Wet-Dry-Wet Setup with Helix

Mick and Dan’s last That Pedal Show in their old location, That Pedal Show – Awesome Wet/Dry/Wet: 3 Amps, Space Echo, Echorec, CE-1 & More, inspired me to explore wet-dry-wet setups for Helix. I wondered if it would be possible to produce a similar result, at much lower cost, and that would be simple enough to actually gig with. The look on Dan’s face as he was playing around the 20 min mark really shows the impact sound can have on us, and how we play. Its one of their best shows and well worth the time to watch.

In typical guitar signal path has mono tone shaping effects before the amp, and often stereo modulation and time based effects after the amp. Since affected and unaffected signals are both mixed together in the same serial signal path, and output to the same speaker, the effects and dry signal are mixed together, one on top of the other. This can cause the overall guitar sound to become less distinct as the dry guitar and effects are fighting for the same limited space.

Simple stereo patch.png

In this simple patch all the effects following the amp are in stereo and have their mix set to provide a blend between the dry guitar tone coming from the amp, and the wet tone produced by the effect. This works well and is a very common setup for Helix patches into a stereo FRFR or DAW. The dry guitar from the output of the mono amp block has the same level in both the left and right channel, so it will appear in the center of the stereo field. The effects will typically be spread across the stereo field, often using their Spread control set to 10.

While this typical signal path sounds good, there are a few problems with this setup:

  1. Adjusting the mix control for any of the effects will change both the dry guitar and effect levels. That is, turning the mix up will provide more effect but less dry guitar.
  2. The dry guitar is going through four mix controls and is therefore attenuated quite a bit. This can be compensated for using the amp Chanel volume, but that volume would need to be adjust depending on how many of the effects are on and where their mix is set. Snapshots would work for this, but a simpler solution would be nice.
  3. All the post-amp stereo effects are in series, meaning each downstream effect sees the output of the upstream effects in the signal chain, not just the dry guitar. For example, if the delay is before the reverb, you will “reverb the delay”. If the delay is after the reverb you will “delay the reverb”. Either of these may be what you want, but separating them so they are independent makes them easier to control.

All these issues result from mix decisions that are done early in the signal chain with each effect mix control. Each upstream mix decision limits the choices for the next mix downstream in the signal path. A better option would be to delay all the mixing until the last possible moment, when the various contributions to the tone come together in the air in a room on their way to your ears. Wet-dry-wet setups let you do this, but at some expense in therms of complexity and cost. Let’s look at some options to see how we can exploit this to create a context that inspires us to play and our audience to listen.

Simple Wet-Dry-Wet Configuration

Wet-dry-wet is a setup approach that provides greater control on the level balancing and and separation between the guitar dry or unaffected signal, and the effected signal. Sean Halley demonstrates this approach in Using Helix For Wet/Dry/Wet Scenarios | Line 6.

Here’s a modification of the patch above that provides a simple wet-dry-wet setup that only requires a stereo FRFR. We move the time based effects following the amp block  to path 1B so that we can control them independently of the dry guitar output thats on path 1A. The B Level control on the merge block at the end of the signal chain (right before the output) can be used to control the amount of wet signal without having any impact on the dry guitar signal which comes through clear and distinct in both the left and right channels.

Simple w-d-w stereo.png

In this case the Tremolo/Autopan block is kept on path 1A and does effect the dry guitar signal. This is because a volume effect would be lost on the wet path 1B because it could only auto pan or tremolo the effect inputs or outputs depending on where it is positioned in the wet signal path. This would likely have a negative effect on those effects and not be noticeable as an autopan effect. We want to  tremolo/autopan the guitar volume, not the effects.

Wet path effect order and mix

We still have the problem that the time based effects are all in series on the wet path, path 1B. In this simple wet-dry-wet setup, the order and mix levels in this path are critical to getting a good wet-dry-wet sound. What we want to do is make sure path 1B doesn’t have any unaffected dry signal, we want all that coming from path 1A. To accomplish this, at least one block in the wet path needs to be on and have it s mix set to 100%. This will ensure that there is no dry signal on that path. Which block should play this role depends on the block, block order, and how it interacts with other effects. It may take some trial and error to find the best combination.

In this example, I put the reverb at the end of the signal chain. I prefer that because it sounds more natural to add reverb to the delay repeats then to repeat the reverb, which cause the reverb to pulse with the repeats. If you set the reverb mix to 100%, this will sound fine with the delay off because the reverb input is the dry or chorused signal (assuming the chorus is on). However, when the delay is on, you won’t hear the ping pong delay since the reverb is 100% wet. You’ll hear a little big of ping pong of the reverb output, but not the delay repeats. So we have to use the reverb mix to control the relative level of the delay and reverb. That means one of the blocks before the reverb has to be set at 100% wet too keep the dry signal out of the wet path. Note that if you turn off both the delay and chorus, you’ll also need to automate the reverb mix back to 100%. This can be done with a footswitch controller that bypasses the blocks and changes the reverb mix at the same time, or it can be done with snapshots.

We have two choices in this configuration for which block to “block” the try signal, the chorus or the delay. The first consideration is the effect order: do you want to have the chorus see the try signal and then delay the chorus, or do you want the delay to see the dry signal and chorus the delay. Probably either can work, but I think its more natural to put the delay later in the signal path as that is what would actually happen if the delay was produced by nature.

Next is which should be set to 100% wet? I depends a little on how you might use the patch. Assuming the chorus is more likely to be turned on and off during a performance, that would imply the delay should be 100% wet. But it turns out that having both be 100% wet works out pretty well. If the chorus is off, the delay will see the dry guitar input, and produce 100% wet delayed output into the reverb. If the chorus is on and set to 100%, the delay will see a heavily modulated signal in the delay tails. If the chorus is on and the delay is off, the heavily modulated wet signal is fed into the reverb with its mix control set to balance between its input and output, producing a rich chorus tone.

Delay Scale control

Scale on delays can be set to 0% so the right delay is at the same time as the dry thru input signal. If there is a modulation or other effect before a delay setup this way, then its wet output will come through unaffected in at least the right channel. The left channels of the preceding effect would come through only on the left delay repeats. This is probably not that useful.

Scale and Spread controls on delays can be confusing. Spread is simple enough, it controls the stereo spread between the left and right delay repeats. Spread at 0 is mono, Spread at 10 hard pans the left and right delays. Scale sets the time relationship between the left and right delays, regardless of where they are in the stereo field as set by the Spread control.

The delay parameter sets the delay time of the left repeat while the scale parameter determines the time of the right repeat as a percentage of the delay time of the left repeat. This establishes a repeating pattern between the left and right repeats. Scale at 0% sets the delay time of the right channel to 0% of the left delay or 0 msec, meaning no delay. You’ll effectively hear the dry signal immediately in the right channel because there’s no delay. Scale at 100% sets the left and right channel delays to the same delay time, so the left and right repeats will occur together, making the delay sound mono regardless of where the Spread control is set. See why its confusing? Setting the Scale to 50% sets the right delay to 1/2 of the delay time making the left and right delay ping pong between the channels at effectively 1/2 the configured delay time.

This wet-dry-wet configuration sounds really good, only requires a stereo FRFR, and is reasonably simple. But the more we can separate things, and the later we mix them, the the greater impact the wet-dry-wet setup can have on your tone. We can go a step further if we use three amplifiers, or add a center channel FRFR to our stereo FRFR. Richie Castellano shows a great, but somewhat complicated example of this setup using three guitar amplifiers towards the end of Using The Line 6 Helix with Amps and Pedals. Let’s see how to do something similar with FRFR speakers, and maybe simplify things a little.

Center Channel Wet-Dry-Wet Configuration

In this configuration we will add a center channel speaker that only gets the dry guitar signal. If your using typical powered PA speakers for your FRFR, you can just add a third powered PA speaker for your center channel. I have a pair of JBL EON610s that I use as my stereo FRFR. To add a wet-dry-wet setup to this, I just add an EON612 as the center channel.

What we want to do is send the dry guitar output to the center EON612, and send the wet stereo output to the two EON610s, one on either side of the center speaker. To do this we’ll use path 1 for the mono dry guitar signal, and path 2 for the stereo wet signal.

The routing is actually very simple. Take an output right after the amp (or cab or speaker IR) block, and after any post amp mono effects (such as EQ, compression, and possibly tremolo) and feed that into the center EON612 using s Send block. Then route the output of 1B to the Multi output and path 1A to the input of path 2A for the rest of the wet signal path. We’ll use the send block Dry Thru control to control the volume of the wet output.

Center Channel w-d-w.png

This similar to the setup we had before with just a stereo FRFR. But in this case, the wet and dry signal is being mixed in the air in the room, not in the Helix and narrowed down to a stereo output. This will sound a lot bigger and have that 3D tone that Dan loved so much on That Pedal Show.

