Axe-Fx III Firmware Release Version 13.00 Public Beta

That would make no sense. The Quality parameter for reverbs is completely different from decreasing the authenticity of the drives. People always ask for the best possible modeling, and FAS always goes for the highest quality. Nothing comes for free.

I humbly disagree, even if it doesn't exactly map to reverb quality. The drives were "authentic" before (weren't they?) and now they are "more authentic" (accurate) with additional CPU cycles. Higher reverb quality gives more resolution/authenticity/accuracy to the reverberations while lower quality is less authentic/accurate.

Having a HIGH (new alg) vs. NORMAL (orig alg) quality selection would allow us to choose a tradeoff, especially for high CPU presets.
 
A Quality parameter is possible for the Drive block. I could reduce the oversample rate. I'd have to some extensive testing first to make sure it doesn't cause any problems.
And adding an option to also increase the oversampling, would it have any benefits eventually?

I guess it could on those models that add lots of distortion (fuzzes, muff, metal zone, etc..) and if there's a real benefit I wouldn't mind if the cpu usage will increase significantly.
 
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And adding an option to also increase the oversampling, would it have any benefits eventually?

I guess it could on those models that add lots of distortion (fuzzes, muff, metal zone, etc..) and if there's a real benefit I wouldn't mind if the cpu usage will increase significantly.
I don't know a ton about how oversampling fits in, but in general that type of thing reaches a law of diminshing retrurns. For just recording the sample rates audible difference between 24khz vs 48hz is pretty big, 48khz vs 96khz less, 96khz and 192khz, even less audible, and so on. That said, Axe-FX is already 48khz, so if there is oversampling from that point on, I wouldn't think very many levels of oversampling would make a huge difference. I think the difference in oversampling in this context may make more of a difference than it does when you record acoustic instruments with a mic into 48k vs 96k, but even then you can only oversample so far before it won't make a significant difference. There may be a difference in aliasing within the DSP processing up to a point, but not sure how far even that would go.
 
I don't know a ton about how oversampling fits in, but in general that type of thing reaches a law of diminishing returns.

There may be a difference in aliasing within the DSP processing up to a point, but not sure how far even that would go.

AFAIK, yes to both and also the noise floor (SNR) is affected.
 
I don't know a ton about how oversampling fits in, but in general that type of thing reaches a law of diminshing retrurns. For just recording the sample rates audible difference between 24khz vs 48hz is pretty big, 48khz vs 96khz less, 96khz and 192khz, even less audible, and so on. That said, Axe-FX is already 48khz, so if there is oversampling from that point on, I wouldn't think very many levels of oversampling would make a huge difference. I think the difference in oversampling in this context may make more of a difference than it does when you record acoustic instruments with a mic into 48k vs 96k, but even then you can only oversample so far before it won't make a significant difference. There may be a difference in aliasing within the DSP processing up to a point, but not sure how far even that would go.
The point is that the amount of oversampling needed to make the aliasing inaudible is directly proportional to the amount of distortion introduced.
If you use the drive block only as a clean booster then probably you would need no oversampling at all, but there are some drives that add a lot of distortion on their own (see the ones I mentioned in my previous post) and only Cliff knows if the current oversample rate is high enough to make the aliasing inaudible even at extreme settings.

I've done a lot of testing with VSTs and on most amp/distortion plugins I can definitely hear a difference even between the highest settings
 
The point is that the amount of oversampling needed to make the aliasing inaudible is directly proportional to the amount of distortion introduced.
It's filtering that removes the aliasing, not oversampling. The amount of distortion doesn't enter into it. Before your sampling rate gets to twice the Nyquist frequency, it's possible to design a filter that eliminates aliasing. Oversampling beyond that rate won't affect aliasing — but it can be used for noise reduction. And noise can be a concern with drive pedals.
 
Guys after trying the update I noticed my presets sound completely different now. In this case with the gain ones (Is like the low end sounds really boomy like when you crank the bass knob on a mark series amp for example). I run a dual amps set up and a fet boost before the amp but after fixing the values on the drive block again once the update was done still sounds really loose. What am I missing here ?
As a side note when I went back to the last firmware and the problem was fixed so I don’t know what it was ? Thanks !
 
And adding an option to also increase the oversampling, would it have any benefits eventually?
Not IMO. I always select the oversampling to give a "Signal-to-Aliasing Ratio" of at least 60 dB. Any more than that is a waste of CPU.
It's filtering that removes the aliasing, not oversampling. The amount of distortion doesn't enter into it. Before your sampling rate gets to twice the Nyquist frequency, it's possible to design a filter that eliminates aliasing. Oversampling beyond that rate won't affect aliasing — but it can be used for noise reduction. And noise can be a concern with drive pedals.
Wrong and wrong. It's oversampling that removes aliasing and the amount of distortion factors into it.
 
