Wish Axe FX at 96k

Will any future Fractal Axe FX devices work on 96k other than fixed 48k it would be so much more convenient as most studios I work in choose 192k or 96k……..and before any smart arses tell me how to convert that is NOT the question.

I understand the extra cost and complexity of the build but as far as I’m concerned this is a professional product and I shouldn’t have to have technical hurdles hampering the creative process.

Not a grievance ( I still love it for live ) just a genuine question to the manufacturer before the “haters” and “know it all’s” give me irrelevant advice and tell me what I “should” be doing.
 
Use the analog outs into the line in of the session interface.

Problem solved with zero latency issues and zero quality issues.

That's what everyone does with their Kemper or other devices when people want to record at 96 KHz 32 Bit or even worse 192 KHz (unless you are doing a Blue-Ray DVD Audio session).

You are recording a lot of digital black above 24 KHz frequency range when you go beyond 48 KHz sampling recording rate anyways for no good reason with most musical instruments.

There's professional with a reason for doing something and then there's professional falling for marketing hype and using the max sample rate on an interface/software just because you could.

Do you really think any of the previously recorded music of the last 100 years is rerecorded when used for theatrical soundtracks or are they sample rate converted? A lot of that was mastered at 44.1 KHz at some point and still sounds good sample rate converted to 192 KHz for the theatrical/Blu-ray release.

This is not a smart ass answer either, this is a "if there's no info to record in a frequency range, why are you wasting time, storage space, money, and effort recording it?" pragmatic answer.
 
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That's what everyone does
The wish in the OP is ambiguous, but I don't think he's asking for SRC in the AxeFX from 48K to 96K. There wouldn't be much point to that when it can already be done in almost all DAWs on the fly during recording. I believe he's asking for doubling the internal SR to 96K, hence the comment about the "extra cost and complexity". Maybe the OP can clarify what he's asking for?
 
The wish in the OP is ambiguous, but I don't think he's asking for SRC in the AxeFX from 48K to 96K. There wouldn't be much point to that when it can already be done in almost all DAWs on the fly during recording. I believe he's asking for doubling the internal SR to 96K, hence the comment about the "extra cost and complexity". Maybe the OP can clarify what he's asking for?
Even if the OP is asking for internal sampling rate to be boosted to 96 KHz, if there's no sonic information in a guitar or bass past 24 KHz besides digital noise floor, what is one really gaining by that move, other than session sample rate compatibility?

That begs the question, if one runs the analog outs of the Axe-FX III into the analog line ins of a great interface using good cables, can one hear a difference versus the digital outs of the Axe-FX III?

If you can't hear a difference that way, what is one really gaining in sonic benefit increasing the internal sample rate, other than taxing the processor more?
 
Even if the OP is asking for internal sampling rate to be boosted to 96 KHz, if there's no sonic information in a guitar or bass past 24 KHz besides digital noise floor, what is one really gaining by that move, other than session sample rate compatibility?
There are reasons other than session SR compatibility. For example, processing at higher than 48K is routinely done, even in the AxeFX, to avoid aliasing artifacts when doing non-linear computations. Usually the upsampling is done selectively, as needed in the signal chain. Session SR compatibility isn't needed with most DAWs these days anyway, so I'm also confused about what the OP is asking for :).
 
1/20 = 0,05sec at 48k samplerate = 2500 times
1/20K = 0.00005sec at 48k samplerate = 2,5x times
48K = 0,000020sec

Basic tone
the higher the frequency, the faster something changes.
The higher the sampling rate must be to capture it accurately.

With the overtones, which can be a multiple of the basic tone, the frequencies also go up.

Increasing the sampling rate only makes sense when capturing more complex events with extended dynamics.

It is clear, however, that the more accurately the data is recorded, the more accurately it can be reproduced. Whether everyone can hear that?

Examples
Spatial sound with many objects
Mixing of many tracks simultaneously in the DAW
and more...

Decisive for the sound are the AD/DA converters used.
 
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88,2 for mastering guys, only divided by 2, easier conversion .
I’ve read a lot of things about this stuff as I record a lot and like ever, you can read opposite things and you don’t know what to do . I choose to record in low sampling rate (44,1) to have less conversion in the end, no surprises, and my computer say “thank you”
 
Use the analog outs into the line in of the session interface.
That’s what I do 👍🏻. Then you have the choice . at high rate the project is so big in the computer 🤦 . maybe I need to retry one day if you gain that much recording at 96 hz and 32 bit and then mastering at 44,1 and 16… but … that’s strange to work at a high quality rate, be happy, and then doing a mp3 lol (exaggerating) . You understand what I mean ? You have no surprise if you record at 44,1 first and are happy with the sound, it will be the same result in the final product
 
Use the analog outs into the line in of the session interface.

