Always consult the official Owners Manuals first!
January 2024: all pages have been checked and are up-to-date
- 1 Available on which products
- 2 Channels or X/Y
- 3 Amplifier and cabinet modeling for beginners
- 4 About Impulse Responses (IRs)
- 5 Cab modeling progress
- 6 Cab block diagram
- 7 Position of the Cab block on the grid
- 8 Cab modeling on the FM3, FM9 and AX8
- 9 No latency and always in phase
- 10 Disabling cabinet modeling
- 11 DynaCabs
- 12 Select a cab
- 13 Scratch-Pads
- 14 Preset-Cab bundle
- 15 Mic preamp and channel strip modeling
- 16 Mono or stereo
- 17 Cab block and CPU usage
- 18 Cab-in-the-room
- 19 Parameters
- 20 Tips, tricks and troubleshooting
- 21 Videos
Available on which products
- Axe-Fx III: 2 blocks (4 IRs per block)
- FM9: 2 blocks (2 IRs per block)
- FM3: 1 block (2 IRs)
- Axe-Fx II: 2 blocks
- AX8: 1 block
- FX8: not available
Channels or X/Y
- Axe-Fx III and FM3 and FM9: 4 channels
- Axe-Fx II: X/Y
- AX8: X/Y
Amplifier and cabinet modeling for beginners
About Impulse Responses (IRs)
The Cab block relies on Impulse Responses (IRs) to reproduce the sound of speaker cabs.
See these pages for more information:
- Impulse responses (IR)
- IR length: Normal, Standard, HiRes, UltraRes, FullRes, DynaCab
- Far-field IRs and close-mixed IRs
- IR Capture
Cab modeling progress
Axe-Fx III with firmware Ares
FRACTAL AUDIO QUOTES
From the Axe-FX III product page:
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 Picker makes it easier to select cabs, employing filters. To access the Cab Picker on the hardware, press <ENTER> in the Cab Number field.
- Muting an IR in the Cab block decreases CPU usage.
- SPEAKER SIZE parameter and dedicated microphone modeling: not supported.
- The Cab block normalizes its output level when using multiple IRs. The LEVEL parameter of each IR doesn't simply set the output level of that IR, but it determines the relative levels between the IRs when more than one is loaded..
- When set to Stereo Input, two of the IR slots are fed by the left channel, and two are fed by the right channel (Axe-Fx III).
- When powering on, the unit reads all user cabs. This happens in the background.
- IRs can be visually aligned.
- Room ambience has been improved and includes floor reflections.
- Micro Delay has been changed to MIC DISTANCE.
- IR LENGTH lets the user adjust the length of an IR.
- LOW CUT and HIGH CUT and SLOPE can be set per IR.
- Smoothing, Proximity and Delay can be set per IR.
- Much steeper filter slopes are possible.
- A separate IR Player block also processes IRs, but has less functionality and therefore requires less CPU than the Cab block.
More information is available in the Owners Manuals.
FRACTAL AUDIO QUOTES
 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.
About changing IR levels in the Cab block:
 The volume stays constant.
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.
Firmware 17 for the Axe-Fx III
This firmware introduced FullRes IRs.
Firmware 22 for the Axe-Fx III
This firmware introduced DynaCab cabinet modeling.
Cab block diagram
Position of the Cab block on the grid
real analog world, it makes a difference where you put effects, either before or after the speaker cabinet. It's different with digital processors.
Cabinet blocks in parallel rows sound louder than a single Cabinet block.
FRACTAL AUDIO QUOTES
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 not "completely" linear if motor drive is non-zero but it is "wide sense stationary" so you can treat it as linear.
 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.
 Since linear systems are commutative EQ > Delay is the same as Delay > EQ.
Now, if the system isn't linear (i.e. there's some distortion) then the sound will change. Also if the system is time-variant the sound will change but most modulation is wide-sense-stationary enough to not create a noticeable change if the order is reversed.
The most likely explanation is stereo delay into a mono cabinet which will destroy the stereo image.
 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.
 You gain nothing putting it before the cab and risk collapsing the stereo image if the cab is mono.
 Given that most post effects are linear or "wide sense stationary" the order of effects after the amp doesn't matter. Reverb -> Cab is theoretically equivalent to Cab -> Reverb because linear systems are commutative (i.e. a * b = b * a).
However... if the cab block is mono then you'll collapse any stereo effects to mono. Or if the cabs aren't panned fully L/R you'll lose your stereo imaging.
Pitch effects are not linear so putting them before the cab block will sound different than after. Anything that causes distortion is not linear so the order matters. If the distortion is subtle then the order is less important.
 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.
 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.
Cab modeling on the FM3, FM9 and AX8
The FM3 has the same Cab block functionality as the Axe-Fx III, but provides a single Cab block instead of two, supports two IRs instead of 4, and some functionality has been left out, including smoothing.
The FM9 has two Cab blocks like the Axe-Fx III, but supports two IRs instead of four per block, and some functionality may have been left out, like smoothing.
On the AX8, FM9 and FM3, cabinet modeling runs in an CPU accelerator instead of the core DSPs, which saves CPU. On the FM9, the second Cab block doesn't run in the CPU accelerator, so it'll take up more CPU than CAB 1. 
FRACTAL AUDIO QUOTES
FM3 and FM9:
 FM3/9 only support 1K samples for normal IRs since the IR processing is handled by a coprocessor whose max. length is 1K samples.
 The Cabinet block uses an FIR accelerator to do the IR processing. This FIR accelerator offloads processing from the CPU and, as such, doesn't reflect in CPU usage.
 They are done in an a separate accelerator so they have minimal CPU loading.
