Firmware supplied IR's

My experimentation with IRs I've created reveals that the IRs sound closer to the mic on cab when used in a DAW than in the AXE. This has nothing to do with the quality of the AXE convolution. It has everything to do with the length of the IR, the motion of the speaker even after the guitar signal is removed (post transient resonance), the reverberation that occurs within large poorly damped closed back cabinets, and the room reverberation that is captured no matter how much work I seem to put into attempting to dampen reflections even when placing the mic at a close distance. HOWEVER, if I truncate the IRs to 2040 points in my DAW, they sound identical whether used in the AXE or my DAW, and neither truncated IR sounds like the original mic on cab while the non-truncated IRs sound almost identical to the mic on cab. It's unfortunate in my opinion that the AXE can't use longer IRs that can capture the time domain effects of the above. However, by removing the reflections I find it's possible for captured near field IRs to come very close to replicating the mic on cab sound using the AXE as the host for the IR.

With an open back cabinet, a close mic, and a very dead room the length of the IRs becomes less than 40ms and the IRs sound very close to the mic on cab and identical whether they are hosted in the AXE or a DAW.

That's part experience and part speculation on my part. I know the reverb time for a large sealed cabinet is longer than 40ms, and a ported cabinet can have much longer decay times which will be clearly audible, but how much of this sound actually gets picked up by the microphone when such a cab is close miked is dependent on at least several variables. The post transient resonance varies with the type and size of speaker and more importantly the cabinet, with ported enclosures exhibiting long "decay" times, and again I'm not certain how much this contributes to the difference in a truncated IR vs. a non-truncated IR. It's difficult for me to separate these effects but there is something more than just the room that's affecting the length of the IR in my experience and opinion. Take it for what it's worth. Capturing IRs outside is a good way in my opinion to eliminate room reflections from the equation and get an IR having very little if any significant information beyond 2040 points using the right cabinet and speaker.

I'm not an expert by any means on speakers or enclosures as you can probably surmise from my comments but I do have a lot of practical experience with capturing and experimenting with IRs. I have yet to hear a FF IR that sounds convincingly like an "amp cabinet in the room". I've never attempted to create my own but my attempts to use the ones I've purchased have been less than satisfactory. I probably just don't know what I'm doing.

May I ask, what is the best quality software in your opinions for deconvolution and creating IRs? What are you guys and gals using? This question is directed at everyone who has contributed to this very educational thread. This is the most thought provoking thread I think I've come across in a very long time.

Thanks for the informative discussion!

And @AlbertA: Your conversion utility for AXE IR's has been of tremendous help to me over the years. Just thought I'd take this opportunity to say thanks! It rocks!

How do you monitor the AxeFx when you are using an cab block and IR?

And where can you purchase far field IR's?

The 3 factory far fields sound like a cabinet in the room to me especially when I'm monitoring with a good wedge and power amp. But even on my nearfields they sound like it. This is with A/B to an amp in the same room.

Richard
 
How do you monitor the AxeFx when you are using an cab block and IR?

And where can you purchase far field IR's?

The 3 factory far fields sound like a cabinet in the room to me especially when I'm monitoring with a good wedge and power amp. But even on my nearfields they sound like it. This is with A/B to an amp in the same room.

Richard


I monitor the FF IRs in the AXE through a single flat studio monitor in mono, and for close miked IRs I monitor in stereo using two flat studio monitors. I "sort of" hear a "cabinet in the room" effect with the FF, more so than with a near field IR, but only when I run a single speaker in mono, and it still doesn't give me a perfect sense of a cabinet being in the room. Maybe I'm using the wrong FF IRs. I understand the concept but it just doesn't sound good to my ears. The FF IRs in the AXE sound thin and "phased" to me. I just don't like them. Maybe they actually do sound like they're "cabs in the room" if I'm sitting in the right location with respect to the monitor but I just don't care for the sound of the cabinets that were used to create them.

RedWirez BIGBox collection has FF IRs. I haven't used them in some time but when I get home I can tell you how many there are and what cabs and mics were used. I purchased them a year or two ago.

