Revision as of 14:51, 20 March 2018

List of I/O ports

Axe-Fx III

INSTR (front): 1/4” phone jack, unbalanced, conditioned for guitar use, auto-switching, 1 Megaohm (adjustable), +16 dBu instrument level
INPUT 1 (rear): 1/4” phone jack, unbalanced, conditioned for guitar use, auto-switching, 1 Megaohm (adjustable), +16 dBu instrument level
INPUT 2 (rear): XLR Female and 1/4” combo, L/R, balanced, 1 Megaohm, +20 dBu line level
INPUT 3 (rear): 1/4” phone jack, L/R, balanced, designed for unity gain applications such as 4CM, dual stereo inserts or general purpose, 1 Megaohm, +20 dBu line level
INPUT 4 (rear): 1/4” phone jack, L/R, balanced, designed for unity gain applications such as 4CM, dual stereo inserts or general purpose, 1 Megaohm, +20 dBu line level
OUTPUT 1: XLR, L/R, balanced, ground lift switch, 600 Ohm, +20 dBu line level
OUTPUT 1: 1/4" phone jack, L/R, HumBuster, ground lift switch, 600 Ohm, +20 dBu line level
OUTPUT 2: XLR, L/R, balanced, ground lift switch, 600 Ohm, +20 dBu line level
OUTPUT 3: 1/4" phone jack, L/R, HumBuster, 600 Ohm, +20 dBu line level
OUTPUT 4: 1/4" phone jack, L/R, HumBuster, 600 Ohm, +20 dBu line level
A/D and D/A conversion: 48 kHz, 24 bits, 114 dB dynamic range, 20 Hz - 20 kHz frequency response (0/-1 dB)
Digital I/O: S/PDIF (RCA Coaxial), AES (XLR), USB Audio 8x8, sample rate 48 kHz
MIDI: IN, OUT, THRU
Headphone output: 1/4" stereo jack, 35 Ohm
Expresssion pedal ports: 2x 1/4" TRS, 10-100 kOhm, momentary or latching
FASLINK II port: XLR Female

Output 1's XLR and 1/4" ports can used simultaneously. They're buffered.

The XLR outputs are protected against phantom power from the console.

The Axe-Fx III has multiple Input and Output blocks that can be placed anywhere on the grid.

Axe-Fx II

IN 1 (INSTRUMENT): 1/4” phone jack, unbalanced, max +16 dBu (conditioned for guitar use), instrument level
IN 1 (rear): 1/4” phone jack, unbalanced, max +20 dBu
IN 2 (FX RTN): 1/4", L/R, balanced, 1 Megaohm, max. 20 dBu
OUT 1 MAIN: XLR, balanced, 600 ohm, max. output +20 dBu
OUT 1 MAIN: 1/4” phone jack, unbalanced (hum-canceling)
OUT 2 (FX SEND): 1/4", L/R, unbalanced, Humbuster, 600 ohm, max. 20 dBu
Digital I/O: S/PDIF (RCA Coaxial), AES (XLR), USB Audio

Output 1's XLR and 1/4" ports can used simultaneously. They're buffered.

The XLR outputs are protected against phantom power from the console.

"Both outputs should work simultaneously. They are actually buffered so even if you shorted one it shouldn't affect the other." source

AX8

IN 1 (INSTRUMENT): 1/4", mono, unbalanced, 1 Megaohm (fixed), max. 16 dBu, instrument level
OUT 1 (MAIN): XLR, L/R, balanced, 600 ohm, max.20 dBu
OUT 1 (MAIN): 1/4", L/R, unbalanced, Humbuster, 600 ohm, max. 20 dBu
IN 2 (FX RTN): 1/4", L/R, balanced, 1 Megaohm, max. 20 dBu
OUT 2 (FX SEND): 1/4", L/R, unbalanced, Humbuster, 600 ohm, max. 20 dBu
S/PDIF digital out, 24-bit, 48 kHz (fixed)

Output 1's XLR and 1/4" ports can used simultaneously. They're buffered.

The XLR outputs are protected against phantom power from the console.

FX8

IN [PRE] (INSTR): 1/4", mono, instrument level, unbalanced, 1 Megaohm (depending on Input Impedance setting), max 16 dBu, instrument level
OUT [PRE]: 1/4", L/R, (L/Mono), unbalanced, Humbuster, 600 ohm, max 20 dBu
IN [POST]: 1/4", L/R, line level input (+4 dBu), 1 Megaohm, balanced, max 20 dBu
OUT [POST]: 1/4", L/R, unbalanced, HumBuster, 600 ohm, max 20 dBu

Things you must know

Balanced versus unbalanced

Unbalanced audio signal: carried over a two-conductor cable. The most common cable is a 1/4" guitar cable, where the ground wire wraps around the positive wire. Unbalanced cables are generally good for running signal up to several meters (< 10).
Balanced audio signal: carried over a three-conductor cable which connects a balanced input and balanced output. The two signal wires carry identical copies of the signal, with one of the wires 180 degrees out of phase with the other, creating a differential. At the receiving side the two signals are brought back into phase with one another, and the induced noise will be canceled. These cables usually use XLR or Tip-Ring-Sleeve (TRS) connector end-types. The ground wire wraps around the signal wires and acts as a shield. A balanced connection supports the use of a ground lift switch and noise-free long cable distances.

A DI box converts signals between balanced/unbalanced.

More information in this Wicked Wiki article and Wikipedia.

"XLR is only necessary for long cable runs where there is the danger of picking up interference on the cable. Anything less than, say, 10 meters and unbalanced is fine." source

(FX8) "For most uses an unbalanced TS cable is fine. The inputs are balanced so that you can get even more hum rejection by using a TS-to-TRS cable from the amp's send." source

(Axe-Fx II) "The unbalanced and XLR outputs are the same level in the Axe-Fx II. In the original Axe-Fx the XLR outputs were 6 dB hotter. This is not the case with the II." source

Instrument level versus line level

Audio signals operate at:

Microphone level: lowest output level, used with microphones and microphone inputs on mixers.
Instrument level: level of guitars, basses and effects pedals.
Line level: highest output level. Line level can be:
- Consumer level: -10dBv.
- Professional level: +4dBu. Commonly used in 19" processors, line inputs on mixers and monitors.

Outputs on Fractal Audio gear operate at line level. Some are adjustable between +4 and -10.

More information in Wikipedia.

Humbuster

The Axe-Fx III, Axe-Fx II, AX8 and FX8 support Humbuster outputs. More information.

Unity gain

What is unity gain? Unity gain means that the input level is equal to the output level.

When does unity gain matter? It is not important when connecting the device to an amplifier or mixing console. It is important in setups where the device is being used as an effects-only processor (e.g. as a pedalboard or in an amp's effects loop) or when using the Four-Cable-Method (4CM) to connect to a guitar amplifier.

How to set up for unity gain? To set up an input/output for unity gain, set the corresponding Output knob to its maximum position. To test: fill the grid with shunts, and you should get exactly the same signal at the output which you put in.

"Unity gain mode is a special mode designed for use with the 4CM. When you turn the output levels all the way up whatever you put in you get out (assuming all unity-gain blocks in the chain). If you have an amp block in the chain then you have tons of gain and therefore no longer have unity gain." source

"With the Axe-Fx volume all the way up you would be pushing +20 dBu into the amp which could clip the inputs to the amp. Unity gain mode is only desirable for 4-cable-method." source

FX8 and unity gain (from the Owner's Manual):

Why do I care that the FX8 is designed for unity gain?
A: The FX8 makes it EASY to achieve unity gain. This can be important because amplifier tone, distortion amount, dynamics and noise are level dependent. With unity gain:

The level of the signal from your guitar output can reach your amp input without being altered. Therefore, your guitar-amp interaction sounds and feels the same, offering a transparent playing experience while using the FX8.
The level of your FX SEND can reach your FX return without being altered. The entire system can therefore perform optimally, without unpredictable changes to level, dynamics or noise when you engage True Bypass or bypass all post-effects.

Q: How do I set up the FX8 for unity gain?
A: You don’t need to! Just set up according the basic instructions in Section 3. A default empty preset should sound have the same level as True Bypass Mode.

Q: What might I do to inadvertently upset unity gain?
A: Many SETUP and EFFECT parameters change the gain level. Some of these are intended to change gain levels (how else is a boost supposed to work, after all?) Here is a short list of things to consider:

The LEVEL parameter of every effect increases or decreases the overall level.
Changing MIX on certain effects changes both dry and wet levels. This is to prevent signals from “stacking up” and causing clipping. You can compensate with your ears by turning effects on and off and comparing the level with True Bypass engaged.
If you’re going to change a block’s BYPASS MODE from the default setting of THRU, it is best to check its levels when you engage/disengage the effect BEFORE you switch to something like MUTE FX IN.
The level parameters on the OUTPUT page of the main mode menu increase or decrease overall levels. Incorrect settings on the I/O: AUDIO page can result in gain changes.
The NOISE GATE has a level control.
If your rig is MONO, every BALANCE or PAN control can affect levels.
The Global Graphic EQs affect overall level.
The I/O LEVEL page settings DO NOT affect unity gain. Each setting is compensated internally.

Q: Any last words of advice?
A: Use the TRUE BYPASS switch as a way to make sure your presets and scenes are on track. In general, it is better to be in control of your levels than to be fixated on the “concept” of unity gain. Do what sounds best to you and learn as much as you can about your gear.

