DunedinDragon

I'm not sure the article says anything different than what I said. It's making a reasonable guess and exploiting non-processing time (which there is a LOT of in HTML operations). My second sentence distinguishes between the programmatic choice within a browser of what to fetch out of cache and what to guess at for pre-fetching operations (which your article doesn't even deal with). In most modern browsers the cost of prefetching is FAR greater than accessing a page in cache and simply comparing it to decide if the cost of pre-fetching is worth it or even called for. As far as education, I'd suggest you learn a bit more about the differences between simple HTTP and HTML interpretive functions and real-time operations using very specialized DSP-centric operations requiring near-zero latency in pretty much continuous processing interaction with the user. One of these things any 7th grader can understand, and one of these things only a handful of people have experience with because of it's complexity. So I'd take your bet given you clearly can't seem to distinguish between the levels of complexity between the two different types of systems being compared here. Technically this is all kind of silly anyways as we're talking about two fundamentally different architectures such that "prefetch" isn't necessary with the Helix because all presets are always constantly in memory all the time (unlike a web page), but only the active one has any resources allocated to it such as DSP objects or working storage. That's the part that takes the time and that's the part of the system that's limited.

Wrong. Prefetching only works if you know (or can reasonably guess) what preset will be selected next. Even in the case of a page in the browser, it's never "pre-fetched", it's held in cache from a previous visit to the page and checked for consistency when loaded to make sure it hasn't changed. Not exactly "immediate". For prefetching to be immediate on the Helix you would not only need to prefetch the blocks and signal chain setup, but allocate the DSP for the blocks, and there's a limit to the DSP available.

Let's not forget to make sure the gain levels on the left and right speakers themselves have the same settings. And that may be the way they want it set depending on the coverage they need in the room itself.

If you read the previous responses to this, it's pretty clear that you have to mute your sound to get the Autoswell to reset. It's not that hard to do. I just personally use hybrid picking with a pick and fingers and palm mute to get all the strings to swell together smoothly.

I'm not sure not having the capability is actually the case. Certainly seeing it stated on an internet board doesn't make it so. But it's certainly not a show stopper, even if it's not in the box. Signal metering is easily and economically available as an add on to to anyone's rig that wants it in a number of different ways. Would I probably use if it were there...maybe. But the fact is I have constant metering available to me at all times in any patch without having to do something special to turn it on or off. I just play and watch the meter.

THIS is the only really compelling argument for having built-in signal metering. Although I also have to say I've never run into this kind of issue on stage because my process for getting ready for a gig would have revealed it and allowed me to correct it long before I ever went onto the live stage. However with the ability to provide for various levels of integration with outside devices (which I also don't use), I can see the value in troubleshooting in the same way as metering gets used at a mixing board in a live situation.

The reality is with modeling you're not going to hear what you used to hear on stage unless you use that type of configuration with the Helix. What you're going to hear on stage is what your audience has always been hearing if you mic'd your equipment and sent it through the mixer. The fact is your audience has never heard what you heard on stage whether it's live through a PA or recorded since it was captured by a microphone. And that's what they expect to hear because that's what they've been hearing their whole life.

Well, the fact is I worked with amps and guitars for 50 years without a meter other than the meters on mixing boards or DAWs. In fact, I don't think I've ever seen a meter built into guitar amp or a multi-effects board.

If you simply cannot under any circumstances get by without a meter, here's your affordable answer. Plug in and measure to your heart's content. https://www.amazon.com/Mackie-402VLZ4-4-channel-Compact-Quality/dp/B00EDRUQXC/ref=sr_1_80?keywords=mixing+board&qid=1577924375&sr=8-80

