DunedinDragon

To be clear here, feedback is a factor of certain frequencies that a pickup is sensitive to and causes feedback. If you slice out that frequency, you can achieve more volume. That is what the Feedback Wizard is for on your Touchmix mixer. The feedback wizard is an automated process that can be run at the board during sound check. It will increase the volume until it detects feedback, then reduce that frequency to cancel the feedback, then repeats the process over and over through the 12 allocated cancellation filters as it increases the volume so that you can maximize your volume without encountering feedback. It's a built-in feature of your touchmix mixer. In other words, it's doing the same thing as your Felix preamp blender was doing but in a more automated and extensive fashion. What I would probably do in your case if I wanted to do this is to mute all other channels other than the acoustic guitar and then run the Feedback Wizard and let it find and slice out the appropriate frequencies that would cause feedback. However that might be overkill since there's nothing you can accomplish with the Felix preamp blender that you can't do with a parametric EQ on the Helix. If you know what frequencies you were cutting on the Felix, you can do the same with the Helix's parametric EQ block.

Actually I'd probably try out the chorus/phaser in all sorts of places and see what fits best. That's one of the hallmark of those type of effects. They can go pretty much anywhere and each position will have a little bit different effect.

I'd bet with some pre-planning you could settle on just a handful of key presets based on music styles that would get you the versatility without overwhelming you. When I do open mic nights or jam sessions I have a set of about 7 or 8 presets I can choose from that cover 50's, 60's Brit Rock, 60's Flower Power, 70's Rock, 70's Funk....and so forth.

First thing I'd do is run the Feedback Wizard on your touchmix. That's the easiest and fastest way to gain headroom. The other thing that's worthy of attention may be to use a sound hole cover on your guitar which is what I do when I use an acoustic guitar in our band. Also I'm not sure the Jazz Rivet is doing you any favors in your signal chain with an acoustic guitar. Generally with acoustics I simply run a simple gain block along with the 3Sigma acoustic IR to maintain the natural sound of the acoustic. The Jazz Rivet may be causing some of your gain staging problems with it's tone stack and pre amp.

I'll give you my take on how I'd approach building that particular sound for 'She Bangs the Drums' in hopes it gives you some general ideas for how to approach these kind of things. First, that's a fairly clean and compressed guitar sound with some rather heavy reverb and phaser/chorus effects to my ear. There are a number of amps that would likely work with for this sound such as the HiWatt, Twin, Interstate Zed, Soldano Clean. Archetype or Placater Clean, but the one that jumps out at me would be the Matchstick Channel 1 with a fairly low gain setting around 2.5 to 3.0 to give it a small amount of bight. I think that amp combined with an Alnico Blue speaker from the Ownhammer VC30 set of IRs in your Core Tone Bundle would give you the traditional BritPop sound or close to it. Most likely you'd want to start with the IRs in the QuickStart set. This is one area you'll have to play by ear. I'm a particular fan of ribbon and dynamic mic mixes so I'd start with the OH2F-07 or OH2-07 in that group. If that's too dark for what you want you can always go over to the Mixes directory and try out the same OH2F or OH2 with the mic's closer to the cap (03 - 05) and see what suits you best. Once you get the base guitar sound right, then I'd start adding the effects. I'd first focus on the compressor probably at the beginning of the signal chain using the Deluxe Comp. To my ear it sounds like it's moderately compressed, but present enough to be noticed. After that I'd play with some different reverbs (heavy on the mix parameter) and phaser/chorus...both after the amp and IR. Since you're working with the stomp if you have to sacrifice a block due to the Stomp's limitations you could probably lose the compressor if necessary. At least that might give you some ideas of where to start.

