DunedinDragon

The one problem you might have with you approach is that the recommendation from Line 6 for equalizing volume on the amp models has always been to use the amp's channel volume, not the master volume as the the channel volume is guaranteed not to affect the tone of the amp. Master volume on the other hand can and does affect the overall tone in terms of overdrive characteristics. To give you an example, the WhoWatt is a very nice clean amp with the master volume at 2 but begins to get a bit of a crunch to it at a master volume of 5. In theory I like your approach, but it could get in the way of selecting the right amp for a given preset as a master volume of 5 may not give you the best indication of how a specific amp will sound. Again looking at the WhoWatt that particular amp was designed specifically for being very good at working with external overdrive pedals for adding crunch and overdrive while retaining it's underlying tone and that underlying tone changes significantly when you adjust the master volume.

It's really not something Line 6 is in charge of. HX Edit (like most installable programs) depends on a lot of underlying runtime components in the operating system that must be installed and installing those components is a matter of having administrator rights to the computer on both a PC or a Mac. It's all part and parcel of the security system of the computer which you can't bypass...for very good reasons.

What you want is to cram everything into one preset as if you're dealing with a traditional pedalboard. But you have 1,024 presets all perfectly capable of adapting to different, unique configurations but you only have 8 snapshots and a limited amount of DSP available in that snapshot. Maybe taking a look at the Helix as a different paradigm from the traditional pedalboard might open up a wider range of possibilities.

Or maybe just a switching setup for speakers. Honestly I think you're overanalyzing this. What you're wanting is the best in all categories, but it doesn't really exist because you have two very diverse categories. One is a consumer category and one is a professional category. Consumers could care less about the tedious intricacies of dialing in a modeling environment, they just want a sweet sounding setup to play their recordings. But a professional endeavor demands control over those kind of intricacies if you want the best in accuracy. I don't think this is as much of a "which product will fit the situation" as it is "what do I want to accomplish" for each of the divergent activities. If I were planning on dialing in my Helix for use in a live gig or a flawless recording, the last thing I would want to work with would be a consumer stereo product. However, I'm perfectly happy with my studio speakers being the main way I listen to music on the Internet when I'm at my desk. In my bedroom I'm happy with my Bose bluetooth speaker, for listening to tv, music or audio books. But I only answer to my needs.

High end HI-FI speakers are generally "sweetened" a bit for more casual listening so in that regard they might not be as accurate in their representation of modeled cabinets within the Helix. However, it may not matter as much in a casual listening environment if you're just casually playing the Helix in that environment. Larger studio monitors might give you more of what you want in a casual listening environment, but you're exactly right that listener positioning as well as speaker positioning is much more critical with most higher end studio speakers. In professional studio environments they typically have both near field and far field monitors used together in the mixing board area to help fill the space as that's where they often listen to the mix with the musicians to evaluate a mix which might also be an option for you. The thing is that consumer HIFI is meant for general consumers which is an entirely different audience than professional studio gear so a lot of latitude is taken to make it more appropriate in a situation where people aren't in a generally fixed location or in a non prime listening position so it might not work for both in your application if, for example, you want to use that same system for critically evaluating a preset you're trying to dial in. Also, it's not that a live PA speaker would "bring the house down", more that it's designed specifically to fill a large space evenly both horizontally and over a longer distance. If you've ever been to a concert or a club and stood near a PA speaker you've already experienced how bright and harsh it can sound. That's because PA speakers are designed to have a consistent sound across a very wide horizontal radius (sometimes as much as 180 degrees) in order for the sound to be consistent for listeners at the far left or far right. They are also designed to limit the vertical sound radius so they conserve sound energy from being lost in the ceiling or floor to reduce sound pressure loss across longer distances. All of that combined with the way the high and low frequencies are handled between the main speaker and the high frequency driver can make them quire harsh up close. In my application in my rehearsal space I generally give my DXR12 about 6 feet of clearance in order to get an accurate feel for how it will sound in live applications. The bottom line here is that there are always engineering tradeoffs in order to be "best" at doing something. That's why you can't buy a car that has both the top of the line performance as well as the most economical mileage. As a consumer you have to make a choice.