Now that we are using path 1B and all of path 2 for the wet signal path, we can separate the chorus, delay and reverb into separate parallel paths so we don’t need to deal with the reverb mix anymore. All the mix controls in this configuration are set to 100% wet. We just have to be sure that at least one block is on all the time in path 1B, path 2A and 2B in order to be sure there’s no dry guitar signal going into the wet output. For the chorus block on path 1B, we can use the level control to turn the block all the way down to turn the effect of instead of bypassing the effect, which would send the dry guitar output to the left and right speakers, defeating the wet-dry-wet setup.

Unless of course you want the dry guitar signal in all three speakers. In this configuration, if you turn off all the effect blocks on say path 1B, 2A or 2B, the dry guitar output will be going to the center channel and both left and right side speakers. That could be great for a normal stereo reverb setup where you have three speakers instead of two. This will sound fuller and carry further simply because the guitar dry signal is able to move more air with the extra pair of FRFR speakers. So putting the dry signal into the left and right speakers is not a bad option to have too.

Another difference is the Tremolo/Autopan is now moved to the wet path and put first in the path 2A signal chain. An autopan effect is likely to be used by itself, not with chorus or delay, so with this configuration we get the dry guitar coming out the center channel speaker, and the autopan shifting back and forth in the left and right speakers at the same time. That’s pretty cool!

Additional Considerations

Where things get a little complicated is in the wet path when using multiple wet effects. Lets go through some different cases. These cases apply to any wet-dry-wet configuration.

One wet effect: This is the simplest case. The effect block should be stereo, and with the mix set at 100% so you only get the wet effect through the stereo outputs. Any block that has a Spread control should have that set to 100% as well in order to spread the stereo effect hard left and hard right. This is the typical wet-dry-wet configuration. You can control the balance between the wet and dry effect by adjusting the Dry Thru parameter on the send block. Note that if the effect block is turned off, then the Dry Thru output of the send block will go directly to the Left and Right wet speakers. In this case, you just have a a three speaker FRFR setup instead of one or two (for simple stereo). That’s probably ok, but you may need to adjust the Dry Thru level so that the volume doesn’t jump too much when the effect is turned off.

Two wet effects: If you’re only using two wet effects at a time, then you can put one on path 2A and the other on path 2B, turn both effects on and set them to 100% wet. This keeps the effect separate and independent, and can result in a more distinct tone.

Four or more wet effects: This is where things get tricky since Helix only has  three additional parallel paths. The problem is that any downstream block is only going to see the wet output of any upstream block, not the Dry Thru. This may or may not sound good. So you need to think about the effect placement on the parallel paths and in series on either path. Effect order applies to dry (mono) wet-wet (stereo) or wet-dry-wet (stereo with center channel) configuration. The only additional consideration for the wet signal path is that at least one effect in the has to be 100% wet.

Here are some rules to help avoid issues with effects only seeing the wet output of preceding effects:

  1. At least one effect block in each wet path needs to be on, or the Dry Thru will also go to the web left/right outputs. That may or may not be bad, but it reduces wet-dry-wet to be closer to just stereo.
  2. At least one block on each wet path needs to be set 100% wet to prevent the Dry Thru from going into the L/R wet speakers.
  3. Any block that is off will have no impact on the wet output of any preceding block.
  4. Any block that is off will have no impact on the dry input of any succeeding block.
  5. Any block that is on, but has its mix set to less that 100% will have controllable impact on the output of any preceding block.
  6. Prefer putting modulation effects in series on a path because its less likely to have more than one on at the same time.

In the example above, the effects are placed as they are because:

  1. Reverb is on all the time and is usually the last effect in an effect chain, so its placed at the end of the wet path.
  2. Reverb mix is set to 50% so that it will allow the output of any preceding upstream block to come through.
  3. Delay is usually used with reverb, but we want both the delay and reverb to see the Dry Thru input: we don’t want to “delay the reverb” or “reverb the delay”. So delay and reverb should be put on parallel paths when possible.

For a more pure wet-dry-wet, snapshots or footswitch controllers can be used to modify the reverb mix when another effect is enabled on the same path.


We’ve spent a lot of effort in the sections above trying to split the wet effects into separate amps, and in pure stereo. While that is useful because it keeps each effect as separate from others as possible, it’s probably not all that important to ensure the wet amps have only wet signal, and we’re not putting too many effects in series. The main thing with a wet-dry-wet setup is to separate the dry amp in the middle from the wet effects on the left and right. It doesn’t matter that much if the wet amps have dry signal too, it matters more that the dry amp doesn’t have the the wet signal too. You may also find that having dry signal mixed into the wet amps will provide a fuller, more cohesive sound while also being a lot simpler to setup.

No matter how you setup your guitar signal path, its often going to end up mono in a live FOH mix, or maybe stereo in a recording. So all the separation and spatial effects you get from a wet-dry-wet setup with three amps in a room are going to be somewhat lost in the mix. But the separation of the wet and dry signals into different, independently controlled signal paths, and attention paid to the effect order of series effect paths can really help produce clear and more controllable tones, especially live in the room. Give it a try!


Selecting an IR

Impulse responses depend lot depends on the selection of the speaker and cabinet to capture, the technique and gear used to do the capture, and the ear of the person doing the capturing. For example, S-Gear comes with a number of carefully chosen Redwirez IRs. In a recent update, Mike Scuffham created a few of his own IRs, some using similar speaker models to what was available from Redwirez. I think Mike has an amazing ear for guitar and it shows in these speaker models.

To me, its hard to tell sometimes which IR model is “better”, in many cases they’re just different. You can waste a lot of time trying to pick the best one, only of find the next time you try you’ll get a different result. Here’s some guidelines to consider:

  1. Use only 16 bit, 48 kHz IRs for Helix (don’t depend on Helix to convert them)
  2. Pick a small set of speaker IRs to audition based on what’s typically used in the amp your using, for your style of music, and your guitar (single or double coil pickups make a big difference). I started with Robben Ford and Matt Schofield’s live rigs because I love the tone they get.
  3. Get the speaker choice right first, or at least a very small set, using a Neumann U87, CapEdge and 2″ for a relatively neutral, uncolored mic. Then zero in on the mic and mic position
    1. For the very small set of chosen speakers, pick a small set of mics: an SM57, Neumann U87, and Royer R121 will sound quite different, the other choices can be pretty subtle.
    2. For each mic, pick a small set of mic positions, CapEdge being the most typical
    3. For each mic position, pick a small set of distances from the cabinet: from say 0″ to 4″ with 2″ a good starting point
  4. Import the IRs into Helix using some unused or expendable IR index slots
  5. Audition in a typical live setting. What works well at low volume by yourself might not work at all in a gig situation, a lot depends on volume level, feel and how the speaker fits in the mix.
  6. Audition by using pedal edit mode to increment and decrement the IR block while playing. Try a range of songs, pickup combinations, effects, etc. Get a feeling for the whole, not just one specific thing.
  7. Keep notes on each IR, how it sounded, how it felt, whether it was muddy, fizzy, scooped, articulate, etc. so you can remember how they compared. Use a table in Apple Notes or Evernote to capture your notes
  8. If you still can’t decide which IR to use, try this simple selection process:
    1. Select an IR
    2. Compare it with each other IR until you find one you like better
    3. Replace the first IR with that IR and repeat the process with the remaining Its until there are no IRs that sound better. You now have your favorite – but possibly only for that situation.

Getting the right IR can have a big effect on tone and feel, perhaps nearly as important as the guitar. But much of the variability between IR models will be lost in the mix, and imperceptible to your audience. So don’t worry too much if you can’t decide which one you like best, probably may choices are good and are just different. And don’t worry too much if you keep changing the IR in your goto patch, its likely the change will be subtle.

IEMs and Digital – With Feeling!

I just watched That Pedal Show – No Speakers! Can We Go Direct & Enjoy It? and it was clear from the pain on Dan’s face that he did not enjoy playing through headphones with a silent amp behind him. Those of us who have migrated to digital understand this pain. Over the last couple of years, I’ve done a lot of live experimenting to figure out how to get a manageable stage volume with a tone that helps inspire me to play while protecting my aging hearing. Its been a challenge that’s also somewhat expensive and not entirely successful. I’ll share what I’ve tried here in the hopes that maybe it will help someone, and that maybe someone who has discovered a better solution can help me.

Guitar Amp, Cabinet and Pedalboard

This is the traditional backline setup. The base tone is heavily influenced by the cabinet and speaker choice, and is driven usually be a clean amp, or one that is on the edge of distortion. Then most of the effect including compression, wah, distortion, chorus, delay, reverb, etc. are mono and going into the front of the amp. An emerging variation is to get some tone and distortion flexibility through preamp distortion and channel switching in the amp. Another is to split the effects into tone/drive effects in front of the preamp distortion, and modulation and ambient effects after preamp distortion using the amp’s effects loop.

This gives the best amp in the room tone, and a nice thump that both you and your guitar  feel. This can result in some nice natural sustain as the guitar body and strings intact with the sound coming from the speakers behind you. For the guitar player, this is the best option. For everyone else, not so much.

