jeremyn

The ISOXL will work with any Helix output level. Use mic if you want a lower level into the desk.

The Palmer PAN 01 should work. I have a bunch of Radial Stagebug SB-2 boxes that I use for generic interfacing. Any passive DI (ie. transformer based) from a reputable company will be fine for live use. No extra noise with phantom power on or off, and protects all equipment (including itself) from being damaged if accidentally plugging into a live phantom powered input. They also soften the 'bang' if you plug into the system while phantom power is on and the channel is not muted. It also depends on what you're trying to connect. If you want to go from a high level line output to a super sensitive low level mic input, you'll probably want a box with a built in pad, or a step down transformer. If you just want to isolate the two ends without changing the signal level, use something like the Whirlwind ISOXL, which also lifts the ground. However, if you're in a studio, I'd avoid going through a desk preamp at all, and just go from the Helix XLR outputs into the balanced line (not mic) input on the desk (most desks seem to have 1/4" balanced line inputs - which almost never have phantom power). This may only need an XLR to 1/4" balanced cable.

Also, if you really wanted to, you could load a symmetric filter kernel as an impulse response to achieve a zero phase filter with latency equal to half the filter length (approx 10ms for the 1024 point IR, and 21ms for 2048 points). But, it would be a fixed filter and you wouldn't be able to change it on the fly.

The most significant drawback to zero/constant phase (symmetrical) filters when it comes to live audio is that they introduce additional latency into the signal chain, proportional to the length of the filter. Longer filters with detailed low end frequency / full bandwidth response introduce 5 to 20ms of delay - which is huge, especially when you add it to all the other non-avoidable latencies in a digital signal chain. Super short (low latency) zero phase FIRs have low frequency bands that are too wide for a general purpose audio EQ. So basically you have to live with the extra delay in the signal if you want good low frequency control. Also, symmetrical filters introduce a 'ping' or pre-delay to the signal that can sound a bit unnatural - but I suppose this might be a benefit for someone trying to go for something 'different'. However, FIRs don't have to be zero phase / symmetrical, they can implement any type of filter response with linear or non-linear properties (ie. they can emulate a completely analog filter with close to identical properties including near zero latency). A linear phase (ie. minimal phase distortion) filter isn't exclusive to the digital domain, it's just much easier (less effort and a lot less componentry) to make high order filters with DSP (and be able to change them on the fly). A good analog Bessel filter can be made with a constant group delay and minimal phase distortion if that's what is desired - and it'll have extra latency, just like the FIR - but, will take a lot more space, parts, and still require tuning during production to compensate for component tolerances. Likewise, just as an FIR can be used for zero phase symmetrical filters, they can also be used to implement typical analog filters including minimum phase, or any of the standard models where frequency response is more important than phase response or group delay. A skilled DSP engineer has enormous control over filter design and implementation. The hardest part is selecting the appropriate characteristics, which is a bit of an art when pertaining to the human perception of 'tone'. What may seem ideal in a text book sense, may not have a particularly desirable sound over and above a more common less ideal approach. (Edit) Obviously there are specific uses for zero phase filters outside of live audio general purpose EQ, such as the decimation and reconstruction filters that are used for sigma delta A-D/D-A converters and sample rate conversion. And where latency is not an issue, they allow very detailed (and easy) frequency control using FFTs. Also, an FIR can be used to implement very complex non zero-phase filters such as the impulse response block in the Helix.

Nice. Very effective use of rhythm/wah to get it sounding like a punctuated didge. I use an EHX Freeze into a distortion pedal then into a CryBaby to get the didgeridoo drone going.The wah gives it that 'vocal' character and brings life to the drone. I'd love to see a Freeze effect added to the Helix. I've thought about using the Freeze in the loop, but that's an extra pedal to bring along, when I may as well just bring my full board.

This is true, but the pitch instability of the string is not randomly jumping all over the place. You can see clearly with a strobe tuner the pitch 'wobble', and that wobble is easily seen as averaging around a central pitch. The average changes as you turn the tuning machine, but the string still varies around the central pitch. This is normal, and is the physics of the string. However, the Helix indicator jumps around erratically and it takes longer to determine the average point. It still is usable, and allows the guitar to be tuned way more accurately than the pre-V2 tuner, but it's a bit nasty to look at. Think of the string pitch invariance as a violinist's vibrato. The vibrato will sound out of tune if not centred around the correct pitch. A strobe tuner will show the pulsing vibrato as a nice low frequency movement around the centre pitch. The Helix tuner shows random points of that movement. Reminds me of my first attempt at making a string vs sine pitch comparison back in the '90s for my first semester DSP course. I fixed the issue before submitting. This is not rocket science. Is it usable? Definitely. Thanks Line6 for fixing the tuner so I can use it to tune my guitar accurately. Can it be made more aesthetically pleasing (less jumpy)? Yes to that too.

