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Showing content with the highest reputation since 11/29/2019 in all areas

  1. 7 points
    A fine, impassioned and informative rant, but paragraphs are to prose what mix quality is to music. :-)
  2. 2 points
    This is a very accurate representation of how a turkey views its annual nightmare, Thanksgiving :-)
  3. 2 points
    Try the global EQ. Yes, it's the best idea to tweak your patches at gig volumes and it's an even better idea to tweak them on a system as close as the PA/monitoring system you'll be using. But we all know that quite often that just isn't possible. That's the very moment to become good friends with the global EQ. Personally, I usually only use it for my own monitor and leave it to the FOH folks to adjust the sound properly for their needs, but in case that's not a viable solution, it's time to take matters into your own hands. The easiest way IMO would be to only use a broad (low Q factor) mid band and raise it. Mine sits there waiting to be used at something between 300 (or 305 as the Helix doesn't allow dialing in 300) and 700, usually I go for a rather low center frequency. In case that's not enough, add a high cut to taste, it's very easy to adjust. That's precisely what global EQs are for, so don't shy away from giving it a try.
  4. 2 points
    I'm no doctor, but I'll bet that there are pills for this...;)
  5. 2 points
    You're problem is two-fold: 1) You're changing speakers... and speakers are half your tone, if not more. Anytime you change what you're monitoring through you can expect the tone to change... sometimes a little, sometimes a lot, depending on how different the speakers in question are. 2) Volume. Tones that you dial in at comfy bedroom volume will sound drastically different at stage volume... even when using the exact same speakers. The perceived loudness of different frequency ranges changes dramatically with volume. At lower levels mid-range frequencies are prominent, and at higher volumes both the lows and highs will be perceived as louder, and the mids drop out. It's called the Fletcher-Munson curve, and it's just how our brains operate... there's nothing you can do about it. The only solution is to EQ your patches at or very close to the volume at which you intend to use them, preferably through the same speakers, or at least through ones that are as similar as possible. A pair of small near-field studio monitors are quite different than a full stage PA system designed to throw sound around a large room at high volume...toggling back and forth between the two, with a big volume difference thrown in, is a recipe for disaster.
  6. 2 points
    Done repeatedly. For the sake of redundancy: Every living thing is born, grows old, and dies. That's life. Helix is not alive. It sounds good today, it'll sound good in 10 years. Buy it because you like it. In X years there'll be a new model, but you can't buy it now. Putting off your purchase waiting for it is a never ending cycle. FWIW - L6 just spent a great deal of time and money standardizing the underlying architecture of the entire Helix line. The purpose being to make updates and improvements easier and faster across the line. Does that sound like they're about to drop the line for a new latest greatest? Lastly, analog in this context pretty much means tube amps. Tube manufacturing is environmentally irresponsible. IIRC, there's only four major tube manufacturers left in the world. Many of the transistors and other components used to make the classic effects are no longer manufactured. NOTHING is forever. Live in the moment, tomorrow is not guaranteed.
  7. 2 points
    I just wanna clarify something that is a huge problem with the mentality of most musicians. First of all, there should never NOT be a PA. If the venue is small where some might think “we don’t even need a PA,” you should still have at least a small personal PA system. This is not about needing power or volume...it’s about control and having direct sound. Sound waves are very directional, until it hits a wall, then the ceiling, then the floor, plus the hard finished surface of the drum kit, and a million other surfaces. These reflections create thousands of other waves that all reach the listener’s ears at different times. If you want to sound good, at all, you simply play quieter so that you can deliver single mix, from a direct source, pointed straight at your listeners. If they start hearing stage volume with the PA then it is going to be a good 15-45 ms behind and all the reflections from the stage sound mask the direct sound of the PA and that’s how you get phase/comb filtering/chorus(not in a good way), it’s a muddy and incoherent mess. This is what drives me crazy about musicians...they think that playing loud and pushing air somehow makes them cool or feel like a rock star but to the audience you sound like s%#t. Secondly, there is no such thing as a “stage mix”. There are monitor mixes, sometimes yeah, but only in very large venues where there is room to have that much sound without interfering with a PA. And this is more about control of feedback and helping the front of house than it is providing whatever mix every single band member wants. So unless your playing arenas, there is likely only room for one mix before hitting 120db and that mix should be for your listeners, not you! This is not the studio, it is live sound, and your not using headphones (most likely) so you don’t get to just blast whatever you want and expect that noise to somehow not interfere with the P.A. A 60 hz sound wave is nearly 40 feet long, that’s going to the ceiling and all 4 walls, and back again, at least once in most rooms. Now the kick drum is producing those waves about every second...can you understand how fast this will max out sound pressure in a room fairly