question about tracking too hot......

[miroslav]

So what exactly do you think varies at different levels between those extremes?

Not looking to answer for Glen....but I kinda gave that perspective some thought...and the only thing I could come up with was that he was suggesting that when levels are real hot=LOUD...it may affect our mixing decisions VS when levels are lower=softer.

BUT...that really isn't much of an issue as you would simply turn down your monitor level if the mix is LOUD....VS a song where everything was tracked at lower levels and you would turn up your monitors.
I mean...in either case, you will always adjust your monitors to whatever listening level is comfortable for you...so the individual/overall level of the tracks becomes totally irrelevant when inside the DAW....doesn't it???
You simply adjust your monitors and then balance out the individual tracks levels as you mix.
Now...when it comes time to sum out a finished mix...you still have to set to overall mix level to a point below clipping, since inside typical DAWs it's pretty much impossible to actually clip the mix while it sits in 32bit float mode.

Granted...if you always track following proper gain staging and you work within acceptable margins of typical analog gear...then your DAW levels will usually reflect that...without much need for concern.

IMO…the problems arise when people are operating their front end at the outer limits of its acceptable margins JUST to try and hit the DAW with a hot level because they read somewhere that was desirable….and while the DAW has no problem accepting a -1dBFS level…their front end is probably soiling itself putting out that kind of level…hence the crappier sounding tracks.

 

[SouthSIDE Glen]

Just forget it, Ethan. You just don't get it and you never will.

Stick to making bass traps. *That's* what you're good at.

This thread needs locking.

G.

 

[Ethan Winer]

in either case, you will always adjust your monitors to whatever listening level is comfortable for you...so the individual/overall level of the tracks becomes totally irrelevant when inside the DAW....doesn't it?

That's exactly my point.

I really don't want to piss folks off, but I thought we were very close to a resolution, with either me or Glen changing our opinion, when I asked:

So what exactly do you think varies at different levels between those extremes? Please be very specific. "Easier to blend" doesn't count, especially because it's entirely anecdotal and can't be proven or backed up with logic. "Because I say so" doesn't count either, nor does "I already explained this five times."

But it seems it's easier for Glen to bow out than answer that direct specific question. That's okay. Again, I'm not here to piss anyone off. I do have an agenda! But it's not making people hate me.

while the DAW has no problem accepting a -1dBFS level…their front end is probably soiling itself putting out that kind of level…hence the crappier sounding tracks.

Maybe, maybe not. Define "soiling itself."

Seriously, this is exactly the sort of precision (in wording) I'm trying to enforce by being such a hard-ass. The OP's question asked about tracking at -12 versus -2, and the correct answer is there's no difference assuming common sense is used when setting preamp and converter levels.

--Ethan

 

[bozmillar]

Off hand, the only reason I can think of a preamp acting not as nice at higher levels would be low slew rate. This would make it so high levels at high frequencies distort. Also, as someone mentioned in another thread on this subject, if you are applying gain on your preamp, the noise contributed by the preamp may not be directly related to the amount of gain provided. I don't know if this part is true or not, but it would be a simple thing to test. This is just saying that when you are reaching the max of preamp gain, providing 3dB more gain may be causing 6dB louder noise from the preamp. Again, I've never tested that before, but it could be possible.

The linear range of an amplifier is not completely linear. How non-linear it is before clipping should be easily measurable though.

Other than those reasons, Ethan, you are right. I've never seen anyone say more than "because I said so."

 

[miroslav]

Define "soiling itself."

It's a creative term..... though I guess the more technical definition would be stuff like...distorted/pinched/compressed w/possible effect on frequencies...etc.


Yeah....I kinda thought there was some semblance of a general consensus in this thread...but I think there are some disagreements about minute details and/or terminology and expressions that are preventing that.

*shrug*

 

[Bristol Posse]

Just a thought

When I need advice about refitting brakes on my cars I talk to a mechaninc who does that for a living.
When I want advice on which plants should go in to the garden in which season I talk to a gardener who does that for a living.

Now with the question of level setting We are seeing an argument between several folks who set up complex signal chains and set levels professionally, for a living, day in day out who's advice is all saying the same thing about how to set levels,and someone who makes audio furniture and wall fittings for a living who disagrees with them

I know less than all of them but I think I know whos opinions I am going to be listening to more closely in terms of level setting and who's advice on room treatment/design I would rather have

 

[bozmillar]

good point, but I've seen many a mechanic screw up a car. I've eaten a lot of nasty food made by great chefs and a lot of good food made by my wife.

 

[Bristol Posse]

good point, but I've seen many a mechanic screw up a car. I've eaten a lot of nasty food made by great chefs and a lot of good food made by my wife.