 An accelerator is a DSP unit dedicated to performing a defined task. In this case it performs convolution. The difference is that a "DSP" as we commonly call them is really just a microcomputer optimized for Digital Signal Processing. The term DSP most correctly refers to any device that does signal processing using numerical computations. That device can be an FPGA, ASIC or fixed hardware unit. In the AX8 it is a convolution processor.
No latency and always in phase
Cab processing does not increase processing latency.
The phase of IRs is always correct.
FRACTAL AUDIO QUOTES
 Cab blocks do not add any latency.
 The Cab and IR Player blocks automatically "phase correct" IRs so that the peak signal is positive.
Disabling cabinet modeling
If you never use cabinet modeling, turn it off in the Global Settings menu which will decrease CPU usage. You can also simply not add the Cab block to the preset.
You can bypass the Cab block in a preset to disable cabinet modeling, but this will NOT decrease CPU usage.
Firmware 22+ for the Axe-Fx III, firmware 5+ for the FM9 and firmware 7+ for the FM3, feature
DynaCab cabinet modeling. This allows us to freely position the microphone. Set the Cab block to DynaCab mode to be able to select a mic type and set its position and distance. Behind the scenes the appropriate IR is loaded. A graph visualizes the settings.
The available microphone types in the stock DynaCabs are:
Dynamic 1 (Shure SM57),
Dynamic 2 (Shure SM7),
Ribbon (Royer 121?),
Condenser (Soyuz 023).
Additional DynaCab packs with more microphone choices can be bought in Fractal Audio's DynaCab store.
The built-in factory DynaCab IRs are 2048 samples on the Axe-Fx III, but the resolution can be lowered to Standard (1024 samples) to save CPU. On the FM9 and FM3 the factory DynaCabs are 1024 samples. Fractal Audio's commercial DynaCab Packs contain Ultra-Res IRs.
DynaCabs IRs are time-aligned without destroying phase information, allowing them to be mix-and-matched. They have 0.3 ms of leading silence.
The Cab block offers parameters to let you select the speaker cabinet type, the type of microphone (condenser, dynamic, ribbon), the position (cap to cone) and the distance between mic and speaker. DynaCabs have no Proximity parameter.
Depending on a configuration setting in the Amp block, the Amp block automatically selects the correct Speaker Impedance Curve for the selected DynaCab in the first IR column of the Cab block.
DynaCabs support smoothing.
Auto DynaCab Impedance in the Amp block to automatically select the correct Speaker Impedance Curve in the Amp block when selecting a DynaCab in the first IR slot of the Cab block. If the preset contains only the CAB 1 block, Auto Impedance controls both AMP 1 and AMP 2 blocks. If the preset contains CAB 1 and CAB 2 blocks, CAB 1 controls AMP 1, and CAB 2 controls AMP 2.
FRACTAL AUDIO QUOTES
 You can mix up to four DynaCabs. You can mix-and-match any combination of cabs and mics.
Position sets the radial distance of the microphone from the center of the speaker.
Distance sets the distance from the grill cloth. As you move the mic away from the grill the bass typically rolls off and more room is heard.
 We actually used six mics during the captures, two of each type. I picked one from each type that I thought sounded best. The dynamic is an SM57 (naturally). I actually prefer the SM7 captures we made but given how iconic the 57 is on guitar cabs I went with that instead.
 While the SM57 is harsh on its own that extra brightness is useful when blending mics. Yes, I prefer the SM7 on its own but the SM57 is ubiquitous in studios when blending two or more mics.
 I found it's worth exploring each cab to find the sweet spot. Some of the cabs I like the mic near the edge whereas others I like it closer to the cap.
 […] the closer you are to other speakers the more interference you get from them. It's even more noticeable as you pull the mic back from the speaker (again, not surprising).
We actually captured the entire speaker for this Dyna-Cab stuff. I then auditioned radially and found that I preferred the IRs that moved away from the other speakers. I.e. for a 4x12 I preferred a radial line from the center to the nearest corner of the cabinet.
Even single speaker cabs I heard a difference in some cases. For example, our Tweed 1x12 the speaker is off-center on the baffle board so it doesn't sound the same moving left vs. right.
 0.0 represents as close as possible to the grill cloth without touching it.
 The files are initially aligned to the same reference point -- a universal standard across all DynaCapture IRs. In other words, IRs for mics that are farther away have been time compensated to be aligned with IRs for mics that are closer.
You can use the controls on the ''Align page in the usual way, however, to add time to any individual IR.
 The Amp block now features “Auto Dyna-Cab Impedance”. When set to ON the speaker impedance curve of the Amp block will follow the Cabinet Type in the first mixer slot of the associated Cabinet block. I.e., if the Cab Type in the first mixer slot of Cabinet 1 is, say, 4x12 5153 and the Mode is Dyna- Cab then Amp 1’s speaker impedance will automatically be set to 4x12 5153.
 Cabinet 2 Slot 1 controls the SIC of Amp 2. Cabinet 1 Slot 1 controls the SIC of Amp 1.
 All Dyna-Cabs have corresponding impedance curves that were taken from the actual cabs.
 Best sounding condenser mic I've ever used.
 It's because they are not minimum-phase transformed.
 If only Cab 1 is in the grid then that controls both Amp 1 and Amp 2. Otherwise Cab 2 controls Amp 2.
 Dyna-Cabs IRs were captured with a Neve preamp. Legacy IRs were shot with an API. API preamps have more mids. I prefer the sound of Neve.
 I made some measurements of our Neve vs. API preamps. The difference is virtually nil. For a condenser mic the difference IS nil which is unsurprising since a condenser has its own built-in preamp.
