mdmayfield

Ah, sorry I missed this thread until now! I am in a Pink Floyd tribute band and tried the same thing with similar results. Two octaves of pitch shifting seemed OK for strumming, but just too much for such an exposed and important part. I ended up dedicating a separate guitar to the "high strung" gauges and tuning for Hey You.

Spot on, 100%. Mmmmm, space sauce.

The Channel Volume doesn't work too well for a master patch volume control, because if you have any effects post-amp model (reverb, etc.) there's a chance that they could "virtually analog-clip" and cause unwanted distortion. This caused me clipping / harshness problems when I was running amp Channel Volumes up around 7.0-8.0 or so (on certain sounds) into delays and other effects post-amp. Reducing the amp's Channel Volume and increasting the Gain on the Output block helped this.

That sounds like a cleaning product :-) I'm not at my Helix at the moment but you might try using the Reverse Delay with a fair amount of feedback into a reverb set at 100% - that will take away the attack and make it sound more pad-y.

This isn't a big problem, but it's an annoying curiosity - I'm seeing a few of my patches immediately light up with the "E" symbol next to their names to signify they've been edited before I edit them, immediately after calling them up. So far I haven't isolated any common factor to which ones or why, and even if I save them again, or make a small change and save them again, they persist in this odd behavior. Has anyone else seen this, and do you have any idea why this would be?

I've never heard the StageSource speakers in person so I can't compare them, but I do own QSC K8s and K10s, and am very happy with them. Even the little K8 has always had more than enough power for me to use it as backline, and occasionally when necessary, as the only source of guitar sound (not through PA). It has never let me down. (For reference, I play mostly in classic rock and pop cover bands and tribute acts - so, medium-loud stage volume at most. Your mileage may vary if you play a style that has > 100dB SPL stage volume, 16th note double kicks, or other busy and complex arrangements.)

Yes, it works on vocals but doesn't sound very natural. There's no formant correction so voices pitched upward sound a shade like The Chipmunks, and voiced pitched downward sound weird in the opposite way. Also, it will intelligently change intervals between say major and minor 3rd intelligently, but won't go to 4ths or other appropriate intervals without manual (footswitch) parameter changes.

It'll help to think about this more generally. Instead of it being a Whammy pedal model, think of it as a general-purpose pitch shifter. Like any effect, you can select parameters (settings) to be controlled by the expression pedal. To build a whammy effect: 1) Set up a pitch shifter effect 2) Set its Mix to 100% 3) Hold down the parameter knob for the amount of pitch shift (half-steps) and assign it to the expression pedal 4) Set Min to where you want it when the pedal is in heel position (for example, -12 for one octave or -24 for two octaves) 5) Set Max to where you want it when the pedal is in toe position (probably 0 for no shift)

Although I don't know exactly how every DAW calculates its latency, I've come to believe that they don't always report it accurately. I'm not sure if they're just guessing based on buffer size or what, but if 8-10 ms is what your DAW was reporting, I wouldn't be surprised if the true effective latency of your setup was actually double that (16-20 ms) or more.

Time for my favorite analogy... Can you create an IR of a cab without a microphone = Can you take a photograph without using a camera?

I'm glad jnysen mentioned something about this above - there's nothing inherently bad about the phase shift that happens in a traditional IIR / minimum phase EQ, and nothing necessarily always good about avoiding phase shift by using an FIR / linear phase EQ. One of my favorite sites to talk about audio stuff is HydrogenAudio, because as part of their Terms Of Service any claims about subjective audio quality must be backed up by double-blind test data. We all tend to hear not with our ears, but with our eyes and expectations - even when we're trying very hard to be objective. A nice quote from an acoustics expert in a thread on there which is appropriate here: That's from here: https://hydrogenaud.io/index.php/topic,103316.0.html

I've tested the Helix with two different Variax 300s. With one of them, the Helix did not control the Variax model - but with the other, it does. Not sure why, and unfortunately I don't have the first Vax 300 anymore so can't test further.

I'm using my Helix for vocal effects, and it seems that every time I turn it on, in Global Settings, the Mic Gain parameter *says* that it's at 15dB where I left it, but the actual gain is much lower (probably 0dB). If I turn the knob one click away from 15dB, to 12 or 18, and then back to 15dB, it sounds right. There seem to be a number of these issues being reported lately, where the front panel says one thing but the parameter is actually set to another.

If this were implemented, I'd hope that it would have a Preferences option to turn it on or off. I strongly prefer the current way with percentages.

Modeling a guitar cabinet without including a microphone is like trying to take a photograph without using a camera. It just doesn't work that way!

Did you reset the unit to factory settings? The instructions say in big bold letters: IMPORTANT! YOU *MUST* FOLLOW THESE INSTRUCTIONS TO INSTALL 2.00! ... YOU MUST RESET GLOBALS AND RESTORE PRESETS, OTHERWISE HELIX 2.0 MAY NOT WORK PROPERLY If you didn't back up your patches, reset the unit, then restore the patches with the new editor software, then that's very likely why things went wrong.

OS X 10.10. Followed the instructions exactly including patch backup, firmware update, reset to factory, update Helix application, restore patches. No issues at all. I suspect that a majority of upgrade issues have been due to forgetting or missing/skipping something, and a tiny minority (if any) otherwise. On a forum like this you're going to see 90-100% people who run into issues, and 0-5% people who have no problems.

