Old 11-06-2010, 08:16 AM   #1
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Default "better mixes with track levels @ -20dBfs"

I felt like I was hijacking the McGurk Effect thread (http://forum.cockos.com/showthread.php?t=68122), so here's a new one.

In there, yep claimed,

"Doing basically the same mix twice, except doing one where I keep the average per-track signal level at -20dBFS or so, I'm finding way better results with the "quiet" mix, and ugly, "digital" brittleness when I disregard gain-staging even when it's supposedly irrelevant."

So, let's say you've got source tracks of varying peak levels. You then either use a volume plugin first or adjust the pre-fx gain for every track to place it at about -20dBfs? I assume it would be best if the signals were recorded at that level rather than adjusted to that level?

I'm somewhat understanding the analog/digital/electrical background to this concept but am having trouble understanding the method, techniques or workflow to best take advantage of it.

BTW, I'm continually amazed at the levels of knowledge, community and support in these forums. Infinite thanks...
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Old 11-06-2010, 10:24 AM   #2
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So, let's say you've got source tracks of varying peak levels. You then either use a volume plugin first or adjust the pre-fx gain for every track to place it at about -20dBfs? I assume it would be best if the signals were recorded at that level rather than adjusted to that level?
You're right, best is to record them that way, but in order to prevent further damage (plugins/summing) I'd trim them down to average -18db / peaks -6 approx.
No need for a trim plugin btw, just change the item's level itself in the properties (next to "Normalize").

EDIT:

btw. what helped me enormously is to set my meters to a bigger range! So even lower levels will give you some kind of "satisfying visual feedback". One thing you couldn't set up in Logic and one of the reasons I had to quit using it. You see 1 (one!) segment of the meter flickering when you touch -20db – WTF? And I always wondered why my console's channel meters were notoriously in the red!
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Old 11-07-2010, 02:23 AM   #3
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This technique (thanks Yep and Paul Frindle) has made a very noticeable difference in my mixes...

I wish I had stopped drinking the "it doesn't matter as long as the master isn't clipping" kool-aid a lot sooner. Sure that may be the case with a simple null test using ReaPlugs only, but in practice headroom is important, digital or not.

There's a reason mixers have trim, and I really wish Reaper's did too.
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Old 11-07-2010, 05:01 AM   #4
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This technique (thanks Yep and Paul Frindle) has made a very noticeable difference in my mixes...
Absolutely! Though your statement is something to think about: in the analog world, would you have called gain staging a "technique"? I don't think so – I'd call it a "physical rule". Nobody in the 60's to 80's would have asked why a recording sounds bad, while the levels were obviously slapped to overdrive nirvana.

True that in the 16bit era your converters had been most happy when fed close to the 0dbFS point, but 24 bit is a completely different thing! Coming from that era, I had to learn and accept it, too, but the question is why DAW developers don't include some easy accessible workaround for this simple fact??

Like I pointed out in a FR regarding this: the Sony Oxford console had a "virtual zero" or "analog calibration", so meters on analog outboard would visualize the same values as the digital I/O's. There's no way, in no DAW, to decide "my headroom shall be 18db(FS), so my 0db(VU) is shown at -18dbFS, everything above will be visualized as 'red'".
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Old 11-07-2010, 06:34 AM   #5
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Originally Posted by beingmf View Post
There's no way, in no DAW, to decide "my headroom shall be 18db(FS), so my 0db(VU) is shown at -18dbFS, everything above will be visualized as 'red'".
Let Reaper be the first one with this, then...
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Old 11-07-2010, 09:13 AM   #6
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Let Reaper be the first one with this, then...
That's what I thought, too, when I wrote the FR, but nobody seriously jumped on the wagon. And that was 1 1/2 years ago ):

So let's carry on with this thread until we've convinced enough people that metering choices are absolutely crucial. Then let's start a new FR, and collect 1000 votes before 4 is released (: It'll be called "The groundbreaking Reaper metering FR" - LOL
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Old 11-07-2010, 08:15 AM   #7
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OK, I wish to mix my new smash hit. And I wish it to have the highest possible "warm" "analog" sound quality. It consists of multiple tracks of a -20dBFS RMS pink noise WAV file, uncorrelated no less, at 44.1kHz. (thanks, Bob Katz) Trust me. This song is going to be awesome.

