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Extreme Mixing Messages: 945 Registered: April 2004 Location: Los Angeles, CA
Gold Member
compasspnt wrote on Thu, 12 May 2005 06:03
My firm belief is that if users of Protools, and the other DAW systems, would do the following, then MANY of the "digital" or "in the box" audio problems would vanish:
?STOP RUNNING YOUR SIGNAL SO HOT! Do not use the built in peak meters as you would use a VU meter. If you will keep your input levels lower, your sound will improve. You are not really gaining anything by trying to squeeze out that last little bit.
?If you can, find a way to also meter every input with a good old fashioned analogue VU meter...whether it's with an analogue console, a tape machine, a dedicated "meter box," whatever. Let these meter indications rule, while of course cross referencing the peak DAW meters as well. And use MUSIC as your general guide for I/O levels. Setting up with a 1k tone from inside Protools for your reference level to an analogue console following will not give accurate results on all types of program material.
I absolutely agree with this, and with Paul Frindle's statements.
•If you can, find a way to also meter every input with a good old fashioned analogue VU meter...whether it's with an analogue console, a tape machine, a dedicated "meter box," whatever. Let these meter indications rule, while of course cross referencing the peak DAW meters as well. And use MUSIC as your general guide for I/O levels. Setting up with a 1k tone from inside Protools for your reference level to an analogue console following will not give accurate results on all types of program material.
You are right in principle in everything you say But a VU meter won't help you because it's too slow and will allow peaks to go unnoticed that could cause problems in the digital domain. They were fine when one used them with experience in the analogue domain, where healthy a degree of overload margin was implicit.
Every time this comes up I am left with exactly the same totally frustrated feeling.
The reason people do not get best results from ITB mixes and digital processing in general is that the whole cultural environment of metering, level control, overload and production styles within the digital domain is based on SAMPLE VALUE and not SIGNAL
However many times I re-itterate this very important fact it seems impossible for people to grasp exactly what it means and what the gravity of ignoring it actually is in repect of their audio results. And this is NOT even the user's fault, they cannot be expected to grasp it because they are totally buried in systems that are wholly based on sample value misconceptions and always display values which are NOT signal
For a really fair analysis, this is not primarily a user problem - it is an equipment problem that the user must make himself aware of if he is to avoid it.
People who hear differences are not wrong - the equipment is lying to you - it is ecouraging you to produce illegal results that you are not made aware of.
IMHO & LE this is the sole reason underpinning ALL the arguments about ITB mixing, sample rates, 'resolution' - you name it.
Hi Paul,
Can you expound on this a little bit. How do you suggest ITB mixers set levels both when recording and mixing? What can the equipment makers change to make sure their equipment isn't "lying" to us?
David Schober Messages: 298 Registered: April 2004 Location: Brentwood TN
Active Member
It's been my experience that it's just not DAWs that have this level issue. Anytime I've gone to digital, RADAR, Sony, or a DAW, the issue of level is a problem. Signals with lots of transients like a snare drum usually work fine with digital meters.
However, things with more sustain like a vocal do not. Most preamps and certainly the Neve 1073 I use a lot don't have the headroom to fill up the digital meter. When I first started working in digital I found that I'd get distortion from the mic pre way before I could get the digital meter fill up. Even with things such as a string section, I found that turning up the mic pre too far would make things sound a bit ugly. Even though the mic pre hadn't overloaded, it just didn't sound good.
For me, when working on vocals, strings, and sustained sources I set the mic pre as high as I can as long as it still sounds good. The digital meters are useful to make sure I don't get too soft, but long ago I stopped trying to make all the lights come on. I'm sure there are other mic pres that have more gain and can put more into the D/A, but it's clear to me that while I understand why one would want to do this, I've not found a correlation between good sound and hot levels to digital.David Schober
djui5 Messages: 1511 Registered: May 2004 Location: Phoenix, Arizona
Platinum Member
compasspnt wrote on Thu, 12 May 2005 07:03
My firm belief is that if users of Protools, and the other DAW systems, would do the following, then MANY of the "digital" or "in the box" audio problems would vanish:
•STOP RUNNING YOUR SIGNAL SO HOT! Do not use the built in peak meters as you would use a VU meter. If you will keep your input levels lower, your sound will improve. You are not really gaining anything by trying to squeeze out that last little bit.
