DunedinDragon

You do get the effect of the amp model through the modeler. But the Helix amp model is just the amp itself, not the cabinet as if you were just using an amp head. Beyond just the modeled amp head, you still need to have a modeled cabinet selected along with how you want it mic'd (or alternatively an IR) in order to give you a fully formed amp and cabinet model to produce a fully formed model of a real world setup. The Helix provides a signal output that contains the results of the modeled amp and cabinet. In your case that signal is sent through the Helix internal headphone amp and you listen to it through your Bose headphones. If you go out through the 1/4" outputs or the XLR outputs you will simply get the un-amplified signal containing the modeled amp and it's selected cab model. That then needs to go into some form of real world amplifier and speaker. There are many ways to do this but ultimately the Helix doesn't contain a real world amplifier. It's providing your modeled signal that still has to be amplified in a traditional sense in the real world.

The first thing that pops out at me is where do you have your Helix master volume set (big knob on the floor unit)? That kind of determines the output of all of your presets. Bear in mind there's no guarantee that any preset you might download or that comes with the Helix will be built with a specific master volume setting, so you may have to adjust each amp's channel volume so they match up appropriately with YOUR specific master volume setting. Where people set their master volume is up to each person. Line 6 recommends running it full open, but that's not practical for most people. For myself I keep mine at around 3 o'clock when developing patches and around 12 o'clock when I'm playing through my FRFR monitor on stage. The main thing is to be consistent so your patches can be consistent. In my experience, the higher you set the master volume on the Helix, the more consistent the amp settings will be with their real world counterparts for achieving a given volume. As far as IRs, those are just an alternative to the stock cabinets in the Helix. Typically you'll use one or the other, but in both cases it's just a specific cabinet with a specific set of mics positioned in specific locations on the cabinet. Stock cabinets or IRs can be used on any amp block and will affect the overall sound of that amp in the same way as different cabinets, speakers, and mics do in the real world. They really don't affect things like sustain which would be controlled by the amp gain or potentially master volume on whatever amp you're using. They only affect the overall tone of the amp.

Well the good news is that because Helix Core was an integral part of this update, they got a real life field test of updating all the platforms simultaneously in one code base. Looks like it worked out for them.

That's a pretty bare bones powered speaker so I'm not sure how indicative it will be of modern FRFR setups. It appears to have more in common with late 1990's and early 2000's models of powered speakers in that it uses a simple EQ knobds rather than a pre-programmed DSP driven design tuned for a flat response, and at only 300 watts it's pretty anemic. It's a very good price though, so as far as an entry level it might be okay.

As cruisinon2 points out above, there's a world of difference between an integrated amp setup and a component setup with a modeler separate from the output which is far more like what you would experience in a studio or professional concert setup. There are just more things that you get control over that you can't control on an integrated amp. For example, from your description it's not clear what kind of amp you're using which can make a lot of difference in your overall tone, but I'm assuming you're using one of the several different high gain amps either from the Marshall family or one of the others. There's nothing wrong with the selection of the 4x12 V30 cab, the question becomes how do you want to translate that sound because even on an integrated amp, if you place microphones on it to send to the FOH or to a DAW, your mic choice, and especially their placement, is going to be critical in how your sound comes out. Just like on any integrated amp and cabinet, your sound will vary to your own ears as you move off-axis from the center of the speaker cabinet. With microphones it's that much more variance based on not only what mic you use (all of which have different tonal characteristics) but where they are placed relative to the cap or cone edge. This is one of the reasons you'll find a lot of people such as Jason Sadites using dual cabs so he can combine microphones such as a dynamic like the 421 and a ribbon like a 121 at different locations and mix to get a better overall fuller tone. This is the most natural way to get a good tone which can then be tweaked to taste with slight adjustments to the EQ as you would in the studio. The next component you have to be concerned about which is a given in an integrated amp setup is your FRFR, but FRFR can mean a lot of different things to different people ranging from a FRFR cabinet, to studio monitors, to a PA style speaker, all of which also can have tonal implications. For example, studio monitors tend to require fairly precise arrangement between where the speakers are placed and where the listener sits to get a good overall feel for the tone, whereas PA style speakers such as a Yamaha or QSC speaker will pretty much sound the same within a very wide horizontal range if the speaker is placed upright on a pole, but a very narrow range if placed in a floor monitor position. Additionally if placed on the floor there will usually be problems with what's called "bass coupling" which is the low frequencies building up due to the proximity of the speaker with the floor. Some FRFR PA style speakers provide DSP contouring controls that allow you to adjust the speaker based on the placement of the speaker and what it's being used for. Clearly it's much easier and simpler to account for most of these things with an integrated amp, but you then lose the opportunity to precisely manage the components for your use. If managing all of these things isn't something you want to be burdened with then maybe you'd be better of with a simple integrated amp. I personally use the Helix and a Yamaha DXR12 for my stage sound, but I also have well over 200 presets to precisely control my sound for each song. However, the rhythm guitar player in my band wants a very simple, pre-adjusted sound that doesn't require much thought or adjustment, so he uses a Spider V with about 4 different sounds dialed in and he's a happy camper.

