qwerty42

Oops, I'm sorry! My brain clearly wasn't working yesterday. I thought tcamuso was the OP, but he just piggybacked on your thread. I didn't realize you were using the Stomp. I don't own one of those, but I'll do my best to help. My advice below assumes USB1/2 is carrying your guitar and USB7/8 is your mic, based on my understanding of the Stomp's user guide. Thus in your DAW you'd have a channel set up receiving the guitar on USB1/2, and a channel for your mic on USB7/8. First thing I would check is make sure your guitar track in Reaper isn't also being software-monitored. This will double the guitar level you hear (because you're hearing the direct hardware monitoring from Stomp, *and* the loopback from Reaper). I don't use Reaper so unfortunately I can't give you specifics there. You said you can't turn down the guitar level at the Stomp output because it makes a low signal in Reaper. Are you sure that's actually a problem? (How low is it? Are you referring to the Helix-processed mix or the dry guitar?) The SNR on these devices is quite good, so a low signal isn't as bad as it seems compared to the old analog days. The "USB In 1/2 Trim" only affects the volume of things coming from your computer back to the Stomp. USB1/2 from the computer is hard-routed straight to the Stomp outputs and the headphone port, assuming it's like the Helix. In your case that could help, because your mic is being software-monitored in Reaper, which means you're hearing it come back on USB1/2. To clarify: if you are *only* monitoring your mic with software, and the guitar is direct monitored, then increasing the USB In 1/2 Trim should make your mic louder without affecting the volume of the direct-monitored guitar. Worst-case scenario, you could just not hardware-monitor your guitar in the Stomp, and use software monitoring for both the guitar and mic. This would let you set the volume levels with the mixer in Reaper. Unfortunately you'll have latency on the guitar audio if you do that. Edit: Another thought; again only guessing because Stomp is a lot more limited than the Helix. Can you split your signal path into Path A and B, and put a Return block on Path B that brings your mic signal there? If it's like Helix, you could adjust the mic level with that block. From there, it looks like your guitar would go out on USB1/2 still, and then the mic would be on USB3/4.

I'm afraid I can't help much on the Mac side of things, except give some general advice that hopefully still applies. In Windows, you could get rid of the iPad, and host your backing tracks and click track inside a DAW. From there, you can route those back into the Helix, have direct hardware monitoring for your guitar (avoiding latency), and then have everything except the click go back out on USB1/2, which would then be picked up by Windows as the audio source for FB Live. This would all be done on 1 computer. More specifically, as an example: Your backing track (stereo) is loaded up inside your DAW. It sends out via USB3/4 to Helix via the track/channel send of your DAW. (Disconnect it from any master tracks, you only want it going out on USB3/4). Your click track also comes from your DAW. Could be a metronome; could be a pre-recorded audio file that syncs with your backing track. Send this one out on USB5/6. In Helix, set up your 4 paths: 1. Guitar in > amps/effects > Multi Output 2. Mic in > effects > Multi Output 3. USB3/4 in (from DAW) > this is your backing track > Multi Output 4. USB5/6 in (from DAW) > this is your click track > either 1/4" or XLR outputs. You will hear this through the headphones, but it WON'T go back to the computer. Now for FB live, you select the Helix as your mic source (in Windows, it will only let you use USB1/2 from the Helix, which is where 'Multi' sends it in the assignments above). Your audience hears your mix, with everything except the click track. On Mac, I've got no experience with how it integrates audio devices into outputs for something like FB live. The same general idea should apply, though. If Mac lets you route channels from a DAW directly to a mic source for something like FB Live, it might be easier than Windows. There are probably better ways to do this in Windows, too -- this is just how you'd do it with only a DAW and no additional software.

I don't have one of these units, but I've been following these threads and I'd be pretty irked if this happened to me too. Line 6 has a pretty good track record for customer service and doing what's right -- in this case, they hobbled a product in a way that causes it to not meet its original specs, and in some cases turned working units into hardly-working units. It's clearly a band-aid fix because they had no better option, and weren't willing or able to recall them. If I were in their shoes, the right thing to do would be to offer customers free or heavily-discounted replacement with a newer revised model, whenever that comes out. You can't just neuter somebody's product they bought for a hundred dollars or more and expect them to say, "oh well." Edit: I'll just add, considering how many consumer products that use Li-Po or Li-Ion batteries have had these kinds of expensive failures over the years, it boggles my mind that built-in temperature sensing that feeds back to the charging circuit still isn't standard in everything that uses them. It won't fix every possible failure, but it would take a big slice out of overcharging-related failures.

