FREE Custom IRs for Your Guitar and Its Pickup (Acoustic Guitar IR Impulse Response)

[caballero59]

The Atomic Amplifire makes a nice platform for loading custom IR's. It has extensive parametric EQing, pre-filter for rolling off boomy low-end, various boost functions, and the ability to scroll through different IR's with various levels of feedback resistance. Especially nice if you also play electric since one piece of equipment will do the job for both.

Because of impedance issues, I use either the Sunnaudio DI or a Cali76 compressor for a preamp. Cali76 works very well for this job btw.

[GuitarLuva]

Originally Posted by jonfields45

At this point I've made IRs for one person, Cuki made my IRs for another, and one person installed Octave and made their own (not including Doug Young who had them almost forced on him by me).

Only 3 customers and it's free. So far all three report very good results.

Either the market is small or the barrier to use them is too high :~(.

I actually had everything setup for this a few days ago and was about to make an IR and got interrupted. However, I just sat down a few moments ago and made one IR for one guitar. I created the initial wave file using reaper and I gain matched the "mic track" and "pickup track" and exported the wav file and used the IR generator. I than loaded that into reaverb and started messing around. In the end I settled on a dry signal of 0 and wet at 6. Very nice! What I'm listening to sounds very convincing. I used a Slate ML-2 mic to do the wav file and the IR sounds just like the guitar mic'd up. I'll be sure to do the remainder of my guitars in the near future.

A big thanks to Jon and Cuki for all the hard work they put into this and sharing it with everyone for free. If you're the type of person who's into this kind of stuff I recommend that you take advantage. Some pretty amazing stuff here and it's pretty quick and easy to do!

[Cuki79]

Hi Lezz60, Caballlero59 and GuitarLuva,

Jon and I, we are building a free IR database. If you are happy with the IRs you made, please send us your IR files with your guitar and pickup model (pm us a link with a google drive or dropbox link).

If you have time, you can even send us the original mic and pickup recordings so we can update the database when algorithm improvement occurs.

Thanks,
Cuki

[caballero59]

Originally Posted by Cuki79

Hi Lezz60, Caballlero59 and GuitarLuva,

Jon and I, we are building a free IR database. If you are happy with the IRs you made, please send us your IR files with your guitar and pickup model (pm us a link with a google drive or dropbox link).

If you have time, you can even send us the original mic and pickup recordings so we can update the database when algorithm improvement occurs.

Thanks,
Cuki

Will do. Next thing is get more serious about the recording.

[caballero59]

Been thinking about these topics a bit lately and have a question about the terminology "IR". It seems to me that in the electrical guitar world, the term IR being used to model the signature response of a speaker, or if you activated an acoustic box and miced the response, the term "IR" would be pretty accurate, but with the way it is being used for Tonedexter, etc, it is more of a mapping between two domains, i.e, the mic and pickup domains.

Any thoughts?

[jonfields45]

Here is my attempt at an explanation that is published on the website Cuki is shepherding. IR can properly refer to how to program a filter or how to measure the response of a system to an impulse. For linear, time invariant, finite response systems they are exactly the same.

In simpler systems like speaker cabinet modelers or hall reverbs, an Impulse Response is relatively easily measured, BUT it is not the only way to "deconvolve" those systems and measure the IR to program a FIR (finite impulse response filter) to duplicate the system. Sine sweeps are another way.

The problem of acoustic guitar pickup correction is actually a two step problem. First you need to get to the real guitar and then you need to color that with the microphone's characteristics. Direct impulse measurement is not easily applied to this problem. But my script, and the training means of ToneDexter and Soundscape take the most direct route I can think of to solving this problem.

----------------------

There is nothing in this article that is required reading to use an IR to make your acoustic guitar pickup sound more like a mic, but if you are curious...

IR stands for Impulse Response. A very complex filter (equalizer) compared to what we are used to with sliders and knobs can be made with digital computer logic and (for the subclass of linear, time invariant, and finite response) they are programmed with their "IR".