I think what Datacommando was referring to was probably powered studio monitors such as Yamaha HS7 speakers which would work fine on both the stereo as well as with the Helix as they're made for studio use. DXR10 and similar speakers are made for live PA use and would probably be overwhelming in the setting you're talking about. The Powercab, studio monitors or live PA speakers are all viable and accurate for model and cabinet simulations. It's just that they're all made specifically for different usage situations. Powercab is ideal for something like a rehearsal space or as a live traditional style stage amp, but is a full range flat response (FRFR) system that can provide the accuracy needed in modeling. Studio monitors like the HS7 or similar are again FRFR style speakers but meant for critical listening in controlled environment. Live performance powered speakers like the DXR10 are designed specifically to fill large spaces evenly and in close up applications would sound pretty harsh. I actually have both. I have HS7's in my studio for both recording purposes as well as general music listening on YouTube and such. I also have a DXR12 but that's in a rehearsal space with my mixing board where I prepare and dial in my presets for live performances. Hopefully that puts things in perspective for you.

Here's a PDF document that came from Redwirez you might find useful: DialingInYourTone.pdf

If you want Line 6 to consider this you need to post it to Ideascale since no one from Line 6 monitors or typically responds to this forum. There it can be viewed and voted on by other users and considered for inclusion in future product updates. You might want to specify these things as a global option as changing them permanently would likely break a TON of things people have already integrated with via MIDI and there are a lot of things in the MIDI and DMX worlds that work with zero based numerical systems more than one's based numerical systems.

I'll have to get back to you on that as I've got everything packed up right now for a rehearsal, but I'll check when I get it set up.

I've had my K10.2 for about a year and a half and use it every single week in live performances and have yet to have this failure. In researching this fault it's clear that the fault is typically caused by a direct signal that's consistently too hot so the fault is triggered until a test is run on the speaker through the speaker's built in interface to determine if there's actual damage and to clear the fault. I'm relatively certain that the reason I've never seen this fault is because I meticulously gain stage my signal levels on all my presets and snapshots coming out of the Helix through my QSC TM-30 mixer and that mixer is where my K10.2 is connected. The Helix volume knob works as an attenuator for adjusting the analog output signal from the Helix after it's been converted from the digital signal chain used internally in the Helix. I've occasionally gone direct to the K10.2 from my Helix with no problems, but again I'm constantly checking both my internal digital signal levels using the internal Helix signal meter that gets displayed when you select the output block, as well as the analogy signal level coming out of the Helix. In my case I set my Helix XLR outputs to Mic signal level because it's going to a mixing board, but you can use Line level as long as you specify that adjustment on the K10.2 menu. Mic signal level is considerably lower than a Line signal level. By selecting a Line signal level on the K10.2 it makes the appropriate adjustments on the speaker for the hotter incoming signal. You control the signal level coming out of the Helix !/4" and XLR outputs via the global ins/out settings. You should carefully read the instructions in your user manual on pages 10 - 12 to better understand these settings. You should be setting these signals in such a way that you get no yellow signal lights when playing. All that being said, there's one area other than a mismatch between signal level being sent and signal level being expected on the 10.2 that could potentially cause problems and that's the digital levels used within the Helix signal chain and can be monitored via the built-in signal meter on the Helix by selecting the output block and is controlled on a preset by preset basis generally by adjusting the channel volume parameter of the Amp model within the preset. In my case I never exceed about 60 to 65% on the internal Helix signal meter.

Great job at describing a lot of the details involved in taming and integrating pieces using MIDI. I've been doing much the same with my Helix Floor, Beat Buddy and Ableton Live over the last year. The biggest evolutionary change came when I decided to move away from the Helix as the center of control to a dedicated MIDI controller which may be the path you eventually end up at as well. It's very enticing to think of using the Helix to control everything, but unfortunately the MIDI capabilities of the Helix can be very limiting when compared to dedicated controllers such as a Morningstar MC8 or other various models they, and other manufactures have. Helix can handle very simple interactions but I quickly discovered it can be quite limiting when you need to coordinate multiple MIDI interactions amongst a set of other devices. Ableton Live is perfectly capable of sending multiple simultaneous MIDI commands, but has a fairly tedious implementation to work with. By inserting a MIDI controller such as a Morningstar MC8 I have a very robust MIDI "traffic cop" that can be triggered by a single MIDI command from Ableton and can initiate multiple simultaneous interactions between all sorts of MIDI and even non-MIDI devices. I suspect you may end up finding yourself in the same situation eventually. For example, I normally have a different preset for each song. When I change songs I need to change the preset on the Helix, select the matching drum song on the Beat Buddy and prepare my setup so when I initiate the playback of the song the correct track (scene) plays on Ableton and it will trigger all subsequent interactions that need to be accomplished during the course of the song such as changing snapshots on the Helix as well as initiating a drum fill or song ending on the Beat Buddy. The way I handle this is when I change banks on the MC8 it sets the correct preset on the Helix, selects the appropriate drum track and the new bank on the MC8 has all the footswitches defined with all of the subsequent commands that will be needed to be triggered during the course of that song. I trigger the playback of the Ableton scene from the MC8 and from that point on during the song when something needs to change Ableton send a single command to the appropriate MC8 footswitch which will trigger any and all MIDI commands that might be needed at that point in the song. In effect, once I start a song there's no more toe dancing until the song ends and I need to select the next song, and the next song is a single footswitch. This model can be easily expanded to incorporate any number of other interactions with external effects or even stage lighting or running overheads or videos.

