LOGIC: are the track faders the same as a real board's??

[DrummerMan]

OK, I've seen people talking about this recently and I wanted to clarify.

I've always assumed that the faders in Logic (or DP or any DAW really) acted pretty much just like a fader on an analog board would, in that if you jam it up really hot and loud and it distorts, no matter how low you bring down the master fader, that track will still be distorted/clipping.

I've seen people suggest on other threads recently that if you've got all your tracks in your DAW too hot, then you could just bring down the master fader and all will be right in the world.

I understand that this would be the case if it was just your master section that was clipping, but what if your individual tracks were in the red? would bringing down the master fader "fix" that clipping?

If this were true, then it would seem that the only fader with a threshold in the DAW would be the master.

Am I making sense? I'm trying to think of a clear way to phrase this, but DAW-speak is not a language I know too well.

Any thoughts?

[kingtoad]

I think (although I'm not certain) that this might be part of the reason why there is a dedicated "master fader" in Logic 8. I don't think there was one in 7, just normal "output 1-2" faders or similar. Logic 8 has both, and I assume the master fader works as something to control all of them at once on a mathematical level.

[farview]

Most DAWs have a 32 bit floating point summing buss. That gives it something around 1500db of headroom. You would be hard pressed to clip the buss. You will, however, clip the converters if you go above 0dbfs.

It works the same way on the channels. The thing is, some plugins will distort. Especially plugins that emulate analog designs, these are really sensitive to gain staging.

The bottom line is: if you are working in 32 bit float, you really can't distort the buss or the channel. It doesn't become 'real' until you try to render the file in a normal, fixed point format. Like when you render a 24 or 16 bit wav or when it comes out of the converters.

[DrummerMan]

Most DAWs have a 32 bit floating point summing buss. That gives it something around 1500db of headroom. You would be hard pressed to clip the buss. You will, however, clip the converters if you go above 0dbfs.

I don't really get what this means exactly("floating point summing busses" and "dbfs" fall into the category of DAW-speak I was talking about), but it seems to confirm my suspicion, anyway.

I did a little test, which I suppose I could have done to begin with. I took a drum track and did 4 different bounces:

1st bounce: Track, Master and Out1&2 Faders @ 0db

2nd bounce: 24db Gain plugin on drum track(so it was really crankin'), Track Fader at -24db, Master and Out1&2 Faders at 0db

3rd bounce: 24db gain plugin on drum track, Track and Out1&2 Faders at 0db, Master Fader at -24db

4th bounce: same thing except the Master was at 0db and Out1&2 was at -24


The result? They all sound exactly the same. So, I guess it doesn't even make a difference between the master fader an the Out1&2. As (I think) farview said, the only clipping is happening at the converter level, ?which is, I guess, where bouncing takes place???

Anyway, thanks for your input.

[farview]

I don't really get what this means exactly("floating point summing busses" and "dbfs" fall into the category of DAW-speak I was talking about), but it seems to confirm my suspicion, anyway.

OK, there are two ways that digital info is stored and calculated (for the sake of this painfully simplified explanation), fixed point and floating point.

Fixed point is like using a cash register as a calculator. The decimal point is always in the same place and you can only calculate to the nearest 100th. If you have a 6 digit display, you can display every number between 0000.01 and 9999.99 in increments of 0.01

Floating point is like a normal calculator. The decimal point can move around on the display so it can display a larger range of numbers. With the same 6 digit display, you can display every number between .000001 and 999999 with the increments varying along the way.

In this (somewhat dodgey) analogy, the display digits are the bits. With fixed point math, each bit always represents about 6db. With floating point, each bit can represent anything it has to. This gives you a larger dynamic range to work with as well as finer measurements.

Dbfs is the scale that all digital meters are reading. Db, by itself, really doesn't mean anything. It's a ratio. There are a bunch of different db scales, depending on what you are measuring.

The VU meters on your mixing console read dbVU. If you are measuing the voltage coming out of your mic preamp, you would measure it in dbV or dbu. In order for db to mean anything, you have to know which scale it is refering to.

The result? They all sound exactly the same. So, I guess it doesn't even make a difference between the master fader an the Out1&2. As (I think) farview said, the only clipping is happening at the converter level, ?which is, I guess, where bouncing takes place???

The clipping takes place when you convert the signal to a fixed point format. That happens when you bounce, or mixdown, to a 24 or 16 bit file. It also happens when the sound comes out of the converters. The converters are either 24 or 16 bit fixed.

The converters have nothing to do with the bouncing process. All they do is convert the digital signal to an analog signal that you send to your amps and speakers. Bouncing in the box doesn't touch the converters at all.

[Vogon]

The result? They all sound exactly the same. So, I guess it doesn't even make a difference between the master fader an the Out1&2. As (I think) farview said, the only clipping is happening at the converter level, ?which is, I guess, where bouncing takes place???

Anyway, thanks for your input.