quickly? And with each wave bouncing off multiple surfaces, multiple times, every second...all your audience is getting is mush. The louder you are on stage, the mores noise you are introducing and fighting with the clear, direct sound of the P.A. and your amps/wedges/reflections are now muddying everything up. It might sound good to you on stage but that’s because your monitor is 3 feet away pointed straight at your face. You are getting DIRECT sound just like your audience was supposed to be getting, and it might have sounded good to them as well. But get a wireless rig and step out in front of the mains and your loud a$$ amp/monitor is now reaching your years at all different intervals between 10-50 ms after the main PA is and now you essentially just put a ton of chorus and phase on your whole sound. Congrats. I’ve sat and watched guitar players tweak their amp and pedal settings for 20 minutes or more trying to dial in whatever they consider to be a good sound and then completely negate everything by blasting it through a full stack which is complete overkill even in arenas, and now your guitar is as loud as the PA and it turns an entire mix into absolute chaos. It’s funny because on stage they are so sure that they sound awesome and that they are rock stars and nobody can even hear if you are playing a solo because its so loud yet chaotic with no articulation whatsoever. All they hear is noise. Also, the only thing that should really be in your monitors is vocals anyway. The drums are the loudest thing on stage, you set your volume to match the drummer and your done. There’s no reason you need “more guitar” in your monitor because you can’t hear it over the other guitarist...well then tell the other guitarist to turn down genius! Always turn everything else down guys instead of turning yourself up. Play with as little volume on stage as possible, and let your sound guy mix the PA because he knows a lot more than you, and you can make his job 100x easier by not throwing your crappy, muddy, stage volume bouncing all over the room and nearly causing feedback at every turn on top of destroying a perfect mix. How this concept is so hard for musicians to grasp is beyond me...you have a PA, use it! Let it do the work! Rely on it to deliver sound to your audience not your amp. It has panning, EQ’s, onboard compression, and other fx...why would you WANT to take a loud, crappy, mono signal from your monitor and blast it, and all of its reflections, throughout the room after the sound guy just spent the entire soundcheck Fine tuning everything and creating a nice stereo image based on the room? And all this nonsense about the audience near the front at center stage only hear drums? Where are you playing, Wembly? I promise if you if you turn up as loud as the drummer, then your just as loud as the drummer. Period. How is that even a logical question? And besides that, most PA systems like the QSC K series have 120 degree spray. L and R speakers spitting out 120 degrees spray will cover everyone just fine unless your stage is over 100 ft wide which I doubt. And line arrays aren’t for the sake of providing “volume” differences to people in front of the stage. They are designed and setup to provide even frequency response. Low end travels extremely far, high frequencies are very short. Therefor, you need more precise directional control with higher output to throw high frequencies to the back vs low frequencies. These line arrays will typically have different speakers in each cab all the way up the chain to compensate for this. But like I said, this only becomes a factor if your playing arenas and therefor have no reflections. Because there is enough room for the sound wave to die before hitting a reflective surface. In most venues, however, reflections are going to way too excessive, loud, and obnoxious to ever say “center stage needs more volume” much less “more STAGE volume”. That’s absolutely ignorant. If they can hear drums but not guitar it’s likely because of TOO MUCH stage creating a lack of clarity due to phase canceling. Drums are quick transients so they will seem audible even when they are extremely muddy vs string instruments which basically become noise at high decibels with lots of reflections. It’s not that they can’t hear it, they just can’t hear it as a “guitar” because without the proper ear training they just perceive it as noise or simply “not there” because they can’t make out notes or phrases. You never want your audience to hear your stage volume, you want them to hear a direct delivery from the EQ’d and finely tuned mix that your PA is designed to produce. The audience will NEVER benefit from hearing two mixes with a ton of reflections added in, no matter how “premium” your instrument is. That’s absolute ignorance and it’s exactly why musicians need to stop thinking they are sound engineers just because they know how to adjust a fader. And for the record, any large production event you go to will always have passive speakers so why someone would blame a crappy P.A. on that aspect also proves they know nothing. Passive speakers will be more accurate and allow you customize the power settings vs being stuck with whatever internal power supply was built by the lowest bidder. So to re-cap....stop using the term stage mix...there is no such thing as a “stage mix” and it just keeps other musicians thinking that there is and justifying the ignorant way they approach live sound. And the final point, the universal truth to remember, and this applies to every single live sound event throughout the entire world...I promise you, the quieter you are on stage, the more amazing you sound through the PA. No matter what venue, no matter what gear, no matter how good you THINK you sound...it will always sound better with even less stage volume. That’s just the physics of a sound waves.
