DunedinDragon

Have you checked to make sure you're on the appropriate version level on the HX Edit? I'm not sure what Freman recommends for version levels, but you may need to check with them. Scrolling should work fine.

Chances are your audience isn't going to notice the difference whether you play in stereo or not. But if you have the speakers on the floor behind you I'm not sure you'll even be able to hear the stereo effect as well as you would were the two speakers mounted vertically on poles behind you and turned at slight angles. I personally don't play in stereo because very few people in a live environment benefit from it given they aren't positioned optimally to hear the separation, and the acoustics of the venue typically jumbles up the stereo effect within just a few feet given the reflections of sound in the room. I have run my vocals through the Helix, but the biggest issue for me was in sending my vocals and my guitar to the mixing board as separate lines so they could be managed individually. If you aren't sending your guitar to the mixing board it would be easier. But it would limit your guitar to using a single DSP on one signal chain so your other signal chain could be dedicated to the vocals. Whether the drummer hears you is based the same as it is with a traditional amp. It depends on the positioning. This is one of the reasons I placed my DXR12 on a half height pole behind me and turned it slightly toward the drummer and the rest of the group so I could take advantage of the wider dispersion pattern. The primary issue with placing any FRFR speaker on the floor is you're prone to bass coupling which accentuates bass frequencies. On many higher end speakers you have specific DSP contouring options to correct for this. There is a single contour button on the Headrush that simply states it "corrects for excessive bass on some stages". I'm not sure if that refers to bass coupling as it's not really a factor of the stage as much as it is how the speaker is positioned on the stage.

Exactly...!! People have to remember the reason for snapshots in the first place is to avoid the overhead that comes with unloading and re-loading blocks when switching presets which takes a moment to complete. So the idea is to be able to only endure that loading process once and be able to have a variety of configurations within that preset with no overhead delay. Once you begin modifying that model and start allowing loading and unloading of blocks within a preset you're right back to introducing the delays involved with switching presets which makes nobody happy.

I think you may be confused about what snapshots actually are given your questions. Snapshots are not a replacement for presets, or some form of "super" preset. It's simply a way of managing a preset so you can easily get different views or pictures of that preset. The preset is still there as the foundation and all the rules for presets still apply. If you can't change a block from one thing to another thing in a preset without editing it, you can't do it in a snapshot either.

Here's the one that's been around quite a while now and generally gets updated after each FW update: https://helixhelp.com/

I appreciate the detailed post because it makes it much easier to respond to you. You have no idea how rare that is around here... I exclusively use a FRFR setup using a Yamaha DXR12 as my speaker. I'm in an 8 piece band and we have a similar setup to yours in terms of a normal backline situation. Although we commonly go direct into the mixing board with our instruments, there are several cases in which we only send the vocals, harmonica, and congas through the PA and depend on the backline stage sound for everything else in smaller venues. My setup is the same either way in that I have my DXR12 on a half height speaker pole positioned behind me. There's a lot of good reasons for doing it this way. First, these monitors are designed to have a wide horizontal spread of sound and a much more limited vertical spread of sound. If positioned in a wedge position either in front of you or behind you, your bandmates far to your left or right won't get much of the benefit due to what would normally be the limited vertical spread. However, positioned upright on a pole it has exactly the same effect as a normal PA speaker as that what it is. So in a smaller venue it simply augments the FOH speakers in that it has the same kind of long distance throw as a normal FOH speaker. It makes for the best of all worlds for both you, the audience, and your bandmates. As far as going with a full FRFR setup, you'll likely need to make some adjustments to what you're used to with a conventional cabinet. I've been doing this type of setup for so long (about 6 years now) that I get a bit frustrated when I use a cabinet because I'm so used to the clarity and definition of the FRFR speaker cabinets seem muffled to me. I'm a bit concerned about your choice of speaker quite frankly though. In the powered speaker market you pretty much get what you pay for, and the Headrush is more or less the lower end of the market. You may be fine with it, and many people do use it, but it all depends on the style of music you play and how critical you are about exacting tones that you dial in. We play a very wide range of genres including rock, blues, jazz, rockabilly, R&B/Funk, and I use a variety of guitars, so I'm pretty much OCD when it comes to getting the tone right for a given song. If your band is more specific in genre it may not be that big a deal to you. The best way I found to adapt to doing FRFR live is to think of it like I'm dialing in my sound for a studio session and what I'm trying to achieve is a studio sound live. If you think of it that way you won't be thrown off by the whole "amp in the room" thing that bothers a lot of people, because there is no "amp in the room" sound in studio recordings. This also means you'll need to be much more attentive to the things that are relevant to studio production such as compression, cabinets/IRs, mic usage, mic positioning, and EQ to get the finished, polished sound you would normally want in a studio. You'll hear a lot about using high and low cuts due to the expanded responsiveness of a FRFR speaker, which is true to a certain degree, but not always a given depending on the amp, the cabinet, mic and mic positioning. In effect you need to develop your ability to hear what frequencies are problematic and the different ways you can go about correcting them. The biggest transition is in dialing in your tone. It's important to be as near as you can to performance volume levels in order to not be fooled by the Fletcher-Munson effect. And you definitely want to dial in your tones through the FRFR. The main thing to remember is FRFR speakers are specifically designed for long projection of sound and will be much more harsh close up to the speaker. This is the reason you don't seat people right in front of a PA speaker, because it takes a certain amount of distance to resolve the mix between the horn and speaker. I normally stand 6 feet or more away from the speaker and often off axis to check my tone when I'm dialing it in. My biggest issue is often getting too aggressive with high cuts. What sometimes sounds right in isolation is far too muffled when combined with the rest of the band. Hopefully that helps you or gives you some ideas about how to approach this.