For what it's worth, I've gigged almost exclusively with Variaxes for 11 years - first using 300s and 500s (sometimes transplanted into other guitars) and now with a couple of JTV-69s. I have used one Line6 VDI cable for more than 500 gigs and uncountable rehearsals over the years, and it still works perfectly. I've always carried a backup, but never once have had to use it. EDIT: are you maybe confusing the Line6 VDI cable (with Neutrik Ethercon connectors on both ends and a nice flexible cord) with the crappy cable included with Variax Workbench (with a regular RJ45 on one end and Ethercon on the other, and a crappy regular computer-style Cat 5 cord)?

I've run plenty of shows with tracks, and have never seen a need for this. Although, that may be because I have only prerecorded tracks, use one preset per song, and don't change tempos during a song. Are you using Live's actual *live* functionality, like recording new loops and warping audio, or is it just a playback device for pre-recorded tracks? If the former, I can see running into dealbreaker issues without sync, but if the latter, you might try one or more of these approaches: - Send a few Tap Tempo MIDI CC messages in the correct tempo at or before the beginning of each song / tempo change - Include a Tap Tempo MIDI message on every beat of parts of the DAW project that have live tempo changes, if any - Use Tempo Per Snapshot in the new 2.0 firmware, and switch snapshots with MIDI sent by the DAW - If 8 tempos are not enough for one preset, then between songs or musical phrases, use MIDI Program Change messages to change patches to an otherwise-identical tone

I have seen similar behavior when controlling the Helix with Program Change messages from a DAW. If the PC messages happen too quickly one right after the other, the Helix ignores the second one. It's not quite the same as a PC followed by CC messages, but the general behavior seems consistent with what RicksterUK mentioned above.

Also don't forget that any time you measure the response of speakers, you're also measuring the response of the space they're in. Most rooms that are found in houses (i.e. much smaller than most performance spaces) have crazy acoustics, with some frequencies that ring out, and many different frequencies that cancel in some physical places and reinforce in others. It's not uncommon in a small room to find differences in room response of 20dB or more between just a few Hz, especially in the bass and low mids. And - since it can't just be simple can it - the frequency responses changes completely if you move the mic (or your ears) just a couple inches one direction or another, so any EQ is at best a compromise. For example, here's a spectral decay plot (EQ response + resonance in time) of a domestic room I once recorded in as measured by a sine sweep and a measurement mic in one specific position. This is even after extensive acoustic treatment: Note the 17ish dB difference between about 80 Hz and 115 Hz, and the way that some frequencies ring out longer than others. That's with quite a bit of broadband absorption; with no treatment the room was like this: Note the much longer decay times of much narrower frequency ranges. This caused really bad "one-note bass" where certain notes would all sound the same because they would all excite the room acoustics near the same frequency. In an untreated small room like that, EQ is limited in the amount it can help because it can't control ringing/resonance - only relative volume of frequencies. This has rambled quite a bit, but my point is, take care when making global EQ settings based on measurements, because you're compensating for the speaker PLUS the room.

It will power it and pass audio through the VDI cable, but will not force the model or other controls. This is as of the current Helix firmware - it's possible they might implement old-school Variax full support in the future, but there's no guarantee they will, and it seems to me that it's not all that likely.

In my experience, dual 15" active speakers can be all over the map in quality - the good ones are REALLY good, and the bad ones... ecch. I've never heard that particular speaker in person, so I can't offer a specific opinion. I'd strongly recommend auditioning speakers before buying one, though, unless you're comfortable with turning it over on Craigslist or similar. I own and personally recommend the QSC K-Series (one K8 has been more than enough for me, even in fairly loud bands). I've heard the Electro-Voice ELX series, and those are good too; there are a few other Class D active speakers in that general price range that I've at least auditioned once at a store, and all of them seemed acceptable. The Behringer B2xxD series are less expensive, but their performance is not great; I own a couple of them as backups / extra monitors, and they run out of steam fairly quickly - at least my B208Ds do. Any low mids/bass causes them to hit their internal limiter at too quiet a level to be useful on a loud stage. They also have a boxy/midrangey sound that I've not been able to EQ out. I wouldn't recommend them; at the price you can get better quality used.

:blink: and :D ! I really appreciate the info, and don't worry, I'm already a huge fan of the Helix as is; I won't raise my expectations too high or read too much into this. Of course, I can't speak for the rest of the Internet. :ph34r:

Well, yes... but those things have vast tracts of RAM, and none of them are meant to be simple enough the average guitarist can use it. :) If such a thing is being worked on or explored, I have to assume "the devil is in the details" as far as how to implement the UX of it so that it doesn't hopelessly complicate things. For example, touch and hold footswitch, click a new soft button that says "take snapshot." Then go and change all the parameters/bypasses/etc you want, then touch and hold another footswitch, and click "take snapshot," then stomp to switch between the two. But I have no idea how much space is available per preset; it almost seems like each snapshot would take nearly as much storage as a full preset itself. So then there's how to "implement it technically" - like maybe instead presets that have all the same blocks would automatically be scenes instead... But then the UI would need to be different. The whole thing sounds like a huge (but kind of fun) challenge for the developers.