Preamp disablement

[Bruce Graham]

Hey Y-MY-R;
Thanks for the detailed explaination. Once again I have learned more! (I am begining to think I have forgotten more than I learned ).

Cheers
Bruce

[Old School]

Hi all,
Just want to take this opportunity to take this discussion even further off topic. I believe that frequencies over 20k, even though we can't hear THEM, interact with and shape the way we perceive the frequencies that we can hear. Just consider how you cannot see light in the ultraviolet range, but put on a pair of uv blocking glasses and your visual perception is greatly changed. I think that there was a good reason that designers of some of the most legendary consoles insisted that the channel strips pass signals up to 50k. I'm not insisting this is true, but certainly worth considering. Since higher sampling means higher frequencies, the only fair blind test would be to obtain music mixed on equipment capable of recording these frequencies and making masters at each sampling rate. A tall order in these times when much of the equipment available obtains good signal to noise ratios by rolling off the top end with a built in filter to make up for substandard circuitry. And you might say that its a moot point as no commercial recording contains info over 20 Khz, and our speakers don't reproduce it. So please read https://www.cco.caltech.edu/~boyk/spectra/spectra.htm and pay particular attention to paragraph X.

[Y-my-R]

Thanks Bruce and Old School!

A couple of things to note, before I type another word:

I am not an authority on any of the stuff I share my opinions on, on here. While I had some insights from working in the MI industry (but in non-engineering positions where scientific backgrounds are not usually part of the information I received), in the end I’m just a hobbyist/amateur musician and “recordist” who was influenced by stuff I read from reliable, but also completely unreliable sources. I’m not immune to (wrong) outside influences or even hyperbole, as much as I try to filter that out for myself.
I am, however, always happy to change my mind and opinion, when presented with information that is compelling and seems to make sense (at least to me - but if something comes with a great set of scientifically collected background information like here, that’s a best-case scenario, really!).

I do like a good exchange of information on topics such as this, as I always strive to get a better and more complete understanding of how audio (in various formats from analog to digital at it’s many rates and bandwidths or conversion methods, etc.) works, and what the best practices are to get the best results in my personal attempts to apply what I learned.

Having said all that, for MYSELF, in the end it’s always my ears (well… and some types of meters and value read-outs from gear the audio passes through) that will make the decision for ME, if a change in approach is worth it for ME (e.g. recording at high sample rates).

For me PERSONALLY, I came to the conclusion that recording at higher sample rates does not bring ME any benefits.

There’s a few caveats to that, though:
- Since the idea of creating/producing music is to share it with others, there likely would be SOME who could (at least subconsciously) perceive a difference in audio quality, if given the opportunity to A/B the material in an otherwise neutral setting. I understand it, if this argument alone compels some engineers to work with such technology (even for the case where they may not really (consciously) hear the difference - like, how I don’t hear it myself, for example).
For me personally, my opinion is that if I can’t “control” what’s going on in high frequency ranges because I can’t hear it, then I’d rather not have it be present since it could make things worse.
Let’s say I have tons of disharmonic overtones above 20 kHz that I PERSONALLY can’t hear, but that may make others with a finer perception cringe…? If I’d know, maybe I could treat that frequency range to leave primarily harmonic overtones… but if I don’t hear it, I can’t treat it in a meaningful way… and I don’t want to let loose on others, what I can’t judge and control myself… So, along this philosophy, I usually roll of the low end and often also roll off the high-end - sometimes even on the cymbals (i.e. “high energy above 20 kHz” per that article/paper).
- I mostly work on rock and (even “extreme”) metal, and the average listener in this area, isn’t TOO discerning from my experience, hahaha. (Even though some people in that crowd “think” they have golden ears - and maybe some do - but those who I know think they do, clearly don’t, hahaha (example: I know someone with a high-end Hi-Fi system with dumb stuff like a $100+ S/PDIF cable with gold connectors and custom built amps and speakers, who has listening sessions with his “Audiophile” buddies to criticize (or mostly praise) if/how much of a difference a new expensive cable makes… and at least that one guy I’m thinking of right now, doesn’t hear SH*T (e.g. not-so-subtle changes in mix revisions completely fly by the guy and I have to point them out. He just likes to talk like a snob and be “part of the cool crowd” I think).