In my experience the Retro Reel is one of those effects you use very delicately...a little bit goes a long way. I would assume that's even moreso on Plexi style amps that are already fairly dark. I tend to get the best results for my ear when I place it at the very end of the signal chain when using something like a HiWatt amp that's being boosted a bit with a Teemah with a Fane cabinet using a R121 mic. Wow/Flutter at 1.5, Saturation at 4.5 and Texture at 4.0. When I use it more toward the start of the signal chain those parameters are considerably lower. On some setups I actually prefer the 15 ips over the 30 ips, but that's really dependent on the type of sound I'm going for.

I exclusively use IRs, not because they're any better than stock cabs, but just more convenient. That's because I use a LOT of presets for different songs and guitar configurations and it's much easier to simply select from a list of selected IRs than to constantly dial something in. It's also easier to make comparisons of how different cabinet and mic setups will affect a given preset.

...or worse yet...trying to solve every problem with EQ. Try going to the basics first (amp/cab/mic/mic positions)...then see if you REALLY need all the EQ you've been dialing in.

You select which sample length you want and then you select which specific IR you want. The same IRs are in both lengths.

Just a quick level-setting of your understanding of modeling as you move forward with your evaluation since you're new to the game. The TRUE difference between modelers is relatively insignificant. The brains behind all of these modelers are pretty much equivalent as far as processing capabilities despite those that tout newer generations of chips. DSP processing chips aren't like microprocessors in computers. They have a very specific task in life and there haven't been any huge leaps in newer architectures in a while now but those are the differences you've noted as improvements and that's in all modelers. You can increase the overall power by adding additional chips to a unit and that adds to the price tag of the unit, but usually won't make much of a difference in the actual performance of any individual model of the component amps, effects and so forth. Modeling is nothing more than mathematical computations that simulate the behaviors of a circuit in the digital world based on measurements taken on those circuits. So there's no real "magic" in how one builds a model, they're all done the same way. What varies from modeler to modeler can be the level of complexity you neeed in any given preset (which directly relates to the number of DSP processors), the ease and level of integration the unit provides with the outside world, and the simplicity or complexity involved in building a preset. It's abundantly clear from the enormous numbers of blind tests between units, there's no particular single modeler that consistently can be identified as better when judged by a wide audience. Therefore, your best bet in determining what's right for you is to determine how you intend to use it and what kind of user you are. If you want to maintain a typical paradigm where you have everything you'll ever need for a performance baked into a single preset, then you should probably look at something with more DSP chips. If, like me, you prefer to tailor your presets to a given song or genre, or have simple needs in what you need for performance you will have more, but less complex presets, and you can probably get by with fewer DSP chips. What tends to be the most important aspect for a lot of newer users of modeling is how complex the unit is as for as building a useful preset. Not only the complexity involved in using the modeler's interface but also the depth of technical details the modeler allows you to adjust. A good example of this is the Helix compared to the Fractal. The Helix is pretty easy to navigate compared to Fractal and allows you access to a significant amount of technical adjustments on any given model, whereas the Fractal doesn't have the easiest interface but does allow you to access and adjust significantly more technical details. So this really boils down to evaluating what's best for you in terms of how technical you might be or how little you care about the deep technical details that could be involved in building your presets. Personally, I've always told friends of mine that want to get into modeling that the issue is more about evaluating YOU than evaluating the different modelers. If you understand what will make you happiest and how you like to work, that will naturally lead youto the best unit for your needs.

Folks...It's abundantly clear to me that SaschaFranck isn't here to contribute or discuss...he's here to argue. Why feed him??? It goes nowhere and accomplishes nothing...

Because that is where the differences in signal levels (mic, line, instrument) will be actualized and have to be properly compensated if it's going to be consistent with the signal that will reach the end point analog device.

Bear in mind that would require that level setting to be done within the digital to analog conversion process which is a crucial element in maintaining minimal latency, and I suspect that's why most modelers avoid it and rely on you to manage your output levels within the digital environment of the modeler. I'm not advocating for or against such a thing, but I really don't find it that much of a problem as I commonly manage my modeled signal level on every preset I use as a part of gain staging my presets so it's never been a problem for me.

