DunedinDragon

The general approach by most people that use a traditional amp as an on stage playback system but send a direct signal to the mixing board is to branch off from the preset right after the amp model but before the cabinet block and output that signal to 1/4" out to the amp because the amp is providing the cabinet. The rest of the preset continues on with the cabinet or IR block and goes to the XLR output to the mixing board. I'm not sure if that would work with the keyboard amp because it's a little bit different animal. I can understand why the keyboard amp is not responding like a normal FRFR would respond because it's not a PA speaker, but I'm not completely sure it's going to respond like a traditional amp either since it's kind of a hybrid.

And therein lies the problem. HXEdit doesn't need to know anything about all the possible configurations, nor does it need to keep track of DSP usage for adding blocks or managing splits and merges, and so forth and so on. That's because it's written in a function/response format (similar to HTML) and relies entirely upon the Helix unit for the enforcement of logic. If that logic changes, HXEdit isn't affected because it doesn't know or keep track of anything. It simply reflects the results of the action to the user. The only real logic HXEdit is responsible for are interactions with storage such as exports or backups, etc. Of course it's possible to do what you're talking about, but that means the logic rules in the Helix unit have to be duplicated within HXEdit, which is a time bomb waiting to happen should they get out of sync, and architecturally speaking a very bad design. Fortunately HXEdit does have the ability to identify when the logic of a preset doesn't match up with the rules of the Helix. It's called a "bad preset" error...and those are lots of fun to try and track down and correct. Ask anyone whose had one.

This has been an ongoing, and sometimes passionate, area of discussion sometimes. I can only relate what I've discovered in my search for consistency over the last 4 years. The Helix manual recommends setting your Helix master volume at maximum, although most of us don't want to give up the convenience of having a way of globally having some control over our volume across all patches and snapshots, particularly as it pertains to our stage volume. However, the Helix volume setting does have an effect on the behaviors of your amp models as far as where you end up setting them to get an adequate level for your patches. I originally settled on setting my Helix volume at 11 o'clock, but that ended up forcing me to use unusually high channel volumes on many of the amp models which is really the ideal way of getting consistent volume levels without affecting tone. Exacerbating the problem is that I send my 1/4" outputs at line level to my stage monitor (Yamaha DXR12), and I send my XLR outputs to the mixing board at mic level. In order to not affect my mixer signal if I adjust my stage level, I disconnect my XLR outputs from the Helix master volume, which then sends that signal to the board as if I had the Helix volume level set on full. That's not really a problem as it can be gain staged at the mixer, but it bothered me that my gain staging at the mixing board had to be so severe due to the way I had my amp model channel volumes set. So in order to bring all of this back in line and have more consistency I changed my default Helix volume setting for dialing in my presets to the 3 o'clock position which then allowed me to bring my amp model channel volumes back in line with what I'd been used to on most physical amps I've ever played through, and brought my XLR output more in line with the levels for other channels on the mixing board. The added benefit which I hadn't counted on was in the effect this had on many of my post amp effects such as compressors, reverbs, and delays which began to behave in a much more manageable and predictable way due to the lowered input signal from the amp. I'm not saying this is right for anyone else, but I can say this has made me MUCH more comfortable with managing both my stage volume signal as well as my mixer signal independently and getting overall better results across all my patches and snapshots. This may be something you might want to examine for yourself though.

Cabinets, by design, project sound in a wide radius both vertically and horizontally and the sound changes as you move your position relative to that speaker (i.e. stand off to the side). Powered PA speakers, by design in a vertical position, have a wide sound radius and a limited vertical radius. This accomplishes two things. It maintains the tonal characteristics of the sound across that wide horizontal spectrum, and it conserves what would otherwise be wasted sound energy that gets lost in the ceiling and floor so that the sound energy is used more efficiently and projects further. This is why PA speakers can sound very harsh close up, but that harshness blends away pretty rapidly as you put space between yourself and the speaker.

Just to clarify. Assuming Windows 10, when you have the laptop hooked up to the Helix USB and the Helix is turned on, when you hover for a moment over the speaker icon on the bottom right side of the laptop screen, what's the name of the driver that pops up?

