jeremyn

Analogue has virtually no latency. Digital systems generally have a bit to allow error correction. From what a friend who has a pair has said, the RS185 has around 20 to 30 milliseconds. Much better than Bluetooth, but might add up with other system latencies to be noticeable on fast rhythm passages.

Or they give you some consumer grade headphones with hyped bass and treble that distort when plugged into the headphone output of the Helix making it sound like dollars hite.

So loud enough with your loud drummer! Were you using this for FOH duties or just as local foldback for yourself with a send to the house PA? I've had mine up pretty loud, but never really pushed its limits as I use it mainly for local foldback or relatively small venues.

I love mine too. Great horizontal dispersion, and a good flat response. It's not super loud like a QSC K series or Yamaha DXR, but in most venues it gets loud enough beyond which you get told to turn down. Not quite loud enough for competing with a loud live drummer or a second guitarist with a 100W Marshall and 4x12" half stack. But, perfect for normal listening music in most small to medium venues.

My Audio Technica ATH-M70X headphones sound great and are 35 Ohms, so the Helix is definitely capable of cleanly driving impedances lower than 250 Ohms. Those M70X cans are rated a 2 Watts, and are deafening with the Helix volume past noon. But, at normal to loud listening volumes they sound great. Then again so do my Bose QC25s, but the Bose have the usual hyped bass and treble of consumer headphones, whereas the M70X are reference level flat.

I wouldn't say the Helix tuner is 'unusable', but it's definitely more jumpy and nowhere near as stable and quick to tune with as my Turbotuner, Boss TU12 or any of my clip on tuners. At least with the update the tuning resolution was greatly improved, even though I have to hit the string a bunch of times before I'm confident that I've brought the string into tune. It's annoying enough that I still use my Turbotuner. The thing about all my other tuners is I can tell when I'm turning the knob at the right speed to end up at the centre after seeing the how fast the tuner is moving towards 'perfect'. The fine tuning part of the Helix display seems a bit random while the coarse section isn't smooth enough, which means the "movement of the tuning indicator" (ie. tuning 'vector') isn't fluid. So it takes more effort to get into the 'tuning zone', and then once I think I'm in tune I still need to manually 'average' the tuning point by hitting the string a bunch of times to make sure it's jumping approximately evenly above and below the central point. This is not an unfixable problem, but if none of the beta testers can reproduce the problem and they all think it works perfectly, then it's going to take much longer to get it sorted out.

My Audio Technica ATH-M70x headphones have a very flat response and sound great with the Helix. They are 35 Ohms, but sound about the same being driven by an inline headphone amp as they do being driven directly from the Helix's headphone out. I've also tried my Bose QC25s and they sound good, but they have the classic bass+treble boost that most people expect from their music listening phones. So the tone needs to be treated with that expectation in mind and the patches made with these don't really translate across to the FRFR monitors without having to make further EQ tweaks.

I know this is an old thread, but I thought I'd answer the original question. Tubescreamers and many other low to medium gain overdrive pedals mix a portion of the clean signal with the distorted/compressed signal at the output. This is what gives them the ability to sparkle and still have clarity while still having that feel of a solid distortion. The attacks come through with clarity to give punch to the crunch so to speak. However, if you send an extremely hot signal into the input of a TS style pedal, you end up with a loud clean plus a much smaller distorted signal mixing. Unfortunately, since the distorted path is compressed and is not proportional to the input signal, a loud input will overpower the distorted portion with too much 'clean' signal. This happens in the purely analog world if you use a strong clean boost pedal before your TS pedal, or even if you have insanely hot pickups. And similarly, you'll get the same effect if you have any blocks that boost the signal that is being sent into the outboard OD pedal. So the solution for the original poster is to make sure that the gain staging is set in the Helix so the TS-808 is fed an appropriate signal level. This might involve sending it a lower signal level that is more akin to what is coming out of the guitar and boosting the return signal into the next stage of the digital chain. It may even be as simple as putting the OD earlier in the signal chain before other blocks have boosted the signal.

