Axe-Fx II "Quantum" Rev 8.00 Public Beta

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All right, here we go:

ODS Clean and TX Star Clean: kind of like the unit's in bypass mode, very dull.

Nuclear: has a nasty buzzy distortion, almost like digital clipping.

Not sure about the Boutique now, haven't played the previous version for some time.

Fenders, Marshalls, Wreckers etc. all sound great.

I'm on an XL.
 
Nuclear: here's a clip and the preset.

 

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I've always felt the Fractal is a little dark, overall. Which is much better than the fizzy harsh high end of the Line 6 stuff. The Q7 changes added some low end depth and definitely made the amps sound fuller and less "flat". If Cliff can tweak the high end to be a little more present and open it would be really hard to see how that could be improved upon.

I completely get what @DreDawgie is saying. Since I've owned the AX8 (only since December), the primary focus has been refining the amp modeling rather than adding models. That's a great thing. But new toys are fun too, especially since there are quite a few effects that aren't in the device currently.
 
yek said:
Alright, here we go:

ODS Clean and TX Star Clean: kind of like the unit's in bypass mode, very dull.

Nuclear: has a nasty buzzy distortion, almost like digital clipping.

Not sure about the Boutique now, haven't played the previous version for some time.

Fenders, Marshalls, Wreckers etc. all sound great.

I'm on an XL.
Click to expand...

with just some cursory listening to freshly made stock setting presets using the above amps, I am forced to agree with Yek's assessment of the adverse tone changes in them. X/Ying between the TX Star(previously one of the clearest of the clean models) and a clean Double Verb is especially telling., same with the ODS Clean. Anyone else test these yet?
 
bradlake said:
with just some cursory listening to freshly made stock setting presets using the above amps, I am forced to agree with Yek's assessment of the adverse tone changes in them. X/Ying between the TX Star(previously one of the clearest of the clean models) and a clean Double Verb is especially telling., same with the ODS Clean. Anyone else test these yet?
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Haven't tried the other 4 yet, but I can confirm my experience with the buzziness and the Nuclear Tone model is the same (reported a few pages back)

h.c.e. said:
Tried the beta out a little bit ago, and am getting some odd behavior with the Nuclear Tone model in Q8. Seemed really boxy and fuzzy to me, and I couldn't get it to clean up even with Input Drive turned way down to 1. Temporarily installed 7.0.2 just to make sure I wasn't going crazy and It's definitely cleaner with input drive: 4 then Q8beta is at input drive: 1 (reinstalled 8 beta, and am still getting the buzzy clipped sound on the model). Anyone else having a similar experience with the Nuclear Tone?
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Here's a quick recording I made last night comparing what I got with the Nuclear tone when I rolled back to q702 (first chord) vs q8beta (second chord). Third chord is q8 beta with the input drive rolled down to 1, and it's still more grit than the q702 with input drive on 4. Amp was reset before the q8 recordings, and only Input Drive and treble setting was reset. In the q702 clip would actually have slightly more speaker drive (default value of 0.5) than the q8 clip where speaker drive would have defaulted to 0.

 
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All that being said, the Plexi's sounded good to me in q8, def not an across the board complaint, i'm still running it, I'll just be skipping my nuclear tone based patches for now ;)
 
DigiSage said:
How many hours do you guys think before someone like @nojyeloot says some new parameter has "changed my life"? ala https://forum.fractalaudio.com/thre...-firmware-release.125625/page-17#post-1498260

I give it 0.25 to 3 hours.
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Not sure what the expected value is other than to say with 100% certainty that it will be infinitely sooner than the expected time frame in which someone exclaims "DigiSage's whiny sarcasm on the interwebz changed my life!". ;)

Thanks FAS for your epic and ongoing work. I think the silent majority of users appreciate the effort and support for both the big and small updates. I know it means a lot to me.

As for those reporting real issues, like Yek, hce, and others above that's a great thing for everyone and what public beta is all about. It is awesome to be able to provide this kind of feedback direct to a manufacturer/developer and have the head honcho directly plugged in.
 
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Appreciative and happy to wait as long as it takes for Yek's findings to be resolved and a final AX8 version.

For the first time in a long time, I feel no anticipation. I'm just really REALLY satisfied with how my AX8 sounds right now. Not much desire to mess with it!
 
I have a digital signal processing question...

If humans can only hear up to 20kHz, that means we cannot hear a discrete sample of sound that is shorter the 1/20000 = .0005 seconds = 0.5 milliseconds. What is the point of having a delay resolution that subdivides the smallest chunk of discrete time that the best humans can hear (0.5 milliseconds) into 500 finer subdvisions (.001ms cab delay resolution). I am aware of Nyquist frequency and the benefit of oversampling for signal processing to get higher resolution of the final output signal with reduced propagation of errors in intermediate calculations... but is this taking it past the point of diminishing returns? Wouldn't the output digital signal be identical when then the difference in cab block latency is below say 0.1 milliseconds?