Decibels

Cliff's tech notes about decibels:

"The decibel is a unit of measurement that gives the ratio of the power of one signal relative to another. The formula for the decibel is dB = 10 * log_10(P1 / P2) where P1 and P2 are power measurements. The reason it is called a decibel is because it is 10 bels. One bel would be log_10(P1/P2).
The important thing to understand is that the decibel is a RATIO of powers. A dB is meaningless without a reference power. So if someone says "that signal is 86 dB" that is a meaningless number as it has no reference.
Decibels are convenient because they convert logarithmic perception to a linear scale. Human hearing, for example, is logarithmic. Many other natural phenomena are logarithmic which means that the phenomena exists in the "multiplication domain" as opposed to the "addition domain". For example, human vision is logarithmic. We perceive light such that the light must double for it to appear twice as bright. If we were to plot that we would have an exponential curve of light intensity vs. perceived brightness. If we take the logarithm of the intensity instead we get a straight line. This is why cameras use f-stops which are a base-2 logarithm.
So, back to reference levels. There are many reference levels used in dB: dBm, dBu, dBV, dB re. kPa, etc. dBm refers to the power referenced to one milliwatt. If the measured power is, say, 100 mW then that would be 10 * log10(100/1) = 10 * log10(100) = 20 dBm. dBV is a voltage ratio and not really a true dB but, regardless, is still commonly used. The formula for dBV is 20 * log10(V1/V2) since we need to square the voltage to get the power.
In audio a common unit is dBu. dBu is the power relative to the voltage into a 600 ohm resistor that is dissipating 1 mW. This is roughly 0.77 volts. Back in the early days of telecom 600 ohms was the standard termination impedance, hence the dBu. Most pro audio gear runs at +4 dBu. What does that mean? 0 dBu is 0.77 volts so +4 dBu would be 4 dB greater, or about 1.22 volts. To go from dB to volts the formula is 10^(dB/20).
Consumer audio gear usually runs at -10dBV, or roughly 0.32 volts.
When recording your goal is to get your signal level near the nominal signal level of the equipment being used. This ensures the best S/N ratio. Many recording consoles use VU meters which are calibrated such that "0 dB" is +4 dBu. The goal is to get your signal level around 0 dB.
Well-designed gear has some amount of "headroom". Headroom is the difference between the maximum signal level and the nominal signal level. For example, the Axe-Fx II has a maximum signal level of +18 dBu. If operating at +4 dBu nominal this gives 14 dB of headroom which means that any signal peaks can be over four times higher.
In digital gear we encounter the dBFS, which is dB relative to full-scale. Full-scale is a term that indicates the maximum signal level into or out of an A/D or D/A converter, respectively. With digital converters the best performance is achieved by operating the converter such that the nominal signal level is close to full-scale. The exact voltage is unknown and irrelevant. Most digital gear will have indicators that measure the levels relative to the converter's full-scale value. For example, the input meters on the Axe-Fx indicate the input signal relative to the A/D converter's full-scale value. The "tickle the red" advice aims to operate the A/D converter near its full-scale value as the red LEDs light at 6 dB below full-scale, or -6 dBFS."source

"Decibels are decibels. There is no such thing as "root-power decibels".
By definition a decibel (dB) is a ratio of two powers. The formula is 10 * log10(P1/P2) where P1 and P2 are the power of two signals, respectively.
In electronics, however, we usually manipulate and measure voltage levels. It's convenient to represent the ratio of two voltage levels in dB. To do this you would need to square the voltage to get the power (since P = V^2 / R). We also assume R = 1 for convenience. With a little math you get dB = 20 * log10(V1/V2).
Therefore if we reduce the voltage level of a signal by a factor of 0.1 then the signal is now -20 dB relative to before.
dB is simply an easy-to-read logarithmic-to-linear mapping. Music, human perception, and many other things in nature typically have a logarithmic response. The decay of, for example, a cymbal is logarithmic. If you plot this on a linear axis it's hard to display because of the dynamic range. But if you use a logarithmic axis you "compress" the data into something that's easier to view. Decibels are just a widely accepted mapping. You could use any base for the log; log2, ln, etc but since we have 10 fingers log10 is nice.
The point is that X dB is X dB. If you reduce a signal by 20 dB you've reduced it's voltage to 10% of what it was previously. You also reduced it's power to 1% of what it was previously. These are the same things: 20 * log10(0.1) = 10 * log10(0.01)." source

Connecting instruments and other devices

Instrument input

Axe-Fx III: INSTRUMENT (front and rear, auto-switching).
Axe-Fx II: INSTR (front).
AX8: IN 1 (INSTRUMENT).
FX8: IN 1 (PRE).

The Instrument input uses a proprietary circuit and a dedicated A/D converter to lower noise. It's conditioned for guitar through hardware and software ("Secret Sauce"). For best results, use the instrument input for guitar, whether wired or wireless, electric or acoustic, except when running a line level signal.

"The input buffer is designed for ~10Vpp max." source

Axe-Fx III: The Axe-Fx III has two instrument inputs: front and rear. The specifications are the same. The rear is meant to be used with racks, wireless units and such. Using the front input, for example with a cable, ALWAYS overrides the rear input. This does not require configuration in Setup.

Secret Sauce: The AX8, Axe-Fx II XL and XL+ feature “Secret Sauce III”. The Axe-Fx III features "Secret Sauce IV" circuitry on the front and rear instrument inputs. This lowers the noise floor using a proprietary technique along with special analog input circuitry.

"The "Special Sauce III" uses a combination of things to get a lower noise floor. One of these things is new, premium Burr-Brown op-amps in the signal path which have extremely low noise and distortion (and are very expensive). As always I don't design stuff to be cheap, I design it to be good." source

"The spectrum of a guitar is pink(ish). Above 800 Hz or so the energy rolls off dramatically. As luck would have it, humans perceive noise above 800 Hz or so to be most objectionable as it manifests itself as hiss. So the front input pre-emphasizes the high frequencies and then does the inverse in software. This has the net effect of a flat frequency response but pushes the noise floor down by the amount of the pre-emphasis. It's an old trick, used in FM radio and vinyl records. The basic premise is to optimize the data conversion to the information content of the source."

(Axe-Fx II) "You have to set the input selection to match the input you're using. If you're using the front input then you must set the input selection to front and vice-versa. If you plug something into the front and set the input selection to rear it will get MUCH brighter. The front input is optimized for guitar level inputs and has spectral shaping and more gain than the rear input. The front input is optimized for guitar pickups. This is a combination of hardware and software processing. If you set the input source to Analog Rear this turns off the software processing part. If you are plugged into the front it will change the tone since you're still going through the hardware processing. This is why I say you must match the input selection to the input you are using. The rear inputs are standard line-level inputs and can be used with any program material. The front input, as stated above, is optimized for guitar pickups. As such it has more gain and less headroom and may clip if used for non-guitar program material. If you plug a guitar directly into the rear you may find you don't have enough signal level."

Input impedance:

Axe-Fx II, III and FX8: adjustable input impedance on the instrument input (Axe-Fx III: front and rear).
AX8: fixed input impedance (1 Mohm).

More information about input impedance.

Switching between different guitars

When switching between guitars, there will be differences in level and tone. That may be okay, because why would you use different guitar otherwise? But if you want to adapt presets to accomodate different guitar, several things are possible:

Use different presets for different guitars.
Use a separate input for each guitar and assign each its own signal chain.
Adjust Level in the Input block. This parameter controls the loudness of the signal entering the grid. It was specifically introduced for this purpose: compensating output level differences between guitars. It works per preset only, so it needs adjustment per preset, unless the setting is stored as part of a global block (Axe-Fx II only). It's not possible to attach a modifier.
Set up X/Y or channels in the AMP block for different guitars, using different values for Input Trim for example.
Attach a modifier to Input Trim in the AMP block, connected to a pedal or switch.
Set up a different AMP blocks (not on AX8) for each guitar.
Use scenes and scene controllers, attached to Input Trim in the AMP block.
Adjust Amp Gain ion the Global menu (not on Axe-Fx III).
Axe-Fx II only: add a low-CPU block to every preset. Like FILTER or VOL or PEQ (PEQ and FILTER allow additional EQ-ing). Put it at the start of the grid to make it affect the amount of gain in the Amp block. Keep the block neutral and set its Level at i.e. -6 . Make it a global block, so you can easily change a setting and have it applied across all presets immediately. Attach its Bypass parameter to an external controller. Engage this block by going into I/O > MIDI and toggling EXT CTRL xx INIT VAL between 0% and 100%. Or assign a general function footswitch to the external controller’s CC and use that for toggling instead (set the switch to Global:Yes in the MFC). It works across all presets.

Wicked Wiki: Settings for Different Guitars.

See section 4 of the Owner's Manual for a preset example for a dual output guitar, such as magnetic + piezo.

Multiple instruments simultaneously

Axe-Fx III: Guitar #1 connects to the Instrument input (front of rear), guitar #2 connects to Input 2. Even a 3rd and 4th instrument can be connected. Each can have its own signal chain on the grid, and its own output if desired. There are two Amp blocks, so 2 instruments can make use of amp modeling. But you don't need an Amp block for an acoustic guitar, or piezo, or an electric that should sound like an acoustic. Perhaps neither for bass, with the help of the B7K drive model. You may even get away with a Drive block and a Cab block for clean guitar tones (at the cost of dynamics). Also see section 4 of the Owner's manual.

Axe-Fx II:

Set Input 1 to Stereo.
Connect one instrument to the front, and the other to rear Input 1 Right.
Use two rows on the grid.
Add a VOL block to each row. Set one to Input left (for the instrument that goes into the front input or rear input 1 Left) and the other to Input Right (for the instrument that goes into rear input 1 Right).
Add an AMP block after each VOL block if necessary. You can also leave out the VOL blocks and set Amp 1 to Input Left, and Amp 2 to Input Right; this only works with the Amp blocks in the first column.
Continue the rows to the end, adding a CAB and effects to each one if necessary, or merge them if desired. Keep the signals separated by using Balance controls.

AX8: To use the AX8 with 3 devices: guitar, and two other devices such as piezo or synth:

Add an Amp and Cab.
Put the FX Loop block after the Cab block. This sends the regular guitar sound to Output 2, and lets signals from Input 2 (left and right) enter the grid.
Split the signal after the FX Loop block into two rows, and add a Volume block to each one.
Set Input Select in one Volume block to Right Only. Set Pan and Balance as desired.
Set Input Select in the other Volume block to Left Only. Set Pan and Balance as desired.

More information:

Acoustic instruments

The instrument or Input 2 can be used to connect an acoustic instrument. On the Axe-Fx III Input 2 can also be used.

The IR of an acoustic body can add acoustic resonance to the tone. Always use Ultra-Res IRs when using acoustic IRs when available. There are no acoustic IRs among the stock cabinets. You can find some here:

A preset for an acoustic guitar with good pickups can be kept simple: compressor, EQ and reverb. An Amp block is not required, but the TUBE PRE model can help warm up the tone.

Audio clip and preset for a piezo-equipped guitar.

You can also use Tone Matching with great results. Tutorials on G66.eu.

Bass guitar

The Axe-Fx III, II and AX8 provide stock bass cabs and some bass amp models.

The tuner supports bass guitar tuning. On the Axe-Fx III, pitch detection on bass guitars has been improved.

Forums:

Pedals

Between guitar and processor: If you want to connect an effects pedal to the Axe-Fx II, III or AX8, with the processor configured for amp and cab modeling, connect it between the guitar and the instrument input on the processor. Remember to check the input impedance on the Axe-Fx II, III and FX8, and also verify the input level.

