DunedinDragon

There is no source code available nor documentation for the protocol used by HX Edit to write your own version of HX Edit on another platform. I'm not even sure there's any type of reactive monitoring capabilities in the HX Edit protocol as it's more of a simple terminal program and that's why the signal meter could only be implemented in the Helix unit itself and not in HX Edit.

Pretty much everything I mentioned applies to the kind of places I normally play which really aren't that big...maybe 150 - 200 max occupancy. But I have to admit I've yet to worry about doing an RTA on the rooms even though the capability is already built into my QSC TM30 mixer with the exception of a good area mic. I suppose if I were walking into places that were horrible acoustic nightmares I might think differently about it. It's just such a hassle to go through the process with people hanging around while you're loading in and setting up. Especially if I'm running into feedback issues and I need to run the feedback notching wizard. That's already invasive enough and always draws complaints from people in the venue. Plus, by the time we get everything setup all I want to do is a quick soundcheck, gain stage and take a break before we have to start playing. I've been doing this long enough in the same rooms that I pretty much know all my PA and monitor settings by heart.

In theory that might work. As a practical matter that works fine in a recording scenarios but for live sound it really comes down to the differences in how PA speakers are designed to work and how studio monitors are designed to work, and the difference really is significant. Studio monitors allow for very little latitude as to where someone is positioned relative to the speaker, in fact they tend to give pretty specific guidelines about speaker placement and listener's position relative to the speakers, but that's a HUGE consideration when it comes to PA speaker design because speakers need to be positioned in various ways to cover the area as completely as possible or tuning the speaker based on how it will be used (front speakers or stage monitors). Because of that they have very wide horizontal coverage areas and very limited vertical coverage in order to conserve sound energy in a big area from being lost in the floor or ceiling as well as specific tunings that can be applied to the speakers to compensate for differences. But in all cases the sound stays pretty consistent within those limits but can sound overlly harsh when sitting too close. Volume plays a factor with PA speakers in that PA speakers need a bit of space to blend the highs and lows as well as blend appropriately with the subs. That's why no one in their right mind would want to stand next to a typical PA speaker or place a sub in such a way that it causes problems with the mains. Aside from those differences, you can certainly run into acoustical differences in a live show, but there's a process built into most reputable PA mixers called RTA or Real Time Analysis that is broadly used to make global adjustments across all channels before the final output to correct for serious deficiencies. That's why I'm a big believer in making sure you know what you're audience will be hearing by using powered PA speakers positioned vertically at chest height at a reasonably higher volume level (around 90 or 95dbSPL for dialing in live tones if you want to really get a feel for what people will be hearing.

This is again comes down to the same old problem of thinking that "flat response" is somehow a standard that crosses the boundaries between all applications, which couldn't be further from the truth. I myself use a pair of HS7 speakers in my studio, but I NEVER dial in my live performance tones using them. That's because they're made for a different purpose than a live performance modern powered speaker such as a QSC K.2 series or a Yamaha DXR series which is what I use for dialing in live sounds. And they're different because they're designed to be different because they serve two different situations. Studio speakers are designed for evaluation of a sound or mix and require (by design) a very precise listening environment to be effective in which the speakers and the listener are positioned correctly in an equilateral triangle and the speakers themselves are positioned appropriately relative to surrounding walls as they often depend on bass reflex actions to provide an accurate representation of low end frequencies. Live sound is concerned with covering a wide and long space with a consistent sound regardless of where the listener is positioned. Very low end frequencies are boosted through the use of subwoofers (which all operate well below the frequency levels your typical guitar produces). What this means in a practical sense is you shouldn't be surprised if you dial in your tone on studio monitors and then plug into a PA that's using modern PA powered speakers. There are differences because there are SUPPOSED to be differences. If you really want to evaluate the way your tones will sound in a live environment the quickest and easiest way is to plug your Helix direct into a modern live powered speaker meant for that purpose and listen to it in the way that people in the audience will be listening to it. In effect, at longer distances and off-axis around the room and at a reasonably higher volume levels probably around 85 to 90 dbSPL. THAT, or something very near to that, is what your audience will hear in your live performance. And that, oh by the way, may be different from your stage sound depending on what your live setup consists of. That's one key reason people choose to go direct to the mixing board and monitor either through high quality powered monitors or in ears. It reduces the complexity and increases the consistency of your sound both on and off stage.