At this time I can't do an A/B comparisons of a cab and a FF IR created from the same cab since I don't have a FF IR of a cab I own. When I get a chance one day perhaps I can take a small cab outdoors and take a FF IR outside with the elevated speaker cab pointed at an angle toward the sky to eliminate almost all reflections and put a mic on a long pole. That seems like it would work, although I've never tried it.

I think I would probably like FF IRs if the right cabinet were used. I tend to prefer very close miked speakers with very little phase cancellation and solid bass response, not the "thin" "phasey" sound of the FF IRs i've tried.

I'm not knocking FF IRs, just relating my personal taste and perception. When I get home today I'll try out some of the RedWirez FF IRs and give them another chance.
 
I monitor the FF IRs in the AXE through a single flat studio monitor in mono, and for close miked IRs I monitor in stereo using two flat studio monitors. I "sort of" hear a "cabinet in the room" effect with the FF, more so than with a near field IR, but only when I run a single speaker in mono, and it still doesn't give me a perfect sense of a cabinet being in the room. Maybe I'm using the wrong FF IRs. I understand the concept but it just doesn't sound good to my ears. The FF IRs in the AXE sound thin and "phased" to me. I just don't like them. Maybe they actually do sound like they're "cabs in the room" if I'm sitting in the right location with respect to the monitor but I just don't care for the sound of the cabinets that were used to create them.

RedWirez BIGBox collection has FF IRs. I haven't used them in some time but when I get home I can tell you how many there are and what cabs and mics were used. I purchased them a year or two ago.

At this time I can't do an A/B comparisons of a cab and a FF IR created from the same cab since I don't have a FF IR of a cab I own. When I get a chance one day perhaps I can take a small cab outdoors and take a FF IR outside with the elevated speaker cab pointed at an angle toward the sky to eliminate almost all reflections and put a mic on a long pole. That seems like it would work, although I've never tried it.

I think I would probably like FF IRs if the right cabinet were used. I tend to prefer very close miked speakers with very little phase cancellation and solid bass response, not the "thin" "phasey" sound of the FF IRs i've tried.

I'm not knocking FF IRs, just relating my personal taste and perception. When I get home today I'll try out some of the RedWirez FF IRs and give them another chance.

I think our definitions of far field IR's aren't the same. That was my confusion.

Richard
 
I think our definitions of far field IR's aren't the same. That was my confusion.

Richard

My definition of a far field IR is one that is sampled by a microphone at a distance from the cabinet in preferably a perfectly anechoic chamber so that no reflected sound is captured - only the direct sound of the cabinet itself is captured, and at a distance far away enough such that the the sound of each speaker contributes equally and the overall frequency response or power response is captured (the microphone is far enough away such that it is essentially equally distant from all of the speakers in the cabinet). The distance from mic to cab typically ranges 2 meters or more, the object being to capture the "overall response" of the cabinet, not the response at a certain point that's dependent on where the mic is placed.

The further away from a cabinet you are, the less the response will be affected by moving the microphone. At a great distance, movement of the mic by a even an entire foot would be inconsequential due to the geometry. When a cabinet has the microphone close to it, moving the mic by even a fraction of an inch can radically change the sound, and the captured sound at close range is not indicative of the overall power response of the cabinet. I apologize for incorrectly using the definition of far field.

Perhaps the RedWirez "far field" IRs are not truly far filed IRs. I don't know - I didn't create them. All I know is that's what they call them.

The FF IRs in the AXE sound like cabinets with the microphones at a distance such that the overall response of the cabinets are captured. My understanding of the logic behind this is that when a FF IR is used with a flat full range speaker, it will produce the same spectral power density (response with respect to frequency) as the actual cabinet that was used to create the IR would, and hence, "sound" like the cabinet was in the room. It seems to me using a single speaker as opposed to a stereo pair would more realistically recreate this effect, but that's speculation on my part.