  8. 2 points
    The magic of a tube amp is all of it's imperfections. If L6 didn't provide these options, there would be people complaining the amp wasn't noisy enough :) I'm with you.... just remove it completely. But... it is 1 (of many) imperfections people think they need.
  9. 1 point
    I'm having some weird issues after the Helix 2.80 update. My joystick is now wonky. I can no longer "push" the joystick in to open the model list when making or editing blocks. I have to use HX Edit or manually turn the joystick to go through every effect or amp I'd like to use. While not mission-critical it's annoying to have to turn the knob for 3 minutes until I get to the amps. Any advice or experience with this issue would be greatly appreciated. And it's not a DSP issue, I've tried it on new/blank presets as well
  10. 1 point
    And fwiw, that SOS article is pretty dated and not exactly well written. The guy doesn't even mention the socalled "safety buffers" each and every interface will introduce. These exist to warrant proper functionality and they're usually not documented, plus, apart from a few rare exceptions (I know MotU offers low safety buffer options - which doesn't seem to help their interfaces, but that's another story...), you can't access their settings as a mere user, so they're baked into the interface. And quite often they're responsible for a large percentage of your overall RTL, especially on cheaper interfaces. Which is also why you will get vastly different RTL values from different interfaces, even if you're using the same samplerates and buffersizes. The SOS author is not mentioning this, which is a bit of a shame, as SOS usually is a pretty reliable source. In case you want some extensive but interesting reading, I recommend this thread: https://www.gearslutz.com/board/music-computers/618474-audio-interface-low-latency-performance-data-base.html Anyway, it still hasn't got much (if anything) to do with the recording offset the Helix introduces. That's all about proper host-driver-communication.
  11. 1 point
    Wishing all a Happy Thanksgiving!
  12. 1 point
  13. 1 point
    Hi, Not sure where you got the info about FS4 & 5, but it is wrong - that is not a valid combination. If you meant FS3 & 4 “Test Mode” that is only available on the Floor and LT models. For your entertainment here is the complete list of Reset/Test options for all the HX hardware from the website - https://helixhelp.com/ Helix Floor/LT Button Combination Description 2 LED light Fun 3 & 4 Test Mode 5 & 6 Global reset 7 & 8 Reset setlists to factory, Keep IRs 8 & 9 Reset setlists to factory, Clear IRs 9 & 10 Reset setlists to factory, Global reset, Clear IRs 10 & 11 Upgrades existing presets to latest format 11 & 12 Clear current preset 5 & 12 Clear setlists, clear IRs, Return system settings to default 6 & 12 Safe boot mode To use Reset Options: Power off the Helix Hold the desired button combination Power on the Helix Release the button combination Please note:Buttons are numbered from top left to bottom right. Helix Rack Since the rack doesn’t have footswitches, you’ll instead hold down a combination of the 6 encoders that are below the main LCD screen as you’re booting up. Button Combination Description ENC2 & ENC3 Reset globals ENC3 & ENC4 Restore stock presets and setlists ENC4 & ENC5 Restore stock presets, setlists, and IRs ENC5 & ENC6 Reset globals and restore presets, setlists, and IRs (entire factory restore) ENC1 & ENC3 Rebuild all presets ENC1 & ENC5 Clear all presets, setlists, and IRs ENC1 & ENC6 Will enter Update Mode (This feature is only available on firmware v1.06.5 and higher) HX Effects Button Combination Description Holding FS3 + MODE Will clear all Presets and IRs Holding FS4+FS5 Will reset Global Settings Holding FS5+FS6 Will reset Presets only Holding FS6+TAP Will Perform factory restore (reset, Globals, IRs, and Presets) Holding the right arrow ">" Will put the unit in a fail safe update mode. Use this when the HX Effects is in an un-recoverable state HX Stomp Button Combination Description All 3 Footswitches Clears all presets/IRs FS1+2 Resets presets and IRs FS 2+3 Factory restore (globals, presets, IRs) Have fun!
