mdmayfield

Thanks a lot guys! Glad it's useful.

I'd start here: http://lmgtfy.com/?q=frfr+site%3Aline6.com :D No seriously, there are dozens of threads on this, and a lot of good advice - take a look through those for many different suggestions.

This is interesting - another manufacturer dealing with a similar issue. The news on 10.11.4 so far looks promising: http://dj.rane.com/support/knowledge-base/el-capitan-update/ They describe that in their USB 2.0 audio class compliant devices (of which the Helix is also one), they're using a specific clocking technique which had been recommended by Apple several years ago: generating its own clock asynchronously (i.e. independent of the OS) to serve as the master, and using the "implicit feedback meethod" which I'm not familiar with in detail, but it sounds like one of several ways to implement this. Pops, crackles, and other glitches along those lines are nearly always clock issues, and so it seems reasonable to assume that the Helix is probably using the same technique since it has the same issue (under the same circumstances, OSX 10.11.0-10.11.3). It sounds like the developers at Rane are also seeing an improvement in El Capitan 10.11.4... So if that means the Helix's audio is also fixed, maybe Line 6 won't bother writing a custom driver (or whatever it is they've been working on) after all.

One quick clarification on this... You can't turn off the mic simulation only. The cab simulation includes the mic. It's essentially an IR, that is, an actual recording of a mic'd cab - kind of like an audio photograph of the sound of a specific cab through a specific mic in a specific position in a specific room. The only way you could create a cab simulation without a mic simulation would be to make a recording of the cabinet without using a microphone, which is not possible - it's like trying to take a photograph without using a camera.

Are you using Amp blocks, or Amp+Cab blocks?

Dang it, every time I check the forums I see this thread title, and it makes me all excited before I realize it's a joke! You got me twice just today! :angry: :D

I've been using a QSC K8 for many years, and it's served me well through an XT Live, HD500, and now Helix. It's loud enough to cut through in the pop/rock bands I've played in. I've never played with very loud drummers, but also have never even approached the K8's limits - not sure where its limits are.

I think that this thread reveals who has worked in tech support or software development, and who hasn't. :-)

The Helix has been 100% rock-solid stable for me over approximately 150-200 hours of use in the last few months in these circumstances: - As a standalone guitar + vocal processor - As an audio interface with OSX 10.9 Mavericks using Logic 9 and CoreAudio built-in drivers - As an audio interface with Windows 8 using Reaper and WASAPI drivers The only issues I have had have been with ASIO drivers under Windows 8 (Reaper would crash); the Helix patch librarian application on OSX (it would scramble patch names vs. patch contents); and one time that I accidentally set up a MIDI feedback loop, which caused the Helix to stop responding to MIDI.

It's two guitars and a bass - so that's why I was assuming that, if it's a fairly traditional rock sound, it should be practical not to use an amp simulator at all on the bass. Add some EQ, maybe some compression, and the bassist already has a fancier sound than they would if they were plugging in direct through a DI box as many bassists do.

This is because 1A and 1B share DSP #1, and 2A and 2B share DSP #2.

No, it works fine. You just need to spread the load, with one amp on path 1(A or B ) and another amp on path 2(A or B ), rather than trying to put two amps on one (1A and 1B).

Yes, it's possible, although there are a few disadvantages: - You'll have a single point of failure for all three stringed instruments. - It would be hard for multiple instrumentalists to switch on/off their effects without (literally) stepping on each others' toes. - Editing individual people's tones becomes inconvenient, because they're all saved in one patch together. - Unless your setup is fairly light on effects and amp sims, you'll have to carefully budget your DSP resources. To clarify, there are four potential paths: 1A, 1B, 2A, 2B. 1A and 1B share one processor, and 2A and 2B share the other. There's a limited amount of DSP resources per processor, and some effects (like EQ) use up less than other more "expensive" effects (like amp+cab sims, pitch shifts, and distortion). If all you're using for effects is an amp sim, some reverb and delay, and maybe a distortion pedal, you'll probably be OK.

I believe it would work like any classic Variax 300/500/600/700 - over VDI the Helix will power it and pass audio, but will not control its model selection, volume, or tone. This might change in a firmware update, but no guarantees.

The difference in quality between A/D converters these days is microscopically insignificant compared to volume, EQ, amp/effect settings, and other mixing choices.

Yes, that's right, those are marketed as for a Telecaster. They're the only ones I know of that are available in a complete set of 6 with the screws on the correct side, AND the back "open" so the string could be routed through them like the original JTV saddles. I don't use the vibrato at all - I tightened the screws all the way, damped the springs' vibrations with foam, and blocked the whole mechanism off with wood - so I never had any problems with tuning stability and don't notice any change. No pictures at the moment, sorry. Someone else had some about a year ago, maybe Charlie_Watt? Not sure who it was.