The easiest and most bulletproof way to do this is to download and install the most recent version of HXEdit which will not only install HXEdit but also the most current version of the Line 6 updater and the appropriate device drivers.

It's probably easiest and most accurate to think about "flat response" as the lack of coloration a speaker adds to the sound. In reality no speaker is totally flat as that would actually sound pretty unnatural, but things like studio monitors or live speakers tend to be admired for how accurately and evenly they can reproduce a signal across the entire frequency range of human hearing.

Acoustic amps are an entirely different animal than typical powered speakers. They may tend to be more full range and less flat response and are tuned specifically for acoustic instruments often times with features specific for that such as anti-feedback adjustments. They also tend to be lower powered than your typical powered speaker just due to the nature of amplifying acoustic instruments on stage. Another key difference is that most powered speakers in this category are bi-amped using one amp to drive the lower frequency ranges through one speaker and the other to drive the compression driver for the high end speaker. Acoustic amps tend to be simple crossover designs from a single amplifier circuit. Of course using the FX return bypasses the tone circuitry, but what comes into play in most of the powered speakers are DSP contouring features that help maintain the flat response which is necessary to support ALL types of inputs whether it be instruments, voices or percussion. This is why many of us choose to use these type of powered speakers because it matches up better with the FOH speakers in terms of response so we feel more confident that the sound we've dialed in and hear on stage will be a closer match to what the audience is hearing. I truly doubt that sort of thing is designed into acoustic amps since they are more single purposed in their design. Generally many of us that use powered speakers like these go direct from the output of our Helix (such as an XLR) to the board and go direct into the speaker with different output such as the 1/4" out. I'm not sure you're going to get the same effect going that way on an acoustic amp, but the only way to know is to try it.

Yeah, Frank posted it on the Facebook group to wide acclaim. I really appreciate your efforts on this. I've used it on both my Helix units and had my IRs cleaned up in no time. One question did come up in the Facebook group though as far as whether or not this was dependent upon your setlist export files coming from a certain level of firmware. I'm pretty sure they changed the layout of the export files some time back, but I thought I'd ask you here if you know of any limits in that regard.

That sounds exactly like a situation that occurs when the HX Edit version and the Firmware versions don't match. Are you sure you updated BOTH the HX Edit and the Firmware?

Did you try the CelestionPlus V30 Creamback IRs? That's the most accurate ones I've heard.

Thanks so much. It's nice to know there's someone out there keeping all this updated for everyone's benefit. Much appreciated!!!!!

Another thing to check is if you have a delay effect in your patch you might want to check the Headroom parameter depending on the type of delay it is. Simple delays and a few others don't have that parameter, but it's a pretty common place where, if not increased, will cause clipping.

I think you need to be a bit more specific here to determine if it's in the headphones or Helix. What level do you have your headphone out volume set at on the Helix? Have you tried it through studio monitors or some other output? Is it only when you listen to the recorded track or is it there when you simply play through the headphones? Does it happen on different patches? You need to tell us a bit more about what steps you've gone through to troubleshoot it.

Well...that depends on if you buy into the "tube voodoo" associated with tube amps. The only thing that really matters is what sounds best to your ears. Whichever way you go you're going to have to account for the fact that the patch will ultimately be going to the PA if it's a larger venue. If you dial in your tone on something like a Spider Valve, it may not sound the way you expect once it goes through the PA.

Just a stupid thing that's caught me a couple of times. Have you checked that your guitar is plugged into the guitar input on the Helix and not inadvertantly into the Aux In on the Helix? I can't say it sounds muffled when you do that as much as just kind of dead.