Here's the frequency response difference between a Neve 1073 and an API 312: .
If you ignore the difference at 20 Hz, which is inaudible and only about 1.5 dB, the difference is within a couple tenths of a dB across the spectrum. This is with an SM7B. The results were similar with an R121.
Therefore the only logical conclusion is that the mic preamp has very little influence on the IR. Now, that's not say mic preamps don't sound different. The nonlinear characteristics are definitely different and will manifest depending upon how hard the preamp is driven. However, IRs are inherently linear and don't capture these nonlinear characteristics and, as seen in the graph above, the resulting IRs are basically independent of the preamp used.
Select a cab
How to choose
It’s a matter of personal preference which cab(s) you want to use with a specific amp model. You can choose a traditional combination, or think out of the box. The differences between cabs can be huge. The cab has at least 50% impact on the sound of an amp+cab combo, more than adjusting amp controls.
When comparing cabs, don't judge too quickly. Each time you select another cab, your ears have to adapt. Also, you may need to adjust the amp settings to suit the selected cab. You can use the Looper to playback a recorded clip while you switch between cabs.
Traditional combinations of amps and cabs are:
Legacy or DynaCab mode
Select whether you want to use Legacy cabs or DynaCabs.
See DynaCabs for more information.
The firmware of amp modelers contains many built-in factory cabinets, also referred to as stock cabs or internal cabs. These can be selected with the Cab block in Legacy mode.
Number of factory cabs:
Axe-Fx III + FM3 + FM9 – 2048
new factory cabs (2 banks of 1024), and 189
legacy cabs (same ones as in the Axe-Fx II XL+ and AX8)
Axe-Fx II XL + XL+ – 189
Axe-Fx Mk I + II – 132
AX8 – 189
All stock cabs are time-aligned, which means that you can mix them without phasing issues.
See Detailed list of all stock cabs for more information.
FRACTAL AUDIO QUOTES
 When I was capturing IRs I specifically chose to obfuscate the names to force people to use their ears.
 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.
Note: Stock cabs can ONLY be mixed with other cabs on the hardware itself. Cab-Lab can't mix stock cabs with other cabs, unless you capture a factory cab and turn it into external SYX and IR files using Cab-Lab.
Yek tells us:
 I'm using an Axe-Fx III to capture a stock cab from the FM3:
- Connect Axe-Fx III / OUT2 LEFT to FM3 / IN2 LEFT
- Connect FM3 / OUT2 LEFT to Axe-Fx III / IN2 LEFT
- Turn up the OUT2 knobs on the Axe-Fx III and FM3 hardware
- Build a preset on the FM3 consisting of [IN2] - [CAB] - [OUT2]. Select the desired stock cab. Disable additional processing in the Cab block including Low/High Cut
- Select MinPhase etc.
- Enter a name for the IR that will be created (press Enter after entering the name)
- You can do this on the hardware only, but if you want an .IR file too for mixing, you need to use the IR Capture tool in Axe-Edit.
- On the FM3 you can use OUT1 instead of OUT2 if desired.
- If you're running Axe-Edit and have captured the IR to an user slot, its name will appear only after the cab slots have been refreshed.
The method above has provided me with .SYX and .IR files of the famous Basketweave TV Mix stock cab (Legacy #103), as well as #102 (AX Mix), #105 (EV-12L) and #106 (EV-12S). The captured IRs sounds exactly the same as the stock cabs. I can now mix these into a single IR with Cab-Lab.
Or use one of these methods:
If you want to look beyond the stock cabs, try external impulse responses. In Legacy mode, the amp modelers provide user cab slots which can be filled with external cabs (imported impulse responses), using Fractal-Bot, Cab-Lab, the editor or a MIDI librarian.
External IRs for Fractal Audio processors are files with a
SYX filename extension, sampled at 48kHz, 24 bit.
Files with an
IR filename extension are also IRs for Fractal Audio processors, but restricted for use in Cab-Lab. It's a proprietary Fractal Audio format. You can't load these directly into the hardware.
IRs in WAVE format can be imported directly into the Axe-Fx III and FM3 with the Manage Cabs tool in the editor. Simply drag-and-drop any number of .wav files into the Browser pane to allow the editor to convert them. To convert a batch of WAVE files in one go, use Cab-Lab.
You can choose between MPT and auto-trim, and create UltraRes or non-UltraRes cabs.
The Number of user cab slots by unit are:
Axe-Fx III Mark I – 2048 (2 banks of 1024)
Axe-Fx III Mark II – 2048 (2 banks of 1024), plus a bank for 64 FullRes IRs
FM3 – 1048
FM9 – 1048
Axe-Fx II XL and XL+ – 1024
Axe-Fx II Mark I and II – 100
AX8 – 512
The amp modelers and software editors display the names of the external cabs in the user cab slots. The name is contained in the sysex data of the file. Impulse responses can be renamed using the editor or Cab-Lab. The name is shown in italics or a different color when it's an UltraRes impulse response.
To empty a user cab slot on the hardware, use the software editor or Cab-Lab. Some modelers (including Axe-Fx III and FM3) allow you to delete ALL user cabs at once through the front-panel's Utilities menu. Be careful and consider making a backup first!
Also see Sources for commercial and free IRs.
Scratch-Pads are the very last user cab slots. These are reserved
dummy locations, meant to temporary load impulse responses. This allows auditioning impulse responses without overwriting any of the user slots. The number of Scratch-Pads depends on the hardware. Scratch-Pads are erased when powering off the unit.
Firmware 17 for the Axe-Fx III adds a dedicated Scratchpad for auditioning FullRes IRs.