Bear in mind that input levels can vary significantly for different mixing boards depending on how they handle their preamp stages. Some boards have an automatic compensation on each channel depending on whether you're using 1/4" (which will be assumed to be at Line level) or XLR (which will assumed to be a Mic level). That's because those are the typical standards in the industry for 1/4" and XLR. On the Helix you have a choice of signal levels, but I always go with the industry default which is Mic level for XLR and Line level for 1/4". In my experience across a number of different mixing boards you'll be safest doing what rd2rk said. Set your XLR output to Mic level and disengage your Helix volume knob from the XLR output in global ins/outs so that it sends the full unity signal at mic level. This will be consistent with how most other inputs coming into the mixing are handled such as vocal mics or mics being used on cabinets. You can still manage the output levels inside each of your presets using the channel volume of the amp model or the output block and you can estimate your signal level by selecting the main output block of your preset which will display the presets signal level in the lower right corner of your Helix display which I normally keep at about 60 to 65%. That will allow anyone working the mixing board to easily gain stage your input signal and all your presets/snapshots will adhere to that consistent level. Configure your 1/4" outputs to Line level and reserve their usage for going direct to powered speakers which is exactly the same signal level all passive (non amplified) mixing boards use for going to monitors or front of house speakers.

I would be hesitant to recommend a Helix to a brand new guitar player. It's a real handful as far as the depth of the technology and a complete paradigm shift from the typical "plug your guitar into and amp and play" situation most people are familiar with. The modeling paradigm simulates the real world amp and effects situation, but it would likely take away from your primary joy of just learning to play...unless you would consider yourself to be a pretty technically inclined person. It is fairly easy to find resources to download fully functional presets and just play, but the most reliable ones aren't the free ones typically. At some point you still have to learn how to tweak them or troubleshoot them if you make adjustments and things go wrong. That's probably fine for someone that's pretty technically inclined, not so good for someone that's not.

No...I'm at the stage in my life where the only concerts I go to is the live performance of 'Frozen' at the Straz with my granddaughters.....LOL

One VERY important reason for using a real mixer for the Helix is you get access to a legitimate analog signal meter on each channel. That's actually one of the main reasons I always have my mixer in my live setup. The USB output of the Helix sends a digital representation of your sound whereas the XLR or 1/4" outputs send an analog signal after the internal Helix output is converted. The internal signal meter in the Helix gives your a rough idea of the digital signal level, but isn't representative of the actual signal level that is used in live performances and is based on whether you specified the output to be sent at Line, Instrument, or Mic level from your analog outputs, which can dramatically change the actual output signal. Because everything I do ultimately ends up in a live performance it's very important that I have consistent levels across everything I'm sending to the live mixing console so it can be easily gain staged in a live setting. That's MUCH harder to do using a audio interface which typically doesn't really provide an accurate and easily measured representation of the output signal especially when it comes to ensuring all your presets, snapshots and recorded audio outputs are at consistent volumes for a live performance.

The only real use I find for dual cabs is when I use stock cabs and want a specific pair of different mics and mic placements. I never use dual IRs as IRs can be captured with various mic combinations and placements, so there's no need. I do use a lot of IRs that are shot with dual mic setups. Other than that I'm not sure I see any real benefit from dual cabs. Multiple mic IRs is a different matter however. I do occasionally use IRs with multiple mic and mic placements, in one case as many a five different mic and mic positions, as that can make a significant difference in the overall realism of the cabinet's sound. However, as is the case often with the Helix, I think people use multiple cabs because they can...and then convince themselves of the benefits.

My first question is what are you trying to accomplish here? I only ask that because you mention using the Headrush for output instead of the LSR308's which would be uncommon for a setup that will be actually used for studio recording, and more indicative of a setup for live performance, and in my experience those two worlds tend to be very different. I have mine separated on two separate systems, but they wouldn't necessarily have to be. In my case I use the Helix as my direct audio interface to the DAW via USB for all inputs (mics, guitar, keyboard, etc) which I record track by track. My studio monitors are also connected to the Helix for playback either through my monitors or through my headphones for recording and playback purposes while in the recording process. For live performance playback I have a very simple audio interface connected to my computer via USB which sends the playback of the recorded output of the DAW via two 1/4" outputs to two stereo channels on the mixer which is connected to my Yamaha DXR12 FRFR speakers. Likewise my live vocal mic is sent directly to that mixer on a separate XLR input as is the Helix XLR output with it's own separate mixer channel so each source can be independently gain staged and mixed for live performance. I don't know if that's what you're after, but that's how I do it.