1 Available on which Fractal Audio products
2 X/Y and channels
3 Impulse responses (IRs)
4 Enabling and disabling cabinet modeling
5 Cab block in the Axe-Fx III
6 Position of the Cab block
7 Mono and stereo cabinets
8 CPU usage
9 Choosing cabs
10 Scratch-Pads
11 See which cabs are being used in presets
12 Substituting an IR with a Tone Match block
13 Preset-Cab bundle
14 Recording 4 different cabinet signals
15 Parameters
16 About microphones
17 About guitar speakers
18 Tutorials and more information

H cabs.png

Available on which Fractal Audio products

Axe-Fx III: 2x.
Axe-Fx II: 2x.
AX8: 1x.
FX8: no.

X/Y and channels

Axe-Fx III: 4 channels.
Axe-Fx II: yes.
AX8: yes.
FX8: n/a.

Impulse responses (IRs)

What is an impulse response

For recreating the tonal characteristics of speaker cabs, the Cab block relies on Impulse Responses (IRs).

An IR is a collection of data representing sound measurements taken from a speaker cabinet or system. A test signal is played through the actual speaker, recorded, and used to generate a profile. The profile (IR) can then be used by an IR-loader to recreate the sound of the speaker.

The terms "cab", "user cab" and "IR" are often mixed up.

The Axe-Fx II also uses IRs to reproduce the characteristics of specific microphones.

Axe-Fx III Owner's Manual:

"Fractal Audio Systems speaker simulation technology is incredibly accurate, yet some listeners find the sound of IRs unfamiliar at first. This is because impulse responses typically recreate the sound of “close miking.” When you mic a guitar speaker, the mic “hears” something very different to what you might hear. Our ears are by definition “neutral” whereas a mic has distinct “color.” We typically listen at a distance (and speaker tone is very different as we move around) while a mic is inches away and stationary, focusing on the desirable sound at a specific spot.
As guitarists, we are accustomed to the sound of a speaker “in the room,” but this is not what our audiences hear. For recording and performing, the close mic’d sound is essentially a universal standard. THIS is the sound that the Cab block is designed to reproduce, and this explains why not only guitarists, but recording and front-of-house engineers have embraced its use. (Of course our Amp models can also be used with a traditional guitar speaker as demonstrated in many of the rig designs in Section 1: Setting Up). If you are new to using mics on a guitar amp, you will find the Cab block is a fantastic way to learn more. To get started, listen to single IRs, or explore the factory presets which combine several at once. For almost a century, artists, producers and engineers have honed the craft of placing or blending mics to achieve a desired tone. Many classic techniques are easy to recreate. Try a tried-and-true “recipe” blending one “bottomy” and one “bright” mic, or try something totally original.
The factory content includes dozens of speakers with multiple different mics in different positions. You may also enjoy “Mix” IRs by Fractal Audio or 3rd-parties, which bring a producer’s experience to you in a single IR. In any case, recognize that the sound of IRs is the very sound of the speakers and mics they capture."

"An IR stands for "Impulse Response". In mathematical terms it is the time response of a system to a Dirac delta function (also known as an impulse). An IR can be used directly as the coefficients for an FIR (Finite Impulse Response) filter. In the modeling world IRs are obtained from real speakers and when processed using an FIR filter produce extremely accurate results. In essence an IR is a "sample" of the speaker and microphone and uses very similar principles. However the quality of any IR is subject to the talents of the individual(s) capturing the IR. Mic placement, preamp choice, etc., etc. are important as you are essentially recording the speaker. In the old days modelers used EQ to emulate speaker response but I don't think there are many left that still use that technique. So the quality of the IR is really the issue here. The original Axe-Fx pioneered this technology which has since become almost ubiquitous." source

UltraRes, HiRes, Normal Resolution

Fractal Audio devices and software support IRs of various lengths, measured in number of samples (points) and milliseconds.

Normal: 1024 samples, 20 ms.
You can often use Normal Res without a noticeable impact on the tone, compared to Hi Res. The Cabinet block's Stereo mode supports two Normal Res IRs.

HiRes: 2040 samples, 40 ms.
HiRes processing requires more CPU power than mono or stereo Normal Res, and also more than UltraRes.

UltraRes: up to 8000 samples, 170 ms.
Ultra-Res IR processing requires more CPU power than mono or stereo Normal Res but less than HiRes (about 4% less CPU usage AND higher resolution)! The Cab block's Stereo Ultra-Res mode supports two Ultra-Res IRs. Non-Ultra-Res IRs will be processed in Stereo UltraRes mode as Normal Res. If one IR is Ultra-Res and the other not, then the Ultra-Res IR will still processed as UR and the other as Normal in stereo mode.

Ultra-Res speaker IR processing is a Fractal Audio proprietary technique that enhances the spectral resolution of an IR without adding CPU burden or storage requirements.

Note: Ultra-Res IRs do not support size warping. Therefore, the Speaker Size parameter is unavailable in Ultra-Res mode.

Ultra-Res lRs are displayed in italics in the editors.