I have to say it's fairly rare for me to end up using two different amps and cabs within the same song to get a dramatic shift in tone and dynamics. It happens occasionally, but it's rare. Most of the time I can accomplish it with the same amp and just tweak gain settings or add a drive pedal. The only one that really comes to mind is the is one in which I use the Soldano Clean with an Alnico Blue vox style cab combined with a Brit Plexi Bright with a Marshall celestion cab because it really needed a more modern clean sound and a vintage overdriven sound. But I have to go along with a lot of others here in that the Placater Clean/Dirty or even the Archetype Clean/Lead are pretty versatile setups for such things, but I'll use the same cab for both. I'm not sure how many different styles your cover band covers, but for myself the rock sound from the 70's, 80's, and 90's are really quite different. I'm not sure I could do them justice with only a single setup. Especially not if your multi-decade also includes the 60's and 50's.

Oh I'm sure it will work and be consistent. It won't show up at the mixing board as -12db as it's a different signal than what's being sent via the USB, i.e. an analog Line level or Mic level signal depending on how you configure it in the global ins/outs. But once the sound man gain stages the signal where he wants it, it will be consistent and that's all that matters.

It all depends on how you're measuring the signal level. Measuring it in a DAW through the USB doesn't directly correspond with the output level from the 1/4" or XLR outputs that are controlled by the Helix master volume knob (if enabled) which determines the signals being sent to your on stage output device and to the mixing board. However, a consistent signal level is a consistent signal level which is all that ultimately matters.

All of my patches are stored on my hard drive. When I put a show together I simply load them up to a setlist or sometimes a series of setlists. My most recent version of any patch will always be on my hard drive and automatically backed up using WD Smartware.

I never use HX Edit for building my patches. I build all my patches (well over 200 of them at this point) solely from the interface. I do use HX Edit for it's archiving/saving/backing up capabilities. When I'm editing patches I have my Helix up on a bar to work with it so I'm not constantly bending down. I guess my reason for this is I don't want to be hampered if I need to do something at a live gig, so I'm much more familiar with working on the unit than I am through HX Edit.

That's generally not necessary. Most DI boxes have attenuation settings, but I've never had to use them and just send a Line level signal. I will often go with Mic level when sending direct from the XLR through a phantom power blocker to the board simply to be consistent with the other XLR inputs coming into the board.

In spite of what some people claim, there is no discernible difference in having your Helix master volume knob dimed. I would probably say most of us don't do such a thing. Lots of people cite arcane technical reasons, but I'd hazard to guess that in double blind test no one would ever be able to consistently pick out a Helix with it's big knob dimed and one that wasn't. The Helix isn't that complicated. Use your ears, dial in the tone that works for you and play guitar....and stop worrying about minutiae that has no bearing on your tone.

Smaller isn't necessarily the problem with the JBL's. As mentioned previously they're studio monitors, meaning they're meant to be listened to in a very specific equilateral triangular arrangement between the listener and the two speakers in order to get the full effect. They will most undoubtedly sound very high fidelity, and will probably sound okay as you move about the room, but the stereo effect will be limited. Since they're a bass reflex system you'd need to follow the manufacturers directions regarding placement. That's simply the nature of the design of those types of speakers. In terms of the other direction of using FRFR live performance style powered speakers, it's important to understand that you get what you pay for in terms of quality. Generally speaking, other than professional large scale concert type speakers, you have three basic levels in the market right now. The low end systems which have a good range of sound but very little flexibility in terms of adjusting for different placements and types of usage (spoken lectures, recorded music, live music, etc.), and fairly high volumes. The Alto's or Headrush would fall into this category. The next level of speakers would be more mid-range priced, have more flexibility for adjusting for usage, but a bit lower maximum volume, and use a bi-amp design for separating power to the speaker and the horn. The speakers that would fall into this category would be things like the QSC CP12, EV ZLX-12P, or Yamaha DBR12. The premium level speakers have all the general features of the mid-range speakers and with more clean and precise high volume performance, and more advanced electronics and DSP capabilities for contouring the speaker's response. The type of speakers falling into this category would be things like the QSC K12.2, Yamaha DXR12, or the Line 6 Stagesource speakers. As you mentioned, volume isn't a primary concern. That being the case I might suggest you look more into the mid-range speakers such as the ZLX series, DBR series, or CP series. You might also consider looking into the smaller versions of these speakers as bass response really isn't a major consideration when it comes to DSP driven powered speaker designs. I personally own a QSC CP8 that I use for smaller open mic nights or jam sessions which is very capable with a great sound and consistent peformance to my DXR12 which I use for professional gigs. I also own a ZLX-12P which I use for a side fill monitor, but have used it on occasion for my Helix.

