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I'm reasonably sure that Dice chip is handing the digital-audio-to-and-from-the-computer part. 

I don't quite know how willing TC might be to offer (that much) support to Joe Random off the street, but i guess it can't hurt to ask? :)
Just don't be expecting them to offer you a ready-made firmware, customized for THAT particular device, etc. This latter part was either done by programmers at A&H, or perhaps they just contracted that particular job out to TC themselves.

Not sure what generation Dice is in that Zed mixer, but the Dice II at least, can definitely do 192k, in certain interfaces, depending on the firmware. For example, my TC Konnekt 48 can do 192k, but the Focusrite Saffire Pro 40 / 24 can't, despite using the same MCU.

PS: For the record, or at least in my humble opinion, "AD/DA" isn't necessarily the same thing as "(computer) audio interface" - some units decidedly aren't :) I'm thinking of stuff like the Focusrite Octopre, Presonus' various Digimax'es, the venerable(?) Behringer ADA-8000/8200. All AD/DA's, but not interfaces (no computer connectivity).
 
Quite often, increasing the SR involves some changes in the hardware "glue", like a different crystal.
An.d BTW, going from DS to QS does not bring a level of improvement comparable to going from single to DS. In most cases, the frequency response goes from about 44k for 96kSR to less that 70kHz for 192kSR, while the S/N and THD hardly change (typically due to the limitations of the analog path). I've never been convinced of an undisputable superiority by any demonstration of QS or DSD. Intended for audiophools IMO.
 
+1

there are plenty of arguments for the very diminishing returns of 192K out there if you google around. Working in a large professional studio for almost a decade now, I can't stand it. I often cut vinyl from hi res 192K files and its just plain inconvenient for me...And silly. I'm of the belief it actually is more degenerative to a typical audio system than beneficial. Although I haven't proven that to myself with tests.

I have talked about this before. I have done extensive blind A/B tests between sample rates and bit rates in this studio with the house engineers and NOBODY could tell the difference between any bit rate or sample rate above 16bit 44.1K listening to stereo files.

Multi track recording is another story but I still think 96K is pretty optimum in that realm. And only if your working in the box with a lot of plugins.
Just think about how many awesome sounding records were recorded on those Digidesign 888's in the 90's.

I was actually in one of the rooms when Neil Young was here geeking out over the 192K thing and I had to just stand in the back of the room subtly shaking my head... I mean I love Neil Young, but considering his age and the amount of time hes spent standing next to live crashing cymbals and guitar amps, there's no way he's hearing above 10K...In fact he would probably enjoy 16bit 24K audio. A nice steep filter at 12K for a little distortion right at the edge of his hearing. Known as "air" to audiophools ;D
 
Things would be a lot less complicated today if the original digital audio format was 20 bit / 64 kHz (or was it 60 kHz?) as some had proposed back in the day.

I've done my own abx tests and could reliably tell the difference between 44.1, 48 and 96k. It seems converter dependent though,  each one seems to have its own sweet spot.  192 was not necessarily an improvement over 96, some ways better,  some worse. I mostly use 48k, as a compromise between sonics and resources.

Regarding the original question,  my guess it is probably not an easy fix to get 192. If it was it probably would have been a feature to begin with.  Mostly just speculation on my part though.
 
I've done it a few different ways over the years generally with foobar. I've wondered if there is actual a totally correct way since everything introduces some variables.

I've compared commercial tracks that were available in HD,  high rez bought from HD tracks vs CD version.

I've recorded raw material at different rates to compare, using the same source material.

I've taken 96k recordings,  SRC down,  compare.  Then also SRC back up to play back both at 96k.

I think so much is probably the filters,  maybe with a top shelf prism it won't matter as much,  best I've used is lynx aurora and d/a in dangerous dbox .

Interesting side note,  I remember a shootout where a song was tracked / mixed twice at 96k and 44.1k, at the end the 96k was converted to 44.1k. Even there I thought starting at 96k sounded better.
 
80hinhiding said:
96Khz seems to be doing a decent job of preserving a signal that comes off of tape.  I figured 192Khz might do even better..

But that's the thing, there is nothing to capture from tape with 192K. If you have ever done a sweep on a tape machine (nice Studer or Ampex) into the record head and out of the play head while rolling tape, you will see how the signal dives aggressively downward between 20K and 30K. So technically 88.2K sampling rate could capture all audible bandwidth tape is capable of reproducing. Not to mention the limitation of the tape deck play electronics. What do you suppose your capturing up at 96K (192K sample rate)? I would say if anything you would be capturing ultrasonic noise that you would probably be better off not replaying through a console, amp, and speakers.

In the real world of mastering, Some times applying steep filters at 24Hz and 18-19K does something nice to audio, I know there might be resonant peaks at the extremes but it can be pleasing. For some reason in peoples minds " openness" and "air" conjures up wide bandwidth all the way out to 100K baby! When in reality a wide 2db bump at 10K is what produces air and openness.

Anyhow if recording at 192K makes you "feel" better than do it. You are absolutely right about music and feel. In the end, feeling is most important.
 
john12ax7 said:
I've done it a few different ways over the years generally with foobar. I've wondered if there is actual a totally correct way since everything introduces some variables.

Yes, the way we did it:

We had the original 1/2" Master tape of Norah Jones's first album. Very nice recording with lots of dynamics. We had an Studer and Ampex ATR machine in tip top shape. We did test a lot of different converters, but again no one could tell the difference.

So  using the stock Digi 192's we would transfer a song into pro tools at 192K and 44.1K.  into two different sessions.

