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 #67485 is a reply to message #67484 ] |
Sun, 22 May 2005 10:51   |
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 bobkatz
Messages: 2926
Registered: June 2004
Location: Orlando |
Platinum Member |
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| innesireinar wrote on Sun, 22 May 2005 11:43 |
Thank you Bob.
It's easy to understand that working at 96 it's better, but why at 48? Is there a particolar reason why 48 is better than 44.1?
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With A REAL GREAT A/D the difference between 44 and 48 is extremely small, perhaps inaudible. This is because of the quality of the low pass filtering and other aspects of the design. An exceptional low-pass filter would have thousandths of a dB of ripple in the passband or less, be calculated at very high precision and dithered cleanly to 24 bits. You only get that in a "roll your own A/D" and the number of manufacturers actually rolling their own filters may be counted on a couple of fingers of one hand!
With a medium class A/D, it helps to get the frequency of the low pass filter up there even a few kHz more, just enough so there is less interference with the audible band due to phase shift, possible preecho, and so on. So, I tend to recommend the higher sampling rates "just to be safe." In most cases, coming from the mid-class converters that most of my clients can afford, I find that work I get in at 48 K and above sounds more "open" and a bit "clearer" than work I get at 44.1K. This is a GENERAL statement, not a rule, just an average. How can you compare things which are not equal anyway 
But hell, some of it might be my D/A! A recent improvement in an experimental filter that is being used in a DAC that I can't talk about yet has ALMOST levelled the playing field in that 96K sounds very very close to 44.1 on this DAC. While previous models of this DAC the difference was much larger. We have so much to learn. And 44.1 kHz can sound much better than most of us have had the opportunity to hear; I feel privileged to have heard the first model of a new DAC whose dimensionality and clarity at 44.1K is significantly better.
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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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67486 is a reply to message #67484 ] |
Sun, 22 May 2005 10:55 |
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bobkatz
Messages: 2926
Registered: June 2004
Location: Orlando |
Platinum Member |
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| innesireinar wrote on Sun, 22 May 2005 11:43 |
I would like to know your point of view how are good rev plugins compared to hi-end outboard rev. Can rev plugs like Altiverb, Rewibe, TL space and also hi level no-convolution rev compete to, let's say, a System 6000?
Thanx
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The convolution reverbs are coming MUCH closer to the quality of the System 6000 than any previous reverb plugin of "standard technology". But the low level resolution and versatility, dimensionality and lastly, the ability of the 6000 to do accurate early reflections still make it the connoisseur's reverb. It used to be intuitively obvious to the most casual observer why a plugin sounded like cheese and the 6000 sounds like the real thing. Now it takes a bit more sophisticated listener to know the difference.
Anyway, now I wouldn't kick a convolution reverb out of bed and with care it can produce superb results. In fact, for convenience, I use the IR1 a lot and I do praise it. The last 50 ms of decay though, leave something to be desired, as do the early reflections. We're talking B+ as opposed to A+ grades here.
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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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67552 is a reply to message #67485 ] |
Sun, 22 May 2005 17:36 |
Paul Frindle
Messages: 360
Registered: May 2004 |
Active Member |
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| bobkatz wrote on Sun, 22 May 2005 16:51 |
| innesireinar wrote on Sun, 22 May 2005 11:43 |
Thank you Bob.
It's easy to understand that working at 96 it's better, but why at 48? Is there a particolar reason why 48 is better than 44.1?
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With A REAL GREAT A/D the difference between 44 and 48 is extremely small, perhaps inaudible. This is because of the quality of the low pass filtering and other aspects of the design. An exceptional low-pass filter would have thousandths of a dB of ripple in the passband or less, be calculated at very high precision and dithered cleanly to 24 bits. You only get that in a "roll your own A/D" and the number of manufacturers actually rolling their own filters may be counted on a couple of fingers of one hand!
<snip>
BK
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I would agree with all you have said here It's the transition bandwidth that counts - the difference between the highest wanted freq and half the sample rate.
