Digital Igloo (Eric Klein, YGG)

chicken said:
I'm far from DSP field but I guess IR length dictates segment size and n in O( n ). That's why it can limit number of 2048-point IR - knowing length gives estimation of load
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Well, I was only stating that because you were talking about phase issues.
 
AFKAEjay said:
Cool product…never heard of it…probably made too much sense for peeps to buy it ;)

Why does 96kHz makes sense in mixers?…cause it’s real-time consumption which doesn’t go back to 44 in the end?
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A few reasons:
  1. Once the first digital console advertised 96kHz, the competition felt they needed to follow. These kinds of specs are much more important when trying to sell to venues, houses of worship, universities, etc.
  2. If one's tracking the dynamics of a 70-piece symphony orchestra with $100k+ worth of mics and pres, one could make an argument that 96kHz can make an audible difference. For guitar processing, however, it's completely wasted.
  3. Vocalists are especially sensitive to latency. 7-8ms is imperceptible for almost all guitarists, but I've produced singers where 4ms was noticeable.
  4. Helix runs the same DSPs as our 20-channel Stagescape M20d mixer, which are something like twice the DSP of B£#®!n%£®'s wildly popular X32 40-channel console. (Stagescape sounded incredible. It was just too expensive for the channel count and like many of Line 6's products, ahead of its time.) That's ~4x the DSP per channel. It doesn't take a whole lot of juice to run a bunch of compressors, EQs, simple delays on a send, a reverb on a send, etc. If you're running amps, distortions, and poly pitch algorithms, DSP is much more important, and running at 96kHz effectively halves your available horsepower.
Sascha Franck said:
Yes and no. Because in case RTL is 5ms, using a loop (splitting up the digital path of course) would already result in 10ms. And at least for me (plus for quite some other folks I know), this is when things start to feel sort of distracted, at least when using headphones. Using 2 serial loops would be pretty much impossible already. With the GT-1000, even with both loops set up serially, RTL (at least without any blocks adding their own latency - no idea about which those would be in Boss land), we'd still be kicking at 3ms (or even lower, according to Leo Gibson - I rounded his 0.7ms up to 1ms), which is just fabulous.
Now, if you never use digital FX loops, if you never run your signal into another digital device, into an additional digital mixer and/or IEM system, any DSP controlled monitors, all that while using a guitar cable instead of a wireless system - sure, in that case, even 5ms is a decent value.
But personally, I'm constantly running into various scenarios where it's not possible to avoid some of these things, sometimes in combination.
So, even in case you're not Steve Vai (which I certainly am not), it's not exactly too uncommon running beyond the "uh, that's not too much fun anymore" point.
It's the reason why I always bring my own little mixer with me on IEM jobs, so I can at least monitor my own signal without any irregularities (and believe me, I've been there multiple times... FOH folks apparently aren't always the brightest candles in my neck of the woods). And it's also why I rather start with the lowest possible latencies. It's also one of the reasons I kept using Logic, because at least for a long time it's been the only DAW where buffer sizes were pretty much a "set and forget" thing (even if you dialed them in pretty low).
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Yep, if you utilize a lot of loops, every D/A/D stage adds latency. So if one were building a higher end pedal switcher where each loop utilizes a conversion, latency creep will be a real thing. Which is why no one does it that way. Even if BOSS were to make an 8-channel digital (not relay-based) switcher, the latency would quickly creep toward 10ms—double that if digital pedals are used.

