DunedinDragon

Yeah...I think it's called "not so global settings"........lol

The SLO has been one of my goto amp models ever since I got my Helix over 7 years ago. But then I just evaluated it on it's own merits not compared to something I may have used in the past on other modelers. Everything it does, it does quite well IMO.

Since my both of my Helix units are used in production live or studio environments I'd prefer a longer wait with a better chance of stability. For me updates are a MAJOR pain because the thought of rolling back an update is a nightmare for me. That's why I'm usually content to wait a couple of months once it's released before I wade into the waters of an update.

The simplest way would be to define the mixer MIDI action to a separate stomp button rather than a snapshot and control it individually.

I've been through a number of approaches over the last 7 years to get my direct to PA tone consistent and gain staged appropriately, but nothing like what you went through. I think everything came together for me much more quickly once I disengaged the Helix volume knob from my XLR output and set my output to Mic signal level. From there I can gain stage my presets easily using a combination of amp channel volume and output block level neither of which will affect my tone. I only use the amp master volume if I'm interested in getting the tone change it offers to the patch, but that's still going to be normalized via the amp channel volume or output block volume. With my gain/trim knob on my mixing board channel set to noon I simply target a consistent signal level of 0db on my mixer channel with a few peaks above that. That signal can then be adjusted at the gig using the channel's gain/trim knob if necessary. I personally never use any global settings on the Helix as that opens the possibility of being affected by someone at the board applying an RTA process to correct for room EQ, which would affect my global settings. If the room acoustics are affecting me, they're affecting all channels, so that's something I leave for the sound crew to address. As far as high cuts, that all depends on the preset and the guitar I'm using. Generally I always have a final parametric EQ in my signal chain which is where I make final adjustments. Most commonly my high cuts are in the range of 8k to 10k depending on the guitar, the amp and the style of the preset. That's also where I make any final adjustments such as a 4.2k cut with Strats or Teles to remove some twanginess. The VAST majority of my core tone adjustments come from my configuration of the cab/IR and which mics and placements I use. If I ever find myself trying to adjust EQ too much I generally go back to the cab and mic models and that normally addresses it. The one thing that seemed odd to me was how you could achieve a good sound through the mains but it was off in your monitors, but that depends a lot on your monitors or in ears I guess. Since I use high end stage monitors that could just as easily be used as mains, I never have any problems getting an accurate representation of what the mains sound like to the audience. The mix may be different in the monitors from the mains, but tone is pretty much consistent. As far as phantom power, I always carry a phantom power blocker with me just in case, but I've never really had the need to use it as that's something that can typically be taken care of on the mixing board.

Just to be complete and give you a better understanding...a key aspect of doing a sound check is to gain stage each instrument channel coming into the mixing board so they're all at a consistent signal level with adequate headroom so they can then be mixed with the faders and not cause have any problems clipping. This level varies board to board and engineer to engineer, but is at the core of how a mixing board is setup.

If you've been playing live you've been gain staging whether you wanted to or not regardless of the equipment or setup you're using.. Always have and always will. This is because all mixing boards in my lifetime of almost 70 years requires each input signal to be within a certain consistent range so that it won't distort or cause problems on the channel it's coming into. The only difference with the Helix is you can ensure every preset is gain staged accurately on the output side so that the sound man only has to adjust your input level once and it will be good for every preset or snapshot you play. As rd2rk mentioned the whole "not exceeding the signal level of an empty preset" is nonsense and has been dismissed for many years now. What you do have at your disposal is a signal meter on your output block that, if selected will give you a rough idea of where your signal strength is sitting on your preset. When I was depending on that for gain staging I'd typically shoot for around 70% and the soundman can then adjust it up or down to fit his needs at the board. You have to bear in mind however that only measures the digital signal level, not the ultimate analog level that gets sent out of the XLR or 1/4" output. For that reason the most common guidance and simplest to deal with is to configure your global settings so the main Helix volume knob is disabled (which sends a non-attenuated audio output signal) and set your XLR output to send Mic signal level which is the most common signal level used for mixing boards. I have hundreds of presets that I've built over the last 7 years and I wouldn't be afraid of plugging any of them into any PA system and would expect all to work perfectly and be easily gain staged on any mixing board.