A Preset-Cab bundle is a single file containing a preset and user IRs used by that preset. You can save and load Preset-Cab bundles with the editor. Bundles make it easier to share sounds and are easy to export and import.
Warning: Do not share IRs which are protected by a EULA, license, copyright, and such.
To export a bundle, use Preset > Export Preset-Cab Bundle menu in the editor, which will display a dialog allowing you to include IRs by checking them.
To import a bundle, use Presets > Import Preset menu in the editor, or drag-and-drop a bundle file onto the editor's main Preset display. This will display a dialog, then the editor will unpack the Preset-Cab Bundle and save it for use to your device. A bundle contains one preset plus all of the IRs it depends on. To proceed, you must select a location to save each of the items within the bundle. The preset will be updated automatically to reference the location(s) you selected for the IRs.
Also see Presets for more information.
Mic preamp and channel strip modeling
Microphone preamps, channel strips and tape can create pleasing musical
distortion. This might range from subtle
warmth to full-on
nasty. Mic preamps and channel strips also offer tone controls which change the sound. The Cab block in the Axe-Fx series and FM3 includes controls to produce these effects. This is not supported on the AX8.
Types include: Tube, Bipolar, FET 1, FET 2, Transformer, Tape 70us, Tape 50us, Tape 35us, Vintage, Modern and Exciter. The three TAPE types have different equalization time constants. 
Some of these are showcased in the FM3 factory preset NASTY PRE SLAM.
The simulation is switched on/off through the PREAMP TYPE. Set it to NONE to switch it off and save CPU usage. 
If PREAMP TYPE is set to anything other than NONE, it's active, But it will have no impact on the sound if DRIVE and SATURATION are both turned fully counter-clockwise. PREAMP TYPE affects only DRIVE and SATURATION, not the tone controls. 
DRIVE controls the gain. SATURATION controls the ratio of even/odd harmonics.
PREAMP MODE selects between ECONOMY and HIGH QUALITY. In High Quality mode, oversampling is employed to prevent aliasing which results in higher CPU usage. 
Important: LOW CUT, HIGH CUT and FILTER SLOPE are independent of the preamp. If the preamp simulation is switched off, LOW CUT and HIGH CUT are still fully operational.
FRACTAL AUDIO QUOTES
 It sets the oversampling rate for the preamp emulation.
 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.
 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.
 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.
 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.
 I always use a little bit of preamp drive in the cab block. All the venerated real preamps add distortion. There's nothing clean about them. It's the distortion that gives them character.
 If you are trying to achieve amp > cab > compressor > tape, then you need to change the order around. A Drive block (ostensibly to mimic tape distortion) after the Amp and before the Cab block sounds a bit dull. The tape distortion adds high end but putting before the cab block filters out those highs.
The correct sequence would be:
Amp > Cab > Compressor > Drive.
The way I would do it is simply Amp -> Cab and use the compressor in the Amp block and one of the tape simulations in the Cab block.
Mono or stereo
The Mono/Stereo mode of the Cab block is important.
The Cab block will sum the incoming signal to mono when using a single IR. It will process and maintain an incoming stereo signal when set to stereo or when using two panned Cab blocks to process the left and right channels.
If a stereo Cab block is followed by a mono effect, such as Drive, the resulting signal will be summed to mono.
If the Cab block on the Axe-Fx III is set to Stereo Input, two of the four IR slots are fed by the left channel, and two are fed by the right channel.
FRACTAL AUDIO QUOTES
 In Mono left and right are summed and fed to all four cab slots. In Stereo left goes to slots 1 and 3, right goes to 2 and 4.
Stereo: left goes to slots 1 and 3, right goes to 2 and 4.
Left: Left goes to all slots.
Right: Right goes to all slots.
Sum L+R: Sum of L+R goes to all slots.
See Mono and stereo signal for more information.
Cab block and CPU usage
CPU usage of a Cab block depends on its configuration and the unit.
Axe-Fx III + FM3 + FM9
- mute an IR to decrease CPU usage.
- use the Axe-FX's IR Player block instead of the Cab block to decrease CPU usage.
AX8 + Axe-Fx II
- A mono Cab block uses less CPU than stereo.
- An UltraRes IR uses more CPU than a mono or stereo Normal IR.
FullRes IRs use more CPU than Ultra-Res IRs.
When importing presets created on the Axe-Fx III or FM9 into the FM3, IRs are set to Standard resolution to save CPU.
Cabinet modeling usually reproduces the sound from a close-mic'd speaker. The player hears the sound through headphones, FR amplification, IEM or studio monitors. This differs from the
amp/cab in the room sound, where the player hears the guitar sound coming from a traditional guitar rig.
There are multiple approaches for achieving the
amp/cab in the room sound through FRFR amplification:
- FRFR and amp-in-the-room
- FullRes IRs are Fractal Audio's approach to adding room ambience to the sound through headphones and IEM, and recordings
- The Filter block can be used to create the sound of an
amp+cab-in-the room. See these pages for more information:
If you crave a real
amp/cab in the room tone from your modeler, amplify it through a power amp and a traditional guitar speaker cabinet.
FRACTAL AUDIO QUOTES
 You're never going to get a full-range monitor to sound like an amp in the room regardless of the IR used. One reason for this is dispersion. A traditional guitar cabinet has a beam pattern that decreases with increasing frequency. This means less high frequencies when listening off-axis. A full-range monitor will have more highs. Now some will argue that if you capture the traditional cab off-axis in the far field then you'll get the same thing but you won't because the monitor is not interacting with the environment in the same way. The traditional cab will send less frequency content to off-axis which is then reflected off the floor, walls and ceiling. The monitor will send more highs off-axis that are reflected. Our hearing relies a LOT on the spatial cues of reflection and the reflections will not be the same.