I tried the FF IRs in the AXE, and although I can't disagree that they sound like cabinets in the room with me as opposed to close miked speaker cones being amplified through a sound system, I simply don't care for the way they sound. That's simply my subjective opinion and shouldn't be taken as an insult.

I would imagine creating a true, quality FF IR is a lot of work and takes some special equipment. Anechoic chambers are hard to come by, especially large ones allowing distant placement of the microphone so that the power response of the cabinet is accurately captured. My idea of placing a small (sealed back) cabinet above the ground pointed upwards so that all of the sound is directed upward toward a totally anechoic sky with a microphone on a pole holding the mic several meters above the cabinet seems to me like it would produce a reasonably good FF IR, but that's according to my incorrect understanding of what a FF IR is supposed to be.
 
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Putting aside whatever a far field IR is supposed to be, I use Voxengo as a deconvolver and a 15 second frequency sweep to create my IRs, both when I used to mic cabinets and now when EQ matching tones with Ozone. I don't know if there's anything better out there. I've been told that one of the contributors to this thread uses a Linux based program that he modified - that's far beyond my abilities or expertise. I'm looking for something relatively easy to use but with quality being the main concern.
 
No apology necessary. Jay was the only source of far field IR's that I knew of... until the soon release of the OwnHammer far fields (
Kevin is good people :))

I just thought someone else out there was already selling far fields and wanted to check em out... no harm no foul.

Richard
 
Perhaps the RedWirez "far field" IRs are not truly far filed IRs. I don't know - I didn't create them. All I know is that's what they call them.

I didn't like them either. We'll see once OwnHammer releases their FF versions to see if they fare better.

The FF IRs in the AXE sound like cabinets with the microphones at a distance such that the overall response of the cabinets are captured. My understanding of the logic behind this is that when a FF IR is used with a flat full range speaker, it will produce the same spectral power density (response with respect to frequency) as the actual cabinet that was used to create the IR would, and hence, "sound" like the cabinet was in the room.

The same frequency response (this includes amplitude and phase, not just spectral power density). But yes, the resulting IR would emulate the CAB and the CAB alone. When you use this IR and play in a specific room - if you compare this against playing with the physical CAB in the same specific room it should sound the same; If you use headphones though, you'll get the sound of the cab completely dry.

It seems to me using a single speaker as opposed to a stereo pair would more realistically recreate this effect, but that's speculation on my part.
That choice is strictly up to you. Nothing forces you to use a stereo pair for reproduction. Using a stereo pair would mean there's a sweet spot where an apparent phantom image will be coming in between the stereo pair.

I would imagine creating a true, quality FF IR is a lot of work and takes some special equipment. Anechoic chambers are hard to come by, especially large ones allowing distant placement of the microphone so that the power response of the cabinet is accurately captured. My idea of placing a small (sealed back) cabinet above the ground pointed upwards so that all of the sound is directed upward toward a totally anechoic sky with a microphone on a pole holding the mic several meters above the cabinet seems to me like it would produce a reasonably good FF IR, but that's according to my incorrect understanding of what a FF IR is supposed to be.

An-echoic chambers are not needed. You can use a ground plane measurement technique instead.

No truly special equipment is needed either;The only thing remotely special is a properly calibrated measurement mic, but other than that it's just normal equipment (an audio interface, a computer, a power amp and the cabinet in question).
By far the biggest hurdle is having access to a space big enough to avoid reflections in the window of interest (i.e. at least 20 milliseconds). Most homes and apartments most likely don't have such space.
 
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I don't know if there's anything better out there. I've been told that one of the contributors to this thread uses a Linux based program that he modified - that's far beyond my abilities or expertise. I'm looking for something relatively easy to use but with quality being the main concern.

I guess that would be me. I create my own software - currently on a linux 64-bit environment (ubuntu). They are all command line based so it's not something I would release.
I'm brewing some ideas to put a UI together for this purpose, but as always I don't have enough time available :)
 
I guess that would be me. I create my own software - currently on a linux 64-bit environment (ubuntu). They are all command line based so it's not something I would release.
I'm brewing some ideas to put a UI together for this purpose, but as always I don't have enough time available :)

Yes, that's you! And I've always loved your software for converting and examining the IRs for the AXE with the plots of the IRs themselves, the frequency response AND the phase response. I wish I knew how to do all that. I'm an EE, but only took entry level DSP courses and that was in the 80's.

