jeremyn

I do this with the Helix floor. Piezo (pre-amped in the guitar) goes into the aux input (or one of the effects returns), and guitar into the main input. The guitar path has a guitar amp and some effects, stomp boxes, cab, eq, etc. The piezo path has a 3 sigma IR on it, some compressor/chorus/reverb/etc options. Look for Peter Hamm's most excellent youtube examples of this on Helix rack. I assume the same would be true for the Helix Effects just without the guitar amp options.

Tech 21 Sansamp Blonde or Liverpool are great 'end of chain' amp/cab emulators for people with pedal boards that want to go straight to the desk.

The issue with the Helix is only tangentially due to the increased granularity. If the tuner displayed the exact offset of the string instantaneously with a display bandwidth greater than the movement of the string/noise of the input signal, then it wouldn't 'jump' around, it would 'move' around. Big difference. One amounts to 'what on earth is going on here', and the other is 'wow this tuner is so accurate I can see the movement of the string'. A strobe doesn't have to come to a full halt. When tuning quickly on stage, I just let it approach the 'zero' speed as it slows down in a predictable way to how much I turn the knob on the tuning peg. I can very quickly get to that point, as the 'ideal/optimal' tuning peg position is really all I'm after. It might get to zero as it crosses the threshold for a 1/10th of a second or less while I'm spinning the knob. It helps that the Turbotuner is instantaneous, with no frequency discrimination, hysteresis, or hold time between the input and the LED display ring. I have the Peterson iOS App and also their little headstock tuner. They are sometimes also both a little 'jumpy' compared to a true stroboscopic display. It's like the Petersons are trying to momentarily hold or discriminate the current input frequency as it gets close to zero (or too weak?) until a threshold is reached and then it resamples. It's slight 'jumpiness' is not just because it is 'stroboscopicaly' accurate. But, this might be due to overall sensitivity as I don't plug my guitar directly into either of these devices. Unlike the Turbo, the Petersons don't show the actual input waveform since they transform it into their sliding bar display. The Turbotuner being a full bandwidth display of the input signal is so smooth that I can instantaneously see the actual string change. When it 'jumps' around (usually a sudden small drop in frequency), it's almost always accompanied by the string slipping in the nut to whatever micro degree that happens. As you say, old strings seem to do this more than new ones, but probably because I always lube the nut and saddle when changing strings. I still don't call this 'jumpiness' when tuning, as it's readily apparent the instability is in the strings and guitar, and not an artefact of the tuner. I also tune to the attack, and don't wait 5 seconds for the vibration to completely settle - the longer you let it ring, the more the sympathetic vibrations from the other strings become visible. As that happens, on the Turbo you can sometimes see two separate signals rotating around the display as the primary signal gets weaker. The eye can easily track this and ignore the second - this doesn't happen on the Peterson, so I suspect it is internally trying to discriminate the signals internally prior to displaying the bars. The Helix on the other hand is all over the place compared to the Peterson and the Turbotuner. For me anyway. There's probably an optimal level and input filtration (eg. volume at 3/4, neck pickup, tone rolled back slightly) where it happens less than all knobs wide open. But, I don't like to reset the guitar when checking tuning in the middle of a set. I can deal with the jumpiness because I can still get the guitar sufficiently in tune by guessing the median point as the bars jump around. But, it takes me a little longer than I'd prefer - especially compared to other tuners like the Turbo/Petersons/Polytune/Korg/Boss that I use or have used over the years. Clearly you don't have any problems with the Helix tuner, and if you did, you'd probably mention that as an issue while you were beta testing the product. Unfortunately, that is not the experience of many other people including myself.

Accuracy includes a resolvable output. The Helix display is sample aliasing to such a degree that you can't claim greater accuracy since it isn't predictably resolvable on the display. The Turbotuner is incredibly accurate and precise (as are many other non-strobe tuners), and yet smooth and easily resolvable visually - even when tuning to the attack which peaks sharp and then predictably flattens over time as the string settles. A tuner can be considered 'stable' as long as it is smooth enough to allow the eye/brain to interpolate the information it needs.

It also depends on the mic. Some condenser mics might have been designed to minimise the impact of hot-plugging into a unit with inadequate input protection. Also, never plug an XLR output from powered equipment (eg. wireless receiver, outboard effects units, etc) into a phantom powered input (Helix or otherwise).