Guys after trying the update I noticed my presets sound completely different now. In this case with the gain ones (Is like the low end sounds really boomy like when you crank the bass knob on a mark series amp for example). I run a dual amps set up and a fet boost before the amp but after fixing the values on the drive block again once the update was done still sounds really loose. What am I missing here ?
As a side note when I went back to the last firmware and the problem was fixed so I don’t know what it was ? Thanks !
I don't now, there were no changes to the amp block.
 
It's filtering that removes the aliasing, not oversampling. The amount of distortion doesn't enter into it. Before your sampling rate gets to twice the Nyquist frequency, it's possible to design a filter that eliminates aliasing. Oversampling beyond that rate won't affect aliasing — but it can be used for noise reduction. And noise can be a concern with drive pedals.
That's incorrect. Filtering doesn't remove aliasing, it just removes content above the Nyquist frequency before an A/D conversion or downsampling. Aliasing can be anywhere below the Nyquist frequency (even as low as 20 Hz depending on the amount of reflections) so a high-cut filter can't remove it.

Oversampling on the other hand pushes the nyquist frequency further from the audible range and you need a lot more harmonics (aka distortion) to achieve the same amount of aliasing
 
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I humbly disagree, even if it doesn't exactly map to reverb quality. The drives were "authentic" before (weren't they?) and now they are "more authentic" (accurate) with additional CPU cycles. Higher reverb quality gives more resolution/authenticity/accuracy to the reverberations while lower quality is less authentic/accurate.

Having a HIGH (new alg) vs. NORMAL (orig alg) quality selection would allow us to choose a tradeoff, especially for high CPU presets.

Reading Cliff’s reply, I stand corrected.
 
Guys after trying the update I noticed my presets sound completely different now. In this case with the gain ones (Is like the low end sounds really boomy like when you crank the bass knob on a mark series amp for example). I run a dual amps set up and a fet boost before the amp but after fixing the values on the drive block again once the update was done still sounds really loose. What am I missing here ?
As a side note when I went back to the last firmware and the problem was fixed so I don’t know what it was ? Thanks !
I came from the previous build to the beta and nothing changed tone wise except loving the drive block. I'm even changing my presets from all amp for my distortion to clean amps and pedal. Seems more fluid and better tone.
 
Guys after trying the update I noticed my presets sound completely different now. In this case with the gain ones (Is like the low end sounds really boomy like when you crank the bass knob on a mark series amp for example). I run a dual amps set up and a fet boost before the amp but after fixing the values on the drive block again once the update was done still sounds really loose. What am I missing here ?
As a side note when I went back to the last firmware and the problem was fixed so I don’t know what it was ? Thanks !
What firmware where you coming from? Can you roll back and follow https://forum.fractalaudio.com/threads/how-i-maintain-sounds-across-many-firmware-revisions.164335/ to make a reference point before rolling forward?
 
I wondered whether the drive in the amps was changed any - I actually thought the tweeds were really collapsing in a (better) way than they had before when you really crank the gain - like more realistically...but it could totally be placebo...I'm not sure I've cranked them that much in a while, so maybe there's been a change in a previous update. Regardless, it's a sound you can't get with other units IYAM.
 
since installing this, most of my block libraries seem to have disappeared, so I cannot change amps, etc in any of my settings.

Any ideas? Help...
 
Wrong and wrong. It's oversampling that removes aliasing and the amount of distortion factors into it.
Okay, I guess I need to revise my understanding.

The way I understand it, Nyquist tells us that in order to perfectly reconstruct a signal, we need a sample rate that’s at least twice the highest frequency in the signal. And we need to leave enough room between that Nyquist rate and our sample rate to build an artifact-free filter that will knock down higher, alias-causing frequencies that would otherwise show up at more than half of our sample rate. IIRC, @FractalAudio posted that ideally, that “magic number” is somewhere around 62 KHz — that would accommodate a 20 KHz bandwidth, which is the audible spectrum, whether there’s distortion or not.

As I understand it, further oversampling can be used for noise reduction via sample averaging, where uncorrelated signal components (noise) would not add as strongly as the desired correlated signal.


I am open to re-education.
 
Okay, I guess I need to revise my understanding.

The way I understand it, Nyquist tells us that in order to perfectly reconstruct a signal, we need a sample rate that’s at least twice the highest frequency in the signal. And we need to leave enough room between that Nyquist rate and our sample rate to build an artifact-free filter that will knock down higher, alias-causing frequencies that would otherwise show up at more than half of our sample rate. IIRC, @FractalAudio posted that ideally, that “magic number” is somewhere around 62 KHz — that would accommodate a 20 KHz bandwidth, which is the audible spectrum, whether there’s distortion or not.

As I understand it, further oversampling can be used for noise reduction via sample averaging, where uncorrelated signal components (noise) would not add as strongly as the desired correlated signal.


I am open to re-education.
You're talking about aliasing wrt ADCs, but the thread's talking about aliasing wrt non-linear signal processing.
Look at the webpage I linked in the previous page, the gist is all in there.

Also, relevant post from the past from the man: https://www.thegearpage.net/board/index.php?posts/26507338/
 
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