Anyway, I add the first track, with the fader at 0, and say "Aha!" "This track is peaking at around -11.5dBFS. It's too hot!" So, I access the media item properties and bring the gain down -8.22dB. There. That looks better.

Now, on the master track I've inserted "JS: Meters/vumeter" "JS: Meters/dynamics_meter" and "JS: Liteon/vumetergfx" Pretty! Everything looking good!

But I have unlimited tracks! Let's use them! Now I duplicate the track seven times, for a total of eight tracks. The meters are showing about -13 RMS and about -1.5 peak. Hey, I even left a little headroom for the mastering engineering! This is guaranteed to sound fantastic and super-analog!

OK. Thanks for indulging me. That was pretty ridiculous. But is this the technique (oops! I mean "rule") we're talking about?

Project file attached...
Attached Files
File Type: rpp 20dbfs_test.RPP (12.4 KB, 231 views)
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Old 11-07-2010, 10:01 AM   #8
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OK, I wish to mix my new smash hit. And I wish it to have the highest possible "warm" "analog" sound quality. It consists of multiple tracks of a -20dBFS RMS pink noise WAV file, uncorrelated no less, at 44.1kHz. (thanks, Bob Katz) Trust me. This song is going to be awesome.
Release date? I'll buy it!*

Quote:
Anyway, I add the first track, with the fader at 0, and say "Aha!" "This track is peaking at around -11.5dBFS. It's too hot!" So, I access the media item properties and bring the gain down -8.22dB. There. That looks better.

Now, on the master track I've inserted "JS: Meters/vumeter" "JS: Meters/dynamics_meter" and "JS: Liteon/vumetergfx" Pretty! Everything looking good!

But I have unlimited tracks! Let's use them! Now I duplicate the track seven times, for a total of eight tracks. The meters are showing about -13 RMS and about -1.5 peak. Hey, I even left a little headroom for the mastering engineering! This is guaranteed to sound fantastic and super-analog!
And that's not even a musical (aka "dynamic") signal! In a real project you have tracks with different peaks and dynamics, different spectral content, so you should read less than 1,5 db then! There should be some headroom left, even with 30 tracks of recorded music!

Quote:
OK. Thanks for indulging me. That was pretty ridiculous. But is this the technique (oops! I mean "rule") we're talking about?

Project file attached...
Yes it is (:



* Don't laugh – when I was 15 and had my first sampler (Yamaha FZ-10, I wish I still had it), I actually made a track only from different noise samples layered, hehehe...

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Old 11-07-2010, 01:54 PM   #9
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OK, and I *think* this is the point of beingmf's FR...

I'm using "JS: Liteon/vumetergfxsum" to check a kick drum level. It's peaking around -20 on there. But that's -20dBFS, NOT dBvu, which would show about 0. So, the FR is just change the numbers so the levels look as if you were working on an analog desk. Yeah?

The current plugin is really monitoring the peak or RMS levels as measured in dBFS.
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Old 11-07-2010, 04:36 PM   #10
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OK, and I *think* this is the point of beingmf's FR...
So, the FR is just change the numbers so the levels look as if you were working on an analog desk. Yeah?
The current plugin is really monitoring the peak or RMS levels as measured in dBFS.
When I think about it: this is the proper description, since a real dbFS > dbVU conversion doesn't seem to be correct. Don't ask me to explain this, all I understood myself was that as an example different converters are calibrated to a different value (somewhere between -22 and -16dbFS). But there's more to it AFAIK...

All I want is a different visualization – if I know my converter's* specs, I just want to tell Reaper "0=-18" –> all meters switch to the new values, like in an analog console, yes. My main concern is of course the MCP meters, with a dbVUish zero point, and the headroom indicated green < 0dbVU/-18dbFS > yellow < -6dbFS > red.

The downside of this is that 3rd party plugins' meters would always show a higher level, but I think one could get used to it pretty fast.


* or converters' (pl.)... Even better would be if I'd be able to assign different values to different track objects! Why?
I have 2 different converters (one 8ch. interface and one ADAT connected 8ch. ADDA), which are routed to my console. I have (as default) set up folder tracks with their outputs assigned to the respective mixer channels. So all I need to do is to drag a track into folder "3-4", and it will play on console channels 3-4. But: the DA levels of the converters are slightly different! So wouldn't it be handy if folders 1 to 8 would convert the levels to a different value than folders 9-16?? Not that I can't live without it, but it would be one step towards highly professional customer needs...

