Home » R/E/P » Reason In Audio » DAW & Desks: Is ANYBODY actually still mixing on their desk?
Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #66549 is a reply to message #66479 ] |
Tue, 17 May 2005 19:56   |
Curve Dominant Messages: 775 Registered: May 2004 Location: Downtown Philadelphia |
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| Extreme Mixing wrote on Tue, 17 May 2005 17:30 | I think you have to look at it this way, Eric. Any signal processing that you do in a mix requires some digital headroom to get the math done correctly. So leave a little room.
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Got it, I will...uh, I mean...I do.
| Quote: | You wouldn't want to use subtractive eqing because you HAVE TO in order to keep from clipping. Print at lower levels so you can do what you want in the mix.
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Excellent point.
| Quote: | And give it to mastering the same way, so their options are open, as well.
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Very much with you on that one as well.
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #66836 is a reply to message #66487 ] |
Wed, 18 May 2005 23:09   |
Extreme Mixing Messages: 715 Registered: April 2004 Location: Los Angeles, CA |
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| David Schober wrote on Tue, 17 May 2005 10:20 |
| Extreme Mixing wrote on Tue, 17 May 2005 11:30 | Make loud copies for the producer, the artist, and yourself, for reference listening, because we all know that you can't send clients out the door with music that's 12db lower than everything else they compare it to.
Steve
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I'm with you there Steve. However my only fear about making a hot rough is that sometimes people can fall in love with a rough like that.
When turning a rough mix for a sales conference or A&R meeting I'd be a fool to turn in a soft, but good sounding mix, knowing that other mixes will be smashed and louder than mine.
However, my worry is that someone, the artist, record exec, or whoever, has tin ears, used to hearing smashed crap may prefer the rough over the master. Despite the fact that after a proper mastering the mix would sound better, it's hard to make people belive that will be true.
Any thoughts, comments?
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People do get used to what they have been listening to for a while. When i print mixes, I don' t usually turn them up til it hurts. I just don't have the stomach for it. My mixes usually sound like I wanted them to, so there isn't that much heavy lifting for the mastering guy. He just needs to balance the levels and make the mixes sound like they belong together. I always make two mixes, one with the limiter and one without. The one without is for mastering. I'm willing to let the guy with half a million dollars worth of gear for processing stereo mixes do his job. If the guy knows what he is doing, it's pretty hard to beat his mastering job with a $500 plug. Most of the time, I like what they do.
Sometimes producers will "L2" their own projects. I hate that. It's funny how you can always tell, even months later, that someone has tampered with your mix.
Steve
Steve
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67036 is a reply to message #67026 ] |
Thu, 19 May 2005 19:20   |
Curve Dominant Messages: 775 Registered: May 2004 Location: Downtown Philadelphia |
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| bobkatz wrote on Fri, 20 May 2005 00:19 |
| Eric Vincent wrote on Tue, 17 May 2005 11:46 | OK. But let's for a moment assume that my levels are not clipping, that I've kept my inupts and levels at a conservative position.
In that scenario, doesn't one get less risk of clipping by EQing subtractively, rather than additively?
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All other things being equal, I would tend to agree. But Paul (who is more expert on this than I) might intervene and say that if a high pass filter is considered "subtractive" then beware of the intersample artifacts of the high pass. Let's see if Paul catches this post.
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Notwithstanding Paul's views on the subject, I have my own observations of the effects of ITB high-pass filtering: Beware. I have switched settings on EQ plugins from a high-pass to a low shelf cut many times, because the high-pass setting introduced "something" undesirable, whereas the low-shelf cut alternative did not. Antectdotal, yes, but antectdotes sometimes serve a purpose.
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67111 is a reply to message #67036 ] |
Fri, 20 May 2005 04:57   |
Paul Frindle Messages: 372 Registered: May 2004 |
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| Eric Vincent wrote on Fri, 20 May 2005 01:20 |
| bobkatz wrote on Fri, 20 May 2005 00:19 |
| Eric Vincent wrote on Tue, 17 May 2005 11:46 | OK. But let's for a moment assume that my levels are not clipping, that I've kept my inupts and levels at a conservative position.
In that scenario, doesn't one get less risk of clipping by EQing subtractively, rather than additively?
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All other things being equal, I would tend to agree. But Paul (who is more expert on this than I) might intervene and say that if a high pass filter is considered "subtractive" then beware of the intersample artifacts of the high pass. Let's see if Paul catches this post.
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Notwithstanding Paul's views on the subject, I have my own observations of the effects of ITB high-pass filtering: Beware. I have switched settings on EQ plugins from a high-pass to a low shelf cut many times, because the high-pass setting introduced "something" undesirable, whereas the low-shelf cut alternative did not. Antectdotal, yes, but antectdotes sometimes serve a purpose.
