Axe-Fx II "Quantum" Rev 8.00 Public Beta

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Quick story for perspective.
My band's Keyboard player recently bought a new and pretty expensive keyboard (a well known & respected brand), and for some reason it was my job to get the thing connected to our PA, I saw it had two 1/4 inch jack line outs on the rear panel, for connection to an external source. I thought i'd check the manual to see if these were balanced or not. The manual says nothing about it. As if the outs were just an afterthought anyway. Also, there appears to be no expression pedal socket on it either, just pedal sockets which are for the brand's own multi piano pedal only. All this before we even get to (lack of) updates for the product.

It's instantly clear that the manufacturer has built this thing up to a price point, so it can compete with other products on the market. Any appointments deemed superfluous are binned in favour of the bare minimum requirements.
Now, that's when it becomes instantly clear how spoiled and very lucky us FAS users are. If we want to route or connect the unit a certain way, we can. It's all there if we need it, plus a whole lot more.
Then we get these updates, and okay, some like them, others don't. You don't have to do the updates, but the option is there which is very cool.

I have bought keyboards assuming there would be an expression jack OR pedal jack. I mean, how unlikely it would be that these jacks would not be included. But they sometimes aren't. This is a great thing to be wary of if you're a trusting buyer and are shopping for one. My Korg Micro Key 61 has no jacks - I sometimes use multiple keyboards on the same MIDI channel (only one of my keyboards, the Impulse 61, has jacks.) There may be a workaround for people who want to split the expression CC to multiple instruments within a softsynth environment. So far I haven't done that, but it should be possible using a MIDI utility. (I mean that I could, for instance, octave shift one of the keyboards and be able to use the pedal for a range beyond the 61 keys, or switch from regular sized keys to small ones if I simply wanted to play some stretchy piano chords that are impossible on a regular-key-size keyboard.
 
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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.
Would computing these small values still re-create some cancellation effects that change audible frequencies, and Lord only knows might be considered useful by somebody??? A cancellation of a low frequency by a high frequency would seem to me to effect audible frequencies. for instance, 22k may be inaudible, but we may hear the cancellation as influencing the 11k or 5.5k, or 2.75k frequencies. I could be wrong.

Not sure, but interesting point you make and I'd love to understand it better.
 
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I dialed it in from scratch and getting better results now.
My main gigging tones are ODS Clean 100-based now. I'm updating tonight. Appreciate the early heads up and I'll let you know how it goes for me.

Edit: no problem tweaking post-update on the ODS Clean 100. I ended up dropping back input drive and bumping up master volume. Some very slight tweaks to BMT and the presence and everything was sounding right again.
 
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Amazeballs!!!

Actually I haven't updated yet. I just learned that word from some young people and wanted to use it somewhere. But I'm with y'all in spirit.
 
I downloaded 8beta and it keeps crashing axe edit so I rolled back, I'm guessing it's a corrupt download on my end (it happens). Will try a fresh download tomorrow. Thanks FAS as always. Can't wait to try.

This might not be the cause of your crashes, but FWIW, Axe Edit updates come after the full non-beta versions are released, as applicable.
 
ODS-100 Clean sounds fine here. In fact it sounds amazing. Maybe the best clean model IMO. You have to really crank the MV though (just like the real amp).

Is a cranked MV the default ? Sounds like it should be.
....This won't get answered. ;)
 
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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.
I don't know how, but this is an oversimplification or a mistake in your calculation. I don't think converting cycles to seconds works as a comparison in this anyway. The speed of sound is somewhat constant at around 1125.33 feet per second. There is some variation due to frequency (and other factors) but it's no where near a one for one. 20Khz doesn't travel anywhere near 1000 times faster than 20Hz, otherwise we would never be able to hear properly. And physics wouldn't work. And we would all die in a cosmic cataclysm. But I digress.

Try this-- Use headphones to monitor. Open a preset with a stereo cab block, identical cabs left and right. Pan both 100% in opposite directions. Now add delay to one side. Immediately you will notice a difference in sound, so your brain is on some level perceiving the timing difference at levels below 0.1ms. At about 0.1ms you will begin to perceive a shift in volume away from the side that you have added the delay to. The more you turn it up, the more it shifts away. Now take one headphone off at a time and try listening with only one ear, then the other. The levels should then seem identical, because in reality, they are identical. When you listen with both, your brain is perceiving the timing difference and telling you that one side is closer, hence louder even though the signal levels are identical.