As was implied earlier, the important thing is not to slam the signal into any plugins, so a "clean" eq after the 24 dB gain (or on the output channels) should sound "different" to one that's getting a lower signal (some use a -18dB peak regime for recorded signals). It may not be "bad" one one channel, but on multiple channels, you'll hear it degrading the mix.
There's a slew of posts regarding "Overs" too, when the build-up of waveforms creates spikes that momentarily overload plugins even when a DAW meter doesn't show it. There are some free meter-plugs which are designed to diplay the "real" levels, sonalksis free-g being one AFAIK.
Be aware also - the "master Fader" in L8 appears to affect all outputs, so if you use it whilst sending signals to outboard or reamping, you'll find your compressors or amps suddenly sound wrong - I tend to avoid it and use careful gain-structuring within the mixer section. Keep an eye out for anyone else you may be working with using it without your knowledge too. I've had a few moments where things have gone fully-pear-shaped and had to trace it back to someone dicking with the master fader.

[farview]

(some use a -18dB peak regime for recorded signals).

It should be -18dbfs RMS, not peak.

-18dbfs refers to 0dbVU on your outboard gear. VU meters don't measure transient peaks, so using a digital level of -18dbfs peak could very easily measure -12dbVU in the analog world, which is a little low and is keeping you out of the sweet spot of your preamps and other outboard gear.

[Jay Reynolds]

Most DAWs have a 32 bit floating point summing buss. That gives it something around 1500db of headroom. You would be hard pressed to clip the buss. You will, however, clip the converters if you go above 0dbfs.

This happened in Logic last week:
I was trying to find a very narrow-Q sweet spot on an 808 "bass-drop". I usually apply a good amount of gain while looking for it, than back it down to more reasonable levels once I've found the freq where it rings out. Going back to check my mains, I see the clip light above the meter on "Output 1 and 2" is lit. Not hard to imagine, because of all the energy that I was just sending through the channel and the main buss.
So this clip indicator represents my DAC clipping, not the bus itself?

[farview]

The clip indicator is telling you that you are clipping the conversion from floating point to fixed point. That happens after the master fader and before the converters. If you bring the master fader down, you won't be clipping anymore.

[Jay Reynolds]

Makes sense.
Here's my next one:
Cubase SX doesn't have clip indicators on the channels on the mixer, just the 2 mix. And I never heard a difference in sound quality, even when I had a fader at the tops of its track a the channel meter hammering away. As long as the 2 mix wasn't clipping, that channel just sounded...louder. Logic does have clip indicators. And I do hear a difference when the Logic's channels are "in the red". The SX setup appears to be more like 32-bit float to me. Can't clip the channels, can clip the mains. Or did Steinberg just omit the channel clip light on purpose?

[farview]

I know that Cubase is 32 bit float. It's virtually impossible to clip it internally.

I'm under the impression that Protools is 48 bit fixed. So you can run out of headroom, just not easily.

I don't know about Logic. (I assumed it was 32 float)

[Jay Reynolds]

I don't think it is.

[farview]

HA HA HA! I just googled 'logic summing buss' and the 3rd result was this thread!

[DrummerMan]

Thanks for the explanation. That clears things up mostly for me. I'm pretty stupid with math in general, so the simple analogy was appreciated. I think I got enough of that to keep me going, anyway.

Be aware also - the "master Fader" in L8 appears to affect all outputs, so if you use it whilst sending signals to outboard or reamping, you'll find your compressors or amps suddenly sound wrong - I tend to avoid it and use careful gain-structuring within the mixer section.

The main lesson to me is to use proper gain structure to begin with, and none of these things will be an issue. I sometimes get into that stupid thing where the tracks all start out fine, but then I keep mixing everything up that I want to hear more of, instead of mixing everything else DOWN, like I should (usually). By the time I realize it's over the top, I've put in all sorts of automation, and I (to be honest) haven't figured out if there's a way to simultaneously bring down all (or a selescted group of) faders AND bring down volume automation across the board with a single function. I know I could do all the tracks individually, but that runs into the realm of breaking up my workflow (determined by the laziest, simplest course of action, of course ), so I tend to just turn down the the Master fader.

I don't do this all the time, by any means, and I've gotten alot better over the years, but I need to be reminded of it from time to time.


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The funny thing is, the reason I was asking/thinking about this first was because I had a drum track that was being used on a little Webisode Comedy thing. The director was over at my place to finalize some things.

He says to me, "I like the drums.... but, is there something you could do to make them more fucked up and drunkenly dirty?"

I went to put on a distortion plug, and the level jumped up severely, very fast, of course.

The director yells, "YES!!! That's IT!!!!!"

I'm like, "Cool! Great!" That was easy! Then I look at the screen and realize that everything was maxed out in the red. I start turning down the track fader and he says, "Wait, wait. That's not it. What did you just change?" I of course realize that the sound he was loving was MORE the channel clipping than the distortion plugin. Of course, I have other instruments going on as well that need to contend with this fucked up drum track, without getting fucked up themselves.

My first solution? Do what I'd do with an analog board: I bussed the drum tracks to an Aux track, rammed up the orig track's fader so it was clipping hard, then started to bring down the aux track. You can imagine what happened: "No, no! I liked it how you had it BEFORE!"

I tried compensating with pushing the distortion plug more, but that had a completely different sound that wasn't doing it. My only solution in the end was to bounce the drum track with the master clipping to all shit, then use the bounced track in the mix. Ended up working fine, but seemed like more work than necessary to do something that would have been really easy and quick on a live board.

I guess DAW manufacturers (or at least the folks at Logic) aren't counting on people who would want to use an analog board in this way. Hmmmm....





Anyway, thanks, everybody, for all the input so far.