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The answer to this depends entirely on the actual signal content. Eq can cause 2 types of 'clipping error' - boosting Freq ranges increases level and changes phase so level clips are obviously more likely. Cutting EQ reduces gain (which intuitively seems safer) however it also changes the phase - so overs are still possible from cutting something.
This situation is worsened by processing already clipped signals (even at LF) as the max sample value is totally reliant on the phase of the harmonics. You can test this by getting an oscillator plugin set to say 200Hz squarewave at -6dBr then filtering below 100Hz - the sample values will almost double depending on the steepness of the filter.
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67224 is a reply to message #67111 ] |
Fri, 20 May 2005 17:45   |
Paul Frindle Messages: 372 Registered: May 2004 |
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| Paul Frindle wrote on Fri, 20 May 2005 10:57 |
| Eric Vincent wrote on Fri, 20 May 2005 01:20 |
| bobkatz wrote on Fri, 20 May 2005 00:19 |
| Eric Vincent wrote on Tue, 17 May 2005 11:46 | OK. But let's for a moment assume that my levels are not clipping, that I've kept my inupts and levels at a conservative position.
In that scenario, doesn't one get less risk of clipping by EQing subtractively, rather than additively?
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All other things being equal, I would tend to agree. But Paul (who is more expert on this than I) might intervene and say that if a high pass filter is considered "subtractive" then beware of the intersample artifacts of the high pass. Let's see if Paul catches this post.
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Notwithstanding Paul's views on the subject, I have my own observations of the effects of ITB high-pass filtering: Beware. I have switched settings on EQ plugins from a high-pass to a low shelf cut many times, because the high-pass setting introduced "something" undesirable, whereas the low-shelf cut alternative did not. Antectdotal, yes, but antectdotes sometimes serve a purpose.
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The answer to this depends entirely on the actual signal content. Eq can cause 2 types of 'clipping error' - boosting Freq ranges increases level and changes phase so level clips are obviously more likely. Cutting EQ reduces gain (which intuitively seems safer) however it also changes the phase - so overs are still possible from cutting something.
This situation is worsened by processing already clipped signals (even at LF) as the max sample value is totally reliant on the phase of the harmonics. You can test this by getting an oscillator plugin set to say 200Hz squarewave at -6dBr then filtering below 100Hz - the sample values will almost double depending on the steepness of the filter.
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Sorry I didn't answer the last part of the question about why an LF shelf in cut sounds different to a High pass. I too have done this switch when mixing. I have a rather odd explanation for it.
I'm not sure what you have experienced but if we leave the clipping issue aside, the most obvious thing is that the shelf EQ will aim towards a specific max attenuation perhaps -20dB whereas a HP filter basically aims at total cut (at DC at least). This means that some of the cut freqs remain with the shelf which allows the effect to soften somewhat - the whole thing is more gentle. Which leads me to the next bit.
IMHO the most important thing is that for any order of HP filter above 12dB or so the rate of change of freq (and phase) around the turnover point will be higher than the LF shelf. This is significant IMHO since fast rates of change (high orders) produce the effect of a distinct ringing (or boosting) near the turnover point (this is why you can apparently increase the bass sound of a bass guitar by rolling off the LF just below the lowest freqs in the instrument). This is some of the reason why valve gear with transformers (that almost always roll off at LF extremes) actually end up sounding more 'bassy' and full.
This is most interesting as it isn't down to Gibb's effect of the filter itself (i.e. signal ringing) - it's actually a psychoacoustic effect. It seems that if our ears encounter an unnatural removal of a freq range it retaliates by making us hear an apparent rise in the freqs in the immediate range just before the loss is encountered. To counter this effect in the old days when I mixed stuff, when I wanted to dramatically remove either HF or LF in something that was prominent in the mix, I would place another EQ section just below (or above) the turnover freq of the HF filter (or LF filter) and put a small fairly high Q (around 3-4 dip) of around a dB or so in the response. This sounds flat whereas the filter alone certainly doesn't! I have done many trials on this over the years and it always holds true. I have no formal idea why this should be the case (only personal theories) - but it's one of the many anomalies of our hearing process (I have a mental catalogue of them!).
You can try this by sticking pink noise into an EQ that has a filter section included (or use 2 plug-ins) and experimenting with this arrangement by ear - it's very odd to hear.
Getting the right sound is a question of what we actually hear, rather than blind adherance to any expectation based on plots on a graph 
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67226 is a reply to message #60868 ] |
Fri, 20 May 2005 18:38   |
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Very interesting, Paul. It appears obvious that this odd effect holds true universally, both analogue and digital.
