DunedinDragon

The rebuild process is a one time only process for reconfiguring the presets to match the format expected by 2.82 both on the Helix and in HX Edit. 10&11 should be the same process. Actually subsequent to the reconfiguring of presets you should find your bootup process is significantly faster than what you've experienced in the past.

As cruisinon2 stated above, if you love the sound you're getting going direct to the PA, then why are you trying to do something different on stage? You're adding complexity when it's not necessary. The vast majority of PA system are using FRFR powered speakers of one variety or another. All you have to do is add one to your stage setup and run direct to it from the Helix with no changes at all to your presets. With the tiny amount of money bands get paid nowadays why complicate things and add to the time and expense of setup and teardown??

And such is the burden one bears when buying something without conducting due diligence, huh?

In my experience using studio resources to gain stage your live output signal is touchy at best. It just seems easier to me to use live resources such as a real mixing board with active meters to see what's really going on with my Helix output signal so that it's a simple straight connection to my mixing board channel with no intervening software or hardware configurations to distort the results. By doing it this way I've never had any problems with the levels on my on stage direct monitor or on any of the hundreds of different mixing boards in the field. It's always consistent among different patches and snapshots and only takes a moment to gain stage my rig for both stage monitoring and FOH monitoring. Here's the setup. I have my Helix configured so that my XLR output is disconnected from the Helix volume knob and that signal (which is the one I'll send to the mixing board) is configured for Mic level output in the Helix global ins/outs. The means that my XLR signal will always be sent at full volume from the Helix to the board at mic level so I can use the meter on my mixing board channel to adjust the levels of the blocks in my preset to present a consistent output signal level with no clipping. My 1/4" output is set to Line level output and uses the Helix volume knob to determine the output level. My standard FRFR on stage is a Yamaha DXR12 which has its input gain always set to Unity or the 12 o'clock position. The concept being that if I configure the blocks in my preset such that they are all at appropriate output levels with no chance of clipping on the XLR output to the board, that level won't clip on my 1/4" outputs to my on stage direct monitor, unless I turn up my Helix volume level too loud which won't be clipping from the Helix signal, but will be clipping (or in my case, invoking the built in limiter) on the DXR12 which would never happen given the level of stage volume we use. This has the added advantage of ensuring all of my presets and snapshots have similar outputs so if you gain stage one preset live on the mixing board at a gig, you've gain staged them all. The other added benefit I found is it makes all of my amp model volumes act more consistent with what I would expect from the actual amp being modeled. For example Roland Jazz Chorus model is consistent with a Jazz Chorus amp, or a HiWatt model is consistent with a HiWatt amp. As you can see from the picture below I currently use a QSC Touchmix-30 live mixer when I'm dialing in presets, but I've used smaller and cheaper ones in the past which work just as well as long as they provide a decent signal meter display (there's actually an older small mixer I used to use positioned behind my Helix in the picture). Maybe there are some ideas in my setup that might help you get to what you're trying to achieve as well.

I'm not sure there's anything normal in your description, especially in terms of gain staging output from your Helix. I use Cakewalk Sonar Platinum and have for many years. There are any number of ways you can adjust the USB recording output level, but most of the time I simply take the output as presented by the Helix and adjust the trim/gain on the track after it's captured to get the signal level where I need it, but there is also an adjustment in the globals to directly adjust the level of the USB output on the Helix. I'm not monitoring the track as it's being recorded anyway, I'm monitoring the source. First of all your use of "gain" is very confusing as gain applies at several stages on the Helix, but I'm assuming you're talking about the Helix output block. If so, that has nothing to do with the performance of your FRFR speakers. What should be presented to your FRFR speakers is a consistent, managed output level normally at Line level to an FRFR speaker or Mic level to a mixing board. In both cases you manage the signal level using the gain on the speaker or on the mixing board. In the case of your speaker, setting the gain at 12 o'clock or midway is defined as "Unity" for the speaker, meaning the amplifier in the speaker is operating without any preamp gain or preamp reduction to the signal. Generally this is where most speakers should be set to so you have ample headroom to avoid any problems with inadvertently engaging the limiter on the speaker. Most people I know adjust the Helix volume output if they want more volume depending on the specific needs at the time. Louder does not mean a better signal. In fact if the signal isn't managed louder can mean compressed when it comes to an FRFR as many have built in limiters in their input circuits to protect the speaker. The normal convention is to ensure the Helix is outputting a consistent signal level across all patches and snapshots and leave it there. I personally measure this output on a mixing board using the signal meters to show me the output level which I try to keep at unity on the mixer. If it's too high or too low I lower or raise the amp model's channel volume which doesn't affect the tone, but the same can be done with the output block if you choose to do so. Jason Sadites has two VERY useful and informative videos on gain staging with the Helix he just added this week to his YouTube channel. I'd highly suggest you go view those to get a better understanding of the concepts.