As for the article - I ended up reading much of it since I found it quite interesting (skipping through to some extent, but probably read over 50%). I completely agree, that audio at the source doesn’t “end” at 20 kHz, and that instruments very much create frequencies in higher ranges (obviously). I also don’t doubt, that these frequencies could contribute to the over-all experience of how the audio/music is perceived by a listener (e.g. even indirectly, such as through vibrations in the bone, as mentioned in the article), if recorded at high sample rates. (That the author had to jump through that many hoops, just to prove that the measurements aren't false readings, IMO emphasizes just how SUBTLE such differences are. This makes me wonder how much of a "practical" difference there really is (as in... with all other things theoretically being equal... would I really "enjoy" one version more than the other, without specifically trying to hear the difference).

I just personally don’t (consciously) hear it, suspect that there’s more people like me than people who do (but that’s speculation), and personally don’t like the idea of uncontrolled audio above 20 kHz.

A little side-story on that:
My subwoofer died not too long ago, and I don’t have a replacement, yet (any recommendations? Also in the market for new monitors). I tried to move fast, and missed that there was unintended low-frequency energy in some of the tracks that I was sent that I didn’t hear, and thus didn’t filter out. I should have caught it, but apparently never looked at the frequency response for that track (shame on me).
The guy I sent it to for further processing DOES have a subwoofer and heard that garbage in the low end immediately, kicked it back and told me to filter that. Glad he caught it.
Now, with energy above 20 kHz, I’d think the same thing as my subwoofer story could happen… except that my guess is, that the next 20 people in line, listening to that mix, would ALSO not be likely to hear it, since either their hearing, or their equipment/speakers aren’t up to the task. But eventually, it might hit that one “real” audiophile who hears those unintentional high frequencies that may be clashing in uncontrolled ways (i.e. maybe even an over-emphasized room-mode/resonance frequency somewhere above 20 kHz), and likes the music less, because of it.
So, taken this way around, I think there’s also a good argument NOT to work with high frequencies, and roll off anything you can’t hear.

But again - that’s just my personal opinion, and I may be terribly wrong about everything. Also, just to note - I don’t get any pleasure out of typing "annoying wannabe smart-a$$" stuff, but rather feel bad for maybe coming across for stepping on other’s toes. Not my intention at all, and sorry to anybody who feels like I did that to them! I’m just hoping that public discourse like this, is in the end, perceived as beneficial to anyone who cares to read along. But at least anything that comes from me, is always just personal opinions and never meant to shut anybody down...

Having typed this much AGAIN, though… I’m seriously considering to take my responses off of this forum and move them to some sort of blog, then just respond to a topic with a “new blog entry - here’s the link” post, so I don’t disrupt the forum topics so much with my ramblings. I often wonder if I’m doing more damage than good on here, because of my long posts that end up making these threads somewhat unreadable…?

[Bruce Graham]

Hi Mike;
Thanks for posting th artical. It's an informative read! I will continue to that. Much to learn.

Y-My-R;
I spent 25 minuetes typing "words" and went to look something up and when I came back to continue, everything I wrote, which BTW was poetry HaHa, was gone! S*it!

Here is the Coles Notes version of 4 paragraphs!
- Please, please continue to write here! I learn a great deal from your writing as well as form better opinions for how I think, and generally enjoy reading what you write even if it doesn't apply to me, so Please continue.
- Beside mixing finished products, CD, TV shows, etc, the other aspect of recordimng it for archieval purposes! I wrote a long story here! My reason for saying this stuff I had recorded 30-40 years ago, like fireplaces, Trains, Wind, Wavwes, Ponds, forests, etc, in highr quality digital audio are still being used today. (received royality cheque a month ago). In these situations recording the raw audio is important so that, the end mixer, can make the sound, sound the way he/she needs them to sound for there project. I'm not saying 96kHz is the way to go, but it's great to know we can go that way and leave things the way they sound for such purposes. My opinion!

I love this love this Forum! "Don't go changing,,,,,,"

Cheers
Bruce
(Wished what I wrote was still there/here... Damn!)

