DunedinDragon

It took me about 3 minutes to google a YouTube video for switching Snapshots on Studio One and the secret sauce came down to the value being sent for CC#69 which he had written down on a notepad and here's the screen shot of it: I only did it this way because this guys video was almost unendurable as he first wanted to make sure your listened to his original composition before he'd hand in the details. I thought I'd spare you that. But if you want to view the whole video, God bless your patience. It's at the following link but I'd strongly urge you to fast forward to the 7:31 time in the video to get to the meat of the subject:

The real test is not your monitors, but how the production sounds once you have it mixed and mastered then played on your car stereo, your laptop speaker or your Bluetooth speaker. It's not very likely anyone else that listens to it will be using a studio monitor.

Well...what a mess we have here...but I don't think it's insurmountable. The key thing is your last statement. Assuming you're using some decent playback monitors or headphones when you record, you're getting good results from your USB connection. The difference here is that the USB connection is a digital connection to your computer, whereas the connection for live performance is always going to be converted to analog for either 1/4" or XLR connections, and that's where the differences lie. There are a lot of things that can affect this conversion with the primary one being signal level and the methods you use to manage that signal level. I don't know if you've used it or not but if you select your output block you'll see a signal meter displayed at the bottom right hand corner of the screen. That is the digital signal level of your preset which would normally be in the range of 60 to 65% at it's peak levels. At those levels it's common that many people going live to a PA via XLR to Mic signal level and disengage it from being affected by the large Helix volume knob when sends the full level signal to the PA which is what most PA's expect to receive. In this arrangement you get the best digital to analog signal conversion with the lowest signal to noise ratio going into the mixing board and can be easily gain staged at the mixing board. In this situation you control the level of your output using the Helix built in signal meter at roughly 65% (along with what you hear with your ears) to manage the volume level in your patch by adjusting things such as the channel volume on your amp model or your output block, neither of which will affect your tone but will get consistency in your output levels between the patches and between different snapshots with the patches. I use the amp model's channel volume to control my levels, so that channel volume level varies depending on the amp as well as other factors such as what type of effects are contained in my patch such as boost, gain and compression. But now you get to the PA's which can be an unholy mess depending on how old the PAs are and the architecture being used. When it comes to phantom power that's really pretty easy to control just using a phantom power blocker which plugs directly inline between your Helix XLR output and the XLR cable going to the mixing board and is readily available from Amazon or other online music dealers. The bigger issue is the significant architecture changes that happened to PA's starting in the 90's and made significant leaps in the 2000's and is at the core of what most people are referring to when they use the term FRFR. In the past such as the 70's, 80's and even 90's PA's typically used a passive mixing board that routed to a central amplifier and a separate crossover for lows and high frequencies which then sent the amplified signal to powered speakers and subs. What changed was the use of powered rather than passive speakers which used built-in amps in the speakers which were better tuned to the speakers and eliminated the need for a centralized amp. However the next stage of development in powered speakers in the 2000's was the one that began the use of the term FRFR because they incorporated DSP intelligence (yes the same DSP in your Helix) to better manage not only the frequency allocation between the lower and mid frequencies for the speaker and higher frequencies horn, but also the better allocated the amplification and tuning of the speaker based on how it would be used. The problem that quickly jumped out at me when you recounted the different PA's you were using is that none of them from what I could tell are of the FRFR variety. Even the newest one which is the Dynacord is a new model that still uses a centralized amp. That doesn't mean it can't be tuned in the same way as a typical FRFR speaker if it has tuning options in the amp, but that's pretty rare except for very expensive amp setups used in major concert setups. The vast majority of us that go direct to mixing boards are using the more modern style of setups that incorporate FRFR technology in both our main speakers as well as our floor monitors. In the past I've dealt with this same problem in some smaller venues that have older PA's and passive speaker systems and how I dealt with it was to use my own Yamaha DXR12 powered speaker mounted behind me in the backline on a pole mount to augment the PA and allow me to not have to make adjustments to my presets based on the difference in the PA. It wasn't a perfect solution, but it worked well enough in uncertain circumstances where the house PA technology was unreliable. This is one of the main reasons most of the bands that do the direct to PA thing also own their own PA setups so they can control such things gig to gig. It's not impossible to accommodate older PAs or passive PAs but it does mean you may have to adjust some things in your presets as most of those older setups tend to have quite a big drop in frequency response in the upper mid level frequencies at the crossover point. Maybe it could be accommodated through the global EQ although I've never attempted it.