I don't know the circuit details of the Helix headphone output, but if it is not properly protected against short-circuit conditions, then it may end up driving excess current (beyond the rating of the output amp) into the load. That could eventually cause the amp output stage to fail. Solid state amps don't have a problem with driving higher impedance loads, they just get quieter as they can't usually drive high enough voltages to get the full power across to the speaker. On the other hand, tube amps with output transformers become inefficient outside of their nominal impedance operating range. If you leave the output transformer winding incorrectly terminated (ie. with a short circuit or with an open circuit) then the transformer can be damaged (and/or output tubes due to flyback). It is never advisable to significantly mismatch the impedance on a tube amp, as the amp can be damaged. Some amps protect against catastrophic failures to some degree and will pop a fuse. But, older style amps will just end up with blown tubes, fused circuitry, and melted windings in the output transformer.

4 is luxury. On my MacBook Pro (i7 quad core, 2.7GHz) with 16GB of RAM, I can't even get a single instance of Native running on Logic Pro X without it glitching out on complicated dual amp (factory default) presets. Works ok on simple presets, but dies on playback with anything more involved. And that's with no other plugins, buffers/latency set to maximum size and everything set to optimise/minimise CPU impact. One CPU core is fully saturated, while the other three are basically not doing a lot. It looks like Native is only trying to use one core. So if Native on a single core works, then 4 instances would eat 4 cores and should still work. I have no idea how people are getting huge numbers of Native instances going unless they have big machines with a bunch of grunty CPUs.

That cut may be there to compensate for the cable resonance of that particular pickup/cable pairing. Different cable, different cut (or no cut depending on where the resonant frequency ends up).

Actually, I was just thinking earlier this week how many extra Mesa Lonestar and Friedman BE amps will get bought just on the awareness and strength of the models in the Helix.

A 'fixed tuner' isn't going to totally change the behaviour, it'll most likely just properly damp and anti-alias the display of the frequency delta to make it behave like pretty much every other tuner out there. I don't think the few people that are happy with the current tuner will complain if it starts to behave more like a Korg, Snark, Boss, TC, etc. Yeah, I'd love if they added a properly anti-aliased strobe tuner, but I doubt that'll ever happen, and I'm not fussed either way if they never do that. I have a bunch of Turbo Tuners that I use regularly, and love the strobe tuner concept. I get by with the existing Helix tuner; the top bars jump around but I can work around that. To me it's kind of like a glaring spelling mistake on the home screen. There'll be a bunch of people that don't care, others that will say the spelling mistake 'adds character', and a final contingent that will complain that it ruins the rest of an otherwise great product. When someone fixes it, there'll be cheering from some, indifference from others, and some booing from the vocal minority. It'll settle down and everyone will forget about it. This isn't like a major rewrite of folk history where people still complain that Han Solo shot first and Greedo never shot at all.

Depends how hard you hit the strings. But, otherwise, yes. Don't underestimate the monitoring/amp output level. When the sound guy turns up (within reason) the monitor channel for an acoustic player, the player will play softer to compensate to get their level. Same applies to a singer's monitor channel. Louder the monitoring, the softer the singer will sing. Same applies if there is a lot of ambient or background noise. People subconsciously (and sometimes consciously) tend to play/sing a little harder to improve the 'signal to noise' ratio. EQ applies here also, make the person unable to hear themselves over the 'mix' and they'll play/sing/talk harder/louder. So if you want a consistent isolated recording level for a channel, you need to have a consistent setup and environment.

If you're using a Mac with OSX, you can set up an aggregate device to combine multiple USB interfaces and simultaneously record as many channels as you want. For example, you could record the Helix direct through USB on some channels, use the Focusrite for your other mics, and possibly even a third USB interface for other mics and instruments. I have no idea if this is possible under Windows. Also, latency shouldn't be an issue if you're not trying to monitor the recording inputs direct to an output in a live setting. For monitoring as the recording engineer with headphones or in a control room, a sub 40ms round trip in to out latency that most USB interfaces achieve even on really old hardware isn't really a problem.

For me, automatic noise gates work best when the guitar has quiet/noiseless pickups in it. That way the thresholds can be set low enough that the gate cuts in when there is only non-string related noise coming out of the guitar. Automatic noise gates never work for me to quieten down a single coil (or noisy humbucker) that is humming/buzzing away in the background. Usually that noise is at the same level or greater than the trail out volume of the note or chord. Then the gate thresholds need to be set high enough that it affects the note trail out or when playing softer passages. There are many solutions to noisy pickups, but IMO, a noise gate is on the bottom of that list. The EHX Hum Debugger is a pedal that filters out the 50/60Hz hum, and it wasn't too bad when I tried it years ago, but it added a bit of a metallic ringing to the sound. My solution these days is to get quiet pickups for the guitar. The Zexcoil offerings are my current favourite for Strat single coils. I used a Tom Anderson guitar with his PQ series pickups (quiet P90s) and they sounded as good as any standard P90 that I've tried. Active EMGs are really quiet too, even their ones aimed at vintage style tone. Proper shielding and grounding helps too. I only enable the noise gate block in the Helix for high gain chugging, and set the threshold high enough to give it that silent mute effect. This is definitely not a setting I'd ever use for normal clean or crunch tones.