Electric guitarists love their distorting (nonlinear) tube amps and IRs are largely useless in reproducing distortion. But electric guitarist also like the frequency response of vintage speaker drivers and cabinets. A filter programmed with an IR can model any of those cabinets. Not really that surprising if you can hear the difference between a D-18 and a D-28. Cabinet IRs are plentiful for purchase and free download on the Internet.

For us acoustic guitarists, products like Aura and ToneDexter use an IR programmed filter to make your pickup sound more like your mic. Is that problem linear, time invariant, and finite? It is close enough for most piezo pickups (skip to the end for a discussion of the exceptions). One problem is loading and playing back the IR in performance and the other is creating that IR in the first place (Aura library, ToneDexter training, etc.). This website is addressing the problem of sourcing IRs to allow Aura like results using an inexpensive IR loader pedal.

It might be easy to imagine sampling a signal and feeding back some of the delayed signal into the live signal. A delay pedal is an example. If you set the delay pedal to feedback the delayed signal back into the delay line input, you've got the possibility for something that could make a signal last forever. The first of these two possibilities, no feedback of the delay output into the delay input, is an example of a very simple Finite Impulse Response (FIR) filter. The other is an example of an Infinite Impulse Response (IIR) Filter. The length of a FIR filter that might make a good reverb is typically very long with compromised SNR and frequency response (to keep the memory to store the echo reasonably priced). The FIR filter we will be discussing here is very short and very high fidelity. It is being used as an EQ and will not be long enough for the more familiar time effects such as reverb or delay.

All these samples going into delay lines are examples of things you can do in what is called the “Time Domain”. But you are very familiar with the tone controls on your equipment. They are examples of things where the controls we use to manipulate them are labeled in the “Frequency Domain” (bass, treble, etc.). A French mathematician, Fourier, invented a transform which can be used to move mathematical formulas (representations of something you are interested in) between these two domains. His actual goal, like how a slide rule uses logarithms to turn multiplication into addition, was to simplify all sorts of hard stuff which in his case was heat transfer.

An impulse is a signal infinitely high in magnitude and infinitely short in duration. Of course, something like this is impossible in the real world, but close enough is good enough. The Fourier transform of an impulse in the time domain becomes "1" in the frequency domain (very cool and simplifying). If you put something like an impulse into your electric guitar amp speaker and sample the speaker output, what comes out is the impulse response. Transform that IR to the frequency domain and you’ve got something like the tone control settings to duplicate that cabinet’s frequency response. In the frequency domain EQ filters simplify into multiplication and that the transform of an impulse is a simple “1” means all that is left is the equalizer’s programming. Multiplication in the frequency domain becomes something called convolution in the time domain which is something you can do easily with a FIR filter (a FIR filter is really a convolution machine). But how to program that filter? It can be easily mathematically derived from that impulse response. It is as simple as the impulse response is the programming of that FIR filter. IR and FIR filter coefficients have become interchangeable ways to refer to the same thing in the music industry's lexicon.

If you record your acoustic guitar pickup and a mic pointed at the guitar, you now have two time domain responses of your playing. Taking the digital version of the Fourier transform (usually referred to as an FFT, or fast Fourier transform) and transform them to frequency domain, divide the mic by the pickup, reverse the transformation, and you’ve got the programming for your acoustic guitar IR pedal. To make this sound good it is much more work than this simplified view (and much harder than cabinet modeling). This process is often referred to as “training”. There is a lot of art to choosing what parts of the original recording to use and how to post process.

Since the IR is only about programming a very complicated equalizer with a fixed setting, there are systems with which it will be less effective or not work at all. An example of a system that is not compatible with an IR is the Baggs Session VTC. Because the Session distorts the pickup output (intentionally nonlinear) it generates frequency content that was never in the original guitar pickup's output and will be absent from the mic recording. The training process outlined above will fail (not linear and not time invariant). Magnetic pickups not placed adjacent to the bridge have a different frequency response as you play up the neck (not time invariant). A fixed filter can not deal effectively with this changing pickup output.

[jonfields45]

Some more food for thought :~):

The script I wrote takes 2^17 long segments of the pickup and mic, converts them to the frequency domain, takes their ratio, and then transforms that back to the time domain.