I can't say I've ever had to do anything to adjust my presets to an outdoor event. The only thing that could come into play theoretically is if your presets are designed to exploit the natural reflective ambient sound of an indoor venue in the place of reverb, but that's not a worry for me. As long as you've normalized your preset/snapshot volumes equally and you get a good soundcheck and gain staging from the mixer it shouldn't be of any concern as far as your guitar channel. Vocals or mic'd instruments like harmonica or conga drums would be a bigger issue without a dedicated sound man.

When it comes to vocals that's probably more true than it is with instruments. But it's certainly not unknowable by ear in either case if you do it enough. I do think until you can develop an ear to hear it you probably need a signal meter so you can identify when it's beginning to kick in and when it's beginning to squish to help train your ear to hear it. With instruments however compressors tend to be used more as an effect than a fix to problems of inconsistencies in the volume of a singing voice, so it makes it easier to hear because that's a particular sound you wanting. Kind of like that "chunky" sound of Pete Townshend or Keith Richards guitars.

There may be several factors operating here. First, you mentioned getting rid of the speaker and mic, but you used a D130 cabinet IR. You do realize these IR's are created with mics, right? So the mic used and it's position on the cabinet when the IR is captured will make a difference. These should all be documented on your IR and the mic and it's position may be feeding into some of the brightness you're hearing. As AMSDENJ mentioned you may want to use choose an IR that uses a ribbon mic such as a Royer 121 and it's location either off-axis (if available with your IR collection) or further out from cap toward the cone edge. My preferred setup tends to be mixes that use Royer 121 and an MD421, but that's a subjective thing that may be different for you. Another thing that can happen is the quality of the IEMs you're using as well as the powered speaker. All of these things feed into the end product of the sound. You didn't specify which JBL EON you were using. Some of the older one's aren't terribly accurate like the newer JBL EON612 which will have some DSP contour settings which you need to match up to how the speaker is being used (floor monitor, on stand, etc.). In any case in my experience there are going to be differences between in ears and powered speakers simply due to their design and what they are designed to do. Powered speakers are specifically designed for projection over long distances so will sound much more harsh up close than when you get 6 feet or more away from them which allows the speaker and the high frequency driver to blend better. All in ears are going to sound more like studio speakers than live sound speakers so they will be naturally a bit more mellow. But I always use the same patches for either and they work fine, but it depends on understanding how to work with modern full range live speakers and the IRs you use with them. When working with modern live full range live speakers many of us employ high cuts and low cuts in our signal paths to take some of the harsh edges and low boominess off of some patches depending on what amp, what speaker, what mic and what mic positions and of course the style of music. I tend to try to adjust these things best I can with the right mic, mic placement or mic mixes if provided by the vendor of the IR before jumping too quickly to using high and low cuts. Remembering, of course that powered speakers in a smaller room and in closer proximity will tend to sound harsher and more bassy sometimes than what they will sound like going through an actual FOH PA setup where the audience will naturally be further away. I use a Yamaha DXR12 due to it's accuracy and consistency with most modern FOH systems in my experience and generally listen to my tones with the speaker at about chest height vertically about 6 foot away both facing the speaker, off axis from the speaker at an oblique angle, and with my back to the speaker to get a good read on it. Even then I accept a bit more harshness because my experience is it will sound about right once I put it on stage in a real environment. If I take too much of that harshness out it will sound somewhat dull and lifeless when I get to a live setting. If you're positioning the speaker as a floor monitor you definitely need to use the DSP contouring option on your speaker (if you have it) to correct for bass coupling with the floor which will make your preset sound more boomy. None of these things will generally have an effect on your in ears like they do with the live speaker. Hopefully that gives you some ideas of how to best dial in your patches to get what you want. Again, in ears and powered live speakers will never sound exactly the same, but the difference isn't all that drastic. About the same as the difference between a live powered speaker and powered studio monitors.

They could call that the cork sniffer tone variation switch!!!!!

I guess I'm not as much of a "cork sniffer" as some when it comes to every nuance of my tone, so I can be equally happy using either and get what I want out of either of them. What it boils down to for me is simple workflow. Now that I've identified the cabinets, mic placement and mixes that tend to work best for me, IRs are a much quicker tool to just drop in and move on when building a preset. Kind of like shorthand for cabinets.

For a supposed "grown up" MarkJarvis, we're not much impressed either....

It does you no good to write it up here. Line 6 really doesn't monitor this forum. But you can add your comments to IdeaScale and see how much user support you get for it. Your situation is so one-off I'm not sure how much user support it might draw which is the defining factor for the things Line 6 is going to focus on ideally.

Good luck with that one given the Helix is a USB 2.0 Type B non host device and requires a specific device driver.