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Offline georgeh

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normalize vs eq soundforge
« on: July 25, 2006, 03:03:31 PM »
Just curious about this. If my levels are all low then I usually normalize. but sometimes I lower the level when the band gets louder. Ya i should probably leave them lower to start, but i don't. It seems fine to just raise the whole show by say 2db's. I am correct in assuming that with normalize you don't change as much of the dynamics.
I don't raise each song, some are meant to be lower.
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Re: normalize vs eq soundforge
« Reply #1 on: July 25, 2006, 04:50:17 PM »
Yep, you are right.

If you are recording in 24 bit, try to keep the peaks around -10 to -15 or so.

Teddy


Offline tapeworm48

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Re: normalize vs eq soundforge
« Reply #2 on: July 25, 2006, 05:13:25 PM »

Teddy,

couple of questions:

1.  are you talking about peaking the levels between -15 and -10 during recording, or in post processing?

2.  i've heard alot of talk about low levels w/ 24-bit not being a major issue because it can be corrected in post.  jacking up levels for 16-bit recordings in post, not that good.  any idea why that is? 

thanks.
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Re: normalize vs eq soundforge
« Reply #3 on: July 25, 2006, 05:33:53 PM »

Teddy,

couple of questions:

1.  are you talking about peaking the levels between -15 and -10 during recording, or in post processing?


During recording.

Quote from: tapeworm
2.  i've heard alot of talk about low levels w/ 24-bit not being a major issue because it can be corrected in post.  jacking up levels for 16-bit recordings in post, not that good.  any idea why that is? 

thanks.

Yes, anyone who tells you to peak around 0 with a 24 bit word is giving you very bad advice.
Buckle up, long answer..(this is from my post in the Audio reference material thread..)


if you record with a 24 bit word, the noise floor is so low that setting levels that peak well below full scale is fine, still way above the noise floor.

Each bit you add to the word doubles the available values the word can represent, and therefore doubles the dynamic range (signal to noise ratio from full scale down to noise) that you can record.

A doubling of dynamic range equates to 6db. Therefore, each bit in the word contributes 6db of dynamic range. A 16 bit word therefore has a 96db signal to noise ratio, and 24 bit word can express 144db of signal.

In the real world, the audio electronics in the converter provide a higher noise floor than a 24 bit word can represnt, so a good 24 bit converter will give, lets say conservatively, 110db of signal to noise.

This means that if you record your audio with peaks no higher than -14db under Full Scale, you'll still be experiencing a recording with 96db of dynamic range, which is the best any 16 bit CD has every accomplished.


To make the point even more graphically - this all assumed that the source signal has a dynamic range in excess of 96db too. I would bet you a beer that it isn't even close. There's no tube mic that operates that cleanly. Your studio room has noise higher than that. All your hardware compressors and EQs operate with a much higher noise floor.

If you were very careful, and ended up having a source with 70db of dynamic range (congratulations!) you could record it with peaks at -26dbFS (-26 under full scale) and still have preserved every ounce of dynamic range.

So its obvious that hitting full scale isn't necessary at all - why not preserve some headroom just in case?  Let's say you do make it just under full scale. No harm in doing that if you don't go over, right?

Well, what do you do when you want to EQ something +2db? Where does that 2db go? Into clipping of course, unless you lower the input level of the plug in, which is going to lose any hypothetical S/N benefit you had preserved anyway.

Even more importantly, when you record this hot, I've got to ask - what did you do to your preamps, and analog chain to get this level? Most converters are set so that 0dbVU = -18dbFS.

That means that if you're getting -6 below full scale on your converter, that you're +12db over the 0VU point! Many analog electronics can crap out here, but almost all will sound different at least. Some times it may be "better" but usually, its a small accumulation of distortion that builds into a waxy fog that makes people blame "digital recording" for its pristine playback of their slightly distorted, but "pretty on the meter," tracks.

If you record with levels around your 0 point, some thing like -18dbFS or -14dbFS, depending on how your converter is calibrated, you'll have your analog electronics in their sweet spot, headroom for plug ins and summing, an appropriate analog friendly level if you use analog inserts later in the process, and on a modern 24bit converter with 110db S/N, the ability to faithfully record signals with a dynamic range of over 90db.

