DunedinDragon

I think you're missing out on some great potential. A lot of great recordings you've probably listened to over the years were done by mixing different cabinets and amps and mic's. It's great to be able to do what used to take hours or days in a studio in just a few minutes all in one box.

I think one of the best design elements in the Helix is the ability to easily audition various amp/cab/mic combinations when using the amp+cab block. Once you've selected an amp you simply move over to the cab segment and you can cycle through any of the Helix hybrid cab selections as well as cycle through the various mic settings. All of this can be done before committing it to your signal chain. I use this a LOT when laying out the fundamental model I want to use which can greatly minimize the amount of tweaking one has to do with the amp.

My setup has been pretty stabilized now for about a year. We're a 7 person band with two female vocalist, a male vocalist and harmonica player, a rhythm guitar player who occassionally sings, myself as lead guitar, bass and drummer. What I've settled on has a few drawbacks, but I've adjusted to it as has the rest of the band, and it seems to work well across pretty much every venue we've encountered. I use a Yamaha DXR12 as a floor monitor positioned behind me and turned slightly toward the center since I stand a bit toward stage left. I try to keep it positioned about 4 feet behind me which helps to give me the fully formed sound from the speaker rather than be overwhelmed by the horn. Our own PA system consists of a QSC line array using a floor setup with two stacks (one left, one right) each consisting of a KW181 sub and 2 KLA12 liine array speakers. We use 2 traditional floor wedges at the front of the stage, but they only have voices and the harmonica in them. All of our instruments and voices go to the FOH. The Bass and I both go direct XLR to the board and the rhythm guitar player mic's his amp. On occasion we do play some larger venues either outdoors or in larger auditoriums with their own professional sound people and systems. In those cases we'll get separate side fill monitor mixes for the rear of the stage area. Regardless of the venue we stay at a fairly consistent stage volume. I normalize the volume of all of my patches to right around 100db with the Helix master volume set at 50% and the volume of my DXR12 set at 50%. Depending on the situation our drummer may use a Roland TD-15K drum kit which he sends to a Mackie Thump15 on a speaker stand and angled across stage toward the center for stage sound with a separate direct line to the FOH desk. He's a pretty active drummer but he stays pretty consistent in his volume levels whether it's an electronic kit or acoustic kit so we rarely vary our amp/monitor stage volumes regardless of the venue. The bass player stands next to me and faces his bass amp angled toward center as I do, and the rhythm play stands on the opposite side of the drums from me and faces his amp (Mustang III ver 2) angled toward the center as well. In this setup we can all hear ourselves pretty well as well as hear each other which allows for better blending of our playing. We've occasionally discussed the possibility of IEM's but the band has been together so long (10 years with only a couple of changes in personnel) that we've got things down pretty solidly with our setup and we pretty much have come to the conclusion that if it ain't broke, don't try and fix it. I could see it if we were having the problem of controlling our stage volume or an inability to hear each other, but that is just never the case. We rarely play in really small venues, but on the occasion that we do, we simply take the instruments out of the PA and our typical stage volume seems to be about right. Here's a video someone took of one of our performances from last weekend that will give you some idea of our setup. We weren't terribly pleased with the job the sound crew did in mixing as they seemed to be adjusting it throughout the performance. In this clip they never did get the rhythm guitar level correct and pretty much messed up the female vocalists at the beginning of the song, but got most of it somewhat right after a minute or so. Just ignore the silence at the beginning of the clip. The person taking the video didn't turn it on for about 10 seconds or so. https://www.facebook.com/SalvationSaloonPosseBand/videos/1694896814167297/

I think this is one of those areas that is a "must learn" thing that comes with these type of modelers. I know when I first got my HD500X I was a bit upset when I encountered the harshness and boominess my patches would have going direct into the PA, which started me down this road of understanding how to best deal with the issue. And I've learned a lot about how to best deal with this over time and it's modified the way I build patches. Surprisingly, I could have learned it much easier if I had just paid attention and watched a few videos of how some people were building their patches. The good news is, having learned some of these things I honestly feel like, for the first time, I'm getting some of the best performance out of my patches I've ever had. Like most I originally focused on hi and low cuts which do a pretty good job of getting the patch presentable through FOH FRFR speakers. I've recently learned how cabinet selection, mic selection, mic placement, early reflections, and final EQ can really polish the sound to be exactly what you want. Hi and low cuts still play a factor, but not as dramatic as it was in the beginning. What's remarkable to me is how much of an effect cab, mic, and mic placement can have in mitigating the harshness and boominess just by themselves. I've also started to incorporate some form of final EQ on the signal in the form of the 10 band graphic EQ or the parametric EQ to limit or enhance the frequencies I really want to hear. I still use a hi cut set around 10,000 hz, but I use the EQ to gradually dial down the upper frequencies which makes the sound much more natural. The same holds true for some of the lower frequencies between the low cut at 125 hz (which is handled by the FRFR speaker) and 1000 hz. Depending on the cab, mic, and mic placement I may also adjust some of these low and mid-range frequencies up or down. It's about 4 or 5 minutes of additional work, but the sound is remarkably better not only to my ears but to the ears of my bandmates and our sound guy based on their feedback. Normally I do all of this before adding any additional effects to the signal chain. I figure if I get the base tone where I want it, all other things will fall into place, and that seems to prove out in practice.

