njglover

It's all digital anyway, that really only matters if you are using analog effects (in which case it absolutely can make a difference, especially with fuzz).

I have the original G10 and have used it with a passive guitar (PRS Custom 24) and an active bass (Stingray). If I went to, say, the opposite end of the room at the bar we were playing at, I could get dropouts, but I never had any issues on stage. Or at least, the one issue I had was that the jack on the guitar/bass (can't remember which) didn't have the nut flush against the end of the jack, so at some point the transmitter rotated and the little button thing that detects if it's plugged in or not popped back out, so the sound cut out. Simple fix, just adjust the jack on the guitar/bass. I've never felt that the roll off was particularly egregious, and this is coming from someone who has bought just about every cable available and sat in front of an amp swapping between them to see how they sound. For reference, my general preference for wired is either Mogami Platinum (yeah, the $100 one) or Mogami Gold. By comparison, I have no issues with the sound of the G10. I also understand that some people have had issues with the USB power jack on the original G10 receiver, but after a couple years and many gigs, I still haven't broken mine, so I dunno. I really liked that I could power it off a USB battery pack. Overall, the G10 has been my favorite wireless system, including over the original G50, G30, and Shure's GLX-D. But YMMV depending on how many other people in your band are using wireless, the general wireless noisiness of the environment, and how far you like to run around.

I think this question actually comes out pretty often around here, or at least I've seen many threads about it in the past couple years, though everyone seems to call it something different. Fizz, static, clipping, distortion, etc. The last thread I remember looking at suggested messing with the bias values for the amps to see if that helps, but ultimately it's just something you kinda have to live with. Some amps are worse than others, some people are more bothered by it than others. Anyway, try messing with the bias and see if that helps. If not, you could search around and see if there are other ideas (I honestly can't remember), but just know that it's normal. Edit: Here are two threads. This one doesn't really offer many solutions, but might be interesting reading: This one is better on the solution front, they suggest messing with bias and also with hi cuts on your cabs:

One of the legacy synths (not explicitly an octaver) can be configured as a killer sub octave generator, too. Can't remember which one it is off the top of my head, but boy does it make a room rumble.

Not exactly sure why it behaves quite like that, but ground loops are weird things. Have you tried messing with the ground switch on the back of the Helix? That would isolate the ground going to the KRKs from the rest of the circuit and could help.

Honestly, for live performance I think it's probably better to not use the volume knob at all. I always set it so that the volume knob controlled only, say, 1/4" outputs while sending the XLR outs to FOH. That way, the engineer gets exactly the same signal every time I play. I found that doing that and setting the XLR outs to mic-level out was generally the way to go. If I had my own stage wedge (which I never did), I could still control that via the volume knob since I'd use the 1/4" outputs for that.

I thought about the wireless thing, but I guess I kinda figured that wasn't still a thing. Used to be a thing like 10 years ago, I just hoped they had figured that out by now... guess not.

I'd also uninstall ASIO4All to make sure it's not causing any conflicts, then install the official driver (if you haven't already) and use that one. Some of those settings might be trying to apply even when you're not using it. 128 or 256 buffer size ought to get you pop-free sound, but you could also try putting the Helix into a different USB port and see if that helps. To be clear, not just like the next port over, but one that's on a different host controller. If it's currently plugged in to a USB 3 port, try a USB 2 port. No guarantees, but that could help.

Les Paul experimented with balanced output on guitars. I think there's one weird model of LP from the 60s or 70s or something that has XLR out. But obviously it never caught on because there's no real reason to do it. The fact that every wireless unit has some sort of "cable tone" feature should be evidence enough that guitarists don't actually want an unadultered signal.

I don't understand. Right on the download page it says, depending on whether you look at the Windows or Mac downloads: Compatible OS: Windows 7, Windows 8, Windows 10 Compatible OS: Mac OS X, Mac OS Yosemite, Mac OS El Capitan, macOS Sierra, macOS High Sierra, macOS Mojave

I don't know that there is anything you can do about that, then. I believe that's the way most real delay pedals work, too, so naturally the models will have the same behavior. Consider that a delay is basically a recording device that can record a short length of audio and then play it back at a later time. When you bypass it, you are just stopping audio from going in and out of it. You are not actually stopping it from operating, so any audio that has already been recorded and is queued to play back is still going to play back, you just won't hear it unless you turn the effect back on. This will eventually go away because there is no audio coming in to be recorded, but the brains of the pedal are always operating as long as the pedal has power. If you need to be able to turn it off and then back on again so quickly (though I am not sure what the musical reason for this would be), you will probably need to use two separate delay blocks and toggle between them.

This topic comes up a lot. I think my experience mirrors many others. Initially, I was not getting what I wanted with stock cabs, so I got frustrated and downloaded a bunch of IRs. I demoed them for hours until I found a few I really liked and then used those pretty much exclusively. But when the Stomp came out, I wanted to see if I could get sounds I like with the stock cabs (in the interest of making my setup Stomp-friendly). I ended up not getting the Stomp, but I learned a lot about the stock cabs and came to find that they are really just about as good in most settings. With headphones on I can definitely tell the difference between the two, but through an FRFR or in the context of a mix, there's virtually no difference, so I've moved to mostly stock cabs these days (for DSP savings). Just depends what I am feeling.

Honestly, I have done it by figuring out what cuts I end up applying in the final mix and then going back and applying those to my patches. I used to cut at like 11 kHz and then discovered that actually I prefer cuts all the way down to maybe 7.3 kHz most of the time. For cleans I find I don't need to be particularly aggressive, but distorted and fuzz tones definitely benefit from generous cuts. YMMV, of course, but my preference is to cut at the source and not have to do it in the mix later. But it's still better to leave a little too much in and not need it than to cut too much and want it later.

It's... not. As suggested, possible there is something with the model that sounds "off" to you or the OP, but it is not a "problem" in the sense that there is nothing wrong with the signal chain or the Helix. It is either an issue of playing style or misunderstanding of what it is supposed to sound like.

As I understand it, Bias X is essentially your bias multiplier, so it controls how much the Bias parameter affects your signal. I'm not sure if there's a specific formula for it, but I imagine that a low bias with a high bias x would be similar to a high bias with a low bias x. Or something like that.