Home » R/E/P » Reason In Audio » DAW & Desks: Is ANYBODY actually still mixing on their desk?
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #65900 is a reply to message #65856 ] |
Sat, 14 May 2005 18:34   |
Paul Frindle Messages: 372 Registered: May 2004 |
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| Eric Bridenbaker wrote on Sat, 14 May 2005 19:19 | Thanks, Paul for clarifying this issue.
Recently I've found that running the mixes at lower overall levels has yielded far better results, a certin clarity and punchy transparency that is quite audible, compared to a "hotter" mixing approach.
Even though an analog VU meter is too slow to catch these overs, I find it to be a helpful tool, to have some form of metering OTB.
I have a question about ITB meters though: Is it not possible to get better metering of these anomalies ITB?
There is a plugin by Elemental Audio calle Inspector, for example which is supposed to show clipping instances.
http://www.elementalaudio.com/products/inspector/index.html
Is something like this of any use in getting a visual on this stuff ITB?
Cheers,
Eric
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Yes it is quite easy to make a meter plug-in that reconstructs the signal and shows you the value (I have made such processing as integral parts of other applications) - and you could put it where you like in the mix - i.e. the main outputs for a start. It is fairly costly in terms of processing load though, so you certainly wouldn't want to pepper such a thing all over the tracks in your PT mix!!
Such a meter would considerably help you to avoid these problems - IF you could afford the processing load and were prepared to reduce modulation levels to satisfy it's readings (in reference to the points made by others about the 'target volume level culture')? I have certainly pondered on how readily the industry might accept a meter that sometimes read higher than another on the same programme, that caused people to produce mixes that - although more accurate - sounded less loud? As a designer in the present climate however, I am understandably more interested in somehow providing people with the best of both worlds 
All that aside though, it is much cheaper to simply modulate at lower levels. After all, in the days of analogue ssytems, VU meters and PPMs, even with 10dB of headroom on the average tape machine, no one would attempt to record a cymbal crash or tamborine at 0VU would they - if they were after an accurate rendition of it's character! In those days we were used to inaccuracy and system limitation, we dealt with it using experience and even subverted it's side effects for our own artistic purposes. As I have said in many previous threads (everyone yawn!) - IMHO the whole human perception of math being something of a religious higher cause - greater than reality itself, is the main reason we are reluctant to accept that digital systems could legitimately posses limitation and imperfection. Or was it that we have been brought up to believe it by decades of (religiously crusaded) marketing spin? Whatever - we still discuss to death the possibility that digits cannot add up - when in fact thats the easiest of functions that it can perform to an arbitarily higher accuracy than we could ever need. When the very issues with which we actually cause it to falter (because lately we WILL stress it so) are hidden by dint of metering that suffers from the very same issues.
Please note that my experiment using a conventional IIR filter isn't good enough to form the basis of an accurate measurement method, because it has a large amount of natural phase shift across it's freq range (being a music programme filter). Also, it will still misbehave (wobble about) around integer divisions of the sample rate cos its output signal is at base rate - i.e. it's output still requires reconstruction! It was only an illustration.
BTW to witness the wobbling peak sample meters at divisions of sampling rate, set a sinewave to 11.02KHz on a system running 44.1KHz sampling (or 12.01K at 48KHz) and watch the meters vary cyclically as the sample values wander in and out of phase with the sampling rate - (you may need to fiddle very finely with the osc freq to see it at it's worst depending on meter fall rates etc). This is a graphic illustration of aliassing caused by displaying unreconstructed sample values. This is also a taste of what your DAC would do - if it suffered from insufficient filtering.
But to finally answer your question - I had not heard of the device in the link you just posted, but a quick look at the site doesn't reveal a presence of the all-important reconstruction processing? So it may not fit the bill? However there may be others that might.
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #65960 is a reply to message #65856 ] |
Sun, 15 May 2005 03:25   |
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zetterstroem Messages: 765 Registered: December 2004 Location: Copenhagen |
Gold Member |
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| Eric Bridenbaker wrote on Sat, 14 May 2005 19:19 | Thanks, Paul for clarifying this issue.