I doubt you'll get a definitive answer on this, but I'm not sure why that would matter either way quite frankly. The two cabinet processes are clearly different, but both serve different purposes, and are useful in different ways. Method One provides a very exact duplication of speaker, mic, and mic placement, but it can't be changed without changing to a completely different IR. It's also subject to the environment it was recorded in and carries with it those artifacts from that environment. Method Two is probably not as precise in it's duplicaiton but offers the advantage of being able to quickly and easily change microphones, placement and certain room artifacts. For myself I prefer the convenience of Method Two as it allows me to quickly audition various setups to get to the sound I'm looking for. Some people may prefer to be purists and are willing to pay the penalty in terms of lack of convenience and flexibility. I'm just glad I have the ability to go either way if I want to.

The Helix application has no control over how your text appears. It just hands it over to the operating system for display, so you need to look at how you have your display drivers setup and if you have the appropriate drivers.

I'm not sure whether you're talking about a personal mixer prior to sending to the main mixer, or direct into the mixer. In either case I would assume you would be connecting into the mixer via an XLR input on a single channel (if you're using mono, two channels for stereo). I don't see any particular advantage to using a personal mixer. You have all the facilities and more of a mixer built into the Helix. It's quite easy to control the output volumes between different patches simply using the volume control in the output block of your patch. Of course this should be done ahead of time so that all patches are roughly the same volume assuming, of course, that you're managing the increase in volume for leads. The only thing you would need to adjust at the gig is the main volume knob on the front of the Helix to match up with the room. To be honest, I would be hesitant to depend on any sort of EQ from the mixing board. Most mixing boards don't have anywhere near the facilities to manage your tone effectively, whereas the Helix has everything you could ever need to do such things. This is the reason so many of us use FRFR monitors on stage and at home to develop and setup our patches. You need to have some way of gauging how your patch will sound through the FOH to get the tone correct. Depending on the FOH guy to make the adjustments is a disaster waiting to happen...and this is coming from someone who's run FOH sound for 4 decades. Your best bet is to send the front mixer a fully formed, equalized, and volume managed direct XLR signal and just leave the guitar channel on the mixer set to flat. Once you've set your master volume on the Helix to match what's needed for the room, the person at the mixing board will manage the level of the signal you send him to blend it appropriately with the other instruments and voices.

I suspect that plate is just connected to a microswitch that tells the helix there is no left output connected, therefore it needs to sum both channels to the right. Most XLR male ports don't have anything like that. The lock on the end of a female XLR cable should still be functional without that plate.

Actually the reference was 80 to 100 db to overcome the limitations of Fletcher-Munson. And 100 db (averaged) is certainly a reasonable level for most of the instruments. Peaks will naturally be higher on things like drums, but those are instantaneous so they don't really interfere with a fairly solid 100db from a lead guitar. The fact is you can go higher, but that sound dissipates rather quickly across space because traditional cabinets (and particularly drums) drop off very rapdily due to the sound energy being lost in their unfocused output (into the ceiling, floor and off to the sides) unlike FOH or FRFR speaker which have much more confined and focused energy in a much tighter sound projection scheme. This typically ensures that if the instruments are too loud, the front few rows of people won't hear vocals very well, and if the sound guy compensates for this the back rows won't hear the instruments very well. The theory that the sound man will be able to correct things like shrill highs and boomy lows is pretty much fantasy. Although the newer digital boards have most of facilities one would use to correct such things, it's highly unlikely that most sound people in real life (who often struggle with the simple gain staging of all the inputs) would allocate the time to apply the hi and low cuts much less the graduated EQ slicing on the highs on your particular channel to get the guitar tone to to something reasonable. Clearly on an analogue board most aren't even equipped with the appropriate outboard equipment necessary to do such things. Most of those boards are equipped with a fixed low-cut, no high-cut, and a single sweepable mid with a fixed spread (Q) on each channel if you're lucky. The may have a graphic EQ with can be used in final stage to correct all channels prior to being sent to the FOH speaker, but those adjustments would apply to everything (voices, cymbals, high hat, harmonica, etc.). That's kind of like saying, "I don't need to use a compressor, or a delay, or a reverb on my guitar. I'll just have the sound guy do it". Good luck with that....

