Always consult the official Owners Manuals first

Cab block

From Fractal Audio Wiki
Revision as of 10:23, 23 March 2015 by Yek (talk | contribs)
Jump to navigation Jump to search

File:Amps2.png

Cab: Amp & Cab Quick Reference Guide by jma

  • Forum member jma provides an handy reference guide, covering Amp and Cab block parameters and descriptions, list of CCs, Drive block descriptions etc.

Cab: Impulse Responses (IRs)

  • This page is about the Cab block and its parameters. The separate page Impulse Responses (IRs) provides information about IRs / user cabs and IR Capture.

Cab: factory cabs (stock cabs)

  • Selected Red Wirez, OwnHammer and other IRs are included in the Axe-Fx II firmware as stock cabs. Also included are some farfield IRs by Jay Mitchell. Additional IRs are included as attachments with the firmware.
  • All stock cabs are time-aligned, which means that you can mix these inside the Axe-Fx II using stereo cabs.
  • The Axe-Fx II XL has 159 factory cabs, more than the previous Axe-Fx models.
  • Cliff: "The factory IRs were hand-selected by me after auditioning thousands of OH and RW and other IRs. Some of the IRs are custom mixes of mine. My rule-of-thumb was to select as neutral sounding IRs as possible. However, what I like may be much different than what others like. Some people complain the Axe-Fx sounds too bright. Others say it's not bright enough. It's a no-win situation. This is why I've been harping on capturing IRs. It's personal preference. Producers probably spend more time perfecting mic placement than anything else when getting guitar tones to tape. An IR is the same thing, it's capturing the mic and placement." source

Cab: matching amps and cabs

  • It’s a matter of personal preference which cab model you like to pair with an amp model. You can choose for traditional combinations. Or be creative and innovative. The differences can be huge.
  • When comparing cabs, don't judge too quickly. Each time you select a cab, you may need to adjust the amp settings to dial in a tone.
  • Common combinations of amps and cabs are listed here: Amp: all models.

Cab: position of effects and Cab block

  • In the "real" analog world it makes a difference if you put effects before or after the speaker cabinet. It's different with the Axe-Fx II. 
  • Javajunkie: "You can place the effects loop anywhere in the chain (just add the fx loop block). Unless you are running a stereo cab or 2 mono cabs panned hard L/R, you may want to place stereo effects after the cab. The cab is a linear time invariant effect (unless you add drive) so effects like delay and reverb will sound the same before or after it. As Cliff and others have stated on numerous occasions LTI effects can be placed before and after each other and they will sound the same. Only when placed before or after non-LTI effects (drive, amps, et. al) it really matters. The one caveat there is that some effects are mono, placing effects before and after that makes a difference."
  • Cabinet blocks in parallel rows sound louder than a single Cabinet block. Here's the explanation. Bakerman: "It depends on how you're panning. Assuming a mono signal sent to cabs: Stereo cab w/ Pan L and Pan R fully left & right will be the same output level as 2 mono cabs w/ balance L & R. If pans/balances are centered the 2 mono cabs will be 6 dB louder. Balance elsewhere would be between 0 and 6 dB louder, and balance doesn't correspond 1:1 to pan L/R for the same placement. Balances will need to be further toward -50 or 50." source
  • Cliff's comments:
    • "The difference in having the cabinet before or after the effects is usually subtle. It depends on how non-linear or time-variant the effect is. For effects like EQ, which are linear and time-invariant, it doesn't matter at all. For slightly time-variant effects like chorus and flanger the difference isn't very pronounced. For highly time-variant effects, like pitch shifting, the difference can be marked."
    • "Linear means that the output is related to the input by a straight line: y = mx + b. Filters are example of linear systems. A cabinet IR is a filter. Distortion is an example of a nonlinear system. Linear systems are associative and commutative. Associative means that a * (b * c) = (a * b) * c. Commutative means that a + b = b + a or a * b = b * a. Therefore you can do cab -> eq (a * b) or eq -> cab (b * a). The cab block is "completely" linear if motor drive is non-zero but it is "wide sense stationary" so you can treat it as linear." source
    • "The cab block is level-dependent if the Motor Drive is non-zero. So if you turn up/down the level out of the amp block you may need to compensate by doing the opposite with the Motor Drive." source
    • "You gain nothing putting it before the cab and risk collapsing the stereo image if the cab is mono." source