Don't worry, there's no way you're going to blow the guitar input (or AUX and loop return inputs) with a typical active guitar pickup or wireless kit.

The wireless kit acts as an active buffer which can be plugged into any of the inputs. The AUX input is fine for the G70 as is any other input. The input impedance function won't affect your pickups because the wireless kit is effectively buffering the signal. Whether or not your guitar has active pickups won't make any difference once it's gone through another buffer (ie. the wireless kit). The reason they comment on active pickups is probably because there is a little more headroom in the AUX input as it doesn't have to process ultra small passive input signals. It might be a bit quieter for this reason too. That doesn't mean you have to. As long as the active pickups don't have such a massive output signal that the digital input clips at full scale, the guitar input should be fine. If it sounds good with heavy strumming and doesn't have any nasty digital distortion, then you're good to go.

A properly designed phantom capable input channel should not have a problem. There were many desks I've seen back in the day that had 'dead' channels. Almost always the ones with phantom power on them. These days, most desks have good protection (or input transformers) to avoid problems. Hot plugging into phantom powered inputs has always been discouraged. The classic where a singer plugs their active mic (or an instrumentalist an active DI) into a phantom powered input while the system is active usually involves some serious pant lollipopping from anyone in earshot of a speaker when the channel isn't muted. Regarding the dynamic mic. A typical dynamic like an SM58 or e835 doesn't have a current path between the shield/ground line and the signal lines, so it shouldn't cause a problem unless you short the ground to one of signal pins while plugging in (which doesn't happen on inputs as the input side XLR is female). However, it could be a problem if you have one of those 1/4" unbalanced output mics that effectively shorts one of the signal pins to ground. Again, this shouldn't be a problem if you're already plugged in while the Helix is off (and has been off long enough for the coupling and supply capacitors to discharge). With insufficient technical information from Line6 on the electrical characteristics of the mic input, there is no way to make a call one way or the other. Maybe just having phantom power enabled at all is enough to fry the input if components haven't been properly rated and the circuit correctly designed. Who knows? (Well Line6 does, but gone are the days of published schematics and technical repair manuals in a throw-away society where it costs more to repair something than to replace the entire unit.)

Phantom power shouldn't be a problem when unplugging, unless you take a long time to unplug and keep 're-plugging' it back as it makes/breaks contact. But, that should be rare. The biggest problem is the action of hot plugging a condenser mic (or any other powered device) into the unit. At that point, all the charged coupling caps are discharged through the protection diodes of the input stage (ie. the Helix), and into the mic's power cap. Condenser mics are usually designed so this is not an issue for the output stage of the mic. Dynamic mics usually don't cause a problem as they don't have a high current path between the shield and the balanced terminals. This is why dynamic mics aren't (generally) affected by phantom power. I'm too scared to turn on the phantom power on my Helix so I only use dynamic mics with it. I was super careful the couple of times I used a phantom powered mic to make sure everything was off before plugging in the mic (just in case phantom power was already on), then turned on the unit, and turned on phantom power. When the phantom is enabled (or the unit powered up) after everything is already plugged together, the phantom pull up resistors limit the current to a few milliamps as the caps are slowly charged and the phantom voltage rises to full rail (48V). Hot plug after the phantom power has charged those input coupling caps and the current is no longer limited by the phantom pull up resistors. A properly designed interface will limit inrush current flow and/or protect itself from huge inrush currents.