True enough which is why it's important to get more than one opinion (as in these threads). If they all agree (which all of the recording engineers on this and various other recording boards both pro and hobbiest I have read appear to) it would seem to be a no brainer

I've seen far more cars f*cked up by enthusiastic amateurs than by talented professionals

 

[NYMorningstar]

I really don't want to piss folks off, but I thought we were very close to a resolution, with either me or Glen changing our opinion, when I asked:

"Originally Posted by Ethan Winer
So what exactly do you think varies at different levels between those extremes? Please be very specific. "Easier to blend" doesn't count, especially because it's entirely anecdotal and can't be proven or backed up with logic. "Because I say so" doesn't count either, nor does "I already explained this five times."

But it seems it's easier for Glen to bow out than answer that direct specific question. That's okay. Again, I'm not here to piss anyone off. I do have an agenda! But it's not making people hate me.

--Ethan

Glen could you 'please' answer that from the perspective the recorded signal is already in digital? Unless I have it wrong, Ethan is looking at it from that perspective. I think he agrees with the need to do proper gain staging in analog. Correct me if I'm wrong Ethan.

 

[bozmillar]

Glen could you 'please' answer that from the perspective the recorded signal is already in digital? Unless I have it wrong, Ethan is looking at it from that perspective. I think he agrees with the need to do proper gain staging in analog. Correct me if I'm wrong Ethan.

Glen did mention about some tests that pipeline did, but the search function on these forums are impossible to use. Can you point us to that thread?

 

[noisewreck]

Glen could you 'please' answer that from the perspective the recorded signal is already in digital? Unless I have it wrong, Ethan is looking at it from that perspective. I think he agrees with the need to do proper gain staging in analog. Correct me if I'm wrong Ethan.

Yes, the whole mess is because Ethan is looking at it from the point where the signal is already digital. And everyone else is screaming that they are NOT talking about that.

The crux of this whole Meduza of a thread is BECAUSE everybody else is talking about the audio signal in the analog domain BEFORE it hits the converter.

It is about finding the sweet spot for a given preamp, mixer, analog EQ, analog Compressor, analog whatever that you have in your signal chain BEFORE it hits the AD converter.

Some equipment is linear and has high tolerance. Some equipment has high tolerance, but is purposely not linear (take a 1176 compressor for example). Some gear doesn't have any tolerance and can't hope to be linear. Some feed their mics into a Grace preamp, while others feed it into an Electro-Harmonix preamp which does color the sound quite considerably.

If we listen to Ethan, we should all just use a Grace preamp and then feed that into a Prizm converter

But the world doesn't work like that. (I am not even sure that a Grace preamp will be happy pushing +22dBu(?).

So, if you have a device in your analog domain that can't handle signals that can be pushed to -0.001dBFS once converted to digital, who the fuck cares if your converter can handle it or not?

Bottom line, and the lesson from this thread is:
Keep your analog chain happy, and the digital side of things will reward you.

 

[SouthSIDE Glen]

This isn't the DHS. You guys should be able to get off just measuring the individual local test dots and connect the dots of the big picture by yourselves. Since that seems to be asking too much, let me paint some pictures for you.

I wish to pose four different potential real-world studio situations for consideration. In each, there is a studio operator operating a typical digital mixing station (computer-based or DAW, it doesn't matter). No theoretical laboratory - no oscilloscopes, no calibrated microphone hooked to audio frequency response analyzers, none of that fun stuff that one finds in an anechoic chamber or laboratory, but virtually never finds in an actual real-world studio, whether home or Big Box:

1.) The operator is an inexperienced newb with little-to-no knowledge of gain structure gaming other than the mythical knowledge that you should use all the bits you can without clipping, and he is given a project comprising 10 instrument/vocal tracks each of them tracked to the highest possible RMS values available without causing peak values to clip. Such values were attained by gaming the gain structure to push all equipment and converters hard under the theory of keeping the S/N as high as possible while at the same time avoiding clipping in the converter.

2.) The same rookie operator in #1 is given the same project, except he is given no preconceived notions about making things hot without cliping, and the tracked levels tend to be a bit lower, with tracked RMS values somewhere in the upper-teens to lower twenties negative RMS and peak values typically ranging several decibels below clipping. Such values were attained by paying attention to the VU readings on the analog side and - unless some kind of tube or tape saturation were purposely desired at one stage or another - the signal average was never pushed too hard past 0VU, and such nominal signals were kept that way when pushed through the converter.

3.) It's the same situation as #1, except you replace the operator with a seasoned, experienced engineer who will do what is necessary to make the mix sound as good as possible.

4.) The same experienced engineer is given the project from #2, with the more nominal track levels, and will do what is necessary to make the mix sound as good as possible.


Two questions: In which situation are we most likely to find the best sounding resulting mix?
and
Which situation will require the least amount of work and signal management on the part of the operator to do so?

Not theoretically; theory and test measurement are only part of the whole equation. In which of these real-life situations are those results most likely to happen?