  14. 1 point
    Fwiw, unlike Mr. Sadites I don't exactly recommend to start with a compressor as a sort of standard at the end of your signal chain. Yes, it may help to polish your sound, but it may as well cut off some of the delicate dynamics of an amp that you would otherwise like. Guitar amps by themselves already add a lot of compression (defenitely true for anything overdriven), so there's often not too much of a point in adding some more - at least not straight "out of the box". Especially when playing live, I often want to maintain as much dynamics as I possibly could instead of squashing them yet some more. This is not meant to say "never do it that way" - but I'd simply not start with it. It's nothing you'd do with your real amps, either. It's also not how things are dealt with in studio recording situations. When dialing in your sound, there's usually *zero* tweaks coming from the engineer operating the console. He/she simply won't slap a master compressor on your sound (unless you explicitely ask for it). As you're just starting out, there's these things I highly recommend (I would probably do all this with the Powercab set to flat mode): - Insert a looper as the very first thing in your signal chain. This allows you to record something into it and do all tweakings while someone else (the looper) is playing. If anything, this is by far the biggest time saviour in my world when creating guitar sounds (I was using external loopers just for that purpose since decades already, can't be beaten, really). It's an entirely different thing from playing something, have a note/chord ring out, adjust things, play again, adjust things - etc. Especially as we're not dealing with a nice amount of physical knobs (but just 6 you need to switch to or the editor), this will save you time. And I'm not talking about saving you some seconds but literally *hours*. Record a riff, adjust, stop the looper, record another riff, to see whether that would work with the same sound, stop the looper, record some single notes, etc. -- Insert just an Amp+Cab block and go through *all* the models while the looper is playing. Sometimes the default settings programmed by Line 6 aren't ideal, so I strongly suggest to spend a little time on each amp to at least adjust the drive(s), master and sag parameters, these are the most crucial to adjust the drive amount (obviously) and the amps dynamic behaviour. In addition, you may want to turn down hum and ripple (they're basically nonsense IMO, at least when it comes to selecting sounds, plus they add noise even in case you're not playing, which is annoying - I have no idea why they're turned up that much by default, if it was like that on a real amp, you'd possibly think about having it repaired). - Once you found an amp that's in your tonal ballpark, play through it. Then go to your global settings and check out the leftmost entry under "Ins/Outs", it's called "Guitar In Pad". It's off by default. Switching it on will globally attenuate your input level. I think the main reason for this to exist is to compensate for guitars with pretty high output, but many people (myself included) think of it to work pretty well on lower output guitars, too. My main axes really don't throw out much level and I keep the Input Pad on all the time. There's still tons of gain if I need it. In fact, I think most amps are a bit too much on the gainier side of things, so I absolutely welcome the additional headroom for clean and slightly sizzling sounds. YMMV, but this might be an important decision to make. - Back to the "ballpark" amp, have the looper playing a bit more and check out all the cabs without changing the amp - or at least the similar cabs. So, in case the Amp+Cab block comes up with a 4x12, make sure to check at least all the 4x12s. This makes a *huge* difference. Like *really huge*. Depending on the amp model, using a different cab might even sound like using an entirely different amp. Gets even more drastic once you change the base cab model (such as from 4x12 to 1x12). - Once you found the cab you like best, you might want to fool around with the mics and their positioning (IMO close is fine for a start, regardless of the mic). For mic selections, I find it to be particularly useful to actually play around and enter pedal edit mode, then select "Mic" and use the "Value +/-" switches to go through the mic models. Note: Pedal edit mode in itself is extremely useful for plenty of fine tunings, as it allows you to play around while adjusting parameters simultaneously - and as great as using a looper is, patches always feel a bit (or even more than a bit) different once you play them yourself. Personally, I'm always using the looper to get me as close as possible and then do fine tunings via pedal edit while playing around (ideally with some backings). In my case, doing it like that allows me to come up with very useful patches in a very short amount of time. As usual, YMMV. Of course, pretty much everything applies to further edits (such as adding other blocks), too.