This is very true. I've found that when recording/mixing ITB, that if you keep your signal lower (in Pro-Tools it would be a little above where the green meets yellow) that the sound quality is vastly improved. Especially when mixing ITB, if you keep the master fader at 0 and the signal on the master fader down, the whole mix will oepn up and is less aggressive. It seems when you hit the buss hard it starts to get "digital aggressive", like a console would if you pushed it hard, but not anywhere near as pleasing. Morale of the day? Stop looking at what you're hearing.
yngve hoeyland 07'
compasspnt Messages: 15133 Registered: December 2004 Location: Compass Point Studios, Na...
Diamond Member
Paul Frindle wrote on Thu, 12 May 2005 10:38
compasspnt wrote on Thu, 12 May 2005 14:03
•If you can, find a way to also meter every input with a good old fashioned analogue VU meter...whether it's with an analogue console, a tape machine, a dedicated "meter box," whatever. Let these meter indications rule, while of course cross referencing the peak DAW meters as well. And use MUSIC as your general guide for I/O levels. Setting up with a 1k tone from inside Protools for your reference level to an analogue console following will not give accurate results on all types of program material.
You are right in principle in everything you say. But a VU meter won't help you because it's too slow and will allow peaks to go unnoticed that could cause problems in the digital domain. They were fine when one used them with experience in the analogue domain, where healthy a degree of overload margin was implicit.
Exactly Paul...that's why I included:
"...while of course cross referencing the peak DAW meters as well."
compasspnt wrote on Thu, 12 May 2005 20:21
You are right in principle in everything you say. But a VU meter won't help you because it's too slow and will allow peaks to go unnoticed that could cause problems in the digital domain. They were fine when one used them with experience in the analogue domain, where healthy a degree of overload margin was implicit.[/quote
Exactly Paul...that's why I included:
"...while of course cross referencing the peak DAW meters as well."
Thanks!
Actually (at the risk of putting the cat amongst the pigeons) I can suggest a simple experiment people can do themselves to illustrate this in action in the most graphic way, which should dispel any lingering doubt that it's important.
The aim is to show that what looks like a legal 'signal' way below any red light in your system can still represent something that cannot pass even remotely correctly out of your digital mixer at full level. And also to illustrate how this may affect your sound quality in practice when mixing ITB. It's a kind of worst case scenario - but it illustrates the problem.
You need a W/S like ProTools, a signal generator plug-in that has a good filter section that actually goes flat to 20KHz and rolls off at 24dB/oct or so.
- In Pro tools get a mono channel up,
- stick the PT generator plug-in at the beginning of the channel and set it for sine at say 1-2KHz.
- Follow this with a good filter plug-in set for the max slope at 20KHz. (For example the Oxford EQ plug-in has 36dB/oct at 20KHz and illustrates this well - any other good HF filter should work as well).
- As an initial test set the channel fader at 0dB and note that the PT meters shows the sinewave signal at -6dBr and that putting the filter plug-in in and out using bypass has no effect.
- OK now switch the signal genny to white noise and note that the level on PT is still -6dBr.
- Now un-bypass the filter plug-in and watch the signal level rise dramatically!! In the case of the Oxford 36dB/oct filter the meter level will rise a full 5dBr to nearly flat out.
Ok so what's happening - how is this possible? Well the digital genny plug-in produces sinewaves correctly - but when in noise setting it is just a random number generator driving the output. So although when set to -6dB peak value no sample ever gets to be greater than 50% modulation - a reconstruction of the undecoded SAMPLE VALUES produces nearly full level SIGNAL. Reconstruction means filtering and so the filter plug-in is acting like a partial reconstruction filter (much like a DAC) - which in turn is now feeding a more legitimate SIGNAL which the sample value only meter can read more correctly.
Ok now if this SAMPLE train is passing out of your DAC it too is being reconstructed correctly - so this -6dBr noise from the genny would a produce nearly full modulation SIGNAL if you fed this to the DAC directly - even though no sample gets to be bigger than 50% and no reading say's it's bigger than -6dBr.
If your filter is a good one you should be able to switch it in and out and hear no difference in the sound of the signal from your DAC - despite the PT meter reading wildly different. The filter has neither added nor taken anything significant out of the intended audio signal - but you have nearly doubled the sample values within the PT channel!
Ok, now wind the genny level up to say -2 or -3dB (still less than only 75% full level) and do the same thing. What happens? Well it now clips when the filter is in (samples bigger than flat out) - now the sound definitely changes when you switch the filter in and out - because it is mathematically limited and in error when the filter is in - cos it cannot pass through TDM slot at the output of the filter!!
