SaschaFranck

Still defenitely no beauty but doing exactly what I want it to do. Some cables could run tighter but that's what you get when you try to cram things into existing arrangements. Also, I want some of the things to be more or less freely exchangable (such as the Stomp, should I need it for smaller gigs and what not, even got a fly gig in India next spring...). Adding that little mixer (which was intended to be a custom-built thing but my electro-dude didn't get back for me so often that I just gave up) apparently was a neat idea, I can now instantly use the board for IEM gigs, route some backing tracks into it on boring soundchecks and what not. Soundwise, I couldn't be any happier, either. Still waiting for my Harley Benton Drop Kick, apparently a Suhr Riot clone, to arrive (was broken, no instant replacement due to the usual supply shortages) which will take the place of the Hustle Drive. But there's way more than enough fun to be had with what's on there already (my last addition, the Rockett "Dude", is just awsome). Would've loved to add a patch bay (in fact, I have one sitting on the shelve already), but with this design it's impossible. I'm defenitely not going to ever take the actual board out of the case, too many people have been stumbling over my stuff in the past already. Well, they can stil do that, but at least they will now break their ankles in return, hrrhrr...

No, that won't work. You need to do it the other way around. Your headphones won't work with the send, so the upper path has to have the amp signal, so you could use the normal HP out. Unfortunately there's something i forgot before: You can only create a split in case there's a block on the splitted path. IMO that's really an oversight on Line 6's behalf as there's plenty of reasons why you'd like to split something without using any blocks. In addition to that, unlike on the Floor, re-assigning output blocks isn't possible on the Stomp at all. So I'm afraid that if you wanted to get this thing to work, you would have to "waste" a block in any case (or connect an external headphone amp to your send, less than ideal I'd say...).

Method 1) You would have to use the split path for that and create an output for path B. That way, path B would run to the send jacks. You could use that to feed your amp and use path A for your headphone duties. The modulation block would have to go for that purpose. The advance of this would be that you wouldn't have to waste a block for a send. Method 2) You would insert a send/return block before the Stomp's amp and feed your amp from there. You'd lose a block that way but could leave anything else intact. For both methods, you defenitely want to make sure the amp (or at least the cab block) are the last thing in the signal chain, otherwise you would only hear, say, your delays and verbs through either your amp or your headphones. Fwiw: You're pretty much wasting the cab block in the patch you've posted. In case you'd be using an A+C block, you could ditch the dedicated cab block.

Voted.

Kind of. Or rather the other way around... Nah, at least kinda finalizing things today, got all things together and need to do some new pedal arrangements and some housekeeping. Will look pretty similar to the last iteration, though. I'm sure that I'll be building an entirely new board next year or so, but that needs some serious tinkering (and possibly some serious investment, too), so until then I just want to call it a day (perhaps even longer, obviously needs some longer term tests).

Good morning, fellow pedalboard!

Did you post that on Ideascale? Not that I'm using any loops much anymore (also, as said, I usually stick with 230k), but I'd vote that up instantly because it's an excellent idea.

We could possibly argue about which modeling we like better, which is more authentic, more "complete" and what not until the cows come home, but I gotta say one thing about the Fractal units: IMO (which might be terribly different from other opinions) they got their factory presets, their model defaults and their model presets just right whereas Line 6 hasn't. Some of the model defaults in the HX-verse just sound absolutely horrible to my ears (Mail Order Twin, what's that? A completely broken amp they pulled out of some trash? And did someone try to play metal on the Fullerton and Grammatico amps and saved their attempts for us to enjoy? George Benson would run away when plugging his guitar in this version of a JC-120 and Brian May might file a lawsuit for wrecking up his favourite amp so much... etc.). I have missed the 3.5 factory presets, maybe at least those are a little better, but so far anything straight out of the box sounded *way, way, way* below the league the Helix is trying to play in (which it actually in fact is playing in - or at least could). And yeah, I checked with a whole variety of kinda standard guitars. Once you've deleted all the factory presets and set up your own templates and model defaults, things are an incredible lot better.