Problem solved with zero latency issues and zero quality issues.

That's what everyone does with their Kemper or other devices when people want to record at 96 KHz 32 Bit or even worse 192 KHz (unless you are doing a Blue-Ray DVD Audio session).

You are recording a lot of digital black above 24 KHz frequency range when you go beyond 48 KHz sampling recording rate anyways for no good reason with most musical instruments.

There's professional with a reason for doing something and then there's professional falling for marketing hype and using the max sample rate on an interface/software just because you could.

Do you really think any of the previously recorded music of the last 100 years is rerecorded when used for theatrical soundtracks or are they sample rate converted? A lot of that was mastered at 44.1 KHz at some point and still sounds good sample rate converted to 192 KHz for the theatrical/Blu-ray release.

This is not a smart ass answer either, this is a "if there's no info to record in a frequency range, why are you wasting time, storage space, money, and effort recording it?" pragmatic answer.
No. I record almost exclusively at 96k. It's a real problem. Reamping analog creates hiss. A little hiss but hiss nonetheless. I'm 100% behind the OP. I wish they allowed for multiple sample rates. I accept it's a problem, but still.
 
That’s what I do 👍🏻.
If you have a choice (depending on who is producing the session, sometimes you don't), your best option is to record your AxeFX digitally at 48KHz, then, at the end of your project, use a high quality non-realtime SRC to the desired sample rate. That way you avoid unnecessary D/A/D conversions coupled with unnecessary on-the-fly sample rate conversions.

Recording analog output into an audio interface is useful as a last resort when you are in somebody else's studio and you can't find any other way to connect your AxeFX to their DAW.

The idea that if the AxeFX had realtime SRC your problems would be solved doesn't make sense, when almost all DAWs are already capable of doing that. Moving that SRC from the DAW to the AxeFX wouldn't improve anything. But I suppose it would help if you're using a DAW that isn't capable of that.
 
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Using the axe in a sound card is globally easier for my set up, as we record other instruments in the same room (vocals, drums…) , I don’t want to use the axe to listen to music from my computer for example, so having the studio monitoring plugged in the axe is “annoying”
 
Using the axe in a sound card is globally easier for my set up, as we record other instruments in the same room (vocals, drums…) , I don’t want to use the axe to listen to music from my computer for example, so having the studio monitoring plugged in the axe is “annoying”
It's easier for many people. But that doesn't mean you need to give up recording digitally. You can record digital output from the AxeFX while using an audio interface.
 
If you have a choice (depending on who is producing the session, sometimes you don't), your best option is to record your AxeFX digitally at 48KHz, then, at the end of your project, use a high quality non-realtime SRC to the desired sample rate. That way you avoid unnecessary D/A/D conversions coupled with unnecessary on-the-fly sample rate conversions.

Recording analog output into an audio interface is useful as a last resort when you are in somebody else's studio and you can't find any other way to connect your AxeFX to their DAW.

The idea that if the AxeFX had realtime SRC your problems would be solved doesn't make sense, when almost all DAWs are already capable of doing that. Moving that SRC from the DAW to the AxeFX wouldn't improve anything. But I suppose it would help if you're using a DAW that isn't capable of that.
I don't understand this. Having a DAW that can change isn't the issue. Having an interface that can do different sample rates isn't either. I have many devices - effects boxes and reverbs that have to match the sample rate. Bricasti and Lexicon PCM92 reverbs. THEY have to match the rate of the interfaces, not the other way around. If they can't I either can't use them or I have to use them analog, like I do with my Lexicon PCM 91. Fortunately there's not loss in fidelity like there is in Axe Fx III.

There's no practical way to reduve the DAW project's rate to 48k. I can bounce the whole project to 2 track and then record my guitar parts, up sample of my guitar and import them into the 96k project. But holy hell, that's a lot of extra work. It's just not worth the extra time. Speed is of the essence with studio work.
 
I tried my damnedest to change my Scarlett 2i2 to this sample rate, but it just won't let me. Disappointing, I wanted to see the light.

IME I previously had an Scarlett 18i20 gen 2 that, no matter what, couldn't sync with my AX8 via S/PDIF... There's a well number of posts by me rambling about it on the AX8 sub-forum. In the end, I ditched the 2nd gen as soon as the 3rd gen was available and... the new one just worked out of the box.

My point is that there's something going on with the Scarlett series as it works for some people, it doesn't for others... YMMV
 
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