However, there are things one can do to help mitigate that:
  • Run analog drives before conversion, and if they have MIDI control, bypass them that way. Zero latency accrued
  • If you always run certain pedals in a specific order, have them share a loop and use MIDI to bypass them independently
  • The big one—for any time-based effects, keep their mix settings at 100% and control the mix from your modeler's FX Loop block. That way latency is only accrued on the wet signal, which is effectively inaudible.
chicken said:
  1. About helix paths/cores synchronization - if I play with powercab AND real cab I have two paths - one with IR for powercab and second without for real one. Are they in phase? Generally speaking if two paths have different effects sets will they be synchronized at the output?
  2. Please add poweramp simulator
  3. Please keep worldclock in the helix 2.0
  4. Have you considered "advanced" editing mode where you expose more model parameters?
  5. Is it possible to make model editor public? a tool a-la ltspice?
  6. I can't RTFM my way through this but how to make helix auto assign bypass to footswitch as soon as I add new block?
  7. Please give us global blocks.
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  1. For most people, they should certainly be close enough for rock n' roll. An exception would be if you used Poly Pitch algorithms on one path but not the other. If there's an issue, the Simple Delay block has 0.1ms resolution for compensating for any phase mismatches—run Feedback at 0% and Mix at 100%. If you're still worried, you could also use two identical Synth > 4 Osc Generators and the Mixer block (panned hard L-R) with the polarity flipped on one side to perform phase cancelation tests.
  2. This would require a major retooling of our amp DSP. Not insurmountable, but a big ask nonetheless.
  3. You mean a new Helix Rack? In one of our polls, almost no one is using BNC WC in with Helix Rack; it was included because previous racks had it. I doubt we'd be able to warrant adding WC in to any future floor processors, but I'll add your comment to our spreadsheets.
  4. Yes, we've discussed this at length.
  5. Line 6 closely follows the 80-20 rule. If we felt a good portion of our user base would take advantage of something, nothing's off the table. But if a feature requires a lot of work (and this includes documentation, support, QA, CS, etc., not just development), it better be something we feel the majority of customers would use.
  6. POD Go and POD Go Wireless do this. We don't auto assign stomps in Helix/HX because A) cap sense switches let you assign a block to exactly the switch you want in 3 seconds, B ) there are far more block locations than switches, and C) unassigning a block you don't want on a switch is currently pretty time-consuming (8 seconds vs. 3, but still). Also note you can swap all assignments by touch-holding two stomp switches.
  7. We've discussed this at length as well. Actually, there isn't a legitimate feature request we've encountered that hasn't been discussed at length. No promises of course.
 
Thank you! I know you get same questions sometimes over and over. I'm sorry :-/ But please use me as a data point for the spreadsheet. Fractal has myriad of params and it seems to fly well 🤷‍♂️ I don't know who is your user though. Regarding Worldclock, I don't care about floor units. For racks it kinda makes sense? gives professional look? synchronizes clocks? If you have digital something you have clocks and if you have spdif you have to accommodate to external clock source anyway. But I can imaging one of those planning meeting where it will be discarded :-(.
 
chicken said:
Fractal has myriad of params and it seems to fly well 🤷‍♂️ I don't know who is your user though.
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Your average Fractal user is more technically proficient than your average Helix user. Have said this before, but if I never worked at Line 6, I'd probably own an AxeFX III, as I'm totally a Fractal-type customer.

We focus on Helix users (and potential new-to-digital customers); attempting to steal Fractal customers away would likely alienate our own customers, not to mention kinda being a dick move. Cliff is doing a great job answering the needs of his user base; we'd like to think we're doing the same with ours. Admittedly, the venn diagram overlaps quite a bit, but we know where we need to go and where we need to stay.
chicken said:
Regarding Worldclock, I don't care about floor units. For racks it kinda makes sense? gives professional look? synchronizes clocks? If you have digital something you have clocks and if you have spdif you have to have accommodate to external clock source anyway. But I can imaging one of those planning meeting where it will be discarded :-(.
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S/PDIF is automatically clocked from the source, so there's never any need for BNC in there. With AES/EBU in, however (which Helix Rack has), it can be helpful. I run BNC WC from my Lynx Aurora n to Helix Rack, which sends its AES/EBU out to a Dangerous D-Box+, which doesn't have a BNC WC in. Haven't experienced any jitter whatsoever.
 
