DunedinDragon

It might help if you gave us a bit more info about what you're trying to accomplish with the harmonizer and which one you're using. If you're trying to accomplish dual or triple guitar leads and are using the Twin Harmony (Eventide H3000), the first problem might be that you have it first in the chain. I get much cleaner results if I position it after the amp and cab, particularly if you're using any kind of overdriven signal. Also, if you only want dual harmonies and not triple harmonies, set the second harmony on the twin harmony to 0 as the offset for the harmony (which will be the same as the lead line you're playing). Personally, I start with a single harmony and expand it if necessary, because once I have the settings for one harmony correct, any additional harmony will be correct as well. You need to know a couple of things about the key of the song (and lead scale being used) in order to set it correctly such as major key or minor key, as that determines the scale being used for the harmonies in most cases. Beyond that you just set the harmony offset for the traditional harmonic triad. In other words, the first harmony would be the third (+3), second harmony would be the fifth (+5) in most cases. It's hard to give you exact settings because the selection for key, scale, and harmony offset is really dependent upon the song, but those are the basics.

I've always used a Vox or a HiWatt for Keith Richards sounds. But the guitar you use makes a HUGE difference. I use a Strat with single coils.

That really is odd that you would be clipping unless the wattage rating is way off. I run my DXR12 (rated at 1100watts) about half volume and my master volume on the Helix is also at 1/2. All of my patches are normalized using a sound meter to 80db at those settings. I never come close to clipping even it I bump up my Helix to 60 or 65%. I don't think I could stand to be in front of it at that level for long, and that's just with a single DXR12. Certainly no one ever complains they can't hear me. I did notice that the Yamaha specifically states it can produce an SPL maximum of 132db whereas there's no such specification on the 112's, so it makes me wonder if there's not some loss of actual SPL in the Alesis. Either that or maybe some of your patches are at a high output level and causing the clipping. Have you tried using a sound meter to measure SPL? Wattage really isn't the final say in what a speaker produces, but SPL is really the only thing that matters because that's what you and your audience and bandmates hear.

The only thing that occurs to me is how you may have had your 112's placed on the stage. If you had them down on the floor in a traditional floor monitor way, you would probably be losing a lot of sound energy into the large upper space of the barn that would normally get reflected back to the audience in a more typical room. I always keep my FRFR speaker as a floor monitor in a normal backline arrangement which is fine for myself and the band, but that's because we always send instruments and voices through the PA so I don't need to project into the audience. In the situation you describe I might have chosen to use a couple of speaker stands so the energy would be more directed into the audience space.

I think this is the one benefit I derived from going to a FRFR setup that I never predicted. Having exactly the same sound whether it be at my house, at the rehearsal space, or at the performance. If there is any major discrepency in room acoustics I can easily adjust global parameters one time and everything sounds consistent. Who'da thought???

I can't respond to a signal spit after the cab although I suspect it may be related to the type of effect and possibly mixing in a stereo versus mono effect in such a split. As far as the differences between amp/cab types, that's normal on all modelers. Amps are modeled based on how their particular design responds to changes in channel volume, drive, EQ, and main volume, along with how the cabinet responds to it's input. They're just modeling what a real amp would do and they all respond differently. You can easily even these differences out by modifying the level setting on the final output block of your signal chain. The best way I've found to deal with these differences is to start off with an amp/speaker combination that is closest to the sound you want to achieve. Build out the signal chain until you get what you want, then level set it with your other patches using the output block level.

I use the Jazz Chorus (Jazz Rivet) quite often on various styles of songs..mostly Steely Dan type jazz/rock and some BB King blues stuff. Most often it's with my Gretsch Silver Falcon. My most common setup has drive and bass in the mid 3's, and Mid and Bass right around the halfway point. I use the standard 2x12 Jazz Rivet cabinet with a 212 condenser mic at about 4". Bear in mind the Gretsch is a hollow body so you may need to make some adjustments as it has a pretty full bodied sound relative to a solid body. I can't help you with the SoundSource though...never had any experience with it.

To be honest I'm not sure there is a single answer to that. I think most of these manufacturers keep that type of design info pretty close to the chest as it's generally proprietary and can be a differentiator in the marketplace. I'm assuming it's a combination of both cabinet/speaker designs as well as certain DSP elements, but that's just a guess. I also assume the techniques vary depending on what type of design they're using such as traditional speaker versus a line array.