In effects loop: Alternatively, insert the pedal in an effects loop. Make sure to adjust levels where needed. You can include/exclude the pedal per scene, or use the effects loop as an audio switcher.

(about running instrument-level pedals in the loops of the Axe-Fx III)
"The outputs have adjustable pads." source

Between AX8 or FX8 and amplifier: A pedal can also be placed between the AX8 and FX8 and an amplifier, when using the processor as a virtual pedalboard. Make sure to adjust the levels where needed.

Wireless

A wireless receiver can connect to the instrument input or another input.

(Axe-Fx II) "The front input has a better SNR but if you are using a wireless the better SNR of the front input won't be noticeable since the noise of the wireless will dominate." source

Play sound from computer or other device

Through USB Audio: Audio from the computer can be played through the Axe-Fx II and III through USB Audio. The signal will not enter the grid and will be streamed to Output 1. To adjust the volume, adjust the volume in the audio application. The audio can't be routed to another output of the Axe-Fx II. To route the signal to another output of the Axe-Fx III, see section 3 of the Owner's Manual (Mac) or use ASIO (Windows).
The AX8 and FX8 don't support USB Audio (input nor output).

Through Inputs: External audio can enter the processor by connecting a computer or music player to the inputs.

Axe-Fx III: output 3 L/R or output 4 L/R would be the first choice.
Input 1 (rear) on AX8 and Axe-Fx II: make sure to set it to stereo, use a preset with shunts only or set the Axe-Fx II to Bypass mode.
Input 2 on AX8 and Axe-Fx II: the FX Loop block must be used and connected to the grid output.

How to send signal from Input 2 to Output 2.

Microphones

A microphone should only be connected to Input 2 (or 3 or 4). The Axe-Fx II, III and AX8 do not have microphone preamps, so there will be a level mismatch. The microphone's signal is too low and the signal-to-noise ratio will be high. You'll need to increase the level of the source to get sufficient signal strength into the unit. This can be accomplished by using amp external mic preamp, or a device like Shure's A85F adapter.

(Two awesome options for between a mic and Axe-Fx III are the Summit Audio TBA-221 and the FMR-RNP. A nice inexpensive choice is the Rolls MP13. With one or two of these, you could sell the interface." source

Sustain and feedback

It's as easy to get your guitar to feedback as it is with a regular amp and cabinet (with the exception of headphones). If you don't succeed, experiment with the Output Phase parameter in the I/O menu.

There's no dedicated "sustainer" or feedback effect (like Digitech's FreqOut) in the Axe-Fx and AX8. Forum member Simeon created a feedback simulation preset, mimicking controlled feedback.

Setting the main input level

Axe-Fx III: I/O > Input.
Axe-Fx II: I/O > Input > Instr In / Input 1 / Input 2.
AX8: I/O > Levels > IN 1 (Instrument) Pad / In 2 (Fx Rtn) Nominal Level.
FX8: I/O > Levels > Input 1 (Pre) Pad.

What is Input Level or Input Pad for?
Input Level and Input Pad are NOT GAIN controls. They do NOT affect the overall volume level, or clipping or amp gain (unless you go below 5%). It just optimizes the signal-to-noise ratio of the analog-to-digital converters. The adjustment is applied before the A/D converter and is offset by a corresponding but opposite boost at the output of the converter.

How to set Input Level or Input Pad
Set Input Level or Input Pad to make the red Input LED blink only occasionally while strumming ("tickle the red"). Strum hard, on the loudest pickup! There's a substantial range between orange and red. If you can't make it blink red, don't worry.

The red light turns on BEFORE the instrument input clips. It means "Warning, you're approaching clipping" as opposed to "Warning, you ARE clipping". On the Axe-Fx II, AX8 and FX8 the red light turns on at -6dB of the point where the signal is hard limited / clipped. On the Axe-Fx III the red LEDs on the front panel come on at -1 dBFS!

While the word "clipping" is used here, in reality the input signal never really clips, because of a limiter will kick in.

Device-specific comments:

AX8 and FX8: higher Input Pad value means a lower input (padding). Run it as low as possible, because padding increases noise floor.
Axe-Fx III: the instrument input on the Axe-Fx III is a little more sensitive than on the Axe-Fx II, but it has more headroom / dynamic range. Do not set it below 5%, because at this point gain may be affected. Also be aware that the red LEDS on the front panel come on at -1 dBFS. The front panel LED meter bridge provides instant visual status for the inputs.

Note: when using a mono instrument, do not use an input that's set to Stereo or Sum L+R, use Left Only (default)

(Axe-Fx II) "For a Strat, near 100% on the input level is not unusual. I run my Strat around there. It has vintage-type pickups." source

(Axe-Fx II) "To get the best noise performance it is important that the Instr In trim is set correctly in the I/O->Input menu. Set this as high as possible without clipping the input." source

(Axe-Fx II) "You don't HAVE to tickle the reds. Adjust for your hottest guitar and leave it." source

(Axe-Fx II) "The AFXII has digitally controlled potentiometers before and after the A/D and D/A converters. Therefore it knows what the input and output gains are. It compensates for these gains in the digital path." source

(Axe-Fx II) "Full-scale is a term that indicates the maximum signal level into or out of an A/D or D/A converter, respectively. With digital converters the best performance is achieved by operating the converter such that the nominal signal level is close to full-scale. The exact voltage is unknown and irrelevant. Most digital gear will have indicators that measure the levels relative to the converter's full-scale value. For example, the input meters on the Axe-Fx indicate the input signal relative to the A/D converter's full-scale value. The "tickle the red" advice aims to operate the A/D converter near its full-scale value as the red LEDs light at 6 dB below full-scale, or -6 dBFS." source

"The Input Trim control in the I/O menu is before the A/D. You can use that to reduce the level into the A/D. If you want 4 dB of gain reduction: A = 10^(-4/20) = 0.63. So you need to reduce your input pad by 37%. The new value is 0.243 * 0.63 = 0.153 => 15.3%" source

Tutorial by AxeFxTutorials.

Setting the main output level

Output level knobs on panel: The main output levels on the Axe-Fx II, III and AX8 are directly controlled with the output level knobs on the panel.

(about the output levels knobs on the AX8) "The output doesn't go all the way to zero. This was done due to the plethora of support issues where people would say they weren't getting any sound and it was simply due to the fact that they had the knob turned all the way down. So now you get a little signal and we get less support calls." (source)

(about the Axe-Fx II)"We test the output to be flat within +/- 1 dB over the range of the knob. In fact I'd be surprised if there were any measurable variation at all." source

(Axe-Fx II) "The output "pot" is actually a ladder of discrete resistors that is remotely controlled by the knob on the front panel. Other products simply reduce the digital signal going into the D/A converter but this is sub-optimum as you reduce your dynamic range when doing this. The Axe-Fx II strives to keep the signal into the D/A as high as possible for optimum dynamic range and then controls the output level using a programmable output gain. The downside of this approach is that you will hear a small noise when the output switches between the resistors in the ladder." source

(Axe-Fx II) "To place a pot after D/A requires running cables to/from the front panel. These cables can degrade signal quality and pick up noise. The pots on the front panel of the II are remote controls for the digital pots. The signal never passes through them. The digital pots also allow us to boost the level from the D/A and then attenuate it precisely to improve output SNR. The Output X Boost/Pad feature would be impossible without digital pots." source

Prevent clipping the mixing console or amplifier inputs: The nominal level of the main outputs of the Axe-Fx II and AX8 is line level: +4 dBu default on the AX8 and Axe-Fx II, and -10 dBV default on the Axe-Fx III. This is adjustable on the Axe-Fx III and AX8 in I/O > Audio.

ALWAYS connect the device to a line level input on the board, when available. The output signal is too hot for a MIC input on the board.

(Axe-Fx II and AX8) "The XLR output is balanced but it's +4 dBu nominal. The problem is people connect it to a mic input which is way too sensitive for that level signal. If the board has a mic/line switch you want to set it to line level. Or if it has a pad switch turn that on. Otherwise turn the level knob way down. The thing to remember is that XLR is just a connector. It doesn't imply microphone levels. Most pro stuff like eq's, etc. have line-level XLR's."

If only MIC inputs are available on the mixing console, try this:

use a pad switch on the mixer to attenuate the signal and prevent clipping.
decrease input gain on the channel strip.
decrease the output level from the device by turning down the Output knob at the front.

Tutorial by AxeFxTutorials.

(Axe-Fx II) "Optimal gain staging would be with the level knob around noon. Higher than this and you risk clipping the inputs of the downstream device. With the level knob at full the Axe-Fx II will probably incinerate a Soundblaster or other low-cost stuff. The max level out of the Axe-Fx II is +20 dBu. Most pro gear can easily handle that but lots of gear cannot and the trend in newer gear is towards lower and lower maximum input levels (due to single-ended designs and low-voltage/low-power constraints). In the old days, +20 dBu was routine. Everything could put out and handle +20. Not so much anymore." source

(Axe-Fx II) "The II actually has more output than the I. The II can do about +20 dBu, the I was about +18." source

(Axe-Fx II) "Start with amp volume at noon. Bring up Axe-Fx volume until desired level is reached. If you need more, turn up amp. With the Axe-Fx volume all the way up you would be pushing +20 dBu into the amp which could clip the inputs to the amp." source

No DI box needed: There's NO NEED to use a DI box to connect the Axe-Fx II, III or AX8 to a mixing board directly (with amp and cab modeling).

Controlling output levels via MIDI: MIDI CCs 11 and 12 control the levels of Output 1 and Output 2. To reset them without the help of a MIDI controller, change the assignment to "none" in I/O.

Preset level and Global EQ Gain: The main output level is also affected by the output level of the selected preset, and by the Global EQ's Gain control.

Output meters on the Axe-Fx III:

A front panel LED meter bridge provides instant visual status for the inputs and outputs. The red LEDs on the front panel come on at -1 dBFS. This is different than on previous hardware.
There's a Meters page in the Home menu showing all I/O levels.

Input 2 and Output 2

Axe-Fx II and AX8: The Axe-Fx II and AX8 have a single stereo effects loop: Input 2 (Effects Return) / Output 2 (Effects Send). Read this: FX Loop block.

