DunedinDragon

I understand completely. The Helix is versatile enough that you can take a number of different approaches to it, each with it's own individual set of things to consider. I think that's why I chose to simply go direct into a full range setup. It also helped that I had already had a significant amount of time working both as a live sound man and in studios so I had a base of knowledge to build from. You may be more comfortable with a more intermediate step, but as long as you ask clear questions and give specifics about how you have things setup and configured, you'll get very good advice here from people that have done similar things. The biggest impediment is when people come on here and say "my tone sounds like crap", but don't provide any specific information about the specifics of their setup and how they have things laid out in their signal chain. Without that specific information it's very hard to provide help sometimes. What I would suggest is start simple and work up. Get the basics working which is just an amp, cabinet, mic (or IR) and maybe a simple effect or two. There's a lot to learn just in getting a simple thing working right for you. Then build off of that. I wouldn't get too involved in things like talk boxes and such until you're confident with your ability to put together a decent tone and understand what you're doing with that. I HIGHLY recommend going through Jason Sadite's videos on building great tone on YouTube. There's a lot of really great info there.

Some people simply don't connect with modelers and you may be one of them. They are a different breed of equipment and far closer to a studio setup than a live stage setup. Not that it can't be used live, but the manner in which you approach is will be closer to the considerations you apply in a studio environment. If that's not natural to you, it may not be something you want to spend the time getting familiar with and just go the path of least resistance. I would say also, however, in my experience it's been worth it. Not so much to achieve the sounds I had in my old setup, but to achieve sounds well beyond what I've had in the past. But that's because I made a total commitment to it, sold off old equipment (amps and effects) and went to a configuration of equipment that allowed me to take full advantage of the modeling environment. I currently own no amps, no effects, but I do have 5 Yamaha DXR12's and 1 EV ZLX-12P, as well as two Yamaha HS-7 studio monitors as well as some higher end studio level headsets. Is it that same sound as I had with my previous traditional amps? No...but it is the same sound my audience has been hearing when I played through my traditional amps once they had been mic'd and sent to the FOH. And it was that understanding that moved me into approaching modeling in a new and more effective way and getting way more out of it than I'd ever gotten from my traditional setups. I came to terms with the fact that for 4 decades the sound I was so pleased with on stage was NOT the sound my audience had been hearing, so I concentrated on building patches that have more in common with studio setups than live setups. In my traditional live setups I had never used compressors, or Parametric EQs, or layered/sidechained effects and sidechained EQs, or multiple mic and multiple cabinet arrangements. In my modeling setup they are pretty much constants in every patch I play through. What I have now is studio quality sound in my live performances which is the same for me on stage as it is for my audience, which was unachievable with my traditional equipment. This new paradigm is something worth considering I think if you're willing to take that step.

The first thing that jumps out at me is you haven't mentioned anything about how you configured the PRX815 which could potentially be playing into this. Have you connected up the DSP utility program to see what the settings are on the speaker? It's possible some configuration in areas such as the parametric EQ that could be causing clipping in conjunction with the inbound signal from the Helix. Something worth checking at least. Another thing to check is where you have your Helix master volume knob set. Most of us probably keep our Helix master around 12 o'clock, but if you followed the terrible advice in the manual and set it at full volume that could be a source of clipping issues depending on the configuration of the PRX815. My advice at this point is to start very basic and simple and see where that gets you. Start a new preset with a well known clean amp that's hard to make distort such as the Jazz Rivet 120 and just use the stock amp+cab with global EQ off and your input set to auto and your Helix volume set to 12 o'clock and your PRX815 volume set to 12 o'clock and see if you're getting a perfectly clean sound. If you have a clean DSP configuration on the PRX815 and are still getting distortion I'd next try and use another guitar with a standard level of output if you have it just to determine if it's coming from the guitar or from the Helix. If you can dismiss both the speaker output and the guitar input then it's a pretty good chance there's something messed up in the Helix itself. That type of configuration should be perfectly clean.

As far as being "frozen" in a patch, I'm wondering if you aren't inadvertently throwing the Helix in a different mode and all you need to do to fix it is hit the mode button once.

From what you've described I'm not sure the problem you're hearing is related to the audio interface as much as it is the configuration of the buffer size in the ASIO driver in your DAW. That's what I normally associate clicks and pops with. Before spending money on a new interface it may be worth seeing if lowering the buffer size might fix it given you have enough computer processing power to spare.

