What is the base sampling rate?

@vangrieg - I don't know, but I think a lot of oversampling would increase the latency substantially. Currently, it seems the latency is just that of the AD/DA converters.
 
The biggest proof was when I first recorded a drumset at 176k. I could not believe how awesome it sounded.

Well that’s not proof. Scientific tests do not confirm that, or that people can actually distinguish between a properly encoded 320 kbps mp3 and uncompressed wav or even dsd. A lot of mp3s (or m4as, doesn’t matter), are not encoded properly, so there’s clipping whole decoding and that’s where the noticeable difference comes from.
 
Yes to the extent that if your buffer size is 256 samples a higher sample rate will reduce the latency time.

But I am pretty sure that the processing latency in all Fractal products is exactly 0. And that means no oversampling.
 
1) You can't hear the difference, with modern sampling and converters and roll-offs.
2) That doesn't mean the difference doesn't matter.

But, remember that it's always better to resample something with software after the fact, than try to get a single set of converters to work well at multiple resample rates.

The Fractal approach makes perfect sense, given they can work at incredible fidelity at 48Khz, and you can just adjust it to fit your audio project after the fact.

Converters running at a rate other than their optimal rate is always sub-optimal (duh). They're simply not equally efficient at every rate, given the need for roll-offs and so-on.

Digital audio is interesting, and I'd wager Cliff knows more about it than you (where "you" is the reader of this comment ;) ).
 
...There absolutely was a difference then and there is one now. I have yet to not prove it even to the most skeptical if using decent equipment. The biggest proof was when I first recorded a drumset at 176k. I could not believe how awesome it sounded. It was astounding. Every nuance was clear and present...
How are you controlling for placebo there? Are you doing A/B/X, double-blind?

Placebo is incredibly powerful, and if not fully controlled for, it can convince you of anything.

There's still a giant market for acupuncture and chiropractic and all that other pseudoscience. Never underestimate it.
 
The Fractal approach makes perfect sense, given they can work at incredible fidelity at 48Khz, and you can just adjust it to fit your audio project after the fact.

Sigh...

You may use other equipment. Fixed sampling rate creates interoperability problems. You may be forced to use multiple DA/AD conversions, that's worse than resampling within the unit. Nobody outside of Fractal world does projects in 48k by default. And so on.
 
I don’t grok how you can love Fractal products and simultaneously claim that ultra-high sampling rates are the only way you’ll experience audio. The two statements are not in alignment. You should not like Fractal products if lower sampling rates were sonically inferior to these 96 kHz and above rates being thrown about here as the be all and end all of sonic fidelity.

You can always record the analog outs if sample rate matching is proving problematic for you. The output from the unit is a smooth waveform, ready for sampling and re-digitizing at any frequency and bit depth you like.
 
I don’t grok how you can love Fractal products and simultaneously claim that ultra-high sampling rates are the only way you’ll experience audio.

For the record, I personally never said this, and I don't think so. I personally need the ability to switch to 44.1, as a matter of fact.
 
Sampling rate affects only one thing: the highest frequency that can be accurately reproduced. That's it.

With modern anti-aliasing filters, 48 KHz sampling can accurately reproduce just a little over 20 KHz, which is the highest frequency that the best human ears can hear. Anything above that is inaudible, which makes it a waste of resources in an audio system.
 
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Sigh...

You may use other equipment. Fixed sampling rate creates interoperability problems. You may be forced to use multiple DA/AD conversions, that's worse than resampling within the unit. Nobody outside of Fractal world does projects in 48k by default. And so on.
Who would convert something into analog just to change the sample rate within a project? That's a horrible way to do it (although it would achieve the desired outcome).

Again: Hardware is necessarily limited. It doesn't work as efficiently at different rates. If you allowed everyone to choose their rate (including fad rates like 384Khz), you couldn't possibly make it run efficiently at each, and you would run into CPU limitations with big presets.

As I stated, you can do a soft resample without losing anything, and then work with the raw file in the rate you prefer. Or you can use DAWs that resample in real-time like REAPER.

It's just a better outcome, and it provides more flexibility with the creation of the Axe by limiting it at the hardware end. It allows it to be an incredibly efficient D/A conversion at 48Khz.
 
It kind of reminds me how, just recently, people here were arguing that Axe fx has enough power and doesn’t need to be updated.

Use Reaper...

The question isn’t “how do I record Axe into a project with a sample rate other than 48k”. There are ways to do it, and there are pros and cons to each of them. The question is “please make it possible to switch sample rate in Axe FX”. There are multiple reasons why this will be convenient, and not just for recording.
 
The question isn’t “how do I record Axe into a project with a sample rate other than 48k”. There are ways to do it, and there are pros and cons to each of them. The question is “please make it possible to switch sample rate in Axe FX”. There are multiple reasons why this will be convenient, and not just for recording.
Yeah, I get that. But should we limit everyone's presets so that we know they'll work at 384Khz, which takes a lot more processing power than 48Khz?
 
Making the AxeFX III a robust audio interface was a good goal. That would be very useful. But without selectable sample rate and without mic input, the specs fall short of achieving the "center of your workstation" claim on the product page. See an Eleven Rack or a Helix Rack for an example of a modeler that can serve as the center of your workstation.
 
Helix Rack for an example of a modeler that can serve as the center of your workstation.

But Helix works internally at 48 kHz, which kind of sucks. If it accepts higher sample rates externally, it will downsample everything internally. So not a great benchmark for the center of a studio workstation.

But convenience? Yes.
 
Sampling rate affects only one thing: the highest frequency that can be accurately reproduced. That's it.

With modern anti-aliasing filters, 48 KHz sampling can accurately reproduce just a little over 20 KHz, which is the highest frequency that the best human eats can hear. Anything above that is inaudible, which makes it a waste of resources in an audio system.
I'm 28 years old, and I don't here any differences between 48 kHz and 96 kHz processed material. But I do remember something about filter slopes getting worse near the Nyquist frequency.
 
Well that’s not proof. Scientific tests do not confirm that, or that people can actually distinguish between a properly encoded 320 kbps mp3 and uncompressed wav or even dsd. A lot of mp3s (or m4as, doesn’t matter), are not encoded properly, so there’s clipping whole decoding and that’s where the noticeable difference comes from.

That is the beauty of America. We can each believe what we want. I know, that for me, hi res makes a huge difference. I gave up evangelizing that long ago. If you just hear Eddie's guitar in hi-res vs CD... ah.... just has more kick. But... that is just me and most everyone I have played it for. I could go into theory about why the others don't work... but I just don't care anymore. I'm happy.
 
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