The Axe-Fx 3 works only at 48Khz

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I read in another thread that the motivation of fixed sample rate at 48Khz is that Cab IRs need to be resampled to make other sample rate available....... i think they are wrong on this.
It would have helped your original post if you mentioned that you knew this information.

You are of course entitled to disagree with the designer's decision but again, there's no reason to called that decision "stupid". :)
 

Exactly. FAS have specific engineering reasons for this 48k standard. As linked above, I’ve provided the exact text. Although you really should learn to use the various resources before opening yor cake hole and removing any doubt you’re a PITA/troll.
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From the wiki:

The sample rate of the Axe-Fx series, FX8 and AX8 is fixed at 48kHz (24-bit).

Digitally connected devices and DAW software always need to be set to the same sample rate. Example

If required, resampling can be handled by software.

"The Axe-FX uses higher sampling rates (oversampling) during the processing stages. This is how it avoids aliasing when non-linearities are applied. But the sampling rate of the audio that is sent to the DAC is the same as the sampling rate coming out of the SPDIF output: 48khz. In other words, it goes from 48khz (ADC) -> higher sampling rate -> 48khz (DAC).

So just because these higher sampling rates are used for the processing stages doesn't mean it would be trivial to send a higher rate to the SPDIF output. The 48khz signal would need to be sample rate converted (SRC) at the output stage by a hardware SRC chip and Cliff's whole point is that software SRC's provide better quality than what is available with hardware SRC's." "IMHO, the ideal sample rate is 64 kHz but that's not a standard. The nice thing about 64 kHz is that you can have a gentle transition band from 20 kHz to Nyquist which results in shorter filters, lower latency, less phase shift, etc.

I was very tempted to make the Axe-Fx II run at 64 kHz but people probably would have freaked out." source"I've long maintained that 64 kHz is the ideal sample rate for audio. But I can't get the industry to change." source"48 kHz is considered "pro" sampling rate.

The reason for 44.1 kHz on CD's is subject to debate. Some maintain that the sample rate was lowered so that Beethoven's 9th would fit on a single CD. Others claim that it was because that rate was compatible with video equipment. IMO 44.1 kHz is insufficient for professional audio.

Personally I would prefer 64 kHz. Whilst Nyquist theorem is all well and good most people don't understand the details and simply state "the sample rate must be twice the highest desired frequency". The problem with this is as you approach Nyquist the filter demands become extreme.

The more extreme the filter demands the more taps are needed, the more precision is needed, the more latency is incurred, etc. A 64 kHz sample rate would give you a nice, smooth roll-off from 20 kHz to 32 kHz rather than the brick wall you get with 44.1 kHz.

There is no hardware advantage to using 48 vs. 44.1. The costs would be the same in either case. Modern converters use over-sampling techniques to implement the necessary anti-aliasing filters thereby reducing off-chip filtering to simple circuits. MP3s have no native sample rate but are typically 44.1 kHz because they are usually derived from CDs.

MP3 is a psycho-acoustic compression format that exploits frequency masking to lower the data required to store audio information." source"If the Axe-Fx were running at 44.1 all the cab IRs would need to be resampled, or there would need to be an SRC chip on the digital I/O. There is no free lunch.

The problem isn't the Axe-Fx, the problem is studios stubbornly sticking to 44.1 when 48 is a much better rate." source"1. 44.1 or 48 KHz is more than adequate for not only guitar processors but ANY audio processor. 88.2 or 96 K makes for nice marketing but, in reality, performance can often decrease when running converters higher than necessary. This is due to activity at the converters digital I/O pins injecting noise into the converters themselves.

Personally I wish the industry would adopt a 64 KHz sample rate standard but this is for esoteric reasons.

2. The dynamic range of a guitar, UNDER IDEAL CONDITIONS (i.e. inside a Faraday cage) is not much greater than 100 dB. To capture this you would theoretically need 17 bits (17 bits gives about 102 dB). To allow sufficient "overhead" one should add a couple bits. 20 bits is plenty and yields about 120 dB of dynamic range. Anything greater than 20 bits is marketing.

There isn't a converter made that gets much better than 100 - 120 dB dynamic range in the real world. You only need 20 bits for that. AKM has these new 32 bit converters (AKM557x). This is comical as they only have 112 dB of dynamic range so they give 19 bits of data and 13 bits of noise. Once you put a guitar in a real-world EMI environment that dynamic range drops precipitously (60 dB or even less).

