DunedinDragon

Wow...remind me not to volunteer to be your sound man!!! ;)

My personal opinion with the Helix is that if you depend on the sound man to EQ your sound, you haven't done a very good job of dialing in your patch. That wouldn't necessarily be true in the older days of actual cabinets and mics. But in these days with full control over every single element that can affect your tone right there in your hands you should be able to provide a sound that needs no channel EQ adjustments. If there's a problem with the acoustics of the room that needs adjustment, that affects ALL of the channels equally, not just the guitar on one single channel. So that needs to be addressed through the overall final graphic or parametric EQ on the board...not channel by channel. For the last few years with both the HD500X and the Helix I've told all of the sound men to leave my channel flat, and it's worked out beautifully.

A lot really depends on what you're using to listen to your patches. This is one of the many reasons so many of us opt to use FRFR speakers. By and large the sound you get at home using a FRFR powered speaker setup (given you're playing at something close to gig volume) will pretty much sound the same as most PAs. That's because powered FRFR speakers like Altos, Yamaha, QSC, Line 6 ARE in fact PA speakers and are very often used that way in small venues. To be fair there are venues that are still using older non-powered speakers which will lose a lot of the mids and quite a bit of definition. But if the place you're playing is using any type of more modern system with powered speakers you'll be very close if you dialed in your tone with a powered FRFR speaker setup. And I don't know where you're getting the idea that something dialed in at home won't sound the same with the band. I dial in all my presets at home and rarely have to make any adjustments once I get with the band. If I do make adjustments it's typically related to the overall volume of the patch.

Well part of what you say is true in that guitar speakers have a more limited range, kind of. For example, if you look at the specs for a Celestion Creamback it lists the response range as 75 - 5000hz. That more the working range than the technical range if you look at the frequency response graph for the speaker which really goes across the whole spectrum but responds differently throughout the range.

I have it from a solid source that the new Firmware update will contain the one feature that will turn you into the "Guitar God" you've always wanted to be...so it's probably worth waiting for... ;)

I'm no convolution expert and I'd love to have someone who is more versed in this technology give us an idea of what actually causes this effect. But to clarify a bit, as others have noted a speaker cabinet IR is more than just the sound of the cabinet. It's also the effect of the room, the physics of the environment and the effect of the amp producing the signal that plays into what eventually becomes the IR. In researching this situation I've come to understand also that one thing that gets in the way of a perfect representation of an IR is the differences in the impedance load between the signal chain that produced the IR and the one that's reproducing it. This is likely the thing that comes into play particularly when using something like a higher end FRFR type amplified speaker which is specifically designed to try and maintain a flat load across the full sonic range. I've also come to understand that some of this may also be just in the way we hear things. For example, depending on my position relative to my Yamaha DXR12, there can be a significant difference in what I'm hearing due to the design of the speaker. Powered speakers like this are designed primarily for projecting sound across long distances as compared to traditional cabinets. Therefore the closer I stand to the speaker the more I hear of he compression driver rather than the full mix of the compression driver and the main speaker. Standing further away and off axis typically changes this response considerably. That makes sense when you look at how these speakers are meant to be applied as PA speakers. No one puts PA speakers right in the face of the audience because they would be too harsh. When you think of how people use these speakers by placing them right at their feet facing directly into their face it's likely to sound much harsher then it would a few feet away. This is one of the reasons that in live performance I place my DXR12 a few feet behind me as I would a traditional amp cabinet which results in a much more realistic representation with less harshness. These are my conclusions based on my experiences but I'd love to hear a more technical explanation if there is one.

Are there any plans to get the Android app updated to work with non-antiquated versions of Android?? Given that it's still being marketed on the Line 6 Products page with this capability it might be prudent to get it to actually work.

I can't say there are any specific guidelines I follow in terms of reverb other than to match it to the sound I'm going for which depends on the song and the style of song. A lot of styles like rockabilly, old country or early 60's pop songs will require more of it than rock or blues for example. Although there are also very ambient rich rock styles like Pink Floyd where it's more of a consideration. Generally most modern styles only require just a slight touch as defined in other responses, if any at all, just to help with getting a better mix with the other instruments.

I'm not sure that is functionally much different than how current snapshots work other than if that scheme were implemented as two completely separate presets it would limit the sharing of resources between the two signal chains as is currently allowed in snapshots. Ultimately the presets would still have to be loaded simultaneously at the beginning when selected in the same way as snapshots in order to avoid the overhead of the loading process which is what causes the audio gap in the first place.

That's actually why I find no particular reason to use snapshots. I've never felt limited by just using pre 2.0 assignments and such, even using multiple amps in some cases. As far as things like the Aux In, or other inputs...everything adds to the price point, and if you're trying to hit a certain price point, certain things have to go.