If I'm wrong, I look forward to uncovering my ignorance and learning something new.
 
Initially I didn't like the update through my headphones but the increased definition really helps my patches cut through in a mix when playing through studio monitors. I feel like can hear every detail of my playing. So now that my ears have become accustomed to the changes I prefer version 8.00 to 7.02.
 
Just stopping by to say thank you for this update. The AC20 sounds incredible, and the HBE feels like it has a little more air to it without any harshness. Really pleased with how this FW is sounding.
 
6stringscott said:
I have a digital signal processing question...

If humans can only hear up to 20kHz, that means we cannot hear a discrete sample of sound that is shorter the 1/20000 = .0005 seconds = 0.5 milliseconds. What is the point of having a delay resolution that subdivides the smallest chunk of discrete time that the best humans can hear (0.5 milliseconds) into 500 finer subdvisions (.001ms cab delay resolution). I am aware of Nyquist frequency and the benefit of oversampling for signal processing to get higher resolution of the final output signal with reduced propagation of errors in intermediate calculations... but is this taking it past the point of diminishing returns? Wouldn't the output digital signal be identical when then the difference in cab block latency is below say 0.1 milliseconds?

If I'm wrong, I look forward to uncovering my ignorance and learning something new.
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The micro-delay in the cab block is not intended to be used as a delay effect but to change the voice of the mix of two IRs.
When you sum two identical signals one of which is delayed by a tiny amount you get comb-filtering and the delay time determines in which points of the frequency spectrum peaks and notches will be. Having a finer resolution for this is surely a useful thing because those peaks and notches change even with a small change in the delay time.

https://en.m.wikipedia.org/wiki/Comb_filter
 
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6stringscott said:
I have a digital signal processing question...

If humans can only hear up to 20kHz, that means we cannot hear a discrete sample of sound that is shorter the 1/20000 = .0005 seconds = 0.5 milliseconds. What is the point of having a delay resolution that subdivides the smallest chunk of discrete time that the best humans can hear (0.5 milliseconds) into 500 finer subdvisions (.001ms cab delay resolution).
Click to expand...
You got the math wrong. The 20 KHz corresponds to 0.05 milliseconds.

Small shifts in delay cause different phase cancellations/reinforcements that are frequency-dependent and audible.
 
6stringscott said:
I have a digital signal processing question...

If humans can only hear up to 20kHz, that means we cannot hear a discrete sample of sound that is shorter the 1/20000 = .0005 seconds = 0.5 milliseconds. What is the point of having a delay resolution that subdivides the smallest chunk of discrete time that the best humans can hear (0.5 milliseconds) into 500 finer subdvisions (.001ms cab delay resolution). I am aware of Nyquist frequency and the benefit of oversampling for signal processing to get higher resolution of the final output signal with reduced propagation of errors in intermediate calculations... but is this taking it past the point of diminishing returns? Wouldn't the output digital signal be identical when then the difference in cab block latency is below say 0.1 milliseconds?

If I'm wrong, I look forward to uncovering my ignorance and learning something new.
Click to expand...

16.7 Delay
 
yek said:
All right, here we go:

ODS Clean and TX Star Clean: kind of like the unit's in bypass mode, very dull.
Click to expand...

I`m using the TX Starclean with a Mesa 4x12 IR, a very "strong" cleansound for wide open chorus sounds.

In FW 8 beta it´s more open and clear, not so fat at singlenotes. But I like it - doesn´t sound "dull" to me?

As Justin York said - the AC 20 sounds great. Also with the LM IR.
 
Hubi said:
I`m using the TX Starclean with a Mesa 4x12 IR, a very "strong" cleansound for wide open chorus sounds.

In FW 8 beta it´s more open and clear, not so fat at singlenotes. But I like it - doesn´t sound "dull" to me?

As Justin York said - the AC 20 sounds great. Also with the LM IR.
Click to expand...
after a bit of tweaking, I was able to coax some nice tones out of the TX star, albeit with lower input gain settings.
 
Oops, the location of decimal points matter! Good catch. That said, my basic question remains...

At what point does the oversampling in internal calculations stop affecting the output digital signal?

I am curious to understand this, but not enough right now to overcome my math limitations, dealing with propagation of errors or sensitivity to input during convolution (whether with time-based integrals or discrete transforms to frequency domain). I am way too rusty for that kind of thinking, and it would involve more brain pain than the work I am trying to procrastinate right now.

In short: Rex's pithy response captures it best (short of a math geek wanting to explore it and report back for a vanishingly small audience of interested folks).
 
6stringscott said:
...my basic question remains...

At what point does the oversampling in internal calculations stop affecting the output digital signal?
Click to expand...
At the Nyquist rate for the highest frequency of interest. That's why Fractal gear only samples the inputs at 48 KHz, which is a bit more than twice the 20 KHz highest audible frequency.
 
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