  15. 1 point
    There are three primary parameters to overdrive - saturation (drive), tone and output level (volume). There are many approaches to controlling these parameters to get different tones for different purposes. But let's explore three broad approaches and their advantages and disadvantages. The simplest and perhaps most traditional approach is to use the amp for distortion. For traditional Fender, Vox and Marshall amps, that often means turning up the volume until the power amp distorts. Depending on the power rating of the amp and the number of speakers, this is going to be pretty loud and aggressive sounding, with a lot of odd order harmonics from symmetric clipping of the power output stage. This isn't very flexible because all the tone controls are in front of the distortion except presence or the treble cut in the Vox amp. So most of the tone control of the distorted tone comes from the speaker choice. What you loose in flexibility though you gain in feel since louder generally sounds and feels better, provides more interaction with the guitar for better sustain, and introduces sag for additional dynamics. The biggest issue though is that you have to set the amp for the dirty tone, then use the volume control on your guitar to get a clean tone. This often results in a pretty dark clean tone. If you amp has channel switching, then this wouldn't be a problem. To work around these disadvantages, you can use distortion pedals into a clean amp. Now you can get distortion tones at controllable volume levels with tone controls after distortion to tame the fizz/icepick, warming up the distortion. Pedals aren't that expensive, so you can have different ones for different purposes. This works so well that there is an explosion of fantastic distortion pedals on the market to choose from. The drawback is that it can be difficult to get a wide range of tone/saturation options out of a single pedal. To work around that shortcoming, you can use multiple pedals to create flexible gain staging in your signal path. With multiple pedals, you can set different pedals for different tone/saturation combinations for different purposes. The big question is, how should these different gain stages be ordered, and how should they be used together? I think a simple general rule (but one meant to be broken) is to put sustain before tone in the signal path. That is, you want the things that establish the drive tone to be after the things that create the distortion and harmonics, so you have more control of how those sound. For example, working backwards from the clean amp to the guitar: 1. Set the amp for you cleanest tone and overall volume. Use the neck pickup on your guitar with the guitar volume and tone all the way up to create your base tone. Tame an overly bright bridge pickup with the guitar tone control. Do this at typical playing level to get an accurate idea of the tone. 2. In front of this, place your first gain stage. This one should introduce a little breakup and with the tone set to keep the level of warmth you want. I use Teemah! for this as its tone controls work very well for sculpting early distortion tone. Teemah! also provide an option for asymmetric clipping for more even order harmonics that can sound less aggressive. This will often be left on all the time. It establishes early breakup tone without having to turn the amp up so its too loud. 3. In front of this, place your 2nd gain stage. This gain stage plays two roles. By itself, it creates a different tonal structure, often with a significant mid hump with a bit more saturation for a more aggressive distortion that cuts through the mix for solos. Stacked with the first gain stage gives increased saturation for typical leads. Use the volume and tone control on your guitar to easily adjust the amount of saturation and tonal color. I use Minotaur for this, with the gain set relatively low and volume/level set higher to drive the first stage into more saturation. 4. In front of this, place your 3rd gain stage. This one is for over the top sustain when you need it. I use two mutually exclusive options for this stage (I don't ever use them together): hard clipping distortion (OCD) or fuzz (Facial Fuzz). These will often be used with either or both of the first and second gain stages depending on the song. Fuzz especially seems to benefit from having another overdrive after it to provide more control of the fuzz tone. 5. (optional) In front of this, you can add another distortion block that helps establish the fundamental tone of your guitar. I use Heir Apparent for this because it has a lot of flexibility in setting saturation and tone. This one also stays on all the time, and isn't even assigned to a footswitch. It doesn't do that much, but you do notice it when its turned off. I use it mostly to just fatten up single coil pickup tones. I find it is better to get increased saturation by gain staging multiple pedals then it is to get all the saturation from one pedal. This keeps each pedal working in its sweet spot while also giving a lot of flexibility in distortion voicing. Another good rule of thumb is to use the minimum amount of saturation for the song. This maintains articulation and keeps your guitar from turning into an indistinct buzz that disappears in the mix. The gain staging above applies to traditional pedalboards and of course works great with Helix. But Helix has an additional level of flexibility that is generally not possible with traditional amps (unless you use reactive loads). Helix amp models have a level control and cab blocks with low and high cut. So it is possible to leverage the distortion capabilities of the amp models and cab block low and high cut to control distortion saturation and voicing. I use this in HX Stomp to get various distortion tones using a couple of footswitches to control amp drive, treble, bass, presence, and cab high/low cut to get the tones I need without using any distortion blocks. This is sort of like going back to the traditional approach, getting all the distortion from the amp, but using studio controls after the amp to tailor the tone and volume level.