In reality the nyquist freq of 44.1K sampling (22.05KHz) is too close to the wanted 20KHz band for comfort and it requires a pretty hairy filter to roll-off sufficiently to almost nothing within just 2KHz, without causing any abberations in the wanted band. It can obviously be done, but may not be provided in some designs. 48KHz is much better because the requirement for the filter is much more reasonable.
For the 'roll your own' designs we made with a separate processor for the filter - with our best efforts the performance at 44.1K was indistinguishable from the input signal in all the A,B tests I could do. However I use this system daily still and I fancy that there have been odd moments over the last 10 years when I just may have heard a suspicion of difference that was not there at 48KHz, for which the designs were optimised.
IMO a sample rate of around 60KHz or so would provide the optimal answer as a trade off between converter convenience and processing loss costs to the user - providing we could agree on flatness to 20KHz (and not higher).
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67562 is a reply to message #67556 ] |
Sun, 22 May 2005 18:17 |
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| Paul Frindle wrote on Sun, 22 May 2005 18:47 |
24bit digital audio has 144dB or so total dynamic range - so you can easily provide more than 30dB of this kind of headroom before the digital signal noise becomes significant in respect of the DAC SNR.
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Paul,
Would you then say we could record (in a good quality DAW with good converters) at, say, 12 dB below "red light?" Would there be any other trade off penalty (something like "using all the bits" which some people talk about)?
If this level is ok, then most people are digitally recording about 12 dB "too hot" most of the time (assuming a proper, non-sample based meter). This lower level would certainly negate many of the digital/plugin overload problem!
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67613 is a reply to message #67583 ] |
Mon, 23 May 2005 02:18 |
Andy Simpson
Messages: 583
Registered: July 2004
Location: Poland |
Gold Member |
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Not to mention the advantages of being able to run all the faders between -10 and +10, where all the (log) fader resolution and control lies, digital or analogue.
I think alot of digital mixes suffer simply because when each channel is coming in at -6 or above, and on a 24 track mix, the faders need to be set very low. Obviously, the levels can't be set with compelete intuitiveness when the fader is at the bottom ranges of its travel, where each small movement means a large volume change, and the 'perfect' level is impossible to locate, especially with a mouse! 
Headroom headroom headroom headroom, resolution, headroom.
Andy
PS. I also attribute modern music's lack of impact to the fact that on consumer systems (and even pro) it is impossible to get any fine control over playback levels, and therefore impossible to get the perfect level for listening. Too loud or too soft.
Simpson High Resolution Microphones, Poland
www.SimpsonMicrophones.com ** Now updated with faster download MP3 samples **
Orchestra mp3 - Piano mp3 - Gregorian Choir mp3 - String Quartet mp3 - Drum overheads mp3 - Tabla mp3 - Sitar mp3
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67625 is a reply to message #67562 ] |
Mon, 23 May 2005 03:51 |
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bobkatz
Messages: 2926
Registered: June 2004
Location: Orlando |
Platinum Member |
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| compasspnt wrote on Sun, 22 May 2005 19:17 |
| Paul Frindle wrote on Sun, 22 May 2005 18:47 |
24bit digital audio has 144dB or so total dynamic range - so you can easily provide more than 30dB of this kind of headroom before the digital signal noise becomes significant in respect of the DAC SNR.
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Paul,
Would you then say we could record (in a good quality DAW with good converters) at, say, 12 dB below "red light?" Would there be any other trade off penalty (something like "using all the bits" which some people talk about)?
If this level is ok, then most people are digitally recording about 12 dB "too hot" most of the time (assuming a proper, non-sample based meter). This lower level would certainly negate many of the digital/plugin overload problem!
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If you work to an RMS level of -20 dBFS throughout the board you'll automatically not have this problem. I can't think of a single pathological signal I've encountered that would break this rule.