I think I already have, but I'll repeat it if you didn't understand what I was saying. High end guitar speakers and cabinets, are not designed to perform the same function as FRFR speakers. If you want a clear example of it, try pushing vocals through a guitar cabinet and you'll create a very unhappy vocalist. Obviously frequency response plays a part in this, but that's not the only difference. Beyond that,cabinet design and DSP play into it as well in order to provide some very key differences that are clearly audible. The most important attributes are the way sound gets projected evenly and across a greater distance than will a traditional guitar cabinet and at generally lower SPL. Guitar cabinets are designed (or better said, NOT designed) so that sound is projected EQUALLY in all directions, but not EVENLY in all directions. This is what accounts for the fairly significant difference mic placement can make as well as the sound you hear depending on where the cabinet is placed relative to the listener. Modern FRFR speakers address both of these issues in order to provide longer projection and a more even distribution because that's what they're required to do. Unlike a guitar cabinet, sound is projected so that it's wider than it is tall. They also have very significant drop-offs in sound levels outside of that rectangular sound space. This conserves sound energy that would end up being lost into the ceiling and the floor , as well as behind the speaker and therefore that energy is sustained across a longer distance. Likewise, other design characteristics provide for a more even projection of ALL frequencies across the entire physical width of that sound being projected. This better ensures that someone sitting directly in front of the center of that speaker will hear pretty much the same thing as someone off to the side of center. In effect eliminating the on/off axis tendencies you find in guitar cabinets, as well as the fairly rapid drop off of SPL across longer distances. All of this might be interesting, but has little to do with why one would choose an FRFR cabinet over a guitar cabinet. The fact is, the sound is (by design) different when it comes out of a FRFR speaker and when it comes out of a guitar cabinet because it's DESIGNED to be different. And that is exactly why one chooses to use a FRFR speaker, because it IS different. Therefore what you hear on stage will be much closer to the sound being sent to the audience. Additionally, because of the even distribution of sound, other band members will hear a more accurate and articulated representation of your sound at a much lower SPL than the muffled sound they might hear from being well off center from a guitar cabinet. That's why it's important that if you use a FRFR speaker you should design your patches using that FRFR speaker in order to produce exactly what you want the audience to hear. And this is where the real advantage of FRFR speakers come into play. Designing patches targeted for a FRFR speaker has quite a bit in common with the way you mix and master a guitar sound in a studio mix to precisely manage where it will sit within the soundstage of other instruments and vocals. Something that's much harder to do with a less accurate speaker. Granted, some of these things aren't particularly useful in situations where you aren't depending a great deal on the FOH for your sound distribution (such as a smaller bar). But even then you'll gain the benefit of a more even distribution of sound with a longer throw at lower SPL with a better chance (if patches are designed correctly) to cut through the mix.

This is a perfect example of science running headlong into the harsh realities of the world. In spite of all the classroom minutiae of how the sound is processed, that sound will ultimately be heard by human ears, all of which respond independently and are subject to the physical realities of their locations in relation to the sound source. This is where the physics falls apart because we rarely, in live performance, are blessed with an ideal placement for each person in a perfect acoustic setting. So we try to cover as broad a spectrum as we can in the design of PA speakers systems to at least give the best sound we can to the most people in a way that's evenly distributed. What we hear on stage may or may not be an accurate representation of that sound once it's processed for the ears of the audience, but one thing is for certain. A guitar cabinet will not produce that same sound or sound distribution in the same manner as a modern PA system whether that be by conventional powered speakers or a line array. The best you can hope for, which is what many of us opt to do, is to emulate that sound production and sound distribution through the use of similar mechanical/sonic processes so that we can tailor our sound to best match the ultimate output system. I've said this before and I'll say it again. I opt to use the powered speaker I use (Yamaha DXR12) because it best simulates the sonic characteristics of the system I know (most of the time) will be projecting that sound to the audience's ears (QSC KLA12). I don't stand on stage so I can selfishlishly please myself and my ears, I stand on stage to ensure the audience experiences the best sound experience I can give them. And in order to do that I need to focus on what THEY will hear, not what I will hear, and this is my best shot at doing that.

Based on what I hear commented about on here and other boards my guess would be: Some additional or possibly re-tooled reverbs. Some type of IR management system worked into the Helix Editor. Possibly some metering facility for measuring output signal strength. I would expect maybe of few additional amps and effects, but they've got most of the big bases kind of covered so I'm not sure what amps they might choose. I wouldn't mind seeing something like a LabSeries L5 for BB King type tones, possibly a Dumble, maybe a Two Rock. I can't think of many others that might be worthy additions other than some of the more arcane Fender models like the Showman.