And by all means, 0dbVu is no glass ceiling like 0dbFS is. Keeping levels around 0dbVU doesn't mean that peaks won't exceed that by 6 (or more) db. If they do, your ability to record 96db of S/N (if you even have it in the source, and you don't), just like the best CD you ever heard will be preserved, if peaks don't exceed -12dbFS! More if they do.


) A 24 bit PCM word can express a theoretical limit of 144 db of S/N.

2) The analog electronics in the converter limit the performance to a functional 100 db of S/N. (slightly more in some cases, but I'll use a conservative figure and make the point even without those extra 6 db)

3) As long as the noise floor in any recording system is lower than the noise floor in the signal you're recording, you will record the full dynamic range perfectly.

4) No source you've ever recorded had a signal to noise ratio higher than 80 db, and most would be much much lower. Lynn suggests that he RARELY sees the source's noise floor lower than 70 db down, and even then, rarely. Assuming that his peaks are not at full scale, his typical source S/N ratio must be in the 50-60 db range?

This means that if you record your (best ever) 80db S/N source into a converter so that the highest peak just reaches -19 dbFS (below full scale) on the meter, that the noise floor in your signal will be louder than the noise floor in the converter. You needn't record it any hotter than that.

In the real world, you could get away with peaks around -28 dbFS, and be PERFECT. Any higher than that is totally unnecessary.

Conclusion: There is absolutely NO benefit to tracking hot.

But does it hurt to do it? Read on...

1) Your microphone preamp is set to perform best (gritty distorted choices aside) peaking around 0dbVU. This is where you'd have it set if you were recording to analog tape, hitting 0 on the VU meter. Plug that same source into most converters, and you get peaks around -20dbFS to -14dbFS, depending on how the converter is setup.

The scientists who developed this system understood the situation, even if the guys who wrote the digidesign manual don't! They EXPECT you to record with peaks around 0VU (-18dbFS on the digital scale). They KNOW about the signal to noise deal I explained earlier. That's why they chose to put the nominal level so "low" on the meter.

When you record hotter, with peaks at -6dbFS, lets say. You're driving your mic preamp 12 db hotter than you did yesterday in the analog world! That's going to add a subtle layer of distortion to your project. And they say analog sounds so much better than digital - maybe its because most people use their analog gear incorrectly when recording to digital. Maybe the "problem with Pro Tools summing" is really the effect of tracking too hot?

I've heard people say "My Neves can handle outputs +24db according to the spec, so what's the big deal?" My Neve 1073s are great sounding workhorses. They are rated for a LOT of gain. Still, they definitely sound very different even at +12. Very different. Maybe a good choice in some cases, but not the norm.

2) If you have a peak at -2dbFS, and you try to boost a mid range frequency +3db on an equalizer, you're going to clip.

Another unintended detriment to tracking hot is that you no longer have any headroom in your plug ins! It is true that in Pro Tools, you can recover lost headroom in the mix bus by lowering the master fader. This isn't true in an analog console, where the distortion has happened in a summing amp "upstream" on the master fader. In that case, the master fader only lowers the volume of the distorted signal, which remains distorted.

In Pro Tools, the master fader is actually a co-efficient with each individual fader before summing. This means that if you're clipping the mix bus, you can pull the master fader down, and fix it. Great. But what about the plug ins across each fader? They aren't affected by the master fader (thank god, or your compression levels would change etc) but neither are they protcted by the master fader. If you're clipping the mix bus, and have your master fader at values lower than unity, then odds are that you're clipping some plug ins too.

3) Most analog gear doesn't like inputs that are 12db and more over 0, even if the spec says they can take it. If you track hot, you're causing a nightmare for analog gear that you may choose to insert during the mix. Keep your levels around 0dbVU, and you can leave the digital domain freely without adding more sonic grunge.

Conclusion: Tracking hotter than 0dbVU can easily cause distortion in any number of places in the chain.

So, to reiterate:

1) There is absolutely NO benefit to tracking hot.

2) Tracking hotter than 0dbVU can easily cause distortion in any number of places in the chain.

If you want to hear the result of tracking too hot, and what it does to Pro Tools, listen to any Lenny Kravitz record. believe me, he uses all the best vintage gear, with gobs of headroom etc. There is no shortage of Neve, Helios, Fairchild, Neumann, Telefunken or whatever on his sessions. The sound of those records is entirely due to the tracking and mixing levels.