That is what would be happening in your DAC, it would saturate if you sent this at only 3dB setting on the genny - reading -3dBr within the mixer itself, straight to the output!!
Ok now what does this mean for a mix? Well with all those mixed signals, cymbal crashes, HF EQ and limiting etc.. how close do you imagine the output signal can get to being a bit like white noise in places within a real production - even if none of the contributing channels hit the red light? Is this not the exact register of what we term as 'air' and 'resolution'? And people are aiming at max possible mix output levels on meters that do not show SIGNAL.
So why does an OTB mixer apparently sound better than an ITB mixer when you are modulating your digits close to 0dBr (sample value) all over the place? Well all those DACs (flawed as they may be) are acting to legitimately reconstruct your programme - before - you mix them all together and produce too many illegal signals that cannot pass out of your digital mixer! Paradoxically, the loss of sound quality due to all those converters is not as bad as the illegal signals created within the digital mixer by the 'too hot' signals you are trying in vain to pass out of the system.
It is not a summing issue at all (the one thing digits CAN do is add up almost perfectly). It's an illegal output problem caused by the fact that there are no meters that display actual SIGNAL in your whole mixing environment - you simply never see it happening.
So - go back and get your fav test mix back up on your W/S, re-mix the whole thing making sure that at every place in all chains (including between all plug-ins) never gets bigger than -6dBr. Make sure your final output after any limiting etc also never peaks beyond -6dBr. Now do the comparison between this ITB mix and a similar OTB mix. You might have a big surprise
So - go back and get your fav test mix back up on your W/S, re-mix the whole thing making sure that at every place in all chains (including between all plug-ins) never gets bigger than -6dBr. Make sure your final output after any limiting etc also never peaks beyond -6dBr. Now do the comparison between this ITB mix and a similar OTB mix. You might have a big surprise
I am extremely intrigued by your comments. I am trying to reconcile your recommendation with a statement made by digidesign engineering. They claim that it is impossible to clip the internal mix bus. With the maximum number of tracks possible in Pro Tools, all with coherent signals at +6 db on each fader, it is impossible to clip the internal mix bus. Yes it is possible to clip the output, but not the internal mix bus. To avoid clipping the output, they say you can simply lower the gain on the master fader however much needed to avoid the problem. Your statement and digidesign engineering's statement may be completely unrelated, so forgive me if I am linking them together inappropriately. Can you comment?
How could a DAW application be designed to eliminate the problem you point out? Would some kind of an internal reconstruction filter after every track and process be required? Is the problem apparent only in DAW's or does it show up in any digital mixer?
Your experiment recommends never peaking above -6dbr, even after any final limiting. Are you saying it's impossible to ever bring a final mix up to 0dbr without adversely affecting the sound quality? If I put a limiter on a master fader in Pro Tools, the digital summing has already occurred at 48 bits then been dithered to 24 bits before it even hits the limiter plug-in. If I were to sum my mix, never peaking above -6dbr at any stage before hitting the limiter plug-in, then bringing the final level up to 0dbr using the gain on the limiter, would this negate your experiment? Of course I will have to try your experiment for myself to hear what you are talking about, but I'm not at my rig at the moment. I just thought you might be able to add some insight be responding to my thoughts. Thanks.
Curve Dominant Messages: 775 Registered: May 2004 Location: Downtown Philadelphia
Gold Member
blairl wrote on Fri, 13 May 2005 05:19
Paul Frindle wrote on Thu, 12 May 2005 17:54
So - go back and get your fav test mix back up on your W/S, re-mix the whole thing making sure that at every place in all chains (including between all plug-ins) never gets bigger than -6dBr. Make sure your final output after any limiting etc also never peaks beyond -6dBr. Now do the comparison between this ITB mix and a similar OTB mix. You might have a big surprise
I am extremely intrigued by your comments. I am trying to reconcile your recommendation with a statement made by digidesign engineering. They claim that it is impossible to clip the internal mix bus. With the maximum number of tracks possible in Pro Tools, all with coherent signals at +6 db on each fader, it is impossible to clip the internal mix bus. Yes it is possible to clip the output, but not the internal mix bus. To avoid clipping the output, they say you can simply lower the gain on the master fader however much needed to avoid the problem. Your statement and digidesign engineering's statement may be completely unrelated, so forgive me if I am linking them together inappropriately.
My initial take is that you are indeed linking them together inappropriately.
Mr. Frindle has attempted to inform you on how to attain the optimal sonic quality from your DAW.
The Digidesign Answerbase which you quote from, on the other hand, attempts to inform you on how their kit will potentially handle abuse of it.