Cab: mono/stereo output

  • Keep an eye on the mono/stereo configuration. When placing the Cab block at the end of the grid, the output signal will be summed to mono, unless the Cab is set to Stereo mode, or when using two panned Cab blocks.
  • If the Cab block is set to Stereo mode, but it is followed by a mono effect such as Drive, the resulting signal is also summed to mono.

Cab: Hi/UltraRes, Normal Res, Stereo

  • The Cab block has various resolution modes: Hi/UltraRes, Normal Res, Stereo, Stereo UltraRes.
    • Normal Res: supports IR consisting of 1024 samples. You can often use Normal Res without any negative impact on the tone, compared to Hi Res.
    • Stereo: supports either two Normal Res IRs.
    • Hi/Ultra: supports Hi Res IRs (2040 samples) as well as UltraRes IRs (up to 8000 samples). Using Hi Res or UltraRes IRs requires more CPU power than Normal Res / Stereo. But UltraRes is more efficient than Hi Res, which results in about 4% LESS CPU usage with MORE resolution.
    • Stereo UltraRes: non-UltraRes IRs are processes in normal resolution whereas UltraRes IRs are processed using the UltraRes engine. If one IR is UR and the other not, then the UR will still processed as UR, and the other as Normal.
  • If you don't need cabinet modeling at all, for example because you're using the Axe-Fx II for effects only, or exclusively with a power amp and speaker cabinet, switch off Cabient simulation in the Global menu. This will decrease overall CPU utilization considerably.
  • Samples (resolution) translated in milliseconds:
    • Normal Res = 1024 samples = 20 ms
    • Hi Res = 2040 samples = 40 ms
    • UltraRes = 8000 samples (max) = 170 ms (max)

