Meiannatee

You can compare it with another tuner. Or just tune the high E to pitch with the Helix and then see how it reports the 12th fret harmonic as horribly sharp even though it should be more or less a perfect octave from the unfretted note. Being the fool that I am, I was intonating my high E string and wondering why it kept showing that it's sharp even though my ears tell me that it's flat. I kept setting the saddle further back and still couldn't right. Only when it was as far back as possible did I get suspicious and checked with another tuner. Lesson learnt: trust your ears!

I think the flubby bass may have something to do with the cab too? A little bit of flub is normal, and depending on your cab, what little there is will be filtered out. However I still pre-EQ most patches to get it tighter. About the level meter thing, I use a free plugin on my DAW, but wish it was conveniently on the unit as well. Considering the duration that a session can stretch to, would be nice to have an easy reliable meter instead of fatigued ears. Another tip: the built in tuner is not accurate at the higher frequencies! I didn't believe the posts about tuner inaccuracy until I tried to do a quick intonation adjustment on my guitar. Someone can verify? Simply tune the high E string to pitch, then compare it to the 12fr harmonic (or fretted note on a correctly intonated guitar). My ears tell me it's in tune with the open E, but Helix says that it's flat.

I think people just want to know what each parameter does. And how to make usable the very weird modulation in some of the new reverbs. I've also yet to try putting it in a parallel chain 100% wet with an EQ block after. Anyone tried?

Yes. To clarify panning: At 100% L pan: L volume 100%, R volume 0% At middle: L volume 100%, R volume 100% At 100% R pan: L volume 0%, R volume 100% All values in between are interpolations of the points above.

It's not a bug. The dual cabs is treated as 1 block in helix. When changing cab 1, you're changing the whole block to a new one. When you change cab 2, you're changing a parameter in the same block determined by your cab1. To have helix treat cab 1 and 2 separately, use 2 single cabs in parallel paths. DSP usage is the same (i think)

Hey, I think it's not that the Helix is scooped. It's because there's a lot more bass and treble in the helix cabs, and maybe some amps. With more top and bottom end, the mids don't stand out as much. I've experienced coming from the Atomic Amplifire. If you want the mids be more up front, it may not to wise to add more mids with EQ. Instead, pull back the highs and lows before the amp (the low/high pass filter block is useful for this). Reason: I assume that you want more mids for overdriven amp tones. Bass distortion = flubby, treble distortion = fizz, mid distortion = sweet. So instead of sending more mids when the bass and treble are already hitting the amp too hard, reduce the lows and highs so that they go through the amp relatively cleanly. Let the amp work on the mids, so that as the mids are distorted and compressed, they level out again with the cleaner bass and treble. End result: tight bottom end, rich overdriven mids, sizzling highs without too much fizz. Another thing to try is to mess with the cab parameters, especially the high/low cut and mic distance. Helix cabs take some time to dial in right. Or you can just use a more aggressively EQ-ed IR. The plus side with the Helix is that the treble and bass is there if you need it, e.g. for glassy cleans. Happy experimenting!

Actually I read somewhere that Helix implements a make-up delay between stereo paths/channels to eliminate phasing issues. Not sure if it's extensively tested, but I'm using XLR left for live too, and no issues so far. However, for stereo delays, you are intentionally adding "phase difference" between L&R. Usually, the L&R signals are separated far enough to avoid serious phase cancellation, and results in a somewhat pleasant chorusing effect when summed to mono. If that's not what you want, try messing with the delay parameters, particularly turning off pre-delay and reducing mod depth. If that still doesn't work, you might have to make duplicate presets with mono delays. From experience, the other effects seem to sum to mono perfectly fine, so that's a bit better than reworking the entire preset to mono!

Actually, if you like listening to music through headphones, and like your guitar to sound like that, you'll like the sound of the Helix through headphones (assuming you dialed it in the way you like it). Not sure why anyone needs a ton of low end for guitar tone. I usually highpass at 80-100Hz anyways. Keeps my guitar out of the way of the bass and kick drum. One tip is to add a room reverb to dry presets, at a low mix (15-30%, decay to taste). Dry tones sound weird on headphones because you're not hearing the natural reverb of the room you are in, resulting in a sound which is devoid of a sense of space. A little reverb can "open up" the perceived space to a size that our human ears are used to hearing. Whatever the case, I think anyone here will tell you that listening through headphones at a normal level is definitely better than turning down FRFR speakers to a volume as loud as an unplugged electric guitar to not wake your kids.