Generally you don't gain a lot with combining IR's like you do with stock cabinets. Because of the way stock cabinets are designed, the most common use of additional cabinets is to be able to mix different mic's and placements as a given cabinet can only have one mic. That's not the case with IRs. The number and mix of mic's can be quite large within a single IR depending on how it was captured. In many cases it's not unusual for IRs to be captured with different combinations of speakers as well, so combining them in Helix is somewhat redundant and not very efficient.

Although there's some truth to the difference in PA's in theory, in practice they're generally pretty equivalent nowadays as long as they're of a more modern design. FRFR isn't a "standard" so to speak and no speaker manufacturer other than Headrush actually uses the nomenclature FRFR. So there isn't really any "true" FRFR as you referred to it. And you actually wouldn't want a completely flat response on any speaker because that would sound horrible to the human ear. Each manufacturer voices their speakers slightly different based on their target market. Although I normally use the Yamaha DXR series, I also own and have used the EV ZLX series as well as the QSC CP series with very similar results. Primarily because those manufactures all address live music performance as their primary target markets and voice their speakers accordingly. FRFR is just a term that generally refers to a certain type of speaker architecture. By and large that architecture (at least at club level applications) consists of powered speakers using a bi-amp design with DSP processing for allocation of the frequencies across the high and low end drivers, sometimes with tone contouring adjustments for various placements and special applications. Most relatively modern PA speakers within the last 10 years or so would fall into this category. The ones with significant problems are typically older/cheaper style PA's with powered boards and analog crossover systems. However at large scale concert events the architecture is different but still consistent with the small scale design goals of what's generally referred to as FRFR. Given that you're not playing into one of those cheaper/older type of systems, these modern type of speakers will give you a pretty consistent playback not only tonally but behavior-wise since these speakers are architected quite differently from a cabinet and for an entirely different purpose. Look at it this way, how many PA's have you ever seen using full range stage cabinets such as the Behringer as their FOH speaker of choice? I'm not saying whether or not your choice of the keyboard cabinet is bad or not. That's up to you. But it doesn't accomplish the same thing as what's referred to as a FRFR speaker as far as projection and consistency of tone across a wide sound radius. This is one of the reasons why even keyboard players are opting away from cabinets and toward these type of speakers in order to get a more consistent feel for what the FOH sound will be. So it's not "hype", as you put it. There are real engineering advantages designed into these speakers that have caused them to become the default tool of choice in live PA music performances from smaller club level speakers to larger events using line arrays. There's a real reason for that.

The only really important question here is, how are you going to dial in your patches to ensure you get the sound you want through the FOH. Clearly as you saw there is a difference between the sound of the EV and the KXD using the same patch. So your audience is going to hear the EV sound regardless of what you use on stage if you go XLR out to the FOH. So dialing in your patches on the KXD is not going to be an accurate representation of what your audience will be hearing live.

Frequency response is maybe the smallest part of the picture. The term "FRFR" is a somewhat invented term and typically refers to speakers that have cover the full range of frequencies that human hearing can detect AND have a generally flat and contoured response across the entire range of those frequencies. It's that second part that's the really distinguishing factor. This is accomplished by having speakers that can cover that range, and an amplification system that's advanced enough to intelligently allocate the frequencies so that there's a smooth and even level of the frequencies allocated to the two (high and low frequency) speakers. To do this the speaker is generally going to be a powered speaker design most commonly using a bi-amp design and digital signal processing (DSP) chips to manage the frequency response between the speakers. Because of this design the speakers won't typically have any tone controls because that's already pre-determined in the design. On the higher end units they provide contour settings which allow you to adjust the frequency response to different types of applications such as floor monitor, pole mounted, live music, recorded music, speech, etc.

I'm sure it would be, although I don't know if the Sharc is used for A/D and D/A or if they have a dedicated converter since I would think the Sharc is primarily focused on the mathematical manipulations on the digital stream. It's all guesswork from the outside.

I doubt seriously that the USB-B is even capable of supporting a wireless connection. The USB-B interface is, by design and specification, for use with a slave device such as a printer rather than a peer device.