It depends on the board. Many have a preamp that covers the range between mic and line levels, but some have a switch, and I've seen some that pretty much confine line level to 1/4" and mic level to XLR. But most relatively modern ones have a wide range preamp controlled by gain/trim knob. That hasn't been the predominant case on powered speakers from what I've seen. Most are controlled either by a switch or by dedicated inputs. What seems to be coming on as a "standard" for the most part is a Microphone input that's always mic level and a line input that's switchable in some way between line and mic.

Also, if this is your first step into a premium level modeler be prepared to do some learning. As easy as the Helix is to use, there's a TON of technical things to learn and become familiar with. It's certainly no "plug-in-and-play" kind of thing. You're dealing with the intricacies of building a signal chain and it has much more in common with working in a well-equipped studio than it does a typical pedal board. Not only is it effects routing and placement, but amp, cabinet, and mic placement modeling...all of which significantly affect the resulting sound. I would highly recommend you watch Jason Sadites great series on building great tone with the Helix on YouTube as he has a LOT of good information you can benefit from. Jason Sadites Building Great Tone Part 3 - Using a Split Block

The reason you see the Alto speakers referenced so often is because they are the lowest cost unit. That may be fine for you, but in the world of powered speakers you really do get what you pay for. For example if you place the speakers on the floor as a wedge you will typically produce a bass biasing effect which will over-emphasize bass frequencies. You could always correct for that in your patch, but that would likely mean your direct FOH signal wouldn't have enough bass. Most of the higher priced powered speakers such as the Electrovoice ZLX-12P and many others in that category come with a broader range of DSP tuning options in the speaker that allow you to correct for such things, but the speakers run about $400 and up. Most commonly people run 1/4" out to the speakers and XLR out to the FOH. There's no particular reason to run XLR on stage as the runs aren't that long so you really don't gain much. I personally use a Yamaha DXR12 as my on stage speaker and I connect to it via 1/4" L/Mono out and XLR L/Mono out to the FOH with a phantom power blocker to protect from phantom power coming into the Helix. In my global settings I have the 1/4" out set to LINE level and my XLR out set to MIC level and set my Helix volume control to only effect the 1/4" out. That will then send a full volume MIC level signal to the board which will be very consistent with all of the other mic input signals, and any volume adjustments I make on stage won't affect my signal going to the board. As far as running stereo that's more or less a personal choice. The consensus seems to be that your audience is highly unlikely to benefit from it, but it would make a difference for what you hear on stage. As a sound man I tend to agree with that as most rooms you play aren't very optimal for maintaining stereo separation and it all gets muddled together out front very quickly. Additionally only a very tiny fraction of your audience are in the dead center of the room where they would be in a position to even hear a balanced separation at all. Not to mention not all soundmen are open to allocating two channels to a guitar signal particularly if they're short on channels or aren't familiar or comfortable with sub-grouping. It just doesn't deliver typically as you might expect. That's why I just stick with mono as it doesn't really matter to so much to the audience.

I personally own two Helix floor units. One that I use for live performances and one that is hooked up to my DAW and serves as a backup to my live unit. They were purchased roughly a year apart and I can tell no difference when I switch from one to the other.

Exactly...instrument level is what goes into the normal guitar input of an amp.

I was actually referring to the preamps used in mixing boards for live use and in powered speakers. There are certainly much more high end pre's used in recording situations, and in some cases on high end boards, but the ones on mixing boards and powered speakers are relatively generic in nature for the most part. Quite frankly I honestly can't tell the difference in the mic pre's on an Allen & Heath from those in a Mackie board, and I'm not sure the audience can either. But it does make for some good marketing. Of course all of that stuff matters significantly to folks like Universal Audio which have built their company on high end, specialized equipment. But I'm just not that much of a cork sniffer I guess when it comes to that sort of minutia.