(source) "The problem with conventional IRs is that they are too short to capture the detail in the low frequencies. There are those that maintain 20 ms is the maximum length you need to fully replicate the speaker. This would be about 1000 samples at 48 kHz. I disagree with this as I have many IRs here that exhibit significant energy beyond 20 ms. I believe the room has some influence as the low-frequency modes of the room will impact the resulting sound. The amount of this impact depends on the room, the mics, distance, etc., etc. Or perhaps certain speakers have particularly high Qs in the low frequencies. Regardless, it is my opinion that you need IRs much longer than 20 ms to fully capture the "mic'd amp in the studio" sound. My tests show that IRs of 8000 samples are required to fully capture the low-frequency detail. Unfortunately to process an 8K IR in real-time require copious processing power... Fortunately I have developed "Ultra-Res" cabinet modeling. Ultra-Res cabinet modeling provides the frequency detail of a very long IR with little or no added processing power requirements. The following image depicts the response of Ultra-Res cabinet IR processing: ..." (see thread)
"Existing IRs will still be processed as usual. Ultra-Res IRs will be tagged as such which will indicate to the processor to use the new processing algorithms. Note that Ultra-Res IR data is not conventional IR data."
"The frequency resolution of an IR is the sample rate divided by the number of samples in the IR. The window function has nothing to do with frequency resolution (except for making it even less). So a 1K IR at 48 kHz sample rate has a frequency resolution of roughly 48 Hz. If a speaker has a resonance (formant) at, say 80 Hz with a Q of, say, 3.0, then 48 Hz is insufficient to capture that resonance accurately. You need a frequency resolution of several Hz to accurately recreate that resonance. I chose 80 Hz and a Q of 3 because that's what that response looks like. The Q could even be higher than that. It doesn't take much mental energy to realize that if you have a narrow formant at a low frequency then you need fine frequency resolution to reproduce that. An 80 Hz formant with a Q of 3 only spans about 25 Hz. Obviously a frequency resolution of 48 Hz is not going to be able to reproduce that. Windowing only smooths the response even more. This is basic FFT theory. The less time-domain information you have, the less frequency domain information you have and vice-versa. This is the uncertainty principle. I always window IRs with a Hann window."
"Another way to look at it is to think in terms of formants. That particular speaker has a pronounced 80 Hz formant. It takes well over 100 ms for the energy of that formant to decay to the point of imperceptibility. Obviously a 20 ms IR can't reproduce an event that occurs for over 100 ms. Here is a zoom of the original non-minimum-phase IR (IOW raw time response)... (see thread). You can clearly see the 80 Hz formant. There are some room reflections but they are very small. The 80 Hz formant starts well before any reflections. It's obviously a high-Q resonance as it rings for quite a while. The higher the Q, the longer it takes to decay."
"Here's another example. (see thread) This is one of the new OwnHammer IRs. The IR is OwnHammer_412_MAR-CB_D-120_SS_RBN-121. These IRs are 100 ms long (4800 samples). I windowed the original IR to 4K to prove a point. The blue trace is the original IR (windowed to 4K samples). The green trace is the "typical" 20 ms IR (windowed to 1K samples). The red trace is the Ultra-Res version."
"The problem is that human perception is logarithmic and IRs are a linear process. 48 Hz resolution is way more than necessary at, say, a few kHz but not nearly enough at low frequencies. The brute force solution is to use very long IRs, 8K or more. Ultra-Res solves this in a novel way that uses little to no extra processing power and no additional latency."
"Normalization is your friend. Rectangular windows are simply truncation and are generally regarded as bad practice due to extremely high sidelobe levels. The choice of window is subjective. I actually use my own custom window that is not really a Hann window but that's proprietary information. My window preserves more frequency detail while still suppressing Gibbs phenomenon. Windowing trades off frequency resolution for sidelobe suppression. My window is optimized for the unique statistics of IRs. For a random process I tend towards Bessel-Kaiser windows. IRs have unique statistics that aren't addressed by any of the standard textbook windows."
"It is desired that the IR be 8K samples or more."
"Let me state these points:

We don't record guitar amps in airplane hangers or anechoic chambers. We record them in studios.

When we record a guitar amp we carefully set the amp up in the studio to get the best sound "on tape". This involves moving the amp around, placing gobos, etc. When we collected the Producer's Packs IRs we spent hours arranging the amps/speakers, mics and gobos and playing through the amp and readjusting until we were satisfied. This also included adjusting the preamps and mixing board. In one studio we found that we got the best tone raising the cabs off the floor by a couple feet, orienting them towards a particular wall and placing gobos behind (this was the engineer's standard recording arrangement).

At this point our objective of the IR is to capture the sound of that amp/speaker at that position in the room, with the gobos, mics, preamps, etc., etc. The goal is not to capture the raw sound of the amp/speaker in an airplane hanger or outside using a ground-plane measurement and measurement mics. That might be someone else's goal but it is not ours. IOW our goal is to treat the cab, mics, preamps, room, etc. as a whole, as a good engineer/producer would.

Subsequent analysis of the data shows that there is significant energy out to 100ms and even beyond. However there is little energy beyond 200 ms or so (as it should be in a well-designed studio). This observation was the catalyst for the Ultra-Res algorithm. There are other observations about the statistics of the data that I cannot disclose.

Some cabinets displayed noticeable resonances at low frequencies. Others did not. The frequency of these resonances were not consistent and, not coincidentally, matched the measured resonance of the impedance sweep. It is a logical conclusion, therefore, that the resonance was NOT caused by the room but by the speaker/cabinet combination. Furthermore a plot of the group delay for the raw data showed that the delay of the resonance was too short to be a room mode. Regardless, whether the resonance is from the speaker or room or mics or preamps is irrelevant. All we care about is recreating the sound of that speaker as it would be recorded as accurately as possible.

Truncating an IR destroys information by definition. We don't care where the information comes from, be it the speaker or the room or the mics or the preamps. We want all the information. If a plot of the frequency response of a truncated IR differs considerably from the non-truncated version then we have lost information and concomitant accuracy.

NO ONE producing commercial IRs records them in an airplane hanger, for obvious reasons. The best ones are done in a studio using the same technique we used for the Producer's Packs: setting up the cab, adjusting the position, mics, preamps, etc. and playing through the amp/cab and readjusting until the best tone is achieved. The new OwnHammer IRs are an example of this. Many, if not all, of those IRs exhibit significant energy to 100 ms (and likely beyond but the data stops at 100 ms). Truncating them to 20 ms destroys vital information. You can argue the semantics all day long. I've compared truncated and non-truncated and the difference is clearly audible. It is especially noticeable when chugging power chords. You can hear the resonance. It goes "bonggggggg" as opposed to "thuk". Most importantly it sounds "better" IMO.

Ultra-Res is an algorithm that markedly increases accuracy. It gives the frequency resolution of a 200ms IR without additional processing overhead and no added latency."

"Ultra-Res is especially powerful in Tone Matching applications, particularly real-time matches and was another impetus behind the development."
"The myopic only see the IR as a capture of the speaker's "unadulterated" response. As I stated before I believe the future is treating IRs as capturing the entire recording chain including mics, preamps, etc. and have pushing in that direction. We have already seen the fruits of this labor in the Producer Pack and OwnHammer V2 IRs. We used mainly PP and OH IRs at Axe-Fest this weekend and the results were stellar. Andy Wood's tone was among the best guitar tones I've ever heard live and we dialed it up in 10 minutes under far less than ideal conditions. It consisted of the Two Rock amp model and the EV 12L Mix IR. When you include more than the speaker response in the IR you can have low-frequency resonances that persist for tens of milliseconds or more. Truncating an IR destroys this LF information. In many cases this LF information loss would probably not be perceptible. In other cases, from experience, it can be extremely noticeable. The bottom line is that you can always remove the information if you don't want it but you can't add back what isn't there."
"Let me phrase this another way. An IR can consist of the "raw" speaker response plus none, one, some or all of the following: mic, preamp, room, power amp (e.g. you want to capture the response of a tube amp driving the speaker), etc. If you only care about the raw response then a short IR is all that is required. However if you want any of the other elements as part of the IR then a longer IR may be necessary. Ultra-Res gives you the OPTION of processing longer IRs."