[Old School]

Hi Y-My-R,
I personally enjoy all of your posts and have learned much from them, so I hope you don't stop posting them here. As for the 20k thing, please don't misunderstand me, I do not have golden ears, and have never met anyone who did. But what that article was saying was if a track was mixed on equipment that was capable of recording 50k into the master, that the effect that these frequencies had on the lower frequencies would remain even if you rolled off at say 12k when you pressed the copies. I don't think anyone's ever heard a difference in 96, 48 and 44 because in today's world no commercial music (or very little) is recorded on such equipment. So I guess for all practical purposes it is a moot point, but it's still interesting to talk about.

Have a blessed day,
Mike

[captainamerica]

Good dialog folks...love the passion on all fronts!

[doktor1360]

Wow... this was a good thread. and of course I'm late to the game as usual...

I didn't read every post (yet), however scanning thru it was interesting as there was a lot of good points and information on a whole. Without wasting anyone's bandwidth by regurgitating or quoting any of this, I think an important point to take away from the discussion is that there is no way to avoid any DAC/ADC conversion unless your in/outs are routed from the I/O card cage (hardware) in ADAT format - all other inputs or outputs, with the exception (if memory serves) of the digital 2-track, S/PDIF-AES/EBU and phones digital in n out signals that have already been converted. The only other electronic pathway for any signal to avoid this is via the internal eFX cards - kind of a moot point in regards to the discussion...

Anyone can avoid using the pre-amps by not connecting a microphone to channels 1-12 inclusive, but otherwise very few signals going in and/or out of this console avoid the inevitable digital/analog conversion (internal op amps). The various block diagrams of the signal flow and schematic documents clearly illustrate this, I personally had to pour thru them to get clued-in meselfs as to how things are engineered...

Due to this, it's primarily why I'm implementing the Cranborne 500R8 interface into my gear platform (see recent forum threads) with a Frontier Apache optical patchbay... the USB/ADAT interfaces of the 500R8 are the lynch pins. At some point, I'll have moved to 500 series preamp input(s) - among other things. It's just another way for me to squeeze everything outta this older Mackie tech... for everything it's worth...

Apologies for anyone's waste of their afore-mentioned bandwidth... and thx for hearing me out. I'm certainly not attempting to orate. just contribute where I see fitting. I stand corrected if any of the information in this post proves otherwise...

[Y-my-R]

Sorry for following up so late... lots going on in my work/personal life(s), and on top, I finally spent more time using the D8B to work on music, now that I have it all back in running order

Glad to see that I don't seem to annoy anybody TOO much with all those TOO LONG posts. I guess if some of you find it useful, then it isn't as bad as I feared. I'm glad - thanks!!

About the sample rate topic: Totally - when recording anything that should be reproduced as closely as possible to the original sound (e.g. nature sounds, etc.), that would be the perfect application to use high sample rates, IMO. It definitely doesn't "hurt" and you'll capture the stuff "some" people may be able to hear.

But what that article was saying was if a track was mixed on equipment that was capable of recording 50k into the master, that the effect that these frequencies had on the lower frequencies would remain even if you rolled off at say 12k when you pressed the copies.

That part, I must have missed - sorry about that, Old School! I'll need to re-read that article... but I don't doubt that it's possible that it will impact the sound below 20 kHz, even after rolling off the high-end. I'm sure I wouldn't (consciously) hear it, personally, but again, some might...
In the end, I'll still stick to my the decision for myself, to continue working in 44.1 kHz, since the difference (that I personally don't think I could perceive - at least I didn't during that shootout) doesn't seem worth it to me, to replace the D8B and get larger harddrives for the much larger file sizes I'd then have to deal with.
But I do accept and appreciate the argument... I guess to really form my own opinion, I'd have to do some VERY critical listening (and am probably not even set up for that)

That sample rate shootout at my old company... the guy presenting the audio we were listening to, CLAIMED that he carefully selected materials that were known to have been recorded at high sample rates... I'm no longer sure, though, what the format of the files was... he tried to make a point, that the entire chain from recording to playback, was supposedly appropriate for those high-resolution examples.
What I DO remember from that session was, that we listened to a lot of NICKELBACK (AAAAAAAAAAAAARRRRRRGGHHHHHHH) - supposedly because they recorded that album at high sample resolution (I guess I should research if that's true).
But half the intro from the presenter at that session sounded a bit like BS... so, I wouldn't sign what he said, if I was asked, haha

