hefonthefjords

So much wasted breath in this thread. OP, just use an ABY box if you are so precious about using your existing audio interface rather than helix's built in one (which is designed to do exactly what you are doing) for whatever reason. Problem solved. There is no point in having helix do A2D2A through a send block just to A2D your signal AGAIN in the existing interface guys. You are just messing with a signal for the sake of it then. Passive ABY box feeds both boxes the same unmolested phase coherent and time aligned signal, job done.

Metal zone primarily. The Friedman amp secondarily. HX reverbs will nice even though I'm quite happy with the existing ones. Experimenting with all those legacy effects, most of which I've never played with having not owned anything from L6 between the X3 series and Helix.

I find the helix XLR outs are quite hot for most consoles and trim mine back -5dB. As far as mono out from both XLR, just add the stereo width block at the end of the chain and it will allow you to for sure collapse everything to mono before the output.

Phantom power is well known to break most devices which are not designed to make use of or need it. I've seen people fry the headphone output on both expensive macbook pros and mac pros as well as phones by using minijack to XLR cables on digital consoles and forgetting to turn off phantom power before connecting them. Hell, i did it myself to my own phone one time. The only sure way to protect yourself from this lollipopup with helix is to buy/make a phantom power blocker (they are dead simple devices containing only two small capacitors) and leaving them in your helix at all times. Technically the helix has phantom protection built in but the problem is that some devices will supply significantly more than the standard 48 volts (as much as 96 volts!) To support legacy devices or by poor design and it is possible that helix's phantom protection on the outputs is not designed to prevent damage by these higher voltages. In the early helix days many people reported noise and distortion on their XLR outputs and the cause was phantom power on the lines. Apparently their units were ok after a reboot and disabling phantom but it's not worth the risk IMO.

Wait, aren't they just putting a "Legacy" folder within each category of effect? It would be totally dumb as hell to do anything else.

The Amphenol connector is a switching XLR connector. There is a switch inside this connector which detects when it has a cable attached which tells the helix it should output a stereo signal to XLR. When it does not detect an XLR connected to the right connector it sums the signal to mono and only sends it to the left connector.

- global input and output level metering (how the hell can a device of this level of performance and price not have such a basic thing???) - per block input and output level metering (may not be possible because of processing limitations but would be nice) - snap/stomp button config per preset - screen knobs assignable to parameters in an as yet non-existent "no block selected" mode - selectable ability to access EXP3 when no external expression is connected - selectable ability to disable toe-switch on built in EXP. - ability to assign a "parameter reset" to a momentary stomp switch while that parameter is already controlled by something else (for example - to reset a parameter that is controlled by EXP to it's preset level without reloading the preset, or to set the parameter to some user specified level regardless of it's current level) - a marker on all parameters to show the value saved in the preset/snapshot while editing - the ability to disable the joystick scroll functionality in the path view (to prevent annoyingly changing a block by accident while navigating the paths) i'm sure i can come up with more but those are all good ones i'd like to see.

It's possible that the shielding on the longer cable is not adequate enough for the EMF noise in your vicinity. It is often surprising just how much noise the gear around us spews out in the EM spectrum and it can wreak havok on poorly shielded signal cables like this.

Although it is apparent that these devices are not using a standard networking protocol, in standard networking protocol installs the absolute limit for reliable signal integrity in networking is normally 100 metres point to point between devices/switches.

For the sake of contributing something other than speculation, I haven't noticed any audible difference between the firmware versions as yet, however I have yet to test the new firmware at giggable volume levels.

Speaking as an audio engineer, I admire your testing methodology here. I have one question that may complicate your test slightly though... Since helix is modelling analogue gear and by nature of the inherent variability of analogue gear's signal reproduction instance to instance, should you not also test two examples of 2.21's output against themselves and two examples of 2.30's output against themselves to rule out the possibility that the analogue emulation is simply really good and doesn't ever yield the exact same output twice from the same input signal? Food for thought.

The reason is mainly that what the speaker is producing in those ranges outside it's specification is generally considered to be useless noise rather than "tone". We're simply cutting out the parts that are not useful or pleasing so that we may more clearly here the part we like. :) When you look at the response curve of a speaker it will always show you 20hz to 20khz but the speaker itself may only be specified to operate between say 200hz and 6khz. it might be easier said that should be read as "the bit of the spectrum that sounds nice in this speaker is 200 - 6000hz", if that makes sense... Our IRs manipulate the incoming signal across the entire 20-20k range but if the IR is that of a speaker that is only good between 200-6000hz we only want to really keep that bit of the spectrum and maybe a little above it to preserve the presence of the sound. As a live engineer, for me, it is automatic to high pass a guitar at 120hz without even hearing it. Sometimes i'll shift that back to 100 or forward to 250 to taste. If the room is large a low pass may not be needed but in a smaller or particularly reflective room (stone walls, arched roofs) a low pass at 10k is not uncommon on a mic'ed guitar because those high frequencies already have enough energy in them to propagate and reflect violently throughout the room. You might actually be amazed by some of the mental EQ curves that get used in a live situation to make something sound good. I'm mixing tonight, if i remember I'll snap some pics of what rig was getting used and what the curve was like just for fun.