There really are two different perspectives on this cabinet versus FRFR situation. Although I'm concerned about what I hear on stage, I'm FAR more concerned about what my audience hears. They are the reason I'm playing, they are the ones paying the tab for me to play. What good is it if I have the greatest tone ever on stage if the audience isn't getting the benefit of it? This is the reason I made the commitment to using a FRFR rig. This often results in taking more time to dial in my patches to get the sound closer to what a traditional cabinet would give me, but I have absolute confidence that the mix of my instrument and the rest of the instruments and vocals on stage matches quite well with the experience the audience gets, and have confirmed this with recordings I make when we perform that produce 2 stereo tracks of ambient sound and 2 tracks of direct sound out of the mixing board using a Tascam DR-40, as well as occasionally capturing a full multi-track recording off of the board using an Allen and Heath ICE-16 . I can honestly say that this is the closest I've ever come to achieving studio quality sound in a live performance. I've played live for well over 40 years and I've had my share of traditional amps and great cabinets, but this is the first time in my career that I can confidently say the stage mix I hear is precisely what the audience is hearing, and that's the real target for me.

You might be surprised how many of us don't use the editor. It takes me twice as long to do adjustments with the editor as it does directly on the Helix. Not to mention the lag time between the action taken in the editor and it's reflection on the Helix unit itself. I've never had a freeze up, but the Helix is only rarely and briefly ever connected to a running PC or editor. Typically if I'm not storing presets on the PC's hard drive, the PC is turned off, so the USB connection is totally inactive. That's why I'm thinking these freeze ups may be related to the USB.

It may not be the editor that's directly causing the freeze ups, but the USB connection that somehow gets blocked. I know there have been several issues come up with the way USB ports are implemented on PC's that have caused problems for the Helix. Next freeze up you might try disconnecting the USB entirely and see if that helps.

I think rvroberts gave you a pretty good overview of some of the things you may have to deal with in your setup with a Helix, but just to expand on a couple of things here. I'm not sure why, even with the X3Live your presets would run so much louder. An integral aspect of modeling systems is that you don't really use volume to craft the tone. The tone will be the same regardless of the volume setting. Even on the X3Live the Master Volume only controls the level of the signal going out and has no further effect on the tone which is crafted using the other controls such as Drive and Tone Volume. The idea being, with X3Live, HD500X, or Helix, or any other modeler for that matter, you craft the tone you want and use the Master Volume to manage the overall output level (no effect on tone). That is then sent to (ideally) a clean output amp (no pre-amp or gain) and on to speakers. As rvroberts pointed out, you may not have experienced it so much with the X3Live, but certainly since the HD series and with the Helix the precision of the modeling has increased exponentially along with the separation of the effect of the amp, the cabinet, and the mic/mic placement has on tone (along with numerous other things like Bias and Sag). Naturally you can bypass these effects and depend on a traditional guitar amp and speaker setup (bypassing the pre-amp) to provide the effect of the cabinet and mic, but you will then have to deal with any direct signal going to the PA or to a recording system which will be very unnatural sounding because it's not mitigated by the cabinet and mic effects. If you want to keep your current setup with the powerblock and the 2x12 cabinets then you would need to make some choices about how to deal with this situation as outlined in rvroberts post. Alternatively you can go the route of least resistance (which is what many of us have done) and move to a FRFR setup which solves the entire issue. The value being that the sound we hear on stage from the FRFR powered speaker will be exactly what comes out of the PA or gets captured in a direct recording. That's not to say that you won't have to do a bit of tweaking of some very high and very low frequencies that are not naturally heard on most guitar rigs. Most of us have found we need to place low cut filters somewhere around 100 to 140 hz to tighten up the low end and get rid of boominess, and high cut filters generally up in the range of 3500 to 5000 hz to get rid of excess harshness. Of course all of these things really depend on the modeled cabinet and microphone you use on the patch, but generally speaking these frequencies are outside the range of a typical electric guitar rig. It may help if you begin to think of the Helix more as a recording production studio rather than an effect or modeling pedal. With the Helix you're doing the same thing you would do in a recording studio which is develop a signal chain consisting of effects, amps, cabinets, mics to precisely craft the tone you want to capture. Once you've designed that signal chain and it's components you manage the output of that signal using the Master Volume into a clean, full-range recording or live output mechanism. Hopefully that helps give you some perspective.

