DunedinDragon

I would agree that the picking sound tends to be more prominent through almost any FRFR setup whether PA, cab, or direct studio monitor. In a sense I suspect it tends to be related to the greater clarity and articulation of the speakers which actually respond much faster and with less interference because of the division of labor between the speaker and compression driver. In some cases it bothers me, in others not so much. A lot depends on the style of song and often on the guitar being used. It's very useful in finger picking on a Gretsch hollow body but very often annoying on single coil guitars. Fortunately it can be pretty well managed by some of the deeper amp parameters such as bias and sag, but I don't see many people going to that level of detail on their patches very often.

It really has little to nothing to do with digital or tube amps when talking about this subject. It's really about speaker design and response profiles. All of the FRFR designs whether a cabinet, studio monitor, or a true live sound reinforcement speaker such as an Alto TS210 or DXR series from Yamaha have a much flatter response profile across the frequency response range than any typical traditional guitar cabinet due to the addition of the compression driver and often some form of DSP processing of the signal. The "harshness" is simply an artifact of that design factor. Generally the decisions fall into one of three categories, a traditional cabinet based FRFR solution such as a Firehawk, Atomic CLR or Friedman, a live sound FRFR speakers such as a Line 6 Stagesource, Alto TS210, or Yamaha DXR series; and then there are studio monitors such a Yamaha HS5/7/8, JBL LSR305 or other similar designs. Each of these are designed and perform differently based on the job they're intended to do. Most important is how they project the sound energy. Traditional cabinet based solutions and studio monitors are designed to fill a room or space with sound. Both they disperse sound in a generally equal pattern in all directions, but most of the traditional FRFR cabinet designs like a Friedman tend to have more powerful amps in order to fill a bigger space. But in all cases this sound dissipates fairly quickly across distance. The big difference is in the live sound speaker designs based on the fact that they are meant to fill a large space efficiently with less drop in sound energy across distance. Therefore they don't project in all directions equally. Instead they focus the sound energy in a very wide horizontal manner, but a rather limited vertical pattern. This is so that sound energy doesn't get wasted going into ceilings and floors, but directed specifically toward the performance area and crowd. Because of these things the "harshness" is really a factor of two things: the fact that you are now more equally sending sound across the entire listening range (unlike traditional guitar cabinets which have a much more pronounced low and mid response), and in the case moreso of live sound designs being in too close of proximity to the speaker so that the sound between highs and lows don't have the space to combine and blend appropriately. Obviously both of these things can be compensated for either by EQ or by speaker configuration and placement. So really the decision comes down to how you intend to use it and what's important to you. Clearly studio monitors aren't going to work in a live scenario, but provide the FRFR benefits in a home or recording scenario. Cabinet and live sound designs are really meant for live performance. Cabinets will give more of a traditional guitar cabinet feel, but will also color the sound slightly in the same way as all guitar cabinets do (this is what IR's try and model for you in the signal chain). Therefore the stage sound you're hearing may not be a very accurate representation of what your audience is hearing out of the FOH. If your desire is to have an accurate idea on stage of what your audience is hearing, then a live sound FRFR speaker would be best for doing that job. In all cases what you end up with in FRFR setups once you get them dialed in is a much more polished studio sound with greater articulation and clarity. That's more important in some forms of music than in others. I didn't really mean to drone on so long, but I think it might be useful in getting your bearings and making the right kind of decision for yourself and your situation.

I doubt it was your dual amps/dual paths. I run plenty of these types of patches out of my L/Mono output and have no problems. I suspect it's more likely due to changing your position relative to the speakers.