I used a set of PN-8102-R0 saddles which fit my JTV-69 with some modification (the intonation screws were at a weird angle and had too large of heads - I had to drill out the screw holes in the back of the bridge). However, except for the volume gain, I don't find that it radically improves the sound. Any change is so subtle that after matching the volumes with my other unmodded JTV-69, I hear a tiny difference comparing recorded samples one right after another, but couldn't tell which one was playing if you only played me one. My reason for switching was string breakage. I broke a string at the saddle on my Strat just before a solo at my very first public gig when I was 19, and ever since then have used String Savers on every guitar I've owned.

I don't have my Helix in front of me ATM but I could imagine beginning to approach a random-ish delay/chorus kind of thing using a couple different choruses or delays in series and using multiple paths. Hard to describe but I'll try: Path 1: Guitar --> split to path 2 ---> combining point --> amp, reverb, etc. Path 2 --> Mod/Chorus Echo at 100% wet, shortish delay and fast speed ---> Mod/Chorus Echo at 100%, longer delay and medium speed --> combine back to path 1, mix to taste The end result would be random-ish sounding, if the two delay times + speeds are not mathematically related to each other. That is, their measurements should have no common factors, and have a very large Least Common Multiple. You could also set the second modulation effect to <100% mix to thicken the sound, or even expand the general pattern to include another path & mod effect for even more pseudo-randomness.

Line 6 use a service called IdeaScale to keep track of feature requests. Line 6's page on IdeaScale is here: http://line6.ideascale.com Your best bet is to go to IdeaScale (create an account there if you don't have one) and 1) Check to see if anyone else has already proposed this; and if not 2) Create an "Idea" (feature request) for it there. They have to decide how much programming time to invest in which ideas, based on how much effort it would take to add the feature vs. how many people want it. My educated guess is that this wouldn't need very much programming effort/time.

My two cents: - Headphones are already generally "FRFR" in the original sense of how the term was used regarding modelers. (i.e. Guitar Cab vs. FRFR) - Even if you have near-perfectly flat-response headphones, the physics of how we hear are different in headphones vs. with speakers. (due to room acoustics) One reason it's harder to mix (or create patches) with headphones is that they're acoustically isolated. Each ear only hears its one side, and there are no room reflections like you would hear in any real-world situation. I also suspect that by playing sounds on speakers, the changing frequency response of the acoustic space as you move around (even just by turning your head a bit) highlights random different aspects of the sound over time, drawing your attention to things you might not notice with headphones.

Interesting.... Maybe try turning down the master output on the Helix. Check if the clipping goes away (even though it'll probably be quieter than you want). If it does go away with the Helix's output turned down, I suggest you leave it turned down, and turn up the gain on the L2T to compensate - and see if the clipping comes back. If the clipping does come back after adjusting the output gain of the Helix and the input gain of the speaker, something very fishy is going on - unless your speakers just can't get loud enough, that situation should be impossible (as described / as I understand it), so in that case there must be something I'm misunderstanding. Also, I'm not familiar with the L2T, but if it has a choice between Mic and Line level, try switching it to Line. Also, see if there's a difference in connecting the 1/4" outputs of the Helix vs. the XLR outputs. The only other possibility is that there actually *is* distortion happening within your patches, but that it's being masked somehow when you connect via L6 Link. That seems extremely unlikely to me so personally I would recommend trying *everything* else before worrying about that possibility. (FWIW, I have used the Helix analog outputs with QSC K8 and K10 speakers, without ever running into clipping problems, so there's definitely a way to make it work.)

Consider the signal path: Guitar -> Modeled effects/amps -> L2T via L6 Link Because there's no clipping with this situation, we know the clipping is NOT coming from the guitar or the modeled effects/amps. Therefore there is nothing to fix *within* your patches. It's different with this signal path: Guitar -> Modeled effects/amps -> FRFR via analog output The clipping must be happening because the *output* of the Helix, at its current volume setting, is currently too strong for the *input* of the FRFR speaker, at its current Input Gain setting.

What a deal!

Hi Kedhrin, Yes this is possible, but it will be more like specialized programming (requiring a lot of research, preparation, and testing) than any kind of "just hook it up and it works." You'd need to know (or look up) the messages in the MIDI language that the AxeFX expects (i.e. Continuous Controller, Note Ons, Program Changes, or other messages), and then manually assign those to footswitches of the Helix. Then you'd connect the MIDI Out of the Helix to the MIDI In of the AxeFX. If you haven't done much work with MIDI, and don't have a lot of spare time to really dive into the details, it may not be worth the effort... but if you're a seasoned MIDI expert, it shouldn't be too much work. I hope that helps. Matt

Few things: 1) Not every single combination of settings on the Helix will be better than every single combination of settings on the XT Live. 2) Your ears & brain are used to the XT Live, and people naturally like what they're familiar with. Also, the XTL is fine; it's not bad. 3) Volume differences HUGELY mask tone differences. Louder seems better in the short term. Make sure you are matching volumes carefully.