Looking at the frequency domain, and recognizing the math has redundant positive and negative frequencies, you still have 2^16 (65,536) different frequency complex (phase and amplitude) coefficients. At 48 kHz sampling, they range from DC to 24 kHz. An acoustic guitar is going to range from 82 Hz to ~10kHz.

((10,000-81)/24,000) * 65,536 = 27,085

That is an equalizer with 27,085 bands!

[Cuki79]

Originally Posted by caballlero59

Any thoughts?

You can also read what Larry Fishman wrote about it:
https://www.premierguitar.com/articl...digital-divide

Tonedexter applies a Finite Impulse Response filter.
Any EQ, Parametriq EQ, Graphiq EQ uses a Infinite Impulse Response filters.

IR is not a term specific to electric guitar, acoustic guitar... or guitar. It's a term that applies to any physical system that has a linear response... Which means 99.9% of everything on earth. However in this context, the term is used to describe the response of a filter.

You have to separate:

• The signal processing: the filter
• The filter design: the wavemap/IR training

The first point is not new. There is no more signal processing wizardry in tonedexter than in a ZOOM, Line 6, Fishman, BOSS pedal. They all apply linear filters with IRs (FIR or IIR).

The second point is "new" for the acoustic guitar market. However, as everyone can see, Jon's algorithm is not very complex (less than 40 lines of code), the greatness about Tonedexter is how robust, efficient and easy to use it is.

[jonfields45]

Originally Posted by Cuki79

as everyone can see, Jon's algorithm is not very complex (less than 40 lines of code)

In my engineering career the nursery rhyme "There Was an Old Lady Who Swallowed a Fly" was a funny and useful poem to remember.

Often in engineering products we would find ourselves fixing problems caused by the previous new feature or bug fix and then find that next fix caused even more problems not discovered in test.

There is a beauty and goodness in something so very simple that we all can understand how it works. In my case you have to accept Fourier's work on faith and the rest is just high school algebra (like you can't divide by zero).

More complex solutions often have a bit of that old lady hiding in them and are waiting for just the right not included in engineering test problem to motivate swallowing something even larger :~).

I think you can EQ (in your other pedals) my IR generator to taste and get where you need to go. You can run it at home with as many mic recording or playing options as you care to. If you want it to generate longer IRs, it is a simple edit of the script. I am getting positive feedback from people who are running the script at home and I hope that takes off.

I got an email from Mooer saying they are running short of staff due to the Corona virus but do intend to get back to me on my proposal:

A small simple 2 knob (mix & volume) and a bypass switch pedal that can variably mix a 2048 length IR with the input for $100 MAP. The PC/Mac side software would include my script (rewritten to not require Octave to run).

[Cuki79]

Originally Posted by jonfields45

In my engineering career the nursery rhyme "There Was an Old Lady Who Swallowed a Fly" was a funny and useful poem to remember.

Dear Jon,

I did not want to say your algorithm is of lesser quality than more complex ones. I just wanted to say there is no "magic" wizardry in making IRs.

Your algorithm is very efficient. That's why I stole what I thought was a great idea to improve mine.

The beauty of Jon's algorithm is how it almost annihilate the need for blend a low-pass filtering in a very elegant manner.

Both Baggs Soundscape and Tonedexter for example have some kind of recipe to limit the low end. In my case, you must add a parametric EQ, it means for example that the IRs I make needs this feature in the pedal that will host them: Not Jon's IRs.

Also, in LIVE situation you might not want to have a "mic" sound that will lack the punch and directness of a piezo pickup but also will bring feedback. So you'll end up blending some raw pickup sound. That's why for example a dual source like Anthem will always have some pickup in the signal chain.

The challenge in IR training is about finding the relevant information where there is almost only noise. My program spends a lots of processing to do that.... and I can't even guarantee the sucess. Jon who has my program knows that it's not a 100% sucess thing, and that you have sometimes to change a parameter to make the algorithm work. To avoid such a processing, Jon's algorithm blends back some pickup information, when it is difficult to create a correct mic "image".

