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I've been using my M13 for a couple months now and generally love it. But I cannot figure out how to get a volume boost for a solo without using an overdrive or distortion effect. Often I want to do a clean solo, perhaps with just some delay or phaser. But those types of effects don't have a gain or output level parameter that I can adjust to boost my signal loud enough for a solo. Am I missing something? Is there some type of "transparent" effect that has an output level parameter I can use for a "clean boost"? Is there some overall setting/preference I should be tweaking? I don't have a volume pedal and I don't want to be dependent on an expression pedal for this. I want to be able to hit one foot switch and not have to do the pedal dance. I run a mono set up, simple input/output with a Fender Hot Rod Deluxe IV. Thanks for the help and suggestions.

hi guys first of all I'd like to let you know that I read all the threads I found about this same subject and didn't find the complete answers, so I decided to run some tests myself (like I did on the POD HD years ago) and to create a new thread with my results (I'm on FW 2.60 by the way) ok, I'll start from the end results, so that those who just need to have a quick answer can read it immediatly, then I'll give some deeper explanations the main point of this post is: the moment you create a split/merge routing configuration by dragging down a block you get a level change: after the merge block, with all bypassed fx: with split A/B and Y you get +3dB at the output sum with split crossover you get -3dB at the output sum BUT each path by itself gets attenuated by -3dB so if you use the split A/B to switch between sounds, each of those sounds is -3dB qiueter than the same fx/amp with the same settings in a non-splitted path the good news are that the level changes do not actually occur at the split, so the two paths are both an exact copy of the signal before the split, instead the change occurs at the merge block so the right solution is to act on the merge mixer faders --- why? and why those values? • well to answer, I need to clarify two things that I've read someone get wrong and mixed up in other threads 1) when you sum two identical signals in a circuit either analog or digital you get +6dB boost (voltage sum) 2) when you send a signal to a physical speaker and then send the same identical signal also to another identical speaker you get +3dB (acoustic power sum) this is math and physics, not an opinion now, this second point is one of the reasons why something called "pan law" exists: as we said, when you have a mono signal going to two speakers (so the pan knob on a mixer is at the center position) you get +3dB of sound pressure so mixer designers in some cases decided to pad the center position of the pan knob by -3dB, gradually returning to unity gain at the extremes (in some other cases they may use -2.5, -4.5 or -6dB: the actual effect of the pan law depends on the coherence/incoherence of the signals, on speaker placement, on listening conditions, and also on mixing taste I would say, so the designers have to make an assumption and take a plunge, I don't have time to enter into this now, sorry). • now back to the Helix: you have an empty "new preset" patch you have a signal going through path 1 (I used various sine waves and pink noise) you add a gain block at 0dB (if you want you can even bypass it) => nothing changes you drag the block down creating a double path => you get +3dB at the output sum so, what's going on? - the A/B or Y split duplicates the signal to two paths, so each of them carries the same identical signal which goes to a merge block mixer channel, then the two get summed together... so you should get +6dB, but... ... since the pans in the mixer have a -3dB pan law you only get +3dB - the crossover does not duplicate the signal but splits it in two frequency bands, if you sum those two filtered signals you get the exact same signal you had before, so no +6dB boost here, but there is still the pan law, so you get the -3dB - but, as I wrote before, each path by itself gets attenuated by -3dB because of the pan law, so if you use the split A/B to switch between sounds, each of those sounds is -3dB qiueter than the same fx/amp with the same settings in a non-splitted path • please note that these are not theories or speculations, I've tested and measured thoroughly every configuration and I'm sure the Helix routing works like that • all this behaviours occur identically to mono or stereo signals, remember that all the lines representing signal flow in the Helix display are always "double conductor cables" so either stereo or dual-mono (stereo = they carry two different signals; dual-mono = they carry two identical signals) this point could be deepened a lot but I don't have time for this now --- some other considerations: • if for example you open the pan pots of the merge mixer to the extreme L and R, the pan law attenuation does not apply and the level goes back to unity BUT this is not a solution, because if you run in stereo you only get the left from path 1A and the right from path 1B • I'd like to warn you of potential problems I found with the parameters inside the split blocks I) the split A/B "route to" parameter is a balance control, not a crossfader, so moving it to the left attenuates the signal going into B while not touching the signal going into A and viceversa; so it is appropriate to use that parameter to switch from A to B but not too much to find a mix of A with B for 3 reasons: 1- as I said you only have control over the attenuation of one of the two signals you are mixing 2- at center (even position) you have more level than any other position because the two signals are at full level, so the judgement on which is the best mix is compromised for psychoacoustic reasons 3- worst of all: if the blocks in the paths are amps or distortion boxes or compressors, as long as you are not in A100, "even" or B100 positions, you are sending an attenuated signal to one of them, changing its sound and not only its level II) the split Y "Balance A" and "Balance B" parameters (added in fw 2.10 so not covered in the manual) work like this: the stereo or dual mono signal goes into the split, it is duplicated to path A and B and on each of those you can control the balance between left and right, those balance control also have a pan law but different from the merge mixer block pans ! in this case it's unity at center and +3dB at sides ! (I think the reason for this is to avoid an attenuation to the splitted signals at default settings) so if for example you use 2 amps in the paths do not use these parameters or the two amps will receive a different level than if the balance is at center. --- • to finish I'll add two other Helix Routing measurements I found during my tests: - the pan block is actually a balance control with a pan law equal to that applied in the split Y block: at center it's at unity gain, going to the left attenuates the right up to minus infinite and adds +3dB to the left (and viceversa) - unlike the POD HD500 the fx sends / returns on the Helix are all at unity gain levels (+/-0.3dB), thanks Line 6 ! --- thanks for reading, I hope to have been helpful bye Lorenzo