Measuring the performance of the kit by how well it performs under abusive conditions is not really very helpful, nor informing.
It's kind of like asking, "If I crash my car head on into an oncoming truck at 65mph, what are my chances of surviving?"
Mr. Frindle is attempting to inform you how to avoid the head-on collision.
Digidesign is attempting to inform you of your survival chances.
So - go back and get your fav test mix back up on your W/S, re-mix the whole thing making sure that at every place in all chains (including between all plug-ins) never gets bigger than -6dBr. Make sure your final output after any limiting etc also never peaks beyond -6dBr. Now do the comparison between this ITB mix and a similar OTB mix. You might have a big surprise
I am extremely intrigued by your comments. I am trying to reconcile your recommendation with a statement made by digidesign engineering. They claim that it is impossible to clip the internal mix bus. With the maximum number of tracks possible in Pro Tools, all with coherent signals at +6 db on each fader, it is impossible to clip the internal mix bus. Yes it is possible to clip the output, but not the internal mix bus. To avoid clipping the output, they say you can simply lower the gain on the master fader however much needed to avoid the problem. Your statement and digidesign engineering's statement may be completely unrelated, so forgive me if I am linking them together inappropriately.
My initial take is that you are indeed linking them together inappropriately.
Mr. Frindle has attempted to inform you on how to attain the optimal sonic quality from your DAW.
The Digidesign Answerbase which you quote from, on the other hand, attempts to inform you on how their kit will potentially handle abuse of it.
Measuring the performance of the kit by how well it performs under abusive conditions is not really very helpful, nor informing.
It's kind of like asking, "If I crash my car head on into an oncoming truck at 65mph, what are my chances of surviving?"
Mr. Frindle is attempting to inform you how to avoid the head-on collision.
Digidesign is attempting to inform you of your survival chances.
Make sense?
Yes, I am only trying to point out the pitfalls one can fall into with digital mixing. Just like in analogue, digital also has it's foibles and limitations. A combination of metering methods, misunderstanding about the practicalities of sampling and current trends for loudness at all cost is leading you to conclude (understandably - but quite wrongly) that somehow DAWs cannot sum signals together without loss and plug-in processing is somehow fundamentally flawed.
The same effects and observations are also leading people to conclude (understandably) that higher sampling rates throughout the whole system are somehow philosophically better. And what you percieve within the sound of the errors being created is leading people to conclude that 'resolution' is somehow an issue because that's indeed what the errors sound like - even though that's nothing to do with it. This further encourages people to modulate at the extremes and virtually guarantees that the problems will occur that lead you to make those very conclusions - catch 22!
Digi are entirely correct - you cannot mathematically clip the summing bus this way - like I said it is not a summing issue - it's a question of what can occur WHEN you sum together many signals that may limit (from a reconstructed SIGNAL point of view) from time to time without you being made aware of it - cos no red lights occur.
I know the concept is difficult to grasp from sitting in front of a system and using it - but it really is something worth understanding
I am extremely intrigued by your comments. I am trying to reconcile your recommendation with a statement made by digidesign engineering. They claim that it is impossible to clip the internal mix bus. With the maximum number of tracks possible in Pro Tools, all with coherent signals at +6 db on each fader, it is impossible to clip the internal mix bus. Yes it is possible to clip the output, but not the internal mix bus. To avoid clipping the output, they say you can simply lower the gain on the master fader however much needed to avoid the problem. Your statement and digidesign engineering's statement may be completely unrelated, so forgive me if I am linking them together inappropriately. Can you comment?
Yes they are right - the mix summing has sufficient range to add a great many channels at full level without saturating. But I am not talking about a summing issue as such. I am talking about what may occur in the signal domain if you add lots of hot contributions together where either clipping or significant processing may occur within individual tracks - and you try to modulate at full output from the mix using sample value metering supplied with the DAW. And of course comparing this with a chain that reconstructs every channel before you mix them - i.e. OTB etc..
Quote:
Your experiment recommends never peaking above -6dbr, even after any final limiting. Are you saying it's impossible to ever bring a final mix up to 0dbr without adversely affecting the sound quality?
No - and that's the important point, it IS possible to have a full scale sample value signal that is legal.
For instance in my experiment, had the noise generator (set at -6dBr) been followed by a brickwall filter (like a DAC) it would have given a legal SIGNAL output at full level.
Similarly if an ADC (if properly designed with a good filter) is driven to it's max output modulation, this is also a legal signal that a good DAC (with a good filter) will decode correctly. (However this does rather expose those guy's who propose that the ADCs and DACs should have minimal filters etc..)
