jeremyn

Make sure the phantom power is off at the desk for your channel.

And those 12" guitar speaker plots are on-axis. As you move out of the centre 'beam', the higher frequencies roll off even faster. For most guitarists standing next to their amp on stage, there is virtually nothing left past about 6kHz. Definitely nothing that can be heard amidst the rest of the band.

An acoustic guitar amp is basically a mini (or big) PA speaker with instrument level inputs, some EQ control, and maybe some combination of effects like compression/reverb/chorus/delay. The Helix has all of that already. For acoustic tone, less is usually more. But, there are a couple of things that would enhance the acoustic guitar experience: 1. An 'acoustic body' cab. Sometimes a little bit of guitar body resonance can help (eg. Fishman Aura), and the IR module is great for that. I'd like to see a built in 'acoustic cab' that allowed you to dial in a good body image for a given guitar - kind of like how that Bodilizer application works where you can dial up/down various body parameters (like depth, hardness, width, etc) to modify the body resonance to taste (rather than a fixed IR that can only be blended to taste). 2. An automatic feedback filter. This is useful when the volume goes up and the guitar starts feeding back. Being able to just stomp a button to automatically find and kill the howl by setting a notch filter would definitely be helpful in those situations. For acoustic guitar, the Helix already seems to have everything else.

At present an effect can only be controlled by one switch (but, one switch can control multiple blocks). I think everyone's waiting for a feature update to allow either some sort of 'scene' switching, or a more comprehensive switch assignment matrix. There are a few items on Ideascale that describe this. Sign up and vote if you haven't already.

Anything like the Radial ProDI, JDI, Stagebug passive would do if using the 1/4" outputs. The Whirlwind ISOXL looks like an inexpensive compact transformer for line level XLR to XLR isolation. Basically anything that says it includes an isolation transformer. There are products from Art and Behringer that meet this criteria for multiple channels. Google shows heaps when doing a search for XLR isolation transformer.

Typical guitar speaker rigs are handy for personal monitoring for the simple reason that their tone can be effectively equalised simply by changing position relative to the speaker's projection axis. Down the beam and you get lots of frequencies 1kHz+ up to the natural speaker roll-off. Move out 15 degrees off axis and those 1kHz+ tones start to roll off much faster than the sub 1kHz tones. Most guitarists like to stand 45 degrees off axis, which is nice 'mid' position. That's also why guitar speaker rigs aren't good for FOH coverage. You get different sections of the audience hearing vastly different guitar tone. 4x12 and 2x12 horizontal are the worst offenders. Micing up a guitar cab presents similar problems, and that is why mic position is so critical to get right for FOH. Compounding the problem is the speaker responds differently in the near and far fields, so a close mic at 45 degrees off axis doesn't sound the same as standing 6 foot away at the same off axis angle. Player foldback and FOH are also two different requirements. The guitarists needs to hear and 'feel' their tone appropriately otherwise the player is endlessly changing their style / hand technique trying to get 'Their Tone'. What the audience hears is a full band mix plus room and crowd response, and the guitar tone will need to be EQed and other fun things done to keep it sitting properly in the mix. Practicing with a FRFR (or IEMs) helps greatly to re-tune your ear to what you're expecting to hear coming out of the gear mix. That feed can then be further EQed and compressed for the FOH mix. But, like IEMs, a good FRFR box won't tonally change much by moving around it, so you need to get the tone (and mic sim position) right in the modeller or it won't "sound right" - especially if you're used to hearing a subdued 45 degrees off axis to your speaker while the FOH gets a nice bright mic'd signal. With praise & worship live music (and probably every other kind of live music), nearly all guitarists (and singers) complain the first few times they are introduced to a "silent stage" & IEMs. The fact is that their tones sound much much better in the FOH of the same venue than they ever did before. Just not to them - and that is half the problem. So how to get the player happy with their own tone? The first is to understand the varying EQ and mix requirements and adjust them so the player hears something much closer to what they were used to when using a regular 12" speaker guitar cab on stage. The second is from the players point of view, and is really an adjustment in expectations of what they want/need to be hearing. Old school veterans will probably have more trouble with the second part than newer players. I know this may not sound like an ideal solution for a lot of players out there, but the above is what I've found to happen when people move to FRFR and silent stages. It should also give a clue as to why things can't sound exactly the same to the player as an old school live stage with acoustic drums, loud amps, and cumulative ear damage. And, yes, there is definitely some magic about being inside that wall of sound when playing on a loud stage and both feeling and hearing the thump of the instruments. BTW, sorry for the long post.