Cab: UltraRes in detail

  • Firmware 13: Added support for UltraResTM speaker IR processing. UltraRes is a proprietary technique that enhances the spectral resolution of an IR without adding CPU burden or storage requirements. Full support of UltraRes requires the CabLab utility to convert .wav files to the required data format. NOTE: UltraRes IRs do not support size warping therefore the Spkr Size parameter is unavailable for UltraRes cabinets. In Normal Resolution mode size warping is possible.
  • Cliff's comments below are taken from this thread:
    • "The problem with conventional IRs is that they are too short to capture the detail in the low frequencies. There are those that maintain 20 ms is the maximum length you need to fully replicate the speaker. This would be about 1000 samples at 48 kHz. I disagree with this as I have many IRs here that exhibit significant energy beyond 20 ms. I believe the room has some influence as the low-frequency modes of the room will impact the resulting sound. The amount of this impact depends on the room, the mics, distance, etc., etc. Or perhaps certain speakers have particularly high Qs in the low frequencies. Regardless, it is my opinion that you need IRs much longer than 20 ms to fully capture the "mic'd amp in the studio" sound. My tests show that IRs of 8000 samples are required to fully capture the low-frequency detail. Unfortunately to process an 8K IR in real-time require copious processing power... Fortunately I have developed "UltraRes" cabinet modeling. UltraRes cabinet modeling provides the frequency detail of a very long IR with little or no added processing power requirements. The following image depicts the response of UltraRes cabinet IR processing: ..." (see thread)
    • "Existing IRs will still be processed as usual. UltraRes IRs will be tagged as such which will indicate to the processor to use the new processing algorithms. Note that UltraRes IR data is not conventional IR data."
    • "The frequency resolution of an IR is the sample rate divided by the number of samples in the IR. The window function has nothing to do with frequency resolution (except for making it even less). So a 1K IR at 48 kHz sample rate has a frequency resolution of roughly 48 Hz. If a speaker has a resonance (formant) at, say 80 Hz with a Q of, say, 3.0, then 48 Hz is insufficient to capture that resonance accurately. You need a frequency resolution of several Hz to accurately recreate that resonance. I chose 80 Hz and a Q of 3 because that's what that response looks like. The Q could even be higher than that. It doesn't take much mental energy to realize that if you have a narrow formant at a low frequency then you need fine frequency resolution to reproduce that. An 80 Hz formant with a Q of 3 only spans about 25 Hz. Obviously a frequency resolution of 48 Hz is not going to be able to reproduce that. Windowing only smooths the response even more. This is basic FFT theory. The less time-domain information you have, the less frequency domain information you have and vice-versa. This is the uncertainty principle. I always window IRs with a Hann window. EDIT: I broke out my impedance measurements for that Vox cabinet and the speaker resonance is 80 Hz."
    • "Another way to look at it is to think in terms of formants. That particular speaker has a pronounced 80 Hz formant. It takes well over 100 ms for the energy of that formant to decay to the point of imperceptibility. Obviously a 20 ms IR can't reproduce an event that occurs for over 100 ms. Here is a zoom of the original non-minimum-phase IR (IOW raw time response)... (see thread). You can clearly see the 80 Hz formant. There are some room reflections but they are very small. The 80 Hz formant starts well before any reflections. It's obviously a high-Q resonance as it rings for quite a while. The higher the Q, the longer it takes to decay."
    • "Here's another example. (see thread) This is one of the new OwnHammer IRs. The IR is OwnHammer_412_MAR-CB_D-120_SS_RBN-121. These IRs are 100 ms long (4800 samples). I windowed the original IR to 4K to prove a point. The blue trace is the original IR (windowed to 4K samples). The green trace is the "typical" 20 ms IR (windowed to 1K samples). The red trace is the UltraRes version."
    • "The problem is that human perception is logarithmic and IRs are a linear process. 48 Hz resolution is way more than necessary at, say, a few kHz but not nearly enough at low frequencies. The brute force solution is to use very long IRs, 8K or more. UltraRes solves this in a novel way that uses little to no extra processing power and no additional latency."
    • "Normalization is your friend. Rectangular windows are simply truncation and are generally regarded as bad practice due to extremely high sidelobe levels. The choice of window is subjective. I actually use my own custom window that is not really a Hann window but that's proprietary information. My window preserves more frequency detail while still suppressing Gibbs phenomenon. Windowing trades off frequency resolution for sidelobe suppression. My window is optimized for the unique statistics of IRs. For a random process I tend towards Bessel-Kaiser windows. IRs have unique statistics that aren't addressed by any of the standard textbook windows."
    • "It is desired that the IR be 8K samples or more."
    • "Let me state these points:
      1. We don't record guitar amps in airplane hangers or anechoic chambers. We record them in studios.
      2. When we record a guitar amp we carefully set the amp up in the studio to get the best sound "on tape". This involves moving the amp around, placing gobos, etc. When we collected the Producer's Packs IRs we spent hours arranging the amps/speakers, mics and gobos and playing through the amp and readjusting until we were satisfied. This also included adjusting the preamps and mixing board. In one studio we found that we got the best tone raising the cabs off the floor by a couple feet, orienting them towards a particular wall and placing gobos behind (this was the engineer's standard recording arrangement).
      3. At this point our objective of the IR is to capture the sound of that amp/speaker at that position in the room, with the gobos, mics, preamps, etc., etc. The goal is not to capture the raw sound of the amp/speaker in an airplane hanger or outside using a ground-plane measurement and measurement mics. That might be someone else's goal but it is not ours. IOW our goal is to treat the cab, mics, preamps, room, etc. as a whole, as a good engineer/producer would.
      4. Subsequent analysis of the data shows that there is significant energy out to 100ms and even beyond. However there is little energy beyond 200 ms or so (as it should be in a well-designed studio). This observation was the catalyst for the UltraRes algorithm. There are other observations about the statistics of the data that I cannot disclose.