never tried that EQ before. added to my ever growing list of "oooh a new plugin to play with" :)

essentially as yer man above suggests, it's to further tune the output to be more pleasing to the ear. basically at the bare bones of it, an IR block in Helix is just a complicated EQ and compression process that conforms an input signal to a stored reference value. now because your IR is a full range reference of the frequency response curve of a given speaker, and that speaker will be able during a sine sweep to produce low sub tones and sizzly highs as part of it's sound, the algorithm doing the processing is gonna add a bunch of subby tones and a bunch of sizzly high end that you dont always necessarily want to hear so it's a good idea to have the option to roll those off right in the impulse block so that you aren't feeding those frequencies you dont want into your next block or to your output. you should always keep in mind that you are essentially trying to manipulate the sound in the same way a studio engineer might when recording a guitar and cutting the super lows and super highs on the track EQ is very common practice when recording guitars and other instruments in the studio.

XLR mic level out @ 0dB straight to FOH, 1/4" line level @ 0dB out to my poweramp for stage cab. Volume knob controls only 1/4" out for setting stage level. I use a volume block in all presets assigned to EXP2 with EXP2 set to global position to control volume swells and muting during the show. All my snapshots are leveled appropriately for performance relative to each other in the studio and then during soundcheck all I do is set the big volume knob to an appropriate level for how much I need to hear from the onstage cab in order to get good feedback and hear clearly and that's that. Job done. I've only once had to adjust the XLR out master level at a show to lower it slightly because the console in question had really hot preamps. Consoles are designed to accept mic level voltages on their preamps so it's logical to send them mic level signals.

Snapshots can basically do anything. They made Helix for me. Sometimes it requires some time and fiddly programming to get it to do what you want but it will do what you want once you figure out the workflow.

Yup, I'd say that's accurate. They sound soft, even woofy, and lack a lot of detail in the top end. Personally I would go for the ADAM F5 units or even save some more and get F7s instead. They will be far superior in every possible way.

The bigger the drivers, the better is always my opinion. Stay away from KRK though. They sound pretty lollipop even compared to most joe average home hifi systems.

I bought Helix in the middle of last summer (2016) and returned it during the two weeks grace period specifically because of the gaps in preset changes. The very next day the 2.0 update with snapshots came out and fixed the problem I was having, so very shortly after that (maybe about a month later) I bought one again. It did take me several weeks of tweaking and tweaking and playing and tweaking and figuring out routing and playing with things to get the sounds I wanted and required, often times having to ignore what the settings say and just go with what sounds correct even if the numbers don't correlate with what I hear. Helix reacts very differently to any other device in my experience and I moved to it from X3, owning both Live and Pro. It also reacts very differently to my two different guitars, with the same tone sounding WILDLY different depending which guitar I'm using. Now, several months later, having figured out those quirks and had a few bugfix software updates, I wouldn't swap my Helix for anything. I'd still consider buying other stuff to play with alongside it, but I don't think I'll be moving permanently to anything else because the value for money is way too good with Helix.

yep, this is exactly what i was getting at. it's gonna happen. it's probably gonna happen a lot. i feel bad for you guys in advance having to deal with exactly that type of unwarranted complaint. maybe time for L6 to do a dedicated audio interface box? :)

it will generate the exact same signals as helix, however if you are outputting the audio through different hardware the technical performance of that hardware will affect whether it sounds EXACTLY like helix. if your audio interface has the exact same DACs and output amplifiers as helix then it will sound exactly like helix. i suspect that's partly why they intend you to use it with helix. stop me if i'm stating the obvious here.

assuming we are playing at gig volume, if your gate is set correctly it's threshold will be set above the level of feedback but below the level of a picked note therefore your sustained note will drop below the threshold of your gate before you achieve useful feedback unless you move closer to your speaker system. to change this interaction i have different gate thresholds set for each of my snapshots depending on how i intend to use that snapshot and then i also have a bypass for the gate programmed for whenever i might want to override it's interference. for example; i play fairly high gain guitar sounds for rhythm parts in a metal band. during the songs i want the tight snappy stops that a gate provides so my snapshot for rhythm parts has a gate with a fairly high threshold on it above the level at which feedback occurs. now say i'm playing rhythm on this snapshot and we get to the end of a song... i want to hit that big chord or last note at the end and have it ring out into feedback while the singer talks some lollipop and introduces the next song. the gate will step in at some point and have that lovely nice musical feedback of that held chord or note will be choked out quite quickly by the gate leaving an embarassing silence. so i just turn the gate off when i want feedback. :) it's worth noting i pretty much exclusively play at gig volume though and this is obviously aimed at those who do not.

ehh, i'll stick with my switch programmed to turn the noise gate off for $0 thanks.

FWIW, this problem has never returned for me. Don't expect it to either.

It's a ground loop. Put an isolation transformer in one of the lines between the triaxis and the helix and it will fix it.