SaschaFranck

How would you know?

But you can do the same with the big volume knob (as in assigning it to just one output type). Personally, I'd rather keep the Global EQ available for other tasks - yeah, usually it's set up to only work on the TRS outs, but at one gig I had to route it to the XLR out feeding a cheesy and extremely boomy PA, to adress some unpleasant low end rumble.

Yeah, fine - but these are two different issues (even if they're somewhat connected). First thing I always do is to keep my levels kinda "balanced" so I'm getting a healthy signal into my DAW without clipping, even when boosting the overall level a bit (such as for leads). Ever since 24bit recordings that's no issue anymore as we're not losing much (if noticeable at all) dynamic range. Whatever, DAW or not, the level inside the Helix should never clip on the outputs (and a DAWs input meter is a fine tool to check that - perhaps somewhat better than the Helix output LED chains, but I find them to work quite fine, just that you have to select the output block to see details). The level you're feeding your FRFR monitor with shouldn't have much to do with that (unless you're feeding it via SPDIF, which usually isn't the case). That's what the big volume knob is for. And as far as the input level accepted by whatever FRFR monitor goes, just fool around with it. Use a completely clean, very dynamic patch and listen whether there's any clippings. Personally, I usually set the big volume knob to noon and then adjust my monitor(s) to be roughly as loud as I may need them. I can then easily do finetunings straight on the Helix. Some people may now argue that you will perhaps get a better s/n ration when turning the Helix volume all the way up, but I highly doubt this will make a really noticeable difference, at least when using more or less decent monitors. Of course, in case they start to hiss at higher power amp levels (which the Blueamps ones shouldn't, they're of pretty great quality), you might want to turn the Helix output up as much as possible, but usually that shouldn't be required.

Anything but true peak is completely irrelevant as long as we're talking DAW input levels. And your DAW should provide you with a decent enough input metering allowing you to tell whether the input is clipping. Each and every DAW I know of does that.

Upvoted!

But none of all this affects the input signal at all. Yes, I know you can apply effects and print them into the file while recording (by now, pretty much all DAWs allow for that, even if it's rarely useful), but what you're recording in this case isn't the input signal of your DAW but a signal that is already manipulated by your DAW. And I can't think of any DAW which would do this by default - in fact, you usually have to go through at least some smaller hoops to actually achieve this, the default always being that you record a signal in the same shape as it hits the DAW input. Which in this case would be the USB out of the Helix. Also, when you set up your recording levels, the indicators in your DAW usually show you the signal as it comes in. Anyway, I'm really not trying to split hairs here, all I'm saying is that it's not the likeliest scenario that you would accidentally create a signal routing in which the signal would be boosted by whatever DAW-internal tools. Along these lines, with any "empty", "default" or "template XYZ" project file provided by your DAW maker, it's extremely likely that these will a) show your DAWs input levels and b) not further manipulate the signal before it gets recorded. At least none of the DAWs I'm familiar with comes with any such a template. What I'm saying is: By default, each and every DAW would allow you to properly monitor the level of the signal sent from the Helix USB paths. Anything else could/should be done after making sure that these levels are ok.

Interesting. The Floor does.

How would there be anything going on in the DAW? There's no input level adjustment available, so in case the Helix levels are fine, the DAW input should be, too. It can turn red. Just insert a bunch of EQs and boost them all to their maximum. There's tons of headroom, though.