Gotcha. I think there are probably better ways to do this if you're hosting from your PC, which would make full use of all the USB I/O Helix has. But, if you want to stick with this routing setup, do the same thing I said above except change your outputs for paths 1-3 to 'USB 1/2' and set path 4 (your click) to either 1/4" or XLR. You will still hear the click through the headphone jack of your Helix, but the audio stream going into your PC via USB won't have it. Also I just checked this and it seems like the relative mix levels going in via USB match what I'm hearing in the headphones. Also, make sure under the Windows Sound settings, under the Recording tab, your Helix is set as the default recording device, and also click Properties, go the the Advanced tab, and make sure "Default Format" is a 2-channel format (eg 2 Channel, 16bit, 48kHz). One trick here is you may need to adjust your USB output level to match your headphone level: adjust the output gains on each of the paths that are going to USB1/2 so that the levels into your pc (& thus fb live) are good, then adjust the headphone volume knob on the Helix to get a good level in your monitoring headphones. One downside to how you're doing this is that your backing track/loop will be mono... you might not care, but just an FYI if you didn't realize.

Hi, I’m not sure if this is the best way, but here’s one way. I’m assuming that your ‘audience mix’ is the 1/4 Outputs on the back of Helix. If it’s actually the 2 XLR ports, then just swap them in what follows: For each of your first 3 paths, set the output to 1/4”. For your 4th path, set the output to XLR. In global settings, make sure the option for ‘Headphones Monitor’ is set to ‘Multi.’ Now your headphones will hear everything, but the audio output on the 1/4” jacks will get everything except path 4. You can adjust your headphone level via the Phones knob, while the big Volume knob controls the level out of the 1/4” jacks. I can’t think of any simple way to adjust the mix of all 4 in your headphones independently of the other output. If you weren’t using all 4 paths already, there might be some clever ways, but as-is I think you’ll have to take it as it is. I’ll think about it and reply again if I come up with something.

Thanks! In that case, OP shouldn't be using direct hardware monitoring at all, or he'll be hearing both the Helix processing alone doubled with the Helix + extra VST processing applied. So, he should set the output block of his Helix to 'USB1-2' and adjust the output gain in that block as needed. Still don't see why an external mixer would be needed...?

I’m trying to follow this thread, but I’m not quite getting the problem. When I use my Helix with a DAW, I send the USB1-2 to a track and arm it for recording, but I turn OFF monitoring on that track within the DAW. If you don’t do that, then you’ll hear the direct signal from your Helix as well as the same thing looped back through your DAW, with some latency too, which makes things sound weird. If things are configured correctly, you adjust the volume of your guitar with the knob on the Helix, because it’s being monitored direct through Helix. You use the mixer faders or master volume in your DAW to control playback volume of individual tracks or recorded material. If you don’t have enough headroom in your DAW to get it up to a usable level with your direct-monitored guitar, then go to the output block of the Helix and dial the gain back a bit. If that’s still not enough options, there is a “USB 1/2 Trim” level adjustment in the Helix too, which will cut or boost the volume from USB 1/2 without affecting your hardware-monitored guitar volume. Lots of ways to make this work—if you need a mixer, you’ve probably got something off in your approach. Also I’m surprised you say the headphone output of the Helix ‘doesn’t have much headroom.’ I have the floor unit, and use a pair of Sennheiser HD600 headphones (which take a fair bit of power to drive), and never need to go above 50% on the volume knob for ear-ringing volume. Are you using low-impedance headphones? If so, that might be the issue.

Yep! This was my suspicion when I asked him what that icon showed. He needs to select that input for path 2A and change it to 'none'.

You don't also have something assigned to the 2A input, right? In other words, your 2A input should only show a little arrow above it, indicating it has path 1 routed to it. It should say 'None' underneath. If you upload a preset here where you're having this issue, I can take a look. It shouldn't be doing what you describe.

Here you go, here's an example for the routing. Note I didn't try to make this sound good, I just added default blocks to show you an example of how you could do it and how you'd route it. Make sure to pay attention to the panning when you split the signal and when you merge it -- that's important to get things blended optimally. Also if you want it to remain stereo after merging, make sure every block after the merge is also stereo. Oh, also, the two different IR options are how many samples it uses for the IR. Basically one is twice as 'precise' as the other, and uses more DSP because of it. I recommend trying each option for your tone, and if you can't hear any difference, use the one that uses less DSP (unless you've got plenty to spare!) Nothing strange about enjoying learning -- that's how you get the most out of your tools! Good luck! example.hlx

You'll have to read the thread above. It's all explained. The reason is to match the modeled pedals as closely as possible, and for the most part it works very well. Because of the way it's been implemented, in rare-ish cases, it causes problems for some people when bypassing certain pedals.