Make sure you isolate the rattle to the speaker. I had a similar issue a few years back which I thought was coming from the DXR12. Turns out it was one of the pictures on the wall near the DXR12.

THAT is exactly the question here. If it's true that the big knob is a "digital master attenuator" not only would that run at odds with the design of every digital mixing board on the planet, but it would have to be an extraordinary deficiency designed into the system. I seriously doubt Line 6 would ever reveal if that's true since they've always been extraordinarily tight lipped about anything to do with their internal design, but it certainly doesn't act that way to my (and many others users) ears. Case in point being, I often turn down my Helix master volume knob when practicing from it's normal performance position at 11 o'clock to around 9 o'clock if it's late at night or very early in the morning out of consideration for my neighbors. The differences I hear are more consistent with the changes I'd expect in the Fletcher Munson effect, not loss of fidelity. In fact they're quite consistent with the changes I hear when I go through my QSC TM-30 mixing board and perform the same function. If you do the digital math on such operations I would expect it to start sounding more like an AM radio given the bit loss. I would also expect to note a significant loss in fidelity between my stage sound and my FOH sound given my Master Volume knob is disconnected from my XLR outs going to the mixer which would be a loss of more than HALF the bit depth since my stage sound is driven by the Master Volume knob at 11 o'clock, but that's clearly not the case.

There's no technical benefit to having the Helix master volume maxed out. The only thing that really matters is that you have some way of consistently setting your output level, which can be done any number of ways. Having the master volume knob at the 12 o'clock position with a higher channel volume producing -6 db on a signal meter is no different than having the master volume maxed out with a lower channel volume. It's still -6 db on a signal meter....which is ultimately all that matters. We're in the digital world where signal to noise ratios (SNR) and total harmonic distortion (THD) are negligible.

This is the defining factor I think. Some of us are more situational users, so depending on the situation, we'll use different capabilities offered by the Helix. There's another set of users that tend to play the same or similar type of stuff all the time, so those capabilities might not be as relevant. It's great that the design of the Helix doesn't lock you into only one way of working.

I think your impression is most likely flavored by HOW you use your Helix. For example, I'm a preset per song kind of user. That's because the genre and feel of one song can be quite different from the previous song or the next song. In some cases changes in the dynamics or sound of a song can be quite different within different areas of the song involving a number of changes in effects and amps, or even in the way the signal chain is being used. Snapshots are a HUGE advantage in achieving those things with a single foot switch.

The rules people tend to apply to modern bi-amped, DSP driven FRFR speakers about low end response aren't terribly relevant since they come from the days of non-powered, simple crossover designs. Most of these manufactures such as Yamaha or QSC provide frequency response graphs that apply across the board to their 12, 10 and 8 inch speakers. As far as going to studio monitors, in addition to my DXR12 I also have some Yamaha HS7 studio monitors. Both have amazing responses but there are some differences to think about simply based on what the speakers are designed to do. The DXR isn't terribly position dependent and can be adjusted via DSP contouring options. That's because you never really know how those speakers are going to be deployed and used ahead of time. Studio monitors are made for a different purpose which is for detailed listening such as in a studio, and are expected to be placed in a certain way relative to the listener to get the best response. Many of these studio monitors (including the HS series) depend on a rear bass reflex port so there are some specific guidelines for how they should be positioned to get the appropriate response. You probably should examine those requirements and see how that would work in your home setup. Another thing you might consider is simply downsizing to a different FRFR powered speaker that's smaller but with similar response to your DXR. I personally use a QSC CP8 for such things in situations where my DXR12 would be overwhelming and I've been quite astonished at how well it performs compared to the DXR in those cases.