I believe that's why, in the original post you responded to I specifically said "higher end modelers". I'm sure you could get by with little to no gap on a Spider V amp as well, but that's hardly an apples to apples comparison. It would also come with less specific detail in the modeling of the individual blocks as well....such as a comparison between the HD500 modeling and the Helix modeling.

The truth is, you're going to find similar limits on any of the higher end modeling units based on the fact that there's a functional limit to how much processing can be done per DSP chip. When you change presets you start over from scratch in that everything needs to get re-connected from the ground up which causes the gap. A snapshot is nothing more than just a single preset with some additional functionality to help decide what components within that preset are active and what their settings are at any given point in time. In the end it's still a single preset with the same computational limits of how much processing the DSP can handle. What saves most people is that you have two dedicated DSP chips each one allocated to a separate signal path. This allows you to allocate the blocks specifically to each of the DSP chips. By using some creative routing it's relatively easy to share effects between different amps and just change their settings of those effects within the snapshot. It's also necessary to become familiar with the general amount of DSP used by different blocks. As I said you'll find the same limitations on all of the higher end modeling devices because by their very definition they use more computational capability to achieve the level of modeling being done. The practical limit on all of these type of modelers tends to be 2 DSP chips before you begin to run into latency issues. The problem is, in this world nothing comes for free. And the higher quality you want comes with a practical computational limit. It can be worked around as many people have done, but you have to put in the energy and time to make it work for you.

Yep...exactly the same for me.

Personally I'm prone to using the amp's gain in most cases and distortion pedals for leads. It makes the gain feel more natural and gives better definition without adding much volume for leads. However I have found the HiWatt combined with the Teemah has a very natural mid gain sound with the HiWatt gain at about 3 and the Teemah gain around 5.4. I then layer a TubeScreamer on top of that for a nice difference in tone for leads. That's a perfect recipe for a classic Joe Walsh sound by the way. A little less gain on the Teemah combined with some moderate compression up front in the signal chain works pretty well for Rolling Stones, The Who, and even early Doobie Brothers.

Here's a great tutorial from Redwirez regarding how to best dial in tones with an IR. Obviously it's written with a slant toward their IR's, but a lot of what they're talking about applies generally to mic types and mic placements on IRs and should be generally useful in selecting which configuration of IRs might best achieve the sound you want. http://redwirez.com/ir/DialingInYourTone.pdf

+100 The American Elite compound profile neck is one of the finest and most comfortable necks I've ever worked with. Here's some info on it: http://www2.fender.com/experience/tech-talk/tech-talk-american-elite-compound-profile-neck/

It's always a crap shoot when it comes to using an unfamiliar PA and soundman. Personally I always use a modern FRFR speaker (Yamaha DXR12) when I'm dialing in my presets at home. That's how I know what it's supposed to sound like, and will sound like on a capable and modern PA system with little dependency on the sound man to correct it. However, there are quite a few older passive systems out there that were designed to deliver volume rather than accuracy like modern systems do. Fortunately there's only one situation that I commonly play that falls into that category. In that case we simply depend on our on-stage equipment for the instruments and use the PA and PA monitors for vocals only. This is the reason I use my DXR12 as a direct monitor from my Helix in live performances. If you've designed your presets to be used with a PA you probably don't want to be pushing those presets through a traditional amp since presets built to work with a traditional amp tend to be designed very differently. You might be okay if you're sure of the PA system you'll be working with. It's probably worth doing some research on it prior to showing up. But that of course means you need to know a thing or two about different PA configurations and how they work.