Recently I've found that running the mixes at lower overall levels has yielded far better results, a certin clarity and punchy transparency that is quite audible, compared to a "hotter" mixing approach.
Even though an analog VU meter is too slow to catch these overs, I find it to be a helpful tool, to have some form of metering OTB.
I have a question about ITB meters though: Is it not possible to get better metering of these anomalies ITB?
There is a plugin by Elemental Audio calle Inspector, for example which is supposed to show clipping instances.
http://www.elementalaudio.com/products/inspector/index.html
Is something like this of any use in getting a visual on this stuff ITB?
Cheers,
Eric
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i don't think it will work..... it needs to emulate the reconstructio process in the d/a
as this : http://www.tcelectronic.com/Default.asp?Id=3322
it's called gibb's effect...
we talked about it here:
http://recforums.prosoundweb.com/index.php/t/4140/6691/?SQ=1 8552ac44754d36523d0c0da1d75d3b5
i glad someone is debating it again.... i think we all really need to rethink our approach to digital sound.....
Noting the music industry's complaints that illegal downloading means people are getting their music for free, he said, "Well, why not? It ain't worth nothing anyway." (b.dylan)
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #66079 is a reply to message #65960 ] |
Sun, 15 May 2005 17:50   |
Paul Frindle Messages: 372 Registered: May 2004 |
Active Member |
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| zetterstroem wrote on Sun, 15 May 2005 09:25 |
| Eric Bridenbaker wrote on Sat, 14 May 2005 19:19 | Thanks, Paul for clarifying this issue.
Recently I've found that running the mixes at lower overall levels has yielded far better results, a certin clarity and punchy transparency that is quite audible, compared to a "hotter" mixing approach.
Even though an analog VU meter is too slow to catch these overs, I find it to be a helpful tool, to have some form of metering OTB.
I have a question about ITB meters though: Is it not possible to get better metering of these anomalies ITB?
There is a plugin by Elemental Audio calle Inspector, for example which is supposed to show clipping instances.
http://www.elementalaudio.com/products/inspector/index.html
Is something like this of any use in getting a visual on this stuff ITB?
Cheers,
Eric
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i don't think it will work..... it needs to emulate the reconstructio process in the d/a
as this : http://www.tcelectronic.com/Default.asp?Id=3322
it's called gibb's effect...
we talked about it here:
http://recforums.prosoundweb.com/index.php/t/4140/6691/?SQ=1 8552ac44754d36523d0c0da1d75d3b5
i glad someone is debating it again.... i think we all really need to rethink our approach to digital sound.....
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Yes I agree entirely. But we should also remember that this is not new, it's more a question of 're-learning' the correct approach. The possibility for these peaks has always been there and in the 'old days' DACs with analogue filters might have coped with it - but only by accident as a result of the analogue nature of the filter. However in a world where 'more is always better' and spec competition gets ever-harder, sacrificing 3dB or more SNR in an oversampled DAC with a gain reducing digital filter is perhaps something manu's won't risk? Also it has to be said that the unstoppable march towards ever more loudness has made the whole thing much more important anyway - it is exposing other short cuts in the kit i.e. meters, DACs, dynamic limiters etc.
Another aspect worth noting is that DAC manus can attempt to justify a lack of such headroom (dogmatically) on the 'defensible' basis that such peaks cannot be generated by a fully legal PCM audio signal - i.e. any signal coming from an ADC. So the existence of such 'errors' is always the result of some extra processing the user is doing - ergo it's someone else's fault, (even if all they did was EQ the signal a tad) - no need to sacrifice a few dB from their own published specs.
This kind of attitude is increasingly evident in many aspects of the kit manus. For instance the clock jitter sensitivity issue - apparently that's not their fault either - if jitter affects the converter's quality it's not the fault of their product, it's your clock. They save a few bucks in the converter - you shell out hundreds of bucks on expensive clock gennys and lose the battle in the cabling anyway. And of course the blasted 'A' weighted noise figures they publish that preferentially (and IMO misleadingly) improve the SNR numbers on anything that has wideband noise (particularly HF prone DSM converters) and bear little resemblance to what you will actually hear. Stick your mix through an 'A' weighting filter and see if you think that's representative 
There are many other examples, but these kinds of liberties are the direct result of ever more fierce commercial competition IMO. Wholsesale free market competiton alone does not always ensure we obtain the best results. In the last DAC's I designed I sacrificed up to 8dB of potential SNR spec in order to linearise the harmonic distortions people might just hear - and the figures were quoted unweighted of course. But then, these were designed for what was then still a 'professional market' that had professional outlooks and priorites.