I do the same thing...a patch per song. That leaves me plenty of DSP space to get the patch where I want it, although I seldom come very close other than if I use 2048 sized IRs. Here's how I have mine set up when I'm doing my patches and maybe it will help you. I have my speaker (Yamaha DXR12) at about 4 or 5 feet off of the ground. Even though I normally set mine on the floor as a floor monitor when I play, I still leave my speaker settings for floor monitor so it reduces the very low bass levels that would normally couple with the floor, and I have my low cutoff set on the speaker for 120hz which matches the crossover from our subs to the fronts. That's pretty representative of the frequency response profile of most modern PA speakers. I have a small 4 channel mixer (Alesis) that I plug my XLR L/Mono into and I run 1/4 inch L/Mono out to the speaker. The speaker volume is set at 50%. The purpose for the mixer is to allow me to see the line input levels visually on the mixer's VU display. I set my Helix master volume at probably just a tad below 11 o'clock when I'm working with my patches, the gain on the mixer is set to 50%, so my levels on the meter linger around -3 to -6db. When I turn up to performance level at just below 12 o'clock on the Helix master volume, that puts the meters pretty much right at unity or 0db on the meters. When playing the patches I try to stand about 5 to 6 feet away from the speaker which gives me a pretty good sense of how they'll sound live out of the FOH speakers. On stage I have the same settings on the speakers as I do at the house, but I place the speaker on the floor behind me which allows everyone in the band to get a good mix from my speaker. None of our instruments come through the house monitors, only the voices. Because of the way I do this there's rarely any adjustment that needs to be made at the soundboard because all of my patches come into the channel pretty much right at unity with no need for EQ, which is where they should be with only slight bumps for leads. Hope that helps...

What you may be hearing is more based on where you are located relative to the speaker. FRFR cabinets are designed for long projections over distance. If you're standing too close you may be getting more of the sound from the horn than the blended sound of the horn and speaker.. Get further away until the horn and speaker blend better. It makes a significant difference. Also bear in mind you're listening for the sound as the audience hears it, which is different from the sound on the stage with an traditional cabinet. That in no way means it should sound bad, just a bit different. You might want to try combining cabs in parallel. For example, use a second cab split onto a second path with a ribbon mic placed on the outer edge of the cone about 1 or 2 inches away and mix the two to get the sound you want.

Not only that, but there are many, many cases where combinations of amps, cabs, mics and mic placement don't really require global cuts, or certainly don't need the same cuts as other combinations. If you're tweaking a preset, use your ears to get the best cut, if necessary, out of THAT setup. Otherwise you may be losing tone and articulation you might actually want.

Actually the Spider series was one of the first hardware based modelers I was exposed to at any great length. I was curious when the Spider V came out and went to the store to play with one. The modeling isn't horrible and in some cases it seemed better than the Spider IV. Granted I didn't have much time with it, but if I needed something around the house play and practice on, I'd probably go with it for convenience. And I have definately seen people use the larger systems for live shows. It's absolutely decent enough that most people are going to be happy with what they hear. As far as the distortion sounds it's pretty decent, but then I'm no real connoisseur of such things. I do my share of harder rock like Queen and AC/DC but I don't get into the Djent stuff. I will say, however, there are some key distinguishing factors when comparing the Spider V, or any of the amp based units to an upper end unit like the Helix. And that really has to do with your control over the tone. The Spider is pretty much a "packaged" system. You can put some pieces together like amps, cabs, effects...so forth, and do it pretty easily without having to have a great base of knowlege. You can kind of do the same with a Helix, but you have much more control and precision over the way you craft your sound. That's a double-edged sword by the way. Because in order to do that you need to have a fair amount of knowledge and understanding about amps and parameters and signal routing and live output devices to do that effectively and efficiently. One of the ways I tend to describe this is that the Helix is almost like a full recording studio environment in a box (minus the actual recording device). What I mean by this is that the sound you capture for a professional recording comes down to more than just the amp. It's often a combination of amp, amp parameters like eq, bias, or sag, speaker, type of mic, mic placement, and possibly combinations of multiple amps, speakers, mics, ambient room mics, etc. Not only this but also effects and where they're placed in the signal chain have impact on the sound. For example, a compressor has a very different effect on the sound depending on whether it's placed at the beginning or end of the signal chain. Likewise harshness in a given patch can be addressed in mulitple ways such as with a parametric EQ, or by simply running different cabinets and mic combinations in parallel and mixing the level of signal each one receives. All of those capabilities are what you have boxed up inside the Helix and available for you to use in building a sound. It's not necessary that you use all of these capabilities to get a decent sound, but the more you know and understand the more precisely you can achieve the exact sound you want. In my case the patches I create now look very little like the patches I was building initially on the Helix. I very often find myself re-tweeking older patches with things I've learned over the course of time as I prepare them for a performance. So really the big difference between the Spider and the Helix is the depth of capabilities you have in producing the tones you want. That means the question for you is, does this sound like something you want to invest your time in learning? I don't necessarily mean right now, but rather over time? In the same way that you grow and get better on guitar the more you practice, the better you get at crafting sounds on the Helix the more you delve into it. To me that has been the most exciting aspect of this product because it's something I can grow with. But I have the time to do such things, you may not...so that puts the ball in your court on this one.