ON THE SUBJECT OF PC-BASED MODELING IN LIVE SITUATIONS: I thought it might be appropriate for me to clarify where I stand on such things given the turn of this discussion. First, I want to clarify that I agree this is the natural progression of this technology over time. The market has proven time and time again that open, integrable systems create robust markets that everyone (consumers and vendors) benefit from. My concerns in this regard are more about the state of the industry and the state of the technology in getting us there. I realize there are many people at this point that are willing to do make the leap of using laptops in live performances in order to incorporate off-the-shelf capabilities found in the DAW world into live performances. I wish them luck with their endeavors, but I'm just not willing to make that leap yet for a number of reasons. I built a successful career as a systems programmer from which I retired in 2001. But I was fortunate to be in the business during the development and growth of the current crop of operating systems we see most used prominently today such as Windows, the Apple OS, various versions of Unix, and Linux. In addition to those I worked fairly extensively in real-time operating systems for military applications such as HP's RTE. I point this out because this is really the core difference between Helix floor units and Laptop applications such as Native. The difference between a commercial grade, general purpose OS and a real-time, dedicated OS comes down to whether you want flexibility or dependability. As amazing and capable as laptops are you have to look no further than the services listing in task manager to understand all of the capabilities and utilities being actively supported in an OS such as Windows, all of which steal processing cycles and have interdependencies on other component services to be well behaved in order for the system to remain stable. These are the things that make a PC integrate effectively and safely with the world of things built by outside vendors, but are also the core reasons for latency and instability. Having a real-time, dedicated OS such as that in the Helix reduces the potential for such problems at the cost of a lack of flexibility to incorporate outside components. I do think the industry will eventually lead to a happy medium between these two being developed eventually, but we're not there yet and we probably a few years away from it right now. In the meantime, were I Line 6 I would be hesitant to support Native being used in a live situation simply due to the fact that they and their product would be blamed for any failure in terms of stability of latency when in fact it very easily could have nothing to do with them but rather with the specific configuration of the service components of 3rd party vendors on that specific laptop. Think about it. How many times has Line 6 taken the heat and the blame for FW updates failing or Editor problems which were not due to their software, but rather to the specific configuration and operation of the PC being used to update it? Were you to use Native in a live performance and it resulted in a system error that required a full reboot of the system, or that suddenly your latency went crazy, who are you going to blame? The likelihood is not that Native was doing something wrong, but that some well-behaved request for services on their part such as memory allocation or access to load a component caused a memory fault because someone in the chain of events didn't follow all the specifications for being a well-behaved application or service...in other words nothing that Line 6 could ever have accounted for or have any control over. This is the world of a general purpose operating system. If people want to take those kind of chances for using Native live that's up to them. But they should be responsible enough to assume the repercussions of that decision and not blame the vendor who never designed that application to be used that way.

This is kind of like asking, "The Ford cars in NASCAR are based on the Ford Fusion, could I take my Ford Fusion I just bought from my local dealership and run it in the Daytona 500?" Well....yes you could. However, you're talking about a version of Helix designed for the recording studio doing the job of one that's specifically designed for live use. The same as a street Ford Fusion is designed for running to the grocery store and not for a 500 mile race. To me the ultimate question is always about "what is it designed to do?" If you want to press it beyond what it was designed for and use it in areas it wasn't designed to do, that's your choice, and your consequences.

I have a 2017 Strat Elite which is an excellent state of the art Strat. Although the 4th generation noiseless pickups are great as is the S-1 switching, I'm not sure those are as much a factor in a great tone as is the right amp, cab/IR, and mic setup along with appropriate EQ. Although for live purposes the 4th gen noiseless pickups do have considerably more punch and character than any of my previous Strats.

Are you sure you don't have a DSP setting on the speaker that takes care of that? On the Yamaha and QSC (as well as several others) speakers it's just a preset that changes the low end frequency response curve to account for the bass coupling.

One of the reasons I settled on placing my dedicated speaker behind me rather than in front of me was that it felt more natural in a stage mix with the band. There's at least one gig per month in which we don't rely on putting the instruments through the PA and simply rely on our backline to fill the space. I've never had any problems with my guitar not being heard even with my speaker behind me in the monitor position. I've never done it, but I'm relatively certain you might have a problem being heard by the audience if you just have your guitar going through a normal monitor in front of you while everyone else is using a normal backline setup to fill the house sound.

I can't say I've ever run into that kind of limit. I do use the parametric EQ on occasion but it's typically just a minor tweak to surgically address some irritating isolated frequency. But then I'm not very prone to doing a lot of EQ. I try to manage the tone as best I can using the amp's EQ and then the cab/IR and mic positions. Beyond that it's just very minor EQ tweaks if at all.

I use a dedicated Yamaha DXR12 as a guitar speaker/monitor. I position it on the floor in a wedge position behind me the same as I would an amp in the backline. I have the DSP setting on my dedicated DXR12 set to "Monitor" in order to alleviate the bass coupling effect of a floor monitor. We're a 7 piece band and typically send all our instruments through the FOH PA, but whether we go through the PA with our instruments or not, I keep the same dedicated monitor on my backline as it's easier for all of use to maintain our stage mix with a normal backline. We're a 7 piece band but normally all we route through our stage monitors (which are also DXR12's) are just the voices and the harmonica. I also occasionally play keyboard which is also routed through my Helix to my dedicated monitor as well as XLR to the mixing board. We also occasionally use electronic drums which are routed to a Mackie Thump12 monitor on the backline as well as DI to the mixing board. In that case the drum monitor is mounted on a pole along the backline to get better coverage on stage for everyone.

I suspect this is exactly why it's been held up. That and the popular outcry from the user community is always about more amps and effects, and it's not clear to me how much of the user base this would be useful to, so it's questionable how much of a priority it is in the feature list based on demand. As to why people would downvote it...that's just a mystery. If it's not something you would use it doesn't impact you for it to be there.