Hi all! So I'm enjoying my HD500 (yea i only just bought one in 2018 in great condition second hand) for use with something not generally thought of but for solo acoustic shows. I will also use it for electric gigs with a band (1% of gigs) in the same way but this question is about specifically getting a unity gain signal out of the 1/4" output for Acoustic gigs so venues that have an XLR to In-House can get neither a too quiet nor a too loud signal. These venues take this signal and plum it off to their amp and usually control their level from an iPad. If the signal is weak i guess i'll have to readjust my master output coz they'll not have enough level and if its too loud it could result in a lot of bleeding ears and equipment damage etc. I have a pedalboard with the unit and Roland kick trigger in it where everything, vocal mic, acoustic and kick trigger (into effects return) is setup. I have done 2 gigs so far and it already sounds good but just with my own equipment not into a venues in-house. My paths are as follows: Path/Input #1: Guitar (with active pickup at max) > Vintage Pre > EQ > Tube comp (for boost) > AMP DISABLED > Both join at Mixer > Reverb > Effects Loop > Looper Path/Input #2: Mic > Vintage Pre > EQ > AMP DISABLED > Both join at Mixer > Reverb > Effects Loop > Looper Roland Kick: Effects loop return > Looper (this only seems to work when the Effects loop is disabled, why haha?) (if there is some way to get the Looper to only record the guitar id love to know!) Note: Unit switches = Guitar in is at "Normal" and 1/4" output is at "Line" I then use a mono 1/4' to XLR cable to my powered speaker, the XLR out of this speaker will go to the venues In-House. So i did notice that when i disabled the effects on the guitar path it was a lot louder signal, and the mic had distortion and some signal noise so i should be able to fix that no worries... My real question is that without the HD500 having a sound meter or clipping meter how do i know where to set the guitar pickup level and whats the correct way to achieve this? Is it at the mixer? Maybe its at the master output? Obviously on a real mixing desk you set the fader at 0 and bring up the gain pot so its kinda just in the orange. Is there some software on the computer that emulates a mixing desk that i could hook up via USB and play into to test signal level? Can the HD500 Edit software do it? Sorry for all the questions haha! My thought is to start fresh with the 2 paths (no effects or amps), leave guitar pickup at max, adjust desired Mic volume output to match with the trim pot, then add effects preamps/eq's etc that are not boosting or cutting volume. Would be great to hear everyones thoughts if that all makes sense, Thanks :)!

Hi folks, being a newby on Helix I dare another question here. I am using the HELIX just as a multi-FX-board and as a control panel for my Marshall JVM410H. I love the amp's sound and would not have another one (as long as the Marshall won't go down. If one day it would do so, I plan to use the HELIX' DI-OUT with an alternative set-list with amp, cabinet etc. directly into the mixer). First step was to control the amp's sounds by MIDI Program change mode which worked absolutely perfectly. No need for the amp's footswitch anymore. :) Second step was using the 4CM in order to put the Wahwah at the beginning of the signal chain and delay etc. after the MARSHALL's pre-amp. I've been watching this video for instructions: https://youtu.be/Wvz3L8bJb1w Well, I've been doing as was said here, but the results were so bad, so that I had to switch back to standard send-/return (3 cables). So, maybe someone can help me with this problem, by answering a few questions: My amp has 2 effect loops, one serial (preamp out, return) and one that can be used in a serial (by putting the mix knob on 100%) or a parallel mode (mix 50%). The latter has the big advantage, that the original signal stays in the amp while the Helix (when its effect mix configured correctly) delivers just the FX signal, while the Marshall mixes the original with the Helix' FX to an optimal sound ... in theory. Now in the tutorial it says, that the amp's SERIAL FX loop should be used. Well ... is that so, that good results can get achieved ONLY with the serial FX loop? I ask, because MARSHALL does explicitely not recommend this, if you are not using hi class studio rack effects. Well, no idea, whether the HELIX qualifies as such ... ;-) I learned, that in the HELIX' preset you have to add a send/return block each. But then, there is a "fx loop" block, too. Is this the same thing as one send with the return block combined? I DID make it work, even in parallel mode by putting the delay's mix to 100% and the amp's return mix to 50%. Sounded great. Problem was: when I chose a Helix' snapshot of the same preset that has no active effects but noise gate (as I do many times for rhythm sound), there is some screeching and loud humming that I cannot explain. Even when I turn the guitars volume to total zero, there is a feedback loop somewhere and I cannot find its cause. So I have no idea, where to search. But like this, you cannot use this setting at all. Anyone any ideas? Sorry, if I repeat some questions for the hundredths time, but the forum's search routine could not find answers to my questions ... I appreciate any help, because I need the Wahwah and the delay. First one sounds crap in Send / Return, second one sounds crap in front of the Marshall ... HEEEELP! :(