Actually a pitch shifter effect would be instantaneous with a regular stomp as long it's in the same preset. The value of shapshots is being able to change a lot of things instantaneously rather than having to switch to a different preset which would have a slight delay.

I also expect that you can't really compare general computer chip architecture and speeds to something like a SHARC since they operate so differently and are applied so differently. The SHARC architecture provides for simultaneous parallel processing of data and instructions through an internal bus in order to achieve real-time processing which is very different from that of a general computer that uses an external bus to access memory and operate in a more sequential manor.

I think what BlueD is referring to are all of your settings both on the blocks in your patch as well as on the Helix in general. Sometimes it's just easier to post the patch itself on some shared site and give a reference to it. Also any elements you might have changed from the defaults in global values and settings. The other thing that's REALLY important are your cab or IR settings which include mic/mic mixes and mic placements. Those have a MAJOR impact on the sound. Although you mentioned you tried all the speaker sims there are a LOT of values on the Helix cabinets which have a significant impact beyond the defaults (which aren't always very good). Also whether you tried using dual cabs in order to mix different mics which is probably one of the more efficient ways of getting the sound you want. The same would apply to IRs in what type of mic mixes and placements you used.

That's more of a question that deals with your situation. If you're going to a mixer run by a soundman that's provided to you by the event or club, he'll most likely want you to send via XLR as that's most common since that's what microphones and most, if not all, other inputs are going to be. If it's your local mixer run by you or someone in your band it's up to you as long as the mixer isn't too far away. For myself, if I go to a board I go via XLR simply because I reserve the 1/4" for my stage monitor.

Okay...seriously speaking. Many of us paid $1500 and you paid $1000 to get all of the benefits we waited for (and got) over the last 4 years including features like snapshots, an HX Edit that allowed you to edit your patches rather than just save and restore them, and multitudes of various amps and effects. Updates aren't something we're ENTITLED to, it's something they graciously provide us with for free. So how about instead of whining you enjoy all the features you've got, and never had to wait for, and give them whatever amount of time they need to provide a rock solid, non-entitled, expansion of features rather than a rushed out, bug-laden atrocity just to appease your curiosity? By the way, summer doesn't start until June 21 when the Sun reaches it's northernmost point.

Well, since that's not the advised way to do an update, that's probably where your problem lies. You need to first download and run the install on the most recent version of HX Edit. That will install all the appropriate device drivers and Line 6 updater versions. You then exit the HX Edit and run the Line 6 Updater to update your firmware. In all likelihood you have a mismatch between the FW and the HX Edit version.

Hit your Mode button.

It's kind of hard to draw any conclusions about the Gemini speakers as the company doesn't provide much technical information. The fact that it doesn't provide anything as far as contouring selections as well as the fact that in lieu of that provides an equalizer for adjusting tone response suggests it's not of the more modern DSP designs used by most Helix users. I'm personally leary of them simply due to the fact that they're primary market is, and historically has been, for DJ's and not live musicians. That doesn't necessarily mean you couldn't get a reasonable representation of the Helix's capabilities through them. Especially if you were to compare it to going through the Twin with no amp or cabinets in your signal chain. It would suggest it's not likely to be the flat response type speakers you mostly hear about on forums such as this for modeling use. That being said I tend to agree with rd2rk that maybe modeling may not be your cup of tea. What you have to recognize is that the Helix (and other modelers) paradigm is that of a guitar playing through a live PA with a mic'd cabinet. That's a different sound than what you may be used to if you've spent most of your time playing through a traditional amp on stage. It IS the sound you would get from your traditional on stage amp once it's mic'd and sent through a PA. But that may not be the sound you're used to. If you do mic your traditional amp and send it through the PA, it is the sound your audience is used to. But some musicians have a hard time getting past that situation and don't feel comfortable with that sound. For them, the best option is to use the Helix in a 4CM configuration using an amp with an effects loop using a signal chain that can send the amp and cab signal to the FOH and only the Helix amp model through the effects loop of the amp and it's cabinet. Quite frankly I'm a bit surprised given your statement of how much you had investigated the Helix that this would be something you hadn't uncovered in your research.

