mdmayfield

Some kind of "scene mode" - hopefully a brilliantly clever, elegant implementation that nobody else has thought of - would solve this issue and more.

From the bits and pieces of info I've gathered, the inability to save loops is most likely a hardware limitation. Chances are, the Helix stores the looper audio in short-term memory (i.e. RAM) which goes away when the power is turned off. It probably doesn't have enough free long-term storage (i.e. Flash, the equivalent of a hard disk or SSD) to hold audio loops.

Hi Thomas, I see this is your first post. Welcome to the forum. It's considered good forum etiquette to start a new topic for a question about something other than the thread's subject - that is, not to "hijack" the thread, or "necro-bump" an old topic that has run its course. For myself, I've found that if I start a new topic for a question, with a relevant title, there's better chance that people will see and answer the question that way. It might be worth considering doing that for your question. As far as the question itself, the only helpful thing I can think of is to keep regular backups of your presets, but of course that's not too useful in the heat of the moment at a gig. Thanks & good luck, Matt

That's correct. If you switch your speaker to +4, that means it's expecting a hotter signal, so it'll turn down its input sensitivity and be quieter. -10dBV and +4dBu are just analog voltage level standards. Any given analog audio signal (such as the analog output of your Helix plugged into the analog input of your speaker) may be at, between, or outside these levels in amplitude at any given moment, as determined by the nature of the sound itself, plus any volume controls it passes through. As long as the signal does not overload the speaker's input and cause unwanted clipping distortion, you should be OK using either the -10 or +4 setting. More info (maybe more than the average person would care about) here: https://en.wikipedia.org/wiki/Line_level

I made a quick YouTube video explaining two Helix feature requests: 1) Allow the use of FS4 and FS5 during looper mode http://line6.ideascale.com/a/dtd/Assignable-switches-within-LOOPER/799749-23508 2) While initially recording a loop, let the UNDO button cancel recording http://line6.ideascale.com/a/dtd/Looper-Undo-to-stop-recording-but-remember-it/789994-23508 http://line6.ideascale.com/a/dtd/Helix-Looper-Stop/789484-23508 If you agree, please consider upvoting those IdeaScale submissions. Thanks! Matt

OS X prior to either 10.8 or 10.9 (I forget which) has an earlier version of the generic Class Compliant USB audio drivers which can't connect to Helix audio. MIDI works fine, but audio is not accessible. I've tested using the same Mac with 10.6 and 10.9, and Helix audio works perfectly in 10.9, but is not seen at all by the Mac in 10.6. If you can't or don't want to upgrade OS X, it might be possible to install the custom Line 6 driver on 10.7, but I have not tried this, and have no idea whether it's compatible or not. http://line6.com/support/topic/19809-helix-mac-driver-100-public-beta-now-available/

Nope, it can't, unfortunately.

Yup, so they're great as cab+mic models, including maybe a handful of early reflections, but not usable as reverbs the way DAW-based IRs can be. Thanks for posting your results for others who might be curious!

Sure, I'll buy one, if it makes me able to play like that...

I don't have my Helix in front of me at the moment, but it should be possible to set up a split, with one "bypass" path and one "effect" path. Then use a switch to only allow one path to be heard at a time, either with bypassing Volume Pedal blocks set to 0% volume, or if you're out of blocks, manipulating the Output parameter of the last block in the chain.

:o :blink: I thought it was 1997 and I was running Netscape Navigator there for a second. :D B)

#1 is definitely on my wish list. I would even be OK with requiring a block for a level meter - which you could "pick up" with the Action button and move around to different points in the chain. I've had to actually create a path with all effects bypassed and at unity gain, route it into USB 1/2, and connect it to a DAW --- just to check my mic gain level (using the DAW's meters)! This kind of MacGuyver-esque chicanery is tiresome. Helix would be immensely more satisfying to use with a basic metering block.

You could even make your own IRs of your specific cab/mic/room, too!

If you're interested in a quick reference to how EQ works, check out this video at 6:14: And this video to understand all the types of EQ included in the Helix and other devices: I hope those are useful.

Simple answer: no. :-)

After reading your description of the symptoms more closely, it sounds like you're saying the Helix always sounds like that, regardless of what it's connected to, and whether or not a computer is involved at all. If so, there is definitely something very wrong with your particular Helix. I would suggest: 1) Firmware reflash 2) Experiment with changing the sampling rate in Global Settings If those don't help, I believe you will need to take it in for repair or exchange. (In my non-official opinion - I don't work for Line 6.)