  16. 1 point
    Yeah, the Zoom units feature an autosave function and it can be a blessing. Ok, here goes: Let's take the example from above. I want to have all my clean sounds a bit louder. And I'm using the same amp for these clean sounds throughout my 26 presets. That amp would be set to "global". Now, of course, I'd have to get down (or turn around, in case we're dealing with a Helix/Rack) and adjust things. I wouldn't have to press "save" though - I mean, that's one of the points of global blocks. There's no "auto save" required for that, either, as global blocks never get saved. They're simply treated globally. Absolutely the same as with an analog amp channel. Alright, now, if you would screw up the 25 other presets - then the amp would be the wrong choice for those presets. The way I would use global blocks (and IMO it's the single most relevant purpose for them to even exist - if they would exist), is to set everything up every bit as a bunch of analog amp channels mixed with some programmable FX and what not. It's what tons of players did ever since somewhen in the 80s when all the big rack things (Bradshaw controlled and what not) appeared. Basically, what I'd like to do is to have a set of "items" that could have their parameters adjusted globally but could still be switched per patch/snapshot plus a bunch of things that would be completely changed between presets. In the "semi analog" world, this is a piece of cake to set up and there's plenty of people doing it just like that. You may have your Gigrig loopswitcher which would control your amp channels and add pedals whenever you wish (semi programmable, maintaining global parameter access), while at the same time switching patches on, say, your H9 (fully programmable, no global parameter access). Not an unusual setup at all. I'm not sure about that. But it's also why I suggested introducing a new kind of "grouped" patches or whatever you might call them. Anything outside of these patches could work every bit the same as it currently is. "Tagging" something as a global block would only work within these new grouped patches - and, again as said, I could for instance imagine that the basic layout of these grouped patches would have to be the same all throughout. Same routing, same block arrangement, all you would be allowed to alter between the patches of a given group would be to change blocks (in case they're not tagged "global") and their parameters. Thinking about it, this could perhaps even be realized within one single patch. What would have to be added would be, well, "sub presets" (similar to snapshots) within one patch, just that, unlike with snapshots, you'd also be allowed to change block types. You might then end up with, say, a dual amp setup, one clean, one overdriven, a pedal for each and some "I always use the same settings anyway" FX (maybe delays and reverbs) - and at the same time, you might have a filter, phaser and vibe combo on one "sub preset" and a flanger, chorus and pitch combo on the other - which would otherwise exceed the CPU power available within one patch. During soundcheck however, all you'd have to adjust would be the "group wide" things. And you could still do ad hoc changes while playing. When you think about it, this - at least inderectly - would also free up CPU ressources, hence allowing for more wicked patches. Because instead of cramming everything into one patch, even if you'd only need your bubbly-phasing-filter madness once per gig, you'd just spread that load over different patches without losing quick, instant control of your overall sound.