And how, exactly would you know that? Were you watching what his guitar tech was doing backstage to accommodate the guitar change? Quite frankly I doubt you would even notice any suspect actions on my part when I change guitars on stage since the only thing you might notice is I changed my preset...which I do for every song. The real fact is, the statement "those that fail to plan, plan to fail" shouldn't bother anyone other than someone who's insecure about their lack of preparation. There's not a single instance of anyone in ANY field of endeavor who's a globally recognized success story that wouldn't agree with that statement as they're all well-known for their diligent work ethic.

I guess I just don't see this as a huge issue given I've been dealing with this quite well for the last 7 years on my Helix. The difference being that I accept different guitars are going to sound different so I simply accommodate it by having a different preset that incorporates the adjustments necessary. It also means I have to actually "plan ahead" (oh the horror!!!) about which guitar I'll be using BEFORE I get on stage. The fact is different songs sound different to a large degree depending on the type of guitar being used. That's the primary reason I have four different guitars to accommodate those differences (Les Paul, Gretsch Silver Eagle hollow body, Strat and Tele). This doesn't work in all cases (I often finger pick on the Gretsch which sounds horrible if done on any of the others) and in some cases I can directly substitute a Les Paul for the Gretsch and vice versa and the same for the Tele and Strat without any adjustment to the preset, except for maybe minor adjustments on the guitar itself (tone knob, pickup selects, Gretsch dirt switch, etc.). However in some cases when I'm more or less forced to use a Les Paul on a song rather than a Strat due to the flow of the show, I'll simply do some surgery on the preset which includes WAY more than a simple adjustment to the input impedance or PAD. Often it involves changes in the amp, cab, mic'ing and definitely EQ and gain components. But the overall preset still works in the same way and I just have it exported to my hard drive with a code at the end specifying which guitar it's for. I realize this isn't an accommodating approach for folks that don't want to prepare or plan for their performances, but it works fine for those that live by the motto: Those that fail to plan, plan to fail.

You can assign the Interval parameter of the Poly Capo to your expression pedal, but any assignment to a stomp or snapshot would only allow an absolute value to be assigned to it. There is a possible way you could use something like a Morningstar MIDI controller pedal and define it as an expression pedal and use it's increment and decrement function on one of the footswitches...possibly?? But that's only in theory.

I guess I don't understand what the big deal is here. To my knowledge there has NEVER been any kind of specialized speaker used for a resonator. In all cases it's always been used with a traditional amp. So pick an amp and speaker and mic and play. The resonator produces the correct sound and tone. All you have to do is amplify it appropriately. My most common setup for my Gretsch resonator is using a US Deluxe Nrm amp model with a Matchless IR with a Celestion G12M25 Greenback speaker with a R121 ribbon mic. Dial in tone and mic placement to taste. There's nothing complicated here......

Well that's why it's Beta. Your only recourse is to report it as a bug to Microsoft and wait for a legitimate release to see if it's fixed.

And where on the frequency response curve is 31hz for that speaker?? If it measures very low in db's at 31hz (as the VAST majority of any and all speakers do), then what's the point? You can throw around specifications all day if it makes you feel better, but whether it's likely to SOUND better isn't measured with that kind of myopic and meaningless spec and that's my point. Go take a look at the Celestion web site and examine ALL their speakers and look specifically at the speaker response curves which they publish for ALL their models and examine the level of response you get at those low levels. But ultimately it all comes down to what will the audience hear and what will make them not only hear but feel the bass? That's all part of what a dedicated subwoofer does which is to specifically boost those very low end frequencies which need to be felt more than heard. As I mentioned we use a QSC KS112 subwoofer which gives everyone in the room (including stage and audience because subs are omni-directional unlike your cabinet or speaker) a more than adequate low end bass feel for ALL instruments that need the low frequency boost including the bass, the low end of the keyboard, the kick drum and lower toms even though that subwoofer is spec'd at 38Hz as it's lowest response, yet at a volume level setting of 3 all low bass frequencies can not only be heard but felt in a room with a capacity of 150 people. Have more people than that...get the next bigger sub or get multiple subs. Take a look at the forest....NOT the trees.