It's interesting in my case how rarely I end up using snapshots. Not that they aren't a great tool, but in my particular case of using different presets for each song it's not terribly common that I really need them that often. Clearly they're useful when there are some significant changes in dynamics and tone in a given song. I just don't seem to run into those types of situations all that often. The weakness I see in using them in a wider setting is the complexity involved in getting everything dialed in with precision. What I mean by this is more based on the way I build patches in that each and every patch (i.e. song) is going to have different settings. For example, although I might use the LA Compressor across multiple patches, the settings are going to vary with each and every use in the same manner as would be the case were I to be recording the songs in a studio. The same with amps, or parametric EQ, or distortion pedals, or anything else really. And of course I use a LOT of different amps for different types of songs not to mention IRs and other various effects. Whatever best fits the song, style, and specific sound I want. THAT to me is why I own the Helix...for the precision and distinctive tone for each song. I guess I'm not a big fan of using snapshots more extensively because it seems like I would end up being much more generalized and maybe less specific just to preserve DSP...which in my case I never have to concern myself with.

While interesting, it's not too useful given the fact that with a snapshot you can just change the amp from a Cartographer amp model to a Vox amp model like most people do in a snapshot or a simple pedal assignment. That would take about 20 seconds to describe.

The only real difference between a mic preamp and a non mic preamp is the signal level it's prepared to work with. It certainly doesn't color the sound or else pretty much every vocal you've ever heard live or on a recording would not sound like the actually vocalist's voice since they all work with MIC level inputs rather than LINE level inputs. MIC level is a significantly lower voltage level signal than is a LINE level and it's primary value has always been less loss and interference over very long cable runs. Obviously that's not an issue with an on-stage monitor connected to a 1/4" output on a Helix. Interestingly enough, LINE level signals are what's normally used as the output levels coming off of a mixing board, so the fact the FRFR-112 uses only LINE level signals doesn't preclude it from being used as a normal stage monitor or even speaker from a mixing board. As far as bypassing a mic pre, that may be an design decision more than an industry standard although I will say that most higher end powered speakers have a switch or toggle that determines the level being used on a given input which suggests a different circuit at least. In my experience, Alto's use of the gain knob position to determine levels is unique and may be that way due to cost saving considerations. Bottom line the main thing you lose with the FRFR-112 is the ability to directly plug a microphone into the speaker. So if you're planning on loaning it to someone that wants to use it on a pole to publically address an audience at a meeting, don't do it because it won't work without some additional equipment.

Personally I attribute what many folks are identifying as "fizz" is really an artifact related to an over abundance of gain/distortion. I can replicate what I think they're hearing by combining too much drive on both the amp and a distortion style pedal. Turning down one or the other seems to fix it. I don't run into it because I'm fairly conservative with such things. Just enough drive to give it some beef and no more. But I've heard what I think they are describing as well at clubs with bands using traditional amps and pedals with lots of drive/distortion. I've always assumed it has something to do with the way the limiter is set up on the speaker responding to an overdriven highly chopped signal. This would be consistent with what jonandtice was describing. In his case the monitor speakers were hitting the internal limiter while the FOH speakers weren't. In the case where I've heard it in clubs was the opposite, where the FOH speakers were engaging the internal limiter while the band was only hearing their amps. All just guesswork on my part I admit.

I'm not sure anyone was complaining about the IR's being too complex. I think people complained about it being time consuming to find the IR's that will best serve your purposes, and the simplistic approach to IR management. I certainly never thought of them as more complex. In fact to me it's just the opposite. Once I have them in my IR list it's MUCH more simple to dial through and find the best fit for the sound I'm after. Certainly much less complex than it would be to monkey around with stock cabs and settings. To be honest in all my time building presets on the HD500X and the Helix I've never had a problem with "fizz", in spite of having built quite a few heavy rock style presets. As is the case with most things in the modeling world I think this is one of those things that is more of a user lack of understanding problem than a problem with the modeling itself. The reason being that the frame of reference a lot of new modeling users have is plugging into an actual amp and stomps and dialing things in. What you hear in the room while doing that is very different from what you hear when recording it in a studio or in a live setting through a PA. If you have those type of experiences as a frame of reference it becomes much easier to avoid the doing the things that create "fizz" as well as any number of other unpleasant artifacts.

I'm guessing we may be seeing more and more of this type of powered flat response cabinet over the next couple of years as the popularity modeling systems grows. Looks fairly similar to the Line6 Powercab setup but probably with different options and such. Not a whole lot of info here to base anything about it's performance characteristics on, but I would guess the internals are likely to be similar to the Powercab.