Sorry I misunderstood. But that also makes me curious why it's such an issue as we also have two guitars going direct to the PA from modelers so it's hard to imagine why it's such a problem for you. As long as you both have decent tones set up the rest comes down to differences in technique, and to some degree some difference in tone. But tone and EQ can't completely fix the problem as it generally requires differences in technique and approach. In our case the rhythm player is basically doing fundamental strumming with very few open chords, whereas as the lead player I provide the texture, and variety of techniques to help fill the song. We may be playing the same basic chords, but I'll tend to use different chord variations along with techniques like palm muting or picking techniques and inter-phrasing fills to help distinguish between what the two guitars are adding to the mix. The worst thing you can do is have both people trying to play the same thing. That's when it becomes a garbled mess. Study closely the way multiple guitar bands tend to do things like the Eagles or Allman Brothers. If you watch the Eagles documentary Hell Freezes Over, there's even a bit of a reference to it with Joe Walsh and I think it was Bernie Leadon discussing who was going to play high and who was going to play low, referring to fretboard position. We also tend to use different amp models and cab/ir setups as well as guitars which also helps. How the two guitars are mixed also comes into the picture. Generally speaking the rhythm guitar is always a bit lower in the mix than the lead guitar, but not by much. This allows the lead guitar to help manage the dynamics of the mix by dropping back a bit during the singing and coming up for fills and accents while the rhythm stays very steady behind it. This is a bit of a more extreme example of different techniques, but it should give you an idea from a live recording we did one day of one of our songs.

The part of the process you can't account for in doing it the way you're describing is that both guitars are going into the mixing board on a single channel. That means both guitars will have to be gain staged on one mixer channel together and managed via a single fader rather than individually as would normally be the case, and it's highly unlikely you'll get each guitar signal correctly gain staged on a single output from the Helix. I suppose if you separated each guitar onto their own individual signal chain and routed one signal chain to the mixer via XLR and the other via 1/4" you might be able to pull it off. I've never tried it that way, but theoretically it makes sense.

If I'm understanding you correctly I think you may be over complicating things here. I too have noted the differences in various IR outputs, but I generally compensate for them in the same way I compensate for any difference in volumes of my patches by simply adjusting the amp model's channel volume. In the end all that matters is the output signal level needs to be consistent across patches regardless of different amps or IRs. I personally can't perceive any difference in the finished tone with this approach and it's surely less complicated and easier to manage and maintain my patches if I adjust the volume of my patches in a consistent way, and Line 6 has assured us many time that the channel volume won't have any effect on tone.

As Codamedia mentioned, it's been my experience that more often than not I come across people using WAY too much overdrive and gain, and it may sound louder but it really isn't in terms of the actual signal. It's just ends up with more noise and less melodic content. You can easily turn down the gain/overdrive and increase the channel volume on the amp and drive the signal without losing articulation and clarity even if it's high gain. And in the end it's articulation and clarity that cut through the mix in terms of what the audience hears. In a way your second point is really also related to gain in that high gain and overdriven signals are already compressed via controlled clipping of the actual signal, so compressors tend to be less necessary than they are on cleaner tones with lots of dynamic range. I still sometimes use a final compressor at the end of some of my high gain presets, but it tends to be very light and absolutely adds no gain to the signal

The Helix volume knob has nothing to do with the level of USB signal going into a DAW. That's controlled by the global values of USB output. The Helix volume knob becomes relevant when going live through the XLR and 1/4" outputs. The most common way of ensuring you're not adding too much gain is to individually and listen to your patch while you enable and disable individual blocks in your preset to ensure they aren't adding too much gain to the base level(all blocks disabled) of the preset. The optimal spot for the Helix volume knob is based on what volume is required for a given setup such as an FRFR speaker connected to the 1/4" outputs along with what level signal you're sending (Mic, Line or Instrument). For example, I detach my XLR outputs from the Helix volume knob in global ins/outs which then sends a full strength signal through my XLR outputs (at Mic signal level) to the mixer, and I use my Helix volume knob to control the feed to my on stage Yamaha DXR12 speaker through the 1/4" outputs and I normally have my volume knob set at just above the 12 o'clock position. The feed to the mixing board will be gain staged appropriately at the mixer and I control the level on stage based on the on stage blend of instruments and voices.

What the HECK is phone volume?? Maybe try and explain your setup and what you're trying to do first.