First the tone from a guitar amp is very directional. Anyone in the audience standing directly in the beam of a 12″ speaker will get their head blown off. Move a few inches to the side and all the high end disappears. Second, its usually loud, really loud. This not only exposes you and everyone around you to potential hearing loss, it also can contribute to an overall poor mix of band. Its hard to bring up the PA mains enough to get above the stage volume produced by these 100 Watt monsters without having the overall band volume get completely out of control. Then the club owners get upset because the band is driving people out of the club, and the bartenders can’t hear their drink orders.

One workable compromise is to use smaller amps like a Fender Blues Junior, and face them away from the audience. But then that magic thump can go away because there’s not enough power to move air.

What we need is to find a way to get better control of stage volume, improve the overall band mix, minimize stage footprint, reduce setup time and protect our ears, all while maintaining a tone and feel that inspires us to play. This is not a simple problem.

Helix with Wedge Monitor

The first solution I tried was to ditch the backline amp and use Helix into a wedge monitor. We all had JBL EON610 powered monitors that were used as vocal monitors. So it was simple enough to use them for this purpose and most of us performed without stage amps (one guitar, bass, keys).

This worked ok. It reduced the stage volume a little because instrument and vocal monitors were pointing away from the audience. Although we had to be careful with the positioning of the monitors and keeping the monitor levels down or they would reflect off the wall behind the band back out front and compete in a very bad way with the FOH mains.

But there were issues. Its hard to find space for five floor monitors in some small clubs, and they tend to be right where we and our fans would trip on them. Having my guitar tone in front of me, and looking right into that horn didn’t feel or sound right. It was too bright and I was just use to having my guitar tone behind me and warmed up by being off axis. I suppose what I was hearing more closely represented what the audience was hearing through FOH, but it tended to choke me up instead of inspiring me to play.

I still had to wear ear plugs to protect my ears. This warmed up the tone a little, but not in a way that sounded good helped inspire me to play.

Helix with IEM

The next thing I tried was to get rid of the wedge floor monitor and go IEM. This turned out to be pretty expensive, but well worth it. I got Gorilla Ears dual driver IEMs and love them. Unfortunately they are no longer in business, but there are plenty of good options for quality IEMs.

IEMs had the dual benefit of protecting my ears while also giving me much better control of what I was monitoring. Each member of the band can have their own stereo monitor mix (we use an X32-Core with two SD16s for the PA).

The biggest issue with this approach is that we still had acoustic drums, and one traditional backline guitar rig in the band. So this made it very difficult to get a good FOH mix that blended the stage volume with the FOH mains. The band tented to sound like one lead guitar and drums a lot.

This also didn’t have any feel at all. My guitar didn’t see anything from a speaker, so the sustain was reduced. As a result, this tended to be the worst option for me personally in terms of having my tone inspire playing.

Helix with Stereo FRFR and IEM

What I’m using now is a compromise that sort of combines all of the above in order to leverage their benefits and minimize their issues. I use two JBL EON610s as a stereo FRFR for my Helix, but positioned as a traditional backline behind me and on the floor. The big volume knob on Helix controls the volume of the backline through the 1/4″ outputs. I send a fixed stereo signal to the FOH using the Helix XLR outputs. I try to set the FRFR volume to match, or be a little below the drums, using a dB meter to determine the typical range for the the big volume knob.

What I like about this setup is that the FRFR backline provides a traditional feel for me, and a balanced stage mix with the drums and the other guitar. I can still rely on the FOH to project the guitar into the room, and to preserve the high frequencies that are often lost when the floor fills up with dancers.

I use the IEMs to protect my ears (mostly from the drums) and to bring back the high end that is lost through the ear protection. The way I set them is I start with my headphone amp (a Behringer P2) turned all the way down. During sound check, I turn it up until I can just hear my guitar and vocal in my IEMs, balancing the low frequencies that naturally bleed through the IEMs. This is a great combination of feel behind me and consistent and controlled high end tone in my ears. Setup isn’t too bad, and the two EON610s provide a nice platform for the PA, reducing stage footprint a bit.

So now the everyone in the band is using a backline, and three of us are using IEMs. We’re making a conscious effort to keep the stage volume down, but realize the drums are the limiting factor. We’re protecting our ears while maintaining tone and feel.

I’m pretty happy with the results so far, but it is a different feel that takes some time getting use to. Having your guitar directly in your ears instead of indirect from bouncing off the walls in the room isn’t, as you would expect, that “amp in the room” tone. So its not perfect, and is something I need to adapt to. But this seems to be a good compromise, at least the best one I’ve been able to come up with that meets my needs and fits with the rest of the band.

Controlling Stage Volume

A common problem for semi-professional bands is controlling stage volume. Acoustic drums, plus a couple of big guitar amps and a huge base amp can produce a lot of sound. This can result in the stage volume that overwhelms the FOH volume, kills people on the dance floor near the band, and produces an overall unbalanced and poor mix for the overall audience. Being too loud also makes your singers work harder. They can’t explore and use their voices to its greatest potential because they always have to push it to 10 every time the band plays.

Getting stage volume under control is the first step in improving the overall sound of the band. Learning how to get your sound, at a manageable level for a sound guy to deal with it, and the room to sound good, should be a minimum requirement for a performing band.

Here’s some ideas on managing stage volume I’ve collected from around the web.


The first step in controlling stage volume is to know what it is. A band playing in a small club is going to be quite close to the audience. To protect their hearing, and provide an exciting but pleasant listening environment that encourages people to stay, the band should keep their maximum sound pressure level under 100dB. The sound man can use a dB meter on their phone to measure the SPL at various points in the room and use that to assess and manage the stage volume level.

Here’s a very useful site on hearing loss; http://www.dangerousdecibels.org/education/information-center/decibel-exposure-time-guidelines/: For every 3 dBA over 85dBA (8 hours exposure time), the permissible exposure time before possible damage can occur is cut in half. So at 106 dBA you can get permanent damage after 4 minutes of constant exposure.

Note: dBA levels are “A” weighted according to the weighting curves to approximate the way the human ear hears. dBA is what’s relevant to hearing loss (has a high pass @ ~100Hz), dBC is relevant to noise pollution and includes nearly all sub frequencies which generally don’t damage ear hair cells (but can damage ear drums if they’re crazy loud). So use a SPL meter with dBA reading capability.

Going full DI with IEMs a great way to minimize stage volume, and the whole band gets sent to the audience via the PA. However, it’s going to take a lot of persuading to get everyone to stop using a backline. Guitars and guitarist both respond to and rely on the interaction with their backline amps. IEMs can’t provide this physical interaction: the thump of the low notes, string feedback and a feeling of space. IEMs do protect your hearing, and can provide a good monitor balance for the whole band without having to contribute to stage volume with holdback speakers for vocals. However, this isn’t a practical solution without a great sound engineer, as it’s impossible to sort out the band mix from behind the PA. Also if using acoustic drums, or any backline instruments, the sound close to the band will be mostly just those instruments and will not sound good at all. Its better to be consistent – all backline or all DI and no backline.


Another source of stage volume is your monitors. Monitor wedges can contribute to stage volume if they reflect off a wall behind the band right into the audience. In small clubs, these reflections can be louder than the FOH speakers, but a much lower quality. Start by getting rid of anything in your monitor that you don’t absolutely need – especially if you are able to have your own mix, or even if several people have their own mixes. Try moving to IEMs to reduce stage footprint and volume as well as setup and teardown time. But IEMs are not a solution for overloud and unprofessional stage volume. Minimize monitor volume (IEM or not) by making sure each performer gets exactly what they need and nothing more.

I some cases, especially in small clubs, it is possible that the need monitors indicates the set up is wrong, making everyone feel the need to turn up even more.


As for cranking up guitar amps to get good tone, it’s really a loose-loose situation. If your amp is running loud, it might have good tone for you, but that beam of sound from your amp’s guitar speaker mucks up the rest of the mix. The sound guy can’t turn you up a lot in the house – which means uneven coverage, so some people might be getting a lot of guitar, while others won’t here you at all. So much for your good tone.

Especially for overdriven stuff, as guitar players, we like having the low end kind of rumble from our amp that we can feel, but sound guys (at least good ones) roll that off in their front mix so it doesn’t compete with the bass and kick. By being loud, you just muddy up the overall band sound and your good tone is thrown out the window.

Use a good sounding clean channel and some pedals to get a good overdrive sound without having to raise the volume too much. Or use a digital amp like Helix to produce the amp tone you want at any volume level.

Also, if you can stomach it, rolling off the low end on your amp, and cranking the mids up will allow you to hear yourself better without increasing the volume.


The best way to reduce bass stage volume while maintaining better FOH control is to split the bass signal, sending signal to both the bass amp and to a DI into the FOH. Then EQ the bass amp so it was really mid-rangy/punchy, more than you would normally ever think to be. Then, the FOH could bring out all of the lows and sparkle. What this allows you to do is keep the bass stage volume down, because what you hear on stage really cuts through. And, there is now bass everywhere because it’s in the FOH subs. This also allows you to play a lot tighter as a band, because you can really hear the attack on the bass player.