Yeah...I guess that's the point I was wondering about. That's the reason so many of us go the FRFR route. Once you place the mic in front of the amp, that wonderful sound you're hearing on stage gets changed out front. It doesn't mean it's bad, but definitely different due to the coloration of the mic and it's placement relative to the speaker. We choose FRFR setups specifically to control that sort of thing.

But what do you do when you need to send the amp through the PA?

Right. The recent trend by many people has been to use stock cabinets saying you can get the same tones as you can from IRs. I think you can get useable tones with stock cabinets but it takes some work. But it's not going to compare with some of the more complex mic mixes you can get in IRs simply due to the fact you'll quickly run out of DSP trying to combine cabinets and mics. If you understand the differences in microphones, and understand the effects of on/off axis and various placement/mixing strategies, and are willing to read the directions that come with most IR bundles, it's not really a black hole as some suggest. For example I recently purchased the 4x12 Basketweave cabinet collection from Ownhammer which incorporates what they refer to as the MMMC or Massively Multi-Microphone Collection in which some of the mixes use up to 5 microphones in different positions to get a very realistic and useable tone. Out of that collection I was able to quickly (within about 4 or 5 hours) identify the 7 or 8 IRs that would suit my purposes without auditioning the entire collection because the descriptions were enough to get me close and find the ones I wanted. Given how quickly I can drop in an IR as compared to tinkering with a stock cabinet it's no contest as far as which will be more efficient and practical for my use. I would love to see Line 6 do some retooling on how their stock cabinets work, but I suspect it would be a significant undertaking to do it in such a way that you could accomplish the level of sophistication offered by IR vendors in a DSP efficient manner using just a single cabinet with multiple mics, so I'll continue to go the route of least resistance and the route of less tinkering until they do.

Actually, no. This has been happening to you on every traditional amp you've ever played. The difference is, you dial in the amp and it sounds great to you. You stick a SM-57 on it near the center of the speaker and this is what your audience has been hearing. Why do you think recording studios go to such complex measures in mic'ing guitar cabinets for recording?

Again, no one can help you unless you give us a LOT more detail about your setup. We can guess from now until Christmas but nothing will help unless you tell us specifically about how things are hooked up and what your settings are.

Looks like a really good way to add about 2 hours setup/breakdown time to a 4 hour gig....

One of the more common problems I see for many users is the overuse of gain and overdrive. Once you step into the world of full range speakers where there's a lot more clarity and less mushiness, too much gain and overdrive can have some annoying artifacts. So a little restraint can get you what you want without the trashiness. If you're dealing with high gain sounds the amp BIAS can also be a great friend for either increasing articulation by turning it up, or increasing warmth and blending by turning it down. That applies as well to clean and mid-gain tones as well. When you say clear, tight sounds I immediately associate that with compressors, and mostly with the LA Studio Comp when placed at the end of a signal chain. This is a very common technique used in studio recording, but applies just as well to live performances with Helix using a full range powered speaker. It takes a little bit to understand how to best use the parameters. The default parameters are reasonably close other than gain which I normally need to bring down to between 4.5 to 6. The idea being to have roughly the same overall volume as you have when it's off. But when it's turned on everything is much more full. Peak Reduction is the key parameter for how much compression you want. The default value is pretty good although I will often go higher or lower depending on the sound I'm after. The Mix parameter is particularly useful in determining how much compression you want mixed with the uncompressed signal and that's one that I vary quite a bit depending on the type of sound I want in the patch. Bear in mind that higher gain tones are already pretty compressed so it won't be as useful on those types of sounds. Everyone is a bit different in how they approach building their patches, but for me I'm much more effective if I get most of my overall tone dialed in with the amp, cabinet, mic choice(s) and mic placement(s). This is where starting with the best amp for the sound you want is crucial. You can spend a ton of time tinkering to get the tone you want when a simple change in the amp, or cab, or mic choice and placement would fix it. If you get those things correct, everything else tends to fall into place after that. If you haven't yet viewed Jason Sadites video on building great tone, you really need to familiarize yourself with his techniques. They're very simple, straightforward and very effective at getting the tones you're after. This one is particularly good.

Well the safest choice would be the PA speakers since they're more than capable of reaching whatever volume you'd want, and could certainly be further used with a band or just to jam with other musicians.