Physics doesn't dictate that tuning precision and indicator are mutually exclusive. The exclusion is between tuning precision and tuning speed. There is a difference between the rhythmic pulsations around the desired frequency that you get with a (real) strobe tuner, and the erratic random jumpiness you get from aliased sampling of a high bandwidth capture of a 'particularly wobbly' string being displayed on a slower update display (without using an anti aliasing filter). The 'jumpiness' on the Helix fine tuner can definitely be reduced in firmware, and is not a fundamental limit of physics. That said, the previous suggestions of using 12th fret harmonics reduces the 'wobble', and reduces the aliasing/jumpiness of the fine tuner display.

With a strobe tuner you can see exactly how much variation there is in the string's frequency while you're tuning. The human eye is great at averaging out any vibrato once the pitch starts to close in on the centre frequency of the desired note. The pattern can be clearly seen wobbling back and forth, and the viewer can easily determine when the net travel of the pattern over time has stopped or slowed sufficiently. This works well, because the ear naturally averages out all that vibrato wobble and identifies the note as the centre average of all the vibrato. The Peterson and Turbotuner strobe tuners are great examples of how to build a tuner that lets the player decide how accurate they want their guitar to be in tune. One guy might spend 10 seconds per string trying get things as close as it is possible for a vibrating string to be. Another guy, might be satisfied spending 1 second per string getting things 'good enough for rock and roll'. Let's talk about accuracy. A tuner should allow a guitar to be tuned 'accurately' within the tolerance desired by the player. The updated v2.0 granularity of the Helix tuner display does help get the string much more accurately tuned than in previous versions. But, the display is way more 'jumpy' than the visual strobe on my strobe tuners for the same guitars. It takes me longer to tune to the same accuracy with the Helix, as the display seems to show an instantaneous sample of whatever frequency exists at that instant time. But, since the update rate is slower than the vibrato movement, the string frequency appears to be aliasing against the display update, causing it to 'jump' around erratically. I assume that a little more low pass filtering on the second derivative of the frequency response would help bring the 'jumpiness' under control. Better yet, just put a little strobe display (with a decent update rate) where those high resolution tuning bars are. But, I'm definitely happy to say that the 2.0 update has allowed me to tune my guitar far more accurately than in previous versions.

The answer isn't necessarily 'no', but the way the distribution works has changed since Yamaha became involved with Line 6. It may be the case, or maybe they care more about the success of the brand than just keeping a local private distributor happy. However, with a product like the Helix, there are local standards/regulatory bodies that required submissions that cost money, and that may cause a problem with just offering local support for all international products (depending on who has paid for that, and how the distribution chain works). It can't hurt to call the local Yamaha/Line6 service centre and find out if they'll repair your product or not. And if so, would they charge you directly, or will it be covered under some kind of warranty (or at least as a 'good will' repair).

It's covered if you ship it back to the original US dealer for repair. Sadly that's pretty expensive for something the size of the Helix. For simple repairs it's probably cheaper to pay to have it fixed locally.

The length of the IR is all about the bandwidth/detail of filter response. The longer the IR, the higher the resolution of detail that can be captured and 'played back'. For real world cabinets, the extra detail of longer IRs is mostly noticeable in the lowest frequencies (i.e. < 150Hz), or for speaker/cabinets that have very noticeable resonant peaks. Many real world speaker/cabinets already have a fairly smooth characteristic response in those areas, and increasing the resolution (ie. length) of the IR won't do much. Same for real cabinets that have been well designed for a smooth response across their output spectrum (ie. no deep/sharp nulls anywhere in their frequency response). Those cabs won't sound much different (if at all) with a longer IR. A rig where the combined (guitar/amp/speaker/cab/mic/EQ) doesn't have much of any low frequency response, probably isn't going to benefit from a longer response, whereas a rig with a lot of low end detail will. Also, a super clean trebly guitar/amp/speaker/cab with a lot of chimey resonance, may benefit somewhat from a longer IR as some of that 'chime' is due to narrow resonant peaks in the speaker. And of course, an IR capture of a far mic'd cab (which includes some 'room' response) will benefit simply from the longer tail of the longer IR (although even 2048 points isn't anywhere near enough to properly capture most room reverberation).