This is a great idea. For example, everyone can hear that Tonedexter cuts the high end of the spectrum. This is because there is very few information there.

In Jon's algorithm, instead of "cuting" the high end because you can't make it sound like a mic, the pickup high end is kept almost unprocessed. It results in a more direct, punchier tone that is great for LIVE performance.

I am sorry if you misinterpret what I wrote.

Cuki

[GuitarLuva]

Originally Posted by Cuki79

Hi Lezz60, Caballlero59 and GuitarLuva,

Jon and I, we are building a free IR database. If you are happy with the IRs you made, please send us your IR files with your guitar and pickup model (pm us a link with a google drive or dropbox link).

If you have time, you can even send us the original mic and pickup recordings so we can update the database when algorithm improvement occurs.

Thanks,
Cuki

Cuki,

I will share all my IR's when I'm completely finished. I plan to do all my guitars and get the best possible IR for each of them. My first crack was just the basic setup, mic about a foot away pointing at the 14th fret. The HFN worked easily. Right now I'm struggling with the anthem. I can get it to sound really good with a low shelf curve of -9db with the IR. I'm trying to get each IR to sound as close to that particular guitar without any EQ afterward. I'll be pressed for time during the next few weeks but this is on my "to do" list.

[Cuki79]

Originally Posted by GuitarLuva

Cuki,

I will share all my IR's when I'm completely finished. I plan to do all my guitars and get the best possible IR for each of them. My first crack was just the basic setup, mic about a foot away pointing at the 14th fret. The HFN worked easily. Right now I'm struggling with the anthem. I can get it to sound really good with a low shelf curve of -9db with the IR. I'm trying to get each IR to sound as close to that particular guitar without any EQ afterward. I'll be pressed for time during the next few weeks but this is on my "to do" list.

Send me the recordings (pm + google drive link or dropbox link). I'll try my algorithm, maybe it will give a different result.

[caballero59]

Originally Posted by Cuki79

I agree with Doug. I sold the EPSI for these reasons. Note that my list does not include multifx like the atomic amplifire, kemper, helix...

I'm using the Atomic amplifire as a one-stop solution. My needs are high-Z, loadable user IR's, parametric EQ, boost. Does any one know of a reason why the Atomic Firebox wouldn't be able to accomplish all this for less money and footprint?

[caballero59]

Originally Posted by jonfields45

Here is my attempt at an explanation that is published on the website Cuki is shepherding. IR can properly refer to how to program a filter or how to measure the response of a system to an impulse. For linear, time invariant, finite response systems they are exactly the same.

In simpler systems like speaker cabinet modelers or hall reverbs, an Impulse Response is relatively easily measured, BUT it is not the only way to "deconvolve" those systems and measure the IR to program a FIR (finite impulse response filter) to duplicate the system. Sine sweeps are another way.

The problem of acoustic guitar pickup correction is actually a two step problem. First you need to get to the real guitar and then you need to color that with the microphone's characteristics. Direct impulse measurement is not easily applied to this problem. But my script, and the training means of ToneDexter and Soundscape take the most direct route I can think of to solving this problem.

----------------------

There is nothing in this article that is required reading to use an IR to make your acoustic guitar pickup sound more like a mic, but if you are curious...

IR stands for Impulse Response. A very complex filter (equalizer) compared to what we are used to with sliders and knobs can be made with digital computer logic and (for the subclass of linear, time invariant, and finite response) they are programmed with their "IR".

Electric guitarists love their distorting (nonlinear) tube amps and IRs are largely useless in reproducing distortion. But electric guitarist also like the frequency response of vintage speaker drivers and cabinets. A filter programmed with an IR can model any of those cabinets. Not really that surprising if you can hear the difference between a D-18 and a D-28. Cabinet IRs are plentiful for purchase and free download on the Internet.

For us acoustic guitarists, products like Aura and ToneDexter use an IR programmed filter to make your pickup sound more like your mic. Is that problem linear, time invariant, and finite? It is close enough for most piezo pickups (skip to the end for a discussion of the exceptions). One problem is loading and playing back the IR in performance and the other is creating that IR in the first place (Aura library, ToneDexter training, etc.). This website is addressing the problem of sourcing IRs to allow Aura like results using an inexpensive IR loader pedal.