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Old 11-07-2010, 04:42 PM   #11
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Just record at or near analog 0 - whatever that is on your digital meter, depending on converter calibration - and it takes care of itself. Many prosumer converters are probably -18 or even -15. Pass a tone though one to find out.

That was a long thread and Frindle was 100% correct, but the thing many people missed is they were talking about PTHD so - some - of that stuff doesn't apply to Reaper or any other native daw.

Record in the normal analog ranges and the other stuff takes care of itself. If your converters are calibrated to -20 you shouldn't be consistently over that - RMS wise (peak wise, sure you'll peak over that) - anyway. Turning your tracks down to - peak - at -20 isn't quite the same thing.

The problem is - imo - that very few prosumer audio devices even have VU meters anymore. Nobody would record +18 analog to tape but people do it in digital, go figure.
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Old 11-07-2010, 04:50 PM   #12
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Just record at or near analog 0 - whatever that is on your digital meter, depending on converter calibration - and it takes care of itself. Many prosumer converters are probably -18 or even -15. Pass a tone though one to find out.

That was a long thread and Frindle was 100% correct, but the thing many people missed is they were talking about PTHD so - some - of that stuff doesn't apply to Reaper or any other native daw.

Record in the normal analog ranges and the other stuff takes care of itself. If your converters are calibrated to -20 you shouldn't be consistently over that - RMS wise - anyway. Turning your tracks down to - peak - at -20 isn't quite the same thing.
Yes, of course (: You know that, and I know that, and kelp knows it, too. Now!
How many long-time engineers do you know, Lawrence, that still record far too hot? I know many many of them.
The easiest IMO would be to give a simple visual reference, that will tell them "Hey, could it be you're driving the converters a little too hot? My Reaper meters are all in the red!"
Wasn't that the secret behind the "magic" sound of the Ensoniq Paris system btw.? To meter à l'analogue? So nobody was hitting them olde converters beyond their sweet spot?
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Old 11-07-2010, 10:59 PM   #13
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[QUOTE=Lawrence;613526]Just record at or near analog 0 - whatever that is on your digital meter, depending on converter calibration - and it takes care of itself. Many prosumer converters are probably -18 or even -15. Pass a tone though one to find out.
QUOTE]

I don't mean to hijack this thread, perhaps another needs to be started? Lawrence, would you mind explaining how to "pass a tone" through converters to find out what they're calibrated at.
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Old 11-07-2010, 06:00 PM   #14
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OK, I wish to mix my new smash hit. And I wish it to have the highest possible "warm" "analog" sound quality. It consists of multiple tracks of a -20dBFS RMS pink noise WAV file, uncorrelated no less, at 44.1kHz. (thanks, Bob Katz) Trust me. This song is going to be awesome.

Anyway, I add the first track, with the fader at 0, and say "Aha!" "This track is peaking at around -11.5dBFS. It's too hot!" So, I access the media item properties and bring the gain down -8.22dB. There. That looks better.

Now, on the master track I've inserted "JS: Meters/vumeter" "JS: Meters/dynamics_meter" and "JS: Liteon/vumetergfx" Pretty! Everything looking good!

But I have unlimited tracks! Let's use them! Now I duplicate the track seven times, for a total of eight tracks. The meters are showing about -13 RMS and about -1.5 peak. Hey, I even left a little headroom for the mastering engineering! This is guaranteed to sound fantastic and super-analog!

OK. Thanks for indulging me. That was pretty ridiculous. But is this the technique (oops! I mean "rule") we're talking about?

Project file attached...
I don't know whether this is a problem on my end or yours, but I don't seem to have any audio with your test project.

That said, in a conceptual sense, it sounds like the makings of a smash hit. I would suggesting recording 120bpm quarter-note hits of an 808 kick sample, then triggering hard compression on the white noise instrumental track with the kick keyed to the side-chain input to get that big pumping club effect. Then find a 16-year-old girl to sing some foul-mouthed "get back" lyrics about how hot she is, apply autotune liberally and you should be all set (after multiband look-ahead limiting to within -3dBFS, of course). The white-noise genre is huge right now.