Compound the above with the fact that 99.9% of IRs are near field captures which sound nothing like the far field.
I believe trying to get a monitor to do amp in the room is a lesson in futility. If you really want that sound use a traditional guitar cab.
 You're not going to hear the same thing through FRFR that you heard from guitar cabs. Your audience will hear something very similar but you won't. What you're hearing through FRFR is a mic'd representation of the cabs. It takes some getting used to. You have to start thinking like a producer/engineer rather than a guitar player. If you start trying to dial out what you call "fizz" and "artifacts" you're going to end up with a tone that doesn't cut. It might sound good to you but it won't fit in the mix. That fizz and sizzle is what makes those classic rock tones work. Listen to some isolated tracks of VH and AC/DC and you'll hear a ton of high-end sizzle. In the mix, however, it's not noticeable. If you remove it then the guitar sounds dead.
 The sound of an amp in the "far field" is quite different than what you get with close-miking. IR's are made using close-miking and therefore sound nothing like listening to a guitar cab at distance from the cone.
Your audience does not hear the far field tone, they hear the close-miked tone as that's what is put through the FOH.
It can be quite an adjustment coming from far field amp tone to close-miked tone. Some people just never adjust.
Fortunately the Axe-Fx was designed to give you the best of both worlds. You can use the FX Loop and Output 2 to a power amp and conventional guitar cab while routing the fully processed tone with IR to the FOH. See the manual for full details. Rather than using your amp you can use a lightweight solid-state power amp and any of the new, lightweight guitar cabs that use Neodymium speakers. This gives you the classic far field amp tone for yourself in a lightweight package and the polished sound for the FOH direct from Output 1.
 Close-miked IRs typically have a lot more high frequencies than what you hear at a distance and off-axis from the speaker.
 […] All speakers "move air", that's the entire point of their design. Guitar speakers are inherently directional at higher frequencies. So when you stand off to the side you hear less highs. If you have two or four speakers the directivity gets even worse. FRFR speakers have less directivity. This combined with IR technology that almost invariably uses samples of a close-miked speaker and you end up with a different listening experience. To confuse the issue further many combo amps have an open back which changes the frequency response at the listening position even more.
Now, if you connect your Axe-Fx to a power amp and traditional 1x12, 2x12, etc. then you will get "amp in the room" but the "moving air" statement has no basis in fact.
 […] You can't compare what you are used to hearing "in the room". The close-miked sound ALWAYS has more highs and lows. This is due to the physics of near-field micing. And this is why a highpass and lowpass are frequently employed at mixdown.
 […] The classic method is "1W / 1m" which is to apply 1W and measure 1 meter away. When you get the microphone close to the speaker the response is much different and you usually get more highs and lows. This is "close miked" and is the technique normally used in studio recordings. During mixdown the producer/engineer will then often highpass and lowpass the signal to remove these excess highs and lows and to make the guitar "sit in the mix".
IRs are almost always made using the same close-miked technique and, hence, will sound like a raw recording. Far-field IRs are possible but very difficult to obtain requiring a large facility and special techniques.
Our primary goal is to model an amplifier and speaker as accurately as possible and the latest modeling is astonishingly accurate. We do not purport to be producers or mix engineers and leave the choice of low cut and high cut frequencies up to the user. Furthermore many users rely on the soundman to apply the filtering at the board, just as they would when mic'ing a "real" amp. More importantly the choice of frequencies is highly dependent upon the IR used.
 IRs are equivalent to close-mic'ing an amp. When you close mic an amp you almost always get more bass and treble than an "amp in the room". The extra bass is due to the proximity effect of the microphone. The extra treble is primarily due to the directivity of the speaker.
During mixdown engineers/producers will typically incorporate a low cut and high cut to help the sound "sit in the mix".
The thing to take away from all this is that an IR represents the close mic'd sound (unless using far-field IRs which are rare) and the close mic'd sound of an amp is much different than the "amp in the room" sound. As such it is common to use frequency shaping on a close-mic'd amp.
 The Axe-Fx is extremely accurate in duplicating the sound of a mic'd amp. Your monitoring thus becomes an essential part of the chain and accuracy is paramount. Many "FRFR" monitors are neither FR nor FR.
 FRFR is just not the same. Traditional head/cab you hear the sound from a bandwidth-restricted speaker at, say, 10 ft. In a typical modeler setup you are hearing what the "mic heard" when the IR was made and that mic was pushed up against the grill cloth.
One approach is to use "far field" IRs which are obtained using a measurement mic at a typical listening distance and angle. These are rare. There are a couple stock far-field IRs. They are indicated by (JM) for Jay Mitchell, who created them.
Even then it's still not the same because when you are using a traditional setup you move around while playing and the tone changes based on the angle. With a far-field IR the tone doesn't change with angle.
When I was gigging I used a power amp and cab behind me and sent the XLR outputs to FOH. More gear to lug but best of both worlds: traditional backline sound, consistent FOH sound.
 It's not the mic per se'. It's near-field vs. far field. Different mics sample the near-field differently. Mic'ing a speaker is sampling the near-field which sounds dramatically different than the far field. The response pattern of the mic samples the near-field and mics each have their unique pattern. Regardless, it's irrelevant. You'll never get monitors to sound like "cab in the room". If you want that use a SS power amp and cab.
FRFR is simply different. It's like mic'ing up the cab in an iso booth and listening from the control room. Therefore it becomes EXTREMELY dependent upon the FRFR speaker. (...) if you have access to some nice studio monitors I'd start there.