      5. Some cabinets displayed noticeable resonances at low frequencies. Others did not. The frequency of these resonances were not consistent and, not coincidentally, matched the measured resonance of the impedance sweep. It is a logical conclusion, therefore, that the resonance was NOT caused by the room but by the speaker/cabinet combination. Furthermore a plot of the group delay for the raw data showed that the delay of the resonance was too short to be a room mode. Regardless, whether the resonance is from the speaker or room or mics or preamps is irrelevant. All we care about is recreating the sound of that speaker as it would be recorded as accurately as possible.
      6. Truncating an IR destroys information by definition. We don't care where the information comes from, be it the speaker or the room or the mics or the preamps. We want all the information. If a plot of the frequency response of a truncated IR differs considerably from the non-truncated version then we have lost information and concomitant accuracy.
      7. NO ONE producing commercial IRs records them in an airplane hanger, for obvious reasons. The best ones are done in a studio using the same technique we used for the Producer's Packs: setting up the cab, adjusting the position, mics, preamps, etc. and playing through the amp/cab and readjusting until the best tone is achieved. The new OwnHammer IRs are an example of this. Many, if not all, of those IRs exhibit significant energy to 100 ms (and likely beyond but the data stops at 100 ms). Truncating them to 20 ms destroys vital information. You can argue the semantics all day long. I've compared truncated and non-truncated and the difference is clearly audible. It is especially noticeable when chugging power chords. You can hear the resonance. It goes "bonggggggg" as opposed to "thuk". Most importantly it sounds "better" IMO.
      8. UltraRes is an algorithm that markedly increases accuracy. It gives the frequency resolution of a 200ms IR without additional processing overhead and no added latency.
      9. Sometimes people can't see the forest for the trees."
    • "UltraRes is especially powerful in Tone Matching applications, particularly real-time matches and was another impetus behind the development."
    • "The myopic only see the IR as a capture of the speaker's "unadulterated" response. As I stated before I believe the future is treating IRs as capturing the entire recording chain including mics, preamps, etc. and have pushing in that direction. We have already seen the fruits of this labor in the Producer Pack and OwnHammer V2 IRs. We used mainly PP and OH IRs at Axe-Fest this weekend and the results were stellar. Andy Wood's tone was among the best guitar tones I've ever heard live and we dialed it up in 10 minutes under far less than ideal conditions. It consisted of the Two Rock amp model and the EV 12L Mix IR. When you include more than the speaker response in the IR you can have low-frequency resonances that persist for tens of milliseconds or more. Truncating an IR destroys this LF information. In many cases this LF information loss would probably not be perceptible. In other cases, from experience, it can be extremely noticeable. The bottom line is that you can always remove the information if you don't want it but you can't add back what isn't there."
    • "Let me phrase this another way. An IR can consist of the "raw" speaker response plus none, one, some or all of the following: mic, preamp, room, power amp (e.g. you want to capture the response of a tube amp driving the speaker), etc. If you only care about the raw response then a short IR is all that is required. However if you want any of the other elements as part of the IR then a longer IR may be necessary. UltraRes gives you the OPTION of processing longer IRs."
    • "Rigid thinking is great for textbooks. To push new boundaries you have to throw the books away."
  • More Cliff's comments:
    • "If the .wav is only 40ms long there is no sense in converting to UltraRes as you won't gain anything. Over 80 ms is desirable. The maximum length supported is 170 ms or so. Anything longer than that is truncated to 170 ms." source
    • "To get the optimum results the length should be 170 ms or more. As the length gets shorter you'll lose information. However there may not be any information to lose. It all depends on the IR. I've seen long IRs where only the first 100 ms or so is actual information and the rest is silence. OTOH I've seen 100 ms IRs where there is obviously more information but it got truncated. You lose nothing with UltraRes except the ability to change the size of the cabinet. You gain better sound and less CPU." "You can't mix UltraRes IRs as the data is not compatible. However... we foresaw that and the UltraRes conversion process produces two files: a .ir file and a .syx file. The .ir file is the raw IR data that can be imported into CabLab for mixing purposes. So CabLab can take .wav, non-UltraRes .syx and .ir files as input to the mixer section and product UltraRes .syx files." "The .ir files are included with our cabinet packs. We will not be offering .wav files. If you have the .wav file you don't need the .ir file. A .ir file can ONLY be used with CabLab. If you use the Axe-Fx II to capture IRs it will only generate .ir and/or .syx files. No .wav files are generated. The resulting data can only be used on Fractal Audio products." source
    • "It depends on the IR. UltraRes improves low-frequency resolution. It is very apparent with some IRs and virtually inaudible with others. It all depends on the low-frequency formants in the original IR. If there are significant, high-Q formants UltraRes will preserve those whereas conventional, short IRs will not. Audibility also varies with the amp being used. The difference is more audible with high gain as this will excite the formants more. Low-frequency formants vary with the type of cabinet and speaker. Some cabinets have a smooth low frequency response. Others have prominent formants. The mic also has an impact. Some mics will accentuate the formants. The room also contributes if it has strong LF modes. Furthermore some people like to capture an IR using a tube power amp. In this case you WILL get a significant formant at the low-frequency resonance of the speaker. A conventional IR will not capture that as the Q of the formant will exceed the resolution of the IR. UltraRes will capture that formant as UltraRes has 8 times the low-frequency resolution. Those who claim they can't hear a difference are correct. They can't. It's nothing to be ashamed of. But because they can't doesn't mean others also cannot. I can clearly hear the difference but I've trained myself on what to listen for. I vastly prefer UltraRes and only use UltraRes IRs in my personal patches (aside from the TV Mix, which is just a magical IR)." source
  • Cliff's comments about Tone Matching and UltraRes:
    • "In Realtime mode the raw internal IR length is 8K which you can dump." source
    • "You can export the Tone Match to CabLab and create and UltraRes IR." source
  • Sources for free UltraRes IRs