Not really, it's changing the way the guitar and amps/pedals interact. Just try for yourself. Gets pretty obvious when you turn down your guitar volume.

Whenever you deliberately want some mismatching. Which is what I prefer.

Personally, I never pick "auto" but pretty much always go for 230k. Seems to work best with pretty much all of my guitars and the vast majority of HX drives, IMO even plain amps profit from it. But then, these days I usually have my Stomp (had a Floor before) running in a buffered environment, so that pretty much equals a constant 1mOhm impedance anyway.

You might be able to get them back through the L6 support.

Just try, there's nothing to lose.

And yet another can of worms... That clearly is possible to a rather large extent (fun test: set your DAW's buffersize to 1024 and try to play through that in time...). But IMO it's raising another question: Do we actually want to compensate for latency all the time, does it do our playing any good? It surely doesn't matter much in case we do it every now and then - but considering anything that might appear logical, if we'd do it all the time, we'd likely get used to playing early. Which I think of as being quite less than ideal. Implication #2, I mentioned this above already: If you compensate like that for a plugin/host-introduced latency (so what comes out of your monitors when recording sounds as if it aligns perfectly), the recording itself will be early (as everything will be recorded the way it comes into your DAW - which in this case will be your early picked DI guitar signal).

No, I wasn't talking about that kinda stuff (which might even be irrelevant, at least for a long time being...), but... Yeah, but the main aspect IMO still would be that, once latency occurs, when using headphones it comes without any further audible clues. In other words: When you move a meter away from your amp, your monitor speakers and what not, the sound is likely gonna change quite a lot due to completely changed reflections - which is an audible clue that the distance to your sound source has changed. And because we're used to deal with these audible clues of distance (or rather even "trained by evolution", because some years ago you just had to be aware of these kinds of distances when that saber-toothed fellow was approaching from behind). As a result, we're likely just able to accept distance in the physical realm kind of easily. But once you get 3ms more of latency through your headphones (which pretty much exactly equals 1 meter), there won't be any change in sound, the latency is just added on top, so that can become irritating quite fast. Add to this that when wearing headphones, you're losing the last bit of audible connection to your actual picking attack even at household levels. Thing is, even when the musical timing issue isn't relevant anymore (and quite likely we could spend another lifetime on debating when exactly musical timing becomes an issue, whether it's only an issue once someone hears it or whether there's "sub-implications", how we compensate for it once it shows up, etc...), which I assume we could at least for now agree on, would be the case once we're talking latency differences of, say, 2-3 ms (otherwise we wouldn't be using modelers at all), our ears and brains are still incredibly sensitive to time-related things. Possibly the most blunt example: When you close your eyes and someone is talking to you from very close distance (like, the heads just a handful of centimeters apart), you will instantly notice when that person will move left or right by just a few centimeters, even in an anechoic chamber (with all reflection clues removed), possibly even more so. Also, as an example out of my personal experience: When playing some rhytmically relevant stuff through a patch with a sharp attack through a sound enhancing all attacks (like a super clean amp) using plugins and through headphones, I can almost reliably tell whether my host's buffer size is set to 32 or 64 samples. We're talking about the difference between 4.6ms and 6.0ms here. And well, while I like to think of myself as being a timing-savy player (who wouldn't...), that's *way* below anything I'd consider myself noticing in whatever possible raw timing tests (in fact, there's some of those tests online, which can be quite revealing). So, how comes I can still pretty much reliably tell about the difference (I'd roughly say in around 80% of all cases in a direct A/B comparison, I once even had a mate switch a delay of 1.5ms on/off while I was playing)? My personal explanation would be that our senses (in this case those of my picking hand) are almost as trained to deal with time and distances as our ears are (which, historically, would make as much sense because we defenitely need to hit our saber-toothed friend in time with our spear - waste just a handful of ms and you're qualifying for further issues...). Interestingly enough (and to take that to yet another level/layer, but I told ya, can of worms and so...), all that is still working with our typical reaction time being *way* above anything we're dealing with here (typical reaction times between perception and reaction are in the range of 0.3-0.4 *seconds*, IIRC), so there must be a higher instance in our brains, lining things up and sorting them in a fashion that we can still perform with proper musical timing, such as in "getting into the groove" or so. Which might as well explain why we (at least that's most common among the folks I know, myself obviously included) often aren't "in the pocket" on the first few beats of, say, a recording, making it a good idea to already start the picking movement (ideally in the air) during count-in. Fortunately, at least we're not singers. Because in that case we'd also be talking about headphone introduced phasing (which can sometimes be avoided by actually *raising* latencies, go figure). Very interesting stuff IMO.