Fortunately MIDI keyboards are the easiest instruments to control and often aren’t the primary contributors to stage volume. Managing the keys is mostly covered by positioning them in the mix. Keyboards do produce a wide range of frequencies. Heavy left hand players can produce a lot of low frequencies that compete with the bass and kick drum. There’s also a lot of competition for mid frequencies between the keys, guitars and vocals, with the potential for low-mid buildup that produces a muddy, indistinct sound. Some EQ might be useful to carve out space for different instruments all competing for the same frequencies.


Vocals should have the low end rolled off quite a bit to reduce bleed from the bass and drums, and eliminate plosives from singing up close to the mic. Turn on the vocal high-pass filter and set the frequency around 100 Hz – that makes things much clearer. Also, your vocalists should just about be eating the microphones – this reduces what the gains have to be set to, and also reduces the amount of bleed to the mics from other instruments.


Avoid using a drum shield because in many situations they don’t help. If the drummer has a wall behind him, or some sort of cement or hard wood floor then sound is just bouncing around until it eventually comes out in a worse form than it would have without the shield. Plexi screens can also make the drummer feel isolated like they’re not apart of the band. The can also take away tone and punch, forcing you to mic.

Look into having lighter cymbals instead of thicker darker ones. Smaller drums help too, as they have less boom on stage. Some drummers are really keen on Yamaha’s Hip-Gig kit, simply because it’s small, doesn’t project much, but can sound huge when mic’ed. If you are mic’ing the drums, the lows can be brought up, so there’s no need for a lot of that on stage.

Another possibility is to use http://www.lidwishsoulutions.com sticks.

The main goal is to keep the drums from going out through the vocal microphones which is what happens to many bands who don’t use shields.

Drummers sometimes play hard. But that doesn’t necessarily mean its really loud. Try tuning drums pretty low (1 high tom, two low toms/bonham set up) so its not such a high pitched and piercing attack. Use muffled heads: Remo pinstripe top/ coated emperor bottom. For cymbals use Zildjian K’s, which are pretty mellow, but cut through and sound beautiful.

And it all depends on the venue. Carpet under the kit, the right head combos, muffling the bass drum. There’s not much you can do for cymbals. The right cymbals won’t be too loud at all. Certain cymbals (A Customs) are loud no matter how light you try to play them. Zildjian K’s and Sabian AAX may be useful in small venues. No matter how hard you play, they always sound even.

Hot rods sticks work great for acoustic settings, but they’re only useful for low volume acoustic/light practice.

Drums are all about mixing in, you want them to be heard, but not in everybody’s face. The mellower everything on the kit sounds on its own the better it can mix at higher volume.

Drummers who use IEMs may end up playing even louder without knowing it since they aren’t hearing the direct kit. To help the drummer using IEMs manage their stage volume:

  1. If he’s using universal fit, lead him toward better-isolated earbuds.
  2. Put very little in his IEM mix besides drums, just a little of lead vocal and whatever he needs to keep his place in the songs. He’s less likely to play louder if he’s already threatening to swamp his own mix at sound-check volumes.
  3. Give him a way to make just his drums louder in his IEMS (if you’re using a wifi mixer, put a smartphone or tablet at his drum position controlling just his IEM mix)
  4. Hard limit his drums just in his IEM mix so that playing drums louder either leaves his IEM volume stable *or* kicks in (intentionally) nasty sounding limiting on his drums in his IEM mix.
  5. Combine one or more of the above with IEMs for the rest of the band so that you and the singers can at least hear what you need.


Finally, the biggest challenge (and often most overlooked) is to play as tight as you possibly can, and everyone should be focused on the song a a whole, not just their individual part. This requires discipline, practice and consistent rehearsals. Often, we find ourselves tinkering with the sound, playing at higher and higher volumes, etc, and not really addressing the thing that will improve our sound the most. The tighter and more under control you play as a band, the clearer the sound will be, the more effortless it will be for the sound guy to make sure everything is heard and the easier it will be to keep your volume down. Playing tighter also allows things to sound clearer when the volume is way up too.

Using a DAW for band rehearsal

Ideally your band should rehearse with the same gear and setup that you’re going to use when performing. That way you can tweak your tones in a realistic setting, ring out the gear, and have a pretty good idea what you’re producing for your audience. But that often isn’t very practical unless you rent expensive rehearsal space.

Finding the time and a place for rehearsals is often the biggest challenge for bands. One of the common causes of band breakups is lack of rehearsal space. What makes finding rehearsal space so hard is there just aren’t that many places you can get five or more guys in a room with all their guitar and bass amps, drums, and big keyboards and let them go. Its way too loud to impose on our families and neighbors. I can’t tell you the number of times neighbors have called the police on me over the years because of loud rehearsals.

So we need a solution. We need to find a way to take up a lot less space, make a lot less noise, and be flexible about rehearsal location so that we can rotate among the band members and not put all the burden on one person. The answer is to go digital.

Using your DAW as a Digital Mixer

Most of us have reasonably powerful laptop computers and a Digital Audio Workstation (DAW) that we’re pretty familiar with. We typically use this for home and possibly gig recording. But that laptop, a good audio interface and DAW can also be used as a digital mixer that can provide the foundation for your Digital Rehearsal System (DRS). Here’s what you’ll need for your DRS:

  1. A Computer with sufficient processing speed and memory, say minimally an i5 and 8GB ram
  2. 8 channel audio interface (or more)
  3. Multi-channel MIDI interface
  4. DAW (e.g., Reaper, Logic Pro X, set.)
  5. Plugins for guitar and bass amps and effects
  6. Software instruments for keyboards and drums
  7. Microphones for vocals
  8. Keyboard controllers
  9. MIDI drums
  10. Headphone amp and headphones for monitoring

A lot of this you probably already have and just need to put together your DRS template in your DAW and assemble all the parts. Creating your DRS consists of creating the tracks for the instruments, configuring the inputs and outputs, choosing audio plugins and MIDI software instruments, setting up MIDI controllers for turning effects on and off, etc. We’ll be going through each of these steps using a specific example.

Digital Rehearsal System

My band, Moonlight Rescue Band, spent some time last year building up our DRS. Our goal was to setup a rehearsal space where we could practice, rehearse and record almost anytime without disturbing anyone else in the house, let alone the neighbors. Vocals in the room are the limiting factor to how quiet we could get, but we wanted that and the sticks on the MIDI drum heads to be the only noise we produced. We had enough spare gear between us that this setup uses very little of the gear we use when performing, and can stay setup all the time. Here’s the components our our DRS:

  1. 2015 MacBook Pro with 8 GB ram and 250 GB SSD. This is a core i7 machine and is perfectly fast enough to run the whole band and then some.
  2. Focusrite Scarlett 18i20 USB audio interface. This relatively inexpensive interface provides two Hi-Z inputs for the guitars, and enough other analog inputs for bass and five vocal mics. Everything else is MIDI.
  3. MOTU Micro Lite USB MIDI interface. This provides five MIDI inputs for drums, two MIDI keyboard controllers and two FCB1010s used to control guitar effects. You can get by with a smaller MIDI interface by daisy chaining your MIDI devices. But this can introduce latency in the MIDI signal chain and should be avoided.
  4. Logic Pro X. This is the DAW we used to provide the tracks, plugins, software instruments, etc. All the keyboards and drums are provided by Logic software instruments. I also created a similar setup with Reaper using SampleTank for software instruments.
  5. Behringer HA8000V2 8-Channel Headphone Amplifier. There are two headphone outputs on the Scarlett 18i20, but that’s not enough. Also the Behringer provides individual treble and bass controls, and a mono/stereo switch for each channel. Everyone is listening to the same stereo out mix, we don’t use separate monitor mixes for each performer during rehearsal. This keeps things simple and makes sure we’re all hearing the same whole during rehearsals. So these extra controls on the Behringer headphone amplifier can be handy for addressing some individual preferences.
  6. Headphones: Since the volume in the room is quite low, you can use open back headphones which are more natural sounding and comfortable. Some of us use our performance IEMs in the studio as these sound even better, and it helps us get use to using them. I use Gorilla Ears IEMs and am very happy with them.
  7. Stereo power amp and stereo speakers. This is something we already had, and so we hooked it up in case we wanted to listen to something in the room. It doesn’t get used very often, but can be handy.
  8. Five vocal mics and stands
  9. M-Audio Core40 and M-Audio Keystation Pro 88 keyboard controllers and stands. The Core40 isn’t a performance instrument as it is just a controller. But it works great as a flexible studio keyboard controller. Any keyboard that has a MIDI output will do though.
  10. TD-6V Roland drum generator and VH-11 Hi-Hat. We only use the TD-6V to send MIDI output to the computer. The actual drum sounds are provided by Logic’s drum kit designer. This way we can easily design the drum kit, and record each drum on a separate track for later mixing if desired.
  11. Two guitars and bass. We have enough extra instruments so these stay in the rehearsal space all the time. They’re not our best gear, but good enough for rehearsal and quick recordings. Having things ready to use in the rehearsal space saves a lot of valuable rehearsal time.