SNAPSHOTSTo be clear, the Snapshots for your Preset are still there in Preset/Stomp. They are just hidden. As long as you don't mess with them, they won't get in your way now. Every time you change Presets, it will load the Snapshot you were in during your last save. If you accidentally switched Snapshots, simply switch back to the intended one and hit save to revert to that Snapshot as default when you load that Preset. UPDATE Like DD has said, if you want to update, follow the instructions word for word and don't deviate, then there should be no problem. Whether it's worth it or not is up to you. Key features: - 2 new amps (the Lonestar is one of my new fav) - New HX reverbs - Added Legacy fx from older Line6 products - New category of FX for all the Legacy stuff. E.g. Mono, Stereo, Legacy. Old reverbs are legacy, so moved to Legacy folder - Read full changelog for more details

You're still confusing snapshots with presets.Before, there were NO SNAPSHOTS. Only presets. If you change a block in preset slot A1, it will not affect slot A2. Snapshot is a "layer" within a preset. You don't have use it at all. Just stick with presets if you don't mind the switching gap. Set footswitch mode to preset mode. Keep us updated on it. Good luck!

When editing from a computer: 1) Right-click on preset > copy 2) Navigate to target slot. Right-click > paste 3) Repeat for the other preset

This. TLDR: DEFINITIONS Preset = Package containing all blocks (amp, cab, fx, etc.) Snapshot = settings WITHIN A PRESET (e.g. on/off a block, modify parameter) Cannot change/add/remove block with snapshots (e.g. cannot change amp from Deluxe to Plexi). Can only play with what you have within the preset. SWITCHING TIME Switching presets: have gap time. Switching between snapshots IN THE SAME PRESET: gapless FOOTSWITCH MODES Preset mode: switch between presets (footswitch 2,3,4,5,8,9,10,11) Stomp mode: toggle blocks on/off, toggle parameter changes (WITHIN THE SAME PRESET) Snapshot foot: switch between snapshots (WITHIN THE SAME PRESET) Composite modes (e.g. preset/stomp, snap/stomp): mixture of the above (e.g. footswitch 2,3,4,5 to change preset, 8,9,10,11 to toggle stomp) For more info, read DunedinDragon's post. Edit: STEPS TO TAKE (Option 1) 1) Footswitch mode In global settings, go to preset mode (or any composite mode with preset in it) 2) Put your custom preset in slot A of any bank in a setlist (e.g. Setlist User1 > Slot 5A 3) Copy Rockabilly preset to slot B of the same bank (e.g. Slot 5B) STEPS TO TAKE (Option 2) 1) Footswitch mode This option is for non-preset modes (e.g. snapshot, stomp) 2) Copy both presets to sequentially adjacent slots (e.g. 4D&5A) 3) Use leftmost footswitches to scroll presets up (e.g. from 4D to 5A) and down (e.g. from 5A back to 4D) 4) The other footswitches can be used to change stomp/snapshot states WITHIN THE SAME PRESET. Changing presets will load the last saved snap/stomp states within the NEW PRESET.

As long as you're within headroom limits, there's no difference. Impulse response convolution is a linear process. If you want a pushed speaker sound, you have to use an IR that was captured with a pushed speaker. For a lot of applications, there's absolutely no problem with mimicking analog signal chains and only balancing the final output volume. The problems arise when the patches get complicated. Say there's a snapshot where the amp and IR are turned off, acoustic IR turned on, for acoustic sound with piezo. Then you gotta use snapshots to control levels. Add more complicated switching in the other snapshots and things get messy fast. Then if I want to manually bypass a block in a live situation, there's a high chance for the levels to jump suddenly and take people's heads off. Good luck dealing with the soundguy after that...

I'm also one of the "weird people" that go for unity gain on every block such that the total patch loudness is the same as a blank patch. Except for optional solo boosts of course. Strange considering a benefit of digital being consistent processing at any level (within the device's operating range). Unfortunately, for dynamic blocks like amps, dirt boxes and compressors, there is no way to set a single default level for every guitar. Pickups can have very different outputs. E.g. An amp is set to unity gain for a vintage output guitar. Bypass lvl = active lvl = reference 0dB. Switching to a hotter output guitar (say +12dB), bypass lvl = +12dB. But with the amp active, there is more clipping than before. Output lvl compressed to say +6dB. Now bypass lvl > active lvl. No longer at unity gain. Short story, you can only adjust to unity gain for 1 guitar at 1 pickup and volume knob position.

