The No.1 Website for Pro Audio
2-Bus Levels?
Old 3rd August 2005
  #1
Lives for gear
 

2-Bus Levels?

I noticed in the "What's on your 2-Buss?" thread that some people use limiters across the digital LR to catch peaks. It started me wondering about what levels some of you guys are mixing to.

I pretty much mix to peaks around -6/-8 on most mixes. I just completed an R&B mix that peaked about -10. I find that it sounds better for digital summing on my d8b and when mixing ITB with SX. If I do any "loudness" adjusting it's always after the mix is done. I'd prefer not to do it at all but you gotta compete in the loudness wars. I also try to recommend against tracking anywhere near zero (-1/-2). I typically track to -8/-6.

There's an interesting paper written by Nika Aldrich that fully explains the flaws in digital metering and why summing (or tracking) up near digital zero (-2/-3?) may not be a good idea.

It's a PDF... Digital Metering ... a good read.

How about you guys, what levels do you track and mix to? Thanks.

Lawrence
Old 3rd August 2005
  #2
Gear Maniac
 
beatzz's Avatar
 

Interesting read, well for me anyways.
Old 4th August 2005
  #3
Lives for gear
 
heathen's Avatar
 

You need a vu meter (ebu)to represent more accurately how the ears percieve loudness as a peak meter is too fast,actually use both and this will give you a better representation of what is really happening.I usually try to mixdown with an rms level of around -18 usually between - 20 and - 16 dbfs,and then catch the peaks with a comp,usuall I like my peaks around - 3 to - 4 dbfs,these types of levels will yeild good loudness when it comes to mastering,when the ME should be able to get roughly minus 12 db rms and peaks bang on zero,when mastering I usually am happy with -14 to -12 rms.With a minus 14 dbfs sine wave the vu should be calibrated to zero vu,then learn how to read it properly and get a feel for it and how audio actually interacts with ears in the real world.Program levels should be around -7 to -5 vu with high peaks reaching 1 or 2 vu,this is just a rough guidline to keep in mind,there are some great articles on this,if you look around
Old 4th August 2005
  #4
Lives for gear
 
absrec's Avatar
 

Here's a tip that I learned recently. Apparently, digital 0 is equal to -18 analog. Like when I'm tracking a vocal into my converters and it goes over that, I do start to hear some audible clipping. A lot of times if I want to print hotter, I'll use the compressor to boost it a little on the way in. Just passing it on...
Old 4th August 2005
  #5
Lives for gear
 

Good stuff, really good advice. I'm going to search some of those documents out.

Although the metering I'm talking about has to do with track levels going to disk and mix levels to the summing bus, not really apparent loudness or how our hearing reacts to those levels. This has more to do with the overall (subjective?) quality of an individual track or a digitally summed mix.

I actually heard a difference when I brought my tracking and summing levels down. I didn't really hear it though until I started recording that way. The tracks and mixes I had previously done sounded noticably harsher. Try it and see if you can hear an overall difference.

I'd suggest anyone recording in a daw leave a good amount of headroom when tracking and mixing digitally. With 24-bits it's all good. You can always make the mix loud as you want later. ME's also seem to prefer it that way.

You don't miss anything but the little bits of distortion that accumulate over the course of many tracks.

Lawrence
Old 4th August 2005
  #6
Lives for gear
 
goldphinga's Avatar
 

alright guys, so in plain terms what should the output meters be reading on the outs of pro/tools or logic on the final mix before i take the track to mastering?
Old 4th August 2005
  #7
Lives for gear
 

Many mastering engineers seem to agree that a peak level of -3 is safe. Here's a long thread on that subject with Bob Katz and Brad Blackwood going over it in detail.

Mixing Levels

Many also agree that that with 24-bits a track or a mix at -20 would sound the same (quality-wise) as a track or mix at -3, except one will be louder. It's funny to hear them talk about the last 4 bits as "marketing bits". Bottom line? Leave some headroom so converters don't distort when it's played back.

The common misconception is that you lower the noise floor by recording hot and get a cleaner signal like when we used tape. You don't with digital. The noise floor is defined by the analog devices you use on the way in or by the converter itself.

I hope this clears it up a little.