Knowing the output impedance isn't likely to help much. You'll still need to know the maximum voltage swing of the output to calibrate your meter for the full scale output. The Helix output impedance is probably around 100 Ohms as it seems to be designed to drive into the line input of a mixer or rack, and lower values provide better noise performance. You should design your meter with a high enough input impedance so that the output impedance of the Helix is no longer an issue. Then you can adjust your resistors to translate the maximum output drive level of the Helix to the maximum display on your meter, so the point of digital clipping (which you never want to hear) is sitting near the end of the "red zone". Trying to measure a power transfer value on an output designed for voltage transfer isn't necessarily the best idea.

EHX Freeze for ambient and Turbo Tuner for tuning. And I have my eye on the Mel9. As PeterHamm said, they'll probably never add in a Mel9, but hopefully they'll add some sort of Freeze effect and improve the onboard tuner.

Midi clock is just a firmware change in the device, so hopefully they'll eventually add this feature.

It also helps for when you have an external expression pedal without a toe switch. (Considering the toe switch would need to be connected into an extra input on the helix if you did have one with a switch.) Saves that extra stomp -especially important when you're on stage singing and looking out at the audience while trying to avoid looking down to hit the right button.

The EHX Freeze or SuperEgo are even better at this. The Slow Gear SG1 is just an automatic volume pedal that detected the attack and faded it in for you, like a gate with a super slow turn on and a quick turn off. Not sure what the GT100 did to emulate this effect, but SG1 wasn't that great at e-bow type lines beyond the first note unless you fed it into a long delay/reverb. The EHX Freeze effect is a true infinite sustain that samples and holds notes indefinitely. For ambient guitar the Freeze is my go to pedal as a base to feed into my ambient chain. I'd love to see that modelled in Helix, then I could stop having to use mine in the loop.

There are two elements for making a fuzz pedal have a particular 'iconic/ideal' sound with a magnetic coil pickup. One is the cable capacitance, and the other is the input impedance (which is mainly resistive) of the fuzz pedal. Curly cords with really high cable capacitances cause the resonant frequency to drop significantly and the high frequencies to roll off heavily. On the other hand, a load resistance causes the resonant peak to be less pronounced and also causes its own high frequency roll off. Since most people use a normal length of standard cable, the resistive load of the pedal itself (around 10k Ohm for an old school Fuzz face) is the element with the overwhelmingly most significant effect on the tone. When a signal is buffered between the guitar and the fuzz pedal, the heavy loading effects of the pedal are taken out of the equation, and the only thing left is the cable capacitance - which, even if you're using a super long curly cord between the guitar and the buffer, isn't going to have a big enough effect on the tone to bring that 'heavily loaded single coil' magic. That being said, if you put an EQ in the chain before the fuzz pedal (or equivalent Helix blocks), you can get extremely close to the desired frequency response that would otherwise come naturally from heavily loading the pickup in the guitar. And IMO is a better option since it is both more predictable, more easily adjusted, and can be enabled/disabled along with the fuzz or wah effect. The G10 is most likely acting as a simple buffer. In the complete void of technical information from Line 6 about the G10, I suspect it would have a typical 1Meg Ohm input resistance with maybe at most a few hundred pF of additional capacitance to simulate some small length of cable. I doubt it has a fixed simulated 'cable EQ' like the ones that can be dialled in on the other Line 6 wireless units, as doing that would make the G10 less usable for acoustic guitars (whereas any extra input capacitance would only affect passive pickups in the same way a short length of cable does). This is why I always add a 560pF capacitor to my short wireless interconnect cables.

It makes no difference as the G10 is effectively buffering the guitar and feeding an active signal into the Helix. The input impedance (and cable capacitance) only makes a difference with passive pickups. So it's safe to just leave the Helix at its default input impedance setting.

It sounds like a compressor is being 'kicked' by a huge input spike in the amp model when it is switched. The attack time will affect how small a 'glitch' will cause the compressor to compress, so if a smaller number here has a more noticeable effect, then the 'glitch' is very tiny. And obviously the release time will affect how long it takes before the compressor fades back to full gain. This is something worth trying to replicate, as it may be an issue with some amp models or effect blocks that glitch after switching. If the Helix mutes the output for a short time after switching, then you don't necessarily hear the glitches. But, a downstream block like a delay, reverb or a compressor that moves the glitch forward in time will end up with an audible affect beyond the output mute time. Try a delay set to 1 second instead of a compressor in the same location and see if you can hear a click or pop. That said, it is possible that it is something that the Line 6 engineers have hidden with a simple muting technique that has been applied to not only the output, but also to some subset of input blocks like delays and reverbs. It is also possible that some compressor models have a small bug in the signal or side chain where they start with a DC offset that hasn't been set to the quiescent DC level when initialising the model.

