yet, cause it seemed, that there's not enough people interested in. It may change due to some threads regarding digital vs. analog levels, and IMO this is where at at least a visual reference point comes into play. I'm still struggling between the original question of my thread, and a simple "0db=-18db" (or -20 or -22) conversion in the meters only...

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Old 11-08-2010, 09:40 AM   #25
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but I like digital clipping... just can't afford the $ for new tweeters...
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