Apples and oranges. You're comparing FRFR to amp-in-the-room. They will never sound the same. And, IMO, those Matrix FRFR cabs sound like garbage but that's another story. When you use cabinet modeling into an FRFR you're recreating the sound of a close-mic'd amp. It's analogous to being in the control room while listening to your cab in an isolation booth. I.e., how records are made. If you want to compare to a head plugged into a cab you need to run the Axe-Fx into a power amp into the same cab. Get a *good solid-state or tube power amp and run that into a Marshall cab. A few tweaks and it should sound nearly identical.
 Far-field IRs are not the panacea some are making them out to be. Some things need clarification:
- A far-field IR will still not sound exactly like "amp in the room". The reason for this is that the dispersion of a guitar cabinet is very different than that of a FRFR speaker. An FRFR speaker has far wider dispersion at high frequencies, by design. With a guitar cabinet the low frequencies are less directional than the highs. This causes the cab to interact with the room differently.
So even if you capture a far-field IR it will not sound the same through a FRFR speaker.
- Most of the time we are not in the far-field of a guitar cabinet. At 10 kHz the far-field of a 12" speaker is about 18 ft. So usually we're in the far-field at some frequencies but in the Fresnel zone at others. At a typical distance of, say, 5 ft. we are only in the far-field at frequencies below roughly 3 kHz. Above that we are in the Fresnel zone.
- Because of #2 the sound at each ear can be quite a bit different. That six inches or so between our ears makes a big difference. When using a far-field IR the same sound will be presented to each ear. Even when in the far-field the sound changes pretty dramatically vs. angle because the dispersion is a function of frequency. One ear will hear more highs than the other.
- A cab with more than one speaker creates significant challenges. For example, a 4x12 has a far-field at 10 kHz that's roughly 100 feet! If you capture an IR of that cab at, say, 10 feet you are nowhere near the far-field. At anything other than nadir (aka boresight, 0 degrees) the individual speakers will contribute with different times of arrival. This results in extremely phasey sound (we were able to get some 4x12 IRs by using a special trick but in general you need to be very far away).
We don't hear this phasiness when listening to the real cab though because of #2. We get very different signals at each ear and our brain processes these. When using a Fresnel-zone IR of a 4x12 the same signal goes to both ears.
- Many guitar cabs are open back. A far-field IR of an open back cab through an FRFR monitor will sound very different because you're not reproducing the sound coming out of the back of the cab and bouncing off the walls.
- The sound of recorded guitar is near-field. This is what most people are used to hearing. So if you're trying to get the sound of your favorite record you won't get that with far-field IRs.
The takeaway from all this is that if you truly want the sound of amp in the room the best way to get that is to use an actual guitar cab. This isn't to say that far-field IRs are useless. They will give you a roughly similar sound to a guitar cab but it's just not the same.
 You'll never get the same experience using FRFR compared to AITR. It's physics. It's not a bunch of internet myth and pseudo-science about "mojo" and "tube magic.
 You'll never get monitors to sound like "cab in the room". If you want that use a SS power amp and cab. No amount of forum discussion is going to change physics.
The Owner's Manuals explain all the parameters.
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 block, and the other side into the other Cab block, for stereo separation.
Read Room ambience for more information.
Low Cut, High Cut, Filter Slope
Most IRs have been captured
close-mic'd, and produce a lot of highs and lows.
HIGH CUT and LOW CUT in the Cab block (
high-pass) allow you to EQ the material, preventing boomy bass and harsh sounds, which is equivalent to using the EQ controls on a mixing board to position the sound of the guitar in the mix. These are VERY important parameters to fine-tune the tone.
Important: While these parameters appear on the Preamp page in the Ares firmware and later, they are still operational when the preamp simulation in the Cab block is turned off.
Common settings are 80-200 Hz for LOW CUT, which cuts bass, and 5-10 kHz for HIGH CUT, cutting the treble. Of course: YMMV, as demonstrated by Justin York's (Paramore) approach:
 I think it all depends on what you’re going for. I always start with 20-20k and dial in the amp. An IR “hears” exactly what the mic picked up so start wide open if you want the sound of a real mic’ed cab. It sounds better in a mix, but for solo playing, cutting the highs may sound more pleasant to you.
For live sounds you may want to high cut, but I find that cutting a little 2k-4k 1-2dB gets rid of the fatiguing frequencies when cranked. Cutting a little top will slightly tame your tone, while cutting too much will take away note separation and you’ll lose yourself in the mix.
For recording, leave it wide open on the top and cut around 80 Hz on the low end.
Low Cut and High Cut are also available per individual IRs in the Cab block, including selectable filter slopes.
FILTER SLOPE in the Ares firmware and later selects between 1st order (6 dB/octave), 2nd order (12 dB/octave), 3rd order (18 dB/octave) and 4th order (24 dB/octave) filters for LOW cut and HIGH CUT, and lets you use different slopes for LOW CUT resp. HIGH CUT. The
pop when switching between the values is normal.
FRACTAL AUDIO QUOTES
 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.
 "LOWCUT FREQ" in the cab block sets sets the -3dB point of a highpass filter at the output of the cab block.
 If at the min/max the filters are off.
 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.
 The slopes are all maximally flat (Butterworth).
 I don't think there's a rule. Sometimes I'll drop it as low as 6K. Sometimes it's wide open. Depends on the cab.
Another way to tame the high end is the studio trick of placing the mics at different distances. Use the Align page to do this. As you separate the IRs in time it will put a notch in the high frequencies.
