[Bob Katz]

Thanks, RBJ!

[Bob Katz]

Thanks, Norman.

[Bob Katz]

Norman’s written a blog about this subject, here

[Bob Katz]

Good to know, Norm. What about a series of diffusers along the back wall of a control room? Is there any advantage or disadvantage of using number theory diffusers in the line of incidence and switching to, say, polys at the sides?

[Bob Katz]

For me, debunking the stairstep myth is an old story, so the guy is preaching to the choir. Hopefully, anyone trained in basic digital audio principles has already learned that after the low pass filter, there is no stairstep, period. You can recover a sine way just below the Nyquist frequency, say, 20 kHz, just as easily as you can recover a 1 kHz sine wave. End of story.

[Bob Katz]

I went on vacation to Italy and then Finland for over 20 days and did not visit the forum once. Until I went away I would singlehandely always give a bit of promotion for the forum on my Facebook page, and it brought lot of visitors. The Board is going to get together shortly and discuss means of promotion for this excellent resource. In the meantime I suggest everyone who believes in the power of deep dialogue here at digido fora to put in a good word for it on your Facebook page! Thanks!

[Bob Katz]

Thanks, Phil, for your suggestion. The thing about “non-Eucontrol” controllers is they can’t keep up with the speed and additional features. Avid basically owns the controller universe. As far as posting pictures, I  can add an image from my iphone or desktop by clicking on the image icon in the toolbar that appears above every posting.

[Bob Katz]

I don’t think you can get around Rosetta other than to use Silicon-engineered plugins. Perhaps you are referring to DDMF plugins, ddmf.eu,  which make a “meta plugin” but to my knowledge it does not get around Rosetta and won’t load Intel-based AAF plugins.

[Bob Katz]

Hi, RBJ. Your idea of using multiple ADCs, each one scaled differently, of course has merit. It could allow perhaps 30 dB of additional headroom over 10 volts, if that were ever needed. But would a cheap company like Zoom implement multiple ADC chips instead of bamboozling their customers into thinking they have infinite headroom? I doubt it. Bamboozlement in progress.

[Bob Katz]

Dear Ferenc: Bruno Putseyz’s explanation in this thread says it all. EVERY ADC has an overload point, no matter how you slice it. EVERY ADC has to output in fixed point, because we’re talking about converting from the real (analog) world, which has defined analog voltages. You can CONVERT (or the device can convert) the output of the ADC to float, but it still had a defined overload point to begin with.

What Zoom has to be doing (there is no other way) is to move the 0 dBFS point to BELOW the overload point of the ADC. They can set it up so that a slightly higher analog voltage than is commonly used would cause the oveload. But that doesn’t make the ADC immune to overload, it just postpones the overload, at the expense of a worse signal to noise ratio. But as Bruno points out, the SNR of a good preamp and ADC is quite good, so it’s not a terrible thing to sacrifice a little SNR. But Zoom is misleading the customer and the article is entirely misleading. You don’t even need floating point to perform this “miracle”, actually. Just attenuate the input to the ADC inside the recording device.

[Bob Katz]

Another way I’ve made a “reference” at times without needing a dedicated reference speaker is to stand just outside the studio door and listened to the master (or mix). Also, playing things softly can be very revealing: What sort of details do you miss? What details remain salient?

I will turn down a questionable soft passage in the monitor, to see if the vocal still comes through and you can understand the lyrics.

[Bob Katz]

I agree. By the mid 90’s, dedicated hardware mastering systems fell to disuse. I don’t know anyone who has been using a Roland hardware workstation or an Alesis Masterlink or a DAT machine anymore. And especially PCM-1630. Exabyte DDP tape. Those are all gone, the all-purpose computer has taken over.

[Bob Katz]

I had a Millennia NSEQ-2. It’s a terrific equalizer with either a transparent solid state or nice tube tone, but for me the fine resolution of the boost and cut knobs was not fine enough for my mastering needs. But it definitely would make an excellent equalizer for mixing.

[Bob Katz]

Faulkner’s setup reminds me of the late David Hancock’s approach to spacing a pair of figure 8 ribbons. I haven’t tried it, but the idea seems dangerous as coincident figure 8’s already have tremendous separation. But David told me that he called coincident pairs “monereo”. And since I have frequently supplemented Blumlein with spaced omnis I’m no purist either 🙂

[Bob Katz]

Hi, Phil. There was another advantage to the way that you overdubbed your voice. Not specifically because of the MS, but important. You used the identical gain for each overdub. As a result, the same perspective of the stereo miking integrates them all.

Reminds me of when I recorded Stravinsky’s L’Histoire Du Soldat in two parts and mixed them later in post. Part one was the instruments and part two the narration and the acting. I insisted to the producer that we keep the same Blumlein pair mike up (An AKG C24) for the narration, at the identical gain used for the instruments. I positioned the actors in front of the mike at positions that sounded good — in the same hall. As a result, everything just flowed together in post. What happens if you decide to record the narration/acting at a different time, is you get a discontinuity, the ambience and the perspective changes. But instead, we got a continuous presentation that makes it sound precisely like the actors are there on stage with the musicians.