  17. 1 point
    Uhm. Ok, here goes. This might get long, so please bear with me... Many of the typical gigs/jobs I play usually involve kinda like "telephone" or "last minute" bands. In other words: I don't know what'll happen until 1-2 days before, sometimes it all happens straight at the gig. Bunch of sheets flying around (or being airdropped, whatever), big piles of tunes everybody sort of knows and off we go. Needless to say that I simply can't prepare properly for such a gig, let alone sort out sounds and create setlists or whatever. So I will have to rely on a bunch of allround tones and possibly modify them as good as it gets in very short time (during quick soundchecks and even while playing). Ideally, I will at least get the main things right, such as gains, channel levels, BMTP controls and all that. With that one-patch-per-gig approach, I'm getting away just more or less fine. I can even re-adjust some things while a chord is ringing out, etc. All this simply wouldn't work anymore as soon as there's more patches involved. So, to turn up my clean channel volume (plus whatever other parameters) would require calling up each patch, doing the adjustments and re-save the patch. But imagine I turned it up a bit too much - there's simply no way of going back during the gig as I can't just hold a note, recall 3 patches, adjust and re-save them and then go back to the patch I'm actually playing. It's absolutely impossible. And the more patches there are, the messier things get (and it's not Lionel Messi...). Ok, now here's how I used to deal with this before going Helix: For around 20+ years, I've been using various incarnations of loopswitcher based setups, everything usually being controlled via MIDI. A typical setup might've been like a 2-3 channel preamp (sometimes even multiple preamps, but those were the 90s...) with a bunch of pedals slapped into pre-amp loops and some FX running in post-amp FX loops or whatever. With any such a setup, the combination of pedals and preamp channels plus whatever FX (rack- or pedalboard-powered) provided plenty of flexibility while still allowing me to adjust some crucial parameters at any time. Clean channel needs to be a bit louder? Well, let note ring, turn volume knob, call it a day. And that action would be valid throughout all my 13 (or 74) clean(-ish) patches. Fwiw, this is also why I didn't use just a single multi FX unit for post-amp processing but usually would at least have a dedicated unit for the final reverb (and sometimes the delay as well). That way, I was still able to call up rooms, plates, halls, etc. to my liking but would as well have global control over just the overall reverb mix. You might get the idea. With the Helix, the only way to do anything similar to that is to stick with one patch. Anything else simply isn't realistic for the things I do. Now, this is working pretty well. As I usually don't need much but a limited bunch of rather generic FX (some delay and reverb, a tad of phaser and vibe here and there - kinda like that), a single patch is often just fine. Yet, with all that power under my toes, I really wish I could do more here and there. But due to the nature of many of my gigs, that isn't possible. Global blocks (or whatever you may call it) would solve that instantly. Here's roughly how it goes in Fractal/AxeFX-land (I never owned one, so I try to recall what people told me or what I've once read in their Wiki): You could "tag" anything as a global block. And once you're using that tagged block in mutliple patches, any changes you do to that block would be valid for each patch using the same block (you can still use the same kind of block without it being global, of course). Back to my proposed scenario, I would very likely set two amps, 2-3 drives, one delay and one reverb to "global". That would keep the most relevant things consistent throughout all patches and still allow for a whole lot of variations (way more than what's possible within a single patch). And oh well, as if this wasn't long enough already: I also think this has a lot to do with aesthetics. I don't think it's the best idea to use entirely different sounds all night long. When you listen to wellknown players, they will have 1-2 core sounds which they will then build up upon, using additional gain things and FX. It's really just the Top 40 et al players who seem to need gazillions of different amp patches (ok, maybe John Mayer, too...). Personally, I think it's great to have a somewhat larger variety of filter, modulation and delay/reverb FX available but keep the core sounds more or less identical. Ok, and finally, I also think that the lack of quick adjustments is responsible for modelers to often not sound as great in a live context as they possibly could. In case you have to bow down (or walk back and turn around, whatever) all the time, dive through menus. press save buttons and what not - it's likely that you rather not do that but live with a less than ideal sound. Global blocks would have a potential to solve that with ease. Add a little MIDI knob box (or a tablet controller) allowing you to control all important global parameters in one centralized small-ish unit (not located at the bottom of the stage, either) and the world is a better place. Well, I told you this would get longer (and I could even add some detailed ideas how all this could be realized...)
  18. 1 point
    On the Obscura stuff, I'm thinking a fretless bass with low action for that sweet mwah, compression, lots of mids, and some tube-y saturation. Could probably reach a similar result in many ways, and I'm pretty new to Helix, but I'd probably try something like LA Studio Comp --> Ampeg SVT (for some mild tube saturation) --> Ampeg 8x10 miked with 421 or maybe 47 if you want a bit rounder tone --> EQ for mids. But bass tones are largely about playing technique/bass setup, compression and EQ anyway - can't give any magic bullet settings. Also, looks like the Obscura bass player plays through an Ampeg fridge in every shot, so might indeed be a pretty good starting point. Another cool trick you can do with Helix (that is annoying and costly in the analog realm) is to split the signal with the crossover split to retain deep and powerful lows while being to add more character to mid frequencies. Can be tricky to use, but also very powerful if you want just a slight bit of grit on the top of your tone without saturating most of your signal. 1.) Split the signal with a crossover split. 2.) Add heavy compression to the lower (bass frequencies) path and leave the upper path with light/zero compression depending on playing dynamics. Some light tube saturation can also work for the low freqs, experiment! 3.) Apply dirt of your choice to the upper path. Guitar amp+cab models can work well here also (it's how they did it in the 70s), I like the P75 model with the stock cab but might not work for your application. 4.) Adjust crossover point and mix to taste. 5.) Apply EQ and low/high cut to entire signal. You often end up with a slight dip around the crossover point, but you can use EQ to remedy that to some extent. One thing I've liked to do in the analog realm is to chain some other dirt box (Rat-style boxes tend to work well IMO) into a Darkglass B7K and use it strictly as a tone-shaping device. The Darkglass has some really nice tone-shaping magic going on, but I've never liked the dirt on it personally. It is great for taming other dirt boxes with more interesting character, though, and I assume this would work with Helix too, as the B7K emulation seems to be pretty good.