I'm probably the wrong person to ask because I always side with giving the audience the best I have to offer them and pleasing myself comes second. If not for them I wouldn't be there at all. My job isn't to inspire me, but to inspire the audience. Personally I don't think it's as big a problem as you're thinking it is. I'm sure you will be able to tell a slight difference, but if you tweak your settings correctly it shouldn't be that much off. Most of the tweaking you're going to be doing is going to be at at the very high end and low end of the frequency spectrum, which your in-ears probably don't replicate that well anyway. And that's why you haven't noticed the issue, because your in-ears don't work that well in that range of frequencies. As far as Fletcher-Munson you don't need to be in the ballpark of the FOH volume. You should simply be at a reasonable stage volume somewhere around 80 to 100 db. Just enough to hear all the frequencies and be able to notice the adjustments you're making. In fact that's a pretty reasonable level to be sending your signal to the board.

The vast majority of the time I'll use the Minotaur but I occasionally use Tube Screamer or ProCo Rat.

Just to be clear the EV-ZLX-12p IS a true FRFR speaker, no different at all from the CLR. Pretty much every powered monitor/speaker on the market would be considered to be true FRFR.

I used to be a much bigger advocate of using Global EQ as a way of taming these issues across all presets, but the longer I study the problem I tend to opt more for addressing it on a patch by patch basis. Much of this comes from more experience in learning how to better exploit different cabinets, mic's, and mic placement which can address a certain level of these issues and then refine the sound using a combination of minimal high and low cuts at the cabinet level and the 10 band graphic EQ to soften the abrupt cut-off on frequencies that high and low cuts provide. I'm much happier with the results because I can get it a lot closer to the tone I'm wanting that way.

This is exactly the reason I use a FOH quality FRFR both at home and on stage, to avoid the surprise of what's going to be coming out of the front speakers. It doesn't surprise me in the least that you found the lower and higher frequencies much more prominent once you went to a powered monitor. And yes, chances are that's exactly what was coming out of the FOH in the case of your other gig where you were using in-ear monitors. That's why many of us opt to use FRFR speakers so we can hear what the audience is hearing. Can it be corrected at the board by a good sound man? If the sound man is very experienced at dialing in guitar tone and has the right equipment to do so, then yes. But in my 40 plus years of working in this industry I figure you have the same chance of that happening as you do of opening up your guitar case at a gig and finding your guitar case has been filled with gold bullion. First, it's not really as much about the level of the bass as it is about the width of bass frequencies being produced. That's why tightening up that bass by limiting that range through low-cut filters is so critical. How much you need to tighten it up will vary based on the sound you're trying to achieve, the amp and cabinet, choice of modeled mic and it's modeled placement relative to the speaker, and will likely change from patch to patch. The same is true of the high end. On FRFR speakers typical of front of house systems, they tend to produce much more energy in the upper ranges particularly because they employ compression drivers and very often horns. Again, it needs to be address by limiting the range of frequencies in that high range that you will allow to pass through using some form of high cut likely in conjunction with a graphic equalizer to gradually trim down that response on the upper end of the frequency range. And once again this all works hand in glove with the amp model, cabinet, mic model and modeled placement of the mic on the cabinet. In my experience tightening up the bass frequency range is simple in comparison to getting the high frequencies where you want them You have plenty of facilities for doing every bit of this built into your Helix. It's not very likely your sound man will have all of those facilities to dedicate to your guitar channel, nor will he be likely to spend the time to get it right. That's why you're much better off doing it yourself with the Helix and just sending the FOH the signal you want sent to the audience. The advantage of this level of effort is that the audience will hear EXACTLY the tone you want them to hear which, in my experience, has rarely ever been true even using a real amp with a mic or a direct out line for the same reasons. But I can now say with a GREAT deal of confidence, because I have checked it out in real life by comparing my stage guitar tone to the FOH sound on various setups, that it's a very precise representation of what I designed that guitar tone to be.

As long as the band can hear themselves well enought to blend the music together, you can hear yourself well enough, and the audience can hear everything correctly those are the only three things that matter. If the equipment does that you don't need any more equipment. If the equipment can't do that, you need more equipment. It's pretty simple really.

I keep my monitor (Yamaha DXR12) volume consistently at 50% even when I'm normalizing the volume on the patches. I don't want to have to adjust it at all. It's easier to make all the adjustments on the Helix master volume knob at sound check time for any room differences.

Personally, I reserve the Master Output volume knob as the way to manage my overall volume across all of my patches. Typically I leave it at about 50% and normalize the volume on each patch to around 100db by using an external sound meter. I set the amp model volume and drive to get the tone I want, but I seldom set my amp model's volume much above 8.5 or maybe 9. Quite often, depending on the amp model, it may be set somewhere around 6. Some of the amp models have a master volume setting which can also sometimes affect the tone, but most of the time that defaults to 100. Once I get the tone I'm looking for, if it's below or above 100db I'll adjust the volume level on the output block as that seems to have very little affect on the tone and simply increases or decreases the volume. Generally 100db is about perfect to mix well with the stage volume of our band. But on occasion on larger stages, I may need to increase it (as will everyone else) in order to hear each other, so I only have to adjust the Master Output volume knob on the Helix and all of my patches will have the same increase or decrease.

