DunedinDragon

I've never heard of these, but they appear to have the key features you would want. Just so you know the Alto does have DSP processing as do most of these type of speakers. The Alto has the contour switch which is a fairly rudimentary function for modifying the response curve, whereas this has some more extensive functions..almost to excess. It seems to have the more important DSP functions such as placement, but it also provides DSP functions for altering bass, mid, and treble response which is very strange for a FRFR speaker and something you typically don't see or want in a speaker of this type. Typically you want the DSP to manage a flat response and only make modifications based on placement or sometimes usage like the QSC and Line 6 offerings do. I would imagine you'd want to simply leave the bass, mid, and treble adjustments set to flat and just adjust for placement and hi pass (for subwoofers) in order to maintain the flat response. I keep my hi-pass setting at 125hz as that matches most crossover points for subs and there's not much useful stuff below that range for typical guitar needs. I suspect the extra $50 was more about the Bluetooth capabilities than the DSP. You see this feature more in speakers that are marketed more toward the DJ audience than toward standard musician FOH or FRFR applications.

IRs (2048 size) and Reverbs tend to be pretty DSP hungry beasts, so you're likely to run out of DSP much quicker if you're using a lot of them so you really can't just count the blocks in a signal path. Some are WAY more expensive than others.

It depends on what you want. If you don't care about the factory presets just restore the bundle because typically they change with each update. Some folks reported problems with bundles early on though. I don't know if that's been fixed so setlists might be safer. I personally don't use either. I always back up my presets consistently by preset, so I restore just what I need at the moment.

To be honest the only thing I really use my volume pedal for is to kill the signal while I switch guitars. It's either 100% on or off. I still clean up my tone with my guitar's volume knob.

No..it looks like he just ran out of room for all the effects blocks he wanted for path 1a, so he extended it by using 1b. Personally I would have just sent the 1a signal to 2a after the amp and cab and had a whole new DSP for my post amp effects.

This is one of those things I've been curious about myself ever since they introduced the ability to set your XLR output separate from the master volume knob. I did experiment with it once when we had a bit of extra time during a sound check. The sound man and I both agreed we couldn't hear any difference at all so I just went back to doing things as I normally do rather than take a chance on a setup I wasn't completely confident in for a live performance. I think if I were dealing with a situation like yours where you often need to adjust the master volume I'd see the benefit. But we've gotten our setups down to the point that we really don't have to adjust any volumes on stage once we dial everything in during sound check, so it's not a pressing issue for us. I still may consider it on a case by case basis when we find ourselves working on a stage or venue that presents some challenges for getting our stage sound balanced and I might have to make some on-the-fly adjustments. It wouldn't be for S/N ratio, just the convenience of knowing the sound man will have a consistent signal to work with.

Everyone tends to have their preferences, but for myself I tend to generally follow these general ideas: Volume pedal at start of chain Compressor typically at start of chain, but can be put after the amp and cab toward the end of the chain for a more subtle effect Modulation effects (a little more organic sounding after amp and cab/iIRs) Distortion pedals just before amp (haven't used fuzz so that may be different) Amp, Cabs/IR Harmonizer Reverbs, Delays (in parallel if used together) That pretty much tends to work for most of the things I need to do.

A meter would aid in normalizing the volume levels between patches. Right now you can do it by ear, but the ears can be deceived so having a meter that gives you a reading on the different patches will let you dial in the volumes a bit more consistently. Obviously it would help in making sure you're not sending too hot a signal out, but as I understand it that's pretty hard to do from the Helix. Where that really aids things is in being able to send a consistent signal level to the board. For example, because I have a small mixer I can plug into at home that provides me with a signal meter, I can make sure all of my patches have a consistent signal level. Therefore when I gain stage during sound checks I only need to have the sound man dial in my channel's gain level from any one of my patches, because all of the other patches will be the same signal level. That saves a lot of time.

EXACTLY!! That's the reason I asked about what the HECK they mean by unity? I know I keep my Helix master volume set at 11 o'clock with the volume on my DXR12 at 12 o'clock or what they reference as unity. With those two settings I design all of my patches so that the line signal going into the mixing board comes in at around -4 sDBu. That gives the sound man more than enough room to bring the level to unity or wherever he wants it for the mix without any risk of overdriving the signal and it provides me with more than enough volume on stage (around 100 db measured by a sound meter). If necessary I can make adjustments right on the front of the Helix but I rarely ever have to other than minor tweeks during sound checks. From what I gather on these forums that tends to be fairly consistent with what a lot of people do. I think this whole thing started when Line 6 made the statement about having your master volume maxxed out because that produced the best signal-to-noise ratio. But this isn't 1970, this is 2017 and those sort of noise floor issues are hardly relevant in the digital age. Maybe if you're in a perfect acoustical setting you might hear some slight difference, but I don't know of anyone that plays or records in that kind of situation. But I do know a LOT of people that, if they needed to adjust volumes on their FRFR speaker would be highly upset at how painful it is during sound checks or performance to have to adjust it on the back of a FRFR monitor. It's simply not practical, nor is it necessary from what I can see.