"If the .wav is only 40ms long there is no sense in converting to Ultra-Res as you won't gain anything. Over 80 ms is desirable. The maximum length supported is 170 ms or so. Anything longer than that is truncated to 170 ms." source
"To get the optimum results the length should be 170 ms or more. As the length gets shorter you'll lose information. However there may not be any information to lose. It all depends on the IR. I've seen long IRs where only the first 100 ms or so is actual information and the rest is silence. OTOH I've seen 100 ms IRs where there is obviously more information but it got truncated. You lose nothing with Ultra-Res except the ability to change the size of the cabinet. You gain better sound and less CPU." "You can't mix Ultra-Res IRs as the data is not compatible. However... we foresaw that and the UltraRes conversion process produces two files: a .ir file and a .syx file. The .ir file is the raw IR data that can be imported into CabLab for mixing purposes. So CabLab can take .wav, non-Ultra-Res .syx and .ir files as input to the mixer section and product Ultra-Res .syx files." "The .ir files are included with our cabinet packs. We will not be offering .wav files. If you have the .wav file you don't need the .ir file. A .ir file can ONLY be used with CabLab. If you use the Axe-Fx II to capture IRs it will only generate .ir and/or .syx files. No .wav files are generated. The resulting data can only be used on Fractal Audio products." source
"It depends on the IR. Ultra-Res improves low-frequency resolution. It is very apparent with some IRs and virtually inaudible with others. It all depends on the low-frequency formants in the original IR. If there are significant, high-Q formants Ultra-Res will preserve those whereas conventional, short IRs will not. Audibility also varies with the amp being used. The difference is more audible with high gain as this will excite the formants more. Low-frequency formants vary with the type of cabinet and speaker. Some cabinets have a smooth low frequency response. Others have prominent formants. The mic also has an impact. Some mics will accentuate the formants. The room also contributes if it has strong LF modes. Furthermore some people like to capture an IR using a tube power amp. In this case you WILL get a significant formant at the low-frequency resonance of the speaker. A conventional IR will not capture that as the Q of the formant will exceed the resolution of the IR. Ultra-Res will capture that formant as Ultra-Res has 8 times the low-frequency resolution. Those who claim they can't hear a difference are correct. They can't. It's nothing to be ashamed of. But because they can't doesn't mean others also cannot. I can clearly hear the difference but I've trained myself on what to listen for. I vastly prefer Ultra-Res and only use Ultra-Res IRs in my personal patches (aside from the TV Mix, which is just a magical IR)." source
"The length of the sweep only determines the signal-to-noise ratio. If the room is completely silent the sweep can be infinitely short (an impulse). To overcome ambient noise you need more energy in the applied stimulus. With an impulse you can only increase the power so much before the amplifier or the speaker or the mic or the preamp, etc. distort. However, if you spread that power out over a longer period of time you can increase the energy and therefore increase the SNR. Think of it this way: a 1 ms pulse at 1000W has the same energy as a 1 second pulse at 1W. Now you can't just put a 1 second pulse into a system because the pulse has little frequency content. A 1 second sweep over the band of interest allows the transfer function (IR) of the system to be obtained via deconvolution. There are other signals you can use like pseudo-noise and MLS sequences but a "chirp" has the best characteristics. In the early days of room IR capturing they used impulses generated by popping a ballon, firing a starting pistol or clapping two boards together. The results were poor due to low SNR. This lead to the development of signals that have higher energy. To get the IR of a room long sweeps are typically used because there is a lot of ambient noise and the "returned signal" is weak (the reverb portion of the response is very low compared to the direct signal). When close mic'ing a speaker the ambient noise is low and the signal strength is very high so a short sweep is adequate. In fact you could probably get away with 100 ms or less in a studio environment." source
"I've never seen a cabinet IR (and I've examined thousands) that has any significant content beyond 150 ms or so. Most cab IRs are under 40 ms. The exception to this would be a "room IR" where the mic is very far from the speaker and the room is significantly reverberant. But one wouldn't normally use that as the primary tone, instead to add a little ambience to the tone and the loss of information would be imperceptible in context. Modeling products typically use IR lengths of 1K samples as this covers 90% of IRs ever captured. We support 2K and Ultra-Res (which is equivalent to 8K) which covers 99% of IRs. The amount of CPU power required to process an IR is proportional to the length of the IR. To support a 500 ms IR (24,000 samples) would require over ten times the CPU power of a 2K sample IR. It also requires over ten times the memory for storage. Given that that vast majority of IRs do not have any information beyond 40 ms it is wasteful of CPU and memory resources to support IRs longer than 2K." source
"The length of time you hold a chord is irrelevant. The impulse response of a speaker cab is typically much less than 100 ms. Only when there is significant room reflections is the length greater. Then you get into the whole argument of whether the IR should contain any room information." source

(about Tone Matching and Ultra-Res) "In Realtime mode the raw internal IR length is 8K which you can dump." source "You can export the Tone Match to CabLab and create and Ultra-Res IR." source

(source)
"I'm a huge advocate of longer IRs. In fact I think I was the first to advocate it despite all the naysayers. I pushed OwnHammer (and others) to increase their IR lengths and they were the only ones who acted on that advice (so far, maybe the other guys will start to follow suit). Ultra-Res was born out of the desire for longer IRs.
For recording you don't need to use the cab block in the Axe-Fx though. Record the raw amp sound and then "re-cab" it later. This way you can try different cabs. Cab-Lab is great for this. Cab-Lab does not do Ultra-Res processing. It creates Ultra-Res files for the Axe-Fx but it does all processing at the full IR length up to 8K samples. You can use other convolution plug-ins as well.
The reason for Ultra-Res is that long IRs have several drawbacks:

They require lots of storage space. Not an issue on a computer but on a hardware product that means expensive non-volatile memory.

They require lots of processing power if you don't want any latency. On a computer it doesn't matter since latency is a non-factor if you are processing prerecorded tracks. On a hardware product we must have zero latency.

So Ultra-Res was devised as a way to exploit the statistics of the data to give the benefits of longer IRs without the usual hardware drawbacks.
In my tests I've found that 8K samples (170 ms) is more than enough. I think 500 ms (24K samples) is overkill and if an IR has significant energy out that far then it has too much room in it. The speaker and cab itself are never more than 100 ms, usually much less. Anything beyond that is the room. I personally don't like IRs with lots of room in them. A little bit of early reflections are nice and make things sound less direct but too much room makes the sound get lost in the mix.
There's no meaningful data beyond 150 ms and if there is, it's the room and you don't want that much room."