First off...I think it's time to get past the juvenile nonsense of tone trivia and web reviews. There are plenty of folks out there that own or have owned Kemper, Axe and Helix and have opted to use the Helix. Generally speaking it has nothing to do with getting better tone out of one or another of the units. Most of that is trivial and has no effect on why they chose to use the Helix over the others. Those are all subjective opinions from people that may or may not actually have the qualifications to offer an opinion that's meaningful. After all, ANYONE can post a review of something on the internet regardless of their technical depth of knowledge, or lack thereof, Likewise, when and what did they review? At best you can only review a tiny slice of what these units do because they are constantly changing with new updates and/or new models. It's a black hole of internet nonsense that produces no viable results. All three units have the capability of delivering convincing results in any form of music with the right hands at the controls. So maybe you'd be better served moving beyond this and understanding what differentiates Helix from the rest of the market. Secondly, the reason said folks who have all three units have opted to use the Helix do so because of objective observations that make a difference in the real-life, day to day use of a unit. None of the other units come close to competing with the Helix as far as ease of use and flexibility of how it can integrate in different ways to different jobs whether that be in the recording studio, on stage, or even in the bedroom. Increasingly it's becoming painfully obvious to the competitors that the modular design of the underlying HX engine is providing Line 6 with a way to rapidly expand the capabilities both in quality and level of models and effects, but also to house all of those same capabilities in a wide array of physical forms be it the full Helix, Helix LT or Helix Native plug-in...all of them consistent and compatible...and all produced roughly within 2 years of the initial introduction of the unit...and all the while continuing to expand the capabilities of the core engine. Imagine where this Helix landscape will be in three or four years. So you see, you mentioned the term "professionals", but the things I mentioned above are the things real professionals concern themselves with. Not the tone comparison trivia game. That's for garage band players who don't see the big picture of having a performance environment that can be adapted to your needs in a convenient and serviceable way. I can develop a patch and record a song, then take that same patch and apply it to my live rig for rehearsal or performance. I can use the same unit to perform in a large venue or a more intimate environment with no changes except where I might plug in the unit and what kind of guitar and mic I might plug into it. Professionals want a system that can grow and adapt to their needs. That's far more important than the subjective characterizations and comparisons of tones in a single genre of music. Will Line 6 choose to offer a style of Helix with a built-in wireless connection someday? Possibly. But at least something like that is a real possibility and will be compatible with whatever work you put into some other form of Helix you used. Sorry if my response seems a bit terse, but I can't help but be bored to tears with the constant minutiae of high gain tone comparisons when the picture is so much broader than that.

Typically the "monitor" mode is a DSP function built into the speaker that flattens out the natural bump in the frequency profile of the speaker in the very lowest part of the frequency range typically between about 50 to 110hz. That bump is typically fine for bass in a full range FOH speaker, but not so good if it's on the floor as the bass gets exaggerated. On the Alto 210's it's marked as the "contour" button I believe.

Putting it into monitor mode will reduce or eliminate the bass coupling, but it may not completely take out some of the very low end (below 125hz) which can also cause some boominess. I know on my speaker (Yamaha DXR12) I place it in floor monitor mode to reduce the bass coupling and I also switch on the high pass filter at 125 hz. This should accurately reflect the sound from FOH as 125 hz is typically the crossover point from the sub to the mains. Just to be sure, in many cases if I'm using a very bass oriented patch I'll often set a 130hz low cut on my cabinets. None of that really affects my FOH because my channel already has 125hz filter on it, plus the crossover from the subs to the mains.

This doesn't negate the possibility that your outgoing signal is too hot. The "clipping" or digital artifacts may be the result of a signal chain with too much gain or output volume being processed for output. This has nothing to do with your master volume setting, but rather with the internal processing of the signal to prepare it for the output. Again, take a blank patch and play through it and see if you get the same artifacts.