What I've found, at least in the case of the dual harmony, is it takes whatever signal it receives and creates a duplicated harmony signal. What mushes things up is when you send a clean harmonized signal to something that overdrives it. That creates a lot of noise artifacts so the harmonies aren't clean. Placing it after the amp and drive pedals ensures the dual harmony will duplicate the exact signal it's receiving but at the harmonized level making it a much cleaner and articulated harmony.

Just curious, what does unity gain mean to you? Technically the term unity gain is what is used to specify that nothing is being taken away from the signal and nothing is being added to the signal. Therefore the signal is in it's unencumbered state, but it only applies to signal strength. For example, when you look at a mixing board the gain knob at the 12 o'clock position is marked with a 'u' to designate unity. Even more telling are the faders which specify a zero (0) position below which are negative numbers and above which are positive numbers which specify positive or negative sDb units. So the signal coming from the Helix is just a signal delivered at a certain signal level. What determines unity is at the mixing board and how that signal (regardless of it's strength) will be handled on the pre-amp side by either adding to the signal or taking away from the signal. Any line coming into the board will be treated exactly the same way whether it be a vocal mic or a amp mic even though the signals from different devices vary greatly in their signal strength. This concept has been applied to some powered speakers as well in that the midpoint of the dial is considered unity and the signal will either be boosted or subtracted before sending it to the amp. I just can't get my head around what people mean when they talk about unity from a device that PROVIDES a signal. Technically it should only apply to a device receiving a signal and whether the preamp should boost or reduce the signal.

That's a pretty flexible set of amps for the most part, but depending on what type of genres you're covering might have some holes. One that jumps out at me immediately is the Vox sound which is so prominent particularly in a lot of british sounds like the Beatles or Queen. You might have another hole if you're doing clean blues like BB King or jazz type sounds similar to Steely Dan which is the domain of Jazz Rivet 120. Of course this is all dependent on how authentic you want your sound to be. I'm somewhat OCD on such things so I use a pretty wide range of amps to emulate the sounds of the artists. But YMMV.

From the documentation this is a traditional insert plug similar to the type used on many PA mixing boards where it's using a 1/4" TRS plug that breaks out into one output (send) line and one input (receive) line. This would probably work fine if you simply wanted to use the Helix as an effects board. In that case I might try first plugging the send line side of the amp into a Helix receive plug and use the 1/4" output line of the Helix to go to the receive side of the Y port on the amp. If you want to use the Helix as a modeler and simply use the FlexWave as a power amp I'm not sure if that's possible since the documentation doesn't specify whether that insert plug is after the pre-amp stage on the amp.

I use the dual harmony effect quite often in my performances to produce harmony leads from Brian May, The Eagles, and Charlie Daniels Band with no problem at all. It absolutely is a polyphonic effect given that polyphonic means capable of producing more than one note at a time. I generally only use a single harmony part, but I have used three part harmonies on occasion when doing Brian May leads. From what I can see it operates no differently from my Digitech Harmony Man which I used for years. In fact the dual harmony effect has several more scale and mode options than did my Harmony Man which increases the accuracy of the harmonies it produces. Of course, the dual harmony is dependent on the person using it to understand and input the correct parameters in order to produce the correct harmonies. It also matters greatly where you place it in the signal chain. I've found it to be the cleanest when it's placed after the amp and cab, given that any overdrive pedal is placed before the amp and cab.

In this case going direct is the only logical choice. Going 4CM into an amp in another room then mic'ing it is like going from St. Louis to San Francisco by way of New York. You're getting exactly the same effect as you would by going direct, so there's no point in not doing it.

It's not just lower tuning it may also be the type of guitar and genre. I use 12's because I commonly play a hollow body Gretsch guitar, often playing jazz as well as finger picked Chet Atkins style. A 10 doesn't really reflect the body in the tone like a 12 does. So I really think this is one of those areas where you have to examine your specific needs for how you specifically play.

First of all, you need to select the 2.21 update and follow ALL the directions in order and precisely. If you do that you'll notice the first thing you do is to use the Helix application on your PC to back up your patches. Then you download the updated Helix application. The updated Helix application will install the appropriate updater. YOU DON'T DOWNLOAD THE UPDATER. THE INSTALLATION OF THEW UPDATED HELIX APPLICATION DOES THAT FOR YOU. I think the problem you have is a mismatch between the Helix application and the updater. Just follow the directions on the 2.21 release notes precisely and all should go well.

There's not enough information here to help you. How did you import them? What's the significance of slot 17? What kind of file management system are you talking about? The Helix has no particular logic for how it manages IR's. It only keeps track of what IR slot is being used on any particular IR block in a patch. At this point it's entirely on your shoulders to make sure that you place the same IR into the same IR slot being referenced by your patches. This is the reason most of us put a 3-digit number that equates to the slot number at the start of the file name for the IR and then save all of our IR's into the same directly. If that directory is organized alphabetically by filename then all you have to do to restore your IR's in order is to select all of the IRs in your directory and drag/drop them into slot one of your Impulses tab and they will all be in the correct order.