Lawrence
Old 4th August 2005
  #8
Lives for gear
 
nukmusic's Avatar
 

Well Lo, i tried that when i first began working with Protools. but the final mix files would be like -23-25db or so.. I found this too low, as my mixes when all analog signal, would be about -16-18bd, which master engineers found to be very cool. So i picked up a tip to use a limiter to boost the signal in Protools and slice peaks at the same time. I make sure that the limiting is about 3db or less. I also listen to the sound of the bus, checking for distortion. works out great
Old 4th August 2005
  #9
Lives for gear
 
norman_nomad's Avatar
Quote:
Originally Posted by absrec
Here's a tip that I learned recently. Apparently, digital 0 is equal to -18 analog. Like when I'm tracking a vocal into my converters and it goes over that, I do start to hear some audible clipping. A lot of times if I want to print hotter, I'll use the compressor to boost it a little on the way in. Just passing it on...
Not always... it depends on how you calibrate your converters. PTHD I think defaults to 0vu = -18dbfs (decibels full scale). You can adjust the converters by turning the small trim pots the back of the unit. PTLE on the other hand defaults to 0vu = -14dbfs. (someone correct me if I'm wrong).

You shouldn't be experiencing audible clipping when running over -18dbfs in your DAW... you might be clipping your pre?

Cheers

Damon
Old 4th August 2005
  #10
Lives for gear
 
JonCraig's Avatar
 

when you say 0vu=-18dbfs... you mean that a pt meter reading of -18dbfs is equal to 0vu, or +4dbu, right? if this is the case, then 0dbfs would be +22dbu analog output, which is pretty hot. and the previously mentioned article touched on the fact that the analog circuits in converters are capable of causing a higher output than +22, although the article seems to refer to this as some sort of magical "reconstruction process", when in fact it's just the analog low pass filters. any filter has a resonance and will ring, and at 44.1, that's one hell of a filter knocking stuff out before 22.05 khz. cheaper converters don't have much headroom (hey, they don't need to reproduce anything above the corresponding analog output of 0dbfs, which we're calling +22dbu here). if the filter rings louder than the available headroom of the device, then we've got distortion in the conversion process. you can't really blame the DAW software... it's the converter that's causing the distortion. i think that you'll find in high-grade converters, there's plenty of headroom to handle filters ringing.

also, a primary advantage of higher sample rates is the low pass filter can be much more gradual, reducing the associated filter ring (although you can never get rid of it... it's science at work).

bob katz's book spends a great deal of print discussing this topic, and he does a much better job of it than i just did.

--jon
Old 4th August 2005
  #11
Gear Nut
 

The filter can never "ring" louder than the transients/signal that precede/succede it. When discussing headroom there is no reason to discuss filters - their "artefacts" (which aren't really artefacts at all, but...) are much lower in amplitude than the signal itself.

Nika
Old 4th August 2005
  #12
Lives for gear
 
JonCraig's Avatar
 

Quote:
Originally Posted by Nika
The filter can never "ring" louder than the transients/signal that precede/succede it.
hey, nika, i see that you're certainly more of an authority than i am. this is how i've understood things. can you clear it up for me?

say we're playing back program material, and a few samples near 0dbfs up around 20khz causes the filter to ring a bit. moments later, we have a few more samples near 0dbfs. the energy of the first sound is no longer present in the digital audio, but is "sticking around" ringing in the filter. add the second bit of sound, and they can have an additive effect, producing a louder tone than each had individually:

"an OVER can be generated even if a filter is set for attenuation instead of boost, because filters can ring."

"One of the most common mistakes made by digital equipment manufacturers is to assume that, if the digital signal "clips" at 0 dBFS, then it's OK to install a (cheap) analog output stage that would clip at a voltage equivalent to, say, 1 dB higher. This almost guarantees a nasty-sounding DAT recorder, because of the lack of cushion in its analog output section."

http://www.digido.com/portal/pmodule...er_page_id=36/

---

"Various filtering strategies are used to remove these frequencies, all of which involve the construction of a very steep low-pass filter, but no filter is perfect and the steeper the response the more the filter tends to 'ring' or resonate at the cutoff point."

http://www.soundonsound.com/sos/aug0...es/digital.htm

---

--jon
Attached Thumbnails
2-Bus Levels?-filterring.jpg  
Old 4th August 2005
  #13
Gear Nut
 

In your original post you said, "If the filter rings louder than the available headroom of the device..." My contention is that the filter can't. Certainly you can get into situations where the accumulation of signals such as the original signal and the filter's dissipation sum together to create a signal that is higher in amplitude than the original signal.