What's not clear to me is whether or not this popping and crackling happens if even if you're not playing...in effect there's no signal going through the Helix, everything plugged in, volume turned up but not playing. I mention this because you said it almost sounds like clipping which would indicate it only happens when there is an active signal going through your rig. I'm also curious about your reference to the worship patch being "scripted" from Glen DeLaune. I'm assuming you mean you modified the patch in some way. However since you also mention it happens, but just not as much with other patches this leads me to think about the signal being processed out of the Helix and into your Deluxe Reverb or even into your headsets. In other words some aspect of gain staging of the signal that's off so that it's causing digital distortion. Digital distortion is different from analogue distortion in that it results in noise, not really a clipping sound like you would have in an amp. The fact that you're running into the preamp portion of your Fender Deluxe is also a question in this regard. Since the Deluxe Reverb has no external loop input you have no way of bypassing the preamp in the Deluxe Reverb and just getting a pure amplification of the signal. I guess the first thing I would check would be to take an empty patch and play through it and see if it causes the same effect. This would tell you that there's some aspect of gain staging of your signal in your patches might be causing this problem.

Given your two methods of listening live (headphones and studio monitors) I'm not sure you're getting an accurate reflection of what your patches will sound like in a live environment. The reason many of us use stage monitors like the TS210's or DXR10's is it's easier and more accurate to set up our patches for a live performance because those speakers will perform in a much more similar fashion to a PA than will a studio monitor or headphone. In effect, what we dial into our patches at home will be the sound we get at the performance. You have to realize studio monitors and headphones are designed with a different purpose in mind, which is really music reproduction typically in a room. In the case of your LSR305's those depend upon a bass port in the rear of the cabinet being positioned appropriately near a reflecting surface in order to reproduce adequate bass. A powered stage monitor, on the other hand, has no such dependencies. Unlike studio monitors stage monitors project sound solely to the front in a very confined pattern. This can make them sound somewhat harsh if you're standing too close to them, but are much better for achieving an accurate sound for developing your patch because the sound you dial in at home will be the sound you get in the live performance...no guesswork involved. That being said It's also important to note that working with live sound equipment requires a bit more knowledge than studio monitors or headphones, especially the newer generation of stage monitors. You have to account for things like bass coupling if they'll be used as floor monitors, and various ways of using low cut filters or contour settings to manage the frequency response curve. To some degree if you want to use the full range of capabilities of the Helix you have to accept a certain amount of responsibility for understanding live sound reinforcement techniques in order to get good results and to understand how to get good results. The alternative is to plug into a traditional amp and get a good sound for yourself, but take your chances on what it will sound like to the audience or how it will blend with the rest of the band on stage. Part of the problem, in my opinion, is that as simple as the Helix makes things for crafting a sound, the concepts of the Helix run very deep and require a certain amount of knowledge and skill to get what you want out of it. Blindly adjusting EQ, low cuts, or high cuts based on someone else's recommendations isn't a terribly prudent way of doing things. They may be using different guitars with different pickups or be playing a different style of music or are using IR's rather than stock cabinets, different mic models with different placements, and so forth. All of these things play into the way you dial in a patch. The approaches and techniques I use for a Chet Atkins patch where I'm finger picking my Gretsch hollow body is quite different from what I'll do for a Rolling Stones patch using a Telecaster. The low cuts and high cuts (if any) will be different, amps will be different, cabinets, mics, and mic placements will be different, EQ (if any) will be different, bias or sag adjustments may be necessary....and all of it will be based on what I'm hearing not what I've been told as far as best settings from someone else. That's the best way to get the tone YOU want given YOUR situation. And all of that takes some decent equipment, knowledge and some time to master.

As I read the known problem notes it seems like this only affects you if you have sound going through the unit when you adjust the master volume. The volume is still correct, it just doesn't move smoothly if sound is going through it. I'm hoping this is the case since I rarely play while setting the master volume..

And this is why we have Google....to search for those terms.... ;)

It's been the stated policy from Line 6 that they only republish the manual on major updates. For example, they updated the manual when FW 2.0 was issued. Generally that's when major functional features have been added such as was the case in 2.0 with the introduction of Snapshots. I'm generally okay with that although it would be nice if they split out a couple of sections like amps and effects that they updated with minor additions and changes such as this one. But honestly, how much information do they actually provide in the manual regarding how to operate the amps and effects anyway? Generally we've all been on our own to experiment and figure those things out.

