DunedinDragon

I think the important factors in managing preset volumes have more to do with the signal that gets produced once it gets converted from digital to the analog world rather than the minutiae of what specific signal level is best within the Helix. Case in point, a preset at 75% on the Helix signal meters but being sent to some older analog mixing boards at Line level through the XLR outputs with the Helix volume knob set to 100% could very easily leave very little room for managing the trim on the signal level at the board where it really counts. The same could be true with a signal going from the 1/4" outputs to a powered speaker at line level with the Helix volume set wide open causing the speaker to be extraordinarily loud or even kicking in it's internal limiter when it's gain knob is set at unity or 12 o'clock. My general rule of thumb has been to try and keep my Helix meters at a reasonable level of maybe 60 to 70% and send my XLR signal at full Helix volume at Mic level to the board (which is where signal to noise levels really matter), and my 1/4" output set to line level with my Helix volume knob set to around 11 O'clock which allows a very reasonable volume on any of my FRFR speakers (Yamaha DXR12, EV ZLX-12P, QSC CP8) with their gain knobs set to 12 o'clock. That ensures ample volume with ample headroom to ensure they won't kick in their built-in limiters if I turn up the Helix volume a bit if I need to. But being a soundman myself, I can also admittedly be a bit obsessive about such things....

Is there a question in there somewhere? It might be easier if we were talking about speaker and amp technology post 1999 though.

Well, that wouldn't be "by my rationale", that would more precisely be "by your rationale". My rationale wouldn't allow me to make that assumption since I recognize I would be comparing AI ( which is based on a simulation of heuristic processes) and modeling (which is based on absolute measurements and mathematical calculations) and are not equally effective at what they do just because they might both be done on computers.

I would suggest you might want to play with the tracking and auto eq parameters on the Poly Capo and make it less rigid as I've found that can produce such artifacts depending on the tone being produced in the signal chain, the amount of change as well as the placement of where the Poly Capo sits in the signal chain.

And why would your subjective opinion be somehow better than the vast multitude of those of us who have been successfully using Helix for years now...some of which own or have owned other brands of modelers and profilers such as those you've mentioned and still prefer the Helix? You, of course are entitled to your subjective opinion, but that's yours and yours alone....which has no comparison at all to the objective process used on ALL modelers of measuring the inputs and outputs of the circuits in the actual amps in question and simulating those exact measurements in mathematical calculations via a DSP processor. Sorry, but I'll take the math every single time over any subjective opinion that's not mine.

I can't speak specifically for the QSC K.2 series, but I have a close cousin to it in a Yamaha DXR12. I've used it in a number of different ways, and in all cases it takes the place of a traditional amp. These type of speakers are all simply PA speakers and can be used as a dedicated floor monitor, or in the backline like a traditional amp, as a normal stage monitor or even as FOH speakers. In smaller, more limited situations where the PA was insufficient or unnecessary for sending the instruments through, I would normally place mine in the backline on a lower height pole to augment the PA which would only have vocals. I've used it in the same configuration on festival stages simply because it's more easily and consistently heard all across the stage by the rest of the band due to it's wide and consistent sound field compared to a traditional amp cabinet because as a PA speaker that's what it's designed to do. I currently use DXR12's as both stage monitors as well as FOH speakers with no amps on stage at all. Again, they're normal high quality PA speakers so they'll present a very precise and full representation of the sound coming from your Helix the same as any high quality PA speaker you'd have FOH. If you want to use them in stereo you just use two of them running direct from the Helix. I actually own six DXR12's. I use four of them (2 monitors, 2 FOH) with the band, I have one I keep at home which I use for dialing in and testing presets, practice and rehearsal, and I have one spare.

I would think those type of differences would be due to the difference in the actual circuitry of the amps involved. They're supposed to act differently because they're modeling circuitry which is different.

The only improvements might be some tighter or or more efficient programming as firmware doesn't improve anything in terms of the underlying capabilities of the DSP engine. The Helix models simply consist of mathematical simulations of the measured inputs and outputs of the real world circuits in a given amp which are computed by the DSP chip, nothing to do with audio. Audio simulation is all about sampling such as in the case of keyboards or electronic drums, not modeling. They may have, over time, figured out ways to do those computations more efficiently, but the end result of the computations would still have to match with the original circuit measurements in order to be an accurate model. But I would imagine a lot of that kind of stuff was done during what was referred to as the Helix CORE firmware updates a couple of years ago.

I'm just guessing here, but I suspect it may be that the Helix itself doesn't really have any direct knowledge about or access to a connected computer's hard drive to do any such operations like what you're describing. Once the computer hands over control to the Helix to do the update, it's all in the hands of the Helix. The reason I suspect that may be the case is because the Helix can't do it's own backup/restore/export/import without intervention from HX Edit which runs within the computer's OS. From everything I've observed the Helix pretty much seems to be in the same role as a printer might be that's attached to a PC and dependent upon the PC for file access. At best it can probably send alerts or messages to the host, but that's about all it can do.