Frequency response is really not relevant, its about how you allocate the response of those frequencies that matter. If a speaker produces 1db of response at 20 hz, it can be claimed to have a frequency response that low, even though it's not really a functional or useful response. What matters is the frequency response curve. That's what you want to know. How evenly does it respond at different frequencies? THAT is what really matters.

Our bass player is just fine using a Yamaha DXR12 FRFR as a monitor and going direct into the mixer. He was using a dedicated bass modeling pedal for a while, but after it stopped working he just went direct through a DI box and thinks it works fine for him. Of course we also have a QSC KS112 subwoofer with helps a LOT with the low end, so he's quite happy with his tone. But that's a premium quality FRFR so I can't vouch in any way for a cheaper monitor like a Headrush112. In our band's situation none of us use stage amps and we all go direct, so that makes a difference as well. The more amps on stage the more noise on stage so even a subwoofer might not be as much of a help in that situation.

Personally I don't see any real advantage for most people to rush to Windows 11 at this point. There are some interesting new features, but most would be more useful for someone in a busy workplace using multiple active applications in a variety of different jobs or interacting with multiple co-workers in team video conferences. There are some more consumer oriented features for gamers as well as those that have some kind of driving need to run Android apps on their PC's, but the android stuff is on a controlled rollout based on the age and capabilities of you PC hardware. The capability of having pop up widgets is interesting I suppose but I already have that on my phone which is always with me even when I'm at the computer. I guess my question comes down to this. What advantage is it to take the risk running a BETA version of an operating system on a PC that you're totally dependent on for certain activities in your life if there aren't features you just can't live without?

I own both Yamaha DXR and QSC K series speakers as well as a couple of other types. Both are great options, but I have to admit I really prefer my QSC K10.2 as my live monitor more than my DXR12 which I use at home to dial in my tones and often use them as front PA speakers for the band. There's something about the presence in the QSC that just makes it easier to work with in the context of a group as floor monitors. BEST infers a subjective opinion, so you can get plenty of those. I only know what's worked perfectly for me as a live musician playing every week in a wide range of genres for many more years than I like to admit to. Any of these speakers will work equally well for amp models as well as acoustic guitars or vocals. That's what they're made to do and why they're so prevalent and popular. I personally play a wide range of instruments live through mine including various electric guitars, banjo, mandolin, bass, keyboards (piano, organ, synth), pedal steel, bluegrass fiddle...and of course multi-part vocal harmonies. They are general purpose, top of the line, live performance speakers. Since going to the Helix 7 years ago and using high quality speakers like these I have sold all my amps and have none in my house. The thing that's important to me, ultimately your audience is going to hear your performance through one of these types of speakers. Even if you play through the Mesa Boogie, once you mic it and send it through a mixing board, it's going to come out to the audience through one of these types of speakers sounding the same way as the Helix modeled speakers going direct to the mixing board...so what's the point? I don't use these type of speakers in my studio because that's not what they're made for. For that I use Yamaha HS7 speakers or a DT770 set of headphones. I do, however create certain backing instruments for our live performances on that system and play them back live through these type of PA speakers and they sound pretty much the same as a live musician would sound.

I have to say I TRULY enjoy these interchanges where someone with 67 posts questions the validity of answers given by someone with 8,770 posts. I often just go microwave some popcorn, sit down and wait for the entertainment to begin. I really need a new hobby......

You could use a return on the LT as an input with an adapter, but you would need to reserve your second signal chain for the mic with it's own reverb/delay and the ability to control it's output level separate from the the guitar which you would have to confine to it's own signal chain. You would then send the guitar signal chain to the digital output and the mic to the PC as you're doing now. It would kind of be a pretty clumsy setup but if it's a low key, smaller gig people might not notice your PA is pretty lame. The PC doesn't make for a very good PA system. That's not what it's designed for.

