Wish Axe FX at 96k

I don't understand this.
Don't get me wrong. I'm not suggesting there is anything wrong with adding SRC to the AxeFX. It would be quite useful in situations like yours. I'm just pointing that for simpler use cases it merely moves the on-the-fly SRC from one point in the signal chain to another and doesn't accomplish anything. That is precisely why you don't see SRC in the AxeFX today.

In any case, as I mentioned above, SRC doesn't seem to be what the OP is asking for, but it's hard to tell.
 
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
It's probably more to do with the ASIO drivers, I have the 3rd gen 2i2 and it works great for everything I need it to, really great interface that I can't complain, I just can't set it to 64k. Not that I would since the Axe runs at 48k, which basically dictates the projects I do.
 
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?
The answer to this is 100% yes. I wish it wasn’t. This was how I planned to work around this, and what I did at first. But once I hooked it up for a level matched, blind A/B test, I had to go the digital route. The difference is not subtle. Once you hear it, you can’t unhear it and it’s hard to listen to stuff I recorded that way before I switched.

The workaround that I’m using is a $1,500 format converter from RME ($800 on eBay if you’re patient). It was tricky to set up at first, but I haven’t thought about or touched it since. Other than the cost, I’m very happy with this solution. I can set my session to any sample rate, and no matter what the converter is doing to the sound, it still sounds a whole other league better than the analog path with the D/A/A/D conversion. If you don’t like this answer and don’t want to buy the external converter, just don’t listen to them side by side. The analog path sounds fine until you hear it next to the digital.

Most of the session files that I play on were not created in my studio, so I don’t get the luxury of deciding what sample rates they come in as. Most of them are 96k, but not all. To me, the question of what sample rate is best doesn’t play into it. It’s not a philosophical, technical or religious discussion. The only decision is, do I want to take the job or don’t I?
 
Once you hear it, you can’t unhear it and it’s hard to listen to stuff I recorded that way before I switched.
Can you describe in words what exactly you perceive?

Was it a double blind test?

I was part of a double blind test with a great control room monitoring system and I could not perceive a marked difference except in cymbal decay and reverb tails. It was so subtle, I can understand why the consumer market for music considers the juice not worth the squeeze.

If there's no info to record in a frequency range, why are you wasting time, storage space, money, and effort recording double or triple the frequency range of a non existent harmonic?
 
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The answer to this is 100% yes. I wish it wasn’t. This was how I planned to work around this, and what I did at first. But once I hooked it up for a level matched, blind A/B test, I had to go the digital route. The difference is not subtle. Once you hear it, you can’t unhear it and it’s hard to listen to stuff I recorded that way before I switched.

The workaround that I’m using is a $1,500 format converter from RME ($800 on eBay if you’re patient). It was tricky to set up at first, but I haven’t thought about or touched it since. Other than the cost, I’m very happy with this solution. I can set my session to any sample rate, and no matter what the converter is doing to the sound, it still sounds a whole other league better than the analog path with the D/A/A/D conversion. If you don’t like this answer and don’t want to buy the external converter, just don’t listen to them side by side. The analog path sounds fine until you hear it next to the digital.

Most of the session files that I play on were not created in my studio, so I don’t get the luxury of deciding what sample rates they come in as. Most of them are 96k, but not all. To me, the question of what sample rate is best doesn’t play into it. It’s not a philosophical, technical or religious discussion. The only decision is, do I want to take the job or don’t I?
That's a common dilemma, but an uncommon solution. Most DAWs can simply do the SRC for you when recording (or even allow you to record audio at mixed sample rates), which of course is better than doing D/A/D when recording.
 
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 :).

For the people who like me have no idea what we are talking about here, I found this video extremely helpful to understand aliasing and the impact of sample rate.

 
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”
In my Mac, I can set up an "Aggregate Device" comprised of all the Axe's and my Scarlett audio box's inputs and outputs (and any other audio I/O devices, including the laptop mic and speakers if I want). This lets me leave my audio monitoring attached to the computer and record from the Axe....
 
The answer to this is 100% yes. I wish it wasn’t. This was how I planned to work around this, and what I did at first. But once I hooked it up for a level matched, blind A/B test, I had to go the digital route. The difference is not subtle. Once you hear it, you can’t unhear it and it’s hard to listen to stuff I recorded that way before I switched.

The workaround that I’m using is a $1,500 format converter from RME ($800 on eBay if you’re patient). It was tricky to set up at first, but I haven’t thought about or touched it since. Other than the cost, I’m very happy with this solution. I can set my session to any sample rate, and no matter what the converter is doing to the sound, it still sounds a whole other league better than the analog path with the D/A/A/D conversion. If you don’t like this answer and don’t want to buy the external converter, just don’t listen to them side by side. The analog path sounds fine until you hear it next to the digital.

Most of the session files that I play on were not created in my studio, so I don’t get the luxury of deciding what sample rates they come in as. Most of them are 96k, but not all. To me, the question of what sample rate is best doesn’t play into it. It’s not a philosophical, technical or religious discussion. The only decision is, do I want to take the job or don’t I?
Format converter?? Do tell.
 
Reaper. $60 for a non-commercial license. Handles on-the-fly resampling and offline sample rate conversion like a champ.
But Reaper is a DAW. Doesn't it require hardware for sample rate conversion? I already have and use Digital Performer, which does automatic aggregation. YOu can use as many interfaces as you want but they're all stuck at the hardware sample rate, Pro Tools, Logic and Ableton 11.
 
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.
^^^THIS^^^
 
Another thing to keep in mind is that many devices that offer different sample rates are simply resampling the audio as well. So your interface that outputs at 96kHz may actually be processing the audio at 48kHz internally and then upsampling it to 96kHz.

It's kind of like upsampling DVD players. They may output at 4K or higher resolution, but the source from the disk is still Standard Definition video. You can interpolate to some extent, but you can't really create more detail where there was none in the original sampling of the waveform.
 
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I not going to frigging argue this stupid debate. I can hear the difference regardless nyquist and null tests. If you have high end interfaces and sample rate conversion you can hear it. I don't mean to insult anyne at all, but please, this is not part of the thread, or shouldn't be.
 
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