(Ultra-Res 2.0) "No big deal, just some improved processing algorithms. The UltraRes cabs in Quantum 2.0 were all reprocessed with UltraRes". source. UltraRes 2.0 is the next level of evolution for our patent-pending speaker simulation technology, with even greater accuracy than the original version. UltraRes 2.0 cab files are backwards compatible with previous Axe-Fx and AX8 firmwares supporting UltraRes 1.0. source

"1" from the speaker is the near field. The response of a speaker in the near field is very different than the response in the far field. In the near field the response changes (drastically) across the face of the transducer. Even moving the mic a fraction of an inch will result in a very different sound. 10 ft. from the speaker is the far field and the response changes smoothly as you move across the field. If the near field were the same as the far field then the sound wouldn't change as you moved the microphone and you could place the microphone anywhere on the face of the speaker. Anyone who has mic'd a speaker knows that this isn't the case." source

Sampling rate

IRs are tied to the sample rate of the hardware. This is 48 kHz (fixed) in Fractal Audio devices.

Differences between IRs for Axe-Fx II Mark I - Mark II - XL - XL+

"The XL has a different sysex ID and therefore requires different cab files. The only difference in the files is the sysex ID.” source

Near-field and far-field

Most speaker IRs represent the tone of a speaker that was recorded with the microphone close to the speaker, aka "near-field" or "close-miked".

"Far-field" IRs represent the sound of a speaker that was captured at a longer distance. There are a couple of far-field IRs among the stock cabs, created by Jay Mitchell ("JM"). These IRs are better suited for recreating the "amp-in-the-room" sound of a traditional guitar speaker.

Min Phase and Auto Trim

Read this: Min Phase and Auto Trim.

Creating your own IRs with IR Capture

Read this: IR Capture.

IRs versus Tone Matching

"Tone Matching is a nifty feature and certainly useful but you'll get far more satisfaction by concentrating on capturing good IRs. The single most important aspect of recording guitar amps is micing the amp. Therefore the single most important aspect of using your Axe-Fx is the IR. People are too hung up on "matching" or "profiling" an amp but fail to realize that when you are doing that you are basically capturing an IR. If you capture the IR separately now you have an IR that is fully separated from the amp and therefore can be used with all models. Matching and profiling cannot mathematically separate the amp's frequency response from the cabinet frequency response. Once you do this you'll be surprised at how accurate the amp models are. I do this all the time and find Tone Matching is unnecessary now (in fact many of the amp models have had their built-in matching data removed in the latest firmware). Any differences between the model and the real amp are so minuscule as to be immaterial. A little tweak of the tone stack or EQ is usually enough to remove and differences. Besides, once you get into mixing you'll realize that you'll be applying EQ anyways so tiny differences in EQ are irrelevant. Moving the mic just a small amount drastically changes the sound. The best producers have mastered micing. You can only fix so much via EQ since EQ is essentially painting with a broad brush where mic technique is akin to using a fine-point brush." source

IRs of acoustic sounds

To emulate acoustic instruments (acoustic guitar, cello, violin etc.), an IR of an acoustic body can help. You can find some on Axe-Change.

Acoustic sounds benefit from long IRs, so Ultra-Res IRs are preferred.

Future developments

"The Axe-Fx III (and II) actually capture the Volterra kernels when doing an IR capture (it's hidden in the firmware for possible future use). I've studied dozens upon dozens of them and the kernels above first order (the first order kernel is the linear IR) are so small as to be inaudible. The distortion from an amp is orders of magnitude greater even when using a clean amp. The only significant nonlinear thing I've measured that speakers do is thermal compression (that we model already) and "cone cry" which sounds like sh*t. Jay Mitchell is probably the leading authority on speaker design and he has stated pretty much the same thing. I'm all for improvements but they need to be real improvements. I've sat here countless times comparing an IR to the actual speaker with a mic on it doing blind A/B tests and can NEVER tell the difference and I think my ears are pretty good. I dug through my Matlab stuff and found this. (graph) This is an IR of a speaker taken twice. The first time the drive level is around 1W (in red). The second time the amp was turned way up, I would estimate at least 50W (in blue). As you can see the difference is extremely small. There's a small difference from 10 Hz and down which is way below the reproduction range of any system and a difference way up at Nyquist (24 kHz) but that's 100 dB down (!). Furthermore we don't know if the tiny differences are from the speaker or from the amp or the mic or the mic preamp. I should add that speakers can and do distort (when Xmax is exceeded) but it's not a pleasant sound. Since the displacement of the cone is the inverse of the frequency the low frequencies are distorted which is the opposite of what you want when creating "pleasing distortion". Speaker distortion is flubby, flabby and farty. The Axe-Fx II and III can simulate that, if desired, using the Speaker Drive parameter in the Amp block. I always set it to zero. There are probably some other modes that cause distortion but, again, these are dwarfed by the distortion of the amp. The only other significant one I've experienced is cone cry. Manufacturers go to great lengths to prevent it from happening. I have a speaker here that does it. Whenever I play a high F it cries and it's annoying." source

Axe-Fx Standard and Ultra IRs

IRs for the Axe-Fx Standard/Ultra must be converted to be able to use these with the Axe-Fx II.

It's no use converting 1024-point IRs to 2040 points because they don't contain the necessary data. You need an original WAV-file of sufficient length to create a 2040 point IR.

Enabling and disabling cabinet modeling

If you never use of cabinet modeling, turn it off in the Global menu.

You can also leave out the Cab block per preset. Or bypass it, but this will not decrease CPU usage.

Cab block in the Axe-Fx III

Product information: "The redesigned Cabinet block features a 4-channel mixer based on our popular Cab-Lab software, providing the capability to mix and remix IRs on-the-fly as you would with real mics on a speaker cabinet. Factory content includes selections from the best of today’s IR producers and artists, including Fractal Audio, AustinBuddy, Celestion, ML Sound Lab, Ownhammer, Chris Broderick, John Petrucci, Chris Traynor & James Santiago, Valhallir, York Audio, Dr. Bonkers, and more. An additional 2,048 “User Cab” memories allow you to load Cab Packs (including any of those compatible with the Axe-Fx II) or 3rd-party IRs, and a built-in utility allows you to capture and save your own speaker tones (now with 16 “Scratch Pad” locations!) Our celebrated Tone Matching block is also improved, now with the impressive ability to clone the tone of an amp or recording in UltraRes."

The Cab block now features a Cab Picker, making it easier to select cabs, employing filters.
Muting an IR in the Cab block decreases CPU usage.
The Speaker Size parameter and microphone modeling are no longer supported.
The Cab block automatically normalizes the output level of the Cab block when using multiple IRs. This means that theLevel parameter doesn't simply set the output level of each IR anymore, but it determines the relative levels between the IRs when more than one is loaded.
When the Cab is set for stereo input, two of the IR slots are fed by the left channel, and two are fed by the right channel.
When the Axe-Fx III is powered on, it reads all user cabs. This takes about 45 seconds and happens in the background.

Read the Axe-Fx III Owner's Manual for more information.