  19. 1 point
    Yep, Line 6 problem from the start. They knew about this in Beta For IOS 13,,,I use my ipad for 1 thing...MUSIC . Now I can't...I registered every line 6 product I own...Yes, c'mon , register your product. Give all your information. But when a critical update will cripple your Line 6 DEVICE, your on your own with no warning! You have my email, serial numbers from products that I own, how about, *Hey Sonic Port Owners...Don't update your IOS because we know your device will not be compatible".....I could have lived with Apple 12.4 . Didn't need Apple IOS 13. Now almost 4 weeks ....with a sonic port used for a paper weight. I have purchased about 5 to 6 guitar Apps from Itunes that CAN NOT BE USED because I have a non working guitar interface. Sorry to vent my frustration here....Thanks for listening...
  20. 1 point
    Problem Solved! Thanks Craig for pointing me in the right direction. Turning down the HUM control didn't fix it. Turning down the Ripple made the hum go away! Further experiments reveal that the HUM control will bring in a low octave hum and the Ripple brings in a buzz like a bad connection. It effects some amps more than others. It may be authentic but who really wants buzz and hum in their sound? Thanks again Craig. Bill
  21. 1 point
    There is a hum parameter as part of most of the power amps, it might need turning down. It struck me as weird at first because I'm in the UK and I am used to 50Hz hum. I don't know if it's relevant but in a tube amp, power supply hum will be 2x line current frequency (100/120Hz) whereas ground hum will be 50/60Hz. Craig
  22. 1 point
    Hey, I have good news! I found out how to fix it! So, My AMPLIFi 150 went out of warranty so I decided to open up the back and have a look. According to some of the stuff I read, the SD card with the firmware on it is the culprit for the problems, so I removed the retaining glue dob and then the card, booted the amp without the SD (It gets stuck in the process), Shut it down, then replaced the SD and booted it again. I haven't had any trouble for well over a year now. I love my Amplifi now and I use it all the time, I even perform with it now because I trust it. If your Amplifi is playing up and is out of warranty, open it up. Nothing to lose, right? There are 6 main screws on the back plate and I can't remember what's on the inside. I do remember that there is a white paint-like substance on the back of the back plate that sticks to everything, so wear gloves or be careful not to lean against it. The SD slot is in the centre of the mother board if I remember correctly and has a little bit of glue to hold the card in. Good luck! Riley.
  23. 1 point
    Hate to tell you this, but trying to set your tones this way is only going to lead to you not being happy with the result. It has been an ongoing debate on this forum about headphones - which are best for etc. Maybe you should research these threads: https://line6.com/support/topic/23319-helix-through-headphones-sounds-awful/ https://line6.com/support/topic/26959-the-mother-of-all-headphone-complaint-answers/ From what I have read about your Sennheiser Momentum, wireless version or not, they are low impedance 18ohm with mini jack connection that seem to be aimed at the mobile user eg: phone, tablet etc., so I’m not sure that they are ideal for setting your tones, but that’s just my comment.
  24. 1 point
    I you are sending Helix to the Mixer Direct - putting the speaker cab behind you will muck up the mix - depending on how loud you set it. My FOH engineers will let guitarist and bass players setup anyway they are comfortable - if we are part of the Rehearsal before a tour - we work out with the musicians how to achieve the best mix. If you are in small venues - i.e. clubs, bars etc...I would never put the speaker behind you - the volume gets up too loud - especially if you are trying to 'feel' it.
  25. 1 point
    Press down on the “Preset” button/knob on the left, then navigate with the joystick.
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