G.

 

[noisewreck]

1.) The operator is an inexperienced newb with little-to-no knowledge of gain structure gaming other than the mythical knowledge that you should use all the bits you can without clipping, and he is given a project comprising 10 instrument/vocal tracks each of them tracked to the highest possible RMS values available without causing peak values to clip. Such values were attained by gaming the gain structure to push all equipment and converters hard under the theory of keeping the S/N as high as possible while at the same time avoiding clipping in the converter.

I am going with #1.

 

[bozmillar]

I'd say 3 and 4 would give equal results. I'm not going to lie, I don't see what this has to do with anything. We've already gone over the fact that we can just turn down the master fader to get the volume at a comfortable level.

 

[bozmillar]

I think Ethan's question is basically asking what "sweet spot" means and what happens if we go outside the sweet spot. I think the answer is quite simple, but for some reason, nobody is saying it.

Do you have more distortion outside the sweet spot? if so, how much more? And what does the distortion look like?

 

[Sound & Vision]

Wow

IMHO a home recording forum is no place to try and outwit eachother over technical aspects of the analog/digital systems we all use. Nothing is proven or dispelled by winning an argument in this setting. It only serves egocentric purposes. Many of the topics being debated have a place in a more professional setting where REAL research is presented and reviewed; not just opinions (which are like assholes). Scientists and Engineers have peer-reviewed journals and conferences for this sort of thing. Might I recommend taking any agendas there?

I think at this point you guys should agree to disagree and move on.

P.S. There is a topic on the gearslutz message board that delves into many of these issues. It features comments by a respected designer of analog and digital audio gear/plugins named Paul Frindle. Wouldn't you know that it turned into a pissing match for a while as well: http://www.gearslutz.com/board/so-much-gear-so-little-time/420334-reason-most-itb-mixes-don-t-sound-good-analog-mixes.html

 

[SouthSIDE Glen]

I don't see what this has to do with anything.

And that's the brick wall that the professional audio engineers that do populate this board keep banging their heads against. I see no way of resolving that impasse, which is why I'd like to put an end to this baloney.

The only reason I even answered again with that last post was because nymorningstar asked me to, and I felt I owed him at least that for all the help and support he has provided me and this board in general in the past.

Anyone who reads my previous post will either get the point or they won't. We all can debate it until we're all blue and cold, and it won't get any further. So let's just stop it, OK? We're just wasting time and bandwidth.

G.

 

[noisewreck]

http://www.gearslutz.com/board/so-much-gear-so-little-time/420334-reason-most-itb-mixes-don-t-sound-good-analog-mixes.html

He brings up some interesting points that I hadn't thought of.

 

[Bristol Posse]

IMHO a home recording forum is no place to try and outwit eachother over technical aspects of the analog/digital systems we all use. Nothing is proven or dispelled by winning an argument in this setting. It only serves egocentric purposes. Many of the topics being debated have a place in a more professional setting where REAL research is presented and reviewed; not just opinions (which are like assholes). Scientists and Engineers have peer-reviewed journals and conferences for this sort of thing. Might I recommend taking any agendas there?

I think at this point you guys should agree to disagree and move on.

P.S. There is a topic on the gearslutz message board that delves into many of these issues. It features comments by a respected designer of analog and digital audio gear/plugins named Paul Frindle. Wouldn't you know that it turned into a pissing match for a while as well: http://www.gearslutz.com/board/so-much-gear-so-little-time/420334-reason-most-itb-mixes-don-t-sound-good-analog-mixes.html

Yeah I loved about the first 4 pages of that thread before it turned into a pissing contest. The stuff that Paul Frindle and others wrote really opened my eyes to the importance of proper gain staging and how badly designed DAWs are to enable that to be done or even considered by those of us whithout analog recording experience was pretty amazing and it was pretty interesting to hear it from the guys who were instrumental in the SSL and Sony Oxford consol development as well as in designing many of the analog emulation plugs

I just wish it had been posted sooner. Oh well next project I'll do better

 

[bozmillar]

And that's the brick wall that the professional audio engineers that do populate this board keep banging their heads against. I see no way of resolving that impasse, which is why I'd like to put an end to this baloney.

I am a professional audio engineer, and I don't see what that has to do with the thread. What you stated there is only referring to digital levels, which is not what the thread is about, unless I completely read the post wrong. I'm not taking the stance that Ethan is taking in that I don't believe that -18dB can be a sweet spot, but I really haven't heard any technical explanation as to why it would be. I have no agenda here, other than to find out why.

If you don't know the technical explanation as to why, then that's fine. It is something that we should all look up and figure out. I've thrown out a few ideas that may or may not be the issue (slew rate, noise, distortion), but as soon as anything even close to technical is mentioned, everyone avoids the subject.