I found the files that RedWirez claims are "far field". Here's a picture of the directory of the ones for the AXE. I suspect they are not what they say they are from what you're telling me.

RedWirez.jpg

Some of them indicate a 2m distance, some don't, some are off axis, some are on, etc. These were advertised and are labeled as "Far Field" IRs. Don't blame me though - I'm just repeating the claim THEY made!!! Maybe RedWirez needs a little more truth in advertising.

I've read a lot of Jays posts years ago about far field IRs. He seems to really know his stuff. Again, a lot of things I wish I knew more about. But I'm here to learn and relate my experiences and perceptions.

Thanks for the info.

I'm not clear on how a FF IR could be obtained using a "ground plane measurement technique". I'm not going to ask you to explain it to me, but is there a link to an explanation you could give me. I'd really like to learn about this. The more technical the explanation the better. I'm pretty good at understanding the behavior of sound, how it reflects, how reflections lead to phase cancellations or reinforcement, etc. I'm sure I'm capable of grasping the ground plane measurement technique. I also have no issues with bringing cabinets and mics outside to eliminate room reflections.

One more question: Using a frequency sweep method, I assume that reflections that are longer than 20ms will show up in the IR but can simply be truncated to obtain the FF IR without the reflections. Would this assumption be correct?

Thanks.
 
Yes, that's you! And I've always loved your software for converting and examining the IRs for the AXE with the plots of the IRs themselves, the frequency response AND the phase response. I wish I knew how to do all that. I'm an EE, but only took entry level DSP courses and that was in the 80's.

Also an EE here but I specialized in DSP/image processing and communications. The graphs shown in the utility are pretty easy to compute; just do an FFT of the signal of interest and convert the result into polar form.

I found the files that RedWirez claims are "far field". Here's a picture of the directory of the ones for the AXE. I suspect they are not what they say they are from what you're telling me.
View attachment 10187

Some of them indicate a 2m distance, some don't, some are off axis, some are on, etc. These were advertised and are labeled as "Far Field" IRs. Don't blame me though - I'm just repeating the claim THEY made!!! Maybe RedWirez needs a little more truth in advertising.

Yep, I have those. Not to my liking either. I can't comment on how they were produced as I have no idea:)

I've read a lot of Jays posts years ago about far field IRs. He seems to really know his stuff. Again, a lot of things I wish I knew more about. But I'm here to learn and relate my experiences and perceptions.
Thanks for the info.

Yep Jay's the master :)

I'm not clear on how a FF IR could be obtained using a "ground plane measurement technique". I'm not going to ask you to explain it to me, but is there a link to an explanation you could give me. I'd really like to learn about this. The more technical the explanation the better. I'm pretty good at understanding the behavior of sound, how it reflects, how reflections lead to phase cancellations or reinforcement, etc. I'm sure I'm capable of grasping the ground plane measurement technique. I also have no issues with bringing cabinets and mics outside to eliminate room reflections.

Sure here's one reference:
AES E-Library » Ground Plane Acoustic Measurement of Loudspeaker Systems

And here's a small concise summary:
mh-audio.nl - Groundplane Measurement

One more question: Using a frequency sweep method, I assume that reflections that are longer than 20ms will show up in the IR but can simply be truncated to obtain the FF IR without the reflections. Would this assumption be correct?

Thanks.

Yes you just window it out.