DRS Mixer

The tracks are grouped similar to how we are setup in the PA for live shows. Track 1 is used to capture simple stereo recordings of the whole band. More on recording below.

Vocal Tracks

The vocal tracks are pretty simple. I just used the Logic Pro X Natural Vocal patch and made a few changes, similar to what I would do when mixing male vocals. There’s some reverb and delay on the vocals to provide some space in the headphone mix. I didn’t pan any of the vocals becase all five of us sing lead in various songs.

The key to getting good vocals is to ensure the mics are properly gain staged. You want to set the audio device so that you’re getting around -18dB into the DAW track. Then bring the track fader down to about mid scale so that the sum of all the tracks don’t clip the stereo output bus. Use the headphone amp to make up the gain as needed to avoid any possibility of digital clipping in the DAW.

All the vocals are sent to the same delay bus to ensure the echo repeats are aligned with all the voices. A single reverb bus is used too so the all the tracks appear in the same space. This keeps the mix simple and makes it easier to distinguish parts during rehearsal.

Guitar Tracks

There are two guitar tracks. I’ve tried to set them up to be similar to the configurations we use for live gigs. These are panned a little left and right so the two guitars don’t sit right on top of each other. We pan them slightly in live gigs too.

Logic Pro X comes with a number of guitar and bass amp models, and a virtual pedalboard. These models aren’t bad, and would be perfectly acceptable for rehearsal amps. However, S-Gear from Scuffham Amps is one of the best sounding guitar amps (virtual or not) I’ve ever heard, and I already had it for recording. Since I have a Helix, getting Helix Native was also a good option, especially so that my rehearsal and live rigs can use essentially the same patches. Helix Native also works very well in front of S-Gear as a virtual front of the amp pedalboard. Positive Grid’s BIAS FX is pretty good, as is Amplitube, and AMPLIFIKATION ONE. There are lots of options for digital amp models, so there should be no problem finding something that works for you.

Steve’s Guitar

Steve uses a Fender Blues Junior for live gigs with a front of the amp pedalboard for distortion and occasional chorus. For his amp I used S-Gear Wayfarer as it has the most Fender sounding and flexible.

Steve's Pedalboard Amp.png

S-Gear doesn’t come with any front of the amp effects, so I used the free TSE 808 V2 TubeScreamer plugin for an overdrive tone. The distortion tone uses Wayfarer Lead II which is nicely voiced for heavy distortion. S-Gear MOD thing is used for chorus. These are all controlled by an FCB1010. Details on how the MIDI mapping are done are covered below.

Jim’s Guitar

I use a Helix floor into two JBL EON610s for my stage setup, along with a JTV-69S as my main guitar. So I used Helix Native with the same patches for my rehearsal setup.

Jim's Amp.png

See Creating a Helix Electric Guitar Patch (newly updated) for a complete description of this patch. The track includes the typical compression and EQ plugins that would be used for recording and mixing, but these are set pretty neutral.

Since I don’t have a VDI cable to connect the Variax to the computer, I use an older Variax Standard for rehearsals, and have to switch all the guitar models by hand. I kind of wish I had kept my old X3Live or HD500X to use as a Variax input device, foot controller and backup for Helix.

FCB1010 Configuration

Configuring the FCB1010s to control the guitar tracks, and recording the automation was by far the most difficult part of the DRS setup. First the FCB1010 is itself quite complex to program. Then programming controller assignments in Logic Pro X adds a lot more complexity on top of that. I’ll cover the high points of what I learned here and hopefully this will provide a reasonable guide to help you with your specific setup. I’ll only cover the setup for my track (Jim Gtr), Steve’s was similar.

Let’s start with the FCB1010 itself. The FCB1010 as shipped from Behringer is highly programmable and flexible, but its not well designed for controlling a virtual guitar pedalboard. The reason is that the LEDs with the switches show what switch was last pressed, not the state of a virtual effect. The market has recognized the value of the FCB1010 in terms of low cost, reliability, and programmability, while also recognizing the commonly missing features. There are options, but they require changing the EPROM chip in the unit, the FCB1010 does not use a flashable ROM like most modern devices. Its a bit of work, but inexpensive and well worth doing.

The FCB1010 UnO firmware and FCB/UNO C’ontrol Center address some of the bugs and missing features of the original FCB1010. In particular, it allows you to designate a row of switches to be used in “Stomp Mode”. Stomp mode allows the switches to show an off/on state that can be mapped to the plugin bypass/enable state. That’s better, but still not enough. I want to be able to use all 10 footswitches in stomp mode to control all the effects I use in my typical patch.

Luckly there’s another option, EurekaPROM3 from EurekaSound. EurekaPROM3 is not an extension to the original FCB1010, its completely redesigned firmware that although it has more limited programming, it makes the FCB1010 much more useful for controlling virtual guitar effects. The chip comes with a number of selectable configurations for specific devices including Kemper, Avid Elevan Rack, Line6 POD, etc. All these configuration have a similar layout, functionality and features:

  • Four or Five Modes (depending on configuration), easily switched between using the Up/Down pedals:
    • Effects Mode: Turn individual effects on/off (like stompboxs), with effect state reflected on the pedal LED for all 10 pedals
    • Presets Mode: Easily enter and send presets
    • Favorite Presets Mode: Favorite Presets Mode allows you to store and recall your 10 favorite presets with the push of a pedal!
    • Specials Mode: Allowing you to access special modes within your chosen configuration
    • Effects & Presets Mode:  Pedals 1-5 from Effects Mode, and pedals 6-10 from Favorite Presets Mode
  • Expression Pedals for wah, volume, and more for some configurations
  • MIDI activity LED

I use the PP: Programmable mode to control Helix Native and S-Gear through Logic Pro X controller assignments. There is no desktop app for programming the EurekaPROM3 PP mode, but it is much simpler to program than the Behringer or UnO chips, so you don’t really need an app. I use the default CC messages in the PP mode, so I didn’t need to do any programming at all. The EurekaPROM3 chip is well worth getting and installing as it significantly simplifies and improves the functionality of the FCB1010. Its hard to find a better solution at the same cost.

Controller Assignments

Logic Pro X Controller assignments map MIDI messages from the FCB1010 to plugin parameters of a specific track. For example, we might map foot switch 1 to turn the guitar compressor on/off. These controller assignments turn the FCB1010 into a virtual pedalboard controlling all the effects for your guitar channel strip.

Here’s the steps I use to create the controller assignments for my guitar track:

  1. Click on the new parameter you want to assign. For Helix Native you need to do these steps to select the parameter:
    • Open the Helix Native plugin editor
    • Select the block you want to control
    • Use the AUTOMATION/CONTROLLER ASSIGN tab to assign the block bypass or parameter to one of the 16 switches or 16 knobs
    • Switch to Editor > Controls view (selecting the control in the Helix Native editor does not select it for Logic)
    • Select the switch or knob you just mapped to the block bypass or parameter
  2. L or Logic Pro X > Control Surfaces > Learn assignment for “<parameter>”
  3. Press the desired MIDI controller foots witch or move an expression pedal
  4. Turn off Learn mode
  5. Set the the Channel Strip parameter to Audio, and the index to the audio track number (view channel strip Type and Number Label)
  6. Edit the control name, label, channel assignment, MIDI message and value mapping as needed
  7. Make sure the learned controller assignment is in the right zone and mode. If not, cut to remove it from the incorrect zone/mode, and paste into the correct zone/mode.

Creating controller assignments in Logic Pro X can be a bit tedious and error prone. Instead of covering all those details here, I put them in another blog post: Creating Logic Pro X Controller Assignments.

In our DRS, the DAW is essentially being used in performance mode, and is supporting multiple instruments and virtual pedalboards being used at the same time. The key things to remember when creating controller assignments for a DRS using Logic Pro X are:

  • The controller assignment Class should be Channel Strip, and the Channel strip should be Audio with the index matching the audio number of the channel strip you want to control. This ensures each FCB1010 is controlling the right guitar track.
  • Mapping MIDI CC values to plugin parameter values can take some experimenting. If the parameter takes values from 0 – 127, then you can use Direct mode for the MIDI value. If the parameter value is scaled in some other way, you will need to use Scaled mode and determine the Multiplier needed to get the value you want. Use the plugin Editor > Controls view to see the plugin parameter values as you change the Multiplier until you get the value you want.
  • Use -1 for Scale to change from On/Off to Bypass/Enable (they’re reversed) depending on the plugin parameter. This ensures the FCB1010 footswitch light is on when the effect is active.
  • Logic Pro X controller assignments don’t provide any way of creating or editing the selected plugin parameter, except through Learn mode. So if you need to change the parameter of a controller assignment, you’ll have to create a new assignment using learn mode to set the plugin parameter, copy the rest of the assignment information and then delete the old assignment.