Having said all that, I think you might be barking up the wrong tree with your test example. You're not going to get any difference in sound quality by gaining up or down within REAPER, or any other modern floating-point audio engine.

Allow me to try and re-state what I think you're referring to...

Going back to analog (forget about digital for the moment), we have a needle to thread when it comes to electrons moving across copper wire and through vacuums and around iron transformer cores and the like. The more copper and iron and power and so on, the more noise (in the form of random movement of surrounding electrons) we get. So there is always a sliding-scale tradeoff between noise and headroom: the more power-handling (i.e., headroom) we add, the more noise we introduce. And expanding that usable "window" between noise and distortion increases cost exponentially.

So what analog manufacturers did, over the course of many decades and largely driven by the needs of the telephone company to deliver clean audio over thousands of miles of thin copper cable, was to settle on a certain "average" signal level that devices were supposed to be built "around", so to speak. The precise spec depends a little bit on what country you're in and where your gear was made and so on, but the idea was always to design and manufacture gear that was intended to sound best within a certain input range, and to deliver a similar output range (don't quote me, but I *think* the US standard was 1 volt=0dBu ((or dBv, as it is sometimes called)), whereas the UK was keyed to current instead of volts, and therefore something like 1.23V at 0dBu). Anyway, the idea was that every device should basically input and output the same electrical signal at it's "ideal" operating range.

Now, analog manufacturers had and still have very wide latitude to decide what "ideal operating range" means. Some, such as telephones, have very low headroom and narrow bandwidth, and are designed to crank out the maximum signal in the upper-midrange, for maximum speech intelligibility over the noise floor of 3,000 miles of copper cable. Others, such as the unbalanced passive EQs favored by some mastering engineers, are designed to have the simplest, cleanest, and most linear possible circuit for use in short-run, noise-shielded environments. In both cases, the ideal "operating range" is determined by the expected input and output signal strength, measured in terms of average voltage or current.

Now, the nature of analog is that there are no "hard" cutoffs. Noise dissipates gradually, but never completely. Saturation similarly occurs gradually. There is probably no better example of this than when a music-shop owner in postwar London named Jim Marshall decided to make knockoffs of American guitar amplifiers, which were expensive due to exchange rates between then-depressed England and the booming American post-war economy. He copied the circuit from a Fender Bassman, using cheaper locally-available tubes, and sold them in his shop under his own name. A guy named Jimi Hendrix (American, as it happened) came in and discovered that by cranking the volume to overload these cheaper European tubes, he could get a sound he enjoyed even better the "cleaner" more "hifi" sound of the Fender originals.

Mr. Hendrix went on to have some success as a popular entertainer, and Jim Marshall's knockoffs have even attracted some customers who could afford a lower-distortion Fender Bassman.

The point is that analog deals in ranges, not fixed thresholds where pristine sound crosses the line into ugly clipping. And analog manufacturers design equipment to function around these "ranges", and have broad latitude to do so. Some analog devices, such as Neve or Trident preamps, are regarded as performing very favorably when "pushed hard", much like Jim Marshall's amplifiers. Others offer lots of clean headroom for transients but start to sound strangled, fizzy, or bad when average signal levels get much above zero on the meter. Which is another thing about analog: there is a lot of art, not just science, to designing analog audio circuits. Do you favor clean, accurate headroom for transients or a more "punchy" or "firey" saturation? This all comes into play not just with how you design the circuit, but HOW YOU SET THE METER IN RELATION TO THE CIRCUIT. The designer is basically building a circuit to sound good around zero dB on a fixed meter...

Digital, on the other hand, is pretty all-or-nothing. Modern high-resolution digital is basically "perfect" within it's operating range, and then craps out completely past it. This is why digital meters are keyed to "peak" level, or the maximum level that the digital system can handle. There's nothing wrong with that, and it is absolutely the correct way to meter digital audio. EXCEPT...