Axe-Fx III:

Input 2: Input 2 on the Axe-Fx III provides a set of Combi connectors (XLR and 1/4" L/R). Use the Input 2 block on the grid to handle the incoming signal. These connectors support high-impedance sources such as guitars and basses, besides other gear. Because of this, there will be some white noise when Input 2 is connected to an output, and nothing is plugged into Input 2 (this does not apply to ports 3 and 4).
The Input 2 level is adjusted through I/O > Input > Input Trim. See comments above.
Output 2: XLR L/R connectors. Use the Output 2 block on the grid to handle the signal.
Input 2 and Output 2 are LINE level ports.

Meters:

A front panel LED meter bridge provides instant visual status for the inputs and outputs. The red LEDs on the front panel come on at -1 dBFS. This is different than on previous hardware.
There's a Meters page in the Home menu showing all I/O levels.

Input/Output 3 and Input/Output 4 (Axe-Fx III)

These stereo pairs on the Axe-Fx III are designed for inserting outboard gear, for the Four-Cable-Method (4CM) and for other purposes. Use the corresponding Input and Output blocks on the grid to handle the signal.

These connectors support high-impedance sources such as guitars and basses, besides other gear.

Input 2 and Output 2 are LINE level ports. Input levels are adjusted through I/O > Input > Input Trim.

To set these loops to unity gain, set their OUT levels to maximum.

Meters:

A front panel LED meter bridge provides instant visual status for the inputs and outputs. The red LEDs on the front panel come on at -1 dBFS. This is different than on previous hardware.
There's a Meters page in the Home menu showing all I/O levels.

I/O parameters

Main Input Source / Input 1 Source

Applies to: Axe-Fx II and III.

This lets you set select the source of input 1: ANALOG (default), SPDIF/AES or USB (Axe-Fx III: USB channels 5 and 6).

Word Clock

Applies to: Axe-Fx II and III.

See Digital audio (below).

SPDIF/AES Select

Applies to: Axe-Fx II and III.

Only one can be active at any time.

Input 1 Left Select

Applies to: Axe-Fx II.

Input 1 is split: it appears at the front (mono) as well as on the rear (mono or stereo). This parameter tells the Axe-Fx II if the front input or the rear input is used to connect the instrument. Default: front.

The rear port has the same impedance as the front input (1 Mohm, for guitars) but it operates at LINE level instead of instrument level.

The front input does not disable Input 1 left at the rear. Use the menu to select either the front input or rear input. You can leave everything plugged in.

"Input 1 on the rear is a very high impedance input (1 Mohm). It is compatible with low impedance outputs. Most people don't understand the real meaning of impedance and think you need to connect low impedance to low impedance but that's only with passive devices and is a relic of the old days when transformers were used to get the best power transfer. Nowadays we have active inputs with very high impedance which are compatible with a broad range of source impedances." source

Input Mode

Applies to: Axe-Fx II, AX8, FX8.

Axe-Fx II: when connecting a guitar to the front instrument input, set this parameter to Left Only (default). Using Sum L+R can introduce noise (from the disconnected right Input 1) and attenuates the signal level (6 dB). You'd only use Sum L+R or Stereo when connecting a stereo instrument to Input 1 (rear).

Input 2 Mode controls the same for Input 2 on the Axe-Fx II, AX8 and FX8.

IN and OUT Nominal Level, Post Level

Applies to: AX8, FX8.

These parameters can be switched between +4 dB (pro line level) and -10 dB (consumer-grade equipment).

IN 2 Nominal Level: use -10 dB (default) when this input connects to the output of a guitar or stomp box. Use +4 dB when it connects to the output of a pro level device.

OUT 2 (Main) Nominal Level: use +4 dB (default) when the output connects to a mixing console or line level input on another device. Use -10 dB when the output connects to a traditional amplifier or pedal.

Post Level

Applies to: FX8.

This switches between +4 dB (pro line level, default) and -10 dB (consumer-grade equipment). Use +4 dB when the unit connects to a line level on another device. Use -10 dB when the unit connects to a guitar amplifier's instrument input or a pedal.

If you're using the FX8 in a Four Cable Method (4CM) setup, and you're experiencing hiss, try switching to another Post Level value.

NOTE: some users report that the FX8 needs a reboot after changing the value before the changed value takes effect. source

Output Mode

Applies to: Axe-Fx II, AX8, FX8.

This determines if the output signal is stereo, or one of two mono modes: Sum L+R or Copy L>R. Out 2 and Out 2 can be set to different modes.

Output Phase

Applies to: Axe-Fx II, AX8.

This lets you change the phase of the signal if necessary.

Output Boost/Pad

Applies to: Axe-Fx II, AX8.

The Boost/Pad parameter optimizes the output level and reduces noise in certain scenarios, such as a 4CM setup (Four Cable Method) or when placing the device in front of a traditional amp. Its range is 0 - 18 dB.

When putting the Axe-Fx II in front of a real amp, maximize Boost/Pad to make sure the full range of the D/A converter is used.

With 4CM: increase Boost/Pad, and prevent clipping. Turn Output 2 on the front panel fully open. Then use Input 2 Level in I/O to finetune the signal level.

Javajunkie: "Boost/pad is not intended for boosting USB output (although it does have that effect). It is for the four cable method (4CM)".

"Usually you will use this when you send the effect loop out before the amp block. The signal will be weak going to the D/A converters. This allows you to boost the signal to the D/A converters w/ maintaining the same output level." source

Output 2 Echo

Applies to: Axe-Fx II, AX8.

This determines what signal is sent through Output 2.

None: no signal.
Output 1: an exact copy of the Output 1 signal.
Input 1: an exact copy of the direct input signal (unprocessed).

Output 1 (PRE) Headroom

Applies to: FX8.

This parameter can supply more headroom if the signal through Output 1 PRE is too hot. Downside is that the noise floor is increased. Default is 6 dB, for the least headroom and lowest noise floor. The increased headroom is offset by a corresponding but opposite adjustment internally, so “what you hear” remains the same at all settings. Maximum headroom is 18 dB.

USB/Digi Out Source

Applies to: Axe-Fx II.

This lets you determine what is sent out through USB Audio/AES/SPDIF: Output 1, Output 2, or Input (which is the unprocessed DI signal).

USB Buffer Size

Applies to: Axe-Fx II and III.

See USB Audio below.

USB Return Level

Applies to: Axe-Fx II.

A level control for the incoming USB Audio signal.

Setups: Axe-Fx and AX8

See the rig diagram in Section 4 of the Owner's Manual.

FRFR - Direct - Studio Monitors - DAW

What is FRFR: full range flat response

Amplifying a signal with cabinet modeling requires a power amp and speaker system that covers the entire frequency spectrum: 20 Hz - 20 kHz. Also, it should not add any tonal coloring of its own. Those systems are called FRFR: full range flat response. Also referred to as: neutral. What goes in comes out. Tone shaping is entirely left to the input device, which is a Fractal Audio processor here.

From the Axe-Fx III Owner's Manual:

"A Full-Range Flat Response (“FRFR”) system aims to reproduce the entire audio spectrum without compromise. In comparison, most guitar speakers are narrow range, with no ability to accurately reproduce extended lows and highs. A 1×12 open-back combo is never going to sound like a 4×12 stack. In comparison, full-range flat response studio monitors, high‐quality PA speakers, and FRFR speakers designed specifically for guitar should be able to reproduce anything you play through them."

Getting accustomed to the FRFR sound may take some time. It is different from listening to a traditional guitar amp. You may also miss the "thump" of a traditional guitar cabinet.

FRFR amplification means: portability, no tone coloring, reduced stage volume, consistent tone at all volume levels and in all venues, ability to work with cab modeling, ability to produce synth and acoustic tones, and the musician hears exactly what the audience hears.

Which systems are FRFR

FRFR systems include:

Studio monitors.
Active (powered) FRFR cabs and wedges.
Passive FRFR cabs and wedges, powered by a separate amplifier.
Headphones.
DAW.
High-quality PA systems.

Popular manufacturers of FRFR solutions for stage use include Atomic, RCF, Matrix, Meyer, Friedman, XiTone, Mission Engineering, EAW, QSC and others. Quality studio monitor brands include Focal, Adam, Genelec.

Close miking

Using amp and cab modeling with FRFR amplification means that the audio signal includes a virtual cabinet. These cab models use impulse responses (IRs). These are sampled sounds of cabinets and speakers, with the recording microphone(s) placed within inches of the speaker cap or cone ("near-field"). This is referred to as a "close-miked" sound. When compared to the sound of a traditional amp and cabinet (aka "amp in the room" or "far-field" recording), close-miked sounds haven much more bass and treble, due to the microphone's proximity to the speaker.

That sound is often EQ'd at the mixing board to fit into the entire mix. Fractal Audio's Axe-Fx and AX8 modelers provide lots of EQ tools to use.

Wikipedia.

"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." source

"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." source

"Close-miked IRs typically have a lot more high frequencies than what you hear at a distance and off-axis from the speaker." source

"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." source

"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." source

"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." source

"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." source

"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." source

"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." source

"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." source

Fletcher-Munson

Another audio phenomenom to be aware of, especially when using FRFR amplification, is: Fletcher–Munson. This refers to scientific findings that human ears perceive sound at low volume levels differently than at higher levels. This is VERY important when dialing in tones.

At low volume levels people often tend to dial in more treble and bass than at loud levels. The Loudness switch on older home stereo systems does just that. But: when the volume is turned up, e.g. at a gig, those low and high frequencies become much too prominent. The guitar will then compete with cymbals (and lose), and with the bass (and lose). The result: the guitar drowns in the mix. Even turning up the volume does not fix this.

More information about Fletcher-Munson:

How to handle harsh, boomy tones and getting lost in the mix

To avoid harsh and boomy tones when using FRFR amplification, and to be heard in the mix, you need to do this:

Always dial in your "live" guitar tone at gig levels, which is 90 dB and higher. Do NOT expect excellent "bedroom" or headphones tones to translate well to a rehearsal room or stage.
Use EQ! The guitar is mainly a "mid frequency" instrument, so shave off superfluous top and bottom end and add mids. This will help focus your tone.

EQ can be applied in several ways:

Use the Low Cut and High Cut parameters in the Cab block to block undesirable top and bottom end. Common values are low-cutting (high-passing) between 80-150 Hz and high-cutting (low-passing) between 6-10 kHz. Now this may seem to make your guitar sound worse by itself, but it will improve its sound within the entire mix!
Put a PEQ at the end of the grid and block the low and high frequencies.
Use the Global EQ or a GEQ for similar results.
Adjust Depth/Bass and Treble/Presence in the Amp block.
Boost the mids. For example: put a PEQ at the end of the grid, set a band (use Peaking when using the first or last band) to 770 hz, Q at 0.35, Gain between 2 and 4 dB. source
Use the Cut switch in the Amp block.