"Another cool thing is the Cab block. You can mix up to four IRs each with independent Pan, Distance, Proximity, Smoothing (De-phase). And it has four channels so you can switch between four completely different mixes, instantly." source

(about changing IR levels in the Cab block) "The volume stays constant." source

(about normalization in the Cab block) "Yes. This keeps the volume consistent regardless of the number of IRs and their mix levels. For example if you were to use two IRs and set each at -3 dB the volume would be half as loud. Behind the scenes it figures this out and compensates."

"Adjusting IR levels is not possible. Cab-Lab automatically normalizes IRs for "unity energy". 99.9% of the time this results in IRs that are the same volume but every now and then an IR will have energy outside the normal range of hearing which confuses the normalization routine. It's superior to the usual amplitude normalization but not without its faults."

Position of the Cab block

In the "real" analog world it makes a difference where you put effects: before or after the speaker cabinet. It's different with the FAS processors.

(Javajunkie) "You can place the effects loop anywhere in the chain (just add the fx loop block). Unless you are running a stereo cab or 2 mono cabs panned hard L/R, you may want to place stereo effects after the cab. The cab is a linear time invariant effect (unless you add drive) so effects like delay and reverb will sound the same before or after it. As Cliff and others have stated on numerous occasions LTI effects can be placed before and after each other and they will sound the same. Only when placed before or after non-LTI effects (drive, amps, et. al) it really matters. The one caveat there is that some effects are mono, placing effects before and after that makes a difference." source

*"The difference in having the cabinet before or after the effects is usually subtle. It depends on how non-linear or time-variant the effect is. For effects like EQ, which are linear and time-invariant, it doesn't matter at all. For slightly time-variant effects like chorus and flanger the difference isn't very pronounced. For highly time-variant effects, like pitch shifting, the difference can be marked."

"Linear means that the output is related to the input by a straight line: y = mx + b. Filters are example of linear systems. A cabinet IR is a filter. Distortion is an example of a nonlinear system. Linear systems are associative and commutative. Associative means that a * (b * c) = (a * b) * c. Commutative means that a + b = b + a or a * b = b * a. Therefore you can do cab -> eq (a * b) or eq -> cab (b * a). The cab block is "completely" linear if motor drive is non-zero but it is "wide sense stationary" so you can treat it as linear." source

"The cab block is level-dependent if the Motor Drive is non-zero. So if you turn up/down the level out of the amp block you may need to compensate by doing the opposite with the Motor Drive." source

"You gain nothing putting it before the cab and risk collapsing the stereo image if the cab is mono." source

"Since a cabinet is linear (or mostly linear) the order is unimportant as linear systems are commutative (a+b = b+a). However if the cab block is mono your effects will collapse to mono if placed before." source

(Cabinet blocks in parallel rows sound louder than a single Cabinet block) Bakerman: "It depends on how you're panning. Assuming a mono signal sent to cabs: Stereo cab w/ Pan L and Pan R fully left & right will be the same output level as 2 mono cabs w/ balance L & R. If pans/balances are centered the 2 mono cabs will be 6 dB louder. Balance elsewhere would be between 0 and 6 dB louder, and balance doesn't correspond 1:1 to pan L/R for the same placement. Balances will need to be further toward -50 or 50." source

Mono and stereo cabinets

Keep an eye on the mono/stereo configuration. The Cab block will sum the incoming signal to mono, unless it's set to Stereo mode or when using two (panned) Cab blocks to handle the left and right sides.

If a Cab block in Stereo mode is followed by a mono effect, such as Drive, the resulting signal will be summed to mono.

More information about mono and stereo.

CPU usage

The CPU load of a Cab block depends on its configuration. A mono Cab block uses less CPU than a stereo one. Also, the resolution of the selected IR(s) matters.

Muting IRs in the Cab block on the Axe-Fx III also decreases CPU usage.

Choosing cabs

Matching amp and cab models

It’s a matter of personal preference which impulse response (cab) you want to use with an Amp model. You can go for traditional combinations. Or be creative and innovative. The differences can be huge. Be aware that the cab often has more impact on the tone than choosing a different amp model!

When comparing cabs, don't judge too quickly. Each time you select a cab, you may need to adjust the amp settings to dial in a tone and let your ears adapt.

Common combinations of amps and cabs are listed in the wiki's list of Amp models and in Yeks Guide to the Fractal Audio Amp models.

Factory cabs

The processors have many built-in cabinets. These are also referred to as stock cabs or internal cabs.

Number of stock cabs per device:

Axe-Fx III: 2048 factory cabs + 189 legacy (Axe-Fx II XL+) cabs.
Axe-Fx II XL and XL+: 189.
Axe-Fx Mk I/II: 132.
AX8: 189.

All stock cabs are time-aligned, which means that you can mix them using a single Stereo cab or dual Cabs blocks.

Detailed list of all stock cabs.

(about the stock cabs in the Axe-Fx II and AX8) "The factory IRs were hand-selected by me after auditioning thousands of OH and RW and other IRs. Some of the IRs are custom mixes of mine. My rule-of-thumb was to select as neutral sounding IRs as possible. However, what I like may be much different than what others like. Some people complain the Axe-Fx sounds too bright. Others say it's not bright enough. It's a no-win situation. This is why I've been harping on capturing IRs. It's personal preference. Producers probably spend more time perfecting mic placement than anything else when getting guitar tones to tape. An IR is the same thing, it's capturing the mic and placement." source

To get a list of the stock cabs used in presets, use FracTool.

User cab slots

If you are looking for something different, try external impulse responses. The Axe-Fx and AX8 provide a load of user cab slots which can be filled with impulse responses, using Fractal-Bot, Cab-Lab, the editor or a MIDI librarian.

External IRs are files with a SYX extension, sampled at 48 kHz, 24 bits.

Number of user cab slots:

Axe-Fx III: 2048.
Axe-Fx II XL and XL+: 1024.
Axe-Fx II Mark I and II: 100.
AX8: 512.

The Axe-Fx II, III and AX8 and software editors display the names of the impulse responses in the user cab slots. The name is contained in the sysex data of the IR file. Impulse responses can be renamed using the editor or Cab-Lab. The name is shown in italics when it's an UltraRes impulse response.

To empty an user cab slot on the hardware, use the software editor or Cab-Lab. The hardware also provides an easy way to delete ALL user cabs.

Commercial and free IRs

The Axe-Fx II and AX8 come with a lot of stock cabs (factory cabs). More information here.

You can also create your own IRs or get additional ones from sources such as listed below:

Fractal Audio provides a couple of free IR libraries:

Some commercial manufacturers provide free impulse responses. There are also quite a few popular impulse responses available in the public domain. Some examples:

"gtrbody": adds the resonant sound of an acoustic guitar body (Axe-Change).
"cello_body": adds the resonant sound of a cello (Axe-Change).
GuitarHack ThisOne V30 (Axe-Change).