For connecting a line signal to a powered speaker, the shielded cable is what you want. For 1/4" unbalanced signal/line leads, the shield and conductor are coaxial, where the shield prevents noise for entering the high impedance internal conductor. For balanced lines (XLR and 1/4" TRS), the shield surrounds a twisted positive and negative differential conductor pair. Balanced connections reject common mode noise because noise that is picked up by both pairs (which is very likely as the signal lines are twisted around each other forming a helix) will be cancelled at the differential input. Line level signals are medium voltage, but very low current, and therefore have an incredibly small power transfer. Line level output drivers are usually very low impedance and the inputs are much higher impedance. Cable capacitance has an effect relative to the impedance of the output driver, but is usually negligible for active line sources over short distances. (Cable capacitance has a big effect on passive magnetic coil pickups like those found in electric guitars - but can otherwise be ignored for active pickups and active line drivers over normal cable lengths.) Speaker cable is designed to transfer power and therefore carry high currents. This type of cable is not coaxial or shielded (it can be, but there's no real improvement for driving speakers to just using a standard parallel pair when the power amp driver is differential). Speaker cables uses a much thicker conductor that is capable of carrying those high currents, and the impedance of the end points is very low (ie. 8 ohms for an 8 ohm speaker) to allow power to be delivered to the speaker. Power amplifier drivers are usually balanced, but may be single ended. Due to the very low impedances, noise is unlikely to couple into the speaker unless it can generate a high enough power to drive the voice coil with enough energy to be audible (which might happen to a small degree if you run a really long length of speaker cable in parallel with a really long length of mains cable that is powering a lot of gear). Cable capacitance has very little to no effect on amp to speaker connections. So if you use a regular twin speaker cable on a signal level interconnect, it is susceptible to induced noise and much more likely to pickup hum/buzz/RF interference. It will not otherwise affect the tone of the connection. If the speaker cable is coaxial/shielded (no benefit for speaker connection, but looks good in some marketing material) then it will act just like a regular line level interconnect, and you're good to go. On the flip side, it is not a good idea to use a signal level cable as a speaker connection for any speaker above a few watts. The internal conductor is usually much too small to carry big currents, and may heat up. This is because thin cable has a higher resistance per unit length, and longer lengths will degrade the amount of power that can be transferred to the speakers. Think of a fuse with a thin core, versus a high current fuse with a large core. Of course, there may be some super stiff leads that are made with thick cable that is designed to carry high power (no benefit for line level signals) that would be fine to use as speaker cables. In general, use the right cable for the job to limit any unpleasant surprises.

Keep in mind that the 'tone shaping' is designed to compensate for the 'room' effects on the frequency response. As an obvious example, the height from the floor directly effects the final bass response of the speaker. Further to that, room reflections, resonances, and absorption can cause differential peaks and nulls in the frequency response of a speaker that may be perfectly flat in anechoic space. In other words, a speaker may be a perfect FRFR in an anechoic chamber, but when set up in room the size of a typical bar have boomy bass, muddy low mids, and uneven treble. Add to that the ambient noise of people talking, and then the mid frequencies no longer sound strong and full either. In concept, a FRFR should at least get you in the ball park of 'flat' and have the facility to be EQed to a final response that is appropriate to the venue. The same speaker should have a relatively even response at various angles relative to a listener seated on the speaker's frontal axis. IMO, the more recent models of DSP aligned speakers that also have some very useful tone shaping ability can be set much 'flatter' than a non-adjustable speaker that may measure perfectly flat in an anechoic chamber, but need further EQ when used in a real acoustically 'live' environment. That said, a good graphic or parametric EQ is very useful to achieve proper room compensation, even if there is some basic EQ built into the powered speaker.

Fantastic little program. Simple and nice UI.

If phantom power is on, plugging in a device can cause a huge current inrush as the coupling capacitors charge. This doesn't happen if the unit is off or the phantom power is off when plugging in. Some rack mount/powered gear with line outputs can cause huge surge currents, but even mics need to charge up their caps and will cause a higher than normal inrush. If the phantom powered input is not properly designed to deal with this, the opamp/IC on the channel input can be damaged. This inrush is much slower and therefore more gentle when phantom power is powered up with the devices already connected. There is also not a big inrush problem when plugging out, although you might hear some loud noise through your speakers if you do this while everything is live. Some circuit designs are designed to limit or are protected against this inrush and are ok to 'abuse' with live plug-in. However, both sides must be considered when making a call whether this will be safe. Things may seem ok, but the protection and circuits could degrade slowly and fail after a while if not designed to take it. I have not seen the Helix schematic and cannot make a call on its ability to withstand live plug-in with phantom active. This also applies to XLR outputs and the desk it's going into.