I've personally used a Sescom phantom power blocker for years with no issue. I just plug it into the XLR output of the Helix and never have to worry about phantom power. I choose to go this way in order to maintain a separating between the volume of the stage output (1/4") and the volume of the output I send to the mixer (XLR).

I'm pretty sure those of us that have been around a while remember at least one other release that caused this kind of uproar with upgrading back around three years ago. To Line6's credit they have streamlined and simplified the installation process a few releases back which is what we're using now which is TONS better than what we used to go through. Clearly it's not bulletproof and could be improved. But like the release 3 years ago that introduced Snapshots and a number of other key improvements, this appears to be a pretty complex release as well with the re-engineering of the Helix core which requires a much more complex process as it has to reformat both your existing presets as well as globals and backup formats. I'm pretty sure going forward that the process itself will likely have less problems. That being said, there are some ongoing problems with the Helix that I hope will eventually be addressed such as USB connectivity which is still a big part of the problems in both upgrading and interaction with HXEdit. To be honest I'm not completely sure if there isn't an inherent limitation in the hardware that might be the issue, but IMO until that becomes more bulletproof, there are still going to be firmware installation issues with USB dropouts or malfunctions. As for myself I'm waiting for the smoke to clear before I go about installing the update, and having been through three years of pretty much flawless updates I expect this one to go well when I decide to do it. Some of that is due to experience with these updates but a lot of it has to do with keeping my PC and my laptop in top running order with my USB and other operating system elements operating predictably when called upon for something like this. Reading through many of the complaints it's clear to me that's not the case with a lot of the users. Quite frankly if this were my software product I'd probably invest some development time on a utility that can be pre-run from the PC or Mac to determine the state of all the communications and connections necessary as a first step and diagnosing problems before proceeding with the upgrade so users can go about getting their systems stable before attempting an update. This would also be quite useful after the fact as a diagnostic if they're having HXEdit issues or problems with their patches or interface to their DAW due to version problems between the FW and drivers.

Chances are good that back in the day he just switched to the neck pickup to get the sound you're hearing during that segment. Sometimes simple is best for those that remember what it was like before we had all the electronics and effects.

The fact is, if you can't get it right following someone else's instructions for EQ or anything else because: 1) Everyone is playing different guitars with different tonal characteristics. 2) Everyone is using a wide range of output devices, in a variety of settings with different band members and instruments. 3) Rarely will a single tone work for all your songs unless you're a big star and known for a specific sound...which you aren't, or we would have heard of you. 4) Personal technique is likely a big part of the tone you may be chasing, and unless you perfectly emulate the person's technique the sound you're chasing will always be out of your reach. And there's probably many more reasons. If you've been playing for any fairly extended period of time you should know that you can't just "buy" your way to great tone with a Helix, Fractal, or Kemper or any amp or set of pedals. You first have to develop a good ear and then match that with the skills required for your personal setup to get something that works for you. The biggest missing link in all of this is the skills and abilities of the human playing the guitar. Regardless of what the popular sentiment is, you can't "buy" your way around that limitation. You either invest in developing your ear and your skills, or just learn to live with the tone you're able to cobble together.

There's so many things missing in your description it's not surprising your not getting the results you want. You didn't tell us what kind of output system you're using, or how you mic'd the V30 cab (mic types, placements) which all make an enormous difference in how your patch sounds and responds. Since you didn't specify I can only assume you don't think those things matter that much.

IRs aren't saved as part of a Bundle, but they are saved as part of a Backup. Hopefully you've got a backup somewhere that contains the IRs so that can be restored.

I'm not sure there are any quantum leaps coming in the underlying technology used in the Helix. The only gain would be to add another DSP chip like Fractal did, but that just gives you the same result but with the ability to have more blocks, and spend considerably more on the unit. Even with a quantum leap in DSP chip processing power, I'm not sure that necessarily equates to higher definition models. It would mean having to rebuild the existing models, but for what purpose? It seems to me that DSP processing power has reached a plateau in which there's not a whole lot more to be gained quality-wise for the increase in cost.