Somebody should make an inverse IR of various popular "FRFR" speakers. Then you just put that IR in the chain after your cab block to flatten the hyped highs and lows.

Bass heavy comes from having the FRFR near the floor or a wall. If the speakers are on a pole the bass should be fairly flat. Treble comes from the fact that the FRFR has much wider angular dispersion of treble. So if you're used to standing next to your guitar cab 45 degrees off axis (where the treble is significantly rolled off), then the FRFR will sound too bright. If you normally stand with your ear right in the on-axis 'beam', you'll notice that the FRFR maintains that tone relatively evenly across its whole projection zone. Also, the Helix isn't outputting the same sound as a speaker, but of a speaker + mic. So what you hear from the FRFR is a point recording of a mic next to the speaker. Plus a tiny bit extra recording environment response if the IR / cab wasn't made in an anechoic chamber. The FRFR can't sound 3 dimensionally like a guitar speaker because it has a very different (ie more even) dispersion pattern that is producing the tone of one or more fixed point source (mic'd) locations. Multi mic cab blocks or multiple cab outputs can improve this.

Then there must be something wrong with my Helix/guitar combo. Or maybe I'm using it wrong. If you can differentiate the -1c and -2c BFTS offsets with the Helix tuner amidst the +/-3c error that I seem to be getting, then there must be something wrong with what I'm doing. With the Helix tuner, my Tom Anderson Drop Top doesn't seem to bring that BFTS chording magic that I'm so used to hearing. At the risk of being labelled a 'measurebater', I also measured the error to see what was happening, and found the same kind of wide 'in tune' range of around +/- 3c as the two posters that provided detailed evidence. After I've tuned with the Helix, I have to fine tune with my Turbo or iStrobostomp app and bring it much more into tune. Yes, the guitar is inherently never perfectly 'in tune', but there's a threshold at which the ear says, 'this is adequate'. I'm guessing that threshold is different for different people.

Clearly you've never played with a bunch of stuffy orchestral musicians before. Probably doesn't matter much to the audience though.

"Good enough for rock and roll" vs "orchestral dissonance" really depends on what and how you're playing. It also depends who's listening and making the call. I need to tune to the attack, and press slightly harder on legato runs to compensate, otherwise I get flack from the violinist. If the guitar is slightly out during a rhythm section, no one seems to notice or care. Dissonance is most noticeable when playing legato unison and some harmonies. BFTS and a decent tuner makes everyone happy. That being said, the Helix tuner is definitely "good enough for rock and roll". But it'd be nice to take it to that next level. Although, for me it's mostly academic, as I don't expect to see Line6 including BFTS offsets, or adding the ability for a user to program in their own. Compensated nuts and BFTS help with the chording/fingering issue. Individual errors also add up, and generally aren't too bad for most note combinations. But, every now and then, a +2c string adds up with a +3c fret, a +2c fingering error, and a -2c string, a -2c fret and 0c fingering error, for a total of 11c. Having the tuner start with a decent level of accuracy takes out the initial error. Since tuning and intonation are also player dependent, you need to tune (and have the guitar intonated) based on how you play. If you strum hard (and want to remain in tune with other instruments), then it's a good idea to tune to the attack. If you grip hard and/or have deep frets, make sure the guitar is intonated with your grip taken into consideration. BFTS, a careful setup, and a decent tuner get me to the point where I can play an 'A chord' without cringing.

The problem with Helix as it stands is that the impedance option is global. So setting it to Auto is the only way to get it to follow the current patch. So if you want 1M, you just make sure the first pedal isn't a fuzz or wah. And if you want that perfect fuzz tone, then use a different patch where the fuzz is first. That way the Auto setting makes sure the input impedance is correct for the patch in question. Unfortunately, the patch switching delay/gap in Helix makes it less than ideal to seamlessly go between two patches where one is a sparkly clean and the other a smooth fuzz tone. The way I do that now is to stay within the patch, and just use an EQ ahead of the fuzz to emulate the impedance effect on the pickup, instead of relying on the input impedance option to directly alter the pickup's tone. Then I can just stomp on/off the EQ+fuzz when I want the fuzz tone. Doing it that way also allows the use of the expression pedal to smoothly transition from a clean to an effected fuzz tone, which is something I like to do to build song dynamic.