njglover

I don't normally use the top row of footswitches for anything other than switching presets, but last night I thought I'd mess around with some of the factory sounds and I noticed that the footswitch second from the left on the top row (footswitch 2) is not working right. Sometimes when I push it it doesn't toggle, other times it does but then it toggles right back off. I can hold it down and if I move my foot a little bit it will switch, sometimes on then off then on then off... I can hear it physically clicking, so I'm not sure if this is a hardware or a software issue. Has anyone else experienced this? My nearest repair shop is probably 40 miles from me...

I usually do the amp and cab as separate blocks and I'll stick the reverb after the amp but before the cab. When I want a reverb that sounds like an amp reverb, this works great. Setting it after the amp makes it sound too much like an effect and I don't like that (most of the time).

No, it has nothing to do with where the knob is set. It simply outputs a more or less standard line-level or mic-level signal.

In the global settings you can change which outputs the volume controls. You can set it to multi, XLR, 1/4", or digital. If you set it to digital than all your regular hardware outs output at fixed levels (which can be specified as line/instrument/mic level depending on the output).

Haven't had a chance to actually try these yet, but I did finally download them. When you refer to distance, is that distance back from the grille or distance sideways from the cone? I suppose if these work out I could just buy IK's model and do the IRing myself following your guide.

That would actually be a really elegant solution. Angles or even just a second distance parameter, distance from center. It's annoying to me that, in order to make the cab sound "real" I have to use the low and high cuts (though even on the IRs I use I'm using the high cut, but in that case just to tame some fizziness) and the room tone option. But the more annoying thing is that, even after I do that, I can get really close to where I want to be, but just moving the mic a little farther off axis would get me that last bit and then I wouldn't need to bother with all these IRs.

Yeah, I played with those a bit in demo mode, they aren't bad. I'd normally be using either a Rockerverb or the Dual Rec with that cab. Really any amp sounds good, that cab is great...

Nah, that's the Ownhammer ones. I'd like to make an IR of my Orange PPC212OB but I don't think I have the right gear for it.

IRs are dangerous. Yesterday as an experiment I recorded a bit to compare to a track I had recorded of my actual Rectifier for an album and discovered I had not mimicked the tone nearly as well as I thought I had. I spent the next hour going through basically every Mesa 412 IR trying to find the right one...

Now that I think about it, I had been using it in Ableton Live before yesterday just fine plugged in to that same port. Something somewhere is just not reading the settings right.

Yeah, haven't tried 2.0 yet, have to free one up first. So annoying that 3.0 never seems to really cooperate even though it's supposed to.

This morning I thought I'd try using the Helix as an interface and record a quick riff to compare it to my actual amp just to see how it stacks up, but when I first went to play back my old recording it sounded nuts, almost like it was bitcrushed. So I played with my buffer settings and tried to check various other settings, here are some things I discovered: -No matter what I set the sample rate to on the Helix, Windows only gives me the option of 16/44.1 or 16/96. It also doesn't matter than in the driver panel I have it set to 24-bit, Windows still reads it as 16 (I assume the driver the Helix application installs is the correct ASIO driver, anyway). - On a related note, does the sample rate set on the Helix impact the sound coming out of the Helix itself? I'm fine leaving it at 44.1 as long as what's coming out of the actual unit isn't lower quality as a result. -Studio One reads it, correctly, as 24-bit, as set in the driver settings, but does not allow me to change the sample rate. -Finally, after much frustration, I found that changing the bit depth in the Helix control panel to something else and then back to 24 got everything working correctly. So basically, nothing is communicating with each other. Helix says one thing, Windows says another, and the DAW says yet another thing. If it helps, I am running Windows 10 Pro 64-bit, it's an ASRock motherboard, and I was plugged into one of the front USB 3.0 ports. Using Studio One 3.

The Alto speakers peak at 800 watts, if I remember correctly, but are probably realistically pushing more like 400 or so watts. Which is still plenty if the guitar is all you're running through it. I use a K10 which pushes 1000 watts and I haven't needed to turn it up very loud at all. I left the volume knob at 9 o'clock and just controlled volume off the Helix and that was more than enough for a live set. And we generally play very loud. The K10 also has the advantage of having a very wide dispersion so everyone can hear it at practice and live I can use it like a wedge behind me and still hear myself well. Typically, at a nice venue I'd probably leave the volume knob disabled and just run a consistent line out to the board and leave the engineer to adjust from there (I did that with bass recently and it worked really well).

Yup, I run my Stingray bass into the normal guitar input, no problems. In fact it sounded weird in the aux (the different impedence changed the tone a bit) and I couldn't drive the SVT model as hard, so I put it right back in the guitar input.

If you want to do it that way, you could just set the expression pedal to control the A/B split. I usually use an A/B for songs I need to go from clean to dirty and have different amps on each path (but the same cab/IR on the end to save CPU) but in my case I just footswitch between them. No reason you couldn't just fade between with the EXP pedal, though.

It depends what you want, I often find that I WANT the cleans to be a little quieter so the distortion has more impact. You could also just use the volume knob on your guitar like they did back in the day and, if you've got a treble bleed circuit, it will naturally get brighter as you turn it down.