Discussion.

"Resist the temptation to add bass and treble. The amp designers knew what they were doing (well most of them). If you are applying heavy EQ then you will be disappointed at gig volumes. What sounds midrangey and bland at low volumes will sound great at high volumes. Do some research on Fletcher-Munson to understand this." 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

Do not use a microphone on a FRFR speaker on stage

When you're using FRFR amplification on stage and you need to provide a signal for FOH, do NOT place a microphone in front of the FRFR monitor. That would make no sense because the signal already contains miked cab simulation. Direct-to-FOH is the right way to do it: run a cable from the output(s) to the mixing console. For long distances use the balanced outputs.

Create an amp-in-the-room tone with FRFR amplification

To create the amp-in-the-room sound using FRFR amplification:

Select a "farfield" stock cab type. There are the ones with "JM" in the name. Or select a stock cab captured with a neutral mic, such as the Red Wires ones, and set Proximity to its lowest value for far-field coloring.
If you have stereo amplification: use two IRS in stereo and use a very short delay in the Cab block on one of them to create the HAAS effect.
Use Room Reverb in the Cab block (not on AX8).
Increase De-Phase in the Cab block. It adds amp-in-the-room characteristics to the FR sound (not on AX8).
Add a mid-boost, as described above..
Turn up Speaker Compression (Amp block) or Motor Drive (Cab block).

Of course, if you want a real "amp in the room" tone from your modeler, use a power amp and a traditional guitar speaker cabinet.

Tweeter squeal

Some FRFR speakers can cause very high-pitched loud feedback.

"Tweeter squeal is magnetic feedback from the speaker's tweeter. Move further away from the speakers. This is a phenomenon unique to FRFR solutions." source

"Magnetic feedback is an issue unique to FRFR amplification. The tweeter creates a magnetic feedback loop with the pickups. The closer you get to the speaker the more feedback until the point it squeals. The only solution is to move away from the speaker or turn down the gain/volume." source

"The high-pitched feedback is pickup squeal and is caused by electromagnetic feedback from the speaker to your pickups. FRFR tends to exacerbate this since you have a tweeter feeding back high frequencies. A noise gate can help but the best solution is to move away from the speaker." source

Power amp and guitar speaker

See the rig diagram in Section 4 of the Owner's Manual.

Why use a power amp and a guitar speaker

When you need amplification, and FRFR (see above) is not your thing, you can amplify the processor through a power amp and a guitar speaker.

Traditional cabs move "air" and deliver a punch, which is not always achievable through FRFR. That's why many players still use a traditional cab on stage (backline), even with a direct signal going to the PA system.

Tube power amp for guitar (or head or combo)

When using the processor with a tube power amp designed for guitars (e.g. Mesa, VHT, Fryette) and a traditional cabinet:

Switch off power amp modeling in the processor.
Disable cab modeling.
Set the controls on the power amp as neutral as possible.
Turn down Speaker Drive and Speaker Comp in the Amp block, because these parameters are designed to be used with a cabinet model, not a guitar cab.

This also applies when connecting the processor to an effects return port of a guitar combo amp or head.

"If you use a tube power amp and don't turn off power amp modeling in the Axe-Fx you will get the impression that the tube power amp sounds "bigger" and "warmer". This is because the tube power amp will have more bass (and highs) than the solid-state power amp since a tube power amp's response follows the speaker impedance. People will ALWAYS find that more bass and treble sounds "better" when listening alone but in a band context that tone will get lost. Speaker designers have been exploiting this fact of human perception for decades. Many "hi-fi" speakers exaggerate the bass and treble because the uneducated customer will think they sound "better". A truly flat speaker will sound dull in comparison to one with exaggerated lows and highs. Over time, however, those exaggerated frequencies lead to fatigue. It's only in comparison that exaggerated bass and treble sound "better". In an isolated context this aspect of human perception is not evident." source

"If you are using a tube power amp you should set any Presence, Depth, Resonance, etc. controls to their minimum positions on that amp (assuming they are conventional controls). On a Mesa power amp, set them to noon. The Presence control on Mesa amps is most neutral around noon. If you turn it up it boosts the highs, if you turn it down it cuts the highs. On most other power amps it only boosts. source

Tube power amp (neutral)

When using the processor with a "neutral" tube power amp (e.g. Fryette's Power Station) and a traditional cabinet:

Power amp modeling should be turned on.
Disable cab modeling.
The tube amp interacts with the speaker automatically. It is not necessary to simulate impedance/resonance, so set Low and High Resonance in the Amp block to zero.
Turn down Speaker Drive and Speaker Comp in the Amp block, because these parameters are designed to be used with a cabinet model, not a guitar cab.

Solid-state power amp

When using the processor with a "neutral" solid-state power amp (no tubes, e.g. Matrix, Seymour Duncan) and a traditional cabinet:

Power amp modeling should be turned on.
Disable cab modeling.
Turn down Speaker Drive and Speaker Comp in the Amp block, because these parameters are designed to be used with a cabinet model, not a guitar cab.
The solid-sate amp doesn't automatically interact with the speaker like a tube amp does. That's where the Resonance parameters in the Amp block are for. The defaults may not get the best results. Optimize the bass response (Low Frequency Resonance) by finding the resonant frequency of the cabinet, like this:

Put a Filter block after the Amp block.
Set the type to Peaking, Q to 5 or so and Gain to 10 dB.
Start with a Frequency of around 50 Hz. Play some chugga-chugga and slowly adjust the Frequency until you hear and feel the cabinet resonate. You need to do this at loud volume level to notice it. Make a note of the frequency.
Remove the Filter block and set the Amp block's Low Resonance to match.

Alternatively:

Add a Synth block (after the Amp block) to the preset and make sure it is connected to the grid output.
Select Sine wave.
Turn off Tracking.
Turn up the volume of your rig.
Adjust Frequency until you hear and feel the cabinet resonate. You need to do this at loud volume level to notice it. Make a note of the frequency.
Remove the Synth block and set the Amp block's Low Resonance to match.

Gain-staging

Make sure not to overload the input of a connected power amp or active monitor.

"The II actually has more output than the I. The II can do about +20 dBu, the I was about +18." source

"Start with amp volume at noon. Bring up Axe-Fx volume until desired level is reached. If you need more, turn up amp. With the Axe-Fx volume all the way up you would be pushing +20 dBu into the amp which could clip the inputs to the amp." source

Wet/Dry/Wet setup (W/D/W)

Configure a W/D/W rig.

Speaker wire

"The Axe-Fx is designed to recreate the signal at the speaker jack of a tube amp and it does this tremendously well. If I do a Tone Match to the output of the amp vs. the model it's almost always nearly a perfectly flat line. So today I was playing around and did a quick tone match to one of my Plexis and then a Suhr Badger and the results showed a significant mid-scoop (2-3 dB). I was puzzled. Had I messed something up in the new firmware? I repeated the tone match using a DI off the speaker jack and the result was a perfectly flat line. Then I realized that the difference was due to this 30 ft speaker cable I was using because the speaker cab was remote from the amp. Just a bit surprised that that little resistance could have that much effect. Fortunately the new Cab-Lab addresses all this by allowing you to capture reference IRs and we've included reference IRs along with our latest Cab-Pack. To double-check I then captured a reference IR off the speaker and corrected the IR using the new Cab-Lab and viola, perfectly flat." source

"There's a big difference between a long cable between your guitar and amp and a long speaker cord. A long instrument cord loads your guitar's pickups with a reactive load that's mostly capacitive. This changes the resonant frequency of the pickups and rolls off the highs. A long speaker cord increases the resistance between the amp and the speaker which decreases the damping factor. A lower damping factor means the response follows the impedance curve of the speaker more than a high damping factor." source

Combining FRFR and power amp + guitar cab

The Axe-Fx series and AX8 let you combine amplification methods, such as sending a signal WITH cab modeling (FRFR) to one output (like a PA system), and a signal WITHOUT cab modeling to another output to feed a power amp and guitar speaker on stage.

Axe-Fx II and AX8:

Echo Out2 = Out1: Enabling this setting in I/O will duplicate the Output 1 signal to Output 2. The Global EQ on Output 2 lets you tailor the tone, independent of Output 1. The Output 1 knobcontrols the level of Output 1, and the Output 2 knob does the same with Output 2. This method DOES NOT WORK with presets that contain a FXL block. This method is great when you want to control the level of your personal monitoring (Out2) separately from the signal that's being sent to FOH (Out1). Tutorial by AxeFxTutorials.
FXL block: Insert an FXL block and make it part of the routing but don't connect it to the grid output. The signal before the FXL block will be sent to Output 2. This method is more flexible than the one above, because the position of the FXL block determines which part of the signal is being sent to Output 2. For example, placing FXL before or after a Cabinet block determines whether the Output 2 signal includes cabinet modeling or not. Use this when you want your FOH signal to be "direct" (including cab modeling) and your stage sound to come from a traditional cabinet (without cab modeling). Among the factory presets is a template. The Axe-Fx II lets you put FXL in series or parallel, but the AX8 requires FXL in a parallel row to prevent a feedback loop. Here's a tutorial.
Left/Right: Split the signal at the end of the grid into a row with a Cab block and a row with a shunt. In the Output Mixer pan those rows 100% left (Cab) and right (shunt). Now OUT1 Left is the signal with cabinet modeling, and OUT1 Right is the signal without cabinet modeling. This method allows you to use the stereo effects loop for other purposes. source

Axe-Fx III:

Echo Out2 = Out1: see above.
Separate outputs: you can use multiple Output blocks and signal chains to handle multiple outgoing signals.
See the rig diagrams in Section 4 of the Owner's Manual.

Four Cable Method (4CM)

Axe-Fx III: See the rig diagram in Section 4 of the Owner's Manual. Use I/O 3 or I/O 4, set to unity gain, adjust Boost/Pad in the I/O menu for optimal SNR, and adjust levels where needed.

Axe-Fx II: Adjust Boost/Pad and Input Level in the I/O menu to optimize the signal. Also, turn the Output Level knob fully open for unity gain. You can't combine 4CM with cab modeling.