I would also suggest Angelo Farina's publications (it goes into depth about the advantages of exponentially swept sine for measurement):
http://pcfarina.eng.unipr.it/Public/Papers/134-AES00.PDF
http://pcfarina.eng.unipr.it/Public/Papers/226-AES122.pdf

Angelo Farina's Publications

And lastly this is the sweep I generated (along with its inverse):
Index of /sweeps

-This is around 20 seconds, at 48KHz
-It's generated to start and end in sine phase at every octave (useful to extract harmonic distortion products not just shove them into negative time).
-Since it ends at sine phase and the sweep extends to nyquist limit we avoid having to window the end - so we avoid pre-echo and ringing artifacts due to the sweep during deconvolution.
-About first octave (10-20) is used to window in the sweep to avoid low frequency ringing during deconvolution.

If you have a utility that computes convolution (linear convolution) then you can just convolve the inverse sweep with the recorded sweep; Then you just trim the "negative time" portion (number of samples in inverse -1) to obtain the final IR.
After this, you can trim out the time of flight (and any latency introduced by the recording system) to obtain the final IR.
 
Possibly a really dumb question but I have to ask

If you folks can tolerate one more question...

This is something I've wondered about for years and I finally have the opportunity to ask uniquely qualified individuals about the answer to my question.

According to my understanding a 20ms 1024 point IR is sufficient for a FF IR. Does this include bass cabinet IRs?

To get right to the point, would a front ported bass cabinet continue to sound at its resonant frequency for more than 20ms once the signal to the speaker is removed? If so, wouldn't the truncation of this sound at 20ms alter the sound of the IR?

A 41Hz tone has a period of about 24ms. If a bass were playing a 41Hz note through a cabinet that had a resonance near that frequency and the player changed to a dead patch to mute the sound, would the cabinet continue to sound for more than one period of the tone? If not, I have my answer. If so, wouldn't a longer IR be needed to accurately capture the response of the cabinet?

What about the same issue with a front ported guitar cabinet? If a cabinet were resonant at about 82Hz and the low E was immediately muted, how many 82Hz cycles could theoretically be reproduced by the cabinet resonance that would be audible? With a period of about 12ms, if it were more than two audible cycles would not a 1024 point IR truncate information? If the tone had a strong sub octave and the cabinet had a resonance point around 40Hz, how many cycles of that would be audible once the signal was cut off?

Perhaps the effect of cabinet resonance dies out within a fraction of a cycle and this has no effect whatsoever and my concerns are totally without justification. That's what I want to know. Is this anything to be concerned about, and could any of the above concerns have a real effect in the real world, or am I totally off base here?

If I can't get a firm answer (I have yet to get one over the course of the last couple of years) I intend to test a real cabinet which has a very low frequency resonance by simply recording it using the ground plane technique (now that I know how to do it) and measuring the duration and amplitude of any residual oscillations.

Thanks for your patience with my possibly stupid question, but I have to ask so I can sleep at night. :D
 
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Albert,

Thanks for the answers and links - I very much appreciate it!!! I bookmarked every link and will read every word.
 
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Also an EE here but I specialized in DSP/image processing and communications. The graphs shown in the utility are pretty easy to compute; just do an FFT of the signal of interest and convert the result into polar form.



Yep, I have those. Not to my liking either. I can't comment on how they were produced as I have no idea:)



Yep Jay's the master :)



Sure here's one reference:
AES E-Library » Ground Plane Acoustic Measurement of Loudspeaker Systems

And here's a small concise summary:
mh-audio.nl - Groundplane Measurement



Yes you just window it out.

I would also suggest Angelo Farina's publications (it goes into depth about the advantages of exponentially swept sine for measurement):
http://pcfarina.eng.unipr.it/Public/Papers/134-AES00.PDF
http://pcfarina.eng.unipr.it/Public/Papers/226-AES122.pdf

Angelo Farina's Publications

And lastly this is the sweep I generated (along with its inverse):
Index of /sweeps

-This is around 20 seconds, at 48KHz
-It's generated to start and end in sine phase at every octave (useful to extract harmonic distortion products not just shove them into negative time).
-Since it ends at sine phase and the sweep extends to nyquist limit we avoid having to window the end - so we avoid pre-echo and ringing artifacts due to the sweep during deconvolution.
-About first octave (10-20) is used to window in the sweep to avoid low frequency ringing during deconvolution.