Here’s the configuration I used for my FCB1010:


Nate’s Bass

The bass guitar plugs directly into input 8 of the 18i20. The bass track uses the exceptional Cerberus Bass Amp from Kuassa. I have tried many different options for bass, and Cerberus is by far the best. But you can also just go direct with bass and use some DAW compression and EQ to get a good sound.

Bass Amp.png

Lou’s Keys

Lou uses two keyboards for performance, a JUNO-Di for synth and sampled sounds, and an a separate 88 key device for piano. Initially we attempted to recreate this keyboard setup in the studio using an M-Audio Code49 keyboard controller to drive Logic’s EXS24 sampler plugin, and an M-Audio Keystation Pro 88 for the piano and Fender Rhodes tones. We were motivated to do this for a few reasons. First, the Hammond organ and Clavinet instruments, and the Leslie speaker plugin in Logic are fantastic, sounding much better than many professional MIDI keyboards. We also had the two MIDI controllers and wanted to use them so the keyboards could be left set up and not have to be moved for gigs. Finally, we had already used up all of our analog inputs, but had plenty of MIDI inputs available. Using all MIDI for the keyboards and drums allowed us to get by with the 8 analog inputs provided by the Scarlett 18i20.

But ultimately this proved to be less usable than we had hoped. Lou already has a set of patches he uses for songs, and knows where the sounds are on his JUNO-Di for selecting sounds for new songs. We could collect all these patches, and try to reproduce them in EXS24, possibly even trying to reproduce the JUNO-Di. But that would take a long time and wouldn’t provide the same, familiar interface Lou would need to use it. The problem is the EXS24 as a Logic plugin isn’t designed for performance, its designed for recording, editing and mixing. For example, it has no facility to respond to program change messages to go to a specific patch number, it only supports increment and decrement.

So it turned out to be much easier for Lou to use his JUNO-Di. The JUNO-Di has a USB interface, but its for MIDI only, it doesn’t support USB audio. So we needed another pair of stereo inputs. Fortunately we had another stereo Firewire audio interface that we could use for the extra pair of audio inputs. MacOS Audio/MIDI Setup allows you to create an aggregate I/O device that lets you combine two or more physical devices into a single logical device. You can label the inputs and outputs on the aggregate device to make it easier to know which inputs in Logic correspond to which inputs of the physical devices. Aggregate devices ensure audio is time aligned between the physical devices and  manages any differences in latency of the different physical devices. This is a very nice feature of MacOS Core Audio.

We still kept the same configuration of tracks for the keyboards and simply adding an I/O plugin to the Keys track so it would work with a MIDI device or the stereo audio output of the JUNO-Di.

Lou's Keys

The M-Audio Keystation Pro 88 for the piano and Fender Rhodes worked fine since there were only two primary sounds used on this device, and it was simple to change. So we continued to use MIDI for those two tracks. The Hammond and Clavinet tracks are still available and can still be driven by MIDI from either the JUNO-Di or M-Audio Code49.

Note that Logic is a bit confusing regarding monitoring of MIDI instruments. If you have multiple MIDI tracks in a project, and only one of them is armed for recording, Logic will automatically route input from any MIDI instrument or channel to that armed track. This is convenient because it doesn’t require you to worry about assigning instruments and MIDI channels if you’re using one keyboard and recording a track at a time. When you want to play different MIDI instruments on different tracks at the same time, as you would with multiple players, or performing with multiple keyboards, you have to arm all the software instrument tracks you want to monitor. Once more than one software instrument track is armed, Logic will now use MIDI instrument and MIDI channel assigned to the track, ensuring the MIDI controllers are routed to the proper channel strips. This can be a little confusing to setup. And sometimes only on channel strip or track will get armed, and all the instruments will control that track.

Geoff’s Drums

Unfortunately, a MIDI drum kit is a must for an effective DRS. If you only have acoustic drums, there’s no way to bring down the overall rehearsal volume. Fortunately we had a set of MIDI drums that had been unused for a few years that we could put back into service. For a DRS you don’t need a fancy MIDI drum controller. All the controller needs to do is send MIDI messages to the computer. The actual drum sounds are provided by Logic Pro X Drum Kit Designer. This also reduced the number of analog inputs we need. Five mics, two guitars and bass took up all eight of the available analog inputs in the 18i20, so having the drums and keys be MIDI only just fit.

Drum Kit Designer comes in two configurations, normal and producer. The normal drum kits use a single stereo track. The producer kits use a separate track for each drum organized into a track stack for the whole kit. We didn’t use the producer kit just to keep things simple. But if you do, you can record each drum separately and mix, substitute, trigger, etc. during mix down if needed.

Drum Kit.png

Recording Rehearsals

Recording rehearsals is a great way to review how the songs are progressing, see what needs to be corrected, capture the arrangement, and maybe even capature tracks you can use for band marketing. These rehearsal recordings might also be better than what you could typically capture at gigs. Its a bit tricky to play, sing, run the mixer and record all at the same time, so keeping it simple is important, or you won’t get any recordings. Logic Remote can also be very helpful in this case since you can control recording start/stop right from your iPhone or iPad using a simple touch screen. This is a lot easier then bending down trying to click on tiny icons using a mouse while you have an instrument and picks in your hands.

There are two basic ways to record rehearsals: stereo or multi-track.

Stereo Recording

This is the simplest approach. It captures a stereo recording of the whole band in a single track. This is often fine for rehearsals as its all that’s needed to review the material and capture the arrangement. You can also use these simple recordings to help adjust the mix levels, which can be very hard to do while you’re also performing.

Choose a bus to act as the Stereo Rec output. Set the output of all tracks to this Stereo Rec bus. Create an Audio track also called Stereo Rec and set its input the Stereo Rec bus and its output to Stereo Out. Set input monitoring and arm this track for recording.

Set Preferences > Audio > General > independent monitoring level for record-enabled channel strips to off. This way the mix we hear while performing is the same as the recorded mix.

This will record the MIDI for the keyboards and drums too since these have to be record enabled in Logic in order for the tracks to respond to the proper MIDI input channels. But that’s ok, just delete the MIDI content if you don’t want to keep it.

This has the advantage that its small, doesn’t take up a lot of disk space and needs no mixing. We record what we hear, and can use the recorded tracks to help set a performance mix. If you make sure all the instruments are gain staged at the same level, say around -12dB, on the input stage, then the relative fader levels for you mix can be translated to your live mixing console as a starting point for gigs.

Multi-track Recording

You can also record enable all the tracks, and get a multi-track recording of the performance. This is useful if you want to capture rehearsal content that you want to mix and master later into something that can be published on your Web site, Reverb Nation or YouTube.

Just record as usual. The problem is splitting out the different songs. The simplest way is just stopping and restarting recording between songs. This results in the creation of different, easily selected region files, making it easy to create separate projects out of the different song sections.  Or create a new project for each song from a DAW Rehearsal template and do Save As… for each new song. This eliminates the need to break anything up later, at the small expense of having to create and save the song projects with a guitar in your hand.

Multi-track recording has the advantage of capturing the individual tracks so they can be mixed later. It also captures the mix levels so we can analyze the levels and panning during playback and save that for a better mix while we’re playing. The down side is that the tracks need to be mixed down and take up a lot of disk space.

Recording Automation

When doing multi-track recording, the guitar tracks will need to record the automation of the parameter changes done with the FCB1010 foot switches. These parameter changes are recorded as track automation using ‘Latch’ mode. When using latch mode, Logic always applies the current automation data at the beginning of playback, regardless of how the switch or control has been set prior to playback.  This makes sense, because latch mode should only start to record data after some activity on the parameter.  Automation ‘Write’ mode will follow (apply) the current switch position right from the start of the track. For example, say you start recording a track with automation in Write mode. At some point, you press a footswitch to turn on a distortion block/plugin. Logic will write the automation at that point, which will effectively turn on the distortion block all the way back to the beginning of the track. This is clearly not what you want. Use Latch mode instead to avoid this issue.

Note: If you learn an assignment in Logic then you can’t record its MIDI data, but you can record its automation. If you do not learn an assignment for a MIDI controller, then you can record its MIDI data if it is sent to the sequencer. This may apply to MIDI CC messages comming from the keyboard controllers. Since these are going to MIDI tracks, the CC messagers can be recorded directly in the MIDI tracks, along with the MIDI note on/off messages. However, any MIDI messages that are used in controller assignments will not be recorded to the MIDI tracks, but can be captured in the automation for the tracks.


This was a long and complex post. I didn’t go into all the details you’ll need to address to setup your particular DRS using your DAW of choice. But this should give a good overview of the process and help you avoid some of the issues we struggled with.  We’ve been using this DRS for the last year and find it to be very convenient, flexible and really improved our rehearsal experience. To some band members, using the DRS was the first time they’d ever really heard the whole band while performing. This was a lot of work and not a trivial expense. But it was well worth the effort and is making use of some gear that was just gathering dust. Best of luck, and happy rehearsing!