Of all the things you said, THIS is the most key statement and was an early insight for me when I first began modeling. I believe I characterized it as a modeler is really just a recording studio in a box, but without the ability to record, and that really got me past the idea that a single preset was going to be sufficient for every possible use case. This is one of the main reasons one of the first things I'll ask someone when they're having a problem dialing in their Helix is "what exactly are you trying to use it for?" because that truly determines the best fit for what you might need to use to dial it in. It's just not a one-size-fits-all world when it comes to modeling. Even when it comes to live productions no single output methodology will work best for all. That's the real beauty of the Helix design. They took all of that into account allowing the user to decide about what output configuration will work best for them. I've always been a huge fan of using full range powered speakers in my stage productions. But that's because I cover a wide range of musical styles and rarely perform live in a situation where all instruments and voices don't go through a decent/modern PA setup. But my most recent experiences tell a real tale of the challenges of crossing that line between different applications of modeling in the studio and in a live presentation. Over the last couple of months I've begun using a hybrid configuration of components in my band's live performances. We have live instruments which we use to play in a traditional way through a PA, but I've incorporated a process using Ableton Live which allows me to integrate recorded instrument tracks into our live productions in order to broaden our ability to address different types and styles of songs in a more authentic manner. This isn't something new as many live big name performers have been doing this for some time, but it is something new for playing at local club and festival levels. In my home I have both a recording studio setup and a live rehearsal setup. I have to build these instrument tracks in my studio and then bring them over into my live production setup. This means I develop the studio tracks using keyboard/MIDI controllers, instrument sample libraries and studio monitors and that resulting track is then played (along with live instruments) through a live production PA setup. One of the things I've had to take into account in doing this is the difference in how the different systems present the sounds. What sounds great and authentic in the studio such as a Hammond B3 with a Leslie can mix very poorly in a live production using powered full range speakers. Due the the nature of how the system works live, it's not reasonable to make changes to the mixing board between different tracks being used that contain different instruments. Every track has to be contoured with how it will mix and present itself in a live environment because the two environments are dramatically different in how they're SUPPOSED to sound to an audience. Very often I have to iterate between testing the backing track in the live system and going back to the studio to make corrections to adjust the sound of the backing track to better integrate seamlessly and correctly into the live production. My problem in this regard is no different than anyone that tries to use headphones or studio monitors to create a sound intended to be used live...and then are surprised when their sound on the live system doesn't match what they dialed in on the studio setup. That's not the Helix fault because the Helix can do either. It's the simple fact that the two environments are DRAMATICALLY different and have to be accounted for.

This is what my live test setup at home consists of. Basically it's a live system setup that allows me to dial in my presets while gain staging them through the mixer and evaluating the tone of both the main front speakers and/or the floor monitors. What's not pictured here is the Laptop I use for managing the Helix. I don't use all of this all the time and my comon configuration is a bit different with just the Helix, mixer, and DXR12 speaker. If you're using powered speakers such as the ones I use it's very important that you give them plenty of space in order to hear them in the same way they'll be heard in a live environment along with other instruments. I usually stand about 6 to 8 feet away from the speakers when I'm evaluating the tone. As I've mentioned previously I use a Parametric EQ as one of the last blocks in my chain, and it's typically used for high and low cuts and any tweaks in any other frequency problems. I make extensive use of IRs, but that's mostly for convenience sake as I can get the same tone out of stock cabs, but it just takes longer to set them up. I have a set of maybe 50 or so previously auditioned and selected IRs that I work with depending on the type of tone I'm going for on a given preset. My most common cab/IR setups use a combination of a dynamic mic (MD421) and a ribbon mic (R121) My high cuts are typically in the 8khz to 10khz and low cuts range from maybe 80 Hz up to no more than 140 Hz. All of those vary based on the type of guitar I'm using on the preset and the style of the song. I would personally caution against high cuts as low as you're talking about as you'll lose a LOT of articulation and clarity and probably end up being buried amongst the other instruments you're playing with. If you're using the right cabinet and mic configuration you shouldn't need those type of dramatic cuts in either live or studio situations.