If you have a utility that computes convolution (linear convolution) then you can just convolve the inverse sweep with the recorded sweep; Then you just trim the "negative time" portion (number of samples in inverse -1) to obtain the final IR.
After this, you can trim out the time of flight (and any latency introduced by the recording system) to obtain the final IR.


This is a treasure trove of information - I've been reading for hours now and I can't thank you enough. The ground plane method makes sense when I visualize how the sound reflects. I would never have imagined that this would work had I not read the information in the link you provided.

I also came across a paper from a google search by Angelo entitled "Simultaneous measurement of impulse response and distortion with a swept-sine technique". It describes using CooleEdit Pro (which I have) to do this using exponentially swept sine waves. I also know someone who has Adobe Audition and I downloaded all the the Aurora plug-ins to experiment with.

I don't fully understand your last paragraph about convolving the inverse sweep with the recorded sweep and trimming the negative time portion "number of samples in inverse -1" , but before I pester you with questions I want to read Angelo's papers, particularly the ones on the separation of the IR from the distortion and the use of inverse and exponential sweeps, etc. etc. Please realize I'm not a DSP guy but I have the prerequisite knowledge to be able to understand this if I put the work in to study it. I know what Nyquist's theorem is for example, and I understand the concept of convolution and deconvolution conceptually and I used to be able to do the actual math, but I've always thought of convolution as the value of the area of the product of one signal passing through another in the time domain. I haven't yet attempted to grasp the manipulation of sinusoidal sweeps in the convolution and deconvolution of audio signals as a computational tool to determine IRs and now I find out even the distortion of slightly non-linear systems.

It will take me days or more to read through this before I can even begin to ask intelligent questions without wasting your time, but once I get a better handle on some of this I hope you'll have time to answer some very specific practical questions related to accurately deriving impulse responses using the techniques you mentioned above. While the inverse sweep appears to exponentially decay, the other sweep is constant amplitude except for a fade in at the very low frequencies at the start. I also noticed one of Angelo's papers deals with using exponential sweeps as a means of separating IRs from distortion and you touch on this in your post above, but I have yet to read and understand the theory behind this - however, I will.

I think you have instigated a serious interest in this subject that will lead to me spending many of my waking hours in the coming weeks re-educating myself and teaching myself the basics of signal processing and at least some the basic theory that I wish I had learned long ago. It was my intent to go into either electronics or control system design when I graduated in 1985 but I was offered a lucrative job with the electric utility in the city where I lived, was playing in a band at the time and didn't wish to relocate, so I accepted a job in the power industry that has offered no challenge whatsoever to my intellect (or what's left of it after atrophy of the brain from non-use over the years). I've spent my spare time studying analog circuit design and designing and building analog audio processors and tube amplifiers which is quite simple (and useless) compared to the digital electronics and software algorithms in the world we live in today. I'm a dinosaur from an age gone by trying to come up to speed (or a fraction thereof at least) in the area of DSP.

Thanks again, and sorry for such a long post - I'm excited about learning and applying this new knowledge I didn't know existed until recently and the papers by Angelo are exactly what I've been looking for but unable to find prior to the links you gave me.
 
Correction to the above: I'll be back in a few weeks or months with some questions. The math is getting serious, and I'm having to research a lot of the terminology and functions to understand it. I feel like I'm back in school except the text books have been replaced with web pages.

The paper I found by Angelo Farina about Simultaneous measurement of impulse response and distortion with a swept-sine technique was the same one Albert linked me to that was presented at the AES - I thought it was a different variation on the same subject but it's identical word for word. This stuff is awesome.

Back in a few months with a lot more knowledge in my head.
 
Here's another example from another session; this time no adjustment was needed - the recovered IR was just trimmed.
http://guitarlogic.org/comps/AxeFXIRvsMic-evm01.mp3

Here again I'm switching between the Mic signal and the Reamped AxeFx signal (see picture below to see where I switched back and forth).
The Mic signal has some bleed from the guitar strings being strummed so on some sections you'll hear this, on the reamped sections you won't. Other than that, they sound pretty much identical to me.

reaper_setup.png
 
Yet another HUGE thank you to AlbertA for mentoring and all the time spent processing, etc!!!