Cab: microphone models, Proximity parameter

  • Don't underestimate the impact of a mic type on the tone. I.e. adding the R121 model (Royer 121, captured at 6", front) will add highs and lows to the tone. The 57 DYN (Shure SM57) works with almost everything and adds bite. Many studios and users like to combine the Royer and SM57. Cliff: "I almost always use a mic model on the non-mixed IRs. I'll typically use a 57 on one side and an R121 on the other." source
  • The "None" and "Null" settings disable mic coloration. A mic is still involved though, because the IRs themselves are always captured with microphones. Even when a neutral mic was used to capture, such as an Earthworks mic. When capturing IRs, the mic is most often placed very close to the speaker, so the result is a close-mic'd tone. Still, selecting "none" is a good way to prevent adding additional eq-ing to the tone.
  • The Proximity parameter simulates the proximity of the modeled mic to the speaker. Higher numbers translate to the mic being closer to the speaker (nearfield). Lower numbers translate to the mic being further away from the source, with the lowest number providing far-field coloration. Proximity only works when a mic model is selected, including Null.
  • The Proximity Frequency parameter allows tuning the frequency range over which the proximity effect occurs.
  • Cliff's comments:
    • "The mic models are actually IRs. The mic IR is convolved with the speaker IR to create a composite final IR." source
    • "The mic I've been most impressed with for recording guitar lately is the Beyer M160. I don't like SM57's alone for amps. They're too spikey and compressed but mixed with an M160 or R121 they add some nice sizzle." source
    • "The M160 is an awesome guitar cab mic. All the IRs we got with the M160 came out really nice." source
  • There's useful about mics and mic positioning in the document Dialing in Your Tone by Red Wirez.
  • Wikipedia:
  • More information about microphones:
  • Additional information about microphones in this Wicked Wiki thread.

Cab: Input Select parameter

  • This lets you control the signal that enters the Cab block. For example, if you wish to run two panned Cab blocks, you can use this parameter to force one side of the signal to go into one Cab, and the other side into the other cab.