I'd love to see something in the manual or data sheet from Line 6 that specifies the output sufficiently to determine if phantom power is simply just a mere hiss causing annoyance, or if the Helix may be damaged by it (or possibly cause damage). Who knows why there is hiss, I can think of a dozen circuit possibilities that could explain it that it's not even worth speculating. Lifting the ground may help, but only if there is no alternative low impedance ground path between the two pieces of equipment - if there is, then you're back where you started when it comes to phantom power causing problems. Making a low impedance non-transformer based output safe for phantom power needs heavy protection diodes, enormous voltage headroom or a higher output impedance to limit current inrush. Many of these methods affect the signal to noise/distortion ratio of the output, especially for longer cable runs. We know the Helix XLRs are not transformer coupled, which means it may be susceptible to being plugged into phantom power. Heavy protection diodes moves the risk across to the input side (yes, the side that is supplying the phantom power can be damaged too). When an input that has phantom power enabled is suddenly plugged into a device that is not designed to be powered by phantom power, there is a sudden inrush of current through protection diodes at both ends while the output coupling caps charge up. The moment the devices are plugged together, the 48V is applied and charges up the output stage coupling caps with a high surge current through the protection diodes of the output and input stages. If either the output or input is not current limited, or both ends don't have high rated protection diodes (ie. not the ones built into the input/output op-amps), then something is likely to be degraded/damaged and at some point fail. I have no idea what the Helix output circuit looks like, so I can't make a call. But, if it is protected with huge protection diode clamps, there is still a risk of damaging the input stage of the mixer/desk. This is why manuals always warn that phantom power must only be switched on after everything has been plugged in. Plugging in devices before enabling phantom power limits the charge current to the phantom power bias resistors (which is less than 10 milliamps). Even ignoring the inrush surge currents, if a device has a capacitively coupled output, the caps need to be rated sufficiently (and polarised appropriately if they are polarised capacitors) to avoid being damaged by the existence of phantom voltages - even if very little or no current is flowing. Again, with no specs or official statements from Line6, I wouldn't make a call that its outputs won't be degraded, and at some point fail. As PeterHamm says, its always safe to use a DI box. That way you don't have to second guess yourself, and you get all the advantages of a DI box including attenuation, total galvanic isolation with transformer based models, and whatever your box supports. Over the years, I've seen many mixer inputs die and random gear stop outputting on one or more channels, almost definitely due to the stresses of using phantom power where it is not supposed to be used. Maybe the Helix is ok, but why risk it.

That cable is a stereo splitter. If you plug a mono cable into it you end up with signal coming out the left side and nothing on the left. Definitely don't split your unbuffered guitar externally with a cable and plug it into multiple inputs, otherwise your tone will suffer on all inputs. This happens because the guitar will be loaded by the combined impedance of all the inputs. Also, you can use a Y-splitter block in the Helix to split a signal to avoid signal degradation.

Yeah, I'm really hoping Line6 puts a Freeze effect or equivalent into the Helix at some point. Make sure you vote up the idea on Ideascale if you haven't already. It's a simple effect, but a great basis for ambient patches of pads and drones that sit underneath the guitar. Reverb and delay are great, but they can be like holding down the sustain pedal on a piano without the ability to lift your foot off to 'clear' the sound space at the chord change. The people that made the Plus Pedal that someone posted about after NAMM really understand the musicality of this type of sustain. It's not only about infinite sustain that you can solo over for 10 minutes. It's also about keeping a solid background sound scape alive while you're moving through chordal changes. The ability to 'tell' the pedal when you want that to happen is incredibly important to avoid 'chord clash'.

I don't think the Freeze has a kill dry or 100% wet setting. So you need to set up a block in the same split section as the Freeze to kill the guitar signal going into it. That way the guitar goes down the other split path, and the Freeze keeps the frozen output going into the trem. Unfortunately this is a pain to do. The Superego makes it easier, as does modding the freeze to allow 100% wet. edit: here is a link to a mod that will allow the freeze to do this: http://dogisblue.com/gear/diy-effects-ehx-freeze-with-effects-loop-and-expression-pedal/

No, but if you need extra controllers, you can always use external midi devices. That is a much better way to add assorted control devices without worrying about the inputs on the Helix. It's also a portable solution that will work with just about any other digital effects processor.