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Digital Igloo said:
the latency would quickly creep toward 10ms—double that if digital pedals are used
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one might start to wonder why digital pedals don't have digital io

Idea Thumbs Up GIF by ZEE5
 
Digital Igloo said:
S/PDIF is automatically clocked from the source, so there's no need for BNC in.
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I'm a software guy, but as it happen for some reason already have my mental model of how things work lol. I thought the clock source kinda replaces ticks globally, including converters and DSP. If it is true it must mean SPDIF clock propagated further. I imagined it as a switchable thing - internal crystal, SPDIF, wordclock. So having SPDIF also means means having this clock switching logic. And then "simply" (another word for planning meetings =)) having WC sub board, like just another source. Hope at least SPDIF stays.
 
chicken said:
one might start to wonder why digital pedals don't have digital io
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Cost, complexity, size/footprint, and the fact that most pedalboard guitarists wouldn't see the advantage. Alesis tried this with their ModFX line of tabletop processors back in 2003; IIRC they didn't stick around for long.
chicken said:
I'm a software guy, but as it happen for some reason already have my mental model of how things work lol. I thought the clock source kinda replaces ticks globally, including converters and DSP. If it is true it must mean SPDIF clock propagated further. I imagined it as a switchable thing - internal crystal, SPDIF, wordclock. So having SPDIF also means means having this clock switching logic. And then "simply" (another word for planning meetings =)) having WC sub board, like just another source. Hope at least SPDIF stays.
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Not an engineer, but I'm fairly certain that almost all gear automatically syncs when S/PDIF is used. Which means if you run a cheap effects processor into your $5000 audio interface, if that interface doesn't resolve incoming clock well, your interface's A/Ds and D/As may end up with worse sound quality, as the better product is resolving to an inferior clock. These days, however, most of the clocking problems from the 90s and 00s seem to have melted away. Converters and clocking has gotten a lot better, and the only reason for digital connectivity is for those who can't sleep unless they know they're getting the absolute best out of their system... and people trying to eek the absolute lowest latency out of a system.

And again, unless everything's running at the same sample rate, there's no latency advantage to keeping it all digital, as sample rate conversion incurs latency as well.
AFKAEjay said:
What’s the rationale behind that?
In my mind with 96K you get less “filling in the gaps by the algorithm“…which may be audible…but when you go back to 44.1 when you master…it’s all out of the window right?
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I'm probably the wrong guy to ask, as I never record higher than 48kHz. Well-produced classical recordings ($$$$$) have a much greater range between the softest and loudest passages than your typical pop or rock record (which is why they're compressed to hell on terrestrial radio and Sirius/XM, so you're not constantly turning it up and down to hear over engine and road noise), so those who embrace higher sample rates argue that the incredibly quiet parts have better resolution when listening on an expensive audiophile system in a controlled acoustic environment. Is all of that lost when downsampling for CDs or streaming? Maybe, maybe not.

Okay, I'll admit that there may be one additional advantage to 96kHz—if you drop a recorded loop to half-speed, it'll be at 48kHz instead of 24kHz.
 
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Do you think the Stomp XL and HX 1 deserve an upgraded version with more DSP? Even if they don't get any other IO or other physical upgrades.
'Give me more power Mr Scott'!!
From my perspective the current quality of HX content is really good and those two products are screaming to be more robust.

I can see why you wouldn't throw more DSP at HX Floor etc other than continuing QOL support. I'm hoping a next level Stomp XL would be worthy as an interim project.
 
Enchilada Jones said:
Do you think the Stomp XL and HX 1 deserve an upgraded version with more DSP? Even if they don't get any other IO or other physical upgrades. 'Give me more power Mr Scott'!!
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We currently have no plans to replace the DSPs in HX Stomp or HX Stomp XL. I imagine any potential replacements would have a more powerful DSP (if only because those SHARCs will presumably be EOLed by then) but never at the expense of what makes those products special.

There's zero advantage to increasing the DSP of HX One, as its ARM can handily cover any single effect we have now or any new effects we'd create in the foreseeable future.
 
Digital Igloo said:
However, there are things one can do to help mitigate that:
  • Run analog drives before conversion, and if they have MIDI control, bypass them that way. Zero latency accrued
  • If you always run certain pedals in a specific order, have them share a loop and use MIDI to bypass them independently
  • The big one—for any time-based effects, keep their mix settings at 100% and control the mix from your modeler's FX Loop block. That way latency is only accrued on the wet signal, which is effectively inaudible.
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I'm aware of all these (and possibly some more... working around latency has become, uh well, one of my "side hobbies"). It's also why I applaud Fender for their two digitally controlled analog loops in the TMP, allowing you to integrate analog pedals into a programmed realm (most analog pedals simply don't feature MIDI control).
Yet, in case your pedals are working fine in a digital loop (read; surrounded by buffers, so to say...), that's defenitely a way more comfortable thing to deal with as you can mix and match internal digital and external analog things to your likings. And the GT-1000 allows for just that without ever having to think about latency. The HX series aren't all that bad, either - but as said, the smaller, the better.
 