Before it gets to be digital, the audio starts out as analog. Which means, if you are solely using digital meters, there is a lot of potential for the analog front-end to crap out before the digital system ever gets a chance to meter it. There is also the problem of internal digital processors-- are you certain that every plugin you're using has good floating-point internal calculations? And if so, what about your saturation/compression/analog-ifier/guitar effects? How do they know when to start "saturating"? What about the cutoff filters on your AD and DA converters? What about inter-sample clipping that the digital system cannot detect?

this stuff is not always obvious to hear. Your clip LED will light up when you get that screechy white-noise "deep" digital clipping, but you'd probably notice that anyway. This kind of "hidden" clipping tends to have a more subtle effect, just flat-topping the waveform peaks and creating a harsh, brittle, "digital" sound. It's not always easy to hear in the thick of a recording session, but a little on the bass, a little on the high-hats, a little on the snare, a little on the kick, and next thing you know, you've got a "flat", "cheap", "digital"-sounding mix and you're out shopping for ribbon mics and tube preamps.
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Old 11-07-2010, 06:33 PM   #15
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OK, I'm losing my grasp here (again)...

Yep just stated that gaining up/down within REAPER isn't going to affect sound quality. But haven't points been previously made about "plug in headroom?"

I'd like to be sure my OP was understood. There's a subtext there that no matter what the input, having lower track levels will make the final mix "better." This is consistent with yep's claim that better results have been seen when doing a "-20dBFS" mix compared to the same (implied) "0dBFS" mix.

Here's what I'm trying to remove from my brain -- that there's a simple technique or rule that will magically make all my mixes sound better.

I see three general headroom areas: a) the input, which seems to be the simplest to understand (don't push the A/Ds); b) plug ins, which seem to be a huge variable; and c) the final master or D/A, which still kind of confuses me because is all lost when the final mix succumbs to the loss of dynamics and maximized level in the holy loudness war?

Beingmf and PitchSlap seem to be saying, yes, this simple rule makes a difference. Lawrence is saying get it right going in. And Yep may or may not be saying watch the front end and the back end and use your ears in the middle.

I'm going to be a broken record here, but holy crap you people are awesome. I feel like I'm quivering on the edge of some new understanding. Some "new" understanding that's existed for decades...

THANK YOU!

P.S. -- Yep, I didn't include the audio file but gave a link at the beginning of the post to Katz's site, my source. Sorry, I should have highlighted that.
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Old 11-07-2010, 06:39 PM   #16
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OK, I'm losing my grasp here (again)...

Yep just stated that gaining up/down within REAPER isn't going to affect sound quality. But haven't points been previously made about "plug in headroom?"
There are a few plugs that still operate internally at 24-bits... Waves Linear being the most "notorious" (well known). If you encounter one of those by chance and clip it, you'll probably hear it right away because you'll be practicing critical listening methods during the mixing process.

Unless maybe the track is a distorted guitar that was recorded severely clipped already from the guitar amp... you may not hear it. Either way, if you follow analog practices during tracking (and trim the tracks from others who did not) it won't matter either way because you'll have plenty of headroom.

The vast majority of plugs in 32-bit float won't clip no matter what you do to them, for the "science experiment" part of it. But generally speaking you shouldn't be feeding them levels that high anyway. If the mix isn't "loud" enough... that's not the solution.

The common train of thought for tracking with gear that doesn't meter properly is this. If you don't know your converter calibration and you don't have VU meters on your outboard gear and you have no idea what level to shoot for in the daw... use the -10 peak level in the digital meter as a maximum target (for random transient signals) and you'll be good... if everything sounds ok.
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Old 11-07-2010, 07:50 PM   #17
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OK, I'm losing my grasp here (again)...There's a subtext there that no matter what the input, having lower track levels will make the final mix "better."...
No, sorry for the confusion.

The subtext is that keeping lower levels throughout MIGHT make the final mix better, IF one or more things in your signal chain are either not being metered correctly by the digital meters, OR if they are not really capable of being accurately-metered by digital peak meters.

Just to clarify, there is nothing wrong with REAPER's metering (or if there is something wrong with it, it's not what we're talking about here). And REAPER has, for all sane purposes, infinite clean gain, up and down. Same applies to all major DAWs in current use, slightly excepting ProTools HD.

So we're not talking about some kind of across-the-board improvement or degradation that happens as a result of some magic digital gain level.

What we are talking about is a host of potential opportunities that occur for distortion/clipping that never shows up on digital meters, due to the nature of digital peak meters and the sorts of things that they can and cannot detect/measure.