I contend that the real problem here is not so much that the filters' "ring" is the culprit of the clipping problems you are describing. If you look at the impulse response of a typical 44.1kS/s anti-aliasing filter you'll find that the relationship in amplitude between the primary and secondary lobes (peaks) in that "ring" is so vast that it is very unlikely that it would contribute to overs in the way you are describing. On page 294 of my book, Figure 17.30B, I show a set of coefficients used in an anti-aliasing filter. If the maximum amplitude of the transient is 1 then the first compression peak in the impulse response is about .1. Indeed what Bob Katz said is true - that with filters such as are used in EQs this can be a problem, but in the specific application of anti-aliasing or reconstruction filters I think this is completely negligable.

The real problem with those circumstances is the problem I describe on my website in a paper called, "The Consequences of Traditional Digital Peak Meters" that can be found at: http://www.cadenzarecording.com/papers.html

Yes, there are a lot of problems with converters and headroom, but it is not because of the filters and the way in which they dissipate out the transition band frequencies. It is the fact that the signal itself exceeds 0dB regularly even when the samples do not.

I hope this makes sense.

Nika
Old 4th August 2005
  #14
Lives for gear
 
JonCraig's Avatar
 

after an enlightening phone conversation with nika... he's right. while it is theoretically possibly for cumulative effects of filter ring to cause an analog signal higher than what 0dbfs should produce, the inaccuracy of pcm sampling leaves the door wide open for a much larger problem, as described in his paper.

nika, a question i should have asked over the phone... higher sampling rates should, in theory reduce this effect (at least in what we normally consider the audible range), yes?

thanks again!

--jon
Old 4th August 2005
  #15
Lives for gear
 
norman_nomad's Avatar
Quote:
Originally Posted by JonCraig
when you say 0vu=-18dbfs... you mean that a pt meter reading of -18dbfs is equal to 0vu, or +4dbu, right? if this is the case, then 0dbfs would be +22dbu analog output, which is pretty hot.
Yes.. you got it...+4dbu for balanced analog gear. At least this is my understanding. Running your system at -18dbfs, or lower even, allows you to run your analog gear's output stage "hot" if you want to, while still allowing ample headroom in your DAW.
Old 4th August 2005
  #16
Gear Nut
 

Quote:
Originally Posted by JonCraig
nika, a question i should have asked over the phone... higher sampling rates should, in theory reduce this effect (at least in what we normally consider the audible range), yes?
That would be one way to reduce the effect. Another way would be to use better meters.

Nika
Old 4th August 2005
  #17
Gear Head
 
Steved's Avatar
 

Picture's worth a thousand words.

Old 4th August 2005
  #18
Lives for gear
 

Quote:
Originally Posted by JonCraig
after an enlightening phone conversation with nika... he's right. --jon
Nika has this really annoying habit of calling to explain himself. heh

Seriously though, he called me once to clarify some questions I had about dithering. I won't go into detail but I was considering whether or not I should use 24-bit dither at the output stage of my daw mixes.

"After an enlightening phone conversation with Nika..." I had a much better understanding of the concepts and issues and felt reasonably certain I could make a more informed choice.

I still appreciate his taking the time to do that.

Lawrence
Old 4th August 2005
  #19
Lives for gear
 
XHipHop's Avatar
Quote:
Originally Posted by Lawrence
Nika has this really annoying habit of calling to explain himself. heh

Seriously though, he called me once to clarify some questions I had about dithering. I won't go into detail but I was considering whether or not I should use 24-bit dither at the output stage of my daw mixes.

"After an enlightening phone conversation with Nika..." I had a much better understanding of the concepts and issues and felt reasonably certain I could make a more informed choice.

I still appreciate his taking the time to do that.

Lawrence
Did you at least buy his book afterwards to repay him??!?!












I'm a good salesman....hehe.
Old 4th August 2005
  #20
Lives for gear
 

Quote:
Originally Posted by XHipHop
Did you at least buy his book afterwards to repay him??!?!

I'm a good salesman....hehe.
Guilt. The emotional gift that keeps giving.

Actually I'm sorry to say I didn't. Guilty as charged.

Lawrence
Old 6th August 2005
  #21
Lives for gear
Maybe I go about it different..I dont render my tracks in my sequencer.

I use Tape it 2 and capture the 32 bit float file. I then dither it down in soundforge.

When I track, I never pass -6. When I mix, I go by ears..sometimes this means that I have peaks over 0 and I have to put LIGHT compression on the 2 bus to get the peaks back to -2 or so.

but I shoot for -3 when I mix.
Old 6th August 2005
  #22
Lives for gear
 

The thing is... it doesn't matter. -3, -6, -10, -12... it's all good until you get too close to 0. The -3 mix won't be any less noiser or dynamic or smoother or better than the -12 mix. Just louder.