I have no doubt LUFS would be very useful to many people in getting the sound they want in the volume range they want pretty easily. But I'm not sure it would be of significant help to me since first and foremost I rarely dial in my presets in isolation. I spend most of my time dialing in presets and backing tracks while gain staging and mixing them simultaneously with a mixing board signal meter. This ensures the best mix in all shows and all songs, and I've done it long enough to recognize and correct the difference in tone within a patch or within a backing track without significantly affecting the signal meter. In other words, I know what I can get away with. However, had I discovered LUFS 5 or 6 years ago I would have had a much greater appreciation for it probably. Unfortunately I had to go through the pain and frustration of training myself to do it by ear and now I'm used to it and pretty proficient at it. So I'd definitely agree that if someone struggles with those types of differences, using LUFS would be an excellent training tool to help you identify and correct those type of problems quickly by ear.

What you're saying suggests strongly that your PA may be the problem. As I stated earlier I have no knowledge of that PA system but it appears to be a passive (non powered) speaker system. Most modern PA systems use powered speakers with digital signal processing (DSP) chips to provide a more flat frequency response. On most older PA systems that use passive speakers and an inboard amp or simple intermediate amp which depends on the simple crossover in the passive speakers, there is a very prominent dip in the mid range frequencies at the crossover point between highs and lows which results in the kind of behavior you're describing. What I'd suggest is to see if you can borrow or try out a decent powered speaker like a QSC K or CP series, plug the Helix direct into it and see if that fixes the problem. If it does then you probably have one of those older style PA setups and I've never found anything you can do to fix that problem sufficiently on the Helix or any other modern modelling system.

There are more ways to adjust these things than room in this forum to list them. In short as far as what the Helix controls, your amp EQ controls are your first level immediately followed by your choice in cabinets, mics and mic placements. Of course before that comes your settings on your guitar such as which pickups are being used and your volume and tone controls. 90% of your tone depends on those settings, and you have ample other EQ blocks to make any final tweaks. But if you don't have those first level EQ components correct, you can play all day with other EQ pieces and never get it right. I can only tell you that after setting up the guitar and amp/cab/ selection I'm not satisfied with the tone I'll first go back and audition different cab and mic setups till I find something I like better. From there it's just a matter of making some very minor tweaks with a parametric EQ at the end of the signal chain, or in some cases a low/high shelf block following the cabinet to adjust the levels between overall highs and lows. Most of this kind of stuff is covered very completely in Jason Sadites YouTube videos which I'd suggest you spend some time with to get a feel for how to set them up.

Generally speaking I don't tend to use much in the way of compressors if I'm dealing with a high gain tone. That IS a compressor. I almost always have a final LA Studio comp in my chain, but even then it's rarely used or not even present in some cases if the high gain tone is compressed enough as it is. For longer sustains in those cases I pretty much just depend on fingering techniques to maintain longer dwell times.

That really depends on what you mean by real life. Real life as far as live performance or real life as far as recording? In both cases it comes down to your output device, but the output devices are different. In live performance you need to simulate what your audience will be hearing and that can be somewhat complex depending on your live setup. In my case I go direct from the Helix to the mixing board so all I need is a quality, modern FRFR speaker mounted appropriately at chest height to get a good representation of what the audience will hear. That should work in almost any type of configuration as long as you can isolate that sound from what you're using on stage such as a powercab or more traditional amp/cab using 4cm. You're not going to like my next statement, but you really do need to use a higher quality FRFR speaker representative of the type that will be used on a live performance system. I mostly use a Yamaha DXR12 but I've also used a QSC K10 from time to time. In both cases I can set the DSP contouring on the speaker for how it's likely to be used in a live front of house speaker. That's the best and simplest arrangement to get consistent results. The studio is much simpler. I use my Helix floor as my audio interface and have my studio monitors attached directly to it for output. In addition I have a Beyer Dynamic DT770 set of headphones for when I'm doing vocal work. My studio monitors are Yamaha HS7's and are placed on my studio desktop in an equilateral triangle with my chair and positioned relative to the walls (back and sides) as recommended by the vendor. I realize this may not be as budget conscious as you'd like, but ultimately it's about getting a consistent and realistic representation of your guitar sound in the environment you're working in. You may be happy enough with a lower budget approach to speakers, but that's a judgement call on your part.