The most obvious example (what Lawrence is talking about) comes from the fact that there is no such thing as a true "all digital" signal path. Even if your record consists entirely of digital synthesizers, something has to happen before you can hear it through the speakers or headphones. There is a stage between the computer and your speakers where that digital is converted to analog. If you are recording some "real" instruments or voices as well, then there is a corresponding stage on the input between your mic/preamp/whatever, where the still-analog signal is "prepped" for conversion to digital. This stage is NOT just a straight wire, and most people have no metering at all to tell them what is happening.

REAPER will accurately tell you what is happening AFTER the signal has been converted to numbers, but it cannot tell what happens on the way in, or the way out. If your input or output converters are overloaded, distorting, triggering overload protection, experiencing inter-sample clipping on conversion, or distorting the cutoff filters, you have no way of knowing it (other than by careful listening comparisons at various gain levels, not something most people do when they are trying to record a second rhythm guitar track with drums, bass, other guitars, and synthesizers all playing back).

The second type of ugliness, the type than can occur INSIDE the digital system, happens once you've realized that, man, this guitar track (or whatever) sounds brittle and ugly and "digital". So you start running it through all your tube-ifiers and tape-ifiers and magical analog emulation plugins that are supposed to deliver smooth, "saturated" harmonic distortion, just like a real magic box. Problem is, they all sound like cheap digital distortion, and you're sitting there thinking you need new mics or preamps or better magic plugins...

Well hold up there, cowboy. What average signal level is the original "magic box" supposed to run at? 1 volt? Now, what is the equivalent input level of your guitar track (or whatever) that you're trying to "smooth out" by running through it? 9.5 volts? So the processor is doing exactly what it's supposed to do: it's distorting. And you're now trying to fix distortion with more distortion.

At this point, somebody's probably raising their hand and saying, "but I use a POD set to British Funk, which is the same preset Chris Lord-Alge uses, and it's all run through hand-made, solid-gold converters made by a guy so esoteric that NASA has deleted his name from Google. Also I have a book that says you should record to digital as high as possible before clipping. Surely my recordings can't sound bad...?"

Well, yeah. Whatever. good luck with that. This advice is for people who are NOT already happy with the quality of their digital sound, or who think there might be room for it to sound better. If you have the ability, time, and sufficient documentation to read through and suss out the analog and digital topology of your entire signal path, then you probably should. That's how the old analog guys did it. Maybe everything is perfect and recording at lower levels is pointless.

But if you'd prefer to hedge your bets, my advice is to track at lower levels (say, maybe -20 average, -10 peak, or thereabouts), and then KEEP your levels there throughout the digital and analog signal chain. And based solely on the complexity of most multitrack DAW projects, I'll bet that you start getting better overall mixes, since it's probable that at least something, somewhere in your signal path doesn't like being pushed to max at all times.

If nothing else, it at least makes it easier to mix, since you can just throw up all the faders to zero and still have headroom to work with, as opposed to constantly having to crank down the master fader and/or re-mix every time you add new processing.

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Old 11-07-2010, 08:58 PM   #18
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But if you'd prefer to hedge your bets, my advice is to track at lower levels (say, maybe -20 average, -10 peak, or thereabouts), and then KEEP your levels there throughout the digital and analog signal chain. And based solely on the complexity of most multitrack DAW projects, I'll bet that you start getting better overall mixes, since it's probable that at least something, somewhere in your signal path doesn't like being pushed to max at all times.
Thanks for another insightful lesson Yep.

Just to clarify - you're talking about tracking at -20 on the Reaper meters?

Thanks again
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Old 11-07-2010, 11:23 PM   #19
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Well hold up there, cowboy. What average signal level is the original "magic box" supposed to run at? 1 volt? Now, what is the equivalent input level of your guitar track (or whatever) that you're trying to "smooth out" by running through it? 9.5 volts? So the processor is doing exactly what it's supposed to do: it's distorting. And you're now trying to fix distortion with more distortion.
Yep, i asked a similar question in another thread but i have to ask it again here. I also thought that the 'input' level into an 'analogue-sounding' plugin mattered i.e. that it was best to make sure the input was trimmed to the 'zone' the original analogue device was operating in. Again a reason to make sure you leave some headroom on your individual tracks, as you and other have been mentioning.

However, i got told in that other thread that those plugins would 'automatically' trim the input to the 'proper' input range (and applying the appropriate gain after the plugin if needed).

