kylotan

Ah, maybe I was misremembering the range of the Output block. I know some blocks only offer +6dB so I thought that was the case with Output. I think this will be my plan - to push the Output block higher, and aim to keep the volume dial somewhere between 50% and 75%. (If it seems like I'm being lazy and not checking for myself, it's because my home computer doesn't let me post replies to this thread, strangely. Right now I'm posting from work. So when I am at the Helix, I can't post, and vice versa.)

Thanks for the suggestion, but I have many problems with it: I don't always have space for 2 extra blocks, I'd have to remember to do this on every preset I use, I'm already using all my footswitches, I'd prefer more granularity than 3dB anyway, and I'd prefer to be using the same mechanism for turning it up as for turning it down. I appreciate being told that the large dial is essentially an attenuator, and I can try and work with that, but it's a poor design if you ask me. The manual even expects you to be playing with it somewhere in the middle range (as implied by "slowly turn up the volume knob", presumably stopping when it's 'loud enough'), so it is really unfortunate that as soon as I plug it into a line-level device I'm supposed to now have it at 100% with no scope for pushing it a little harder. I will probably experiment with the global EQ later (assuming that has an overall gain parameter - the manual doesn't show one), because I guess that is probably as good as it is likely to get.

Much as I appreciate this thread being bumped, can the mods perhaps do something about this optical-trem obsession? PeterHamm: when you say "there is no reason to have the big volume control at anything but max for this use, and then adjust gains down if necessary in the output block", I think you're missing my overall request, elaborated on in the 14th post above, which is basically - I want an analog dial on the floorboard that lets me move volume up or down, according to the situation. If it's already at full, I can't move it up. Right now, I'm considering pushing it up to more like 75% and removing/reducing the level boosts in the output/gain blocks to compensate, so I still have some degree of freedom in both directions on the volume dial.

I don't have a master post-loop volume on the amp. The current setup is guitar -> helix -> amp loop return. Sometimes there's a mic on the cab going to the PA, sometimes there's not. There are 2 major situations in which I play, and I want to use the same patch(es) for these: Rehearsal room. I dial in the volume, then maybe adjust up or down depending on how things are sounding that week, based on where people are standing, whether we've got everyone in this time, where the cabs are, etc. Gigs. I dial in a volume that I think is right, which may be very different from what works in the rehearsal room. And then the sound person might ask me to increase or decrease it to his/her taste. That's why I want my default volume to be broadly in the middle of the dial, so I have headroom either side to make these adjustments. I don't want to have the dial at full and then have the sound engineer telling me I need to be a lot louder, or for me to discover I'm too quiet during soundcheck, and to then have to fiddle around with the Global EQ which, after being set, is basically out of sight and out of mind. (And to Trolley; no, there is no cab block involved. This is just a preamp block and various effects which I calibrated to broadly unity gain levels.)

I need the volume knob to have some range of motion both upwards and downwards so that I can adjust the level according to the situation. If I calibrate my patches with the assumption that the volume dial is on full, then any time I might end up being too quiet (e.g. standing further from my amp, or working with a weaker power amp than usual), I then have to go in and edit all my patches, which is not practical in a live situation. If the volume dial is solely an attenuator, is there any guide to the relative levels of having that dial at different points? e.g. Is 12'o'clock -6dB, or -12dB, or... ? And is there any other way to create a significant global signal boost? Ideally only for the 1/4 outputs? I might be misremembering but I think the output block only goes up as far as +6dB. EDIT: I mean, this sounds conceptually like a job for the Global EQ... but that is buried in a menu, and the volume dial is right there on the top of the unit. It makes sense to be able to adjust this via the large dial.

Trying to reply from a different computer since the forum wouldn't let me post from home: My 1/4" output is definitely set to line. And my total signal boost is actually +8dB; +3dB on a Gain block, and +5dB on the Output block itself. This seems like quite a lot! If I didn't have this then I'd need to turn the Helix volume up to almost full just to get the same loudness levels as having my regular pre-amp on 3 or 4 out of 10.

I play into my Yamaha HS-7s at home, with a cab sim engaged. These speakers have quite a flat tone so most of the flavour comes from the cab sim, which I try to get to simulate my typical live cab.

Some interfaces will route the Line In values back out to the output by default. I have to mute my Line In on my Scarlett 6i6 for this reason. Additionally, be careful that you don't have a parallel loop in the signal chain if you edited a preset yourself. When I made my first sound I forgot that the signal travels down all paths, rather than just one (as in older Boss products), so I was hearing the clean signal alongside the amped signal.

No idea, but 1/4 output should be line level anyway, right? If the amp expects instrument level and it gets line level then it would be louder than I might want, rather than quieter.