It seems so clear to me now! What took me so long? Here is the fun way to balance your volume across all of your patches. Go to YouTube Pick a backing track in the style that you most commonly play. I am a funk, blues fusion kind of player so my banks are set up : clean, rhythm crunch, lead. Turn up the amp and turn on the backing track and have fun! I start with rhythm and level all of those so i can switch between them without any volume difference. I move to cleans. My cleans typically are the Eric Johnson style: chorus, 2 delays and comp. I arpeggiate a pretty chord and make sure it cuts properly in the mix. I tweak all of the cleans until I can switch between all of them with no volume difference. Repeat for the other banks and patches A couple of observations in this process... This process assumes you have patches set up that you like. You don't want to get side tracked "fixing" a patch. This is about volume adjustment only. Sometimes you don't need more volume to get a patch to cut through the band mix. Sometimes a simple boost in the mids or other eq adjustments will do the trick. Only use one guitar. Pick the one you use the most. Don't use multiple YouTube videos: pick one and stick with one. That way there is no moving volume target since one video may be louder than the next Works with any guitar: Going from Strat to Les Paul? After you have finished leveling your patches just turn down the overall volume of the amp when you pick up the Les Paul. The patch change volumes are in relation to one another. Should be fine! For Variax players, you can level the volume of your modeled pickups so you won't have to deal with any amp adjustments. I use blue tooth through the amp for the backing track, then I'll put the backing track through the P.A. to confirm that my patches are good. Playing with the band is the final confirmation. Uber fun and useful cause you are playin a boat load of 🎸! And that is always good!

I'm thinking of getting a Helix LT next week to use as a 4CM FX only board with my Orange Dual Dark 50. I've been led to believe the amp has a line level tube FX loop. Would the following configuration of I/O and level settings work to maintain correct gain structure through the rig?... Guitar > Helix In Loop 1 (Inst Lvl) Send > Amp In Amp Send > Loop 2 (Line Lvl) Return Helix (Line Lvl) Out > Amp Return Thanks.

Hi. On my Variax 600 the tone of the G-String temporary fades away to 10% and crackles. If I hit the bridge with my palm several times and/or let the string slap hard everything can be fixed and the tone reanimates. I know there are problems described with sweat and dirt, polluting the piezo. Also there are issues with grounding or loose cable-connections. Can somebody please give me a hint or provide some links so I can figure out some solutions?

Hey, I have to question, I use the POD HD 500X via 4CM with my Engl Gigmaster (Effects only) How should I asset the Comprresor (and which COmp I should use) for best results in your opinion. Genre is Metal/ Hard Rock And is there any way to eliminate the levle rise of the analog chorus? Thanks

I don't have an m9 but I am looking at getting one. Since there are two expression pedal inputs (I am not using it for a volume pedal or wah or anything), can I assign a pedal to control something such as reverb decay or mix?

Hi, there, I have some problems to make the volume levels the same for every preset. at home i setup the volume in the presets at almost 100%. but there are so many diverence in level for each amp model. so at live i have to change every preset every time. can some tell me the right way to make t he same level volume for every preset. thanks.

