DunedinDragon

I'm happy for Line 6 in that I think this fills a hole in their line of products. It's not something I would be terribly interested in though as I'm not terribly convinced it would give me an accurate feel for what's coming through the FOH given the difference in design between this and most FOH speakers. Nor do I really care much about an amp in the room feel. I could see how it would be very attractive for some users though.

You realize by the time someone looks it up and responds to you, you could have looked it up yourself in the manual, right???

Better than going to an unofficial pusher who charge $1000, maybe you should try the official ones with the official price....like Sweetwater or Musician's Friend. I'm assuming since you found your way to this forum, you likely have internet access I'm betting... ;)

In this case you're using a standard cabinet I assume. That being the case, were you also using cab/IRs and mics in the preset? If so, it wouldn't surprise me there was a lot of mess there.

As I said before, you do this by doing the following: 1. Hit the Menu button on the Helix 2 . Push down the button underneath Bypass Assign 3. Use the joystick to move over and highlight the compressor 4. When you do that you'll see it's assigned to a specific footswitch number. Turn the dial underneath that counterclockwise until it reads "none" 5. Proceed on with your life. It works like that every time and will always work no matter how many times you have to have someone to repeat those instruction to you.....

Select the block you want to turn off. Hit the menu and go to bypass assignments. You'll see the assignment of that block to the specific footswitch. Just turn the footswitch selector until is says "none".

Pre setting and saving the tempo with the preset is the way to go. Just change to the next song and the correct tempo will be flashing at you.

I can't say I've noticed any appreciable pause in my Sonar Platinum when I arm or record. It seems about the same as when I was using my MOTU ultraLIte MK3.

I never have any latency problems with my Sonar Platinum, but then I only monitor the direct signal when I'm recording while monitoring the other recorded tracks through the Helix. I don't see any real advantage to monitoring the armed track while recording. I suspect this has more to do with the behaviors of your DAWs than it does the Helix. Reaper is pretty tiny in comparison to Sonar Platinum, so you need a lot more computer horsepower if you want to get rid of the lag in monitoring your track while recording.

Nothing to it really, especially with a SM57 since it doesn't require phantom power which has to be setup up in global settings. The most important thing is don't plug and unplug the mic while the unit is on as it can cause an electrical surge that might affect your microphone port. Just plug the mic into the Mic input and turn on the unit. On a blank preset select the input block and change it from 'Multi' to 'Mic'. From there I normally add a Studio Tube Pre from the Preamps and whatever else you want in the signal chain. It outputs in the same was as any other input device.

I'm sure it's just related to unequal grounding when the Helix is off creating a ground loop through the 1/4" connection. You wouldn't hear this on a HD500X unit because it doesn't have it's own power supply as Helix does. I always carry a two outlet surge protected strip with USB ports in my gig bag which I use to power my Helix, my FRFR speaker, and my G10 receiver. That way I plug everything into the one strip and power everything on on at once after I've made all the connections and there's no ground loop.

I use it all the time and it's never been a problem for me. Probably a whole lot easier and quicker to just adapt your skills to work with what's there than to wait for a mod to fix it for you.

As with any product, the chances of anything being added will be based on demand for the feature. That's one of the main reasons Line 6 uses IdeaScale as a way to measure how much demand there is for a given feature. Given that the vast majority of customers of the Helix are more interested in guitar related features than vocal features, I'm not sure this would be an area that would generate a lot of interest. But you never know. Submit it to IdeaScale and see how the users vote on it.

I too am apparently unworthy of such things....but I promise I won't take it personally... ;)

Not something I would ever consider doing, but I suppose if you have an appropriate electronic engineering background so you can do the necessary calculations for heat dissipation and component placement you might get away with it and not end up with a very expensive smoke bomb..... ;)