I'm not sure why this would mystify anyone. There are actually quite a few people that have individual presets for each song especially in typical cover bands. I myself have over 2oo individual presets for different songs. It's a very natural separation for those of us that play a wide range of genres and styles and is one of the distinct advantages of having a modeler like the Helix. Even within a genre such as rock the signal chain for a Rolling Stones song won't be anything like one for Queen or the Doobie Brothers if you want an authentic sound. And there's no reason to not do it that way as the pause between songs means there won't be a problem with with the dead spot in switching between presets. Not to mention the necessity of needing a different guitar to get the Queen sound and the Rolling Stones sound. And that's not even counting going from rock to jazz to funk to country to rockabilly...etc. Snapshots are still very useful in managing the dynamics and sound differences within a song, but there are more disadvantages and limitations to using snapshots across songs than there are advantages in my experience. What is curious in this case is why, if you have different presets for different songs, you would need the type of complexity required in the OPs preset that has been the issue here. That's VERY uncommon in my experience when using a dedicated preset per song, and simplicity being one of the great advantages of that approach.

One thing you'll probably discover as I did when you go to full volume output at mic level on your XLR output, the differences in volume levels between different amps is not as wide as it is when the Helix volume knob was at 1/2. I suspect that has to do with A/D D/A conversion mechanism, but since adopting that approach the channel volumes on pretty much every amp I use in a patch tends to be within a very close range of 1 or 2 notches. Also in terms of metering blocks, that doesn't tend to be necessary if you follow the generally practiced procedure of checking each block's affect on the volume by enabling and disabling it. This helps alleviate the problem of multiple blocks building up the signal level on each other through the signal chain.

If you've gain staged your output SIGNAL level consistently it shouldn't have any problems with your output method. I've yet to run into any problems on probably around a hundred of different mixers and PA systems I've encountered over the last 4 years, and I can use those same presets to go direct to either a Yamaha DXR12, EV ZLX-12p, or QSC CP-8 all set at unity gain (12 o'clock) or direct to my DAW via USB using Yamaha HS-7's with no problems. It sounds to me like you may be confusing volume with signal level in your setups.

Probably most of us play in a variety of places with no changes at all to our patches other than output levels from the Helix. But that really depends on how diligent you are in setting up your patches and having an appropriate setup that will best represent your live environment. It's unlikely you would know if you were having problems with your pedalboard and amp because the sound you were hearing on stage would be nothing at all like what came out through the PA. If you went DI, your PA signal would be missing the acoustic elements of the cabinet. If you mic'd the cabinet your FOH sound would be representative of the type of mic used and how it was placed on the cabinet which could vary greatly, but it for sure wasn't what you were hearing on stage. What varies with the Helix is you're in charge of it all with every patch. Which means leveling the output and gain staging each different setup so that you send an appropriate signal to the board and listening to your tone on a system representative of what you'll be playing through. This is especially true if you're using different amps as each amp has it's own unique gain staging and tonal attributes. I have almost 200 presets (which I don't use for every show), but they each have to be dealt with individually when I'm building them to ensure the performance characteristics are correct. That's not uncommon. You wouldn't set up a Fender Twin with the same gain and tone setting your would for a Marshall or an Orange. The same thing applies to the models which are directly built from those amp's circuit designs. None of this is particularly difficult to understand or do, but you do have to have an approach and a system of working that accomplishes it. Fortunately the Helix provides ample tools and flexibility to do such things. The MOST important thing is to make sure your signal is appropriately gain staged so that it's not clipping and is at a consistent SIGNAL level with your other presets. This could easily account for why some of your patches sounded bad. If the FOH mixer was gain staged to optimize one preset and the next preset was at a different level one of two things will happen. It will either start clipping or engaging the limiter, or if you have a diligent sound man he'll keep adjusting the mixer preamp gain level on your incoming signal to stay consistent...then proceed to cut the tires on your car on the next break. Generally speaking most people have different ways of managing this. In all cases you want a consistent signal and volume level that doesn't affect the tone of the patch which most people tend to use the channel volume on the amp model as it doesn't affect the tone. I personally go beyond that and use a signal meter on my mixer channel to ensure the signal I'm sending to the mixer will only need to be gain staged on one preset and will be fine for all the others. I only do that once, when I'm building it and simply check it when I load up presets for a gig. I also detach my Helix master volume knob (big knob) from my XLR outputs in the global ins/outs and set my XLR outputs to Mic level signals. This does two things, it sends a signal to the mixing board that's consistent with all the other incoming signals from mic's or keyboards, and it allows me to separately manage my on stage volume level to my monitor using the Helix volume knob without affecting the FOH signal. It's also important to understand that when you detach the Helix volume knob from an output, it sends that signal at full strength as if the volume knob was on full. That's the actual recommended practice in the Helix manual, but that also means you need to be very sure about gain staging the signal levels you're sending in some manner, either with a mixer as I do or through a DAW. Some people get by with a simpler approach and often simply depend on their ears, but being a sound man myself for over 4 decades, I'm a little more cautious about making sure my output is bulletproof and is exactly the same as what I'm hearing on stage through my Yamaha DXR12, which is the same setup I use at home for dialing in my patches.