"The very early Axe-Fx II's had more bandwidth than necessary on Output 2. The frequency response extended to hundreds of kHz. When used with certain tube amps this would cause instability in the output drivers. The solution was to limit the bandwidth to a "normal" range of 20 to 20 kHz. We provided the update for free and all units shipped after the first 100 or so had this update included. The Axe-Fx II Mark II, XL and XL+ have a redesigned output circuit that is immune from any of these issues." source

"It is very difficult to minimize the hiss when putting a digital processor in front of a high-gain amp due to the A/D and D/A conversions. The XL is probably one of the quietest processors made but there will still be some residual hiss when using high gain. The Output 2 Boost/Pad feature was specifically intended to minimize hiss in these scenarios by running the D/A converter as "hot" as possible and then reducing the signal level after the converter with an analog pad." source

"The XL+ shares the same amazing low-noise architecture of the FX8. I regularly use my XL+ in 4CM as this is part of the modeling process. It's the quietest device I've ever tried in 4CM." source

AX8: Unlike the Axe-Fx and FX8, the AX8 is not optimized for 4CM, but it will work. The process is the same as with the Axe-Fx II.

Demonstration.

Effects only

POST-effects

When using the Axe-Fx III, II or AX8 as an effects-only device in an amp's effects loop, you probably want it to send and receive line level signals, at unity gain.

Use Input 2 and Output 2 on the Axe-Fx II and AX8, and Input/Output 3 or 4 on the Axe-Fx III.
Adjust Input Level for an optimal signal-to-noise ratio.
Select the correct input and output settings in I/O.
Set the Output 2 knob to it maximum for unity gain.

Test the setup by creating a preset with only shunts. The level should be the same as when leaving out the processor. Then start adding effect blocks (no Amp or Cab).

Axe-Fx III: See the rig diagram in Section 4 of the Owner's Manual.

"You should NOT use Boost/Pad in this configuration." source

Some amps require inserting a dummy jack into the effects loop's Send to activate the effects loop.

PRE-effects

Use Output 2 on the Axe-Fx II and AX8, and Input/Output 3 or 4 on the Axe-Fx III.
In I/O set output level to -10 dB if possible.
Crank Boost/Pad to make sure the full range of the D/A converter is used and turn up the Output knob all the way for unity gain.

Test the setup by creating a preset with only shunts. The level should be the same as when leaving out the processor. Then start adding effect blocks (no Amp or Cab).

Axe-Fx III: See the rig diagram in Section 4 of the Owner's Manual.

Digital I/O

Digital I/O: supported by which Fractal Audio products

Axe-Fx III: USB, SPDIF, AES.
Axe-Fx II: USB, SPDIF, AES.
AX8: SPDIF output only.
FX8: no.

Why use digital I/O

A digital connection skips the analog/digital conversion of the source signal. The analog outputs of the Axe-Fx and AX8 however deliver high-quality results too, similar to digital audio.

Sample rate fixed at 48kHz

The sample rate of the Axe-Fx series, FX8 and AX8 is fixed at 48kHz (24-bit).

Digitally connected devices and DAW software always need to be set to the same sample rate. Example.

If required, resampling can be handled by software.

"The Axe-FX uses higher sampling rates (oversampling) during the processing stages. This is how it avoids aliasing when non-linearities are applied. But the sampling rate of the audio that is sent to the DAC is the same as the sampling rate coming out of the SPDIF output: 48khz. In other words, it goes from 48khz (ADC) -> higher sampling rate -> 48khz (DAC). So just because these higher sampling rates are used for the processing stages doesn't mean it would be trivial to send a higher rate to the SPDIF output. The 48khz signal would need to be sample rate converted (SRC) at the output stage by a hardware SRC chip and Cliff's whole point is that software SRC's provide better quality than what is available with hardware SRC's."

"IMHO, the ideal sample rate is 64 kHz but that's not a standard. The nice thing about 64 kHz is that you can have a gentle transition band from 20 kHz to Nyquist which results in shorter filters, lower latency, less phase shift, etc. I was very tempted to make the Axe-Fx II run at 64 kHz but people probably would have freaked out." source

"I've long maintained that 64 kHz is the ideal sample rate for audio. But I can't get the industry to change." source

"48 kHz is considered "pro" sampling rate. The reason for 44.1 kHz on CD's is subject to debate. Some maintain that the sample rate was lowered so that Beethoven's 9th would fit on a single CD. Others claim that it was because that rate was compatible with video equipment. IMO 44.1 kHz is insufficient for professional audio. Personally I would prefer 64 kHz. Whilst Nyquist theorem is all well and good most people don't understand the details and simply state "the sample rate must be twice the highest desired frequency". The problem with this is as you approach Nyquist the filter demands become extreme. The more extreme the filter demands the more taps are needed, the more precision is needed, the more latency is incurred, etc. A 64 kHz sample rate would give you a nice, smooth roll-off from 20 kHz to 32 kHz rather than the brick wall you get with 44.1 kHz. There is no hardware advantage to using 48 vs. 44.1. The costs would be the same in either case. Modern converters use over-sampling techniques to implement the necessary anti-aliasing filters thereby reducing off-chip filtering to simple circuits. MP3s have no native sample rate but are typically 44.1 kHz because they are usually derived from CDs. MP3 is a psycho-acoustic compression format that exploits frequency masking to lower the data required to store audio information." source

"If the Axe-Fx were running at 44.1 all the cab IRs would need to be resampled, or there would need to be an SRC chip on the digital I/O. There is no free lunch. The problem isn't the Axe-Fx, the problem is studios stubbornly sticking to 44.1 when 48 is a much better rate." source

Axe-Fx II and USB Audio

Windows: The Axe-Fx II is an Audio Class 2.0 compliant device. A class-compliant device requires no drivers. The drivers are provided by the OS manufacturer. Audio Class 2.0 also encompasses MIDI-over-USB. Microsoft does not support Audio Class 2.0. Therefore FAS provides a driver for Windows systems. The driver for Windows contains both the firmware installer and the audio drivers.
Apple: Apple does support Audio Class 2.0, but poorly. To overcome this, you can increase the buffer size in the Axe-Fx II's I/O > Audio menu. The driver for Macs is NOT an audio driver. It is a firmware installer. The Axe-Fx II uses a "soft" USB controller. It gets its code from the host computer. When you turn the Axe-Fx II on it requests firmware from the host. This is superior to a hard-coded controller in that updates merely require a new host image rather than reflashing the controller.
Linux: read this.

See the parameter I/O > Audio > Buffer size.

Lower USB Buffer Size in I/O > Audio for less latency, increase when experiencing distorted audio. You should stop USB audio streaming when changing this value so as to allow the buffer to reset properly. Streaming can be stopped by closing the application sending data to the Axe-Fx or by disconnecting the USB cable.

The meters in the Utility menu display the USB performance. Ideally the bar should be at around 50%. If the bar sinks all the way to the bottom or goes all the way to the top, then the buffer may under/overflow and the USB buffer size should be increased. The number of buffer errors that have occurred since the last buffer reset is indicated above the bar graph.

USB Level in I/O > Audio sets the level of the USB input signal sent to the main outputs. If you don't hear anything when monitoring the Axe-Fx through a computer, check this parameter. Also verify the USB/DIGI OUT setting.

USB Audio rate is fixed at 48 kHz, 24-bits.

"The Axe-Fx II USB is 24 bits. This is 144.7 dB of dynamic range. Full-scale is about +20 dBu. So even if your guitar is -20 dBu (-40 dB re. FS) you still have over 100 dB of dynamic range. A typical single coil pickup can easily exceed -20 dBu. A humbucker can easily exceed 0 dBu. Full-scale of 20 dBu gives you a few bits of headroom in case of very hot pickups. The self noise of a guitar pickup and associated electronics limits its dynamic range to less than 100 dB typically." And: "The digital bit depth on the USB and Digital I/O exceeds both the dynamic range of the Axe-Fx itself and certainly that of any guitar. Furthermore the bit depth is sufficient to fully capture the dynamic range of a guitar while still maintaining +20 dBu as full-scale." source

"The hardware is incapable of doing 4x4. The only choices are 3x3 or 4x2 and Logic doesn't work with 3x3. We also had some issues with 3x3 in Windows 7 IIRC." source

USB features.png

Axe-Fx III and USB Audio

The Axe-Fx III has a dedicated 16-core, 500 MHz USB microcontroller, providing 16 (8x8) channels of USB audio (8 in, 8 out) through USB 2.0.

USB Audio rate is fixed at 48 kHz, 24-bit.

USB In (from Axe-Fx III to Computer):
1+2: Output 1 (regular stereo output).
3+4: Output 2 (regular stereo output).
5+6: Input 1 (copy of signal at front/rear Instrument input, for reamping, mono).
7+8: Input 2 (copy of signal at Input 2, stereo).

USB Out (from computer to Axe-Fx III):
1+2: Routed to physical Output 1 L+R (audio from computer, added to OUT1).
3+4: Routed to physical Output 2 L+R (audio from computer, i/e/ backing tracks that can be processed seperately).
5+6: Routed to the Grid via INPUT 1 block when its source is set to USB (for reamping).
7+8: Routed to the Grid via the dedicated INPUT USB block (for additional computer audio).

A USB audio sound source can be placed anywhere on the grid with its own dedicated block.

Latency when using software monitoring is low.

USB performance can be monitored on the Meters page of the Home menu, or in the Utilities menu.

Incoming sound from USB Audio is mixed with the signal that comes out at Output 1. To change this, adjust the sound settings on the computer.

Mac: see section 3 of the Owner's manual.
Windows: use ASIO.

"The effective throughput of USB 2.0 is roughly 280 Mb/s. One channel of audio is 48000 samples/s * 24 bits/sample = 1.152 Mb/s. Theoretically you could transfer over 200 channels of audio on USB 2.0." source

"The Axe-Fx III has a new driver. I just tested it under Reaper and was able to set the buffer size to 8 with no problems. I typically have it at 256 because I monitor directly from the output but it seems to be working fine on the lowest setting." source

SPDIF

AES and SPDIF can not be used simultaneously.

Axe-Fx III: to be added.

Axe-Fx II: configure your Axe-Fx II for S/PDIF.

AX8: SPDIF output only. The strength of the SPDIF signal level depends on the position of the front panel output knob (unlike the Axe-Fx II).

"The SPDIF is a digital representation of OUTPUT 1." source

This discussion lists AX8-compatible SPDIF interfaces.

AES

AES and SPDIF can not be used simultaneously.