SO if you use the system as a straight recorder (with well designed ADCs and DACs) it shouldn't be possible to produce an error of this kind and the sample value metering - although not perfect - will give an acceptable indication of peak programme level.
Quote:
If I put a limiter on a master fader in Pro Tools, the digital summing has already occurred at 48 bits then been dithered to 24 bits before it even hits the limiter plug-in. If I were to sum my mix, never peaking above -6dbr at any stage before hitting the limiter plug-in, then bringing the final level up to 0dbr using the gain on the limiter, would this negate your experiment?
This is an interesting question - and complex to answer in one go because there are 3 situations working together.
Firstly - providing that all the contributions in your mix were entirely legal at every stage, raising the gain to peak levels at the mix output should not result in an error in itself. But in practice it's risky since any contribution that gets processed after the ADC recording that introduces phase shifts, non-linearities or accentuates distortions that existed in the recording at upper mids or HF could result in a reconstruction error at higher contribution gains. In other words raising the levels of a 'troubled signal' may push it into the reconstruction error zone, where previously it was admissible.
Imagine for instance a loudish instrument with rich HF percussive harmonic content that for one reason or another only just reached peak values at the output of your ADC in record. You then EQ it a bit (perhaps rolling off the HF a bit) changing the relative phases of the freqs in the spectrum, noticing that the peak sample value level has dropped a dB or so, you increase the gain to max once more. The drop in peak sample value resulting from a slight re-arrangement of the phase of the freqs - may still have resulted in an almost flat out signal when reconstructed - before you added the gain - now it could overload even though no red light is on.
Secondly we need to consider HOW the limiter acts on the signal. For instance it is possible (even likely) for the fast peak limitting which is popular today to produce it's own illegal signal by modulating the sample values quickly. Imagine for instance a pure sinewave that has had it's peaks reshaped by the fast action of the limiter - this is harmonic distortion which could result in an illegal signal during reconstruction if freq are high enough, despite never producing full level sample values. There are ways of preventing this but many applications do not include them.
A third and more interesting thing to think about is how the limiter sidechain will respond to the levels of the signal. If we go back to my experiment with the noise genny and the filter; we can see that filtering the noise samples produces nearly twice the peak sample value for the same apparent signal level. Now if you follow this with a limiter or compressor that has a sidechain that measures sample values (i.e. acting like your meters) how will it respond in either case if set with a threshold of -6dBr? Well with the filter out it won't compress or limit at all since no values get bigger than -6dBr. However with the filter in it will compress and reduce the reconstructed output to -6dBr again. I.e. it will compress by 6dBr - and the audible result of this will be to drop the signal level by 6dB. The presence of the filter severely reduces the total loudness you can obtain from the limited signal (and this is another story for another day). Try it - it's a real eye opener
Quote:
How could a DAW application be designed to eliminate the problem you point out? Would some kind of an internal reconstruction filter after every track and process be required? Is the problem apparent only in DAW's or does it show up in any digital mixer?
There are several possible ways one might arrange things to avoid this problem within the design of the whole system (particularly if the production culture was more sensible). But we should appreciate by now that making a mixing app of great quality is not simply a question of providing something that 'adds samples together correctly'. And of course this fact blatantly exposes the fallacy of those who try to compare the quality of mixer apps by setting them up identically and taking the results of one away from the other and measuring the differences!!