I've kind of monitored this from the sidelines and I have to say it's been rather amusing... From what I can tell, this bug may very well be some sort of memory leak that if left alone over a long period of time (looks like around 3 hours) may eventually break down the integrity of the signal chain in some way, but I still go back to what I first said. If it is a memory leak, the quick fix for it is to simply unload the preset then load it back as that would rebuild all memory being used by the preset. But what's absolutely amusing to me is using this bug for the excuse not to use it live. That's kind of like saying if my turn signal breaks in my car I won't drive it. Three hours is a very long time in term of stage performance. Heck, I'm changing presets about every 4 or 5 minutes during a show. I've had some long shows over the last 50 years of playing, but none that was three hours long with no break or no changes in presets. Of course if the workaround for the bug is to simply reload the preset, that takes about 2 seconds at best. I'm pretty sure in three hours one could find time to do such a thing. In fact, a typical break or intermission in a show is more than enough time to reboot the whole unit multiple times. All that aside, I'm not sure what the OP is looking for here. Let's assume Line 6 tells him that yes it's a bug. Does he think Line 6 will interrupt the development cycle and create a special release for him to put out a fix for it? If it is recognized as a bug (which it probably has been), it will be added to the items that need to be fixed in the next formal release as it would for any other company in the world. That's a potentially long time to have a piece of expensive gear sitting around doing nothing. Quite frankly, I don't know what the OP's real problem is, but I think it's safe to say it certainly isn't this "bug" that's keeping him from playing......

If it is a memory leak it would seem to me that simply switching to a new preset then back would eliminate the held memory since that re-initializes all of the blocks.

Without knowing which kind of DAW you're using, this type of behavior often relates to the buffer size being too small, or potentially to background processes on your PC interrupting the processing of your DAW.

In my way of thinking, since you're trying to go live on YouTube you need a live setup. The obvious answer to me is to simply incorporate a small mixer. essentially the same setup one would use in a traditional live setting. One channel for the backing track, one channel for the Helix and one channel for the mic. You should be able to find a small mixer for around $100 that could handle that setup and give you all the flexibility and control you need.

I was an early adopter as well and I can't say I've encountered any of the problems you're talking about on 2.82. Some of them I could write off as being situations in which you may have hit two buttons at the same time and thrown the Helix into a different mode such as with your preset names listed across your stomp buttons (in which case it can be solved easily by hitting your mode button to restore it. I'm not convinced your anomoles are related to 2.82 though. I use momentary switches all the time when I'm doing dual harmony leads and I've never experienced any problems at all. But you may be experiencing odd behaviors from just normal use such as having dirty contacts on some stomp buttons (which you can clean by holding the button down and turning it back and forth). If it were me I might try moving it to another stomp button and see if it makes a difference.

Gig levels is a relative term, but generally anything around 85 to 100 db is sufficient. I think of gig levels as stage volume levels rather than FOH levels coming our of the PA which hopefully are MUCH different levels for you. The settings I use with my DXR12 set to unity or mid-point on the gain knob and my Helix big knob volume at about 1 o'clock is probably around 90 db which doesn't seem to bother any of my neighbors. Many of them have commented they enjoy listening to me practice. But then I don't normally practice at night or early in the morning either.