Cab: room ambience parameters

  • The Axe-Fx II provides room ambience parameters in the Cab block. This is a dedicated reverb effect, which also works well when using the Axe-Fx with headphones or IEM.
  • Enabling the cab's room reverb will turn a mono signal into stereo. So the left and right sides of the signal can be different because of this.
  • Also see Audio topics.

Cab: LowCut and HiCut Freq parameters

  • The Cab block provides low-pass and high-pass controls. These make it easy to tame boomy or harsh sounds. You can also use similar parameters in the Amp block, or apply eq-ing through a PEQ, GEQ or Filter block. Common settings are 80-150 Hz for high-pass (cutting bass), and 5-7 kHz for low-pass (cutting treble), but YMMV.
  • Cliff's comments:
    • "Using Low Cut in the Cab block is akin to what you would do in the studio to carve out room for the bass player." source
    • ""LOWCUT FREQ" in the cab block sets sets the -3dB point of a highpass filter at the output of the cab block." source

Cab: Speaker Size parameter

  • Cabinet Size Warping allows the user to change the relative size of the speaker. This parameter is accessible only when the IR is non-UltraRes and the Cab mode is mono.

Cab: Motor Drive parameter

  • This models the effect of high power levels on the tone of the speaker. The Motor Drive parameter controls the relative drive level and, therefore, the intensity of the effect.
  • Changing Motor Drive from 0.00 to 0.01 and above will cause an audible "click".
  • When using two UltraRes cabs in a preset, don't use Motor Drive on only one of them, because this will cause a hollow sound.
  • Cliff's comments:
    • "Motor drive isn't EQ. It models efficiency reduction due to thermal effects." source And: "What I have found is that thermal compression is somewhat noticeable and measurable. This is modeled by the Motor Drive parameter." source
    • "Motor Drive will cause compression if not set to zero (as it models driver compression). Otherwise the cab block is completely linear and will not cause any compression." source

Cab: Air parameter

  • The Air parameter mixes some of the signal going into the Cab block with the signal leaving the Cab block. The Air Frequency parameter lets you adjust the cutoff frequency of the mixed signal. Increase the Frequency to its maximum value for a straight mix.
  • If you want to listen to just the Air'd part of the signal, set the Cab to an empty user cab, and turn up Air.