The specific sound here is a periodic dropout. You can see it in the sound file - around 70-ish times per second, there's a dropout a short burst of near-silence. It looks like it's happening in the analog domain, or at least somewhere upstream of the ADC, because the dropouts are not perfect digital silence. It could possibly be a driver or other software issue. Does other audio show that same defect when recorded through the AudioBox? And, if you plug the Helix into speakers and not into a computer, does it still sound choppy like that? If you have both the Helix and the AudioBox connected to the computer with USB at the same time, try unplugging the USB cord from the Helix before recording its analog outputs into the AudioBox.

Hi everyone, This year I had the honor and privilege of helping create this petition, which was started by some top-notch professional engineers. I wanted to share it with you all in Line 6 land: https://www.change.org/p/music-streaming-services-bring-peace-to-the-loudness-war If you've heard of the "Loudness War" (which is not really about loudness at all - it's about sound quality), please consider signing it and sharing it with others. The petition has been live for less than 24 hours, and we've already got more than 1,000 signatures! Basically, if music streaming services start normalizing loudness (not peak normalization - loudness normalization), then artists will no longer feel pressured to use more dynamic compression than their artistic vision requires. They'll be able to choose however much (or little) dynamic range they want, without worrying about being too quiet or too loud. Everyone wins! Thank you, Matt

I agree with Gawwuf above. Can you make a recording of what you mean?

QSC K8! Very happy with it and have used it with a variety of modelers over the years. I've used it as my sole guitar rig (even when not running into the PA) and it has always cut through.

The reason for this is that IRs used for speaker cabinets are very short - up to 2048 samples, which is about 45 milliseconds. Those don't take up much memory space or DSP power. Hall, room, and spring reverbs require much longer IRs (up to several seconds or longer), which means they use much more memory and processing power. It's likely that the reverbs in the Helix are algorithmic - that is, generated on-the-fly instead of measured as an IR. It's possible for programmers to create excellent algorithmic reverbs that use much less processing power than IRs.

If you've rehearsed full volume with the exact patches and similar PA, you might be fine, but I'd still recommend you bring your amp or previous modeler as a backup. Expect to have to make some adjustments / iterations from one gig to the next. Good luck!

By analogy, you can imagine a synthesizer that perfectly re-creates the sound of a specific piano, let's say Steinway Model B, Serial Number 123456. If it's a perfect synthesizer, then that means you can take a recording of synthesizer, and compare it to a recording of that actual piano (mic'd the same way as the synth and played the exact same way), and nobody would be able to tell any difference, ever. What's missing here is that there's a difference between a recording (which is played on speakers) and a live piano in the room (which produces sound through strings and wood). If you compare apples to apples by listening to the two recordings, played through the same speakers, then there is effectively no difference. But any time you compare the sound of a speaker producing sound, vs. complex physical objects producing sound, you're comparing apples to oranges. They are fundamentally different. By analogy, a guitar amp modeler might be so good that, in a blind test, you cannot tell a recording of the modeler apart from a recording of that same amp mic'd the same way. I believe the Helix is this good, and you might conclude that yourself after hearing the sounds in the link Silverhead posted above. But, remember what the modeler is doing - its XLR outputs are basically equivalent to the mic cable coming out of a soundproof studio where the amp sits. It's not replacing an amp sitting next to you; it's replacing an amp in an invisible soundproof studio.

That update doesn't contain drivers or anything that would change this behavior. As Digital_Igloo mentioned above, the fix is going to be audio drivers that you install on your Mac.

Hi Megapixel, I'm not Digital_Igloo, but I may be able to explain what you're seeing. I'm not an audio hardware or driver developer, but I do know that there's much, much more complexity inside how a class compliant USB audio interface works than what appears to you or me on the surface. If you check out the link my comment above, http://line6.com/support/topic/17903-mac-os-x-10113-doesnt-fix-helix-usb-issues/?do=findComment&comment=134464there's a partial explanation for this. Basically, there are about a million (well, several dozen) subtly different ways that a particular audio interface might negotiate its communication with a computer. Different audio interfaces use different techniques. Somewhere in a huge PDF or Web knowledgebase is a list of all the available techniques that are considered Class Compliant, and OS programmers and interface manufacturers agree to implement those techniques -- but they each do it very slightly differently. Those protocols and options are so numerous that some unpredictability is inevitable. In the case of OSX 10.11, Apple made some small tweak compared to 10.10, which interferes horribly with the very specific combination of protocols used by the Helix (and a few other devices) ---- but doesn't have any effect on the (different) combinations that many/most other audio devices use. I hope that makes sense, and if there's anyone here who actually designs interfaces, programs audio device drivers, or otherwise knows CoreAudio internals very well, they might be able to shed some more light on this. Thanks, Matt