Make sure you aren't running HXEdit when you run the Line6 Updater and that your Helix is up and running.

Yeah, that's making more sense now with the +3db gain block. When it comes to leads, that's one area where I heavily depend on my ears more than on the signal level or an arbitrary boost. That's mainly because leads, by their nature, tend to have characteristics that already separate them from the mix by their perceived volume. In other words I usually depend as much on a change in tonality rather than just a boost in volume to separate them from the mix, mostly by using an effect like a Minotaur, Screamer, Teemah, or a compressor etc. Although that generally works quite well, I also have to admit I don't get it exactly right all the time and have to adjust a bit when I get with the band. But once I make that adjustment and save it, it's good from there on whenever I use that preset again on whatever FOH system I'm playing through.

Not to hijack the thread but that's a very odd situation you're having. Normally changing the trim or gain is simply adjusting the incoming signal so that it pairs up with everyone else's incoming signal via the preamp on the channel. If you're gain staged at 0.0 and your lead is +2db, it should always be +2db over unity as it passes through the preamp to the faders. It shouldn't be an exponential rise, just a difference in the rate at which different preamps boost or reduce the base signal, but changes in the incoming signal should still be linear boosts and would indicate that correctly at gain staging time. The actual volume level for where the guitar sits in the mix is managed by the faders, so if your fader is sitting at maybe -6, your boosted signal should still only be +2db over that. Unless there's a difference in the FOH speakers or how they have the FOH speakers tuned that might accentuate certain frequencies and create a perceived boost in the signal. That's just really odd. I've never encountered that.

My approach is going to be different than most, but that's because I have almost as much time working as a sound engineer as I do being a musician, so it's somewhat of a combined perspective that's reflective of both. My typical stage system consists of the Yamaha DXR12 as my stage monitor positioned behind me in a backline setup on a half height pole. My Helix is setup so that I feed the DXR12 from the 1/4" output of the Helix, and feed the FOH from the XLR output. I've configured the Helix volume control so that it is only affecting the 1/4" output, Therefore my feed to the FOH is at full volume from the Helix and is gain staged appropriately at the mixing board. This allows me to control my stage blend with the rest of the band separate from the FOH blend managed by the mixing board. Generally speaking I dial in my presets by monitoring the signal output of the 1/4" with my Helix volume set at 3 o'clock. This is the loudest stage volume I could ever imagine needing on stage and I adjust the output of my presets and snapshots by adjusting the channel volumes on my amp models so that the signal level as measured on the mixing board lights is about -8db. This is quite a bit louder than what I'll ever use on stage since in live performance where my Helix volume will generally set at around 12 o'clock. But this ensures I have a signal level loud enough when dialing in my patches to get the appropriate responses from other blocks in my signal chain, particularly ones like reverbs, delays and compressors. It's also at a level that ensures the amp models themselves respond in a fashion similar to their real life counterparts. The point of all of this is to get as close as possible to the kind of signal and behaviors being sent to the FOH via the XLR output. Once that's all dialed in, I adjust by ear for perceived loudness that's affected by the tone of clean, crunchy and lead settings so they will blend correctly with the rest of the mix with the instruments and require little adjustment on the part of the sound guy at the mixing board. All he should have to do is gain stage my signal off of any of my presets to be at unity on the mixing board, then adjust my volume level via my channel fader to mix correctly with the other instruments and voices and sit correctly within the mix. Once I'm gain staged and mixed at the mixing board I can pretty much be hands off by the sound man for the rest of the performance. This process has worked perfectly week after week for a long time and across a very wide range of music genres from rock, jazz, blues, r&b, country, funk, rockabilly, etc. and across well over 200 presets. I've attached a picture of my home setup.

Quite honestly, a good portion of the problems you seem to be having appear to me to be more related to the update process not being executed correctly and the rebuilding of the presets. There are some very specific issues that may be related to what you've experienced but they seem to als0 be mixed in with some that might be fixed by re-doing the update process. At this point however, I'd hold off and maybe just roll back to 2.71 until they get everything ironed out. I agree with Datacommando. I depend on my Helix too much to just jump into an update without fully testing it. A live gig is not a good place to discover problems. I'll upgrade eventually once I'm confident everything's stable.

This has nothing to do with the update. This has to do with a few units which had soldering problems on the jog wheel and has been a common problem for those units. One of my Helix units had this problem about three years ago. I sent it to Line 6 on Monday morning and got it back Thursday afternoon fixed.