To be sure, there have been some changes in the underlying DSP technologies that ALL modelers depend on, but that hasn't been any kind of quantum change like it was 7 years ago when this generation of processors was introduced. What has changed is the availability of programming expertise in working with these technologies. That applies to any and all of the current and older chips and it applies equally well to the development expertise at Line 6. When I got my Helix 7 years ago I really couldn't imagine all the things they've been able to do with this technology. But none of those changes have anything to do with newer versions of the DSP chip. It has to do with learning how to do it better and more efficiently. Some of that applies to the basic tone such as the oversampling that was introduced a couple of releases ago. The VAST majority of the improvements, in my opinion, were the result of Line 6's decision to re-architect the underlying code into what they referred to, at the time, as "Helix Core" a few years ago. This is what newer people don't understand about newer products. All of those products will undoubtedly need to go through that same evolution because you NEVER get it optimized the first time through. The Helix has been there and done that already.

My problem with the Xvive is I already have transmitter and receiver for my headworn mic and the transmitter/receiver for the headphones interferes with it and you can't juggle the frequencies to clear it up. At that point I just dropped the whole effort because we were doing fine with our current stage monitors. I may go back and revisit it later because I did have to suck it up with the premium IEMs I got as they're non refundable.

Wasn't that the same point I was making about using convolution reverbs commonly in samplers in my DAW (which of course would be on a computer)??? Have we come full circle yet?

I'm no expert on convolution reverbs, but I do use them and encounter them all the time when using sampled instrument libraries in my DAW and that tells me something about the difference in how they must be processed compared to a modeled instrument. A modeled instrument is a stream of math computations whereas a sampled instrument is pretty much a fully formed slice of audio that can easily be overlayed with another sample. That may be why you don't see them in modelers but you see them everywhere in samplers and it all comes down to the main issue modelers are always battling which is avoiding latency.

I "pre gain stage" my outputs on every preset I'm going to use live into an actual mixer (either a QSC TM30 or a Yamaha MG12XU) connected to my Yamaha DXR12 so that all my presets are all dialed in at a consistent signal level. It may vary slightly from mixing board to mixing board, but that's really irrelevant. The signal is always consistent so it's just a matter of gain staging one preset on any mixing board at a live performance and they'll all be consistent for that board and normalized in volume levels. It's a big time saver when doing sound checks. You can do the same thing using the built in signal check at the output block, but it's much more accurately measured on a an actual XLR input channel of a mixer. I always disable the Helix volume knob so that my outgoing signal on all presets is non-attenuated at Mic level.

Actually, although I agree in principal with the cost analysis above, to get a decent functioning IEM setup would cost me about the same as my Helix originally cost me once you consider the cost of a premium set of earplugs and the wireless system to support it, so it's not an insignificant investment. But all of this is generally a problem of our own making for the most part. Even in this age of technological innovation musicians still struggle trying to manage their stage volume to the point they create the monitoring problems they complain about. That's understandable to a degree when you're dealing with traditional stage amps and drums. But it's not impossible to get by with just decent stage monitors if you address the core problem. By eliminating the stage amps and acoustic drums and going direct to the mixing board we eliminated the core issue of stage volume and can hear everything just fine including vocals and harmonies for three singers just using stage monitors. The added benefit was reducing our setup and break down time.

I don't know if it's an impedance issue or not, but it's certainly not normal to run in ears out of a headphone output. Most people run them out of one of the Aux outputs on the mixing board which are meant to be used for monitors and can be given their own custom mix from whatever is plugged into the board. That being said I use my in ears at home all the time coming out of my computer headphone jack with no problems.

