DunedinDragon

Well, relatively speaking there can be. Of course we're talking about pricing from $200 to $1,000 rather than $14,000 to $80,000. There are roughly three pricing areas when it comes to these type of speakers. The pricing is really based on what market segment they're targeted at and the features and capabilities that appeal to those segments. The Alto is pretty much the low end pricing and is targeted for general use with a big market segment in mobile DJs and the Dance Mix crowd. For that reason they provide a fair amount of wattage for the money but very little in capabilities to contour the sound of the speaker other than boosting the bass frequencies. The next level up would be units like the Electrovoice ZLX speakers, JBL EON speakers and lower end Yamaha DBR speakers. Some decent DSP contouring options to address a variety of different uses (floor monitor, PA on stand, public speaking, etc.), and pretty decent speaker components. The more premium level would be units like the QSC K series or Yamaha DXR series which again have ample volume for most applications, very good DSP contouring options, premium speakers and components, and great care and quality in the cabinet design and materials for both sound and durability. What this really boils down to as you go up in price is greater clarity, articulation, and more accurate sonic response as well as the ability to adapt the speaker to different uses/applications as well as durability which is an important factor for live/mobile use. For many people the mid level units are more than capable of addressing their needs. Even though I choose to use the upper end components because I use mine in many different situations, I do use some middle level units like the EV ZLX for things like side fills. One of the reasons I choose to use Yamaha DXR speakers is that there are sometimes situations I play in where it's not feasible or necessary to send the instruments through the PA, and the PA is reserved for just vocals, harmonica, and some percussion like the kick drum. So I position my DXR behind me on a half height stand facing the audience like a normal backline. In that way the quality of my guitar sound will be more consistent with the quality and reach of the FOH system for my guitar. I've done the same thing with electronic drum kits in that situation using an additional DXR speaker.

Well not any more value than a Mercedes would bring over a Ford Fiesta..... You get what you pay for...

I don't argue this can be done completely by ear. And as I stated, you still have to do some of that even if you're using a meter. For myself it's really just a matter of workflow when I'm setting up my presets. I have my Helix plugged into a small mixing board all the time at home so I can watch the signal meter and I end up with all the presets and snapshots pretty much where they need to be. Occasionally once I get with the band at practice or sound check I may make some fine tuning on a patch or two, but that's pretty rare. The big value to me is that I know the sound man is going to get a very consistent signal level when he's gain staging and that's going to expedite the gain staging process.

A meter is almost an essential element in normalizing the volume between different presets and snapshots. On a single amp you don't have to worry about it quite as much as you do when switching between wide variations in amps and genres. Many, including myself, use external meters either on a DAW or a mixing board to get an idea of whether the signal level is consistent. Of course you still have to fine tune things by ear, but it's a quick way to get everything in the ballpark. Without some form of normalizing you're going to be driving the sound man crazy with all your variations in signal strength.

Line 6 has an annual renewal warranty program. I believe the details are in the announcement above.

The fact is your signal chain is dependent on the signal input and it's more than just signal level. Everything following it will be reflective of it because it builds off of it. It's just physics and the response of the circuits to a given signal. Even on the same physical (non modeled) amp, different guitars with different pickups will sound different. That's because they're supposed to. I perform with a variety of guitars including a Les Paul, Strat, Tele, Gretsch hollowbody, and a Gretsch rectifier. They all use different presets because I design the preset for that specific guitar. I could play a different guitar through that preset, but it's not going to sound the same without some serious surgery on the signal chain. Even then it's likely to be somewhat different than the original. In my pre-modeling days if I used a different guitar I knew it was going to sound different and made adjustments to the amp. At least now I'll get the consistent sound I want when I change the guitar and the preset.

I'm wondering if you're using the cab sims on both the 1/4" out to the amp and to the FOH? Also, what did you use to dial in the tone originally? Probably the quickest fix in this case would simply be to have your FOH XLR out routed back through the stage monitors.

Probably your best bet is to create a template preset that has a basic guitar path and vocal path. Then just copy it over to the preset location you want to work with, rename it, save it, and you're set to go. As talwinkings points out, you have more than enough DSP available to probably accomplish whatever you want in the other path unless you're going crazy. I have probably 30 or 40 presets I've created that only use a single path. Of course the main limitation on such at thing is if you're going into a PA system with other vocals you can't really split your vocal out into it's own channel. But if that's what you're doing there's not really a lot of sense in using a Helix vocal path. That's what the PA is for.

That's been my personal experience as well having used both.

I think what you're referring to here was that as a part of one of the FW upgrades, Line 6 was making some architectural changes in the way things were being processed. This wasn't so much related to changes in the way amps were modeled but in the way amp modeling and all other features in the Helix were processed. I believe this was just prior to the introduction of Helix Native, which was to increase stability, expedite development, and lay the groundwork for the consistency with future Helix products such as HXFX and maybe allowing them to more easily incorporate legacy effects from previous Line 6 modelers. I don't doubt, however, that simply through the process of creating new amps and effects, they've found better ways of doing things as their experience has grown in developing for the platform. That would be natural.

Personally I keep a set of starter templates in one of my setlists. I go to that setlist and to the starter template I want, hit the SAVE button and change the setlist and preset number for where I want it saved on the lower dials, hit SAVE again and it's done and ready to be worked with right where it should be. You can do this anywhere and anytime since you're not dependent on having HXEdit in the process.