So the question is : what is true ? Or is the answer: "we dont know for sure, and leaving headroom on each track will make sure you will never get the issue in the first place".

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Old 11-07-2010, 06:49 PM   #20
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That said, in a conceptual sense, it sounds like the makings of a smash hit. I would suggesting recording 120bpm quarter-note hits of an 808 kick sample, then triggering hard compression on the white noise instrumental track with the kick keyed to the side-chain input to get that big pumping club effect. Then find a 16-year-old girl to sing some foul-mouthed "get back" lyrics about how hot she is, apply autotune liberally and you should be all set (after multiband look-ahead limiting to within -3dBFS, of course). The white-noise genre is huge right now.
So true its not funny. Although 'get back' lyrics are soo 2010. Kitschy Afrikaans is the new 'thing'.

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There is also the problem of internal digital processors-- are you certain that every plugin you're using has good floating-point internal calculations? And if so, what about your saturation/compression/analog-ifier/guitar effects? How do they know when to start "saturating"? What about the cutoff filters on your AD and DA converters? What about inter-sample clipping that the digital system cannot detect?

this stuff is not always obvious to hear. Your clip LED will light up when you get that screechy white-noise "deep" digital clipping, but you'd probably notice that anyway. This kind of "hidden" clipping tends to have a more subtle effect, just flat-topping the waveform peaks and creating a harsh, brittle, "digital" sound. It's not always easy to hear in the thick of a recording session, but a little on the bass, a little on the high-hats, a little on the snare, a little on the kick, and next thing you know, you've got a "flat", "cheap", "digital"-sounding mix and you're out shopping for ribbon mics and tube preamps.
I have a feeling there's more of this happening than people realize. With some plugins (Waves, BBE Sonic Maximizer come to mind), even the slightest amount of clipping starts to sound really bad.

Basically one can hope that every DSP routine in every plugin is coded correctly to handle clipping and intersample peaks, or simply bring the levels down, and not have to worry about it.

Maybe there's some placebo effect that comes from 'doing the right thing', but after reading about this, I went back to a 'vortex of shit' mix that just sounded undefined, harsh and lacking space. Sure enough everything was pushed right up and the input gain on my limiter (last plugin) was at -9! I brought everything down (making sure no dynamics plugins were effected), and without making any other changes (aside from adjusting the limiter so the RMS was the same is before) exported the mix...

All of a sudden the subconscious 'harshness' and 'fatigue' was gone and replaced by 'openness' and 'clarity'. I was quite impressed. While the difference was not as obvious as when I started properly using HPF/LPF and LCR panning it was a similar 'aha' moment.

The way I see it, trim, like HPF/LPF are on consoles for a reason, because (more often than not) you are supposed to use them!

I wish Reaper's mixer would have these as options too. Not only would the workflow be much better, but less experienced users who only know digital mixing might realize the importance of gain staging and filtering without having to randomly stumble across, and wade though hundreds of pages on why their 'recordings sound like ass' or 'why most ITB mixes don't sound as good as Analog mixes'...
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Old 11-07-2010, 07:01 PM   #21
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I often wonder how much of this - at the lower end anyway - is simply driven by poor monitoring. I mean, if your monitors (or analog monitoring path) is noisy when you crank it up a bit, the tendency is to mix "louder" (with DSP limiters and similar) which we all know is a generally bad idea for that particular use.

It's like trying to squeeze out a better analog S/N ratio by pushing all of the digital music up near "Valhalla". When maybe they just need to deal with the noisy parts of the analog chain... and turn the monitors up for everything.

Looking at commercial music with an RMS of -2 doesn't help. I often wonder what would happen if you feed those signals at unity to a typical cassette deck for recording to tape... what the metering there would look like.
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Old 11-07-2010, 07:50 PM   #22
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"But I have unlimited tracks! Let's use them! Now I duplicate the track seven times, for a total of eight tracks. The meters are showing about -13 RMS and about -1.5 peak. Hey, I even left a little headroom for the mastering engineering! This is guaranteed to sound fantastic and super-analog!"
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This is what made me wonder what you were doing.

If you are running -1.5 peak, now that you are using the 'new method', What were you doing previously?

Were you slamming the outputs over 0 db?
If so, then of course it sounds better now.
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