The ANALOG electronics before and after the converters is far more important. Knowing when to use JFET vs. bipolar op-amps, knowing how to select the right op-amp for the task, etc. far outweigh the sampling rate and advertised bit depth of a converter. Good quality components aren't cheap though.

Internal oversampling determines aliasing performance in nonlinear processing. The higher the oversampling, the lower the aliasing but the more processing power required (= $$$$). Aliasing noise can easily dominate output dynamic range. So, again, sampling rate and bit depth are immaterial in comparison to the things that really matter." source
 
What digital rate converter would be of suitable quality to interface the Axe? Recommendations anyone?

There’s got to be a relatively inexpensive box to do this. It might introduce a small bit of latency. Software conversion would be another option depending on use. I for one don’t want the Axe3 price to go up $100 just to add a sample rate converter box to the spdif output.
 
Exactly. FAS have specific engineering reasons for this 48k standard. As linked above, I’ve provided the exact text. Although you really should learn to use the various resources before opening yor cake hole and removing any doubt you’re a PITA/troll.
:::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::

From the wiki:

The sample rate of the Axe-Fx series, FX8 and AX8 is fixed at 48kHz (24-bit).

Digitally connected devices and DAW software always need to be set to the same sample rate. Example

If required, resampling can be handled by software.

"The Axe-FX uses higher sampling rates (oversampling) during the processing stages. This is how it avoids aliasing when non-linearities are applied. But the sampling rate of the audio that is sent to the DAC is the same as the sampling rate coming out of the SPDIF output: 48khz. In other words, it goes from 48khz (ADC) -> higher sampling rate -> 48khz (DAC).

So just because these higher sampling rates are used for the processing stages doesn't mean it would be trivial to send a higher rate to the SPDIF output. The 48khz signal would need to be sample rate converted (SRC) at the output stage by a hardware SRC chip and Cliff's whole point is that software SRC's provide better quality than what is available with hardware SRC's." "IMHO, the ideal sample rate is 64 kHz but that's not a standard. The nice thing about 64 kHz is that you can have a gentle transition band from 20 kHz to Nyquist which results in shorter filters, lower latency, less phase shift, etc.

I was very tempted to make the Axe-Fx II run at 64 kHz but people probably would have freaked out." source"I've long maintained that 64 kHz is the ideal sample rate for audio. But I can't get the industry to change." source"48 kHz is considered "pro" sampling rate.

The reason for 44.1 kHz on CD's is subject to debate. Some maintain that the sample rate was lowered so that Beethoven's 9th would fit on a single CD. Others claim that it was because that rate was compatible with video equipment. IMO 44.1 kHz is insufficient for professional audio.

Personally I would prefer 64 kHz. Whilst Nyquist theorem is all well and good most people don't understand the details and simply state "the sample rate must be twice the highest desired frequency". The problem with this is as you approach Nyquist the filter demands become extreme.

The more extreme the filter demands the more taps are needed, the more precision is needed, the more latency is incurred, etc. A 64 kHz sample rate would give you a nice, smooth roll-off from 20 kHz to 32 kHz rather than the brick wall you get with 44.1 kHz.

There is no hardware advantage to using 48 vs. 44.1. The costs would be the same in either case. Modern converters use over-sampling techniques to implement the necessary anti-aliasing filters thereby reducing off-chip filtering to simple circuits. MP3s have no native sample rate but are typically 44.1 kHz because they are usually derived from CDs.

MP3 is a psycho-acoustic compression format that exploits frequency masking to lower the data required to store audio information." source"If the Axe-Fx were running at 44.1 all the cab IRs would need to be resampled, or there would need to be an SRC chip on the digital I/O. There is no free lunch.

The problem isn't the Axe-Fx, the problem is studios stubbornly sticking to 44.1 when 48 is a much better rate." source"1. 44.1 or 48 KHz is more than adequate for not only guitar processors but ANY audio processor. 88.2 or 96 K makes for nice marketing but, in reality, performance can often decrease when running converters higher than necessary. This is due to activity at the converters digital I/O pins injecting noise into the converters themselves.

Personally I wish the industry would adopt a 64 KHz sample rate standard but this is for esoteric reasons.

2. The dynamic range of a guitar, UNDER IDEAL CONDITIONS (i.e. inside a Faraday cage) is not much greater than 100 dB. To capture this you would theoretically need 17 bits (17 bits gives about 102 dB). To allow sufficient "overhead" one should add a couple bits. 20 bits is plenty and yields about 120 dB of dynamic range. Anything greater than 20 bits is marketing.