As mentioned above, the default output setting of 'multi' on the output block sends the same signal out of both the 1/4" and XLR outputs. But I suspect from your last statements quoted above is what you're talking about is as much signal strength as it is what's in the signal. It's probably best to understand line levels as they pertain to powered speakers and the mixing boards. In general powered speakers use LINE level as their default signal. Many powered speakers come with the option of setting the input to either MIC or LINE level signals, but the MIC level is generally to accommodate people that want to simply plug a microphone directly into the speaker without a mixing board. Signals that are sent from a non-amplified mixing board to a powered speaker are sent at LINE levels, so that's probably how you should send your signal to the FRFR speaker from the Helix, and that is the default setting for the 1/4" out in global setting outputs. On the other hand the XLR inputs on a mixing board are generally defaulted to MIC level signals, although this can sometimes be changed on some boards. The Helix global XLR outputs is defaulted (I believe) to LINE level signals. This would be fine if you were to use XLR to go to your FRFR speaker. But going directly to the board can cause some problems from some boards that aren't equipped to handle that high of a signal through their XLR inputs. I always set my global XLR output to MIC level as that is what mixing boards normally expect. As mentioned above you can also set the Volume Knob Controls parameter in the global outputs to 1/4" rather than the default value of 1/4" + XLR. This means that the Helix Volume knob will only affect the signal level being sent from the 1/4" outputs, and the XLR outputs will always send a full signal as if the volume knob were turned fully to the right. This isn't a problem when going to a mixing board because the first thing that gets adjusted on the mixing board is the gain or trim knob on the mixing board's preamp to adjust that input to be consistent with all other XLR inputs on the board in process known as "gain staging". With your global settings as specified above you'll be able to freely adjust the volume level of your stage FRFRs connected to your 1/4" outputs while not affecting the XLR signals going to the board and driving the sound man crazy with the changes. to his inputs.

Glad that works for you. It's a great little trick. I don't use it all the time, but often when I just can't seem to get the balance I want between upper and lower frequencies it's just the medicine to fix it.

I never do. Ever since the big architectural re-write on 2.20 I think, none of the old process really appears to be necessary. It just works as long as you do the process correctly.

It's probably going to be a fairly big leap after 8 years because a lot has changed in that time, and Helix is kind of at the leading edge of those changes, so just be patient and willing to learn. Although you wouldn't necessarily need a powered FRFR style speaker at this point, it's probably worth going that route in order to keep things simple initially. I personally believe this is one area that pays off in spending a bit of money rather than going super cheap. There really is a difference. But you can get a pretty good speaker in the $400 range nowdays that will work well at home working with the Helix as well as jamming with friends. I personally use Yamaha DXR12's as my preferred speaker which are fairly pricey, but worth it to me. However I recently bought an ElectroVoice ZLX12P that compared quite well with the DXR12 but priced around $360. So the price/performance has been getting more reasonable in the last year or so. The ZLX is a bit less wattage than the DXR, but more than enough for live performance. Many folks around here use them quite a bit. As far as hooking things up, I prefer to go 1/4" out to my stage monitor and reserve my XLR out for going to the PA (using a phantom power blocker on the cable). It just makes things simpler and more consistent. Just make sure you're using LINE level on the 1/4" out and set the speaker to the same. Another consideration on setup is if you position these type of speakers as floor monitors there is a good possibility you will get a bass biasing effect which makes them sound a bit bassy. So you want to make sure you have the speaker's DSP options set for a monitor to correct for that. A lot of the lower priced speakers don't have these type of options so make sure you check when you're shopping. In my case I set my monitor speaker on a half height pole mount so I can position it in a normal backline setup the same as I would an amp. The value to doing it this way versus a floor monitor in front of you is that it's easier for the rest of the band to mix with your sound as it will be consistent with a normal backline. It's also useful in those situation in which you're playing in a situation where you aren't running the instruments through a PA. By having it on the backline it projects in the same way as a PA speaker because that's essentially what it is. I've included a link to the half height speaker stand if you're interested. Good luck... Half height speaker stand from Amazon

Here's what I do. I put in my amp and cab/IR blocks in place, then almost at the very end of the signal chain I place a parametric EQ block. The reason I use a parametric EQ block is the slopes on the high and low cuts are much steeper than the ones on the cab or IR blocks so it makes it easier for me to hear and make adjustments. I typically start high cuts up around 12khz and work my way down until I hear what I'm after. I then work my way up on the low cut around 60Hz till I get the tightness I want on the bass. One other thing I now commonly do is the trick Jason Sadites showed in his video of doing a crossover split using two gain blocks to adjust overall levels of lows and highs. The way that's done is you add two simple gain blocks and then create a split with one on the top path and one on the lower path. Change the split type to a crossover split and I'll normally specify around 650Hz to start. This allows you to fine tune the overall sound of the lows and highs very effectively by simply adjusting the upper and lower gain blocks up or down as well as where the crossover split frequency starts. Very cool trick.

