DunedinDragon

It seems odd to me that someone with such a great dependency on the Helix for his music work every week would be so foolhardy as to immediately jump in and install ANY FW update without giving it sufficient time to prove itself. I notice he doesn't rant much about why it was so incumbent on him to do so. Heck, QSC just released a new FW updgrade for my TM30 digital mixer over a month ago and I'm still evaluating it's stability before I make the leap. You'd think someone so self-righteous would be more disciplined in his planning....

I think it might help you to understand how things are captured in a professional studio recording session or on a professional stage in pretty much every concert you've ever been to. In all cases you are NEVER hearing the amp. You are hearing an amp that's been captured, most typically, through a combination and mix of microphones. The reason for this is because an amp will sound different depending on where the listener is standing relative to the speaker cabinet. Therefore, to get the most representative sound from an amp in the studio or on a professional stage performance, different types of mics and placements of mics are used to get the best representation. The fact is, you can accomplish the very same thing using only the Helix and nothing else, and that's how most of us capture our recordings, and our live performances. The patches I use to capture my recordings are the exact same patches I use in live performance. The only difference is when recording I capture the output through the USB port on the Helix and on stage I capture the on stage sound from the 1/4" output into my Yamaha DXR12 stage monitor, and send the XLR output to the mixing board for the audience. I'm not trying to compare that sound to the sound of an amp, because that sound of the amp will vary depending on where I stand relative to the speaker. I will have the same sound no matter where I stand on stage and no matter where an audience member is seated. The "big" secret is knowing how to create a signal chain that's representative of the sound I want. That's not just the amp. That's the amp, the cabinet (or cabinets) I want to use, and the mix of microphones and how they are placed on the cabinets. That means understanding the differences in the cabinets, the differences in how different mics capture different sounds and what blends together the best, and how to place those mics to capture higher frequencies or lower frequencies. As complicated as that all sounds, it turns out to be relatively simple and is demonstrated quite well in Jason Sadites YouTube videos on creating great Helix tone. Instead of beating your head against the wall, you might be better off to simply watch some of his videos on the subject and try what he's demonstrating. If you want to just plug into an amp and play, that's simple and that's fine. But one way or another if you want the realistic sound of your amp on a recording or for your live audience you're still going to have to go through the same process of positioning and mixing physical mics....or just do it all through the Helix.

You do realize no one is going to see this wish list unless you post it in IdeaScare, right?

You go through the same process. Download the install program for HXEdit and run the install. This installs the appropriate drivers and version of the Line 6 Updater. Make sure you're not running HX Edit and run the Line 6 Updater and chose the 2.71 firmware.

I suppose it doesn't matter to the Helix how you power your active electronics as long as that power doesn't pass along to the Helix although I'm having a hard time envisioning what kind of DI box performs such a function and what kind of cable. DI boxes typically are used to isolate phantom power rather than provide it. But I suppose there's nothing limiting one from providing it if designed to do so.

That's where the question lies. You shouldn't have had to install any driver as the drivers needed by the Helix would have been installed when you ran the install for HX Edit.

You realize all bets are off if you didn't PRECISELY follow the sequence which is to: 1. Download the HX Edit 2.80 install app, then run the downloaded install app that will install HX Edit, the appropriate device drivers, and the correct version of Line 6 Updater. 2. After that you exit HX Edit and run the new downloaded version of Line 6 Updater which installs the firmware. Anything other than that and there's no telling what you have installed. You may want to just start over and make sure you follow the sequence exactly.

More importantly can you edit the post and say "Solved...because I didn't follow the update directions provided by Line 6"

How deep is the area you're standing in? It's entirely possible you could be reflecting sound from the wall behind you into a mic which creates a feedback cycle.

Feedback has nothing to do with electrical power or digital gear. It's purely an acoustic issue. You can aggravate the situation with effects like compressors and EQ that may cause resonation within an environment and lead to more feedback, but feedback is always caused the same way with the signal being re-fed into the source via mic's or pickups. Normally you would address such issues by "ringing out" the PA at sound check time to isolate what channels are feeding back and correcting for them. You do this in each environment because the acoustics are always different. Without more specific information about the gear you use and how you have things setup, it's hard to guess where your problems are although setting up on the floor, especially a hardwood floor, can cause the sound to be transmitted up through a mic stand and into a mic causing feedback if your amps or speakers are also in contact with the floor.