It depends on your vocal style and how important it is for you to be able to 'work the mic'. Albeit, any mic needs to be 'worked' to some degree, but a live vocal singer that is used to a handheld dynamic mic will have many varied techniques to adjust tonality based on mic angle and proximity. Whereas a large diaphragm condenser can't really be 'worked' in the same way, even though a bit of proximity awareness is still usually necessary if you go from whisper quiet to all out belting. As you get close to the mic the proximity effect tends to enhance the bass creating a more personal 'in your ear' tone. While as you get further away (or move off-axis) the voice sounds thinner and more distant. A good vocalist will use this to their benefit to create appropriate tonality based on the emotions they are trying to convey. Proximity combined with voice tone adjustment (ie. going from gentle low tones and soft head voice for intimate passages, and then moving up into full voice or belt for more energy) can work to great effect. Too close when belting sounds too harsh, back off and it thins it out a little making it sound more natural. The best way to think of it is, in the real world, loud sounds tend to come from further away, and softer sounds (that you want to hear) come from closer. My go-to for live handheld vocal mic is a Sennheiser e935 or e945 (e835 is good it too for cheaper), and sometimes a Rode S1 handheld condenser for more intimate environments. These mics have good frequency response curves and are easy to hold and work. No handling noise, and good off axis rejection, and they don't break the bank. They can be used for both live and recording environments. For recording I have a Rode NT1 large diaphragm condenser with a pop screen that is great for low noise recording environments. Very 'detailed' and sounds a little more 'open' than the handhelds, but I've recorded with all of the above mics and there isn't much between them once you've tweaked the EQ if you just stay 6 inches away and sing. The lower noise floor of the large diaphragm mics is handy, but not a deal breaker if you sing at a decent volume anyway. OTOH it's really useful if you have a quiet singer that stands miles away from the mic. I avoid LDCs for live use. IMO, a handheld dynamic mic (Sennheiser e935, e945, e835, Shure SM58, etc) is always a useful tool to have in your box. So even if you branch out from there, a dynamic will act as an indispensable backup for recording, or as your primary mic for live performances (even if it's not you singing into it). Dynamic mics also don't require phantom power and don't have to be babied to make sure you don't damage the element or blow up a phantom powered channel input by hot-plugging them. And in a pinch, you can point it at your (or someone else's guitar speaker to mic the amp/cabinet.

When our keyboardist gets it wrong, and it's the loudest instrument, the singers automatically sing in the key of that instrument. And the bass player might have just automatically adjusted. Leaving Eddie in the wind trying to compensate. Although, you might be right and the guitar isn't an even semi-tone out. It's half way between keys. So nothing Eddie could do even if he wanted to. It's a train wreck no matter how you cut it.

I have no idea what their actual set up is. But, if the synth is like any other keyboard with a transpose option, it is not out of the question that the synth player had forgotten to reset the transpose from the previous song. I know that's happened more than once to me. But, you're right, the most likely problem there is the guitar was not correctly tuned, or if it was a floating trem, a broken string caused the rest to go way out of tune.

Another possibility is if total system gain/level/volume is too low, it tends to cause the player to hit the strings harder increasing fret rattle in the first place. Small changes to volume or EQ can make things better or worse. This is because the player is part of the guitar/amp/speaker feedback loop. If the guitar is already marginally setup with uneven frets, incorrect relief, or super low action, this effect would be exacerbated. It’s like playing an electric guitar unplugged. It tends to rattle because you hit the strings harder to hear it. Plug into an amp and you play softer which reduces or eliminates the fret rattle.

Logic Pro X on my 2013 MacBook Pro (i7 2.7GHz quad core, 16GB RAM, 1TB SSD) goes from a few % CPU usage to two cores fully pegged at 100% when I enable Native. And then Logic Pro X brings up System Overload messages (or glitches if I turn that off). Latest versions of everything. This is just with a pre-recorded track playing back through the in-built headphone jack and nothing else plugged in or running. Buffer size changes help a tiny bit. What helped the most was changing the Logic project sample rate to 48k instead of 44.1k. That let me run some simple presets without glitching. But more complex ones like 24C Royal Duo still barely work at all. The 48k thing is probably related to the inbuilt hardware only supported 48k in hardware, and forcing some sort of software resample (which is clearly problematic) when changed to something else. Even though this helped (and probably worked around one issue), the problem still remains that I can't even run a single instance of Native (with nothing else happening in Logic) for anything but the simplest single Amp+Cab patches. I'm yet to try it with a live external input source like the Helix floor as the interface. There must be a solution here, because it's hard to believe that this CPU/system is deficient. Might be a MacOS related problem, or possibly even a hardware/driver interaction issue since I am using the inbuilt interfaces which use who-knows-what chipsets.

Even though I have a Helix floor unit and hardly ever record seriously, I picked it up at the discount simply because I'd like to re-amp some tracks and do some tone tweaking when I don't have access to my guitar. It looks like a simpler workflow to play with settings on the computer until they are close enough and then bring them across to the Helix floor for final tweaking for live use. So I just use the Helix as the interface for the initial recording. And then can play with various settings to fine tune while the tracks loop around. My only issue at the moment is that Logic keeps bringing up the "System Overload" dialog after a few seconds with even for the simplest patches in Native. I'm sure I'll get this sorted at some point.

Being able to reliably replicate a bug is 99% of the way to getting it fixed. Back up the 'bad' patch list if possible so Line 6 can try to replicate it too - if they don't already know about it.

See this thread: http://line6.com/support/topic/30581-helix-230-firmware-failure-boot-failure-entered-update-mode