Latency when monitoring

When monitoring audio through the computer's output, latency (the time between playing a note and hearing it) depends on the computer and the USB driver. Higher buffer sizes = higher latency. The Axe-Fx II and III allow you to adjust the USB buffer size.

"The Axe-Fx III has a new driver. I just tested it under Reaper and was able to set the buffer size to 8 with no problems. I typically have it at 256 because I monitor directly from the output but it seems to be working fine on the lowest setting." source

When monitoring audio directly from the Fractal Audio hardware or hardware audio interface, latency is below 2 ms.

Master or slave

"The Axe-Fx II can be a slave. Set the Input Source to AES. It will derive its internal clock from the input stream. The input stream must be 48K. Note that SOMETHING must be the master in this case." source source

Word Clock

The clock source for the A/D and D/A converters is either AUTO/INTERNAL or SPDIF/AES.

Axe-Fx III: Word Clock is recovered from the SPDIF/AES input signal.

"Yes via SPDIF/AES in (which actually works better as a word clock than a word clock input)." source

AX8 and FX8: not supported.

Axe-Fx II:

Auto: uses the internal clock if the input source is Analog or USB, uses the recovered SPDIF/AES clock if the input is SPDIF/AES.
SPDIF/AES IN: uses the recovered clock for all input sources. A valid 48 kHz data stream must be present at the AES or SPDIF input. If a valid stream is not detected, the unit will fall back to the internal clock and display "NO INPUT CLOCK!". The SPDIF/AES select must be set to the appropriate value, i.e. if the data stream is input to the XLR jack then SPDIF/AES SELECT must be set to AES.

"Set Word Clock to SPDIF/AES In. Connect a cable from the ULN-8 to AES In or SPDIF In. Set SPDIF/AES Select to appropriate input used." source

"The Axe-Fx II will derive its clock from the AES/SPDIF when using Digital In. In Analog In it uses its internal clock." source

Simultaneous analog and digital audio

Axe-Fx II: you can use digital input together with analog output and simultaneous digital out. Also, you can use Input/Output 2 while using the digital input. source

How to mute hardware when listening to DAW

When the Axe-Fx II is connected to a DAW through USB, and you have monitors connected to the Axe-Fx and you're recording, you may want to monitor just the DAW signal, not the signal from the Axe-Fx.

To accomplish this, use one of these methods:

Set USB/DIGI Out Source to Output 2, set Output 2 Echo to Output 1 and lower gain in Output 1's Global EQ gain slider.
Set USB/DIGI Out Source to Output 2, set Output 2 Echo to Output 1 and send a 0 value for Output 1 Volume's MIDI CC.
Set USB/DIGI Out Source to Output 2, put an FX Loop block at the end of the grid chain and connect your signal row(s) to it instead of the output block.

source source

The Axe-Fx III's expanded I/O capabilities allow easier control.

Use the processor as an A/D converter

To use the device as an analog-to-digital converter: (source)

Create a preset with nothing but shunts from input to output.
Or set Input 1 Left Select to to Rear and plug the device into Input 1 Left on the back (Axe-Fx II).
Or connect In and Out and engage Bypass Mode (Axe-Fx II).

Connecting the Axe-Fx to a DAW drops the volume or puts it into bypass

Some DAWs send MIDI commands (intended for other devices) which lower the volume of the Axe-Fx or put it into Bypass mode. To solve this, adjust the DAW settings. If not possible, force the Axe-Fx to ignore those commands by setting MIDI CCs 10 and 13 in I/O > Control to "none".

Source: Bakerman.

More information about digital I/O

Re-amping

Read this: Re-amping.

Headphones

Axe-Fx II and III: Use the Output 1 knob to set the headphones level. If you have both monitors and headphones connected and want to listen through headphones only, use one of these methods:

Switch off the monitors.
Feed the studio monitors through another output.
Use a line level attenuator to turn down the monitors' level without affecting the headphones level.

AX8: The AX8 does not have a dedicated headphones output. You can use a Y-cable with the outputs. If you connect your headphones directly to the AX8, you may get less signal level than from a dedicated headphones output on other devices. A popular solution is to use a small headphones amp, e.g. from M-Audio or Rolls.

"It does not have a headphone output but the outputs should be able to drive phones with ease. You'd just need a Y-cable adapter." source

"The output impedance of the 1/4" outputs is 600 ohms IIRC. This may be too high for some headphones. We always use a small output impedance on our designs to help protect the output devices against improper connections, ESD, etc. The outputs were not really designed to drive headphones, they are designed to drive high-impedance inputs (>10K). Headphones will work but it won't be optimum. For optimum results use a dedicated headphone amp." source

Impedance: If the volume level through your headphones is very low, switch to lower impedance headphones (such as 32 Ohm), use a headphone amp or use headphones with a built-in amplifier (such as Blue Mo-Fi). The Axe-Fx III is better at feeding sufficient signal level to hard-to-drive headphones than the Axe-Fx II. Article about headphones and impedance.

Sound through headphones can be dull.

"Because there's no string and body reinforcement. When you play through speakers the sound couples into the guitar body and strings. With headphones you don't get this so the sound is very sterile and lifeless. Now, if you use speakers during recording and then playback through headphones it will sound fine."

"It's lack of acoustic reinforcement. I did a test a few years ago and I don't remember the actual numbers but having a speaker aimed at the guitar adds many dBs of power to the lower mids coming out of the guitar. IOW, if you measure the spectrum of the signal coming out of a guitar alone and then compare that to the signal coming out with a cab or monitor in proximity at a reasonable volume there are a LOT more lower mids with the speaker present. This results in a "thin" sound without the speaker." source

"The problem with headphones is that there is no acoustic reinforcement of the guitar. There is zero coupling between the speakers (inside the headphones) and the guitar. Without that coupling, which is a type of positive feedback, the sound is lifeless, thin and harsh. When your heroes recorded their guitar parts that weren't using headphones. On "Appetite for Destruction" Slash recorded his guitar parts in the control room. To get even more coupling into the guitar a combo amp was in the control room with him pointed at the guitar. A volume pedal allowed him to adjust the volume of the combo amp so he could control the coupling. With the volume pedal all the way up he could get controlled feedback. I've actually done tests comparing the spectrum out of the guitar when there is no coupling (i.e. monitors turned off) and with typical coupling (monitors loud or using a conventional cab). The boost in the low midrange is significant. I forget the actual numbers but it was at least several dB." source

"I did some studies years ago and having a speaker in proximity to the guitar actually changes the final tone considerably. I compared the frequency response with the amp in isolation to the frequency response with the amp in proximity and measured several dB difference in the lows and mids. It was clearly audible when the recordings were played back." source

Tips for improving sound quality through headphones:

Use far-field Impulse Responses.
Increase Proximity in the Cabinet block.
Increase Room Level in the Cabinet block.
Always use a stereo input signal.
Add the Stereo Enhancer to presets, or add Micro Delay in the Cab block.
Increase De-Phase in the Cab block.
Use Speaker Compression in the Amp block.

More information about headphones:

In-Ear-Monitoring (IEM)

In-Ear-Monitoring (IEM) provides a way for musicians to monitor sound through earbuds, instead of floor wedges etc. While this provides a superior listening environment, it takes getting used to the direct sound into your ears. Always use the built-in limiter of your IEM system to protect your ears against sudden spikes and peaks!

The same tips as for headphones apply. The expanded I/O of the Axe-Fx III makes it easier to send/receive an IEM-specific signal.

Setups: FX8

Pedalboard (PRE effects)

Guitar goes into IN [PRE] / INSTR. Note: this input only feeds effect blocks designated as PRE.
OUT [PRE] LEFT goes into the amplifier's guitar input. Use a Humbuster cable to prevent noise.
You can use default FX8 settings. Exception: change the output mode in I/O > Audio (see manual for stereo operation)to Mono (see manual for stereo operation).

You can also use this setup to connect an FX8 to the Axe-Fx.

More information in the Owner's Manual, including a description of the cables required.

In amplifier's effects loop (post-effects)

Guitar goes straight into the amplifier.
Amp's effects loop SEND goes into IN [POST] LEFT.
Amp's effects loop RETURN goes into OUT [POST] LEFT. Use a Humbuster cable to prevent noise.
You can use default FX8 settings. Exceptions:
1. Change the output mode in I/O > Audio to Mono (see manual for stereo operation).
2. Change Global Looper Location to OUT POST.
3. Change Global Detector to IN [POST].

"Are the outputs buffered for long cable runs? Yes." source

More information in the Owner's Manual, including a description of the cables required.

Four Cable Method (4CM)

The FX8 can be set up to put effects before the amp as well as in the amp's effects loop.

Guitar goes into IN [PRE] / INSTR.
OUT [PRE] LEFT goes into the amp's guitar's input. Use a Humbuster cable to prevent noise.
The amp's effects loop SEND goes into IN [POST] LEFT.
The amp's effects loop RETURN goes into OUT [POST] LEFT. Use a Humbuster cable to prevent noise.
You can use default FX8 settings. Exceptions:
1. Change the output mode in I/O > Audio to Mono (see manual for stereo operation).
2. Change Global Looper Location to OUT POST.

"Are the outputs buffered for long cable runs? Yes." source

If you have the FX8 set up for 4CM and want to change this, for example to put the FX8 before a computer, just use a jumper cable to connect OUT PRE L MONO to IN POST L, with OUT POST L going to the computer, amp or whatever. All effects will work and there's no need to change stuff in the configuration.

If you're using the FX8 in a 4CM setup and you're experiencing hiss, try another Post Level value.

More information in the Owner's Manual, including a description of the cables required.

Combine with Axe-Fx II, Axe-Fx III or AX8

You can use the FX8 for "pre" effects (plug guitar into FX8) and the Axe-Fx or AX8 for post-effects, including amp and cabinet modeling (plug FX8 into Axe-Fx or AX8). By adding a MIDI connection you can change Axe-Fx and AX8 presets from the FX8.

Relays: switch amp channels and more

CAUTION: Do NOT connect anything to the relays jacks until you've read the warnings in the manual!

What are relays? Relays are electrically operated switches/connectors, which can be used to switch channels on an amplifier and switch other stuff.

How many relays does the FX8 have? The FX8 has two relays.

How can I control these relays? These are controlled through:

Scenes: you can use scenes to switch amp channels through relays. This is configured on the preset's Config page.
Footswitches: you can assign footswitches to the relays per preset, for manual control. Assign the footswitch and configure it on the Footswitch page.

IMPORTANT: a Relay block in the preset will disable the scene's Relay settings.