Actually, I think modelers in general provide a great laboratory environment for trying out new signal chain possibilities. Unfortunately my time tends to be spent more on getting things ready for the next gig most of the time so I personally don't have a lot of time left for exploration. Although I have no problem with exploring different possibilities, I also recognize that the combining different amps and layouts may help to inspire new sound ideas, the sonic differences in the design of amps tends to be fairly limited and I'm not sure that strange combinations is necessarily the secret to duplicating someone's tone. Over the years I've played Mississippi Queen through a fairly wide range of different gear, but ultimately what gives the song it's signature sound is a pretty simple combination of things that can be accomplished on a wide range of different amps and effects. Generally if you choose an amp that's in the same style or family of amps, with a similar guitar and with the right combination of speakers and mic'ing techniques and effects you can produce a decent representation of whatever sound you're trying to emulate.

If you thought the Stomp was great, wait till you see what the full floor or LT is capable of!!! This coming from another Grumpy Old Sound Guy that's been sitting mixing consoles since the 70's. The accuracy, detail and precision over the sound you're sending to the mixer is nothing short of revolutionary. Especially if you're coming from a perspective of a sound guy who knows all the ins and outs of crafting the sound.

You're not hearing things, you're hearing what you're supposed to hear. Bypassing an amp doesn't break the signal chain any more than bypassing or turning off an effect. The signal is always constant unless you intentionally turn it off by lowering the volume to 0 either on an amp, a volume pedal, or a gain, or even an effect.

I'm assuming you mean you're using Yamaha HS-5 studio monitors for your output. If so those should be pretty accurate as long as you have them positioned correctly. I have HS-7 studio monitors and the guidelines for these speakers is to have them placed at least 12-18 inches away from the wall in order for the bass reflex port to work correctly. All studio monitors require a some accurate speaker and listener positioning to get a good representation of the sound. Basically an equal triangle between the two speakers and the listener. Although that could have some impact on your sound, I suspect it lies more in the presets you may be using. It would help to know which amp models you've been using as some are very bassy. Other than that you might want to look at which mic's you're using on the cabinets. Ribbon mics will be more bassy, dynamic and condenser mics will be more toward mid range. Also the distance parameter on the mics can have a big impact on the tone, so reducing your mic distance will reduce the lower frequencies. And finally you can use the low cut parameter on either the cabs or separately with the Parametric EQ. Generally I keep my low cuts set to between 80 Hz to 120 Hz, but some amp models or tones require that I go as high as 140 to 150 Hz.

The brutal fact is, the sound you're getting out of a FRFR speaker is the sound your audience has been hearing forever from the FOH, regardless of the cabinet you were using. They weren't hearing the cabinet, they were hearing the mic'd cabinet. It's just that you're not used to hearing it. To your audience it's no surprise. That being said, there are differences in FRFR setups whether it's speakers or a cabinet like the Powercab, and generally speaking you get what you pay for. My philosophy is that it's pretty silly to spend the money we spend on sophisticated units like the Helix only to have them sound like crap because we tried to save a few bucks on the output device. That makes no sense to me. However, FRFR speakers and cabinets are also things that require a certain degree of technical know-how. This isn't 1995 any more and technology has moved on, so that's kind of expected. Woe be unto those that aren't willing to read their documentation nowadays....

The Powercab is an alternative FRFR approach butstill incorporates the cabinets or IRs. But it does come closer to the amp in the room sound to a degree. There's nothing wrong with your approach until it comes a time to go through a PA in which case you'll either have to modify your patch with a separate split off signal path that incorporates a mic and cab if you want to go direct to the mixer, or mic your cabinet with a real mic. You also sacrifice the consistency of your tone with the cabinet as the sound varies depending on where you or your audience are positioned relative to the cabinet simply due to the way cabinets work as compared to FRFR speakers.

There's nothing particularly hard to understand about cabinets, mics, and mic positioning. As with any advancement of capabilities you have to be willing to invest some time to learn the concepts in order to get the most out of it. You can't expect any piece of equipment to give you advanced capabilities for modeling a wide range of gear and setups without investing some intellectual energy in learning how to use it. That doesn't mean you can't get some usefulness out of the Helix by doing what you're doing. But you'll never achieve the flexibility one achieves once they grasp the full concept of modeling not only the amps and effects, but also the cabinets and mics which is a MAJOR part of the sound. This really has less to do with getting away from tube amps and more to do with putting the full capabilities of studio quality sound into your live performance. Every song you ever listened to either recorded or live was made possible by using an appropriate cabinet, mic, mic position and mic mix to precisely tune and craft the sound. So it's probably worth your time to invest in understanding it. Especially in these days of the internet when learning how to do these things is just a simple search on YouTube or Google.