Would it then not make sense for a HP or LP filter design to include "automatic" reduction of the nearby "offending" freq range, and that said reduction would be somewhat "hidden" to the operator, in that the labelled choice would not "mention" it? Or have you already done such a thing?
Terry
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67233 is a reply to message #67134 ] |
Fri, 20 May 2005 19:33   |
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bobkatz Messages: 2926 Registered: June 2004 Location: Orlando |
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| innesireinar wrote on Fri, 20 May 2005 08:43 |
bobkatz wrote on Mon, 09 May 2005 21:28
stay away from (too many) digital compressors
BK[/quote |
Bob, can you clarify this?
By using compressor plugins instead of true analog compressors instatiation is there a big difference in a mix done ITB?
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I work with one or two excellent mixing engineers who are doing just that, mixing in the box but routing to outboard compressors and reverbs and other gear. Sounds great to me... the key is they know what to do and how to get the most dimensionality and depth with their mixing technique.
The problem is that to do a compressor plugin right requires tremendous CPU power, more than the manufacturers of plugins believe you are willing to tolerate with today's CPU's. In due time all digital compressors will be 2 to 4 times oversampled internally. One way around this is to track and work at 96 kHz, the sound of non-linear plugs like compressors will be much better, purer, more analog-like. I'll leave it up to you whether you think at that point they are the sonic equal of their analog counterparts.
In that case, if you're working at 96 kHz, and not trying to push those comps beyond their mortal means, that is, using them subtly, 1-2 dB GR, to slightly "fatten" or pump" up a sound, or subtly control an overdynamic instrument, I believe that a number of the plugs, for example, like the Waves Ren Comp, could do a reasonable job on background things and who knows, maybe even lead vocals! It's only when you're trying to "punch" and "pump" and the rest---that's where the rubber meets the road and I still believe that good outboard gear adds that sheen, perhaps because of the unique distortion of certain analog compressors, which has not yet been emulated in digital-land. Or it might be something to do with the time constants. I certainly love my Weiss digital compressor and I can even make it work
aggressively. I was able to add some "attitude" to a rock mix just today. Working at 96K and with the Weiss, I personally was very happy with the sound I got today punching up a rock project. So digital comps have come a long way.
You'd have to show me a compressor plugin that at 96 kHz is as versatile and good-sounding as the Weiss...it may exist, as I say, I do not have any comps outside of the excellent Waves stuff.
I'm not that current with the sound quality of the best digital compressor plugins, but of the ones I have access to, I still believe there are still some sonic tricks which cannot yet match (to my ears) the sound of some analog compressors. And I do love my Cranesong Trakkers, which I still think have a few tricks (including that "analog sound", the additional fattening which those rich harmonics can do) that I think even my Weiss cannot do.
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Is there same differences by using analog eq in insert instead of eq plugins?
For example how is better using an original API 550b in a PT insert slot instead of an URS API replication?
Thanx
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There are some superb digital equalizer plugins. I have tested the GML in the TC Electronic system 6000 and if the TDM equivalent sounds as good I'd use it any day of the week. I'm very fond of the Algorithmix when I'm going for transparency. But if you are looking for particular color or edge there's nothing digital that sounds like a genuine API 550 or many other well-known analog pieces.
BK
There are two kinds of fools,
One says-this is old and therefore good.
The other says-this is new and therefore better."
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67390 is a reply to message #67233 ] |
Sat, 21 May 2005 19:09   |
Paul Frindle Messages: 372 Registered: May 2004 |
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[quote title=bobkatz wrote on Sat, 21 May 2005 01:33]| innesireinar wrote on Fri, 20 May 2005 08:43 |
bobkatz wrote on Mon, 09 May 2005 21:28
stay away from (too many) digital compressors
BK[/quote |
Bob, can you clarify this?
By using compressor plugins instead of true analog compressors instatiation is there a big difference in a mix done ITB?
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The problem is that to do a compressor plugin right requires tremendous CPU power, more than the manufacturers of plugins believe you are willing to tolerate with today's CPU's. In due time all digital compressors will be 2 to 4 times oversampled internally. One way around this is to track and work at 96 kHz, the sound of non-linear plugs like compressors will be much better, purer, more analog-like. I'll leave it up to you whether you think at that point they are the sonic equal of their analog counterparts.
BK
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Actually it's not really a question of processing power, it is perfectly possible to make a good compressor within the bounds of the processing people have these days. And it's not even a function of sampling rates really either - despite what people believe. Such good compressors actually exist IMO.
The biggest challenge facing comp/limiter design at the moment is actually conflicting user expectation and the current 'level madness' paradigm along with a complete misunderstanding of sampling and the somewhat hidden limits of their existing systems. If you will - one is trying to satisfying technically conflicting expectations which CANNOT be removed by any possible means - cos the spin has etched itself on everyone's minds too deeply - and in any case they have no choice in the current production fashion, if they want to stay in business.