Scratch-Pads

Scratch-Pads (the very last user cab slots) are "dummy" locations which can be used to load impulse responses but which are not saved to non-volatile memory. This allows auditioning impulse responses without overwriting any of the user slots.

The number of Scratch-Pads depends on the hardware.

See which cabs are being used in presets

FracTool can poll the Axe-Fx or AX8 and show a list of the cabs that are used in presets. It also shows which user cabs are not being used in presets, so you can decide to delete these.

Substituting an IR with a Tone Match block

When you use an external IR in a preset and want to share the preset, you need to share the preset as well as the impulse response. There are two ways around this:

Integrate the impulse response in an Axe-Fx preset by replacing the Cab block with a Tone Match block, after having captured the tone of the Cab block. Here's how.
Create a Preset-Cab bundle, see below.

It's NOT permitted to share commercial IRs (license violation).

Preset-Cab bundle

Read this: Preset-Cab bundles.

Recording 4 different cabinet signals

This G66 tutorial shows how to create 4 separate cabinet signals in the Axe-Fx II, which you can mix at will. It comes down to using 2 stereo CAB blocks, with one of the blocks connected to a FXL block to feed Output 2. In both CAB blocks the impulse responses are panned hard left and right. The stereo outputs 1 and 2 are connected to 4 separate channels on the mixers.

With the Axe-Fx III it's much easier to accomplish.

Parameters

The Owner's Manual explains all parameters.

Parameters table

Parameter	Axe-Fx III	Axe-Fx II	AX8 (FX8: n/a)
Mode	no	yes	yes
Type	no	yes	yes
Bank, Cab Number (per IR)	yes	no	no
Level (per IR)	yes	no	no
Mute/Solo (per IR)	yes	no	no
Pan (per IR)	yes	yes	yes
Mic Type (per IR)	no	yes	no
Mic Distance or Delay (per IR)	yes	yes	yes
Proximity, Proximity Frequency (per IR)	yes	yes	yes
Smoothing or De-phase (per IR)	yes	yes	no
Preamp Mode, Preamp Type	yes	yes	no
Drive, Saturation	yes	yes	no
Bass, Mid, Treble	yes	yes	no
Low Cut, High Cut, Filter Slope	yes	yes	yes
Air, Air Frequency	yes	yes	no
Room Level, Room Size, Mic Spacing	yes	yes	no
Input Mode or Input Select	yes	yes	yes
Speaker Size	no	yes	no
Stereo Link	no	yes	yes
Motor Drive	no	yes	no

Input Select

This parameter lets you select the source signal that enters the Cab block. For example, if you wish to run two panned Cab blocks in an Axe-Fx preset, you can use this parameter to force one side of the signal to go into one Cab, and the other side into the other cab, for stereo separation.

Room ambience

The Axe-Fx provides room ambience parameters in the Cab block. This is a dedicated reverb effect providing "early reverb reflections" only, which works well when using headphones or IEM. Not supported on the AX8. Note that it turns a mono signal into stereo.

"The room stuff in the Cab block is a stripped-down, simplified version of the early reflections generator in the Reverb." source

Low Cut, High Cut, Filter Slope

Most impulse responses have been captured "close-miked", and produce a lot of high and low end material. High Cut and Low Cut Frequency in the Cab block (low-pass and high-pass) allow you to EQ this, preventing boomy bass and harsh sounds. Equivalent to using EQ controls on a mixing board, to position the guitar sound in a mix. These are very important parameters. Default value of High Cut is 10 kHz (Quantum 7 and later).

While these parameters appear on the Preamp page on the Axe-Fx III, they still work when the Cab block's preamp simulation is turned off.

Common settings are 80-150 Hz for high-pass (cut bass), and 5-10 kHz for low-pass (cut treble) but YMMV.

The “Filter Slope” parameter selects between first-order (6 dB/octave) and second-order (12 dB/octave) filters for Low Cut and High Cut.

"Using Low Cut in the Cab block is akin to what you would do in the studio to carve out room for the bass player." source

""LOWCUT FREQ" in the cab block sets sets the -3dB point of a highpass filter at the output of the cab block." source

"If at the min/max the filters are off." source

"People often talk about applying low cuts and high cuts. This is because the cabinet models used in modelers are almost always (with a couple exceptions) based on near-field samples of guitar cabinets. IOW, the mic is pushed up against the grill cloth. This just happens to be the way that record producers/engineers mic a cabinet in the studio and the way guitar cabs are mic'd on stage. This is done primarily for isolation reasons. The downside of this approach is that the resulting tone will have a lot more lows and highs than when listening to the amp+cab "in the room". What the mic "hears" when pushed up against the grill cloth is not the same thing that we hear standing 10 feet away. The most common technique to deal with this is to simply cut out the lows and highs using blocking filters, e.g. highpass and lowpass filters. Producers routinely do this when mixing as excessive amounts of lows and highs will cause the guitar tracks to get "lost in the mix". Live sound engineers often do the same thing. The Cabinet block has blocking filters built in for just this very reason. You can also use a couple dedicated filter blocks or a parametric EQ block. For now let's use the Cabinet block. My personal settings are Low Cut around 80 Hz and High Cut around 7500 Hz and Filter Slope set to 12 dB/octave but these are just a starting point. Far-field IRs are available but they are rare due to the difficulty in obtaining them. They require a large facility and special techniques making the process impractical in most cases. So, until an abundant source of far-field IRs are available we need to think like a producer/engineer who is dealing with the mic pushed up against the grill cloth. This means shaping the tone with EQ to remove unwanted frequencies." source

Dephase / Smoothing

This parameter controls a sophisticated process that removes the “phasiness” from impulse responses by reducing the prevalence of peaks and valleys in the IR. This yields a more “in the room” experience. This is especially helpful when using multiple impulse responses.

Not supported on the AX8 hardware.

Cab-Lab can apply De-Phase when mixing impulse responses together.

The processing required is extreme and the control can have some lag. No extra CPU usage or audio latency, however, is incurred.

"Close-mic'd speakers can sound "phasey" because you are in the near field. When sampling the near field of any source the frequency response and beam pattern is rough. This occurs due to multiple spherical waves arriving at various phase angles. These multiple waves come from the various modes of the speaker, internal cabinet reflections and from other speakers in the cabinet. In the far field the response is more uniform because the wavefronts get flatter and the phase angles converge. The De-Phase parameter removes some of the phasiness due to multiple wave arrival using a complex FFT technique." source

"The higher the setting the more "character" you remove. De-Phase removes some of the character but that's precisely what you want to do as a cab has less character in the far field." source

(Why is De-Phase necessary?) "You don't listen to a guitar speaker with your ear against the grill cloth." source

"It's so simple that even experts in the field don't realize why it works." source

Motor Drive

This parameter was present in the Amp block, before Quantum 9. Quantum 9 replaced it with Speaker Compression. Motor Drive is still present in the Cab block on the Axe-Fx II only. It models the effect of high power levels on the speaker.