Do the relays support X/Y switching? The relays support X/Y switching.

What are the possible settings? The relay states are:

Off: nothing connected.
Tip: tip to Sleeve.
Ring: ring to sleeve.
Both: tip AND ring to sleeve.

What are the switch modes of the relays? The switch modes of the relays are:

Latching: the selected RELAY ON state remains connected and the switch LED remains ON as long as the switch is engaged. Nothing is connected when the switch is OFF.
Auto-Off: the selected RELAY ON state remains connected only for a moment when you press the footswitch. The relay then automatically turns OFF, as does the LED.

Which cables can be used? Depending on the amp, you can use TS or TRS cables.

"The FX8 will short tip-to-sleeve, ring-to-sleeve, or both. The circuit is designed to handle 200mA of current. If the current generated by that voltage drop is 200mA or less, then the FX8 will not have a problem." source

"The relays of the FX8 are designed for use ONLY with amplifiers that use “short-to-sleeve” type switching. Do NOT connect the FX8 relays to the switch jacks of an amp that uses voltage differential switching or any other type of switching aside from short-to-sleeve, or serious damage can occur to both units. If you are not 100% sure, contact your amp manufacturer to determine whether your amp is compatible with short-to-sleeve switching. The FX8 relay jacks are compatible with TRS cables, TS cables, or TRS-to-dual-TS split cables. The relays are also fully isolated from the electrical ground of the FX8."

"The FX8 features two TRS (Tip-Ring-Sleeve) relays that can be used to switch the channel or other functions of a connected amplifier or device. If the warning above seems stern, that’s because the last thing we want is for anyone to damage their amp or FX8. In fact, short-to-sleeve relay switched amps are quite common, and your amp may well be perfectly compatible. We need to trust and require you however, to understand how your amp works and make the right choices about connecting it to the FX8 relay jacks. Your amp manufacturer should be able to help if you read them the warning above."

"The FX8 relay outputs employ a "Short-to-Sleeve" connection. Each relay output can short Tip-to-Sleeve, Ring-to-Sleeve, and Both. If the pedal connection uses a voltage drop to power an LED, the relay circuit on the FX8 is rated for a maximum of 200mA." source

@@ Line 307: / Line 307: @@
 '''What is Input Level or Input Pad for? ''' <BR>
-Input Level and Pad ARE NOT GAIN CONTROLS. They do NOT affect the overall volume level, or clipping or amp gain (unless you go below 5%). It just optimizes the signal-to-noise ratio of the analog-to-digital converters. The adjustment is applied before the A/D converter and is offset by a corresponding but opposite boost at the output of the converter.
+Input Level and Input Pad are NOT GAIN controls. They do NOT affect the overall volume level, or clipping or amp gain (unless you go below 5%). It just optimizes the signal-to-noise ratio of the analog-to-digital converters. The adjustment is applied before the A/D converter and is offset by a corresponding but opposite boost at the output of the converter.
 '''How to set Input Level or Input Pad'''<BR>
@@ Line 318: / Line 318: @@
 Device-specific comments:
 * '''AX8 and FX8''': higher Input Pad value means a lower input (padding). Run it as low as possible, because padding increases noise floor.
-* '''Axe-Fx III''': the input on the Axe-Fx III is a little more sensitive than on the Axe-Fx II, but it has more headroom / dynamic range. Do not set it below 5%, because at this point gain may be affected. Also be aware that the red LEDS on the front panel come on at -1 dBFS. The front panel LED meter bridge provides instant visual status for the inputs.
+* '''Axe-Fx III''': the instrument input on the Axe-Fx III is a little more sensitive than on the Axe-Fx II, but it has more headroom / dynamic range. Do not set it below 5%, because at this point gain may be affected. Also be aware that the red LEDS on the front panel come on at -1 dBFS. The front panel LED meter bridge provides instant visual status for the inputs.
 Note: when using a mono instrument, do not use an input that's set to Stereo or Sum L+R, use Left Only (default)
@@ Line 409: / Line 409: @@
 =I/O parameters=
-Axe-Fx III: ''to be added''.
+==Main Input Source / Input 1 Source==
+Applies to: Axe-Fx II and III.
-==Main Input Source==
+This lets you set select the source of input 1: ANALOG (default), SPDIF/AES or USB (Axe-Fx III: USB channels 5 and 6).
-Applies to: Axe-Fx II.
-This lets you set select the input source: Analog (this is Input 1, default), SPDIF/AES or USB. This is useful for reamping and other applications.
 ==Word Clock==
-Applies to: Axe-Fx II.
+Applies to: Axe-Fx II and III.
-Read this: [[Connecting hardware and setting levels|Digital audio]].
+See Digital audio (below).
 ==SPDIF/AES Select==
-Applies to: Axe-Fx II.
+Applies to: Axe-Fx II and III.
+Only one can be active at any time.
 ==Input 1 Left Select==
@@ Line 505: / Line 505: @@
 ==USB Buffer Size==
-Applies to: Axe-Fx II.
+Applies to: Axe-Fx II and III.
-Defaults at 1024. The buffer can be monitored in Utility > Status.
+See USB Audio below.
 ==USB Return Level==
@@ Line 789: / Line 789: @@
 * '''Linux''': [http://wiki.fractalaudio.com/axefx2/index.php?title=Axe-Fx_II_and_USB#Linux read this].
-Use the USB buffer parameter in I/O > Audio to lower values for less latency, set to higher values if experiencing distorted audio. Low values generally work fine with Windows machines. OS-X computers usually need higher values due to poor clock adaptation. You should stop USB audio streaming when changing this value so as to allow the buffer to reset properly. Streaming can be stopped by closing the application sending data to the Axe-Fx or by disconnecting the USB cable.
+See the parameter I/O > Audio > Buffer size.
+Lower USB Buffer Size in I/O > Audio for less latency, increase when experiencing distorted audio. You should stop USB audio streaming when changing this value so as to allow the buffer to reset properly. Streaming can be stopped by closing the application sending data to the Axe-Fx or by disconnecting the USB cable.
-The “USB” bar graph in Utility > Status displays the amount of data in the USB FIFO buffer. Ideally the bar should be at around 50%. If the bar sinks all the way to the bottom or goes all the way to the top, then the buffer may under/overflow and the USB buffer size should be increased. The number of buffer errors that have occurred since the last buffer reset is indicated above the bar graph.
+The meters in the Utility menu display the USB performance. Ideally the bar should be at around 50%. If the bar sinks all the way to the bottom or goes all the way to the top, then the buffer may under/overflow and the USB buffer size should be increased. The number of buffer errors that have occurred since the last buffer reset is indicated above the bar graph.
-USB Level in I/O sets the level of the USB input signal sent to the main outputs. If you don't hear anything when monitoring the Axe-Fx through a computer, check this parameter. Also verify the USB/DIGI OUT setting.
+USB Level in I/O > Audio sets the level of the USB input signal sent to the main outputs. If you don't hear anything when monitoring the Axe-Fx through a computer, check this parameter. Also verify the USB/DIGI OUT setting.
 USB Audio rate is fixed at 48 kHz, 24-bits.
@@ Line 863: / Line 865: @@
 ===Word Clock===
-'''Axe-Fx III''': Word Clock works via the SPDIF/AES input.
+The clock source for the A/D and D/A converters is either AUTO/INTERNAL or SPDIF/AES.
+'''Axe-Fx III''': Word Clock is recovered from the SPDIF/AES input signal.
 <blockquote>"Yes via SPDIF/AES in (which actually works better as a word clock than a word clock input)." [http://forum.fractalaudio.com/threads/axefx-iii-whats-missing.134747/page-4#post-1594173 source] </blockquote>
@@ Line 869: / Line 873: @@
 '''AX8 and FX8''': not supported.
-'''Axe-Fx II''': the Word Clock parameter sets the clock source for the A/D and D/A converters:
+'''Axe-Fx II''':
 * '''Auto''': uses the internal clock if the input source is Analog or USB, uses the recovered SPDIF/AES clock if the input is SPDIF/AES.
 * '''SPDIF/AES IN''': uses the recovered clock for all input sources. A valid 48 kHz data stream must be present at the AES or SPDIF input. If a valid stream is not detected, the unit will fall back to the internal clock and display "NO INPUT CLOCK!". The SPDIF/AES select must be set to the appropriate value, i.e. if the data stream is input to the XLR jack then SPDIF/AES SELECT must be set to AES.

Difference between revisions of "I/O connectivity and levels"

Revision as of 14:51, 20 March 2018

Contents

List of I/O ports

Axe-Fx III

Axe-Fx II

AX8

FX8

Things you must know

Balanced versus unbalanced

Instrument level versus line level

Humbuster

Unity gain

Decibels

Connecting instruments and other devices

Instrument input

Switching between different guitars

Multiple instruments simultaneously

Acoustic instruments

Bass guitar

Pedals

Wireless

Play sound from computer or other device

Microphones

Sustain and feedback

Setting the main input level

Setting the main output level

Input 2 and Output 2

Input/Output 3 and Input/Output 4 (Axe-Fx III)

I/O parameters

Main Input Source / Input 1 Source

Word Clock

SPDIF/AES Select

Input 1 Left Select

Input Mode

IN and OUT Nominal Level, Post Level

Post Level

Output Mode

Output Phase

Output Boost/Pad

Output 2 Echo

Output 1 (PRE) Headroom

USB/Digi Out Source

USB Buffer Size

USB Return Level

Setups: Axe-Fx and AX8

FRFR - Direct - Studio Monitors - DAW

What is FRFR: full range flat response

Which systems are FRFR

Close miking

Fletcher-Munson

How to handle harsh, boomy tones and getting lost in the mix

Do not use a microphone on a FRFR speaker on stage

Create an amp-in-the-room tone with FRFR amplification

Tweeter squeal

Power amp and guitar speaker

Why use a power amp and a guitar speaker

Tube power amp for guitar (or head or combo)

Tube power amp (neutral)

Solid-state power amp

Gain-staging

Wet/Dry/Wet setup (W/D/W)

Speaker wire

Combining FRFR and power amp + guitar cab

Four Cable Method (4CM)

Effects only

POST-effects

PRE-effects

Digital I/O

Digital I/O: supported by which Fractal Audio products

Why use digital I/O

Sample rate fixed at 48kHz

Axe-Fx II and USB Audio

Axe-Fx III and USB Audio

SPDIF

AES

Latency when monitoring

Master or slave

Word Clock

Simultaneous analog and digital audio