There's no reason why a digital mix should ever hit 0 and have to be compressed back into range. It's simply not necessary to mix that hot. As Nika's paper explained.. if you have an occasional 0 peak your meters can see, there are probably many more that they cannot see.

But if it sounds good... who cares?

Lawrence
Old 6th August 2005
  #23
Lives for gear
here is the issue I have encountered when i mixed lower at -10 or so..

To get the song to an acceptable volume requires a load of compression and L3 usage..I hate that sound..it just gets all muddy and overdone. I mix hot because I like that sound..I do have a mastering engineer who doesn't ever get on my ass about it (he would) so I know im not hitting it too hard.

Now I started thinking..how hot are my mixes? obviously I dont want any square waves in a mix..so i took a look.

So this is interesting..im my sequencer, your dthink this will be a hot mix w/ huge waveforms..but this is the exact waveform from my last mix....





to me, this doesn't look like a hot mix..is my sequencer's concept of zero a little off here?

Im really not sure now.
Old 7th August 2005
  #24
Lives for gear
 
JonCraig's Avatar
 

what do the meters in the daw say? surely there's an option for meters to peak hold, or something of that sort. what level is your highest peak?

--jon
Old 7th August 2005
  #25
Lives for gear
 

Quote:
Originally Posted by Methlab
to me, this doesn't look like a hot mix..is my sequencer's concept of zero a little off here?

Im really not sure now.
It's hard to tell from the photo becaue most daws allow making waveform displays bigger or smaller irrespective of the actual signal level, so you can perform accurate edits. That could be a -2 mix or a -10 mix. It depends on what the display setting is. It doesn't looked clipped at all though.

FWIW... I also hate maximizing also but mastering guys have better tools than I do. heh

Lawrence
Old 7th August 2005
  #26
Lives for gear
 
ProFool's Avatar
 

So at the end what do you have to aim for?
Lets say you recorded everything at 0vu on the preamps +4 on the convertors and levels are running -18dbfs in the daw, what do you aim for at mixing stage? do you get them up to 0dbfs in digital or keep them at -18dbfs while mixing? im getting a bit confused here, in the Pt's hardisk live from a few years ago i read to get your levels at 0dbfs digital before mixing, so whats true about it? im not talking about the 2buss, i mean the mix channels. i never seem to know what to do with that Bf essential meter , i even doubt if its correct, or should i buy 2vu meters to use in the daw. Maybe ill better buy your book Nika if this will explain it all heh thanks & grtz
Old 7th August 2005
  #27
Gear Nut
 

Quote:
Originally Posted by ProFool
So at the end what do you have to aim for?
Lets say you recorded everything at 0vu on the preamps +4 on the convertors and levels are running -18dbfs in the daw, what do you aim for at mixing stage? do you get them up to 0dbfs in digital or keep them at -18dbfs while mixing? im getting a bit confused here, in the Pt's hardisk live from a few years ago i read to get your levels at 0dbfs digital before mixing, so whats true about it? im not talking about the 2buss, i mean the mix channels. i never seem to know what to do with that Bf essential meter , i even doubt if its correct, or should i buy 2vu meters to use in the daw. Maybe ill better buy your book Nika if this will explain it all heh thanks & grtz
You should track in at levels that are comfortable for your analog equipment. Pushing 0dBFS is probably pushing your analog equipment too high, and you have so much headroom in a digital converter these days that you should sooner pay attention to analog levels than digital levels (except for avoiding clipping).

As for mixing approach it like an analog mixer. All of your signals should be fairly hot but nowhere near clipping. Then mix accordingly, sending signals in and out of "processors" (plug-ins) at these levels. Upon summing, raise or lower the master fader to bring the level to the appropriate level for release.

I hope this helps?
Nika
Post Reply

Welcome to the Gearslutz Pro Audio Community!

Registration benefits include:
  • The ability to reply to and create new discussions
  • Access to members-only giveaways & competitions
  • Interact with VIP industry experts in our guest Q&As
  • Access to members-only sub forum discussions
  • Access to members-only Chat Room
  • Get INSTANT ACCESS to the world's best private pro audio Classifieds for only USD $20/year
  • Promote your eBay auctions and Reverb.com listings for free
  • Remove this message!
You need an account to post a reply. Create a username and password below and an account will be created and your post entered.


 
 
Slide to join now Processing…
Thread Tools
Search this Thread
Search this Thread:

Advanced Search
Forum Jump
Forum Jump