But that isn't relevant at all. You can't magically avoid digital clipping that way, it's gotta be done on the way in. But then, in this case, it actually doesn't even matter. In case the signal sent to the DAW from the Helix via USB is too hot, it's too hot inside the Helix already. The reason likely being a questionable boost somewhere in the signal chain - which is preceisely, what we're dealing with here... This is the crucial question! All members of the Helix family should provide plenty of analog output level with the internal level structure kept somewhat "neutral". Sure, when you boost an amp models input, using plenty dBs is as fine as in the analog domain, but others than that, it's likely a pretty wise decision to keep the overal level of your signal path at least somewhat close to a neutral pass through (no blocks inserted, unitiy gain on all output blocks). The combination of amp and cab blocks will likely raise that level to some extent, which is still fine, even leaving plenty of headroom, but I wouldn't go much higher than that and always crosscheck my patches against one with just an amp/cab chain set to the defaults. In case you go for that approach, the Helix' USB out levels usually even tend to be more on the weaker side of things (I'd guess the rather low-ish levels of the default settings are intentionally chosen by Line 6 to avoid any kind of digital clipping anywhere in the signal chain, especially at the output converter stage), which is why I usually split my path into separate monitoring and record outs, so I can adjust my USB recording levels on a then separated USB output block (I may elaborate on the additional advances of that approach in a decidated thread...). Fwiw, at first, I tried to adjust my overall patch levels to achieve a decent USB recording level, but with a little bit of afterthought, I stepped back from that approach. When recording, I want a USB level as close to 0dB as possible - not exactly required for sonic quality reasons, as the signal stays in the digital domain, but I want a) clearly visible waveforms without any further actions inside my DAW and b) levels I'm used to work with, so potentially following dynamic processors will be fed with a "common sense" level. But: Achieving this solid recording level by raising my overall patch level might become counterproductive because IMO it's a wise decision to keep the levels a tad more on the low-ish side before hitting the output converters. I might run into a situation requiring a volume boost of a patch (for musical reasons), which would then cause clipping on patches with not enough headroom before things hit the converters in case I started with a pretty hot leveled patch already. As a result of that, I started splitting my signal on my dedicated recording patches. Fwiw, IMO it'd be a nice thing to have separate internal levels for the USB outputs without splitting the signal. Should be pretty trivial to adress from a programming POV and would defenitely make life easier here and there. Especially the DI guitar level (on USB 7/8) is pretty weak in case you don't use high output PUs and I always find myself having to boost it inside my DAW to deliver a proper input level for any other amp sims but Helix Native (which, at its default settings, likes a somewhat weaker signal)

How can you set the input level of your DAW when running into it via USB?

- Shut down HX Edit. - If you are on Windows, reboot or open task manager to kill the HX Edit service (which will still be running in the background for no good reasons...). - Download the update as individual flash files. - Open the Line 6 Updater. - Perform the update there, using the downloaded flash files.

I wanted to post this quite a while ago already but only got reminded of it right now through a post on TGP. Fwiw: I forgot about posting it because basically my way to convert stereo to mono and the other way around is to simply use the stereo versions (as long as CPU power allows for it) and add some mono block (gain uses the least ressources) behind it. On a sidenote: That's also a great idea if you want to check your patches for mono compatibility through headphones because unlike what's done when using just the left TRS/XLR outs, namely merging L/R into a mono signal, this isn't an option on the headphone out (I actually wish there was such an option, should be super trivial to program). Anyhow, here's how to do it for individual blocks (note: this will only work in case the blocks are actually providing mono and stereo operation, which some don't. Amps and such are always mono, the legacy reverb/delays are always stereo, just as some examples). - Export the patch to your computer. - Open the file with a plain text editor (any should do). - In the text file, search for "stereo". You may find some lines looking like this: "@model" : "HD2_DelayModChorusEcho", "@stereo" : true, - Replace the word "true" with "false" and save the file. - Load the file via HX Edit, done. Ok, why would you do this? Easy: When you change any block from stereo to mono (or vice versa), the settings will be lost and replaced with the block's default settings. All snapshot controls will be lost, too. With the text file "hack", these will be preserved. Maybe that's useful to some of you. Fwiw: This is something that should be adressed by Line 6. If all that is required to change a block from stereo to mono while retaining its settings is a such a simple modification, this should be absolutely trivial to implement straight in HX Edit. Fwiw #2: For me personally, preserved settings would make even more sense when changing a block from "Amp+Cab" to just "Amp". In fact, that's something I often do, as I like to start out with A+C but often find myself replacing the cab with an IR or using the same cab for two different amps. I may have a look whether that is "hackable" through a text editor as well. But we shouldn't have to go through such hoops.

Won't be able to exactly help you here, but does your DAW of choice (if you have one) support the mic in of the aggregate device? In that case, you could possibly work around that. And fwiw, the line in of your computer doesn't work, either? Because that's something I could possibly check as well (my Mac Pro doesn't have an onboard mic).

I higly second that. Pretty much similar to the Input Pad but with more attenuation options (several 1dB steps would likely be sufficient, something such as 10). Ideally, you could switch between two settings through a footswitch, in case using the same presets but guitars with different output level is your game (it surely is mine). I'm even thinking of rebuilding an old little (active) input booster/attenuator that I once had on a pedalboard. Got that because at the time I was often bringing my Ibanez 335 clone to gigs - which has insanely high output (for reasons unknown to mankind, especially given that it's from the early 70s) but otherwise sounded great through pretty much all of my standard sounds. So I simply attenuated its output with that little box.

Personally, I have a gain block as the very first thing in my chain. I do then create my patches with my main guitar(s), in general that's 2 medium output guitars. When using other guitars with noticeably different output, I usually adjust the gain block so the gain staging is at least sort of identical. Very often, this allows me to use the same patches with all sorts of guitars, just that sometimes I may want to take advance of whatever characteristics and hence manipulate amp settings and the likes. For recording duties, I have a bunch of patches tailored for individual guitars, but they're not much.