I've used it quite a bit with an acoustic guitar and it sounds pretty good. However, if you want something unique, pair the 12 string effect with the polycapo up 12 frets for a surprisingly good mandolin sound.

I owned an actual HiWatt for years and I really didn't appreciate the Helix model until I paired it with the original Fane speaker IRs from Rewirez and it became the amp I loved for so long. I still use it extensively on a lot of 70's and 80's classic rock styles and there is no better amp for getting the Joe Walsh signature sound when it's paired with a Minotaur or Teemah.

Possibly the most famous and recognized amp for accepting pedals is the HiWatt or WhoWatt as it's known in the Helix. This amp really took over the music industry 40 or so years ago among the elite artists at the time like the Doobie Bros, Eric Clapton, The Who and many others and has remained a mainstay amp because of it's unique ability to work well with various pedals, and the Helix model of that amp maintains that tradition.

You only need to download and run HX Edit. That will take you through the entire update process for the Helix firmware. Line 6 updater is really no longer necessary.

I think the only thing good about not publicizing those kind of technical details is it would give users the opportunity to bash or promote one modeler over another based on technical trivia they have no clue about. I often find it hilarious when users that struggle to understand the concepts involved with building signal chains or the use of snapshots and presets, or how to route different signals in and out of the Helix get fixated on the one thing that even highly technical trained engineers sometimes don't completely understand.

There are several different ways of doing this sort of thing but my first impression might be to separate the different instruments into different presets to reduce the DSP footprint needed for each. Since it's all going to the looper anyway you could easily incorporate the appropriate looper MIDI commands into the Helix itself and save yourself some complexity in your setup.

^^^^^^^^^THIS^^^^^^^^ is the key reason I finally abandoned using traditional cabinets a decade or so back. I think sometime back in the 80's I started to get frustrated with the directionality of cabinets. Of course this was back in the day when it wasn't really necessary to mic cabinets, so we generally didn't except in studio work or larger venues. It started to frustrate me that the stage sound and even the overall sound of the band could be so affected by cabinet placement and where you stood on stage or what area of the venue you sat in. In all fairness, most PA systems of the time were pretty limited as far as performance, so there weren't too many options. At some point in the late 90's is when there were some important changes that started to take place with club level powered speakers, subs and mixing boards. I was doing a solo classic rock show at the time using backing tracks and MIDI controlled lights and took advantage of the advances and it totally changed my perspective on the possibilities when I crafted the entire show without any cabinets, just the PA system. Rolling forward a few years better modeling options became available, but they were mostly housed in modeling amps, so I went back to cabinets mic'd through the PA and began to experience the same old frustrations with the on stage sound. Finally with the appearance of high end modelers along with the appearance of next generation DSP driven powered speakers the stage was set to abandon all cabinets on stage for all instruments and go direct through a digital mixer and achieve a polished, studio quality sound both on stage and across the entire audience. It sure seems like a long journey to get here but I'm constantly thankful I stuck with it until the technology finally caught up with what I always envisioned it could achieve.

That would be the only real possibility for getting this type of feature added. However, it's also a double edged sword in that there would need to be a pretty broad interest level within the user community for this to actually get changed and I don't see that. I've operated for several years with two Helix Floor units and never come across any need to use or edit them both simultaneously. I suspect that's the most common situation you'd come across in the user community.

The "CPU" used in modelers are actually real time SHARC DPS processors. Upgrading one wouldn't likely be reasonable since they're not anything like a generalized computational CPU but dedicated real-time processors like those used in radars or other military real time applications. Dropping a newer version or especially a different family of DSP chip would be inherently incompatible with the code base in the Helix. The best way to think of DSP chips is they are high speed computational engines used in the manipulation of digital sound streams. There are data sheets available on SHARC processors from the vendors that make them...all guaranteed to put you to sleep within minutes of reading them.

By reactive monitoring I'm talking about an ongoing data feed or instrument feed. The protocol is clearly two-way, but I haven't seen anything that would send ongoing data without some direct interaction either on the unit or on HX Edit. Thus the implementation of the signal meter...