"But how do I get my product hot?"

There is a point to having a final mix that peaks at -0.1dbFS. if you are going to have a 16 bit version, if you want to be commercially competitive, if you like to see all the lights light up - sure, I do it every time. The point is i bump it up LAST in plug ins across the master fader. That way, the mix is all properly gain staged, with lots of headroom right up until the last thing juncture. Then if I raise the result to just below clipping after having the benefit of proper levels all the way through, everything is beautiful.

If you are a non believer, try it. The amount of air, detail and image is astonishing. In fact, eventually you may find that Pro Tools is actually TOO CLEAN and transparent! Then you'll start introducing purposeful distortion in your mix - distortion that YOU control at the mix is a very different animal than the unwitting accumulation of crud that comes from tracking too hot all along.

So all this means is that the noise is SO low in a good modern 24 bit converter that you can keep gobs of headroom for proper interfacing with analog gear, and still get the full 96+db of dynamic range, just 12 to 18 db lower on the meter. Your analog gear will thank you too.

So all of this results in a pristine, beatiful, airy, detailed 24 bit mix with peaks around -12dbFS? Cool! .[/i]
« Last Edit: July 25, 2006, 05:35:43 PM by Teddy »

Offline georgeh

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Re: normalize vs eq soundforge
« Reply #4 on: July 25, 2006, 06:21:48 PM »
Sorry I jumped off line. I was refering to 16 in my question, stilll running the D8. But thanks for the info!
I've always tried to run my levels damn close to zero, but I have stepped back to shooting for around -2 or 3, watching teh levels for teh first ew tunes, and backing off very slowly if need be. Some folks have said just run them lower and if you want after the show raise them in SF.
I do appreciate the tech facts as to why do what you are saying
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Offline georgeh

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Re: normalize vs eq soundforge
« Reply #5 on: July 25, 2006, 06:27:13 PM »
let me add one more thing. i run a MG210>M148>smb1>d8. the sbm1 doesn't do a very good job on the limiter, clips easier then others. so i keep my eye on the sbm 1, not the d8. if the lights come on the sbm 1, for more then a split second, i lower my levels on it.
thanks again
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RebelRebel

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Re: normalize vs eq soundforge
« Reply #6 on: July 25, 2006, 06:39:41 PM »
let me add one more thing. i run a MG210>M148>smb1>d8. the sbm1 doesn't do a very good job on the limiter, clips easier then others. so i keep my eye on the sbm 1, not the d8. if the lights come on the sbm 1, for more then a split second, i lower my levels on it.
thanks again


Even at 16 bit, I would only shoot for around -3 to -5...there is absolutely no exuse for clipping. I dont understand the practice of wanting to use every bit of headroom....Dynamic range is a good thing. Buck the loudness trend, set the levels and forget em, and use that thing we call a volume knob!

Teddy

Offline pfife

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Re: normalize vs eq soundforge
« Reply #7 on: July 25, 2006, 09:07:28 PM »
I am correct in assuming that with normalize you don't change as much of the dynamics.


this is correct, as long as you are using peak normalization, and not RMS normalization.  Each is an option within the normalization window in the old version of SF that I use (6.0).

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Offline CQBert

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Re: normalize vs eq soundforge
« Reply #8 on: July 25, 2006, 11:05:11 PM »
As a thought... normalize by channel as opposed to the sum.  SF finds the peak in the combined channels and normalizes from it... if you do a L and then a R you get a better balanced sound IMO...

Try it and see what you think - look at the wave forms if you do both at once v 1 at a time. 

In the end it is all personal taste.

I will also echo Teddy, 24 Bit and Hot runs are not necessary to get a quality recording... I personally know/knew my gear and most of my rooms well enough to peak in the -6 to -4 range but never sweated running a little below that and bringing things up in post.

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Re: normalize vs eq soundforge
« Reply #9 on: July 26, 2006, 12:16:44 AM »
Teddy...