As with all pro-audio design at least 50% of the design effort is about how you present the information to the user and the nuances of control you give him over it. And of course the cost of the system is important wrt the quality the user expects to obtain and the resilience of the system under duress and not forgetting how it performs within the popular production culture of the times. This subject is too big to discuss here in great detail, but it can be seen that there are largely hidden performance issues regarding digital systems driven to full metered levels that exist at multiple levels, from the quality of the ADCS and DACs (filtering in this case), headroom within the application at the interfaces, plug-in process quality, metering style etc etc.. It is likely that some combinations of system may sound different from others when faced with high sample levels. It is definitely likely that the users' CD players will vary in the artefacts they produce - and this is perhaps the most worrying aspect of it all. In tests I have done most popular CDs produce reconstruction errors at around 2dB or so somewhere in the duration of the production - and paradoxically these are not necessarily the loudest or harshest styles of music. Many of the worst I have are actually (intentionally) clean crisp sounding jazz style CDs. The most often offending programme includes percussion - cymbals, bells, tamborines etc and highly Eq'd (and intentionally clipped) female vocals. Remember that intentionally clipping sounds to produce bite, attack and 'definition' is commonplace in certain quarters of artistic production. How much more clean and crisp would they have actually sounded if mixed and mastered just 3dB less hot
The simplest practical advice right now (with the kit you are currently compelled to use - and if your paymasters will let you) is to think of sample value levels in the green section of the meter as always legitimate (i.e. repeatable at destination). Those in the yellow area (-6dBr and -3dBr) are most probably ok, but caution should be taken as they're definitely big enough and may just cause reconstruction errors if clipped or intentionally distorted in the digital domain or digitally recorded from an artificial source. Levels between -3dBr and 0dBr are dangerous and those that actually reach the red light are almost certainly broken signals and are very likely to degrade in various ways at the destination DAC - in both your's and the end user's!! And above all - don't assume that any meter anywhere within the system indicates a legitimate signal by dint of it not hitting a red light
And to get back to the original subject and my original reason for posting - be aware that by mixing OTB in analogue and encoding to digits via an ADC afterwards - you are removing the possibility of making reconstruction overs in your master. This is very significant within an industry environment where everyone is currently aiming for absolute max loudness and modulation.
gwailoh Messages: 225 Registered: April 2005 Location: Santa Cruz, CA
Active Member
Paul, many thanks for helping us understand this issue better.
Now, supposing one were to follow your recommendations re staying -6dBr down, and so on, mixing ITB. The mix is great, everyone's happy with it. But, you want it to be competitive, that is, loud, in the clubs and on the radio. What to do? Can level be safely added during mastering, or do the same issues apply there when processing the stereo mix one provides to the mastering engineer?
(Would it be better to provide digital stems to master with?)
Bob Olhsson Messages: 3783 Registered: April 2004 Location: Songwriter Gulch Nashvill...
Platinum Member
gwailoh wrote on Fri, 13 May 2005 20:01
...Can level be safely added during mastering, or do the same issues apply there when processing the stereo mix one provides to the mastering engineer?
The same issues apply however the mastering engineer has the benefit of operating within the final context that the level will be set at combined with a variety of different peak-limiting tools and, hopefully, superior monitoring so that any damage can be minimized. If the audio is clean and punchy to begin with, a lot can easily be done. If it has been clipped too many times, it becomes fragile and breaks easily.Bob's workroom (615) 385-8051 http://www.hyperback.com http://www.thewombforums.com http://georgetownmasters.com
Paul, many thanks for helping us (me anwyay) to understand this issue better.
Now, supposing one were to follow your recommendations re staying -6dBr down, and so on, mixing ITB. The mix is great, everyone's happy with it. But, you want it to be competitive, that is, loud, in the clubs and on the radio. What to do? Can level be safely added during mastering, or do the same issues apply there when processing the stereo mix one provides to the mastering engineer?
(Would it be better to provide digital stems to master with?)
Again, many thanks.
Ahh yes - you have picked up on the comments I made in brackets Now this is the rub, it doesn't matter where the gain increase, clipping, limiting occurs, the same thing applies. If the mastering engineer does stuff that causes this all you have is something that doesn't sound like your original and may behave differently from system to system in the user's environment. This is however also true if you mixed OTB, the only saving grace being that it may be less likely to happening twice
Hopefully a good mastering engineer should be savvy enough to avoid doing this, however as we all know the pressure to get the loudest possible results is a significant influence on him as much as anyone else.
There is no doubt that illegal signal can be made to sound louder than legal signal and annoying broken sounds can attract attention in the short term.
The challenge for design of a digital programme limiter is to somehow address both these mutually exclusive situations simultaneously - because if it reduces the overall percieved volume (or initial surprise factor) in comparison with one that produces illegal signal - no one will use it, no one will buy it. There is no point making something wholly 'correct' if it is of no use to anybody! My job is to struggle with such things - certainly makes life interesting
Bob Olhsson Messages: 3783 Registered: April 2004 Location: Songwriter Gulch Nashvill...
Platinum Member
Paul, a couple questions:
1. does the digi white noise generator produce an illegal signal? I can't get pink noise recordings, even that I eq., to do anything comparable using hi-pass or low-pass filters. OK, I just bounced the digi white noise to disk and see the same thing in both Pre Tools LE and Samplitude.
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But a VU meter won't help you because it's too slow and will allow peaks to go unnoticed that could cause problems in the digital domain. They were fine when one used them with experience in the analogue domain, where healthy a degree of overload margin was implicit.