Nah, you need to be connected to your account when you load them into your Helix, so it can be checked whether you're a legitimate owner. Once inside the Helix, they can be used just as any other IR. Possibly not exactly too comfortable for some people, though. Until not too long ago, my main music refuge had zero options to connect to the net (no mobile internet at all, either), so anything requiring online access had me haul around my Mac Pro. That's how I learned to loathe machine-based online activation. Imagine my happiness when installing Logic, which comes in at a whooping 80GB that you can only download onto the target machine (unless you have a second suitable Mac). For plan authorizations I was always just moving up a floor to some office where my mobile net worked, but that couldn't cover 80GB (it's capped at 10GB/month. quite the usual thing here in communication wasteland), so I had to get the entire shebang home with me. Grrr.

From all I know (which is just based on some observations, so don't necessarily trust me on this...) they're kinda encrypted IRs (I think something similar exists for commercial patches), which likely require some kind of extra authoring software, possibly only available to officially approved Line 6 marketplace vendors. The idea behind them most likely being some kind of better protection against copyright infringement.

Hm, if there were gigs, I'd be tempted to try this thing out. Too bad there's still no gigs (and likely none to come in until open air is possible again, unless the dreaded C19 thing would magically dissapear). So maybe next year...

Well, as said, an open E6 string is kicking in at 82.5Hz. Use a Baritone and you may run into severe limitations already - obviously depending on the slope of the rolloff below 90Hz. Same here, I was especially looking for some stereo cabs, kinda like just in case because all that Corona business had me getting back into quite some stereo love affairs (ever since some crazy 90s wet/dry/wet rack experiments I have turned into a "live = strictly mono!" guy) and a single 2x12 cab is a lot easier to deal with onstage than 2 separate monitors. So yes, it's defenitely very new, not even available in Germany (home of Thomann) yet. I doubt there'll be much first hand reports at this point in time.

Yeah, I'm aware of that, but still, 90Hz seems to be quite high, especially compared to other contenders. Even my lowly Alto TS310 goes down to 50Hz (well, they even say 48), which, given my tests, seems to be pretty much correct, so that's almost an octave already. I'm not saying this will make it unusable, but given the specs, it might rule out some use scenarios. Of course, in case the rolloff below 90Hz isn't too steep, one might not even notice in daily life. Guess only a thorough hands on testride would reveal whether it's really a crucial limitation.

In case the specs shown at Thomann aren't wrong, this might not be a decent thing. Says it's going down to 90Hz. Now, the E6 string on a guitar in standard tuning is at 82.5Hz. Might not be too bad because the specs usually only decribe the range inside which the speaker will work linear, so frequencies below won't be completely lost and things may still be sufficient for any typical electric guitar sounds (real guitar cabs usually cutt the lows around that frequency range, too). But then, in case I'm using a modeler, I might as well use less typical sounds, such as a baritone guitar not running into a guitar cab. Or even a bass here and there. And I could very well imagine this cab to fall pretty short in any such cases. Incredible price, though, in case you only need typical sounds, it might be worth a consideration - but it might as well just be too cheap to deliver decent quality.

Fwiw, 200/400/600 IMO possibly isn't the best idea for this. The delays being multiples of each other will possibly generate note doublings which would stand out too much.

Hah - I never use that function myself (and no, it's no more "incorrectly" but just as valid as the other option, from all I can tell). Yet, it's pretty strange that it doesn't work on that one region. I just tried on various regions and it always worked. Ok, that got me thinking and testing (writing this 5 minutes after writing the previous paragraph). Now: You seem to have stumbled over a really strange bug in Logic. I took one region and lowered the gain (destructively) by several dB, so it was very low leveled, yet still audible. And - lo and behold! - all of a sudden the normalizing function available the way you described *doesn't* work anymore! To me, this very clearly is a Logic bug. In case I find the time I may try to narrow down the "can't normalize anymore" threshold and report it to Apple. Seriously, that's a pretty strange bug as we're talking about something that should just work under all circumstances. Anyway, you didn't do anything wrong, it's clearly not your fault at all.

Fwiw, that is a *very* weak recorded signal to be sure. I'd really think about other ways (as you've described yourself already) to tackle the issue. With recording levels as low as that, you may experience dynamic loss and possibly noise or whatsoever, regardless of using 24bit.

Well, after seeing your screenshot, it actually isn't strange. In the editor, click onto "File" rather than "Track" in the title row. You can only normalize individual files, not an entire track.