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Max headroom?

Eurocide

Active Member
Hi,

Today I found out, that the 1176LN and Fairchild are sounding best in Cubase SX when the input signal is NOT near 0db. With a headroom of -6db before going into the 1176 LN, it unfolds its full power without unwanted clipping. Especially on drums and percussive basses.

Any other opinions on driving the inputs?
 

MASSIVE Mastering

Active Member
I'm on the \"never-bring-a-24-bit-digital-signal-over-minus-6dBfs-no-matter-what-until-the-final-mastering-phase\" side of the fence.

Record with peaks no higher than -6dBfs, mix with peaks no higher than -6dBfs...

I only wish I could spit out (superior sounding) masters like that, but my clients would kill me. :twisted:
 

Andy_T

Member
Hi,

I've never thought about this. Do you have particular reason for this or
explanation ? not that I don't believe, just curious and always looking
ways to make better mixes.

I'm using Apogee converters and often use softclip in very mild manner as 2nd dynamics after pre-comp stage. Softclip isn't doing anything in my setup before -4dbfs. I've found this works very well for me.


Andy
 

Plec

Venerated Member
A big HELL YEAH! here too.. keep them levels down. The more headroom you have, the better. When recording 24bit audio you don't really need to push it. If the project is going to CD later on you could basically record every track at -48dBFS and still get the same useable dynamics in the end.
 

BobYordan

Member
Yup, I also agree, keep the levels medium/low on the mix, then bring them to a final level in the mastering. :) Mixdown is for me more a stage where the independent sound sources are balanced & eq:ed to a relative referens level, with mastering in consideration as the next step.
 

Ashermusic

Active Member
Well this is all fine and good for recording audio such as singers, guitars, reeds, etc. but as a composer (engineer only by default because of budget considerations) AFAIK when the levels get down too much lower on Virtual Instruments there is actual bit loss and therefore sonic vitality lost so it is a bit of a compromise. Also, according to Echo, my Layla 24 has more headroom built into it than most audio cards so that the headroom of your card is something that needs to be taken into account when determining levels.

The bottom line is that there has been a lot of discussion on the Roger Nichols and George Massrnburg forums on whether making -6 the detente point is wise and opinion is far from as unanimous as it is here.
 

Dr_Jones

Member
Hey Ashermusic,

I am also using an Echo Layla 24 (I've had it for about 4 years now and it runs solidly) My question is - have you upgraded your echo drivers to the latest 6.11 yet? I did that and am starting to have a crackle and pop issue with my system. Not sure if it has to do with the new drivers, or perhaps the pops are coming from a new vst I am using called Stylus RMS. Also, can you tell me what you have your latency set to? I am currently hovering around 4ms (which I think is about 256 samples) so I may bump that up to 512 samples. Thanks.
 

spiritman2

New Member
Yes I did load the new drivers

No problems here. But latency is not an issue for me. I record on an outboard device. Then light Pipe it in thew my Layla24....
 

Ashermusic

Active Member
Henchman said:
as a composer, you shoudl definitelt have no reason to be that hot.

I do post for a living. And one of our biggest complaints is receivign music scores that are constantly maxed out.
Wait a minute, help me understand. I am not arguing but trying to learn from your perspective If you are mixing a show, not mastering, and I give you audio files that max out at 0 db but are not clipped why would this be a problem assuming your dialogue and effects were recorded at proper levels. Now if the composer or engineer has clipped the audio trying to get it as hot as possible that is obviously an issue. Neither I nor my enginners have ever had a complaint on this from the post houses here in L.A.

BTW the bit resolution loss of virtual instruments at low levels is a real issue. So lately I have started to adopt the practice of converting the V.I. tracks to audio files before I mix, as soon as I am committed to the part. Then I do not have to worry so much if I need to lower the volume of a track.
 

Ashermusic

Active Member
Dr_Jones said:
Hey Ashermusic,

I am also using an Echo Layla 24 (I've had it for about 4 years now and it runs solidly) My question is - have you upgraded your echo drivers to the latest 6.11 yet? I did that and am starting to have a crackle and pop issue with my system. Not sure if it has to do with the new drivers, or perhaps the pops are coming from a new vst I am using called Stylus RMS. Also, can you tell me what you have your latency set to? I am currently hovering around 4ms (which I think is about 256 samples) so I may bump that up to 512 samples. Thanks.
No, no new drivers and I am generally at 256 wikth no problems. I also have Stylus RMX. Could it be you are simply overtaxing your CPU? Also, what host?
 

Dr_Jones

Member
oops, sorry about that . . . didn't mean to hijack the thread. I am interested in your initial question as well, as I always thought it was ok to mix close to 0db without clipping or using a limiter and then send it off to post house for them to work on it. Is the general consensus to mix up to -6db when working in 24 bit? Also, is the -6db max limit on each channel (ie drums, vox, synth etc. shouldn't go above -6db) or are we talking about just the master stereo out shouldn't go above -6db. Am I correct in thinking that each individual channel can be mixed up to 0db (to cover peaks) so long as the sum of all of those channels doesn't go over -6db on the master out channels? thanks.
- Doc
 

neil wilkes

Venerated Member
Personally, I won't go close to -6dB peak on the mix buss, but then I use the K-system for all metering.
K-14 for stereo, K-20 for multichannel.
Meter like this, and hot levels are just not an issue any more, plus every other house knows exactly what your tracks were mixed to for reference.
 