It's kind of like the old inside joke among programmers when JAVA was a big thing. We thought the motto should be "write once, debug everywhere".....

If you are planning to do any recording at all I would SERIOUSLY look at the floor as it's a TON easier to work with in the studio than is the LT simply due to all the additional inputs and the headphone volume control. The LT just requires you to come up with too many workarounds IMO.

What I would suggest is going to YouTube and watch some of Jason Sadites videos on dialing in great tone out of the Helix. You'll find plenty of exceptional ideas for getting any kind of tone you want. However, you need to be aware that two things will limit you from getting an "all around" great tone for anything from Led Zep to Eric Clapton and those are the type of output device you use, and the type of guitar you use. One of the key reasons many of us use flat response cabinets and speakers is so we can get an accurate representation of the tone we're dialing in because we eliminate the coloration of an external traditional cabinet. The other thing that will limit you is the guitar itself. For example getting an authentic representation of an Eric Clapton tone from his Cream days will be hard to achieve on a Tele or Strat because he was using a Les Paul or a 335 at the time. Not that you can't get a passable representative tone, but the limitation is on the tone differences of the different types of guitar and their pickups, and those aren't things that the Helix can have any affect over. The other limitation is that of the Stomp. You simply don't have enough blocks or DSP to achieve everything in one preset. For example, Clapton's tone on "Sunshine of Your Love" is from a Marshall amp, "Bell Bottom Blues" from the Layla album I believe is on a Fender Bassman, and his "Cocaine" tone came from a HiWatt. That's three amps which would most likely use up all of your DSP in one preset. So with the Stomp it might be more practical to do those different tones on three different presets. What I would suggest is using some of the ideas from the Jason Sadites videos to create a good classic marshall rock preset, and a blues preset (both clean and dirty). And you'll probably find other generic type presets that might work for you such as a Punk rock preset, or even a chunky rock preset for things like The Who, The Rolling Stones or the Doobie Brothers. Those are all very doable on the Stomp.

I'm not sure if you mean your high and low cuts are at 5khz and 90hz, but if that still sounds harsh I suspect it may be more related to the type of mic you're using on the cab and it's placement, or even the cab selection itself. It would help to know what cab you're using and which mic or mics you have selected and how you have them placed. Most commonly if you want a more natural sound you're going to want to mix mics to get the best tone just like they do in most recording studios and on most concert stages. You do that by using a dual cab block if you're using Helix cabs and mix different mics like a dynamic mic and a ribbon mic. Typically the dynamic mic is positioned further from the center to bring out more low frequencies and limit the highs, and the ribbon mic is positioned closer to bring out more highs and reduce low frequencies, but it's all a matter of what you're looking for in your tone. I personally use IRs that are a mix of a MD421 dynamic mic and a R121 ribbon mic in various arrangements on different types of cabinets and speakers. Sometimes I even use IRs that are a mix of up to 5 different mics to get the sound I want going through my Yamaha DXR12. With the right placements my typical high cuts are in the 8-9 kHz range and low cuts are between 80 - 130 Hz using the parametric EQ, all fairly dependent on the type of guitar I'm using. The Helix speaker block cuts are more gradual in their rolloffs than are the ones on the parametric EQ.

Here's a good video for dialing acoustic tones from Jason Sadites. Some very useful ideas here. Jason Sadites dialing in acoustic tone

I think 1 and 2 may be the same problem with some instability in global settings as clearing global settings is a known requirement for this upgrade and is in the release notes, but that one seems be limited to LT for some reason but I could be wrong. It could also potentially be due to people not restoring their global settings from their backup as prescribed in the release notes. It may be too early to say whether broken presets is really a bug or a problem with people not following the correct procedures for restoring such things. I think that one will require more work to find out what situations cause it and if it's a consistent pattern. The one that CLEARLY seems to be a problem and easily replicated appears to be the problem with the LA compressor placed at the end of the signal chain intermittently losing volume.

Well that's the point and should be a good lesson learned. Music is my only business as well and because of the critical nature of the Helix in that business I don't take chances with anything that I don't have control over. That's what pros do whether it's music, banking, or retail. You don't ever jump blindly into anything that could affect the critical systems in your business without having first researched, prepared and planned. There was nothing so critical in 2.8 that would require me to do the upgrade, so why chance it? Especially when you're on tour. It's not like you're really going to have a lot of time to experiment with incorporating the new features until the tour is finished, right? Maybe folks should maybe consider such things when they start whining about when the next upgrade will be.