Probably one of the most influential components in what frequencies get emphasized or attenuated is the cabinet, the mic (or mic combination), and mic placement. As rvroberts points out, it's more about the total signal chain than it is the quitar. Less so if you're routing your output through a 4CM arrangement than to some form of FRFR. One standard way of addressing high and low cuts is the way Jason Sadites does it which is to place a Parametric EQ block toward the end of your signal chain in order to make all of your final EQ adjustments for the signal chain. I find this to be a very easy way to adjust not only highs and lows and individual frequencies that are being problematic in a given preset's signal chain. Another benefit of this approach is you don't make assumptions about high and low cuts. In many cases I end up not needing them depending on the cabinet and mic mix I'm using as well as many other factors. But there are almost always some surgical EQ adjustments on every preset that will make it sound better, and this is a great way of handling it.

Apparently he's asking for you to be able to read his mind....

Watch this video: Jason Sadites instructional video on using Snapshots

That's the question I asked myself..and I answered "I don't". I suppose it would be of use if I didn't have a Helix floor dedicated to my DAW computer. I suppose there could be some value if you wanted to record a wet and dry channel and then play with different sounds on the dry channel, but that was about the only value I could see.

Apparently everyone missed what you said. There's no reason to send the signal to the board using the 1/4" out R output. Just send it using the XLR L/mono output to the board. You could be inviting problems if you happen to use a stereo effect. In that case you'd get one side of the stereo in your monitor and one side at the board.

I'm not sure the demand is there for that compared to other priorities. You can always enter it into IdeaScale and see how much support it gets, but I'd suspect it would pale in comparison to many of the other items being requested.

If you're on 2.54 you can do a copy and paste using the hardware interface. Select the block you want to copy and press the ACTION button. At the bottom of the screen you'll see the options for copy or clear. Push the button below the Copy then move to the location where you want to paste it and push the ACTION button again and you'll see the paste command at the bottom of the screen. Press the button below it and it's done. It may not be showing up in the editor if you have a mismatch in the version of the editor and the firmware. This can happen if the firmware was downloaded separately from the editor. Oh...and I'm a retired software engineer as well, so we can start our own sub group of Helix users!!!

I'd suggest you contact whoever you purchased the presets from. This is their problem, not a Line 6 or Helix issue.

You can use the Dual Harmony for that but you'd probably have to use snapshots to control the settings as the first part is just using an octave, but the later parts are in harmony. It sounds like it might have at least on key/mode change so you'l likely need to use snapshots for the changes.

If you're dealing with an older FOH system there's not much you can do. The physical way one speaker works can't be conveyed to another speaker by simply connecting a wire between the two. Not to mention, of course, you still have to go through a mixer. In spite of what we encounter in discussions here, there are quite a number of older PA setups out there that are 15 or more years old consisting of passive speakers with physical crossovers and driven from a central amp that won't have the clarity and definition of today's modern speakers. In the cases I encounter we typically don't send the instruments through the PA, just the vocals and things like harmonica and congas. I setup our on stage DXR12's on half height poles behind us in a traditional backline and use them to project directly out to the audience. This works well as long as it's not a large venue or event. But I haven't run into any problems like this in large events and most likely won't. Large events like that will normally use state of the art line arrays these days, or at least very large and capable FRFR systems.

You don't really have to go back to using a DI box. You can also just get a phantom power blocker and put it inline with your XLR out. Provides the same level of protection from phantom power. Just for reference, line and instrument levels apply to 1/4" outs because they are the standards used for line inputs into powered speakers or instrument inputs into guitar amps and such. The XLR is not as hot because it's at the same level as a standard microphone input which is normally an XLR that goes into a mixing board. The actual level used is controlled at the mixing board by the gain/trim knob on the preamp for that channel and is set to an appropriate level by the soundman as the first step in gain staging during sound checks. You'll understand the importance of having the 1/4" and XLR outputs controlled separately if you're using DI box for both 1/4" and XLR the first time you reach down and turn up your master volume control during a set to get more stage volume and the soundman comes after you with an axe handle for messing up his front of house mix....

Snapshots works as it sounds like in that it's a picture of what the settings are at any given moment in the patch. But you can use global settings to set the Helix to to snap/stomp mode so you'll have half of your footswitches as snapshots and the other half as stomps. Then just define the delay footswitch as a momentary switch and you're all set.

One of the amps that was specifically designed for this type of thing is the HiWatt (WhoWatt 100). I use it quite a bit with the amp providing the clean tone and the crunchy tone coming from something like a Teemah with a Deluxe Comp using a Screamer for the lead boost. This was the idea behind the design of the amp was to be able to work well with pedals, although on it's own just using gain and master you can get it pretty crunchy. An extraordinarily versatile amp which was why it was so popular for so long.

As far as sending a full signal from the Helix output, that's actually exactly what Line 6 recommends in the manual i.e. turning the Helix master volume level full on. I personally don't do that, but I haven't seen any problems with the full XLR output at mic level being gain staged appropriately at the mixing board whether it be an analog mixer of digital mixer over the last 2 years. I guess I could imagine a scenario might exist on some more obscure mixer situation, but no real world situations I've run across . I could imagine sending a full line level signal might be more problematic if some older mixing board wasn't capable of applying a pad to the signal, but that situation would be pretty rare I would think. As far as using a mic level signal for a signal input into a mixing board that's been the norm in live situations due to less line attenuation over long distances for decades. In today's world of mixers it probably wouldn't matter as the preamps on mixers have improved considerably, so whatever one is comfortable with. In terms of noise or loss of signal, the thinking is that mic level will be less noisy and with less attenuation, but in practice I'm not sure anyone could really tell the difference in a double blind test of properly gain staged signals in today's environment. I see no real reason to use mic level signals to a on stage powered speaker as it's not going to be a long distance, and line level is what is normally sent as output from non powered mixing boards to fronts and monitors anyway. Bottom line, use whatever you're most comfortable with as long as the sound guy understands what he's receiving. He should be able to gain stage it appropriately without losing signal integrity. If not, it's a simple change in global settings to correct it.