As you may imagine people working to the red light that comes on within one part in 2^16 of max sample value with an operating level that is only one in 2^16 less than this is somewhat limiting when it comes to doing 'anything creative' with the design. In fact it makes it extremely difficult to treat their programme as signal at all - cos they themselves are not being allowed to. To add to this, ANY loss of displayed sample value level that people may encounter due to 'corrective processing' is highly unwelcome to the user who is always aiming for max values on a meter that reads only sample value and whose livelihood rests on 0.5dB loudness relative to other people's mixes. IMHO there are awful paradoxes in the way people are being directed to perceive audio which is preventing them from ever realising the quality of the plugs they already have.
To give just one simple example - I have actually been asked by users why a certain plug-in limiter set to -0.1dB doesn't always put out their 'red light'. As you would appreciate 'putting out the red light' this way requires that your limiter operates on sample values too, rather than the real signal it should process (re, matters raised in this thread). Even asking people to reduce their threshold to -0.5dB causes them anguish because of the much-feared career threatening potential of level loss.
What the above dichotomy risks doing is forcing comp/limiter plugs to adopt the same misunderstood principles that are already killing their sound within their existing digital environments. Put simply if you are not very careful and very clever, the above example REQUIRES that your limiter works in the exactly same way as their sample value workstation meters that do not measure signal. Not a good basis to start designing a compressor you would agree?
Ok, but here's the real rub - analogue outboard gear is let off the hook totally. It is outside the system of the wrong but highly critical digital metering - and gets put through a reconstructing ADC before you get to monitor it digitally anyway. Because it's analogue everyone naturally forgives it's inaccuracies and unpredictability. Because it's analogue any foibles it produces are referred to as 'character' - i.e. the horse isn't crazy, it's just spirited. Because it's analogue it always works on actual legitimate real signal. And of course it has all the headroom it has already carved out for itself, in total disregard for the machinations of the challenged digital user paradigm it gets connected to.
As people might imagine - increasing sample rates can never redress these problems - that's an open ended quest of diminishing returns which has potentially open ended costs for the user - that is doing nothing more than 'pasting over' far more simple issues that essentially cost nothing to fix - other than a bit of honesty and a change of attitudes within the industry.
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67456 is a reply to message #67431 ] |
Sun, 22 May 2005 07:56   |
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bobkatz Messages: 2926 Registered: June 2004 Location: Orlando |
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| innesireinar wrote on Sun, 22 May 2005 03:41 |
Bob,
Having said that, I've stated from your words that the goal could be the exact opposite: mixing ITB by using as many good analog outboard (inserted) as we can for processing. Expecially for dynamics. And assuming that digital processings ITB or OTB are pratically the same since a good algorithm works as fine on an outbord DSP as on an accel card (reverbs), the best solution could be having a good DAW, many good AD-DA and tons of good analog gears for eq and dynamic. Better than the scenario I've described at the beginning of this replay.
I'm right?
This thread has become very interesting.
ranieri senni
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That's my personal position. I've received numerous mixes. Some excellent mixes done both in the box (totally digitally but with lots of good hardware outboard through D/A/D patches) and some plugins used conservatively. Some excellent mixes done in analog consoles. My preference, about 7 times out of 10, has been for the former, that is, the digital mixes with considerable analog outboard. I find that I don't personally like the veil that comes from sending EVERY TRACK D/A/D through the analog console. But THREE TIMES OUT OF TEN I prefer the all-analog mix because the coloration is just what the doctor ordered.
It is impossible to be more "scientific" than this, as this is experience-based. All things are NOT equal. I've received mixes from so many different mix engineers that all I can do is "average my responses".
But the bottom line is that I personally suggest you try mixing in the box or with a good digital mixing console (for the ergonomics, of course, there's nothing more frustrating than mixing with a mouse), and with lots of good analog outboard. And excellent converters with good clocking. Absolutely essential to making any of these ingredients work. Work at 48 or 96 kHz sample rate, that helps too!
Another key here is: WHAT WORKS FOR YOU. Each engineer's methods and how aggressively he or she runs their controls can affect your results. I know some engineers who ONLY use the equalizers and compressors in a Yamaha O1V and get good results. But the vast majority of these engineers are running these digital tools conservatively and with largely acoustic music recorded in good rooms with stereo miking techniques. If you were trying to get a "rock and roll attitude" with one of those O1Vs you'd fall flat on your face. So, in the end, you have to find the method that suits your madness!
BK
There are two kinds of fools,
One says-this is old and therefore good.
The other says-this is new and therefore better."
No trees were killed in the sending of this message. However a large number of
electrons were terribly inconvenienced.
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