There isn't a converter made that gets much better than 100 - 120 dB dynamic range in the real world. You only need 20 bits for that. AKM has these new 32 bit converters (AKM557x). This is comical as they only have 112 dB of dynamic range so they give 19 bits of data and 13 bits of noise. Once you put a guitar in a real-world EMI environment that dynamic range drops precipitously (60 dB or even less).

The ANALOG electronics before and after the converters is far more important. Knowing when to use JFET vs. bipolar op-amps, knowing how to select the right op-amp for the task, etc. far outweigh the sampling rate and advertised bit depth of a converter. Good quality components aren't cheap though.

Internal oversampling determines aliasing performance in nonlinear processing. The higher the oversampling, the lower the aliasing but the more processing power required (= $$$$). Aliasing noise can easily dominate output dynamic range. So, again, sampling rate and bit depth are immaterial in comparison to the things that really matter." source

So there is a way to use a software SRC in realtime to be able to use AXE FX 3 with sample rate settings of the DAW different from 48Khz? I mean connecting an AXE FX 3 digitally to the DAW and get the connection work even at different sample rate?
 
FWIW it's not just Fractal Audio.

Just looking around at the gear I have in my studio, my Korg Kronos is locked at 48KHz, Roland Jupiter 80 at 44.1KHz (while the very similar Integra-7 will do up to 96KHz), Fractal AxeFX III at 48KHz, UA OX at 44.1KHz, and the Kemper was locked at 44.1KHz for years but recently got a firmware update enabling up to 96KHz.

Gold star to the Yamaha Montage which can do 44.1KHz to 192KHz.
 
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No, he isn't. Words have meaning.

https://en.wikipedia.org/wiki/Internet_troll

A troll is somebody that says inflammatory things intended to provoke and get a rise out of people. A troll is not somebody that disagrees with you or says something stupid. OP said up front that he had a rant and expressed his appreciation for the other positive aspects of the product, and even admits that he might have wasted 30 minutes of his life. Even though I vehemently disagree with what he has to say his post sounds sincere, if hyperbolic and one-dimensional. That is not a troll. The infamous Romeo Rose was a troll because he willfully disregarded what people said and drew absurd conclusions that were obviously designed to garner the ire of the average forum poster... big difference.

That being said:



Who cares? Hook it up via XLR and go back to making music. Every studio I've ever been to has run my Axe-Fx (all two generations of it; I have yet to upgrade to the III) through an outboard pre or the desk, just as they would a real guitar amp (which the product is designed to emulate). It all ends up as a sine wave eventually, after all...


I care, and you will if you'll be the same situation i'm in.
 
FWIW it's not just Fractal Audio.

Just looking around at the gear I have in my studio, my Korg Kronos is locked at 48KHz, Roland Jupiter 80 at 44.1KHz (while the very similar Integra-7 will do up to 96KHz), Fractal AxeFX III at 48KHz, UA OX at 44.1KHz, and the Kemper was locked at 44.1KHz for years but recently got a firmware update enabling up to 96KHz.

Gold star to the Yamaha Montage which can do 44.1KHz to 192KHz.

Is it possible Fractal can update via firmware like Kemper did?
 
Ok, let's talk about an ipotetic scenario and see what you guys would do:

I've got a project from a client set at 44.1 with guitar DIs recorded in it, I want to reamp those DIs with AXE FX3.
There is a way to do that in the digital domain without going back to analog?
How could i do that?
 
if for some reason my old projects were in 44.1, i would use the axe analog.

once i get the axe, i'd use 48 in the future. seems like an easy solution.

i've reamped using analog several times with different mixers with no issue regarding quality.
 
What’s that old ditty, better to remain quiet and possibly be thought a fool, than to speak and remove all doubt.

If you’re this butt hurt over something being 48khz then you have bigger issues, and may not be as pro as you think you might be.

Of course this is all based upon the original post, is not a personal attack, but a reply to the words in the post.

Cheers
Anthony
 
If I were the OP I would go full analog. Problem solved. Otherwise It doesn't seem like he cares as much as he states. I know I don't.
 
project from a client set at 44.1 with guitar DIs recorded in it, I want to reamp those DIs with AXE FX3.
There is a way to do that in the digital domain without going back to analog?

With something like Reaper you'd open the project and reamp. The 44.1 tracks would be resampled to 48k and sent to the Axe-FX. With a separate interface and SPDIF/AES connection to Axe, you'd want to make sure the interface was set to 48k first, and there's a Reaper setting to prevent projects from changing the current samplerate.
 
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