Creating Logic Pro X Controller Assignments

General Notes

I use an FCB1010 with the EurekaPROM3 chip, or an Apogee GiO with Logic Pro X to control a virtual guitar rig configured in a channel. I typically use S-Gear, Helix Native, BIAS FX, or Amplitube 4 to configure the channel strip, and use Logic Pro X controller assignments to map MIDI messages to plugin parameters to support a virtual pedalboard. This post explores some of the details I’ve discovered in creating and using controller assignments in Logic for this purpose. I have done similar things with Reaper, so these notes should help you configure a virtual MIDI pedalboard for most any device and DAW. At least it will show you some of the things you need to consider.

Creating controller assignments in Logic Pro X can be challenging. The process for creating, organizing and editing controller assignments can be tedious and error prone. In addition, controller assignments are stored as user preferences, and are not stored with the project. I suppose this is because the DAW designers considered the controller assignments should reflect the physical studio setup and be used consistently from project to project. Personally I hope Logic Pro X changes this in some future release as they have in MainStage. Creating controller assignments in MainStage is much easier, can be done with or without Learn mode, allows easy editing of all controller assignment parameters, and is saved in each concert or concert template for easy reuse.

Here’s some general notes on controller assignments in Logic Pro X that I’ve discovered over time.

  1. The Logic Pro X Controller Assignments view does not provide any way to edit the parameter of a controller assignment. The only way to select a parameter is using Learn mode. To change the parameter being controlled, you have to delete and recreate the controller assignment with the correct parameter.
  2. Some Logic Pro X plugins allow parameter selection on the plugin Editor view. S-Gear and all the Logic plugins do this. Other plugins like Helix Native and BIAS FX don’t. You have to switch to the plugin Controls view to select the parameter value when using controller assignment  Learn mode.

  3. A quick way to map a parameter is to select it in the plugin editor or generic Controls Editor view, and do CMD-L followed by activating the desired controller on your MIDI device. This automatically learns the mapping in the Controller Assignments. You can then edit the name and label of the assignment, and move it to the desired Zone and/or Mode using Cut and Paste.

  4. You can leave Controller Assignments in Learn mode and quickly map MIDI messages to different plugin parameters. But you have to do this in a careful sequence of parameter selection followed by pressing a MIDI controller or you will overwrite previous assignments. I find it simpler to do one at a time and turn off Learn mode right after selecting the parameter and MIDI message, and before editing any other controller assignment parameter.
  5. Logic Pro X controller assignments try to guess at what MIDI message to use and what mode (Direct, Scaled, Rotate, etc.) to use when mapping the MIDI message to the selected parameter. These values are often incorrect, but are easily edited in the Controller Assignments view.
  6. When you choose a MIDI input (port) from the pop-up menu, all assignments that use the same input are changed accordingly. If the assignment belongs to a supported control surface, the device’s MIDI input also changes in the Setup window.

    This makes it easy for you to create default assignments for a new control surface. These new assignments can be moved to other computers by copying your com.apple.Logic.cs preferences file. Simply paste this preference file into the Preferences folder of another computer, open the Controller Assignments window in Expert view, and change the MIDI Input parameter of one assignment (as applicable to the MIDI setup on the other computer).

  7. To use a MIDI controller for different purposes in the same or different projects, organize the controller assignments in different zones and/or modes in a zone. Then you can pick the zone you want to use for the specific project or activity.
  8. If you learn an assignment in Logic then you can’t record its MIDI data, but you can record its automation. If you do not learn an assignment for a MIDI controller, then you can record its MIDI data if it is sent to the sequencer.
  9. To see what multiplier is needed for some multi-valued S-Gear controls (e.g., Amp A/B), look at the parameter value in the plugin Editor > Controls view and move the switch – then determine what’s required to scale the MIDI message value to change the parameter value as needed.


This section lists some Logic Pro preferences that are generally applicable to controller assignments and how they behave.

Preferences > Control Surfaces: Multiple controls per parameter – use to set the number of controls that can be mapped to the same parameter.

Project Settings > MIDI > Control Change 7/10 controls Volume/Pan of channel strip objects: should generally be off so that instrument expression pedals can be assigned to specific tracks using Controller Assignments.

Control Surfaces > Setup… > New > Automatic Installation: should be turned off so that control surfaces are not automatically loaded, potentially causing overlapping assignments. Also deleting a Control Surface with Automatic Installation on will cause all controller assignments using the same MIDI device to be deleted, whether they were defined by the Control Surface or not.

Supported Devices

Directly supported control surfaces communicate with Logic Pro via special plug-in files located in the /Contents/MIDI Device Plug-ins subfolder of the Logic Pro application bundle Logic Pro also checks for control surface plug-ins installed in the ~/Library/Application Support/MIDI Device Plug-ins and ~/Library/Application Support/MIDI Device Plug-ins folders.

Many supported devices provide an emulation mode for the Mackie Control (MCU) or HUI protocol. In essence, Logic Pro recognizes the device as an MCU or HUI.

Logic Pro supports automatic assignment of hardware controls for a variety of USB MIDI controllers. This is achieved through use of Lua scripts. Devices supported by Lua scripts do not appear in the Control Surfaces Setup window, nor in the Install window.

These devices appear in Logic Pro > Control Surfaces > Preferences > MIDI Controllers. The MIDI Controllers preference pane applies to USB MIDI controllers supported by Lua scripts. Lua is a lightweight scripting language supported by Logic Pro.

Assignments created by Lua scripts are created within a separate top level, modeless, Zone that is named after the device. These assignments are always active and interact with Control Surface Group 1.


Control surface groups are in Logic Pro > Control Surfaces > Setup – the devices on the same line are in the same group. The order of the icons from left to right defines the order in which  controller tracks and parameters are arranged and displayed on the devices. These establish fader groups for making controller assignments.

Devices in separate roles are treated independently.

Control surface preferences are stored in ~/Library/Preferences/com.apple.logic.pro.cs and apply to all Logic Pro X projects on that computer for that user. Different control surface preferences can be created for different users on the same computer if needed.

Not all MIDI devices used as control surfaces need to appear in the Settings dialog. Those simply have pre-defined mappings that can be reused.

Each control surface must be connected to an independent MIDI In and Out port (or corresponding USB/FireWire port, designated as a MIDI port by the device driver). When the device is added, the automatic setup or scan procedure sets the appropriate MIDI input and output port settings for the device. If the MIDI port settings are incorrect, you can manually choose them from the Input and Out Port pop-up menus.

If a control surface device appears to stop working, then this automatic assignment may be incorrect, possibly because it was changed when the device was disconnected. Use the Setup menu to correct the MIDI ports.

Controller Assignments

You do not explicitly save controller assignments or related preferences and settings. These are automatically stored when you quit Logic Pro.

Generally use Expert mode to create controller assignments. You can only switch back to Easy view if a track or plug-in parameter is selected because that is the only kind of mapping supported by Easy view.

Controller assignments for track parameters must select:

  1. Target Channel Strip: Which track should receive the MIDI message
  2. Target Parameter: What parameter should be controlled by the MIDI message
  3. Value Change: What the MIDI message is and from what MIDI device
  4. Value: How the MIDI message is modified and or interpreted to set the parameter value

Each of these is covered in detail in the sections below.

Target Channel Strip

This determines what channel strip will be the target of the incoming MIDI message. For control surfaces, this would typically be Fader Group and an offset index within the Fader Group. Control Surfaces can be installed in groups (horizontally in the Control Surface Setup menu) to represent aggregate control surfaces. Touching a control on one of the physical devices in a fader group will select that fader group and select the track based on the index in the Mixer view (Single, Tracks, All).

For mapping foot pedals like Apogee GiO and FCB1010 used to control a virtual guitar pedalboard, we want the MIDI messages to go to a specific track all the time, the one that has the guitar plugins. That could be the selected track if you are reusing the same MIDI controller for more than one track at different times depending on what you are recording. Or it could be the same track all the time if the device is meant to work with only one track.  Different zones with different target Channel Strip assignments can be used to enable using the same MIDI foot controller for different purposes.

For a Digital Rehearsal System (DRS), we want the specific MIDI foot controllers to always be assigned to fixed tracks as these are essentially being used in performance mode. The way to do this is to choose Audio as the Channel Strip, and set the index shown in the track Type and Number Label.

Fader Bank or Index is not appropriate for selecting the channel strip in this case because the selected track will change depending on the current Fader Group and Mixer view. That’s fine for physical control surfaces that are intended to be used with multiple tracks, but not useful for foot pedals that are usually assigned to a fixed track.

Target Parameter

It would be nice if you could Click + in the control pane to create a new blank assignment, then manually fill in the values as needed. And that is possible with every other controller assignment class except track parameters. These can only be set using Learn mode, and can’t be edited or re-learned.

Controllers can be reassigned, keeping existing assignments (so the same controller can control multiple parameters) or replacing all existing assignments with an new one.