First, I would say that you're definitely on the right track using your ears rather than just using numbers to adjust hi/lo cuts. So much of this is dependent on so many factors that you really can't give a hard and fast set of rules, but let me outline at least how I approach this. A couple of corrections about your assumptions. A FRFR speaker generally has all the same frequency range as a normal guitar speaker. It's just that a guitar speaker doesn't produce all of those frequencies at the same response level as does a FRFR speaker. that's what the "flat response" part of FRFR refers to. A Les Paul may have a different effect on highs and lows than a strat or tele, but it still may need adjustment depending on the amp, cab, mic, and mic placement. You'll be better served to think of the adjustments you're making NOT as trying to emulate the sound of a guitar amp, but rather crafting the tone for a studio recording. For example, if you were to target an Eric Clapton tone, think about the sound he gets on his recording rather than the live sound from an amp. 1. Any given patch I develop is going to be oriented toward a specific guitar. The guitars I use are a Les Paul Standard, a Strat, a Tele, and a Gretsch hollow body. The Strat and Tele tend to need more adjustment on highs than do the Les Paul and Gretsch. But that's all dependent on the amp and IRs. 2. Hi/lo cuts will be dependent upon the guitar being used, the IR's being used and the style/sound I'm shooting for in the particular patch. 3. I start by setting up the amp and IR(s) and adjust the EQ on the amp for main tone I want. It's not unusual that simply choosing the right IR will alleviate the harshness or boominess. Sometimes I can alleviate the high end harshness by simply combining a second IR in parallel and adjust the amount of signal between the two IRs. Once I get the sound close to what I want I then adjust the hi and lo cuts on the IR(s) if necessary to address any harshness or boominess. 4. I try to not use the amp tone controls to address any harshness/boominess. Those are simply to get my tone where I need it. Hi/lo cuts are the final tweak, although there is often some give and take in this area to get what I want. A note of caution here, it's very easy to fall into the trap of having too much bass in your patch. It may sound fine by itself, but it gets very muddled in the context of the band once you add in the bass guitar and drums. 5. IR's from different vendors can have very different effects on harshness and boominess depending on how much coloration they add to the tone. Ownhammer tends to add the most coloration, followed by Redwirez, and the Celestions add the least. But I prefer to try parallel cabinets first to address harshness/boominess before I add hi/lo cuts to the cabinets. In some cases the hi/lo cuts may only be necessary on one of the cabinets in a parallel setup. A couple of other observations. Redwirez tend to be better at controlling the harsh highs but you have to be careful about combining IRs from different manufacturers as you can sometimes get phasing issues between the IR's that can really mess up the articulation of the lows. 6. On hi cuts I generally start at 3.8khz then back it off until the harshness begins to come back. How high that goes is really dependent upon how everything else in the patch is setup and can often go as high as is necessary, but it's all based on what I hear. On low cuts I have my high pass filter on my monitor set to 120hz to match the crossover between the sub and the mains on the PA system. Therefore low cuts (if needed) will be something above that. So I start at 120hz and work my way up. By and large low cuts are not as problematic as hi cuts with the right IRs. 7. I only use EQ blocks, if necessary, to tame certain specific frequencies in the normal tonal range of the patch and I generally place them toward the end of my signal chain. I don't use them to address hi/lo cuts. I leave that to the hi/lo cuts on the cabs. 8. As far as reverbs I adjust those cuts as necessary by ear AFTER the basic tone tone of the patch is correct. These are typically the last effect I add. 9. Don't be deceived into thinking that turning up the -18db sounds better. Your ears will automatically prefer louder sounds. That's why when you compare speakers in a store you'll automatically prefer the louder speaker, even though it may not always the best speaker in terms of tone. I only use the loudness adjustment on the IR to level the overall volume between my patches so that I don't affect the tone. At any rate, this is my general process which I've come to after much tinkering with the Helix over the last year and a half. It seems to work pretty consistently for me and I think it pretty minimalistic as far as processing of the tone. In quite a number of cases I've had to make no adjustments to hi and low cuts once I setup my IRs correctly. Hope that helps.

I don't think anyone could say it better. For many of us, this wasn't a huge leap because we came at it after several years of dealing with different modeling technologies from modeling amps, to lower end modeling boards like the Line 6 HD500X, or others. And to be honest it took all of us a period of time to really grasp the real concept of modeling and how to make it work for us effectively. So in our cases, particularly mine, I knew from day one of buying the Helix how I intended to configure it into my rig, and what equipment I would use and need (studio monitors, stage monitors, headphones, etc.) For example, when I bought my Helix I already owned a Yamaha DXR12 stage monitor which I was using with my HD500X. So there was no guesswork at all about how I would hook it up and what needed to be done with tailoring it to my needs. So just be patient. It's a lot to take in, but once you get the grasp of looking at your Helix as if it's a traveling recording studio in a box that can also be used for live performances, things will start to make sense.