 Touched on one of my favorite topics here, so I did want to chime in about cuts. Something I discussed a good bit in my AxeFest clinic. Nothing wrong with sometimes very aggressive cuts in the cab block, especially with close mic IRs, as Cliff said. Close mic IRs bring in so much of that amazing depth and complexity we love in the upper mid-range of a tone but often introduce too many other frequencies for a balanced mix. In my experience it comes as a surprise to many how narrow the frequency band of guitar tones is much of the time in pro mixes, whether live at front of house or at the board in a studio. What sounds rockin' to you solo isn't always very relevant to what sits well in a mix. Cutting down to 4-5k and up to the mid or even high 100s not only isn't unheard of but would more be described as common practice. There are a lot of instruments in a band. Concentrate on letting the guitar speak where it speaks best! Just one of the inumerable strengths of the Axe-Fx is that we have the ability to sculpt with this kind of detail with the click of a mouse. Something standard amps can only dream of!
This is a demonstration of slope:
In the Ares firmware and later, IR Length applies to individual IRs in the Cabinet block. Shortening the length can remove room reflections and/or decrease CPU usage.
On the FM3 and FM9 DynaCabs can't be shortened so this parameter doesn't appear on those devices when using DynaCab mode.
Read IR length for more information.
This 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
amp/cab in the room experience. This is especially helpful when using multiple impulse responses.
Smoothing is identical to De-phase in the Axe-Fx II.
Smoothing available for Legacy and DynaCabs on the Axe-Fx III.
Cab-Lab can apply this process when mixing impulse responses together to produce an IR with the effect built-in. Doing this allows the AX8, FM3 and FM9, which do not support hardware Smoothing, can benefit from this feature.
The processing required is extreme and the control can have some lag. No extra CPU usage or audio latency, however, is incurred.
FRACTAL AUDIO QUOTES
 […] 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.
 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.
 It's so simple that even experts in the field don't realize why it works.
 It smooths the IR in the frequency domain.
 Cabinet smoothing does not increase CPU usage. Must be something else you changed.
Why is De-Phase necessary?:
 You don't listen to a guitar speaker with your ear against the grill cloth.
Air mixes some of the direct signal entering the Cab block with the
cab-processed signal leaving the Cab block. This adds some so-called
air to the sound, which some users find to add realism to the tone.
Air is not supported on the AX8.
AIR FREQUENCY lets you adjust the cutoff frequency of the mixed signal. Increase the frequency to its maximum value for a straight mix.
Tip: If you want to listen to just the
air part of the signal, select an empty user cab, and turn up Air.
Warning: Adding air can cause phasing issues when using multiple un-aligned IRs.
FRACTAL AUDIO QUOTES
 Air is just clean signal mixed in. It WILL cause phase issues if the IRs are not minimum-phase or delayed.
 Air is nothing more than low-pass filtered direct signal mixed with the processed signal. Sometimes adding some Air can help remove the boxiness. You typically need to set the Air Freq above 3 kHz before the effect is noticeable. I like it around 3500 or so. It adds a little sizzle to high-gain tones and removes that boxy sound.
 There was a change to the Air stuff. I've been tempted to remove Air or change it to a shelving filter because it causes problems like this, especially with non-minimum phase IRs.
This parameter sets a micro delay for stereo applications. When running a Cab block 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.
The Ares firmware and later use millimeters in the MIC DISTANCE parameter. Before that, milliseconds were used. To convert, multiply the old value in milliseconds by 343.
Firmware 17 for the Axe-Fx III and later increase Mic Distance to 3.4m, which is ~11 feet. This allows delaying room mic IRs that have been trimmed to remove the leading silence. The alignment graph features a Zoom control that changes the abscissa between 3ms and 12ms.
If you want to use this parameter but don't want IR coloring, use a FLAT or NULL IR. The last stock cab in the Axe-Fx III and FM3 is a flat IR.
FRACTAL AUDIO QUOTES
Attributed to Cliff:
 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.
 Mic distance is just a delay control, it doesn't alter the sound if you are using just one IR.
 It's no different than using a delay block and dialing in a very short delay with mix = 100%.
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.
 […] 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.
See DynaCabs above.
The Cab block in firmware Ares and later displays an alignment graph showing a zoomed time series of the IRs. This allows visual adjustment of the mic distance. When using IRs that have not been minimum-phase processed, this facilitates aligning the IRs.
If you're unable to line up the IRs, you probably need to trim an IR to remove leading silence using Cab-Lab or the editor.
Deliberately mis-aligning DynaCabs can add a desirable twist to the sound. Watch the video below.
FRACTAL AUDIO QUOTES
 I've added a time display to the cabinet block which shows all four IRs on the same axis and allows you to adjust the mic distance (delay) of each on that display. This allows you to precisely time align the IRs.
 The graphs show the first 128 samples of the IRs. When engineers/producers record cabs in the studio they go to great pains to make sure the mics are time-aligned. A shift of just a few mm can make a big difference.
In the virtual world we can do the same thing. The graphs show the time series and you can adjust the virtual distance of the microphones to change their time alignment.
 […] Producers/Engineers often use two mics and mix them together to intentionally create phase cancellation. They'll move one mic away from the speaker a bit to get a longer delay which causes comb filtering when combined with the other mic. This is analogous to the Align tab on the Cabinet block.
 It's analogous to moving the mic in and out.
 You should align impulses ideally but sometimes a little misalignment adds character.
 […] Another way to tame the high end is the studio trick of placing the mics at different distances. Use the Align page to do this. As you separate the IRs in time it will put a notch in the high frequencies.
 The files are initially aligned to the same reference point -- a universal standard across all DynaCapture IRs.