In my experience I find that relying totally on any type of meter can be deceiving and it ultimately comes down to the human response to how it sounds relative to the instruments around it. I work solely with standard signal meters which are a true, physical representation of signal strength. I use both the built in meter in the Helix (which measures the digital signal level) and the signal level at the mixing board channel (which measures the analog signal level). Both tend to be relatively consistent. But there are cases when working with more overdriven sounds where the signals levels are consistent with other clean levels, but it's clearly lacking overall body and punch compared to cleaner signals. In those cases it's often surprising how much you can increase the channel volume on the amp model to gain back that body and punch without significantly affecting the overall signal level as measured at the mixing board. A more telling example is when I work with backing tracks. Our band uses backing tracks housed on a laptop running Ableton Live that adds various instruments to different songs such as an organ, piano, strings, pedal steel, etc. I gain stage the outputs of these tracks against my guitar signal and drums (provided from a Beat Buddy) through the mixing board and I'm often surprised at how much lower the signal level as measured at the mixing board these backing tracks have to be in order to blend appropriately by ear with the other instruments. What this tells me is what everyone's been saying around here for a long time. Ultimately the best and final measure is always going to come down to using your ears.

That sounds quite normal to me. You start with a very high gain tone then switch to a clean tone. The actual volume might change slightly, but mostly it's changing the perceived volume more than the actual volume. Your high gain tone sustains longer and is highly compressed whereas the clean tone has little sustain or compression so it has higher peaks and fades quicker. But that's just a hearing test. My guess is if you were to watch it on a signal meter the two tones would be pretty consistent in their peak values.

There really shouldn't be any difference between the makeup of what's in the signal whether you use XLR or 1/4", although there could be signal level differences which might be significant depending on your situation. XLR offers some advantages over longer runs over simple 1/4" lines and it's certainly the preference of choice for connecting up PA mixers and components. I would certainly NEVER advocate using the headphone out as an output as that's not what they're designed for and run more highly amplified and specialized signals rather than the standard line,. mic or instrument level signals as the other outputs on the Helix. But certainly you have to recognize the difference in headphones has little to do with the connection and more to do with the design of the headphones themselves since all headphones use the same type of connectors and there are VAST differences in their sound and quality. I get it that you're simply trying to fix a very specific difference in the sound of the distortion between headphones and monitors, but I'd suggest spending your time looking at options within the signal chain rather than chasing different output methods. I have many presets that I've used both in the studio and for live performances, but there will be some differences in how those same presets are dialed in simply due to the differences in the way those speakers are designed to operate. The very nature of the design used on these different types of speakers will dictate much of that, particularly between studio monitors and live powered speakers. One key component that may help you get the type of gain sound you're looking for is the Retro Reel block in the Modulation section. That allows you to apply vintage tape artifacts to a preset which can help enrich and add more fullness to your signal and that may be the answer for what you're looking for. Try it both at the beginning of your signal chain or at the end depending on how much of it you want to apply.

I know very little about the Dynacord Powermax 5 system and there's not much technical information I could find on the internet, but it looks to be s a passive speaker setup using a specifically designed power amp system. If so a lot of your sound will be determined by what specific features that poweramp provides and how you have it configured. I can only say that using decent powered FRFR speakers such as the QSC K series or Yamaha DXR series I never have any high cuts below maybe 7.5 khz and usually much higher than that if not at all in some cases. Buth "harshness" can come in many forms. One of the most common forms of perceived harshness comes from listening to live speakers from a closer than normal position. PA speakers are designed to project long distances and standing too close doesn't allow the speakers enough space to blend correctly. Also "harshness" is a subjective term but most often seems to be more related to high mids (roughly 4khz to 8khz) than actual high frequencies which are more known for "fizz". That type of harshness is best addressed with narrow cuts at specific frequencies using a parametric EQ. The one most common in my experience is the narrow cut typically used for single coil guitars at around 4.2 khz, but there can be others depending on the speaker system you're using.