The simplest answer I can give here is, if you want your tone as bright as possible, set the input impedance ('Guitar In-Z') to '1M Ohm' and leave it there for your patch. This will prevent any of the blocks in your chain from changing it, regardless of their bypass state. If you do this, and you use certain pedals like the Arbitrator Fuzz, Industrial Fuzz, or Triangle Fuzz, their tone won't match their real-life counterparts because in real life those pedals have low input impedances. If you're not familiar with how those real pedals sound and behave, you might not care. The full, detailed explanation is well-stated several times earlier in this thread. I don't think I can really summarize it any better than it's already been done, so if you want more info than that, you'll have to carefully read the previous replies. It's all there.

I mean this in the nicest way possible, but... did you even look?... To make up for my snarkiness, here are your links: ;) HX Stomp Driver: https://line6.com/software/readeula.html?rid=9135 HX Edit / HX Update Driver: https://line6.com/software/readeula.html?rid=8830

You could probably use some software cleverness to get close, but the thing to keep in mind is that the input impedance interacts with your guitar's circuitry -- the pickups, the pots, all of it. So it changes not just the tone (like an EQ) but also the way it responds to different knob settings, types of pickups, etc. Trying to model the myriad of possibilities there would be a challenge and probably never really get close to working with your guitar controls the same way you'd expect with the real pedal. Also I think you're already aware, but just to emphasize for others -- the input impedance isn't 'modeled' in the Helix, or the AxeFX--at least not in the sense we'd normally think. It's not a software trick. It has an analog circuit on the guitar input which actually changes the real input impedance to a range of different values, so the tonal differences you hear from varying the input impedance are due to real circuit interactions, not software EQ'ing.

Certain pedals like vintage fuzzes won't sound right or respond as intended if you do. The whole reason this circuit is included is because some pedals get part of their characteristic tone from their low impedance.

Whewww... here comes the can of worms, lol. Brace yourself... Yes, auto sets the impedance correctly based on the first block in your signal path. That's the simple answer. For most people, that's probably what they should be using. The longer answer is that, even if you have that first block bypassed, 'Auto' still uses that first block's input impedance, regardless of what the next effect is. For some pedals, this is how it would work in real-life, but for true bypass pedals, that's not how it would work. Thus a lot of people don't like how it behaves and have asked Line 6 to change it so that 'Auto' uses the first *active* block in the chain to set the impedance, rather than just the first block. If you don't like how auto behaves, you can: (1) override it for the entire preset by changing it away from Auto; (2) control the impedance via snapshots, which doesn't help much for stomp mode usage; (3) link the bypass switch for your first block to a pair of impedance values that it toggles between, depending on whether that first block is active or inactive. None of those 3 totally solve the issue if you want all of the pedals to behave as if they're true-bypass and you'll be turning them on and off with stomp mode within snapshots. Edit... darnit, @rd2rk beat me again... must type faster

Yup. In a VST, if it gets close to the sound and feel of the real pedals, it's all via software modeling and assuming a certain input impedance. With things like the Helix and AxeFX, it's actually a switched analog circuit inside the hardware that changes the input impedance--it's not really modeled, it's actually put into the circuit with your guitar. Depending on VST, some might feeling closer, others might feel less so... but I can say personally that I almost never use VSTs anymore (despite owning almost every one) because I much prefer the feel and response through my Helix. It's a combo of the low latency and increased dynamic range which many interfaces don't match, along with things like the impedance circuit.

Also, you're probably aware, but hissing noise is normal to an extent, especially for high-gain amps, and single-coil pickups will add to that even more because they're more sensitive to r/f interference from everything else in the room. You can use the noise gate in the input section of the Helix to quiet it down, after you've looked at the things silverhead mentioned. Also, keep things like mobile phones away from your cables... they emit tons of noisy R/F and can definitely be heard when they're transmitting. Even laptops, computer monitors etc. can inject a lot of noise in close proximity.