I mean, you could just set the expression pedal to control the amp's gain, that way it's clean with the heel down and dirty with the toe down.

I mean, that is the exact purpose of the global EQ. The soundguy will generally have, at best, a 4-band EQ with sweepable mids. The global EQ is more flexible and provides better control so you get the sound right to begin with. I have also never bothered with that sort of thing much, but a good sound engineer at a good venue will let you know if you need to make adjustments for the room and/or PA.

I generally recommend using the high and low cut filters on the cab block. Inserting an EQ block is a waste of block space for that and the global EQ is better reserved for fixing variations between different PA systems (ie you would adjust it when you get to a gig).

I got the Mono M80 case because I love me a good Mono case but there's so much extra room inside... what are people filling that space with? The ideal case should be a good, snug fit for maximum protection, I don't like it sliding around like that. The case they used to sell with the PT1 at Guitar Center is practically a perfect fit but that board was a GC exclusive and they don't even make it anymore, as far as I can tell.

So a couple additional points. First, I don't think those Celestion charts are terribly clear, I interpret that as rolling off somewhere between 10-15 kHz, but it may very well be lower than that. The main point here is that they produce primarily midrange frequencies. I think we all agree on that. The second point is that there doesn't NEED to be profiles based on what you are listening on. Since we can all agree that real guitar speakers do not produce frequencies above, say, 5-10 kHz, there is no need to ever have those frequencies coming out of the cab models (unless someone wants a strange tone). Assuming the goal is a realistic tone, then rolling off high frequencies would produce a nearly identical tone to a real miked cab. In which case, that is what would also come out of a FRFR using a real miked cab. Which is also, therefore, what would be recorded in a studio and also what one would normally hear through home speakers or headphones. So my point is just use the high cut on the cab block FOR EVERYTHING. There is no reason to ever have those high frequencies as no live or recording engineer would ever have those to work with. You would simply want to use the global EQ to adjust for differences between different FRFRs to get you a more consistent tone from gig to gig (because, as mentioned above, no two PAs are exactly alike). Again, only Line 6 can no the reason WHY they set up the cab blocks like that, but the reality of the situation is that if you want a realistic tone no matter what system you are listening on, engage the high cut. As a side note to the above post, I believe the Line 6 speakers are intended to have a similar sort of modeling capability, but I never looked into it much.

Truthfully I have no idea why they wouldn't build it more correctly. Perhaps they focused more of their energy on making the amps sound and feel right (which they do). Someone here suggested that Line 6 is actually using IRs for the cabs on the Helix, but I doubt that. I suspect that they attempted to model them more like they did with the amps, sort of piece by piece, but speakers have very complicated interactions that aren't even remotely linear so it's very hard to model them correctly. Maybe it's something they will work on in future updates.

That is exactly what I am suggesting, though I wouldn't call it a bug so much as it is just designed wrong. I am basing my assertions, like you, by looking at the frequency response charts available on the manufacturer's web sites, not on the actual output of the Helix. If you look at an average guitar speaker's response chart, it rolls off substantially starting at about 14 kHz. Similarly, a Shure SM-57 rolls of at around 15 kHz (the 57 model in the Helix is entirely too bright, IMO). So, theoretically, opening up a cab block and rolling the high cut down to about 14 kHz shouldn't make much difference, if any at all. But it does. I don't notice it as much on clean tones, since there aren't as many harmonics being produced, but on distorted tones it makes a HUGE difference. With the hi cut active, it starts to sound more like a real cab when listening through your headphones and/or FRFR. I highly recommend trying this out for yourself to see if that helps. It shouldn't make any difference to your tone when playing through the DT25, as the DT25 already rolls off those frequencies, but it should make headphone listening much better. Further, I would do that on the cab blocks and not with global EQ a) because you can and b) because now you can still use the global EQ for what it's actually meant to be used for, tweaking your sound to the room. You could also try downloading Ownhammer's free IR sample, which is a Mesa 412, for comparison. I find that the IRs have a much more accurate frequency response (as they should), so they sound great on headphones and FRFRs without having to mess with high or low cuts. As a result, I have replaced all my cab blocks with IRs and now I can barely tell the difference between a real miked cab and my Helix tones.

I don't think you are understanding me. I am suggesting using the high cut as an always on thing, not as on only when using the headphones. As I said, if the goal is to get a realistic amp+cab tone out of the Helix, then it shouldn't be producing frequencies that high anyway, as you noticed in the graphs you posted. The fact that it does produce those is the problem. So you roll off on the cab block anything above 14 kHz to better mimic the response if a real cab, now not only does it sound less harsh in your headphones and on a FRFR, but it also sounds basically the same on your DT25 because it wasn't producing those high frequencies anyway. If the point is that Line 6 needs to fix the cabs, then I agree, but if the point is that there is something wrong with the headphone out, then I very much disagree. The problem in that case is the expectation.

That is essentially my routing. Splits right before the 400+ model so I have no modeling going to my rig on stage and then a full model of my rig going out the XLRs to FOH. I actually set it so that both outputs are outputting at fixed levels, too, which works quite nicely. I'm considering if it might not be better to just have a clean power amp to plug into the cab instead for when I need to bring an amp.