In other words, IRs for mics that are farther away have been time compensated to be aligned with IRs for mics that are closer.
You can use the controls on the Align page in the usual way, however, to add time to any individual IR.
 […] don't miss the fact (as indicated by the small text labels) that the data shown on the Cab Block's ALIGN page (and in Axe-Edit) is windowed from just 0 to 3 ms. That's the very beginning of a full UltraRes capture, "zoomed" so you can easily align the critical peaks in this region for close-miked IRs.
If you are looking at the raw data for a "ROOM" or "FAR FIELD" IR, you will very likely have NOTHING on the ALIGN plot for that IR, because the sound will have taken longer than 3ms to reach the capturing microphone. Not that you'd ever need to align a room, but you will need to use Cab-Lab if you want to visually inspect the contents of such IR files.
This adjusts the proximity of the virtual mic to the virtual speaker. Higher numbers replicate the mic being closer to the speaker (near-field), causing an increase in low frequency response (more bass). Lower numbers replicate the mic being further away from the source, with the lowest number providing far-field coloration.
On the Axe-Fx II, proximity only works when a mic model has been selected, including the NULL type. The AX8 and Ares firmware do not support mic modeling, but do provide a PROXIMITY parameter.
The PROXIMITY FREQUENCY parameter lets you tune the frequency range over which the proximity effect occurs.
DynaCabs do not provide a PROXIMITY parameter because the DISTANCE parameter manages this.
This parameter was part of the Amp block before the Quantum 9 firmware on the Axe-Fx II and AX8. In Quantum 9 for the Axe-Fx II it was replaced with Speaker Compression in the Amp block. Motor Drive is still present in the Cab block in the Axe-Fx II but not on the AX8. It models the effect of high power levels on the speaker.
When using two UltraRes cabs in a preset, don't use Motor Drive on just one of these, because this will introduce comb filering because of phase cancellation.
FRACTAL AUDIO QUOTES
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.
 Motor drive isn't EQ. It models efficiency reduction due to thermal effects.
 What I have found is that thermal compression is somewhat noticeable and measurable. This is modeled by the Motor Drive parameter.
 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.
 Motor Drive simulates power compression due to voice coil heating.
 Guitar loudspeakers are intentionally designed to compress. FRFR speakers do compress a bit but not nearly to the extent that guitar speakers do.
 Makes edge-of-breakup tone stupid easy.
 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.
 The thermal time constant of a typical guitar speaker is about 0.52 seconds. Magnetic time constants are zero.
 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.
 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.
 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.
This parameter was available only on the Axe-Fx II.
It allowed the user to change the relative size of the virtual speaker. This is controlled through the parameter SPEAKER SIZE. This parameter appears only if the selected IR is not UltraRes and the Cab block’s mode is Mono. Adjusting this parameter also changed the resonance of the selected cab.
This was available on the Axe-Fx II and AX8 only (except for the Proximity effect).
Microphone modeling relied on impulse responses.
FRACTAL AUDIO QUOTES
 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.
 The mic models are actually IRs. The mic IR is convolved with the speaker IR to create a composite final IR.
 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.
 The mic options are mostly legacy. I never use them but if we took them out there would surely be much protestation.
See the Wicked Wiki article for more information.
57 DYN — Shure SM57
58 DYN — SM58
421 DYN — Sennheiser MD 421 II
87A COND — Shure Beta 87A
U87 COND — Neumann U87
E609 DYN — Sennheiser e609 Silver
RE16 DYN — Electro-Voice RE16
R121 COND — Royer Labs R-121
D112 DYN — AKG D112
67 COND — Neumann U67
NULL — 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.
Forum member Moke created Tone Matches of the mic models in the Axe-Fx II and saved them as IRs.
Tips, tricks and troubleshooting
IR Player block
The IR Player block on the Axe-Fx III can process a single IR, offers fewer features than the Cab block and therefore requires less CPU. The FM3 and FM9 do not provide this block.
The Amp block provides a number of parameters which are closely related to the Cab block, including:
- LOW FREQUENCY RESONANCE
- HIGH FREQUENCY RESONANCE
- SPEAKER IMPEDANCE CURVE (SIC)
- CABINET RESONANCE
- SPEAKER DRIVE
- SPEAKER THUMB
- SPEAKER BREAKUP
- SPEAKER COMPRESSION
- SPEAKER COMPLIANCE
- OUTPUT MODE
- AUTO DYNACAB IMPEDANCE
Record 4 different cabinet signals
This G66 tutorial on YouTube demonstrates how to create four separate cabinet signals in the Axe-Fx II, to be mixed at will. It comes down to using two stereo Cab blocks, with one of these connected to an 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 four separate channels on the mixers.
The Axe-Fx III makes this much easier to accomplish, because its Cab block supports four IRs per channel.
When you use an external IR in a preset and you want to share the preset/sound, you need to share the preset as well as the impulse response. There are two ways around this:
- Integrate the impulse response in the preset by replacing the Cab block with a Tone Match block, after having captured the tone of the Cab block.
- Create a Preset-Cab bundle, if the firmware and editor support this.
It's a license violation and NOT permitted to share commercial IRs!
FRACTAL AUDIO QUOTES
 IRs are stored in FLASH. FLASH memory can become corrupted though it is rare. If you've ever plugged a USB memory stick into your computer and some of your files were damaged that's the same thing. Reloading the IR is the solution.
FX8 and speaker simulation
The FX8 does not provide cabinet modeling or another way to process IRs. But you can use its tools to get close.
To simulate a speaker, use the approach described in
Combined, these tricks let you simulate both an amplifier and speaker cabinet, without Amp and Cab blocks.