Cab: Delay parameter

  • Micro delay for stereo application. When running a stereo mode, or two cab blocks in parallel, delaying one cabinet relative to the other can achieve interesting comb filter effects. A common practice in studio recording is to use multiple mics on a speaker at different distances to intentionally introduce comb filtering.
  • Cliff: "My secret to realistic cab sounds is Delay. Use two IRs in stereo or two cab blocks and put a small amount of delay on one (using the Delay parameter in the Cab block). I like around 0.06 ms. You may like more or less. Producers experiment with placing mics at different distances to enhance the recorded guitar tones. This is the same as using a small amount of delay. Adding a bit of delay introduces some comb filtering which creates notches and peaks in the response which, in turn, adds a sense of "space" to the tone. Try it." And: "If you have any cab packs try mixing the "Back" IR with one of the regular IRs. I use more delay when doing this, 0.1 ms or more. I lower the level on the back IR by a couple dB. This gives a nice "in the room" open-backed cab sound." source
  • GM Arts: "This is about mixing 2 signals: one without delay, and the other with a very short delay. 0.06mS is way too short to be perceived as a repeat; the effect is filtering caused by mixing these two signals. To keep things simple, we’ll apply an equal mix of the same signal and another delayed by 0.06mS. An easy way to experiment with this in the Axe-FX is with a Flanger block, with depth and feedback set to zero, and mix set to 50%. Adjust the delay to 0.06mS (not 0.6mS) to hear the effect with a mono signal. This produces a notched frequency response with complete signal cancellation just above 8KHz, with the -3dB point one octave lower at just over 4KHz. The signal is restored over the next higher octave (8KHz to 16KHz), but bear in mind that most Cab IRs will not have much response there anyway, so this effect is mostly a blocking filter over the range 4KHz to 8KHz. So if you have a cab IR that has some response over this range, it will be perceived as a loss of some treble response. For many, this will remove harshness in a way that’s difficult to achieve with other filters. Others may find this effect too much. You can soften this effect by decreasing the delay and/or changing the mix ratio. Decreasing the delay raises the frequency at which this cut occurs. For example, a 0.05mS delay blocks response over the octave 5KHz to 10KHz. Lowering the mix % decreases the depth of the notch. Similarly, applying a delay to a different Cab IR than the un-delayed block will “jumble” and reduce the final response to some extent. If you increase the delay (typically from 1mS and above), you’ll hear the combing effects as multiple notches become low enough to hear in the range of “guitar frequencies”. This sounds like a flanger or chorus without modulation, which shouldn’t be a surprise given we’re experimenting with a Flanger block. So why does this delay sound produce a tone more amp-like? Most players prefer their amp tone off-axis, meaning that they’re avoiding the direct harsh sound directly in front of the speaker, where high-frequencies are beamed. This filter simulates that effect. It’s also similar to standing slightly off-axis when using multiple speakers. Sound travels at roughly one foot per millisecond, so there is a very short delay between sound from different transducers. As Cliff stated, it also emulates recording techniques with mics placed at different distances from the cab. How to calculate? To find the frequency where this rolls-off high frequencies at -3dB, it’s simply: Hz = 1000 / 4 /delay in mSec . So for 0.06 mSec: 1000 / 4 / 0.06 = 4167Hz. Complete cancellation occurs at double this frequency, 8333Hz, and builds back to -3dB a double this frequency again, 16666Hz. Bear in mind that with higher delays, there will be audible effects from additional notches above this calculated frequency." source

Cab: Preamp simulation

  • Firmware 17: Added preamp simulation to Cabinet block. The simulations recreate the sound of overdriven channel strips, preamps, tapes, etc. The Drive parameter controls the gain of the simulation. The Sat parameter controls the ratio of even/odd harmonics. The Preamp Mode parameter (on Page 2) allows selecting between Economy and High Quality modes. In High Quality mode oversampling is employed to prevent aliasing but this results in higher CPU usage.
  • Cliff's comments:
    • "The VU meter shows the level into the pre. Select a pre Type and turn up the Drive. As the VU approaches the 0 dB marker you will begin to overdrive the pre." source
    • "Probably not something you would use for clean sounds. A common technique for rock music is to push the pres, console, tape, etc. to varying degrees to get compression and "sparkle". The trick is getting just the right amount. Too much and it sounds raspy and nasty." source
    • "0 on the VU meter indicates onset of clipping. It's not the same as your plug-ins in that regard. The problem with plug-ins is that you don't know where the onset of clipping is since the headroom isn't specified. Our way is superior since 0 dB indicates the point where things are clipping. The other way you have no idea where things start clipping. So 0 dB on the Axe-Fx is NOT equivalent to 0 dB on a typical plug-in." source
    • "I've done a lot of testing with isolation cabs. The big thing that happens is that the mic distorts, especially when using an SM57. This adds some crispness to the high end and some compression. I've found that I can duplicate that effect very closely by using the FET I preamp type in the Cab block and turning the Drive up until the desired compression is achieved. I set Sat to zero." source

Cab: related parameters in the Amp block

  • The Amp block has a couple of parameters which are closely related to the Cab block. Check the Speaker tab.

Cab: how to empty an user cab slot

  • To empty an user cab slot on the hardware, use Cab-Manager in Axe-Edit.

Cab: IRs for Axe-Fx Standard / Ultra

  • IRs for the Axe-Fx Standard/Ultra must be converted to be able to use these with the Axe-Fx II. source 
  • It's no use converting 1024-point IRs to 2040 points because they don't contain the necessary data. You need the original WAV-file to create a 2040 point IR.

Cab: actual speaker characteristics

Retrieved from "https://wiki.fractalaudio.com/wiki/index.php?title=Cab_block&oldid=26185"