The file based storage is JSON (which is a variant of XML). But the internal storage of the data within the Helix may or may not be that. I would think for performance reasons it's likely stored in the helix unit in a more efficient data structure that doesn't need to be translated and keeps the overhead down for switching and loading presets in real time. But that's just a guess.

I don't disagree with any of what you're saying here, but some of that's really all part and parcel of the limitations relating to the way the current editor only loads one patch at a time, and that patches and IR's operate in two separate worlds that depend on manual integration. And maybe we're referring to the same thing given your term "Integrated Solution", or better yet a comprehensive integrated solution. But I think that's a pretty significant leap from where we're at right now with the editor. I wouldn't mind just a couple of baby steps toward that vision which could work on an interim basis with an eye toward a more comprehensive solution, such as a JSON formatted IR setlist (perhaps with longer names and maybe some additional descriptor fields) that retains the IR slot position and name (and whatever other descriptive data) that can be loaded and saved in one operation. Otherwise we might be waiting quite a long time.

By the way...this probably all goes away once you go direct into the board.

This has been talked about several times and I know there have been several IdeaScale entries on it. Typically this subject comes up in response to a FW upgrade when people need to save and restore their patches and IR's. It's seems to be a bit less of an issue since a lot of people have settled on the informal convention of renaming their IR's starting with a 3 digit number that designates which IR slot it should go into so that presets get linked back up correctly to their patches. I personally don't find the IR arrangement all that cumbersome. It's very simple and easy to understand and relatively easy to maintain once you adopt the convention of including the slot number in the IR name when it's backed up. It might be easier and less error prone if the editor would simply allow you to save your IR's in the same way as you save a setlist, so that it's one JSON file which would include the slot number for each preset. That one feature alone would provide a simple way of addressing some bulk maintenance issues like determining which IR's are not referenced by any patches in a given setlist, or which patches use a given IR. Since the Editor application only REALLY loads the data from the Helix one patch at a time for editing, and given the lag time it takes for the editor to load each patch, bulk operations across all of the patches in a setlist might better be served as JSON file operations until the day the Editor application is re-engineered to load the data from all the patches in the current setlist in memory all at one time. But I suspect we're quite a few updates away from something like that...but you never know....

The problem here is you're making an assumption that all your patches are loaded in the Helix, but that's not always true. I keep most of my presets saved on disk and load them as needed. A more likely solution would be to have an external facility in the editor that could go through all your saved presets to show you what presets use that specific IR number. You'd still have to determine on your own whether those are all the same IR's as it's entirely possible to have two patches pointing to the same IR number but were built with different actual IR's when they were saved. No matter how you slice it, it's going to be a tedious and nasty process.

Have you viewed any of the introductory videos from Line 6? They're pretty good at introducing you to the key concepts and how things work. You can also download the manual in PDF format from the line 6 support site. Beyond that just search for "Line 6 Helix" on YouTube and you'll find TONS of videos to keep you busy. As far as IR's, that's a concept that was never present in the X3 and it's simply a different type of cabinet that is constructed by any number of companies and individuals that you can import and use in your Helix patches. You have the stock cabinets that come with the Helix, and you can get IR's from all sorts of outside sources. The difference with IR's is that they're basically a snapshot or model of a specific real-world cabinet and mic response characteristics...so basically a cabinet and mic model. The difference being that the each IR file represents a specific cabinet, microphone, and microphone placement on that cabinet, so the sound can vary quite a bit even on the same cabinet. They're very flexible and you can combine them to really craft the sound of a given amp. It's really basically the same process that's been used by manually by sound engineers on real cabinets in both studios and live performance for producing a specific, polished sound. Most people start out just using the stock cabinets and mics supplied by Line 6 because they're very easy to use and adjust. Once you're comfortable with them you might consider looking into IR's as a way to better refine your sound.

Excellent work tracking all of these down. Very interesting comparisons, although mostly pretty similar. Interesting, however, the Atomic Amps has an on to off axis chart, but that makes sense given it's a more traditional cabinet.