First, that chart which I've seen is somewhat a general chart of frequency ranges, but there are some other considerations. There are overtones and harmonics that come into play as well so the range can extend quite a bit byond that range at times. Also bear in mind when you look at a frequency response chart for a traditional guitar speaker, the frequency range also extends beyond that, but not all of the frequencies are projected evenly across the spectrum. The fizziness you hear would also be heard were you to put your ear up next to the center cap on most of these speakers (which I wouldn't advise doing), so that fizziness can even be heard through any FOH speaker system (which are all pretty much FRFR speakers) if you mic the center cap of a speaker. Just as the fizziness disappears if you mic further out on the cone structure with the added effect that lower frequencies will get accentuated. So even with a traditional speaker you can get rid of fizziness at the cost of a flubby bass. All of these same artifacts are duplicated in the Helix with cabs, IRs, microphones, and mic placements. The reason FRFR speakers use horns is the same reason studio FRFR speakers use tweeters. It enables the speaker to have a flatter response across the entire range of frequencies it covers and treat them all equally unlike a simple speaker. The reason for a horn in a powered FRFR speakers like an Alto, or Yamaha, or QSC is because it's designed for projection over long distances because they are, in effect, the same thing as a FOH PA speaker. They do this by conserving energy by maintaining a very controlled sound cone so that energy doesn't get wasted by sending it up into the ceiling or down into the floor, nor does it get leaked outside of the cabinet enclosure. Also higher frequencies die out faster across distances than do lower frequencies. The horn is a pretty important part in all of this. The high and low cuts are dependent on MANY factors. I have several presets that have no high or low cuts depending on the style of music, the guitar/pickups I use, the amp, cabs/IRs, mic, mic placement and mic mixes. Unless you have a pretty good understanding of speaker behaviors and particularly microphone differences, their placement and mixes, it's sometimes easier for people to compensate for such things by simply using high and low cuts. The fact is, those type of high and low cuts have been used in commercial studio recordings as far back as I can remember into the 60s and 70s. That's because even a home stereo system or car stereo will reproduce highs and lows better than what your ears would hear standing next to an amp, and those recordings are made by mic'ing a speaker, so all of the same physics apply.

One of the most neutral amps in the stable, probably because that's what it's always been known for, is the HiWatt. I've used it on a fairly wide range of genre's and tones because of it's reputation and performance. It was hugely popular in the 70s and 80s due to it's range of tone and capacity for working with all sorts of pedals so it may be a good base to start with.

Well...by my standards your tone in the link sounds pretty harsh and piercing through a pair of Yamaha HS7 studio monitors. Pretty much the same through a pair of Beyerdynamic DT770 80ohm headsets. So if it sounds like too much or even adequate mid range to you it's apparently an artifact of what you're monitoring through because you've corrected out mids WAY too much. If I were to try and simulate the harshness in that tone on my gear I'd probably have to use an IR with something like an SM57 positioned right at the cap edge, as well as probably slice out some mids on either the amp or a parametric EQ.

I'm still struggling to understand what you're describing. By 'default EQ of an amp and cab' do you mean the setup of the defaults on an amp+cab model? If so, that may be true as I've never used an amp+cab model. I only use separate amp and cab/dual cab or IR blocks so I may not have experienced what you're describing since the cab/IR block does a lot of the initial EQ'ing of the sound so I really don't experience the 'eq smiley face' which I assume means higher peaks on the low and high frequencies and a lower valley on the mids.

I'm not sure exactly what you're referring to as a "Line 6 cursed" sound missing mid-range. I get precisely the sound I design in the preset I'm playing through. It's only missing mid-range if the design of my signal chain is such that it's missing mid-range. Maybe what you're referring to is what some people reference as the "amp in the room" sound you get from a real, physical stage amp. If that's the sound you want, send your Helix through the effects return of a real amp. Be advised, however, that once you actually mic that stage amp for the PA or for a recording the sound your audience will be hearing will NOT be the amp in the room sound. It will be the amp being mic'd and will be dependent upon the mic used and it's placement on the speakers. No one other than the person standing near the amp in the room will ever hear the amp in the room sound. Recorded or live through the FOH they will hear the mic'd amp in the room sound. As for those of us using the HELIX with a FRFR style speaker, we're interested in getting a studio or live sound...the same sound our audience will be hearing. And it's only missing mids if that's what we dialed into the preset. The sound is entirely dependent on the expertise of the person building it. That being said, I would NEVER use the global EQ for anything other than correcting for the acoustics in the room, and generally not even that if I were going through a PA. I would depend on the sound man to correct for the acoustics of the room on every channel, not just mine. There is no common or global EQ setup that can work for every possible combination of guitar, pickup, amp, cab/ir, or mic'ing scenario because they will all have different EQ needs. The factors that affect your sound aren't dependent on the Helix, they're dependent upon the guitar being used, it's pickups, the amp model you select, the cab or IR you select, the mic, mic mixture and mic placements you select and any other specific EQ corrections you might apply in the signal chain. The limit isn't the Helix, it's in the expertise of the person designing the patch on the Helix.