Okay...here's my on stage workflow when we go to the next song in the set: Press the preset up footswitch. Start playing the song. And the weird thing about it is, I have yet to uncover any need to incorporate snapshots into my patches. And that's not because we don't have songs that have some significant changes in effects and tones throughout the song. And so far, even with fairly dramatic changes in the tone or dynamics of a given song, it still comes down to a single footswitch stomp to activate/deactivate the changes. Again, no magic here. No snapshots. All using the basic capabilities of the Helix that have been there since day one. And more importantly, there's never any confusion about which song I'm playing in the set because it's displayed clearly as the preset name. How is all this possible? Simple. Every song is stored as a preset and maintained on the hard drive of my tablet PC by the song's name with it's default patch settings for the start of the song. When I set up the Helix for a performance, I simply import the preset into the appropriate preset slot and everything is ready to go. Generally speaking most of the songs have only one or two stomps enabled. I may have many more effects in that song, but if they stay the same throughout the song (such as a compressor, or a reverb), they aren't assigned a footswitch so my stomp arrangement for each song is super simple. Nothing to remember, nothing to screw up during the performance. I often think we're our own worst enemies when it comes to technology. Just because a capability exists, we think we need to use it, regardless of whether it really benefits our workflow or not.

I don't download patches, but the process should be the same whether you create them or download them. I set my Helix master volume at 50% and use a sound meter to set every patch to be around 80 db by adjusting the level on the output block. 80 db should be more than adequate to get past any human ear limitations of Fletcher-Munson, and is generally about normal stage volume in my band. Of course you can choose whaterver level you want on the master volume and adjust your output amp accordingly. Just pick a setting then target 80 db and adjust the output block.

Are you making these changes directly on the Helix or through the Helix editor?

It sounds like you're running basically into a power amp with a traditional guitar cab from the Helix, so it makes sense that if you have a modeled amp and a cab in your signal path it's not going to sound right. Ultimately you have a physical cab already and by the description you've given they're simple traditional guitar cabinets with the normal limitations in terms of range that a guitar cabs have (i.e. NOT FRFR). You would notice something completely different if you were going into a FRFR speaker without cab emulation. It would sound WAY too harsh. FRFR speakers depend on the cabinet emulations to model the sound before they get them. What you're ending up with is an amp, a cabinet model, going into a physical cabinet...i.e. a cabinet going into a cabinet. In essence in this arrangement you can make use of all the amp models, but you would be limited to the physical cabs you have. The other issue is, without the modeled cabs in the signal chain, your direct output to the PA is going to sound very harsh. You'd be better off just mic'ing the cabs you're got.

I'm not sure what you mean by a global speaker emulated output. Every speaker is different. Every PA is different. That's why you see so much discussion in this thread about cutting various frequencies to tame the broader range used by PA and FRFR monitors. There's no one single perfect answer. Probably in your case a second chain might be your best bet, but I doubt it will sound anything like what you're hearing on stage to your audience when they hear it through the PA.

If you're talking about the freezing that occurs if your Helix is attached to the Editor and you scroll quickly through the presets on the Helix then yes, I've experienced that. At least when using Windows 8. It's not an issue for me since I do all my editing on the Helix itself. I just disconnect from the Editor and all works fine.

I know exactly what you're feeling. I keep my HD500X around just in case, but I dread the thought that I might have to resurrect it should my Helix go down. At this point I'm not even sure I'd know how to program it!!!

I figured out about 30 years ago that no amount of gear will ever make me a better guitarist. Saved my sanity and my pocketbook..... ;)

I don't think Line 6 had anything to do with "forcing" you to play through anything. There are tons of users happily playing through traditional amps or through FRFR speakers. Your choice to go through all these various permutations is your choice not Line 6's. The fact is the Helix will work with most any output system you want to throw at it. Personally I had no qualms about going the FRFR route with my Yamaha DXR12's. But I knew what I was going to do BEFORE I got the Helix based on my research. Reading your post is like the old joke about the guy that goes to the doctor saying, "Doc, it hurts when I do this." The doctor says, "Well then, don't do that!"

I think this just highlights the differences in how the Helix can appeal to different types of musicians. For example, I end up somewhere in-between these two thoughts. I make extensive use of various amp models using different guitars depending on the style and feel of a song, so the Helix is crucial to being able to do that in a live environment. Although I'm not a sound experimenter probably as much as WickedFinger, I do make use of some of the more exotic effects if it benefits the song. More importantly though, the Helix provides me the ability to achieve studio-level capabilities in signal chain routing if called for by a given song. So I suppose the point for me is I'm able to achieve a range of capabilities in a live environment based on what each song might require.