The question I have is about the PA you used for the Helix. Not all PA's are made equal, and the wattage is really irrelevant. What is relevant is the type of speakers being used on the PA as well as the way you set up your preset. You mentioned it's a powered PA assuming that means you have powered speakers. But there are older generation powered speakers that still rely on simple crossovers and have tone controls, and more modern speakers with DSP driven crossovers and bi-amp designs that have more flat response such as the EV ZLX series, or the Yamaha DXR or DBR series, or the QSC K.2 or CP series and so on. Those will provide the most accurate representation of the elements in your Helix patches because they won't have the typical drop out in the frequency range where the crossover occurs. If you have speakers consistent with those I've mentioned, you really don't need to go through a complete PA to get a feel for the sound from the Helix, you can simply go from the 1/4" L/Mono output to one of the powered speakers. One important aspect when using these type of speakers is to dial in the proper contour adjustments for how they're being used such as Monitor if they're set on the floor to mitigate bass coupling, or default if positioned on a pole and make sure you provide adequate space (around 5 or 6 feet) between you and the speaker in order to not be overwhelmed by the speaker horn. As mentioned before, it's common to use both high and low cuts to trim out excessive highs and lows on flat response speakers, but that's also highly dependent on the cabinet models, mic models, and mic placements used within your preset. This is nothing new and relates very much to the way these things affect traditional setups. You can achieve a mic mix by selecting a dual cab block and specifying the same speaker on both but with different mics and placements on each. I usually have some form of high and low cuts on my presets, but they tend to be much more subtle due to how I use mic mixes and mic placements on my cabinet models. I normally use a combination of mics such as a MD421 dynamic mic and a R121 ribbon mic typically placed at a distance of around 6 to 8 inches. This mitigates the harsher highs so I rarely have a high cut less than 8 kHz, and more commonly up in the 8.5 to 10 kHz area. My low cuts generally range in the 80 to 125 Hz area. I apply these cuts using a Parametric EQ toward the end of my signal chain. Jason Sadites demonstrates these techniques in his great YouTube series on getting great tone. One of the things that likely affected your sound when using your Fender Twin (aside from not being able to bypass the preamp and tone stack of the amp through an effect loop input) is the fact you were using a physical guitar cabinet. There are many people that use traditional amps rather the powered flat response speakers normally in some kind of 4CM arrangement, but even in those cases you wouldn't use the Helix cabinets and mics because the Helix cabinets would add additional coloration to the sound of the physical cabinet.

You do realize there won't be another manual update until 3.0, right? Line 6 has long established that manual revision publications will only occur with major firmware releases. So I'm not sure what you hope to accomplish with this effort except to provide those that come across this thread with some additional information.

The question for larger live performance is really more about how does it come through on the PA, since the larger the audience the more likely that will be their only reference rather than your stage sound. As rvroberts points out you can really only guage the mix once you get it to performance volumes and stand in front of the PA. I've played in the same band for around 9 years with myself on lead and another guitar player on rhythm. In my experience it's far less about EQ differences than it is differences in what and how each guitarist is playing. It only really becomes an issue when both guitars are playing the same chords in the same way. That's when things begin to sound like mush, and then different EQ's don't really help. Typically in our group the other guitar player tends to play standard bar chord rhythms while I'll offset that with maybe open cords or different variations higher or lower on the neck, simplified accent rhythms, or different chord inversions or palm muting or hybrid picking and so forth.

Snapshots by definition aren't global. They are always part of a preset, so whatever you define as part of a preset will be available for use by the snapshots on that preset. You could potentially define a "template" preset then just copy that to whatever new preset you might need to create. You could have that preset defined with default snapshots.

You might have benefitted from reading some of the initial posts because nothing happens with suggestions posted here. You need to post them in IdeaScale so they can get voted on to determine if it's worth doing.

From what I've heard Line 6 has no interest in developing an HX Edit program for Android or iOS devices. That could always change I suppose if there were enough interest, but as of now they've shown no interest in doing so.