Motor Drive was Revised in Quantum 7.0:

"Improved Motor Drive algorithm. New algorithm more accurately models the compression of guitar loudspeakers by factoring in the reactive aspects of the compression.

"The Motor Drive simulation is available in both the Amp block and Cab block now. It is recommended to use the simulation in the Amp block when using an FRFR configuration as the Amp block simulation uses the speaker resonance information in the calculations whereas the Cabinet block uses fixed values. When using a conventional guitar cab, or a hybrid configuration with monitoring via a conventional guitar cab and speaker emulation to FOH, the Motor Drive in the Cabinet block can be used instead. The simulation in the Amp block also has the advantage of being independent of the block’s output Level control."

"Gain monitoring of the Motor Drive is available on the MIX page of the Cabinet Block and the PWR DYN page of the Amp block. In the case of the Amp block the monitoring is available when the Motor Drive parameter is selected. Note that typical guitar speakers have around 3-6 dB of compression when driven hard with American speakers being on the low end of that range and British speakers being on the high end. Some speakers can exhibit even more compression than this with compression amounts of 8 dB or more depending upon the magnetic materials used and the construction of the speaker motor."

"The thermal time constant of the virtual voice coil is adjustable using the “Motor Time Const” parameter. Typical guitar speakers are anywhere from 0.05 to 1.0 seconds depending upon the mass of the voice coil and the materials used."

"Set it to 4.5 and rip the knob off." source

When using two Ultra-Res cabs in a preset, don't use Motor Drive with just one, because this will introduce comb filering (phase cancellation).

"Motor drive isn't EQ. It models efficiency reduction due to thermal effects." source And: "What I have found is that thermal compression is somewhat noticeable and measurable. This is modeled by the Motor Drive parameter." source

"Motor Drive will cause compression if not set to zero (as it models driver compression). Otherwise the cab block is completely linear and will not cause any compression." source

"Motor Drive simulates power compression due to voice coil heating." source

"Guitar loudspeakers are intentionally designed to compress. FRFR speakers do compress a bit but not nearly to the extent that guitar speakers do." source

"Makes edge-of-breakup tone stupid easy." source

"Speaker Drive models the magnetic compression (which is actually distortion) that occurs due to the nonlinear speaker excursion vs. applied voltage. Motor Drive models the change in power transfer due to heating of the voice coil. When the voice coil heats up the speaker sensitivity decreases, in some cases quite dramatically." source

"The thermal time constant of a typical guitar speaker is about 0.52 seconds. Magnetic time constants are zero." source

"So what I've done for the final release is put Motor Drive in BOTH the Amp block and the Cab block. If you're strictly FRFR then you can use the Amp block. If you are using a conventional guitar cab or a hybrid configuration (convention cab for monitoring and direct to FOH) then you can use the Cab block. Doing it in the Amp block also has the advantage that the speaker resonance information in the Amp block is used to calculate the frequency dependent heating whereas the Cab block uses a fixed set of data that is representative of a typical speaker. Finally I've made the time constant adjustable. I did some more calculations and measurements and found that a typical guitar speaker is actually lower than what I had previously calculated because thinner wire is used than I was assuming. Regardless you can now set the thermal time constant to get whatever response rate feels best. When using the Motor Drive in the Amp block it's before the output Level control so you don't have to worry about the behavior changing when you adjust the Level knob." source

"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." source

"The formula is tau = M * C * theta where M = mass, C = specific heat of the voice coil material (typically copper) and theta = thermal resistance between the voice coil and the magnet gap." source

Air

The Air parameter mixes some of the "direct" signal entering the Cab block with the processed signal leaving the Cab block. This adds some "air" to the sound.

The Air Frequency parameter lets you adjust the cutoff frequency of the mixed signal. Increase the Frequency to its maximum value for a straight mix.

Not supported on the AX8.

If you want to listen to just the Air'd part of the signal, set the Cab to an empty user cab, and turn up Air.

Delay / Mic Distance

This is a "micro delay" for stereo applications. When running a Cab in Stereo mode, or when using two panned Cab blocks in parallel, delaying one side relative to the other can achieve interesting comb filter effects. A common practice in studio recording is to use multiple mics on a speaker at different distances to intentionally introduce comb filtering.

"My secret to realistic cab sounds is Delay. Use two IRs in stereo or two cab blocks and put a small amount of delay on one (using the Delay parameter in the Cab block). I like around 0.06 ms. You may like more or less. Producers experiment with placing mics at different distances to enhance the recorded guitar tones. This is the same as using a small amount of delay. Adding a bit of delay introduces some comb filtering which creates notches and peaks in the response which, in turn, adds a sense of "space" to the tone. Try it." And: "If you have any cab packs try mixing the "Back" IR with one of the regular IRs. I use more delay when doing this, 0.1 ms or more. I lower the level on the back IR by a couple dB. This gives a nice "in the room" open-backed cab sound."

The Axe-Fx II uses milliseconds to measure "DELAY" times in the cab block. The Axe-Fx III has a "MIC DISTANCE" parameter instead, measured in millimeters. To convert Axe-Fx II values for use on the Axe-Fx III, multiply the old value in milliseconds by 343.

If you want to use this parameter but don't want IR coloring, use a "flat"or "null" IR. The Axe-Fx III has one as a stock cab.

(Matt) "The flat IR in the Axe-Fx III is NOT one that was downloaded. We created this file from scratch when we were working on a bass rig for one of our most celebrated endorsers. Intended uses include adjusting a DI and/or IR with Mic Distance, or combining DI and IRs into the same virtual preamp and room sim (which gets you a pretty great Nile Rodgers tone, by the way."