MASSIVE Mastering

Active Member
If every single 24-bit digital signal would PEAK at -6dB, the world would be a better sounding place.

One of the big reasons - Use your imagination here -

Think of an analog sine wave - Now digitize it and run it up to 0dBfs. Simple right?

NOW, play it back through a D-A converter... What happens to the \"curve\" of the signal now? That's right, it's clipped. The analog signal, freed from being simply ones and zeroes, follows the analog rules. The peaks will go below AND ABOVE where the digital signal \"plotted\" the path. There are no straight lines connecting the dots anymore. The now analog (and naturally curved) lines can exceed the plotted points by quite a bit.

Superior D-A converters handle this with much more style than lesser converters (like a standard CD player or less-than-stellar sound card), but the distortion and the \"guessing\" is still there.

This is just the tip of the iceberg on the \"idiot's version\" of the physics behind this. It's about all I'm qualified to explain. :) There are some great white papers and studies on this. If I can find some links, I'll post.

If it makes any difference, I commonly get e-mails (from people who used to try to get as close to 0dBfs at every stage) saying what a HUGE difference this makes in the sound quality of their recordings. From my own experience (I used to worry too much about \"0\" also), I can testify to this also. The open quality, clarity, punch - All greatly improved in the final product.

Of course, after mastering, few would want a disc that had a top level of -6dBfs... But when you think about it this way, if everything is lower and therefore less distorted during this final phase, then end volume will likely be able to be pushed farther than if it was distorting the M.E.'s converters in the first place...

Better-sounding at high volume... What a concept...
 

Ashermusic

Active Member
Dr_Jones said:
oops, sorry about that . . . didn't mean to hijack the thread. I am interested in your initial question as well, as I always thought it was ok to mix close to 0db without clipping or using a limiter and then send it off to post house for them to work on it. Is the general consensus to mix up to -6db when working in 24 bit? Also, is the -6db max limit on each channel (ie drums, vox, synth etc. shouldn't go above -6db) or are we talking about just the master stereo out shouldn't go above -6db. Am I correct in thinking that each individual channel can be mixed up to 0db (to cover peaks) so long as the sum of all of those channels doesn't go over -6db on the master out channels? thanks.
- Doc
Once again, and if I am wrong I am willing to be educated, assuming you audio card has decent headroom, as my Layla does, and assuming your program has decent headroom as Logic does, 0 db, whether on a channel or an output, is the point at which you get maximum volume without clipping. Of course if every channel is at 0 db you will be clipping the crap out of the output obviously. So you have to find the right mix (that's why it is called mixing) to get the best result. IMHO there is no magic formula.

What I have learned, as I said earlier, is that if I lower the volume on Virtual Instrument channels I lose bits and sonic quality but if I convert those tracks to audio I can lower them without that issue.

If I am delivering a mix then obviously I must be concerned with how it is all hitting the ouptut. But if I am delivering stems to someone mixing a show on ProTools for example I should AFAIK be able to have all the stems reach 0 db unclipped and it should not be a problem for them.
 

neil wilkes

Venerated Member
Wrong.
There is nothing special about 0dB on your soundcard as opposed to anybody elses.

What you fail to understand here is basic physics.
Try this:
Run a file at 0dB through an analogue process - DAC - tape - ADC.
You will get back a clipped, distorted wave.

This is because your measurements are done wrong. Do not use 0dBFS as a reference, this is cack handed.

If you insist in this, then tell us, please, just what your 0dBFS is calibrated with reference to, exactly?
 

Ashermusic

Active Member
neil wilkes said:
Wrong.
There is nothing special about 0dB on your soundcard as opposed to anybody elses.

What you fail to understand here is basic physics.
Try this:
Run a file at 0dB through an analogue process - DAC - tape - ADC.
You will get back a clipped, distorted wave.

This is because your measurements are done wrong. Do not use 0dBFS as a reference, this is cack handed.

If you insist in this, then tell us, please, just what your 0dBFS is calibrated with reference to, exactly?
Am I to understand that you are saying that if I open up a softsynth and record a a sine wave and it shows 0 db on Logic's channel meter, 0db on the Layla Console app, and 0 db on Logic's output that I am recording a clipped sine wave?
 

cAPSLOCK

Active Member
MASSIVE Mastering said:
I'm on the
Record with peaks no higher than -6dBfs, mix with peaks no higher than -6dBfs...
I am with you for the most part... but I am of the opinion that once in floating point dont worry about levels in floating point.. worry about overall levels. In other words, if my snare track is peaking high it is OK as long as my master levels are reasonable.

If I am a fool (again) I need to know. ;)

cAPS
 

neil wilkes

Venerated Member
cAPS, you have it right.


Ashermusic, As for what I said earlier, a sine wave is not the same thing as a mixed track - by a long shot.

I repeat - what is your 0dBFS calibrated to please?
 

Joachime

Member
I'm not shure I got the message right. Is this 0dB issue occuring when you run the signal through DA - AD or at all times?
I e, am I affected by the issue just by simple DA converting for monitoring purposes?
 
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