Thanks for that writeup. I really appreciate your contributions here. That was one of the clearest explanations I have ever read on the headroom front. Especially the analog stage parts. Ever since I got my R4 (I know ur not a big fan, but bear with me), I've been recording a lot of 24-bit matrixes or 4-mic mixes, and I've been getting better and better at leaving that headroom both so I don't clip, but also for the summing later in my production chain when mixing the 4 channels together and when using EQ, verb, etc. (Not to mention the quantization benefits of mixing/mastering at 24-bit). I've also been lucky enough with the acoustic jazz I tend to record and the mics I have, that I've been able to run mic-in for most stuff, which lets me leave the gain way down more in the sweet spot of the R4's pres (I think). And I've settled into a workflow just like you described above (learned some of this from the Mastering Audio book too), I always leave the last step in my production chain for compression and dither on the master bus (I use Vegas) right before saving to 16-bit (sometimes I compress more like a normalize that goes just a bit farther, and other times I'll really smoosh it harder, it depends, but let's not start that debate here or I'm libel to be -T'd, LOL!). Anyway, just want to say thanks!
« Last Edit: July 26, 2006, 12:21:46 AM by BayTaper »
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RebelRebel

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Re: normalize vs eq soundforge
« Reply #10 on: July 26, 2006, 03:50:25 AM »
Thanks man. Glad it helped.
Teddy...

Thanks for that writeup. I really appreciate your contributions here. That was one of the clearest explanations I have ever read on the headroom front. Especially the analog stage parts. Ever since I got my R4 (I know ur not a big fan, but bear with me), I've been recording a lot of 24-bit matrixes or 4-mic mixes, and I've been getting better and better at leaving that headroom both so I don't clip, but also for the summing later in my production chain when mixing the 4 channels together and when using EQ, verb, etc. (Not to mention the quantization benefits of mixing/mastering at 24-bit). I've also been lucky enough with the acoustic jazz I tend to record and the mics I have, that I've been able to run mic-in for most stuff, which lets me leave the gain way down more in the sweet spot of the R4's pres (I think). And I've settled into a workflow just like you described above (learned some of this from the Mastering Audio book too), I always leave the last step in my production chain for compression and dither on the master bus (I use Vegas) right before saving to 16-bit (sometimes I compress more like a normalize that goes just a bit farther, and other times I'll really smoosh it harder, it depends, but let's not start that debate here or I'm libel to be -T'd, LOL!). Anyway, just want to say thanks!

Offline pfife

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Re: normalize vs eq soundforge
« Reply #11 on: July 26, 2006, 09:04:36 AM »
As a thought... normalize by channel as opposed to the sum.  SF finds the peak in the combined channels and normalizes from it... if you do a L and then a R you get a better balanced sound IMO...

concur


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Offline sullen

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Re: normalize vs eq soundforge
« Reply #12 on: July 30, 2006, 11:09:54 AM »

Yes, anyone who tells you to peak around 0 with a 24 bit word is giving you very bad advice.
Buckle up, long answer..(this is from my post in the Audio reference material thread..)

you're an angel.
this was a great writeup.
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Offline morningdew

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Re: normalize vs eq soundforge
« Reply #13 on: July 31, 2006, 10:22:33 AM »
To continue this discussion I have a question:

Let's say you were going to do some compression on a track and it peaks at -5 dB.  My software allows me to add gain after it compresses to boost the signal.  Should I do this now to get the track to - 0 dB?

Or should I compress (we'll say anything over -20 dB at a 2.5:1 ratio for arguements sake), adding no gain and then normalize the recording to - 0 dB.

If I did it the first way there is really no need to normalize because the track is now peaked at -0 dB.  Or is it better to just compress with no added gain and then normalize?

Thanks.

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Re: normalize vs eq soundforge
« Reply #14 on: July 31, 2006, 10:50:04 AM »
The only time I do much compression is when I have a really quite show with loud applause - I will compress the applause and then re-normalize the recording to get it where I want it.

I tend to only add gain at the very end if a final tweek is necessary.(maybe 1db or a fraction of one)  I prefer to normalize each channel and see where that leaves me.

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RebelRebel

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Re: normalize vs eq soundforge
« Reply #15 on: July 31, 2006, 11:01:12 AM »
To continue this discussion I have a question:

Let's say you were going to do some compression on a track and it peaks at -5 dB.  My software allows me to add gain after it compresses to boost the signal.  Should I do this now to get the track to - 0 dB?

Or should I compress (we'll say anything over -20 dB at a 2.5:1 ratio for arguements sake), adding no gain and then normalize the recording to - 0 dB.