Like I have said, this has been an absolute game changer for me. By understanding, my world expands. If I want "this type of tone" to be afforded by an IR, I am becoming better equipped to more readily be capable of "sculpting" that tone. I am also far more aware of my situational limitations. But!!! Through the understanding of the science and math, this does not prohibit or preclude playing outside the box. In fact, for me, I feel encouraged by it.

I will be posting my patches and IRs from this journey thus far later today (for anyone interested). I am using a modified version of the stock "Recto Orange" patch (v5); Input Gate disabled, Amp Block: Input default 1.00, grid modeling off, gain reduced, Speaker Drive @ 0, and several other minor tweaks. No other blocks enabled. With the IR, Cab Block enabled, Mono IR, no Mic sim enabled, Motor Drive @ 0.
 
Yet another HUGE thank you to AlbertA for mentoring and all the time spent processing, etc!!!

Like I have said, this has been an absolute game changer for me. By understanding, my world expands. If I want "this type of tone" to be afforded by an IR, I am becoming better equipped to more readily be capable of "sculpting" that tone. I am also far more aware of my situational limitations. But!!! Through the understanding of the science and math, this does not prohibit or preclude playing outside the box. In fact, for me, I feel encouraged by it.

I will be posting my patches and IRs from this journey thus far later today (for anyone interested). I am using a modified version of the stock "Recto Orange" patch (v5); Input Gate disabled, Amp Block: Input default 1.00, grid modeling off, gain reduced, Speaker Drive @ 0, and several other minor tweaks. No other blocks enabled. With the IR, Cab Block enabled, Mono IR, no Mic sim enabled, Motor Drive @ 0.

Looking forward to the presets and IR's...

Rock on,

Richard
 
Looking forward to the presets and IR's...

Rock on,

Richard
Link to zip with IRs and patch. Included is a .txt explaining things (it is actually brief, lol).

I discuss the very basics surrounding them in this thread.
I will be doing a more in-depth write, along with my idea of approaching this. I am hoping it will help others, and also prompt others to share their approaches. I see guides/instructions concerning anything to do with art as points of inspiration, not "you must do this to get the best results," as I do not see "best." However, there are certain "rules" and "laws" that simply "are," such as the speed of sound.

Also, these IRs were not created with the intention of sounding great; rather, they were simply a vehicle as a manner of learning. By not having to "worry" about mic placement, etc and instead making sure everything was setup properly (the single IR from Session 1 was nearly ruined due to my own neglect, lol), the goal of the Sessions was allowed full attention. There are still some experimental sessions left to do before taking the time to attempt some really sweet IRs, but I still like several of the IRs from the two Sessions thus far. The single Session 1 IR is one I am using as my current "go to" IR, and a blend (stereo cab block, pan center as mono) of two from Session 2 is quite nice as well. Switching between the Session 1 IR and the stock ones is... interesting.. IMO... :D
 
Interesting stuff. I'm about to start my own experiments as time permits, but I'm still grappling with the math and technical theory behind some of this.

I found what to me was a very education paper that compares the work of Farina with previous techniques. It helped me to gain a little more understanding of the "why" and not just the results themselves.

http://www.bg.ic.ac.uk/research/g.stan/ArticleJAES.pdf

The theory behind the straight forward swept sine wave technique is now crystal clear in my head. What's still not clear is Farina's exponential swept sine technique and how the distortion products are pushed into "negative time" which removes them from the desired IR by allowing them to be simply truncated from the beginning of the recovered deconvolution. It's pretty cool stuff - I just wished I understood how and why it works, but given time I will.

Would this method not also be an easy way to evaluate the distortion of a system? I don't have an (expensive) distortion analyzer, but by simply sweeping a system with an exponentially decaying sine (Farina's method) one can see the amplitude of the various order distortion components. That seems pretty cool in and of itself.
 
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