Simple question: is anyone else finding the output level of the Helix a little low? I have a fairly high gain preamp-only patch - overdrive, PV Panama - with the master up at around 8 or so - and I send it from the 1/4 mono out into the return of my 120 watt tube amp. I also added a Gain block in there with +3dB or similar. And what I find is that I need the volume dial to be around half-way to get a decent level for rehearsal. If I use the amp itself and not the Helix, I would rarely need to push the pre-amp past 3 or 4 out of 10 to get the same sort of levels. So it seems like I have a lot more headroom on the real preamp, but with the Helix I had to almost max out the master volume and add an extra gain block to get something where I have reasonable headroom using the volume dial. It's not a problem, as such; with this in place, I can gig with the volume dial somewhere between 30%-70% depending on the venue. But I'm surprised that I had to push the levels so hard in the effects blocks to get to that sweet spot. Is this something anyone else is noticing?

In my experience, I wouldn't try palm mutes with EMG 81s into a 5150 without an overdrive to filter the bottom end. It's pretty much the opposite of tight, with massive volume increases relative to typical playing. I can't comment on "warmth vs tinniness" though; I deliberately like a bit of 'bite' to my sound and the overdrive adds that.

I find the UI very good in most respects, if a little awkward in others (snapshot naming, 'moving' blocks between paths, having to use both joysticks to switch between presets in different setlists). Agreed that a bit more information about the parameters would be nice but I don't know where it would be displayed and still be practical. The PC editor is really awkward - dragging a block seems to require military precision. And it doesn't offer much more information than the on-device editor anyway.

Some of the GT-10 amp sims sound no better than my old Zoom pedal from 2002, complete with swirly digital quantization noise. I'm glad to be saying goodbye to it!

That's what I was saying in the 2nd paragraph in the last post - when using the Preamp models, I had the Send output level at the maximum of +6dB and that was too quiet. Could be that I needed to supply Line level instead of Instrument level to that particular amp, but I'm not sure.

I just wanted to come back to this thread and thank you all for the comments and advice. I had my first rehearsal with my new Helix last night and apart from teething problems trying to get the right levels into my amp's effects loop, everything worked perfectly and sounded great. Rather than having my amp in the loop, I just used the modelled one, and it sounds good enough that I might switch to just using a power amp and save myself carrying so much around at gigs.

I think I didn't make myself very clear, so I'm sorry about that. I know what the different line/instrument levels mean relative to each other, and I know what gain staging is (I mixed and mastered 3 of my own albums so I had ample time to learn!) It's just that the ways to significantly alter the gain in this case, such as things like the choice between amp sim or pre-amp sim, or whether I drive sufficient signal into a power amp to start hearing saturation, or maybe sending too much... those start to potentially affect the tone. I wanted to hear from any Helix owners who'd already attempted this, to hopefully save me some experimentation. However, I did manage to try some experimentation tonight; not with the Bugera, but with an Orange amp. (Not sure which model, exactly.) It turned out that the Preamp only models were too quiet, even with their master/channel volumes up between 7 and 10 and the maximum of +6dB on the Send block. I had an identical preset with the Preamps switched to Amps, and they were loud enough. I eventually had them running with the Send at -3dB. It's possible that the amp I was using had a Line level loop rather than a Instrument level, so maybe I just needed to switch Send 1 from Instrument to Line and stick with the Preamp models, but in the end I didn't notice any significant colouration difference between the 2 settings. (Though that's hard to judge with the levels being quite different.)

I don't have a problem with the routing, I have a problem with knowing what the levels should be. (And I don't want separate paths anyway. I want to be using as close to an identical set up for both environments as possible.)

It seems to me that if you have 2 signals in parallel - for example, a stereo signal path - and somehow only applied the IR to one of them, then a phasing effect becomes a possibility. Is that a potential culprit here?

Sorry for the speculative nature of this question but I don't have my amp at home and can't experiment for myself easily. I've created some patches with my new Helix where the intention is that I'll run it into the return of the effects loop on my Bugera 6262 (basically a cheaper Peavey 6505 clone, for the uninitiated). The chain looks like this: guitar in -> OD -> PV Panama Amp -> chorus -> reverb -> Send 1 -> Cab sim -> 1/4 out The idea is that I can test the tones at home with headphones and hear it with the cab sim, but in a live setting, I can just take a cable from Send 1 to the Return of my amp, and hopefully the resulting sound will be much the same (slight cab differences aside). The complicating factors are these: 1) The effects loop on the amp presumably leads into the power amp. So I'd potentially be running a simulated power amp and a real power amp. I know I can swap out the PV Panama Amp for a PreAmp model, but this makes it a bit harder to test adequately at home because the levels from it are too low. (Makes no difference to modulation or reverb I expect, but it's difficult to know whether I'm gain staging correctly). 2) The Send can be Instrument or Line level, and I believe my effects loop requires an Instrument level signal. But it's not clear whether the Instrument level is an attenuated version of what goes in, or whether the Line level is amplified, or whether I am just misunderstanding the concept entirely. I feel like I need to understand this in order to get the gain staging right. Ignore this: I guess that as long as I manage to match the setting to the amp, it'll work out about the same. 3) If I bypass a cab model on a normal amp -> whatever -> cab chain, the eventual output is much hotter and appears to be clipping, even if it doesn't clip at all with the cab model in place. I think I saw this with some of the factory presets too. This makes it difficult to know whether I'm gain staging things correctly in a situation like mine where I have a send before the cab model. To compensate I set my Send block to output at -12dB, but I'm not sure if this is the right approach to the problem. So, I'd like to understand all the above a bit more, but most importantly - what are the recommended settings for sending the output of amp models into the effects loop of a Bugera 6262 or Peavey 6505, or similar? (I'm not using the 4CM method for now - the Bugera distortion is great but if the PV Panama sim is good enough then I won't need it, or indeed the extra complication of the additional loop.)