The tape had calibration tones that would get transferred into pro tools as well. That was key, because it allowed perfect volume matching in play back. Like if we used two different converter to transfer the tape into pro tools, they might have been a little off in volume but with the tones we could make sure the outputs matched perfectly.

We would import the 44.1K file into the 192K session. The 44.1K would be routed to a digi 192 output 1+2 and the 192K file routed to the same digi 192 output 3+4.

Those outputs would go into a relay switched A/B box. Two stereo inputs switchable to one stereo output. The hand held control unit had LED's to show which input was active to the output. Output of the A/B box went straight to a Tape in on the console.

SO people could switch back and forth between the two sample rate files and see which file they were Listening to via the LEDs on the remote switch box.

Remember the outputs of the Digi192 were perfectly volume matched because we had the tones recorded on each file before the song played.

It was funny to see the engineers come in, start switching back and forth looking at the LED's going "oh yea I can hear the difference"

BUT THEN:

we turn out the LED's and press the switch button quickly numerous times randomizing the outputs. Then watch the engineers switch back and forth trying to determine which file was which. When they settled on a file and said "thats the 44.1K" we turned on the LED's an saw if they were right or wrong.

So we did 20 guesses and marked down the the right and wrong answers. I think the best guesses were 12 or 13 out of 20 being right. TOTALLY RANDOM.

There were other versions of the test done, two different rigs with two sample rate sessions, different transfer A/D's etc.
But when carefully volume matched and a tally of right and wrong answers was produced, no one could pick out anything. At least four engineers from new guys to 20 year vets took part. In the very room they had been working in for years.

This only applies to two track stereo audio. I have not done tests with multi track sessions and probably never will. It was hard enough to set up the two track tests.

So I'm just always skeptical when people say they can hear a difference. John perhaps you can, but I always try to share my experience with the A/B tests I did, so people know just how hard it is. And maybe relax and just enjoy that pristine 16bit 44.1K audio coming from they're favorite CD. ;)

 
In your test isn't the 44.1k then getting upconverted at the output since it's only one digi 192?

If someone consistently gets 12 or 13 out of 20 that is actually not random but statistically significant.
 
Lots of hard core tape guys seem to  gravitate towards DSD,  so maybe look into that if it's an option for your tape transfers.

From a technical perspective there are good reasons to go above 44.1k. It's not necessarily about capturing above 20k, but rather increasing the performance below 20k. The 10% margin,  while once maybe being necessary,  is in general poor engineering practice.  2-5x bandwidth would be more reasonable.  So if tape is 30k bandwidth, it may well be necessary to go above 96k samples rates to accurately capture that 30k bandwidth
 
I miss tape.  There is something special about listening to something that has never been digitized. Now you've got me thinking about buying a machine again.  It just became impractical at a certain point.  But I also feel it's much harder to make a great album with Pro Tools.

Other modern day options are Burl converters,  supposedly more tape like sound.  I have a pair of Neve 542 tape modules for 500 series.  Gives a bit of the tape head saturation.

Maybe I'll give 192k another spin too.
 
bluebird said:
+1

there are plenty of arguments for the very diminishing returns of 192K out there if you google around.

The only use of 192K IMHO is in test equipment so you can check analogue equipment is behaving itself above the audio band,

Cheers

Ian
 
JohnRoberts said:
I can't hear 192kHz....  I can't even hear 96kHz.  8)
Neither can i, but I can hear the difference in transient response between SS and DS; that I can't hear between DS and QS.
Somewhat converter dependant, but not that much IMO, since most converter chips use the same basic algos for anti-alias and decimation filters.
Applies to tracking. Now for mixing, I'm perfectly content with 44.1/16, as long as the mix is consistent. I may have a different opinion if I was doing classical recording, although I doubt many people have a listening environment allowing to enjoy 24bit resolution.
 
ruffrecords said:
The only use of 192K IMHO is in test equipment so you can check analogue equipment is behaving itself above the audio band,
+1. Finding a converter that does actually gors above 60-something kHz is rather hard to find. Most of the chips' internal anti-alias/reconstruction filter seem to cut at 0.35SR at 192k.
 
john12ax7 said:
I miss tape.
I don't at all. Having fought against tape noise and distortion for decades, I'm happy with a medium that gives out what it takes in. If I want "colour", I add it in a way I can control. I've tried several of these tape sound plug-ins, it seems they do nothing till they do too much.
 
abbey road d enfer said:
Neither can i, but I can hear the difference in transient response between SS and DS; that I can't hear between DS and QS.
not sure what that alphabet soup of letters is, but that's OK I will shut up soon.
Somewhat converter dependant, but not that much IMO, since most converter chips use the same basic algos for anti-alias and decimation filters.
Back last century I did a lot of work with dynamics processors. If you think the transient response of A/D conversion is difficult, try compressors (but I'm sure you have done that too). Even power amplifier design requires rate of change consideration.

The million dollar question surrounding any discussion of transient response is "in what bandwidth". Back when I was active in audio design (I'm not anymore) I was a strong advocate for LPF any audio path as early as possible to prevent shenanigans from out of band signals that even analog electronics sometimes stumbles over.

Dangling a key chain in front of a mic would often reveal weak sisters who couldn't handle the 50kHz + stress.

=========

My apologies to the OP... It doesn't matter what I can or can't hear... do what floats your boat.

JR

PS: Perhaps more appropriate how many here can actually hear a (decent ) LPF at 30-50kHz....  I recall some long and passionate discussions on Geeksluts about this bandwidth issue years ago with little resolution.

 
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