Give these a shot. I changed all the effects blocks after the reverb to stereo. Note that the legacy-category reverbs in Helix are stereo, so to make them sound that way, every block after them in the chain also has to be stereo. That alone gives a wider stereo field by itself. Another trick I added is splitting the signal and adding a delay for a slap-back type of effect, in a separate path from the reverb. If you listen to real amps recorded in rooms instead of isolation boxes, you can often hear reverb as well as a slapback echo. In 'Become the Wild2", I left the delay as a stereo effect, and mixed it back into the signal centered. In 'Become the Wild3,' I made that slapback delay mono, and mixed it back into the main signal panned a bit right and the undelayed signal panned a bit left. The last one, 'Become the Wild4,' I also added a subtle chorus to the right side before the delay, which can give an even more lush sound, depending on what you're going for. Anyway, those should give you some ideas. You can change how dramatic any of these are by adjusting the mix levels, adjusing the delay times (bigger offset between L/R equals wider field, but also be sure to check them with everything collapsed to mono to avoid phasing issues), etc. Think of your chain as the sound would be created in a real-life setup -- once the sound comes out of the amp speaker, a mono source, you get stereo reflections, stereo echoes, etc. Try to set up your paths past that point to mimic what's going on in the real world. Become the Wild3.hlx Become the Wild4.hlx Become the Wild2.hlx

Just downloaded your patch. If it's studio, stereo sounds you want, the first problem I see right away is your signal chain is almost entirely mono blocks. Keep in mind that stereo width is created by the differences between the left and right channels, so to achieve a stereo sound, you'll want to be using stereo effects after your amp blocks. FWIW, every signal route in the Helix paths is made of stereo channels, but certain blocks like amps will always collapse that to mono. So it's important you pay attention to channel splitting, routing the channels through different effects, etc., and also making sure any blocks after the ones that generate the stereo field are *also* stereo blocks, or else they'll combine both channels back into a mono mix. Let me play with your preset just a bit and I'll upload another one in a minute with an example of how to get a full stereo/studio tone.

And just to emphasize, they are modeled that way to match the input impedance of the real-life pedals, which matters a lot for the characteristic sound and feel of certain ones, like fuzzes. It's an analog circuit that re-creates (somewhat) the interaction between your guitar pickups and the pedal. Most computer interface sound cards have high-impedance instrument inputs (usually 1Mohm), so if you're using them with Native, this is a piece of the hardware modeling that won't match.

No problem! Let us know how they work with the Stomp — sound quality, if they’re loud enough, etc. Then the next person that has the question will already have your answer! :)

Just a heads-up, somebody here had the 80 ohm version of those recently and complained about poor sound quality out of the Helix (full size, not Stomp). I looked it up, and with the 12 ohm output impedance of the Helix, you ideally want headphones with an impedance of ~100ohm or greater. *If* the Stomp has the same output impedance as the full size Helix, then the same would apply to it, meaning you’d probably be better off with the 250 ohm version of the DT-990s rather than the 80 ohm. I looked all over to try to find the output impedance of the HX Stomp headphone out, but it doesn’t look like Line 6 published it anywhere, unfortunately. Also, people usually assume that higher impedance headphones will be quieter than lower impedance ones, but that is not always the case. A headphone output that has high impedance can actually be quieter with low-impedance headphones. I’ve seen several posts of people with the HX Stomp complaining they have to turn it all the way up with their 80 or 32 ohm headphones, and I suspect that might mean it has the same 12-ohm output impedance as the full size unit.

Check to make sure your adapter and headphones are TRS (tip-ring-sleeve) only. If either the adapter or headphones have an extra ring for microphone output (very common on modern in-ear headphones for phone use), that can cause issues. On that same note, make sure both your headphones and adapter are stereo plugs. Again, should have three metal sections, the tip, ring, and sleeve.

I’m gonna have to agree with @gunpointmetal here... after reading all your replies, I don’t really see what your point is. If you want to have those ‘pyscho-acoustic’ effects, then run a stereo dual-amp patch through stereo cabs in a room. How is that any different from using two amps and two cabs in the same room? If your point is that a modeler can’t mimic that effect through headphones, well yeah — neither can two real mic’d cabs through headphones. So, like gunpointmetal, I don’t see what your larger point is. Also, FWIW, there are LOTS of recorded albums that have two different amps mic’d and mixed in stereo. If you’re recording, it makes tons of sense to do this in a single patch, like I mentioned above. My best Helix patches for recording and listening in a small room are mostly stereo amps + cabs.