Well, that could be a can of worms we're opening here. DAW introduced latency through headphones is something pretty different from physical distance latency, at least usually - simply because the latter comes with further audible clues (if we'd really open that can of worms, we could as well introduce visible clues, but lets keep that out for now...), whereas electronically caused latency doesn't. Which is why a bit of a room ambience sometimes does a great job in masking that effect. But that's still not all there is to it. When monitoring through speakers at moderate levels, you will likely still have a bit of sound straight from the guitar attacks left, so that could become your "rhythmic anchor", whereas when listening through headphones, it's likely that the sound coming out of them will become your main reference. Hence, we may compensate for latency or we may not. And btw, compensating for DAW-introduced latency isn't exactly a good idea because that "manual compensation" will be recorded, and as a result, it'll be a tad early on playback (because on playback, the DAW compensates for latency). This is quite different to real life latency compensation, when what you actually hear will be recorded. Now, with modern technology and generally small latency numbers, all that might not be too relevant timingwise, so we may probably only have to get used to it, but still, IMO it's a good idea to be informed about the various aspects and implementations of latency.

People have been reporting scrollwheel related crashes already. Did you use that?

Fwiw, some folks mix up device latency with audio interface latency. Even if partially releated, these are very different things. Just sayin'.

That's true. And it's quite a shame as it goes to show how much L6 ignores their very own forum.

I'm pretty sensitive to latency and my current setup features 2 modelers in series, coming in at 2ms each (Amplifirebox or NUX Amp Academy and an HX Stomp for spatial FX), at least for leads. Works fine, even under headphones. I still take care of those values not getting any larger though, because there's always the odd moment, when more latency is added (digital consoles, less than ideal DSP controlled monitors and what not).

Haven't seen anything, either. But then, my monitor is switched off.

When it comes to just that, there's pretty much nothing like the GT-1000 (oh well, and the Iridium and the NUX Amp Academy, but I guess we can rule those out for now).

Hm, I don't think you'd love an FM3 much more then, at least not regarding the on-unit editing aspect. When it comes to just that, I think it's either a Helix Floor/LT, a Headrush or maybe a Boss GT-1000 along with Boss Tone Studio on a tablet. All others pale in comparison. But as said, that's just the on-unit editing aspect, which defenitely isn't all.

Would you run the unit along with your HX Effects? If so, you could as well consider 1-2 of the mini-modelers along the lines of a Strymon Iridium, Atomic Amplifirebox (even if the company at least sort of s***s) or whatever. You could easily connect two of them in the loops of the HX Effects. And you could still enjoy the easy editing of the HXFX. With all the HX series drives, you should be able to get plenty of great tones just out of 1-2 baseline amp models. As far as the FM3 goes, yeah, I gotta agree, the Fractal amp modeling is absolutely great. And while most people (myself included) often aren't able to distinguish a Fractal unit from a Helix in A/B listening tests, I think playing either does reveal some differences. I absolutely think that Line 6's modeling is fine, but Fractals is (or rather feels) extra-fine. Had an Axe FXIII borrowed for a few days and it was super great playing fun. And yes, I liked it a bit more than the Helix. Anyway, if you need decent on-unit editing, the Fractal stuff is really, really bad. Yeah, it's been improved a bit, but compared to most others it still feels like being thrown back at least a decade or so. For me, that's an important thing to consider.

Also, did you make sure to NOT sign in within the Line 6 Updater but went to Offline Mode straight ahead?