Remember that signal to noise ratio is perceived by us largely in RMS terms, not in peak terms, so it is facetious to be normalizing everything to the same peak level, and the signal to noise ratio of a 24 bit digital system will not perceptibly be deteriorated if you work your entire mix system to an RMS of -20 dBFS. Leave the loudness maximizing to the mastering experts. Of course this does not mean that you shouldn't use compressors to get your "sound", but what it does mean is not to get uptight about having to peak everything to full scale peak when the perceived RMS has already crept up above -20 dBFS due to all the compression you may have applied.
My system of using a calibrated monitor attenuator is designed to replace the old systems we used in the days of analog of having VU meters. Since the VU meter has disappeared, we have to find another way to help keep us out of trouble. Stoppig everyone from normalizing the peaks is the first piece of education we have to work on. And the best tool to replace the old system can and should be OUR EARS. If our ears tell us it's too loud and if we back off, your levels will automatically fall back into the safe zone.
All you have to do is understand the words "RP 200", which has to do with calibrated monitor attenuation. I firmly believe that the more people who start to learn how to use a calibrated monitor attenuator will keep themselves out of trouble. But if you can't wrap yourselves around the calibrated monitor control concept, at the very least keep a set of VU or RMS meters around which are calibrated to -20 dBFS with sinewave.
Hey, I could write a book about this
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.
|
|
|
| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67628 is a reply to message #67562 ] |
Mon, 23 May 2005 05:14 |
Paul Frindle
Messages: 360
Registered: May 2004 |
Active Member |
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| compasspnt wrote on Mon, 23 May 2005 00:17 |
| Paul Frindle wrote on Sun, 22 May 2005 18:47 |
24bit digital audio has 144dB or so total dynamic range - so you can easily provide more than 30dB of this kind of headroom before the digital signal noise becomes significant in respect of the DAC SNR.
|
Paul,
Would you then say we could record (in a good quality DAW with good converters) at, say, 12 dB below "red light?" Would there be any other trade off penalty (something like "using all the bits" which some people talk about)?
If this level is ok, then most people are digitally recording about 12 dB "too hot" most of the time (assuming a proper, non-sample based meter). This lower level would certainly negate many of the digital/plugin overload problem!
|
Yes in essence this is the point. But it isn't a good idea to lose too much gain in the input stages of the recording ADC since you will lose SNR. It is perfectly permissable to record in the first instance at relative high levels (peaking around -3dBFS) because an illegal signal should not come out of an AD converter (as we have said here).
The place where you need to make the gain loss to get headroom and avoid unreported overs is in the DIGITAL domain - right at the start of your mixing channel. In this way you preserve the converter's SNR during recording and optimise SNR and headroom for the whole system - making maximum use of the 144dB SNR the digital domain offers. The only provisor is that you will need to run good quality plugs that are not noisy and function correctly at lower reference levels.
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| Re: DAW & Desks: Is ANYBODY actually still mixing on their desk? [message #67639 is a reply to message #67519 ] |
Mon, 23 May 2005 07:29 |
ted nightshade
Messages: 1272
Registered: April 2004
Location: Southern Oregon, US of A |
Platinum Member |
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| Bob Olhsson wrote on Sun, 22 May 2005 12:26 |
What I'm trying to say is that tube consoles had an astounding dynamic range for something like a kick drum. You could just pull the fader back on that channel without padding the preamp input. I haven't found outboard tube preamps or small tube mixers to have this "effortless" quality and they indeed do compress. I can only assume the limitation is the power supply. The "vintage" tube sound people hear on old recordings was not very compressed at all. I didn't completely appreciate this until I used Deane Jensen's prototype servo mike pre around 1985 which also had a whopping dynamic range.
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Yes, could be the power supply. My beautiful little xformerless tube mixer (thanks David Manley) has definite headroom limits, lower than you might expect. I spend a fair amount of time dialing in the choice level just short of audible distortion, working with the headroom limitations in an artistic way. But I'd happily trade for beacoup headroom forever.
But, now that I think of it, preventing distortion is easy on any input signal I can provide, just by rolling back the gain on the rotary fader. So is that any different than life on the old tube consoles?
Ted Nightshade aka Cowan
There's a sex industry too.
Or maybe you prefer home cookin'?
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