Oh really?? It's amazing they included a master volume knob at all then since according to your "rules of engagement" we don't need it. Heck, at least they need to pass a law if we MUST do it. What do you think? A felony or a misdemeanor? I hate to tell you this, but when it comes to how people are actually using their Helix units, the rules you've established are clearly in minority from the responses I've seen over the last 3 1/2 years. In case you didn't know you can disconnect the signal you're sending to your soundman from the master volume knob in global ins/outs. I'm betting Line 6 included such a feature so it would allow users to adjust their master volume knob without it affecting their signal feed to the mixer. You'd think someone that's making rules for all the rest of us would know something like that....

It may be more effective to think about how things would be laid out in a studio session than on a pedalboard simply because it opens up a lot more possibilities than one would have on a typical pedalboard and amp situation. It's also more in line with how Line 6 has represented the Helix signal chains. Not everything on a pedalboard is necessarily optimal for getting the sounds one hears in many recordings. It's that way due more to technical limitations in many cases which aren't present in the studio environment. For example, in the studio reverbs and delays tend to be more toward the end of the signal chain, but in some cases something like a spring reverb may make certain amps have a more authentic flavor if it's placed between the amp and the cabinet(s). Certain things like phaser and chorus effects can produce significantly different tones when placed before or after the amp. And harmony effects can have atrocious tracking before an amp in many cases whereas they sound much tighter right after the cab or even between the amp and cab. I think you can lose a lot of creativity if you confine yourself purely to the physical world of pedalboards and amps.

I think it took me about one day of working on the Helix to discover any changes to a preset were all in disposable memory until I committed them by hitting the 'Save' button twice. Now I barely do anything without hitting the Save button twice. Sometimes even if I haven't changed a thing. It's like the old joke where the guy says to the doctor, "Doc, it's hurts when I do this" to which the doctor responds, "Well don't do that!!".

To be honest I'm not sure the style of music you play is a major influence on the type of FRFR speaker that will work well for it. It will either give you an accurate representation or it won't. The degree of accuracy can become more of an issue if you cover a wide range of styles which draws some of us to the more expensive options to get a better representation, for example, between a typical rock/blues sound and a finger picked acoustic sound or a clean electric hybrid picked sound. I would say that I wholeheartedly agree with rd2rk that getting the lower end FRFRs up off the ground on a pole will definitely deliver a much more honest representation given that most of the lower end units don't really have any DSP options for effectively overcoming bass coupling when used in a floor monitor arrangement for example. It also has the side benefit of better projection and coverage when not using a PA for the instruments.

Since I've never owned an Eleven rack nor Helix Native I can't offer any advice about tone matching there. What I can offer is patterns tend to be the most useful in using the Helix in most recording or live scenarios. As you'll probably notice in the Jason Sadites videos, he favors using mic combinations on cabinets, which is also consistent with every pro recording studio I've ever worked in. The most common combination tends to be a dynamic and ribbon mix such as an MD421 and a R121. This is most easily accomplished with the Helix stock cabs by using dual cabinets typically with the same cab but different mics and mic placements. What seems to drive many users toward using IRs is the versatility of mics and mic placements used in the various IR mixes which aren't available in the stock cabs. I know in my case one of the more useful IRs tends to be the Basketweave IRs from Ownhammer who have mixes that have as many as 6 different mic and mic placements in an IR. I still believe, as demonstrated by Jason Sadites, that you can achieve a good cabinet sound through the stock cabs, but stock cabs are certainly limited in placement options giving you only distance as an option but no way to simulate positions between cone and cap edge nor off-axis positioning in the way that IRs can.

He's not really comparing using different presets in the place of snapshots. He's referring to the capabilities within a preset that can be assigned to a stomp as compared to using a snapshot. For example, using multiple assignments on a stomp, or non-latching behaviors on a stomp. Many Helix users aren't aware of these capabilities as viable options since they've always had snapshots available to them. It's only us "old hands" around here that remember the days of the Helix before we even had snapshot capabilities.