The difference between a 'snapshot' within a preset and another preset, is that the software doesn't need to re-load any new effects or amp blocks. And therefore, won't have that discontinuity that happens when changing patches (see the other threads regarding patch delay for why something like this is useful). But, I suppose that within a preset, you could still use an external midi controller to fire off/on all the individual effects that you need.

This where some kind of 'scene' snapshot would come in handy. You set the effects the way you want and assign the current configuration to a scene snapshot. Then just assign the snapshot trigger to a button.

If you want to test your sound with the L2T set to PA/Reference mode, then make sure the L2T is on a pole at head height and not on the floor.

An orange squeezer is pretty much the definition of 'splat' when it comes to compressors. I'd also love to see a better typical 'guitar' non-transparent compressor added to the collection. My vote goes in for orange squeezer clone. That being said, I still use transparent ones for acoustic guitars and giving a little 'transparent' boost into the front of a distorted amp model for less 'gainy' distortion, while maintaining saturation at that lower gain setting. Kind of like the best of distorted sustain without the over-the-top 'can of bees' that high gain can produce.

The only problem with the Mission is the shallow throw is more appropriate for wah effects than volume swells. The FV-500L/H pedals have a reasonably deep throw, as do the Ernie Ball VPs. The inbuilt pedal on the Helix is somewhere in between. The whole pedal compatibility thing is pretty silly. I wonder if it's worth just getting a midi box that converts a standard pedal into midi. Does the Helix allow midi expression assignments?

I'm interested in this too. I'd love to see a nice synth pad effect added to the Helix. Think Boss SY-300 or, even better Roland GR-55.

The aggregate device is setup in the operating system. It effectively makes any audio software see a single device with lots of channels rather than multiple devices. As jshimkoski said, once Line 6 fixes the OSX USB audio issues, it should be simple to set up.