A higher setting on early reflections (in conjunction with mic distance) may give it a bit more "in a room" feel, but that's only an improvement if what you're looking for is the sound of hearing one of those amps in a room. Since I've always played live with my amps mic'd what I hear from the Helix is far more indicative of a mic'd Twin or Deluxe than what the Mustang produced. Of course once you mic the Mustang it's pretty much the same sound as the Helix. But conceptually adding the "room" feel via the early reflections and mic distance might be a way to get a "room" feel through the PA from the Helix. I might actually try that at some point.

There are a number of people that are using the TS210 series as it's a pretty decent speaker for the money. The Line 6 stage source L2M's are pretty popular as well as are the Yamaha DXR series (which is what I have). The Line 6 and Yamaha's tend to be a bit pricier, but are a bit more substantial and more representative of state-of-the-art design for live sound reinforcement applications and are many times used as front of house speakers for many bands. The QSC line could be considered much the same. The reality is, in live sound reinforcement applications which is what FRFR really represents, you honestly do get what you pay for generally. I would also agree with your band leader in getting an upper end type of unit. You can always dial out frequency ranges you don't need on an upper end unit, but you can't really dial in a lack of frequency response you might get in a lesser design. I personally can't say enough about my DXR12's that I use on stage. You do have to dial out some of the high end and low end in your patches, but I've found the DXR to be incredibly responsive in reflecting changes I apply to my patches. It takes a bit of time to get the feel for how to use all the facilities (amps, cabinets, mic's, mic placement, Early reflection, SAG, BIAS, and EQ) that are in the Helix in order to dial in exactly the sound you want, but once you figure it out it's incredibly quick and easy to dial things in exactly as you're hearing them in your head.

I think your perspective is still very jaded by a traditional stage and amp setup as being something that needs to be preserved because it's something special. In many ways the high end modeling environments were created in response to the limitations of such a scenario. I don't know this for a fact, but if the general responses I see from Helix users in here and on other boards is indicative, I would say there are far more that have gone the route of the Helix either direct into a PA or into a FRFR rig and direct into a PA. If you examine the outputs that come with the Helix it's obvious it was designed with that in mind. There are certainly any number of ways you can try to preserve the amp environment, but the path of least resistance is clearly to go direct into a clean power amp and some form of full range speaker setup, because that is ultimately what the core of this system is designed for. An analogy would be, if all you know is a hammer everything starts to look like a nail. You might be surprised to find out how much more functional and useful the Helix is if you accept it as it is designed to be, an entirely new way of addressing live performance as many of us have done.

Another option to consider might be to bypass the amp altogether and simply go direct to the PA and use the monitor system for your guitar on stage. This way you could use all of the facilities of the Helix as it's meant to be used. Sometimes the simplest approach is the best.

Do you have trails set to ON in your configuration? It's typically set to OFF by default.

I have no problem if others are happy with the G10. In my case the area that had the most difference were very clean and brighter presets with not much in the way of drive, compression, or effects. I'm no novice to sound production both live and in professional studios in Nashville, Memphis, and Austin for over 4 decades, so I know enough to know how to not be deceived and tp verify my perceptions. For me it was significant enough to stick with the cable. For others that might not be the case and if they're happy with their units more power to them. I only pointed out my experience, and that it didn't fit my needs.

As many have alluded to here, you're making a significant change when moving to a modeler, particularly one with the range of capabilities of the Helix. So the key question is how much time are you willing to spend with it to learn how to master it? The first thing I always tell people that are considering this leap is to put in the forefront of your mind that unlike a traditional setup, the sound will be that of a studio produced guitar sound. Think of it this way, when you go to record in a studio, there is one sound you hear when you're playing and capturing the track, there is a different sound when you playback that track. That's because several things come into the picture when the sound is captured, most importantly the characteristics of the mic and the room ambience at the very least. The same will be true of the sound coming from the Helix generally. You're not simply modeling the amp, but also the cabinet, the mic, the mic placement, and even the early reflections of the room. It's something some people have a hard time getting around. The audience has no problem with it, because that's the sound they've been hearing from you live anyway if you typically mic your cabinet. It's also the sound they're used to hearing when they hear recorded music. But some people can't get past having a finished (produced) sound on stage because they're not used to it.

I noticed it mostly on the pick attack, but generally the crispness and articulation was gone. It was similar to what I had experienced with the G50 that caused me to go back to using a cable. That's the reason I spent so much time A/B'ing the patches, because I wanted to make sure it wasn't my imagination.