If I did it the first way there is really no need to normalize because the track is now peaked at -0 dB.  Or is it better to just compress with no added gain and then normalize?

Thanks.

why bring things up to 0???  As long as your peaks are reasonably set, you are golden.....leave the dynamic range preserved instead of trying to squeeze all the volume out...UNLESS you are trying to be commercially competitive...



« Last Edit: July 31, 2006, 11:03:49 AM by Teddy »

Offline morningdew

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Re: normalize vs eq soundforge
« Reply #16 on: July 31, 2006, 12:24:53 PM »
No not trying to be commercially competitive but I just assummed almost everyone normalized to 0 or a hair under.  Especially if your uploading your source for e-sharing.

For example, I just taped a Black Crowes show (16-bit) that I plan to share.  Because of this thread I decided not to stress over levels and just dialed it in and let it ride instead of getting it just right so it hit 0.  It sure was a lot stressful and easier to do and I ended up with a nice show and it looks like my highest peak was about -4.5.

Normally, I would just fade-in the beginning, fade-out the end, normalize (each channel seperately) to 0 and be done with it.  Because of this thread I thought I would try an across the board compression on the whole show and make two copies for myself.  One copy made using my normal method and a second copy where I tried some compression.  Then I would listen to them both mainly just to learn what it does, how it sounds and to learn something for myself.

However, if I leave the copy that I plan to distribute at -4.5 w/o normalizing, I have a feeling my friends will immediately notice and make comemnts like "hey, this show has low levels compared to other stuff you've given me..."

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Re: normalize vs eq soundforge
« Reply #17 on: July 31, 2006, 01:23:57 PM »
No not trying to be commercially competitive but I just assummed almost everyone normalized to 0 or a hair under.  Especially if your uploading your source for e-sharing.

<snip>

Normally, I would just fade-in the beginning, fade-out the end, normalize (each channel seperately) to 0 and be done with it.  Because of this thread I thought I would try an across the board compression on the whole show and make two copies for myself.  One copy made using my normal method and a second copy where I tried some compression.  Then I would listen to them both mainly just to learn what it does, how it sounds and to learn something for myself.

However, if I leave the copy that I plan to distribute at -4.5 w/o normalizing, I have a feeling my friends will immediately notice and make comemnts like "hey, this show has low levels compared to other stuff you've given me..."

Hey Morningdew,  I am running into the exact same concerns as you.  I have a bunch of old DAT masters, and I feel uncomfortable putting them out there with the levels so low.

General question to people... Do you care if the volumes are low-ish?  Or would you rather
have something normalized +/- compressed so it sounds a little more reasonable?  I'm not talking commercial-level in-your-face volumes, just something a bit above -30/-20 db.  I didn't deliberately
set the gain so low, I usually tried to get the peaks at about -12 db and that's just where things
leveled out.

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Re: normalize vs eq soundforge
« Reply #18 on: July 31, 2006, 01:29:57 PM »
I dont normalize at all really..if I do, it is to about -8 to 10 with the highest peaks. I use a very light compression ratio..1:17 to 1 and set the Limiter threshold at -.30




No not trying to be commercially competitive but I just assummed almost everyone normalized to 0 or a hair under.  Especially if your uploading your source for e-sharing.

<snip>

Normally, I would just fade-in the beginning, fade-out the end, normalize (each channel seperately) to 0 and be done with it.  Because of this thread I thought I would try an across the board compression on the whole show and make two copies for myself.  One copy made using my normal method and a second copy where I tried some compression.  Then I would listen to them both mainly just to learn what it does, how it sounds and to learn something for myself.

However, if I leave the copy that I plan to distribute at -4.5 w/o normalizing, I have a feeling my friends will immediately notice and make comemnts like "hey, this show has low levels compared to other stuff you've given me..."

Hey Morningdew,  I am running into the exact same concerns as you.  I have a bunch of old DAT masters, and I feel uncomfortable putting them out there with the levels so low.

General question to people... Do you care if the volumes are low-ish?  Or would you rather
have something normalized +/- compressed so it sounds a little more reasonable?  I'm not talking commercial-level in-your-face volumes, just something a bit above -30/-20 db.  I didn't deliberately
set the gain so low, I usually tried to get the peaks at about -12 db and that's just where things
leveled out.



 

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