Sascha Franck said:
I'm aware of all these (and possibly some more... working around latency has become, uh well, one of my "side hobbies"). It's also why I applaud Fender for their two digitally controlled analog loops in the TMP, allowing you to integrate analog pedals into a programmed realm (most analog pedals simply don't feature MIDI control).
Yet, in case your pedals are working fine in a digital loop (read; surrounded by buffers, so to say...), that's defenitely a way more comfortable thing to deal with as you can mix and match internal digital and external analog things to your likings. And the GT-1000 allows for just that without ever having to think about latency. The HX series aren't all that bad, either - but as said, the smaller, the better.
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Sure. Again, be wary of products that tout how many cores they have. More cores = more core hops = more latency.
 
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Digital Igloo said:
Oh dang, totally forgot about the Roland A-61. Hardcore keyboard controller, robust DAW controller (motorized faders of course), audio interface, and monitor controller with talkback and cue feed in one box. Designed for MainStage/Ableton Live music directors. This, a laptop, and monitors/phones is all you'd need.
View attachment 16757
Clearly, my designs have mellowed out somewhat since 2007.
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Gonna start calling you Digital Kitchen Sink.

:rofl
 
Sascha Franck said:
How did that not take off? Fantastic concept, especially for all those home studio owners constantly having to deal with space constraints.
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A-61 also had a Quick Start guide with lots of diagrams, Mac/PC editor design/layout, customer target study, and even a marketing campaign. No acknowledgment from Japan whatsoever. And I wasn't sending it in blind, either. Part of my job description at Roland/BOSS US was Technical Liaison to Japan. It was literally my job to communicate with Japan—to send in customer complaints, feature requests, technical problems, bug reports, etc. I understand things are better there now; FWIU, Jeff Slingluff (RIP) was able to push a lot of ideas over the pond.

I was able to get several features added to Fantom-G 2.0, however.
 
chicken said:
would it be simpler to do poweramp sim by having a new amp model with flat preamp?
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This is probably more of a Ben or Brandon question.

My understanding is that our preamp and power amp models are intrinsically tied together, and they were measured that way, so I'm not sure how good the experience of any ol' preamp block going into a generic power amp model (or handful of generic power amp models) would be. I imagine it wouldn't meet our standards of accuracy. Plus, there's currently no way for one block to communicate with another. For example, what would happen if two preamps were fed into a single power amp? Or a single preamp into two power amps? Many opportunities for user error to result in "what is Line 6 smoking?"
 
Digital Igloo said:
so I'm not sure how good the experience of any ol' preamp block going into a generic power amp model (or handful of generic power amp models) would be.
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Well, look at it that way: Plenty of people seem to be pretty happy to run preamp only patches into the returns of their tube amps. And at the same time, some other folks are making use of just the power amp sections of other devices, to run, say, their old rack preamps into them. I'm one of these people. I still own a Soldano SP77 preamp and running it into the power amps of my Amplifirebox is quite a great experience. Even the poweramp sim of my AMT Pangaea is so good that I plan to slap a little fun rack together once I manage to take the SP77 to a longtime overdue service.
I don't see how Line 6 power amps would do worse than that. And well, it'd possibly a decent idea to not name (or even design) them after a certain amp model (even if it might not hurt, but it'll save you some trouble to keep it generic) but do it the Atomic way as in providing a number of power amps featuring the most common amp topologies and their most common functionalities (basically just presence, depth/resonance, master and channel volume). On the AFB there's 4 poweramps, featuring EL34, KT88, 6L6 and EL84 tubes and that was it (with just master and presence as sound shaping parameters), still pretty useful.
 
I mean you have custom virtual amps that exists only as model? so maybe it would be easier to add one more. So no measurements needed. The amp with fully flat preamp.
 
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