Ah yes, I didn't see the looper earlier as I was viewing the image on my phone. If I needed delay blocks for ambient tones I could probably use a different patch and ditch some of the other blocks so it sounds like I shouldn't have any trouble. Thanks, I suspect that answers all my questions. Now, just to wait for pay day...

That looks perfect, thank you! The split/merge example is just how I've handled it with other products which had an explicit channel-switching concept, but the snapshot system is more powerful. The only questions that remain are whether a looper can fit in there as well, and whether 3 delays in one patch would be a problem. (The latter seems unlikely as they're pretty low on DSP requirements, right?)

Yup, I don't strictly have a need for cabs or mics because mostly I'm either using the 4 cable method or going directly into the return of the amp's loop. (Mostly likely both within the same patch, toggling the send block which goes to the physical amp's preamp.) It would be nice to be able to have cabs/mics on there as well, in case I feel like doing some direct recording at home, but I can do without that and use cab sim software instead. For clarity, in a a more traditional channel switching or stompbox type of design, I would have considered a routing a bit like this: Input block -> A/B Split -> Overdrive -> Comp or EQ -> Send/Return to amp -> Chorus -> Delay -> A/B Merge -> Reverb -> Volume pedal -> Out \-> Clean Amp sim -> Compressor -> Chorus / That has 2 separate choruses, but obviously the merge could be brought earlier to share the chorus, at the expense of having to toggle the delay off separately - but I guess with snapshots, that wouldn't be a problem as I'd just have a snapshot with that block turned off. I don't anticipate wanting more than 2 amps (one clean, one high gain) and 1 reverb (probably hall, cave, or particle) in one preset. I do anticipate perhaps having 2 or 3 delays for the weird stuff however. I'd be very grateful for anyone who could go through and mock up these patches, just so that I can buy with confidence.

I'm considering getting a Helix, after skipping the HD500 and Firehawk as they couldn't fully replace my ageing GT-10. I've been waiting for something with snapshots because patch switching time is a big constraint for me, so I think Helix is going to be what I need - but first, I need to be sure it can handle the FX chains I have in mind. They don't seem too demanding in theory, and the examples in the manual suggest I have nothing to worry about, but I know the Helix has some high quality modelling which means some effects might be quite expensive in DSP terms. My main patch would need to support this configuration: Snapshot 1: noise gate -> tube screamer -> compressor, EQ, or boost -> send/return (for 4 cable method) -> chorus -> delay -> reverb -> volume pedal -> out Snapshot 2: clean pre-amp sim (e.g. Jazz Rivet 120 or US Double Nrm) -> compressor -> chorus -> reverb -> volume pedal -> out (I'd probably have a 3rd snapshot which is Snapshot 1 minus the compressor, chorus, delay, and reverb, but it would be a pure subset of Snapshot 1 and therefore not an issue, I'd guess.) Is this feasible, all in one preset? (I'd be happy to share the chorus and reverb blocks if necessary.) How about if I wanted to swap out the Send/Return block for the ANGL Meteor or PV Panama model, meaning I could skip the loop and plug directly into my amp's loop return? And could I have all this, plus the looper block in the same patch? (I don't really have a need for cab sims or IRs, which might help.) Finally, I'd also like to be able to use some more ambient patches, eg.: Elephant Man Delay -> Hall Reverb -> Vermin Dist -> PV Panama -> EQ -> out or: Elephant Man Delay -> Elephant Man Delay -> Vermin Dist (would prefer a Metalzone, but...) -> Spring Reverb -> Vintage Digital delay -> out Ideally with the delays and reverbs all in stereo. Are these possible?

I just got this today. It refused to save to the D: drive. I tried again with the C: drive and it was okay.

Those colour numbers are basically standard 24-bit computer colour values. So, if you know how to generate hex RGB codes for web design, you can generate such a code, convert it to decimal, and probably use it in here.