Everyone has their preferences in these two scenarios, so it's really up to what YOU need and want. I personally prefer my Helix direct into a powered speaker (Yamaha DXR12) because it best represents the sound I will get when I'm live going through a PA. This makes it much easier for me when I'm dialing in my tones to have confidence when I get to the gig as well as getting a good stage mix with the rest of the band. Not having the traditional amp and cabinet setup doesn't really bother me at all. What bothers me more is not knowing if what I'm hearing on stage is significantly different than what's being mixed for the audience at the PA. It's not that solution number 2 will disturb the simulations or modeling, it will just represent them differently because of the mechanics involved. But that can be compensated for in your preset if you send a separate signal direct to a PA at a gig which has all the traditional FRFR configuration in it and are just using the amp for stage sound. However, given the limitations of the stomp I'm not sure how viable that is.

It's definitely something in the way your PC is handling the sound output as the way you have things configured is rather odd. The first thing I would do is disconnect the speakers from the line on the PC (I assume you're talking about the 3.5mm jack on the PC soundcard. If your Helix is connected to the computer via the USB port and is connected directly to the speakers with an XLR cable, that's all you need to handle the guitar output as well as playback from the PC as the USB connection to the PC makes your Helix behave as the PC's sound card. In that configuration you simply have to adjust the sound level from the PC to get the right mix of the backing track from the PC and control the guitar volume with the Helix volume knob.

What I would suggest is what I do. I don't touch any EQ until I get my amp and IR (or cab) and other effects (compressor, reverb, delay, etc.) relatively close to what I can be happy with. Only then do I begin to adjust the parametric EQ toward the end of the signal. When I dial in my tones I'm going through a Yamaha DXR12 which is positioned at about chest height and 5 to 6 feet away and I listen to it both on and off axis as well as with my back turned. That gives me a highly accurate representation of what will be coming out of he PA. I think a lot of times it's not just the guitar and your speaker but your position relative to the speaker that can be deceptive.

I think a key ingredient why Jason Sadites gets such great tones without severe or significant high and low cuts is his attention to detail in how he applies cabinets, mics and mic placements. Although I don't use the stock cabinets, I tend to follow the same general approach as he uses with IRs typically combining MD421 dynamic and R121 ribbon microphones in such a way that there's not a lot of high and low cuts necessary because there's just not a lot of fizziness. Like him I use a final parametric EQ on my patches and rarely need a high cut deeper than 8 khz and generally am fine with high cuts in the 10 khz range. I suspect a lot of folks that require such deep and significant cuts would benefit from cleaner and more authentic tones by following his approach.

A lot really depends on your approach. I personally have distinctive presets for specific to each of my guitars depending on the sound I'm after. That's because the preset I would use my strat on is not the kind of preset I'd want to use my Les Paul on, or my Gretsch hollow body, or my Tele, or my Gretsch resonator. Not only is the guitar different but most of the time the amp is different and the cab/ir/mic setup is different as well as many of the other blocks. The guitar is every bit as much a part of the signal chain as is anything else in the preset.

The Helix isn't really in charge of the volume as it's just a preamp. Performance or stage volume is entirely up to your output device capabilities. I assume you understand the preset you use with the Marshall or Fender won't need to use the cab simulations or IRs, but those will be necessary once you go to a mixing board. That might make things a little dicey to get your sound dialed in correctly without a FRFR that would simulate the performance of a PA before actually going on stage into a PA.

I prefer the 80 ohm. Not so low that you get noise and too much volume, and not so high that you need to turn up the volume to get a big, full sound.

Pretty much around here and in the general Helix modeling world the DT770 is pretty much a key workhorse. There's lots of reasons for that but it's mostly because they are consistent and with very little coloration which makes them easy to integrate with studio speakers and live speakers accurately. Here's a good comparative review of both: https://www.rtings.com/headphones/tools/compare/beyerdynamic-dt-770-pro-vs-beyerdynamic-dt-1770-pro/440/714

Sorry, but that's just incredibly naive. No company and no product in the world is supported indefinitely because it becomes increasingly costly and steals significant resources from current LOYAL customers and current product improvements as time and technology move forward. And that's exactly as it should be. If that's your impression of what is "right", then why would you be hypocritical and suggest Windows 3.1 shouldn't be considered in the conversation?