It's best not to draw conclusions without verifying first with another pair of headphones or monitors. I'm using the Ath-M30s and they sound as good as my monitors. If the problem goes away on the comparison setup, then there is likely a defect on your M20s. I've yet to come across a headphone or IEM over 30 bucks that completely turns an otherwise decent tone to $hit unless it's broken.

1. Dry signal (stereo) > split paths 2. Path1 > stereo to mix ctrl 3. Path2 > sum to mono > stereo processing > mix ctrl 4. Mix ctrl blends 2 stereo signals Edit: I hope this is clear. Tried to draw an alphanumeric diagram, failed miserably... Reason for sum to mono before stereo effect: Reverbs are meant to replicate reflections in a certain space. If you are in the middle of the space and facing a stereo sound source, the left direct (dry) sound reaches your right ear earlier, and vice versa = fully stereo. Now, to simplify things, we'll use a ping pong model. 1st reflection: The left source hits the left wall and reaches the left ear first. 2nd reflection: The reflection from the left wall carries on to bounce off the right wall before reaching the right ear first this time > inverted stereo image. The result is a diluted stereo image of the source. Add more complex reflections and the stereo field of the source is lost after a few reflections. Now explain the stereo processing: Say the left wall is a soft material that attenuates some of the higher frequencies, while the right wall is hard and attenuates less. The direct signal interacts with both surfaces just as much, since they bounce left to right to left (i.e. the walls impart their "flavor" to both left and right signals) However, each new reflection off the left wall will reach the left ear first, whether it is the 1st reflection from the left source, the 2nd reflection from the right source, etc. I.e. the left wall is constantly applying "processing" and the latest iteration of this processing always hit the left ear first. Therefore, at 100% mix, even though the original stereo image is lost, the stereo image of the reverb effects are still clear. Those effects are constantly "refreshed" with each reverb iteration.

Listen to 04.Illumo by Toska #np on #SoundCloudhttps://soundcloud.com/official-toska/04illumo

I'm getting it too. By the way, where can I find some explanation on what parameter does what in the new reverbs? When I mess around with them, the uneven panning sometimes seem to go away as well, but I have no idea why.

This. If you are staying within the linear limits (not too loud that you're clipping, neither too soft that there is signal degradation or gating), there should be no difference. Since the OP is asking about the EQ processes within the same block, assuming the above prereq is true, there should be negligible difference. You can test it out with this signal chain: EQ1>EQ2>EQ1-duplicate Set footswitch to toggle EQ1 off when EQ1-duplicate is toggled on. In the case where there's a non-linear block in between, e.g. amp, distortion pedal, noise gate, etc, then see zooey's reply.

Yea, I didn't realize how much I missed the particle verb after giving away my 500x! Awesome reverb

Yes the pad-y sound, not the piano. Technically it's not functioning as a pad since it's playing the song hook, more like a gentle lead. If you know subtractive synthesis, it sounds like the patch is a sine wave + octave up and down, lowpass filter, mild reverb. I can't figure out how to get a tracking sine wave like tone on the Helix.

I know it's an old thread, but I just got the Helix and tried to replicate the soft synth in the beginning of this vid: For those who have used a HD500 before, I could get something close with Pitch Block > Tron Up/Down > Low+High Pass EQ > some mod > Reverb On the Helix, the note generators sound good but unfortunately don't track. And the synth that does track has the annoying envelope filter on the attack that I can't seem to dial out. Anyone had any success yet?? Even the simplest tone of monophonic sine/saw with tracking? At least then there'll be a starting point to build upon.

Omitting the effects, if we're talking just amps:Single amp: 70*37*16 = 41440 Dual amps: 41440c2 = 858,616,080

So I tried to take a bite out of my LT today, and I can safely say that it's tougher than any overcooked steak I've ever had 😉

So I've read that using dual stock Helix cabs/dual mics, you can get a much fuller sound, comparable to a good 3rd party IR. Single amp, 2 cabs panned center and blended. The problem is, with 37 cabs and 16 mics, that's 174936 unique combinations. Then there's also mic distance which has quite a significant effect. Any vets out there care to share your favorite dual cabs as a starting point for newbies like me? Would be nice if you could specify the following: -cab model -mic model -mic distance -mix % of cab1/cab2 I can think of a few advantages of using the stock cabs, although I already have quite a lot of IRs: - simplicity of not having to manage IRs in specific order - able to do fine tweaks - easier to share presets Much thanks in advance!