Well, since the Helix itself has no built-in output method, it might be helpful if you gave us some information about what you're using for output since it may have something to do with your problem????

And oddly enough I have a 36 year old son that gets upset and critical of everyone when the world doesn't work the way HE thinks it should...I'm hoping he grows out of that too!!!!

I personally think passion is what keeps you on track regardless of what you do, and that applies to guitars as much as anything. The quality of guitars in general were just not very good when I started taking them up in the early 1960s. Even the high priced ones weren't always consistent in quality, much less the cheap japanese guitars sold by Sears and discount retailers after the Beatles invasion, which was the type I had. When I started playing in 6th grade I couldn't keep count of all my friends that were starting to play guitar as the music scene exploded, and only a handful of them had anything better than me. By 9th grade I was one of the few that had stuck with it, and by my senior year in high school I think I may have been the only one in my class still gigging in a band. I didn't have the best equipment, but I had the passion for music which got me a scholarship and kept me playing and performing into college and touring with a band after college for several years (with much better equipment by that time). While most of the other kids got into muscle cars and sports, I stuck with guitars and music because that was my passion...even with crappy gear.

You realize you don't have to have the same speakers for Helix practice as you do for normal everyday work on the PC right?? Just connect some studio monitors to the Helix and a pair of smaller consumer grade speakers for everyday work to the PC soundcard. The PC will handle the rest by selecting the right sound device.

There's a limit to how much beefy rock you can get out of any strat. That being said, I use simple Gen 4 noiseless pickups and I can get some pretty decent rock sounds in The Who, Stones, Tom Petty, or SRV type stuff...but I wouldn't be doing any Queen, AC/DC or Joe Walsh with it. For the beefier strat sounds I tend to turn to the Matchless Ch 1, Cartographer, HiWatt or Placater most of the time with either Celestion Redback IRs, OwnHammer Friedman IRs and occasionally the Ownhammer Basket Weave MMMC IRs, using a mix of MD421 and R121 mics.

I think you'll find most manuals for pro gear are going to be fairly lacking in specific recommendations given they typically are dealing with people that already have some knowledge about live sound. I will say QSC manuals tends to be at the far end of the spectrum in this regard. In term of general guidelines, most of these type of powered speakers are designed with two different channels so it can accommodate both an instrument as well as a vocal input, which are two different signal levels. Line level is the most commonly used as that's what is typically fed to these speakers from mixing boards and is the default output level from your Helix. As far as the gain knob, that's really a preamp for boosting or cutting the signal that goes to the amps in the speaker. It's the common convention to at least start with your gain at unity as that means nothing is being added to the signal nor taken away from the signal. Therefore if you need more volume you can control it from you Helix master volume knob which performs the same type of function as the gain knob on the speaker. There's nothing particularly wrong with adjusting the gain on the speaker other than to invite pushing the speaker too hard and inviting the limiter to kick in, particularly if the signal coming into the speaker is too hot. I personally own quite a number of powered speakers for different uses as monitors or FOH speakers. In all cases (other than my QSC KLA line array speakers) I leave my gain knob on the speaker set to unity and control the signal level from the mixing board, or from the Helix master volume if I'm going direct to it.

I'm thinking this all started with participation trophies....you were entitled to one just for showing up. Sound familiar???

I set mine at 12 o'clock or 0db which is unity and send the signal 1/4" line level output into Input A. Never had any problems at all. If you go into input B make sure the Mic Boost is turned off if you're sending line level input as Mic Boos is necessary for Mic level signals. Input B can be used for either Mic or Line input. Input A is always Line level. Both inputs can take XLR or 1/4", but I see no particular advantage to using XLR if it's just a short run on stage. The Sig light is always on (green) if a detectable signal is coming in. The Limiter light is the one that would light up if your signal is too high. All of this is outlined on page 9 of the documentation. If you don't have a copy I've included a PDF copy of it below. QSC_CP_Speaker_Series_User_Manual.pdf