So ask yourself this simple question. What has changed when you move from headphones to a powered speaker? Certainly nothing has changed in the way the Helix is producing that tone because it has no idea how you're listening to it. The output device has certainly changed and the way it produces sound, but the other thing that's changed is your position relative to the output device and the room in which you're listening. With headphones your ears are positioned precisely in the center of the sound with no interaction from the room. That's the reason EVERYTHING sounds better on headphones typically because it's pure sound with nothing interfering with it, and that's never the case with any other type of output device. Play through a traditional amp and cabinet and stand behind it, or stand to the side of it, or position yourself dead center in front or the speaker cap and your sound will change dramatically. Why do you think studios spend so much money on sound baffling, and positioning monitors so precisely? Why are line arrays flown high above the audience in every concert venue? That's all related to the nature of sound reproduction. It's not an insurmountable problem as it's solved every single day by the VAST majority of people using the Helix. In my studio my monitors (Yamaha HS7 speakers) are positioned in a precise equilateral triangle from my listening position and are located relative to the side walls and the rear walls to get the best results from rear bass reflex ports as prescribed in their documentation. When using powered FRFR live speakers I have the correct DSP contour selected for how the speaker is positioned (which is not available on cheaper speakers like the Alto) and I stand a minimum of 5 or 6 feet away from the speaker as any audience member would or I would on stage using it as a floor monitor. Even then the sound will be different, but I expect it to be and can account for it in how I build my presets. This has nothing to do with the Helix and everything to do with you understanding what an output device is designed to do and how to optimize it and what to expect from it. As I mentioned to my friend who's a product manager at QSC the other day, there are way too many musicians who know nothing about what happens after their sound leaves the output connection of the device they're playing through other than some weird, unknowable black magic.

Probably similar if not a direct copy

I think it really depends on your speakers. In my studio I use Yamaha HS7 studio monitors which seems to be a pretty good match to what I hear with the DT770's...not enough difference to fixate over. I think in all audio output devices you tend to get what you pay for. I think the biggest leap in tone differences can come when you take the preset to a live environment because the speaker designs are so different than headphones or studio monitors. Again, I think the quality and consistency is part of what you pay for with any speakers including live powered speakers. There are differences in the sound from what I hear in the studio and what I hear on stage using my Yamaha DXR12 or my QSC K10.2, but I really don't worry about it because the overall general sound of a live performance is different just because of the requirement to fill a large space consistently and the audiences can't really tell the difference because they don't listen to studio monitors generally.

Here's what I would suggest for getting settled in with building your presets. Spend some time watching Jason Sadites "dialing in" series of videos on YouTube. He just recently re-did some of the originals to exploit some of the newer features in recent releases. The reason this is important is that Jason goes through a pretty full explanation of the different values you can use in the different blocks and their effect on the sound so you're not just turning dials and hoping for the best. His YouTube channel has a very wide variety of topics and I'm sure you can find some that will hit on topics to help you figure out the system. The fact is, everyone is different in what they're looking for in tone, and much of what people describe in their tone such as "chewy, saturated" tone might mean something very different to me than what it means to you. I will point out a couple of important take away impressions I got from reading your post. One thing is that the amp models are all going to be a little different in terms of how they handle the master volume based on the design of the original circuit. However as far as the channel volume they all should be the same in that the channel volume on the amps doesn't affect the tone. It only increases/decreases the volume. That is the primary thing I, and many others, adjust in my presets to get them gain staged to the same volume level. The amp master volume can have a significant influence on the tone by itself as well as when it interacts with other OD type pedals before it in the signal chain. Similar to you, I also use Beyer DT770 headphones when I use headphones. I don't necessarily get a lot of "ice picky" sound, but then I'm pretty moderate in my use of gain and OD on most of my presets because at a certain point you're going to get a lot more noise than definition because you've turned the gain into a limiter with very little dynamic response. You can mitigate a considerable amount of the "ice pick" sound in a couple of ways. The most important way is the use of different mics and position of mics on your cabinets. I tend to prefer a combination of mics such as one dynamic mic such as an MD421 which has a good, round (less harsh) midrange along with a ribbon mic such as a R121 for a more pleasant overall full range. You'll hear a lot of people advocate high cuts which I do use on many presets at the very end of my signal chain using a parametric EQ which has a faster slope than the cabinet high cuts. However, I agree with Jason Sadites that you don't really need drastic cuts if you've got a good mic'ing situation on most cabinets. My high cuts are seldom deeper than 8khz and more likely up in the 9khz or 10khz range and help retain the high end articulation for pick attack on strings. I'm convinced a lot of folks end up cutting down to the 5khz range because what they're hearing is not high frequency harshness, but high mid frequency harshness. That's especially true when using single coil guitars. In those cases I find I get better results by slicing out a deeper but narrow slice in the 4.2 khz range with the parametric EQ. You may also find you get some better results on some amps by simply adjusting the SAG and BIAS on the amp. The first thing I thought about when you mentioned the "chewy and saturated" feel is a new effect in the modulations area named the "Retro Reel". This simulates the natural vintage analog tape effect and might be worth looking at. You can place it before the amp or at the end of the signal chain with different effects, but you don't need much of it to get that lovely, rich saturation you might be looking for. Again, all of these things are covered very completely by Jason Sadites so I'd really encourage you to take a look at some of his videos as they've helped a LOT of Helix users get comfortable with building their own presets.