Well there's an enormous difference between a MIDI utility and native devices and software written to work with the underlying operating system services. MIDI is a simple communications protocol that's been around for decades and was written specifically to be operating system independent. That doesn't compare to calling on underlying native services in a specific operating system that's changed dramatically since Windows 7 in particular. The reason Windows 7 was dropped from support is because the underlying service calls were no longer consistent as far as calling parameters or in behavior with the more modern versions of the operating system. The fact is the internal structure of all arguments passed to services inherently change once you move to 64 bit.

Wow!!!!! Amazing what magic happens when you follow the update instructions, huh???

Clipping isn't really associated with DSP usage. If it were possible to overuse DSP it would result in latency not tonal impact because it would still be performing the mathematical transformations required for the model, it would just take longer so that latency would be apparent. That's why it's controlled.

I suspect the problem here might be that the Mac is not expecting input from your Focusrite because it's not seeing it as an active input device but only as an active output device. You'd need to use a DAW or some other utility for it to see it and use it as both. I'm not a Mac guy, but I know that would be the issue on a PC. On a PC you'd have to tell it you're expecting to use that device actively for recording or at least for active sound input. The real question is my mind is where and what are you using for speakers? If they're connected to the Focusrite you should be able to disconnect the Focusrite from the computer and hear the Helix through the Focusrite with the speakers.

Helix only supports cabinet IRs.

Well, I hate to be the bearer of bad tidings, but the FRFR-112 is, in essence, a PA speaker. Granted it's a significantly diminished PA speaker functionally, but it follows the same basic design principles as other speakers like the DBR10, just without all of the DSP contouring features. As is always the case, you get exactly what you pay for generally. The powercab has some useful Helix oriented features and uses a coax speaker rather than the speaker and horn arrangement on a powered speaker. All of the options should work fine for you, but you'll still need to make the adjustment from a traditional speaker cabinet to a FRFR style arrangement. The Powercab might make this transition a bit easier simply because of it being in the form factor of a speaker cabinet. In general loudness shouldn't be a problem with any of these options unless you're dealing with undisciplined musicians that have no sense or control over how they play. The one clear advantage powered FRFR speakers have over the Powercab is that the design of a FRFR speaker is such that it projects a consistent sound across a wide horizontal axis and is designed to project further simply because that's whats needed to cover the audience. This can be to your advantage if you mount it on a shorter stand behind you in you backline as it will have the same effect as you would have if you were to plug your Helix into the PA as far as audience coverage. I always mount my Yamaha DXR12 behind me in the backline whether I'm going through a PA or not and it works consistently in both small and larger venues.

If everything is on a power strip it doesn't apply. But in a ground loop situation ANY connection can cause the differential in ground to cause interference. Your guitar connected to the Helix via a cord, the Helix connected to an amp via a cord or the computer via USB, etc. Even your body can cause the ground differential to be balanced. Just ask how many electric guitar playing singers have gotten the shock of their life when their lips touched a microphone.

Sounds like a classic ground loop where your power supply is on a circuit that has a differential in grounding with the circuit something else it's connected to (like an amp). Get everything plugged into the same circuit or get a HumX for your power supply.

That's going to depend on how the IR is shot as far as number of mics and mic placements, or alternatively how different IRs might be combined together to achieve at least the room affect. Band level is pretty much irrelevant when it comes to IRs because the difference is due to the behavior of the way humans percieve volume and you can't simulate that in something like an IR without having an adverse effect on all other volume levels.

I believe the Litigator model is a rough approximation of a Dumble. It's called the Litigator because Line 6 would likely be in court defending itself if it stated the amp was a copy of a Dumble. All I know is before this model was introduced a few years ago there were always questions about when we might see a Dumble, and after it's release that all quieted down.....

What good does it do to discuss your EQ corrections unless you specify what configuration of mics and mic placements you're using on the IR and what kind of output device you're using to listen to it? Every bit of the frequency range gets affected when any of those things change and certainly one set of EQ adjustments won't be effective for all setups and settings. Even within the Summary folder of Ownhammer IRs there's still quite a bit of diversity in tone.