It might be easy to imagine sampling a signal and feeding back some of the delayed signal into the live signal. A delay pedal is an example. If you set the delay pedal to feedback the delayed signal back into the delay line input, you've got the possibility for something that could make a signal last forever. The first of these two possibilities, no feedback of the delay output into the delay input, is an example of a very simple Finite Impulse Response (FIR) filter. The other is an example of an Infinite Impulse Response (IIR) Filter. The length of a FIR filter that might make a good reverb is typically very long with compromised SNR and frequency response (to keep the memory to store the echo reasonably priced). The FIR filter we will be discussing here is very short and very high fidelity. It is being used as an EQ and will not be long enough for the more familiar time effects such as reverb or delay.

All these samples going into delay lines are examples of things you can do in what is called the “Time Domain”. But you are very familiar with the tone controls on your equipment. They are examples of things where the controls we use to manipulate them are labeled in the “Frequency Domain” (bass, treble, etc.). A French mathematician, Fourier, invented a transform which can be used to move mathematical formulas (representations of something you are interested in) between these two domains. His actual goal, like how a slide rule uses logarithms to turn multiplication into addition, was to simplify all sorts of hard stuff which in his case was heat transfer.

An impulse is a signal infinitely high in magnitude and infinitely short in duration. Of course, something like this is impossible in the real world, but close enough is good enough. The Fourier transform of an impulse in the time domain becomes "1" in the frequency domain (very cool and simplifying). If you put something like an impulse into your electric guitar amp speaker and sample the speaker output, what comes out is the impulse response. Transform that IR to the frequency domain and you’ve got something like the tone control settings to duplicate that cabinet’s frequency response. In the frequency domain EQ filters simplify into multiplication and that the transform of an impulse is a simple “1” means all that is left is the equalizer’s programming. Multiplication in the frequency domain becomes something called convolution in the time domain which is something you can do easily with a FIR filter (a FIR filter is really a convolution machine). But how to program that filter? It can be easily mathematically derived from that impulse response. It is as simple as the impulse response is the programming of that FIR filter. IR and FIR filter coefficients have become interchangeable ways to refer to the same thing in the music industry's lexicon.

If you record your acoustic guitar pickup and a mic pointed at the guitar, you now have two time domain responses of your playing. Taking the digital version of the Fourier transform (usually referred to as an FFT, or fast Fourier transform) and transform them to frequency domain, divide the mic by the pickup, reverse the transformation, and you’ve got the programming for your acoustic guitar IR pedal. To make this sound good it is much more work than this simplified view (and much harder than cabinet modeling). This process is often referred to as “training”. There is a lot of art to choosing what parts of the original recording to use and how to post process.

Since the IR is only about programming a very complicated equalizer with a fixed setting, there are systems with which it will be less effective or not work at all. An example of a system that is not compatible with an IR is the Baggs Session VTC. Because the Session distorts the pickup output (intentionally nonlinear) it generates frequency content that was never in the original guitar pickup's output and will be absent from the mic recording. The training process outlined above will fail (not linear and not time invariant). Magnetic pickups not placed adjacent to the bridge have a different frequency response as you play up the neck (not time invariant). A fixed filter can not deal effectively with this changing pickup output.

Thanks Jon, that's good stuff.

[jonfields45]

I got asked a question about deconvolers and "can't you just use the mic output" in an email and maybe my reply is useful:

Frequency sweep is an alternative to impulse response measurement, but you need a place to input the sweep (or impulse) and a place to measure. For the problem of making a pickup sound like a mic, you are postulating that there is the real guitar with a FIR filter (IR) to the pickup output, and another IR to the Mic output. Since you've postulated that these two parallel systems are linear, time invariant, and finite, there ought to be a single FIR filter or IR that can go from the pickup directly to the mic (and skip the middleman). This problem requires correlated samples of the mic and pickup (as far as I know and the market does not seem to have a better idea). In this case the postulated real guitar is not the one you are holding :~)!