In this area budget is everything. I've spent almost as much if not more on speakers than I spent on the Helix itself. For live use I use a Yamaha DXR12 and for studio use I have Yamaha HS7's. After many years of being a sound man and working with all sorts of powered speakers I would definitely tell you that you get what you pay for in the area of powered speakers either in the studio or live.

I'm not a particularly huge fan of using headphones as most of my efforts are directed toward live performance. That being the case I'd rather dial in my sound on the system that will actually be used. I do have a pair of Beyerdynamics DT 770 Pro 80s that I use when recording and have tried them out on the live rig. But I just tend to be more comfortable in the results if I dial in my patches using the FRFR speaker (DXR12) I'll be using on stage.

Yes, the Helix can operate as a sound card for your computer. However your computer speakers may not connect to the Helix depending on what they are. Typically most people that do this (including myself) plug powered studio monitors into the two 1/4" L/R output jacks of the Helix, then just plug the Helix directly into the PC via the USB cable and everything works great. If your computer speakers are the typical consumer computer speakers that use the small 3.5mm stereo plug output on the computer, there would be no place to plug them in on the Helix. The dual harmony effect would be able to do things such as the outro on Hotel California. You would have to specify the key and the scale type of the harmony to be used. In the case of Hotel California the key doesn't change. The chord progression changes but the key and scale remain the same.

The things I wish I had known, or knew somewhat but learned better due to experience with the Helix: -The target for dialing in patches is a polished, studio production sound...not a traditional stage amp sound. -The most useful IR's are the ones that are the mixes of mics and placements, not the single mic versions. - Often Helix native dual cab setups can perform as well if not better than many IRs once you understand the nuances of parameters such as mic distance and early reflections. - Cab and IR High and Low cuts can be deceiving when dialing them in using a FRFR depending on your position relative to the speaker. They will often be darker when recorded or live with a band. You should give yourself adequate distance from the FRFR speaker to hear them more accurately. Aggressive high cuts on cabs or IRs can result in less articulation and clarity in live band situations or in recordings. - DO NOT be tempted to use global EQ to high and low cut patches. High and low cuts should be tailored to the type of amp being used, the guitar being used, the cab, mics, and mic positions. - Learning how to best use compressors, and deep parameters such as SAG and BIAS can greatly enhance your ability to fine tune patches. - Chances are any patch you buy or download will have to be significantly modified to work with your specific guitar and pickups, so you may as well learn how to build patches on your own. That's about all I can think of for now. I'm sure there are others, but I'd hate to deprive you from learning things on your own :P In my case I just use the USB 1/2 from the Helix for my recording. I have separate patches setup for guitar, keyboard, bass, and vocals. I simply select the patch and arm the track I want to record. To some degree but the Fletcher-Munson effect will always be a player in how you perceive the sound due to the differences in how we hear certain frequencies at different volumes. Actual live volume has no effect on a modeled patch. Patches are only affected by the modeled amp master/channel volume setting. Increasing/decreasing the Helix master volume doesn't affect the sound. This is particularly true with most FRFR powered speakers as they are specifically designed to accurately reflect the source material across a large range of possible volumes. What specific types of FX pedals or individual pedals (if any) are you keeping to use with helix? Do overdrives work the same as on an amp? Will it clip the signal or respond normally going into the Helix?

After re-reading your initial post and seeing that you also have a full Helix that's been connecting, that isolates the problem to the LT unit. Have you made that clear to Line 6 support? From a software engine perspective there shouldn't be any difference between the two. The device drivers should be the same, etc. Therefore I think all the evidence points to some hardware issue with the USB port on the LT. I'd touch base with Line 6 and make sure they understand your full Helix works fine with the same cables, the same computer, and the same OS, but your LT doesn't. That makes a BIG difference in isolating this problem.

Although there are people that report having problems with the USB port, the vast majority of us have never had any probems. I own two Helix units and commonly plug them into a desktop, a laptop, and a tablet computer all running Windows 10 all from different manufacturers. I've never had a single problem with connectivity. If there were an issue with the design of the USB this would not likely be the case. The three situations I know of that have been common issues with the USB connection for people is plugging into a USB hub device rather than a direct USB (remembering that in many cases manufacturers use a hub design to provide additional USB ports when only one or two are actually direct), the supplied USB cable that came with the Helix is sometimes faulty, and various problems with older versions of operating systems on both the PC and Mac. Whether it's a driver issue or not there's certainly not enough information to determine that from any anecdotal evidence that's been posted here or on other forums. There are way too many variations in the implementations of computers, operating systems, virus protection schemes, etc. to jump to that conclusion.

This is the reason I encourage people using this type of technology to get familiar with PA sound boards. That way you can ask the sound person relevant questions about the setup of your channel.