I guess I've never wondered about why this is the case. I just figure that a firmware update is a fresh installation so everything will revert to a consistent state as defined by the base firmware. I guess I assumed that because, unlike a software update on a computer operating system, there is no real "scratch space" on the Helix to save the existing state of something in order to restore it after the update as there is on a PC with a hard drive. I'm sure it could be done I suppose, but it does present some potentially tough problems considering that the number and type of global parameters change with almost every update so that could present a lot of dicey problems for how one would handle it. I'm perfectly happy with the system of simply backing up and restoring things.

That's why you do a backup as directed BEFORE the upgrade. That way you can just restore your globals from the backup. You did follow those directions didn't you????

Whatever parameters you want the snapshot to control you have to press down the button and turn it so it gets bracketed. Once you've changed all the bracketed parameters you want to be retained in that snapshot you save it. Select the next snapshot and change them there and save.

On the contrary, I think you miss my point. HX Edit has no knowledge of what blocks (effects, amps, cabs, etc.) are available, no knowledge of the impact the inclusion of a block has on DSP, no understanding of what a snapshot is, or the difference between an amp and a fuzz box or a compressor, or an output block, no understanding of the rules of how many splits you can have on a signal chain, no knowledge of parameters, and the list goes on and on. It only knows to how to present things to the user that Helix tells it about and how to respond to the Helix's directions if it says OK or "that's not allowed". Without an actual Helix attached, it's going to be a picture of two signal chains that you can do nothing with. Every action it presents to you is an action that is controlled and authorized directly by the Helix unit. It is the pure definition of a "dumb terminal" program. The only "smarts" it has is it knows how to interact with a computer filing system for backups, imports and exports. But even then it needs a Helix to export from or import to. That's the entire extent of it's intellect...and that's why you get it for free. There is no requirement for you to have a guitar in your hands to build a preset. I rarely have a guitar in my hands when I lay out a preset. Once I've laid out a preset I pickup a guitar and start setting parameters. I don't even need HX Edit for that. Just the Helix itself.

You couldn't really backup to a Helix since the Helix doesn't have any general storage devices like a hard drive. The easiest way I've found is to network my PC and laptop. My studio Helix is on the PC and my live performance Helix is on the laptop. But when I export or backup I send it to a shared network drive on the PC, and that network drive is backed up nightly on the internet.

Just so you understand, you already have such a thing available to you..it's called Helix Native. Anything you design in Helix Native can be imported as a preset into your Helix unit. The problem with trying to do this with HX Edit is that HX Edit has no built-in intelligence or understanding of modeling at all. It's simply a pretty interface that uses the abilities and intelligence built into the Helix unit for everything it does, and that's why it's free. It would be simpler to make Helix Native work as a standalone program outside of a DAW. But you'd still have to pay for it since that's a pretty extensive piece of software.

I've always used presets on a song by song basis, and either stomps or snapshots within each preset depending on the complexity within each song. Because each preset only contains the amps, cabs and effects pertinent to one particular song it minimizes the DSP usage in each preset and allows me to easily take advantage of DSP intensive effects. It also helps minimize the number of footswitches required in most presets as I only need to assign footswitches to blocks that will need to change during the course of a particular song. Most importantly, in my case, It allows me to specifically craft the sound of the preset for each song which is important in my case because we cover a very wide range of styles and genres and I can organize setlists to minimize the number of instrument changes in a performance (different guitars, banjo, mandolin, resonator, etc). The disadvantage being I don't have the flexibility to easily move to a different out of order song at a moments notice should the need arise. Fortunately that's rarely the case as we always have our layout of songs pre-programmed for each performance. I have a library of around 300 presets that I keep archived on my PC's hard drive and import them as needed for each performance. Because all my presets are constantly archived they are backed up on a nightly basis so I never lose any presets. I used to have my Helix setup for 4 snaps and 4 stomps, but I've changed to all stomps recently because the of the flexibility now provided by the command center for intermixing snaps and stomps and MIDI commands anywhere I need them. Because I use less footswitches in each preset I've also been able to incorporate MIDI control of a BeatBuddy drum pedal into our setup. I have plenty of songs that don't require snapshots. I don't see that as any kind of waste as snapshots are just another tool to be used like any other tool when required by the song.

It is still a Stomp, which by definition has one DSP chip. Just 8 blocks, extra buttons and some addition i/o

DSP is allocated when you create a block and the amount is based on what kind of block it is, reverbs and amps use a lot compared to EQ and gain. What you do within that block as far as changing parameters doesn't change the DSP allocation. Once you change presets you start all over with a fresh allocation of DSP. The DSP used for global operations is pre-allocated so changes in global operations won't change that.

Ahhh...now I see what you're getting at. That's a good question and, just guessing here, it could simply come down to what kind of algorithm and computation they used to achieve the digital representation of amplification or db increase given that both are still working on digital representations and nothing is analog yet, and assuming all other things in your signal chain are equal.