First, tweaking your presets for in ears is about the worst idea I've ever heard. If your mixer provides it you can tweak your guitar channel's aux output EQ for your in ears. You're there for the audience who's paying your way. If it comes down to who gets the best sound it should be them. Secondly, the IEMs you're using are NOT tailored for use with an iPhone. They're pro grade and that's why they're sold at Sweetwater and other pro stores. In my experience you can get a serviceable experience with IEMs, but not a live stage experience. The closest I've come is using a much higher quality IEM that incorporates some outside ambient sound and isn't completely isolated such as those by Westone. But I've given up on IEMs and gone simply to high quality stage monitors with everyone going direct to the board and no onstage amps to get a manageable stage volume. It's a much better live experience in my opinion because I'm hearing exactly what the audience hears and exactly what I dialed into my Helix at home, but mixed specifically for me. You're not going to get a guitar speaker in a room sound because your audience is never going to get that from you or from any other live performance even if you use stage amps because you'll be mic'ing them. They'll always get the mic'd version of a live amp just like every recorded album they've ever listened to.

And yet, I have over 300gb of music-related programs, plug-ins and drivers on both of my computers and none (including the Line6 ones) have ever failed in over 10 years of full-time daily use in either my studio or live system. I wonder what the difference could be? Oh yeah..you install and maintain the software on your computer, and I do the same on mine!!!

Wait!!! Hold on..... What were you saying? I was totally distracted by all the snoring going on in here........

From my reading of Microsoft Tech Notes on W11 performance improvements, the only relevant thing they've done is some work on prioritizing the allocation for computing time to active in-use applications over background processes or "sleeping" apps, so there may be some improvement in DAW performance at higher CPU usage levels than what you might have experienced in W10. That might also equate to improving runtime on longer running processes such as converting a project to a stereo image. Other than that I didn't see anything particular about sound devices or audio processing specifics that would relate to DAW work. https://techcommunity.microsoft.com/t5/microsoft-mechanics-blog/windows-11-the-optimization-and-performance-improvements/ba-p/2733299

Amp modeling and cabinet modeling are two very different technologies and methodologies. Amp modeling is done through the mathematical simulation of the behavior of the circuits on an input signal going through a given amp. IR's are a simple simulation of the sonic characteristics/behaviors of a specific speaker, cabinet, and microphone output. It's like comparing the space shuttle to a bicycle.

In 7 years of using my Helix in live performances every week, you'd be surprised how rarely I felt I needed a dual amp setup in a preset. But there's a ton of stuff in that book that relates to things I use every single week in various presets. Honestly if dual amps are a major concern for you then why didn't you opt for the LT, Floor or Rack?

It appears to me given all the reported problems on this update that you might be better off uninstalling this update.

You can move to any preset in any setlist using MIDI. Just press the Presets knob and select the appropriate setlist and preset and you'll see what MIDI CC value (for the setlist) and PC value (preset) to send to load a given preset.

I think you may still have a bit of a problem with vernacular and/or understanding. Generally songs would be assigned to presets within a Setlist. Generally bank numbers are simply nothing more than a reference number for a preset the most part. When you cycle up or down most people are simply changing to the next or previous preset, and that preset might have a completely different arrangement of amps and effects. Also I'm not sure where you're trying to go with Song 1 = Banks 1&2. Forget about banks which are nothing more than numerical references and just think Presets and you only work on one preset at a time. Also, age has nothing to do with it. I was in my early 60's when I got my Helix and I'm closing in on 70 now. You can learn anything you're willing to learn if you have the desire to do it. There are literally TONS of introductory teaching videos on YouTube available to you that weren't there when I started so you have all the help you would ever need to learn this unit.

I do exactly as you're describing. I disable the Helix volume knob (which sends the signal out at full level) and I have my output set to Mic level. Every soundman has their preferences for where they like to set the incoming signal, so I gain stage my outbound signal level through my own mixing board and target the level at just under unity with limited peaks above unity by adjusting my amp model channel volume or the level of the output block, neither of which will affect my tone. Because they're all set to the same level any soundman only has to gain stage my signal on one of my presets to whatever level he prefers and all other presets will be adjusted to that same level. I've worked that way for about 7 years and have never had a problem with any soundman or PA, and have received many compliments from soundmen on how easy I make it for them to gain stage my setup.

Maybe Line 6 is just waiting for some compelling reason why anyone would upgrade to W11 before committing to it.......LOL