As for the audio signal internally passing through various OP-amps... I'll have to carefully look at the signal diagram, but can't imagine that the signal would be converted back to analog, once it has been converted to digital, to be sent between destinations in the console (?). I'm happy with the sound I get out of the D8B in the way I use it... but if the signal would get converted from digital back to analog inside the console for some strange reason, it will make me seriously think about ditching the D8B out of principle (well... if I had the kind of money needed to replace it on the spot). I'll go check that out later... that idea scares me, that there'd be unnecessary conversions happening on the inside... wow.

As for the I/O I'm using on the D8B and how I had "tried" to avoid pres/OP-Amps/converters:

I feed the signal from my DAW into the D8B via 3 DIO-8 cards - so, 24 tracks going in digitally.

I do use the analog inputs 1-24 to monitor my electronic drums (TD-30/8 channels - I never use those sounds in a production and replace them with plug-ins before I start mixing... the TD-30 sounds REALLY fake) and my hardware synths are by default connected to the rest of the channels on the Mic/Line bank.
I only use those for monitoring, though, and usually print stems from the synths before mixing... and for that, I go through the converters on my audio interface, to avoid the D8B's pres/converters (but I'd get latency if I'd monitor through the audio interfaces' inputs... so, the D8B is very useful for this).

I DO have to admit, that my ALT I/O card is currently analog, and I use that mainly for the FX Returns from hardware FX units etc. I mostly use plug-ins in the DAW, though, and just use hardware if I want a specific sound I'm familiar with (e.g. the gated reverbs in my Roland DEP-5). So, "some" of the D8B's converters do end up in the mix... but they're not going to make anything sound "BAD" - I just don't want to "stack" too many tracks that were recorded through those pres/converters, because the STACKING of tracks recorded like that, IMO, wouldn't sound as good as when tracking the majority of the tracks through better pres/converters.

I do intend to put a digital card in the Alt I/O slot, and connect that to "somewhat" better converters - but I'll do that when the opportunity arises. For example, a used M-Audio ProFire 2626 (they go for around $100) can work as a standalone 8-channel AD and DA converter... and at least in my personal opinion, those converters were still better than the ones in the D8B (I have a ProFire 610 that I use as a standalone S/PDIF AD/DA converter, it sounds OK, and has the same converters as its bigger brother) .
So, if I run across a cheap used offering for a 2626 (or something similar that works standalone), the ALT I/O card in the D8B will be switched out for a digital/ADAT card.

From there, I'd put those two ADAT cables on the ALT I/O on my digital patchbay... and hook up the ADAT from that 2626 (that I don't have yet) to the digital patchbay as well... and connect the analog I/O of the 2626 to the (analog) patchbay.

This way, I'd then have a choice if I want to use the ALT I/O to tie in external (analog) hardware devices via the converters in the 2626, OR instead feed another ADAT optical source into the D8B, when the tape send/returns 1-24 are already in use without having to sacrifice playback channels.

I'm currently doing "something like that" but have the Tape Send/Returns 1-8 on the digital patchbay instead of the ALT I/O. By default, that's just mapped to the ADAT I/O on my audio interface/DAW - just as if I'd have them connected, directly - so, just 24 digital I/O incl. the 8 channels on the first Tape card.

But sometimes I want to bring in signals from a different ADAT source that lives back by the electronic drum kit. I have that ADAT source on the digital patchbay as well. So, when I want to use it, I currently patch it to Tape Returns 1-8... but lose the first 8 outputs from my DAW as a result.
Since I often feed more than 16 tracks from the DAW into the D8B via the Tape Returns, but only rarely use the ALT I/O for hardware effects, it makes sense for me to go digital for the Alt I/O and add another set of converters, so I can choose different ADAT sources for the ALT I/O.

Maybe it wouldn't be worth it for ONLY the slightly better converters in the 2626... but since it will give me more routing flexibility, I'm planning to go for it, anyway. But I'm waiting for a good deal on a 2626 (...I was offered one free at some point but didn't realize how useful this could have been... oh well).

...how about now? Too long, yet?