I'm no audiology expert, but I believe this has something to do with "binocular hearing" for lack of a better word. It's probably how some blind people are able to teach themselves to use echolocation when their hearing is good enough. Whatever the case even if you can't hear a frequency it can still have an impact on other frequencies in the mix.
 
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I think it's fantastic that Cliff shares each beta with the community, players pickup any unusual characteristics and report them. We all benefit from this. The fact that FAS are so willing to do this is a real credit to them and what clearly sets them apart from the competition.
 
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.

Checked all the other amps and they sound fine here, great in fact.
 
The FAS Class-A is simply stunning! The Nuclear Tone itself - Whoa! Really liking the Plexi 100w Normal - a bit more top end and clarity.
 
This might not be the cause of your crashes, but FWIW, Axe Edit updates come after the full non-beta versions are released, as applicable.

Ya, I know (good looking out though, bro).

Remedy: Reinstalled Axe Edit. Reinstalled 8b. Rebooted all. Logged on, started Axe Edit (read definitions) and we're in business.
 
If you havent tried the cabs slightly out of phase before, this is my take on it. I cant hear anything different until the delay offset is at least 0.03ms whereas 0.06ms is too much, the sound becomes too dark. The useable range of the parameter used to be just a few wheel clicks. So i would generally just set it at 0.04 or not use it at all. By increasing the resolution of the parameter the useable range is a lot more accessible and able to be fine tuned. Thanks FAS.

Would computing these small values still re-create some cancellation effects that change audible frequencies, and Lord only knows might be considered useful by somebody??? A cancellation of a low frequency by a high frequency would seem to me to effect audible frequencies. for instance, 22k may be inaudible, but we may hear the cancellation as influencing the 11k or 5.5k, or 2.75k frequencies. I could be wrong.

Not sure, but interesting point you make and I'd love to understand it better.
 
OK, updated and back to the hoary old problem of all the levels being far too hot again, it's not me before anyone says anything, I've been in the Modelling "scene" since the year dot and trust me that this is a FW bug, reloaded the amps etc etc ,all the usual gubbins so I'm reinstalling 7.02 until the final release
 
Wow, been holding on to that one a while, eh? Since I rarely post, I'll give you the benefit of the doubt that you didn't grasp my exaggeration meant in a positive way. Ya, motor drive is amazing and changed how I created presets and shifted my standards. Happier now? Now, move along and enjoy your free firmware, and pls try to keep things civil.

That being said, to contribute to the thread: I downloaded 8beta and it keeps crashing axe edit so I rolled back, I'm guessing it's a corrupt download on my end (it happens). Will try a fresh download tomorrow. Thanks FAS as always. Can't wait to try.

I think it was pretty obvious to most normal people that you were exaggerating. And thanks FAS, if I am allowed to say that without the thanks police coming after me.
 
I know about the noise reduction meter. But what about the drive settings...

The actual value for a particular speaker is all over the map. The time constant is proportional to the mass and the thermal resistance of the voice coil. Both these values can vary widely. 200 ms is based on a typical theta of 1 degree C/W and a mass of 10g. I don't really feel like destroying all my nice vintage speakers to measure the voice coil mass.



...and how much gain reduction is desired?
See the release notes. :)
 
First time ever I'm rolling back to a prior firmware from this Beta. I like what is happening on some amps, like the Ford ODS; less congested sounding, and the presence and treble controls interact really nicely.

But I have one preset I use a lot based on the Tweed Deluxe. Where before it sounded killer on Eagles tunes, now it sounds like there is a piece of paper on the speaker cone, or a torn cone. Buzz over the top of everything that fades pretty quickly, leaving the tone that I was after.

I like where you're going with this, but I think there are some bugs to iron out before I'm ready.
 
Would computing these small values still re-create some cancellation effects that change audible frequencies, and Lord only knows might be considered useful by somebody???
It's not just cancellation. It's cancellation and reinforcement. That will make some frequencies stronger and some quieter, by varying amounts. That whole picture changes as you change the delay time.

The result is a series of notches in the frequency response. It's called a comb filter because, if you plot the frequency response on a graph, those notches look like a comb. If you make the delay time sweep back and forth, the notches move up and down in frequency, and you call it a phase shifter. :)


Yes, the new, tiny resolution corresponds to frequencies that are above human hearing—if you tweak it down that low. But that's not what it's for. It's for making tiny adjustments in the amount of delay that you've dialed in, to get subtle, audible changes in the comb filtering.
 
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