This of course all depends on where you play and some styles lend themselves to venues with generally good sound systems and others styles to bad sound systems. A lot of old style churches have a desk in a locked cabinet that may have phantom power permanently enabled. This is not heavy rock, but unless you bring your own PA it's gonna be painful if you don't have a bag of tricks with some problem solvers like a DI, XLR/jack converters, spare leads and a long extension cord.

I suppose when the grounding is bad enough to exceed the output common mode of the Helix, there's a good chance it is dangerous too as it's probably a misconnect on another phase. Happened to me (not with the Helix though) at a particular venue that has since been completely rewired for safety. A transformer isolated DI box saved the day, but no one was kissing mics that day I tell you! Level isn't really too much of an issue because you can turn the Helix output way down to compensate for those times when all you have is a blind 'choir' mic receptacle with no way to adjust gain.

For the Freeze effect, vote it up on Ideascale if you haven't already: https://line6.ideascale.com/a/dtd/Freeze/739609-23508

This is generally a good idea if you're regularly plugging into 'unknown quantity' PAs/desks. Get a good passive DI with a ground lift switch and optional pad to lower the signal should it ever be necessary. The transformers in these boxes will deal with enormous ground differentials far greater than the ground lift on an active output. My favourites are the boxes from Radial. The PRODI2 would do exactly what you want. I have a couple of the StageBug SB-2 units that I use on mono pedal boards.

Unless you have the Eon on a pole, make sure you roll off the lows by putting it in floor monitor mode or manually tweaking the EQ (via Bluetooth). Otherwise you'll end up lowering the bass in the Helix amp sims to compensate, which will cause your tone to sound thin and brittle in your headphones/IEMs.

I've been using a set of JBL Eon 10 G2s for close to 12 years for monitoring (along with the occasional field trip for off site FOH duty). After over a decade, they're a bit bent and battered, but still sound as good as the day they were new. Not the loudest 10" powered speakers, but more than loud enough to deafen anyone standing next to them if they're turned up. I recently had the chance to listen to the EON610s in a similar environment and they sound even better again. The off axis coverage has significantly improved compared to the older model, and they just sound more 'accurate' in the way that is expected of the current generation of DSP aligned powered speakers. From their specs, they aren't quite as loud as the DXR10s or K10s, but for anyone that's not maxing out their speakers as FOH mains these new EONs bring a quality sound. I was also impressed that the price is so reasonable.

Oops, I re-read my first line and you're right. I meant to say the cross-over frequency is lower when the bass driver is bigger. As the cone area grows, the directionality of a given frequency tightens. So larger speakers need to cross over at lower frequencies to compensate for this effect to maintain relatively uniform dispersion over the speakers nominal bandwidth. Without a cross over you end up with the problem that (non-FRFR) single driver 12" guitar speakers have where the trebles 'beam' and people like to stand off to the side to compensate. That guitar speaker may even have a perfectly flat frequency response over it's speaking bandwidth (50Hz to 5kHz) on-axis, while the dispersion is very non-uniform and muddy off to the side. If the flat response is off axis, then the on-axis response will have enhanced treble and presence. 15" speakers are worse for this. That's why FRFRs always have multiple speaking elements once the size gets beyond a few inches.

Something worth considering is that in a two-way speaker system, the crossover is usually a bit lower in frequency when the bass driver is smaller larger (edited). The crossover frequency for these speakers is around 2kHz, so the big speaker is handling the mids and lower. The trebles, presence and warmth are being handled by the high frequency driver / horn. So you won't hear a difference in warmth between the DXR8/10/12/15. It may sound a bit different in the way it drives the mids around the crossover frequency, and of course how deep and strong the bass can get if you put full range music through it. These systems are designed to be linear and I've always rolled off the lower frequencies for live guitar and vocals - considering that even the DXR8 can get all the way down to 57Hz (at -10dB), I don't see how there could be much difference between the difference size FRFRs if you're not putting full range instruments into them. Also, smaller sized units usually have a wider dispersion, but the DXRs are all specced the same, so you don't have to worry about that. In this case, portability is a worthy consideration when choosing which one to get. (edit: fixed incorrect statement above)