(GM Arts) "This is about mixing 2 signals: one without delay, and the other with a very short delay. 0.06ms is way too short to be perceived as a repeat; the effect is filtering caused by mixing these two signals. To keep things simple, we’ll apply an equal mix of the same signal and another delayed by 0.06ms. An easy way to experiment with this in the Axe-FX is with a Flanger block, with depth and feedback set to zero, and mix set to 50%. Adjust the delay to 0.06ms (not 0.6ms) to hear the effect with a mono signal. This produces a notched frequency response with complete signal cancellation just above 8KHz, with the -3dB point one octave lower at just over 4KHz. The signal is restored over the next higher octave (8KHz to 16KHz), but bear in mind that most IRs will not have much response there anyway, so this effect is mostly a blocking filter over the range 4KHz to 8KHz. So if you have a cab IR that has some response over this range, it will be perceived as a loss of some treble response. For many, this will remove harshness in a way that’s difficult to achieve with other filters. Others may find this effect too much. You can soften this effect by decreasing the delay and/or changing the mix ratio. Decreasing the delay raises the frequency at which this cut occurs. For example, a 0.05ms delay blocks response over the octave 5kHz to 10kHz. Lowering the mix % decreases the depth of the notch. Similarly, applying a delay to a different IR than the un-delayed block will “jumble” and reduce the final response to some extent. If you increase the delay (typically from 1ms and above), you’ll hear the combing effects as multiple notches become low enough to hear in the range of “guitar frequencies”. This sounds like a flanger or chorus without modulation, which shouldn’t be a surprise given we’re experimenting with a Flanger block. So why does this delay sound produce a tone more amp-like? Most players prefer their amp tone off-axis, meaning that they’re avoiding the direct harsh sound directly in front of the speaker, where high-frequencies are beamed. This filter simulates that effect. It’s also similar to standing slightly off-axis when using multiple speakers. Sound travels at roughly one foot per millisecond, so there is a very short delay between sound from different transducers. As Cliff stated, it also emulates recording techniques with mics placed at different distances from the cab. How to calculate? To find the frequency where this rolls-off high frequencies at -3dB, it’s simply: Hz = 1000 / 4 /delay in ms. So for 0.06 ms: 1000 / 4 / 0.06 = 4167Hz. Complete cancellation occurs at double this frequency, 8333Hz, and builds back to -3dB a double this frequency again, 16666Hz. Bear in mind that with higher delays, there will be audible effects from additional notches above this calculated frequency." source

Preamp simulation

The Cab block also provides simulation of channel strip preamps, tapes and such. The DRIVE parameter controls the gain of the simulation. The SAT parameter controls the ratio of even/odd harmonics. The PREAMP MODE parameter allows selecting between Economy and High Quality modes. In High Quality mode oversampling is employed to prevent aliasing which results in higher CPU usage.

Supported types include: Tube, Bipolar, FET 1, FET 2, Transformer, Tape 70us, Tape 50us, Tape 35us, Vintage, Modern, Exciter. Set to “NONE” to bypass the virtual preamp completely, saving CPU.

Preamp simulation is not supported on the AX8.

"The VU meter shows the level into the pre. Select a pre Type and turn up the Drive. As the VU approaches the 0 dB marker you will begin to overdrive the pre." source

"Probably not something you would use for clean sounds. A common technique for rock music is to push the pres, console, tape, etc. to varying degrees to get compression and "sparkle". The trick is getting just the right amount. Too much and it sounds raspy and nasty." source

"0 on the VU meter indicates onset of clipping. It's not the same as your plug-ins in that regard. The problem with plug-ins is that you don't know where the onset of clipping is since the headroom isn't specified. Our way is superior since 0 dB indicates the point where things are clipping. The other way you have no idea where things start clipping. So 0 dB on the Axe-Fx is NOT equivalent to 0 dB on a typical plug-in." source

"I've done a lot of testing with isolation cabs. The big thing that happens is that the mic distorts, especially when using an SM57. This adds some crispness to the high end and some compression. I've found that I can duplicate that effect very closely by using the FET I preamp type in the Cab block and turning the Drive up until the desired compression is achieved. I set Sat to zero." source

Speaker Size (Axe-Fx II only)

Size Warping allows the user to change the relative size of the virtual speaker. This is controller with the parameter Speaker Size. This parameter appears only if the selected IR is not UltraRes and the Cab block mode is set to Mono.

This parameter is not supported on the AX8 and the Axe-Fx III.

Microphone modeling (not on Axe-Fx III)

Miced cab.png

Microphone modeling: supported by which Fractal Audio Products

Axe-Fx III: no.
Axe-Fx II: yes.
AX8: no.
FX8: no.

About microphone modeling

Microphone modeling is implemented in the Cab block in the Axe-Fx II and AX8 through the use of Impulse Responses.

Microphone modeling is not supported on the AX8 and AXE-FX III, except for the Proximity effect.

(Axe-Fx II) "The mic models are actually IRs. The mic IR is convolved with the speaker IR to create a composite final IR." source

(Axe-Fx II) "If I were to design a Cab block today I wouldn't even include a Mic parameter. I NEVER use Mic simulation anymore. I simply find an IR I like and EQ as desired." source

(Axe-Fx II) "The mic options are mostly legacy. I never use them but if we took them out there would surely be much protestation." source

(Axe-Fx III) "Not gonna happen. What we found is that convolving a conventional mic IR with an IR obtained with a reference mic sounds nothing like capturing the IR with the conventional mic. The beam pattern of a reference mic is completely different than conventional mics. A reference mic is nearly omnidirectional whereas conventional mics have narrower beam patterns. In the far-field this wouldn't matter as much and "microphone modeling" might work. However in the near field this makes a huge difference and it simply doesn't work. Furthermore in the far field you want the response to be as neutral as possible so in this case there would be no desire for mic modeling anyways." source

Modeled microphones

57 DYN: based on the Shure SM57.
58 DYN: based on the Shure SM58.
421 DYN: based on the Sennheiser MD 421 II.
87A COND: based on the Shure Beta 87A.
U87 COND: based on the Neumann U87.
E609 DYN: based on the Sennheiser e609 Silver.
RE16 DYN: based on the Electro-Voice RE16.
R121 COND: based on the Royer Labs R-121.
D112 DYN: based on the AKG D112.
67 COND: based on the Neumann U67.
NULL: this doesn't apply microphone coloring, but it enables the use of the PROXIMITY parameter.
INVERT: inverts the signal, allowing for interesting effects in conjunction with the delay parameter.
NONE: disables ALL mic processing in the CAB block, including PROXIMITY.

The "NONE" and "NULL" types both disable mic coloring. A mic is still involved though, because IRs are always captured with microphones. Even when a neutral mic was used to capture the IR, such as an Earthworks mic. When capturing IRs, the mic is most often placed very close to the speaker, so the result is a close-miked tone. Still, selecting "NONE" is the best way to prevent adding additional EQ-ing to the tone.

Proximity

Simulates the proximity of the modeled mic to the speaker. Higher numbers translate to the mic being closer to the speaker (near-field), causing an increase in bass or low frequency response. Lower numbers translate to the mic being further away from the source, with the lowest number providing far-field coloration.

Proximity only works when a mic model is selected, including the NULL type. The Proximity Frequency parameter allows tuning the frequency range over which the proximity effect occurs.

The AX8 and Axe-Fx III do not support mic models, but provide a Proximity parameter.

Cab-related parameters in the Amp block

The Amp block provides a number of parameters which are closely related to the Cab block. In particular: Speaker Compression, Speaker Drive and Resonance.

About microphones