It's as I was describing before. Whether it's the gain block or the output level block, they're both just increasing the amount of amplification without any regard to the amount of compression or distortion in the signal itself. Therefore, the more compressed the signal is, the more you would need to amplify it in order to get similar results with a cleaner, less compressed signal on a signal meter. But that's where your ears come into play. The very nature of the compressed signal, especially if it's tone has changed, will often stand out from the background mix even though it doesn't move the signal meter as much. You can see the results of this if you place a LA Studio Compressor at the end of your chain with a peak reduction around 7 or 8 and watch what happens to your signal meter when the Mix parameter is at 100 and when it's at 50 because the peaks start coming through. Bear in mind that all of the sound at the mixer is accumulative. A single guitar will not sound as loud to the human ear as a whole band playing at the same fader/signal levels. When you stop playing rhythm the overall background volume of the band drops, and you're replacing that drop with a lead signal. That lead signal is no longer competing with the whole band's sound so it very possibly could stand out with less boost than you think...especially if the tone is different. I only bring this stuff up because this is where the science of signal level and the practical nature of sound pressure level (SPL) diverge and why you have to use your ears rather than your eyes to get a fix on how that lead will sit in the mix. The bottom line is the behavior of the Helix is definitely in line with how signals are affected in the analog world, so there's nothing wrong with how it's being done. It's just one of those cases in which your eyes can deceive you but your ears won't.

I have quite few Ownhammer products, but there are just a few that I tend to turn to in heavier situations like you're describing. The MRBW BasketWeave 4x12 is a pretty consistent choice as it's what they call a Massively Multi-mic'd collection using up to 5 different mic's on the cabinet so you can get a pretty wide range of tones out of it. Although not OwnHammer I also really like the Celestion Redback for more modern heavier sounds.

I've re-read your posts about six times and I'm still struggling to understand what you're asking...but I kind of think I may understand. When comparing signal levels on your mixer, it takes a gain adjustment of +7.5 db on a heavier, compressed signal to achieve the signal level you get on a clean signal with just a minor amount of increase in the gain block. If that's what you're talking about that's quite normal because you're battling with compression which will naturally inhibit the signal level peaks. The gain block doesn't take that into account, it just increases the amplification by the amount specified. Your signal meter represents the actual value of how much it's been increased. This is where you'll have to rely more on your ears than your signal meter because even though the signal isn't increasing as much when the amplification is increased, the sound is actually "beefier" due to the compression which will make it naturally stand out. To be honest, I don't really rely on my signal meters to tell me how much increase I have on a signal like that, just my ears. Does it SOUND louder, because that will make it stand out. In this case in particular the change in tone is very often more important than the change in signal strength. The other part of that is really about band discipline. Just like when someone is singing, when a lead kicks in the band needs to pull back (get less busy/less aggressive) just a little to allow the lead or the voice to be more dominant in the mix. With a lead bear in mind this is a lot easier because the sound floor is reduced to a certain degree by your rhythm playing going missing in the mix while you replace it with a lead line.

THIS^^^ is exactly the thing I ponder most often when I read posts in this and other forums covering the Helix. Clearly given the more technical nature of modeling it constantly surprises me that so many people still avoid investigating and learning about all the technology pieces that the Helix depends upon in order to achieve a great performance. I've often wondered if learning these sorts of things is maybe just a generational thing of not having interest in it, or if it's just something that takes time and exposure. But certainly people are at great risk of not being able to take full advantage of this type of technology if they just absolve themselves of any responsibility to learn and master these technologies that allow the Helix to be the best it can be. I have to believe I'd be very wary of investing the amount of money one can invest in modeling if I knew I was going to be at the mercy of someone else knowing what to do in order to make it all work well.

When it comes to the trim/gain knob on your mixer, I don't think the position is all that meaningful other than it needs to be consistent when measuring the input level on the channel the Helix is coming into. Generally trim/gain is used to get the incoming signal where it needs to be. On a standalone mic, or mic'd cabinet that's the only way to adjust the input levels to get them to be consistent. In terms of the Helix I set my gain/trim knob on my QSC TM30 to unity or 12 o'clock. That makes it consistent when I'm measuring the input level from the Helix and forces me to set all my presets to be at a consistent level by adjusting them using the amp model channel volume in the preset. That works perfectly for recording or live performances, and it works consistently whether I'm going to my mixer or someone else's mixer. They simply set the gain/trim knob wherever they want my signal to be on any given preset and all my presets will be the same. The output level, or the mix relative to other channels is all done with the channel faders as it should be.

I think you misunderstood when I mentioned the clipping. I was referring to your Helix signal chain level, not your recording level. In other words, the signal level represented at bottom when you select the output block on the Helix. It may not be clipping as much as it is a limiter or compressor intermittently kicking in. Especially if your signal level is very hot.