But the practical point right now is that your programme is destined eventually to play out on 'who knows what' converter in the cheapest possible consumer design produced under massive commercial pressure - so it's essential to remain on the side of caution
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #66081 is a reply to message #66079 ] |
Sun, 15 May 2005 18:03   |
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Very nicely stated, Paul. Thank you for the awesome detail.
Now everyone turn it all down a bit...
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #66152 is a reply to message #60868 ] |
Mon, 16 May 2005 06:38   |
zed Messages: 4 Registered: May 2005 |
Welcome |
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Paul,
I have been running some tests in my protools system that have been nothing but inconclusive.
Basically what i've done is to grab a session with 20 tracks, mix them as loud as possible without peaking individual or master tracks, bounce, and repeat the above but with trim plugins at -6dB inserted in every track. (they all have some plugins that are time constant)
When comparing both stereo mixdowns the difference between them is zero, they basically cancel each other when phased inversely.
I'm sure that i'm not understanding the difference between mixing at high levels or low levels since they are giving me similar results!
please comment what i'm doing wrong!
Best Regards
Zed
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #66167 is a reply to message #60868 ] |
Mon, 16 May 2005 08:36   |
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Both working on an analogue console AND working inside a DAW is fun...Music is Fun...if it's not, you're in the wrong business!
It just comes down to what sounds best, and how much of your music you have to "see"...
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #66186 is a reply to message #66152 ] |
Mon, 16 May 2005 10:25   |
Paul Frindle Messages: 372 Registered: May 2004 |
Active Member |
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| zed wrote on Mon, 16 May 2005 12:38 | Paul,
I have been running some tests in my protools system that have been nothing but inconclusive.
Basically what i've done is to grab a session with 20 tracks, mix them as loud as possible without peaking individual or master tracks, bounce, and repeat the above but with trim plugins at -6dB inserted in every track. (they all have some plugins that are time constant)
When comparing both stereo mixdowns the difference between them is zero, they basically cancel each other when phased inversely.
I'm sure that i'm not understanding the difference between mixing at high levels or low levels since they are giving me similar results!
please comment what i'm doing wrong!
Best Regards
Zed
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Well, I would guess that your mix and processing isn't producing the peaks even at higher levels - it's a clean mix? Offending processing normally includes HF EQ boosting, compression/limiting or distortion addition - coupled with instruments producing fierce HF i.e. vocals, small percussion, brass etc.
In this case the peaks may not be seen within the mix itself but may be created in the final product if you bump up the gain and limit hard on the output buss for A&R acceptance?
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #66218 is a reply to message #60868 ] |
Mon, 16 May 2005 11:59   |
zed Messages: 4 Registered: May 2005 |
Welcome |
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Well, the mix i've been using is a fairly standard pop song with no vocals, only drums, bass guitar and electric guitars, the plugins used are EQ and delay, no compression (does compression behaves differently when changing -6dB on signal input even with the same value applied to threshold?*). I repeated the test, this time i didn't even care if the mix sounded good at all, i tried to get the summing of tracks as hot as possible (on the mixdown with no trim plugin on every track the master fader even peaked once!).
Still, when checking the -6dB mixdown with the hot one the only difference between them was that evil peak on the hot mixdown, other than that, they were mathematically identical!
It's truth that i'm not using compression and the added HF on some of the tracks is rather moderate, but shouldn't this be enough to start getting some differences when comparing these mixdowns?
The recordings were all made with apogee converters not going any further than -5dB on input!
Paul, with this tests i'm not trying to prove you wrong, believe me on this, i'm only trying to understand your statement trough practice, for all our best interest it is very important for us to understand our tools as best as we can!
*- i did try this same experience using compression, in this case the difference was very noticeable, although it seemed more like a compression problem...
Best Regards
Zed
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