Well it really comes down to the cab block and what you're using to listen to your recorded signal and your positioning relative to the speakers. The ribbon mic is a pretty decent choice although I tend to prefer R121's. But I doubt that's the real difference. Putting some more early reflections on it would help give it a better "room" feel to compare to your live speakers. But I think one of the things you are most likely dealing with is trying to match your live cabinet to the modeled cabinet because the tone of the live cabinet will change depending on where you are positioned relative to the speaker. Get directly in front of the cap on that speaker and I'm pretty sure you're going to hear the same fizzy stuff or stand off axis to the speaker and it will likely sound darker or muddier. The good news is you can address the fizzy stuff in the digital signal which you can't on the live cabinet unless you mic it. I might first try using a combination of two mics like one dynamic mic such as an MD421 and your ribbon mic. The MD421 will add more mid presence articulation which might help better mask the fizzy stuff. The other thing that might be affecting it is your listening position relative to your studio monitors if you're using them. The fizziness typically doesn't carry too far when it's overwhelmed in the room when it combines with the rest of the sound, so if you're too close to the speakers or they're not properly positioned in an equilateral triangle to your position, the fizziness could be more prominent. And a LOT can depend on the type of speakers or headphones you're using for the recorded signal as well. Those are the things I'd probably start with.

The simplest and most appropriate approach if you want to manage the output volume going to the K10.2 without using the Helix volume knob would be to disengage the volume knob from the XLR out and to set it to Mic level impedence rather than Line level in Helix global ins/outs the same as you would for going to a mixer. You would also need to set Mic level impedance on the K.12 as well. I believe that can only be done on Chan A of the QSC.

Quite frankly I've never had much use for the Solo Lead OD as it's just too much overdrive and unless you doing something like Van Halen it's just overkill. But I do use the Clean often with pedals to get a bit more crunch to it and still retain the articulation, or even the Lead Crunch once or twice. However I've found better alternatives than the Soldano for higher gain sounds where I want to maintain better definition and tightness. The Archetype and the Placater come to mind in those cases. I don't personally remember it being particularly flubby on the low end, but then that's affected a LOT by the way you mic the cab as well as what you're using for your output. A lot of what you describe as problems may come down to those factors more than the amp itself. Although I do have to say I could never get a tone I liked on the Solo Lead OD.

Without knowing what trouble you're running into and what you're using for output it's hard to give you much in the way of specifics. But I do play a considerable amount of country both old and new and I always have the best luck with amps like the WhoWatt (southern/classic rock), Jazz Rivet for sparkly cleans and the Archetype clean. Any of those will give you a very usable clean tone you can build on using pedals if you need to add a bit of crunch. Cabinets tend to vary depending on the overall tone I want but I'd suggest using ribbon mics on your cabinets rather than dynamic mics like a SM57 to avoid the harsh high mids in the 4.2khz range. MD421 works well for a dynamic mic with a more mellow mid. I always run direct to the PA with no amp on stage, just a good floor monitor (QSC K10.2). A good mild compressor at the end of you signal chain like the LA Studio Compressor can be your best friend for smoothing things out, and the Retro Reel in the modulators section placed toward the beginning of your signal chain can add some nice authenticity to your southern/classic rock tones).

Pssst...my dirty little secret is I pre-record my pedal steel and keyboard parts and use Ableton Live to provide them as tracks rather than cart that stuff to gigs.

Well, it's not a function of whether it's a hard thing to implement or not. It's a function of how many people would want the functionality and how it might drive influence or sales of units. One look through Ideascale and you'll see all sorts of things in there with very high approval numbers. That's your competition for getting the feature you want implemented. If your needs are somewhat particular or specialized to your situation you may not find a lot of people that crave to have a feature like that on their Helix. Even though I'm a multi-instrumentalist (electric gtr, acoustic, banjo, mandolin, resonator) I've never found this to be a problem. I simply move my wireless transmitter which just unplugs and plugs in if I need to tune something else and I tune everything before before the gig and seldom need to do much afterwards, so it's not a feature I'd be needing so much.