HI, I thought I read about this issue before in this forum but I can't seem to find the topic below. I have an M20 (+ L2m, L3m and L3s) and specific inputs are currently dropping out. In my practice room pysical mic inputs 7&8 (which take electronic drums stereo) will drop out after about 5-10min. There is not drop in output from the drum module but there is no input level coming in. Examining further I can see the channel strip inputs for 1,2,7,8 all greyed out like there in no connector in them... when there physically is. If I plug into another pair of channels, they work fine. If I cycle power on teh unit the channels come back... for a while. WTF is going on with this thing? did this sort of thing happen before or was it my imagination? Why that block of channels? After the dramas with my L2ms (which I have only just had returned from the fan replacement - and haven't tested properly yet) I am afraid to take this out in public until my faith in the reliability is restored. Mavril

the HD500 Looper could attenuate the recorded signal by 6dB ... ...depending on your routing and settings. (all of this should be valid also for the POD HD Pro and bean, but NOT for the POD HD500X since its looper routing is different) The point can be summarized saying that the Looper is a "mono wet / stereo dry" fx block; you only need to be aware of this if you use the looper in "pre" position and you are one of those who think it's better to disable input-2 (for example by assigning it to variax) also, please, understand that what I'm trying to explain here is complicated so start reading only if interested, thanks to understand what I'm talking about, you need to read this post of mine and follow the signal on my graphs here: ...so, getting to the point: "the looper is a mono wet / stereo dry fx block" means that if for example you have a situation like this: - guitar connected to guitar input - Input-1: Guitar / Input-2: Variax - Looper in "pre" position - no fx - Amp Block in Path-A (path-A at unity and pan centered, Path-B muted) and you record a riff in the looper, the looped level will be 6dB less than the original signal! explanation: as a "mono wet / stereo dry fx block" the looper attenuates his inputs and sums them, then it records the result and, when playng back, sends the recorded signal to both his outputs so if you have disabled input-2, your recorded loop will have 6dB less than the original signal and will be splitted to outputs 1 & 2 of the looper, WHILE your dry signal will be present only at the output-1 of the looper now, if the first mono effect you have is in path-A, the mixed dry and -6dB looped signals from output-1 of the looper will be splitted to Left and Right channels of path-A and that mono effect will receive them at both its inputs otherwise, if the first mono effect you have is in the pre path you will hear no difference between dry and looped signals: indeed the looped signal will re-gain its lost 6dB because it will be present at both inputs of the effect (while the dry is present only on one input). so, to summarize: if you have Input-1: Guitar / Input-2: Variax and the Looper in "pre" position: 1- if the first mono effect is in path-A, it receives the dry signal at unity and the looped one at minus 6dB 2- if the first mono effect is in pre-path, it receives the dry and the looped signals both at minus 6dB consequences: This meas that, if your purpose was to have the ability to tweak your amp settings while listening to the results on the looped riff, and your amp is in path-A, you're going to get the wrong gain setting! (because you will be pushed to turn up your amp drive pot more than needed) Now, in the same situation (even while your riff is still looping), if you move your amp in "pre" position the loop will play at the same level but the dry guitar will play less distorted! solution? set both inputs to guitar (before recording the loop), and none of this problems will occur

In order to get a feel for the signal levels suitable for the HD500, I've made a few measurements of the gains available from the different inputs to the various outputs. I've also measured the gain of all the FX. A small Excel spreadsheet containing the data can be found here (unfortunately, I can only upload a PDF version of the data, which I've done here). A brief summary is shown below: All measurements were made at 1kHz with Master Volume = max, S/PDIF gain = 0dB, and Input2 = Variax (although this didn't make any difference because I had Inputs 1 & 2 panned hard Left/Right). Guitar input pad = -5.2dB compared to normal. 1/4" (line) output has 6.1dB more signal than 1/4" (amp). Aux & Guitar (normal) have the same gain structure (except Guitar has programmable input impedance). Guitar (normal) input to 1/4"(amp) output has 4.9dB of loss with no FX, no Amp, and Mixer = 0dB. Compared to 1/4" (amp) output, the XLR output is 9.4dB less while the phone output is 15.4dB more. Compared to the Aux/Guitar(normal) input, Mic (min gain) has 6.2dB less gain while Mic (max gain) has 38dB more gain. Aux/Guitar(normal) input clips at 0dBFS (measured at S/PDIF) with 8.3Vpp Mic (min gain) input clips at -1.1dBFS (measured at S/PDIF) with 14Vpp Mic (max gain) input clips at -1.1dBFS (measured at S/PDIF) with 91mVpp CD/MP3 input clips at various levels near 20Vpp, depending on which output is used (XLR, 1/4", or Phones) By far most of the FX have 0dB of gain, but there are several exceptions (e.g., Graphic EQ = 2.9dB, Tape Echo = -5dB, and others). FX were measured with Mix=0% (to eliminate comb filtering effects) and Gain=0dB (usually). The PDF files contain the whole list. Using some of this data you can see the clip level is essentially set by digital clipping, although the mic input stage only gets you to within 1.1dB of fullscale (close enough to call it fullscale in my opinion). The output stages will not get close to their clip levels (unless you use the CD/MP3 input which bypasses the digital engine). That is, it's sufficient to look at the S/PDIF signal to keep your signal below fullscale. You don't have to worry about clipping the input stages. You also don't have to worry about clipping the analog output stages. I hope you find this useful in optimizing the setup of your equipment. IO Gain.pdf FX Gain.pdf