You'd never plug the output of your mixing desk, your wireless mic receivers, outboard compressors or other rack gear into the phantom input of a desk. The problem here is that people are assuming the Helix is an instrument, not a piece of pro-audio 'rack' gear. Whereas the designers seemed to assume it was going to be used like rack gear. Rack gear is probably DC coupled on the output side with drivers that maximise the SNR for ultra high fidelity studio recordings, and allows the user to choose the level of protection required versus any compromises to signal quality (and increased cost) that they entail. A manufacturer may add cost and compromise the output slightly with high quality coupling capacitors to block any DC, but this doesn't block inrush. Another manufacturer may choose to go to even greater expense and put in a high quality line output transformer (which would allow maximum flexibility for live use). If a device does not explicitly state that it is safe to apply 48V phantom power to the output, I would NEVER directly connect it to a phantom powered (or otherwise unknown) input. Even a device that 'seems ok' might be stressed and be damaged over time. One aforementioned example is the inrush currents when plugging in to an active phantom input (even if you use a non-transformer based phantom blocker) - a known problem with all line level outputs. This causes stress on the output drivers / protection diodes, and possibly the device's internal rail for a short burst while various coupling capacitors charge/discharge at high current through protection diodes. A carefully designed output would have appropriate protection and limiting, but there are no guarantees without examine the circuit in detail. A DC coupled output stage will have it's bias point changed by the phantom voltage causing it to behave asymmetrically, therefore increasing distortion and/or noise. This will probably not cause any damage (as long as the phantom power was turned on AFTER connecting the gear). Another problem is applying excess voltage to an AC coupled output stage where the coupling capacitors are rated below 48V. This puts stress on the capacitors and may cause them to fail prematurely. This may eventually manifest as reduced signal quality or complete loss of signal (and further damage to other internal components). This applies even if you take the precaution of turning phantom power on after all devices have been plugged in - as it is the continual presence of the 48V DC bias, not the inrush, that degrades the underrated coupling capacitors. Most devices will likely survive both of the above scenarios (depending on circuitry at both ends) for limited exposures, and seem ok for a while or even indefinitely. But, why chance it. A 'phantom blocker' that just uses a couple of inline capacitors to block the DC is an acceptable protection measure, but you MUST remember to make sure the phantom is off before plugging in, as it won't block the in-rush current. It might soften the 'blow', but again, why chance it. The safest option when connecting line output gear to an unknown (or known phantom) input is to use a fully balanced isolation transformer which are designed for this and other voltage differential problems (like mains current leakages superimposed on the earth line). The transformer will both block the DC bias AND the discharge inrush - so you are still protected even if you happen to plug in while a phantom powered input is active (which can happen accidentally during a momentary lapse of concentration, or a 'helpful' sound guy that turns it on when you're not looking). It also protects the mic input stage on the desk, which is also stressed during the inrush event. It is crazy to expect Line 6 to include free external protection devices, especially since most people will either not have a problem, or be already taking appropriate engineering 101 precautions with better quality gear than a 'free' inclusion would end up being. The kind of device that they include would likely not suit everyone anyway. I have a bag full of 'problem solvers' that include expensive Jensen transformer based isolators, DIs, ground lifters, pads, etc. If I was designing the Helix main output stage circuits, I'd have a DC coupled fully balanced and protected line output on the 1/4" jacks that could also be connected to an unbalanced input. And I'd have but a transformer coupled and pad switched output stage on the XLRs, calling them DI outs. That'd add at least $50 extra to the end user price of the unit - probably not an issue in this price range, but the marketing department would probably disagree. If cost wasn't an issue, then we could also put a set of level translating transformers on two of the send/returns to allow connection to old school tube amp loops that operate at high voltage (70V+). And don't forget a dual channel transmitter for driving wireless IEMs, and a dual channel receiver to allow wireless input of either two separate guitars (or a two voice guitar like the Crowdster). A couple of built in isolated 9V DC outputs for powering external pedals would be cool too.

It is never a good idea to connect line level powered gear to a phantom powered input without first considering a number of things: 1) If the gear has a transformer isolated output, then it is generally safe. 2) If the gear is DC isolated, and has phantom inrush limiting and diode protection on the output, it is also generally safe. 3) If the gear has line drivers that are unprotected or marginally protected, then make sure all gear is plugged in before turning on the phantom power. Plugging in while phantom is already powered on will cause the coupling capacitors to discharge into the output device's drive transistors/protection diodes, possibly damaging the device's output stage and/or the desk's input stage. This damage can be cumulative, eventually causing failure only after a large (or small) number of 'plugins', and then just fail completely when you least expect it. 4) Phantom power is usually available on microphone preamplifiers inputs, and not line inputs. So line devices should be connected to the correct line inputs, other use a DI or level transformer to connect to the mic input. Transformers and DIs will block phantom power and properly match the levels (and line impedance for longer runs) to achieve the best result. 5) 'Good enough' results can be had by playing with gain staging, phantom blockers and using pads, but care must be taken that the DC path is being blocked and inrush current is limited (eg. Use isolators that are designed for this). Again, gradual/cumulative damage occurs if components are over-stressed. 6) In regards to the question above about adversely affecting the signal by adding these outboard protection devices; you are already compromising the chain by not using the right inputs/connections, and it is unlikely you'll notice the SNR reductions compared to having them built into the unit. 7) If possible, always connect line level outputs of powered gear to the TRS 1/4" jack input on the desk when a channel has both XLR and 1/4" inputs (many desks have this and the phantom is only supposed to be on the XLR). Yes, this requires a proper cable or an adapter, but it's better to be safe than have your gear fail when you least expect it. So, yeah, the Helix could (or maybe should, knowing their user base) have had protected and/or transformer isolated XLR outs. But, I'd never have that expectation for any piece of gear where the XLR is marked as a line output.