  • Learn mode creates a new assignment starting with “No message received yet” You can’t create a new assignment without moving a controller. i.e., you can’t enter the message directly and then turn off Learn mode, the assignment will be deleted unless a MIDI message is received.
  • There’s no way to set a parameter without using Learn mode, Key commands can be set manually, but not parameters. The Learn Assignment For “parameter” menu item puts you in Learn mode with the parameter already assigned, but you still have to send a MIDI message for the assignment to be completed. You can’t just edit the selected assignment and terminate Learn mode without re-learning the MIDI message. The assignment will be deleted if no message is received in learn mode.
  • Learn mode only creates new assignments, it can’t be used to edit existing assignments. So there’s no way to edit parameter mappings for track or plugin parameters.

To program track and plugin parameter controller assignments:

  1. Click on the new parameter you want to assign
  2. L or Logic Pro X > Control Surfaces > Learn assignment for “<parameter>”
  3. Press any MIDI controller, it doesn’t matter which one.
  4. Turn off Learn mode
  5. Edit the control name, label, channel assignment, MIDI message and value mapping as needed
  6. Make sure the learned controller assignment is in the right zone and mode. If not, cut to remove it from the incorrect zone/mode, and paste into the correct zone/mode.

To program track and plugin parameter controller assignments without having the MIDI controller connected (remember, you can’t complete Learn mode by moving a MIDI controller):

  1. Click on the parameter you want to assign
  2. L or Logic Pro X > Control Surfaces > Learn assignment for “<parameter>”
  3. Edit the Control Name: MIDI Input Message, etc. any way you want
  4. Before turning off Learn mode, copy C the newly created assignment.
  5. Turn off Learn mode – this will erase the assignment because no MIDI message was received (this seems like a bug, but that’s what it does)
  6. Do V to past the copied assignment to get it back.
  7. Edit the MIDI message directly

To change the parameter of an existing controller assignment:

  1. Click on the new parameter you want to assign
  2. L or Logic Pro X > Control Surfaces > Learn assignment for “<parameter>”
  3. Press any MIDI controller, it doesn’t matter which one.
  4. Turn off Learn mode
  5. Copy all of the parameters of the controller assignment you wanted to update to the new Learned controller assignment, saving the Control Name for last
  6. Delete the old controller assignment

Label fields can have variable fields using @c# where c is one of t=track, r=surround, s=send slot, S=number of sends, e=EQ band, E=number of EQs, p=insert slot, i=instrument slot, and # is the value of the item.

Parameter field: Text description of the addressed parameter. Can only be set by choosing the Logic Pro > Control Surfaces > Learn Assignment for [parameter name] menu item. Note that for plug-in and instrument parameters, Parameter Page offsets apply, allowing you to shift the parameter addressing up and down by page.

Value Change

This sets the MIDI device and value change that will change the selected parameter. MIDI Input pop-up menu: Choose a MIDI input port to change all assignments that use the same input port. If the assignment belongs to a supported control surface, the device’s MIDI input also changes in the Setup window.

The area at the center right shows the following parameters:

MIDI Input pop-up menu: Choose a MIDI input source (MIDI Port or Caps Lock Keyboard). This can be changed by incoming MIDI messages, shown in the Value Change field.

Value Change field: Shows incoming MIDI messages that cause a value change in the destination parameter, and lets you edit these MIDI messages.

The Value Change field displays the message as a sequence of hexadecimal bytes. The plain language meaning appears below the field. The placeholders for the variable part of the message are:

  • Lo7: Low 7 bits of the value (LSB or Least Significant Bits)
  • Hi7: High 7 bits of the value (MSB or Most Significant Bits)

For messages containing only a Lo7 placeholder, the value is treated as 7 bit. For messages containing both a Lo7 and Hi7 placeholder, the value is treated as 14 bit. The order of Lo7 and Hi7 is honored, and there may be constant bytes in between. This allows you to define Control Change LSB and MSB portions. For example, B0 08 Hi7 B0 28 Lo7 indicates a 14-bit message. For messages containing neither Lo7 nor Hi7 placeholders, Logic Pro assumes an incoming value of 1.

Touch/Release field: Enter an integer value to force incoming MIDI messages to change the touch/release state of the selected parameter. This only applies to control surfaces that provide touch-sensitive controls (where touching or releasing a fader, for example, enables or disables reception of data from the control surface).


Constrains the min/max values, or scales the incoming MIDI message value (usually between 0 and 127) and sets how the message will effect the target parameter.

Bypass vs On: Toggle parameters may be Off(0) – On(1), or Enable(0) – Bypass(1), that is bypass on turns the effect off. Typically bypass parameters should be inverted. For S-Gear Mod and Delay Thing for example, that would typically be Mode: Scaled, Multiply: -1. However may not be quite correct to get the parameter to switch properly. Use -1.01 to get values of 0 and 1 in the Controls view. This could be a bug.

Min and Max fields: Enter integer values to set the range of incoming MIDI values represented by Lo7 and Hi7.

Format pop-up menu: Choose the format used to encode negative values.

Multiply field: Enter a value to scale incoming MIDI values. For example, S-Gear Amp A/B takes a values like 0.01 for Amp A and 0.02 for Amp B. You’ll need to experiment with the Multiply field to see what values will result in the right parameter values for S-Gear. Use the plugin Controls view to see how the parameter value changes as you change the Multiply field and press the switch.

  • S-Gear automatically scales MIDI CC values 0 – 127 to 0.00 – 1.00 for many plugin parameters.
  • Amp A/B is currently has two values (but S-Gear was designed to support more) 0.01 for Amp A and 0.02 for Amp B. However 0.00 also happens to choose Amp A
  • So to get Amp B, Multiplier should be 0.02*Lo7/127+0.01. The Amp parameter is 1, not 0 based, so the multiplier has to be a little larger in order to  account for the +0.01 that needs to be added. Logic scaling only multiplies, it doesn’t have a more complex formula to transform the MIDI values. A Multiplier value of 0.03 gives the value 0.02 for the parameter, and selects Amp B.
  • This is what I mean about experimenting to determine how to scale the MIDI CC value to what is needed for the plugin parameter.
  • S-Gear MOD and Delay Thing bypass are also three states:
    • 0.00 is bypass off (effect on)
    • 0.50 turns off the effect input, but leaves effect tails if any
    • 1.00 the effect is bypassed and the effects are off

Mode pop-up menu: Choose the mode used by incoming values to modify the current parameter value.

Direct: The incoming value is used as the parameter value.

Toggle: If the parameter’s current value is 0, it is set to the incoming value (Multiply is ignored). All other values set the parameter value to 0 (i.e., if the parameter value is non-zero, any MIDI CC value, including 0 will set it to zero). This option is useful for buttons that toggle a value, such as Mute or Solo.

    • For GiO, the switches are press 127 and release 0, two MIDI messages
    • The max and min are set to 127 so only the press has any impact on the parameter
    • Parameter is 0 and MIDI value is 127, parameter is set to 127 or on
    • Parameter is 128 and MIDI value is 127, parameter is toggled to 0 or off

Scaled: The incoming value is scaled from its value range to the destination parameter’s value range. This is useful for faders and rotary encoders, or for switches that aren’t simple off/on values.

Relative: The incoming value is added to the parameter’s current value. This is commonly used for encoders but is also useful for buttons that increment or decrement by a certain amount—specified by the Multiply parameter. To decrement a parameter by 1 with a button press: Set Multiply to –1.00, and Mode to Relative.

Rotate: The incoming value is added to the parameter’s current value, cycling between maximum and minimum values. This is useful for button presses that cycle between modes, such as automation mode.

X-OR: The value defines a bit mask (a filter), which is applied to the parameter’s current value with the “exclusive or” Boolean operation. This is useful for enabling or disabling single channel strip types in All view.

Note: For On/Off parameters, Mode is set to Toggle by default. It is set to Scaled for absolute controls (faders and knobs, for example) or to Relative for encoders.

Feedback pop-up menu and checkboxes: Choose the display format of the parameter value (on the control surface display, if applicable). This can be useful to control the lights on a device like the Apogee GiO so that the lights properly reflect the state of the plugin parameter.

Creating a Helix Electric Guitar Patch (newly updated)


Line 6 recently updated Helix to version 2.30, created a new Helix Edit application, and updated Helix Native. So its time for another update of my goto patch. Helix has change a lot in the last year adding snapshots and a number of new amp and effect models. As a result there have been some significant changes to my goto patch I’ve been using for live gigs the last couple of years. I also added a JTV-69S to may rig, and updated its pickups with Amalfitano Daytona pickups. After some fine tuning of the setup and a new nut, this has become my goto gigging guitar.

The new Line 6 Helix amp modeler is an awesome device, capable of creating a wide range of really great tones for acoustic guitar, electric guitar, bass and vocals. However, it can be quite a challenge figuring out how to put all this capability to work for your particular needs. The factory presets are a good place to start, auditioning each one to get some ideas. But these factory presets are generally designed to demonstrate the device’s capabilities, and can be a bit over hyped and impractical for gig use.

In this post I’ll describe some different approaches to setting up electric guitar patches in Helix. Then I’ll go into some detail on my updated goto electric guitar parch, covering the reasoning behind what is placed where in the signal chain, and how each effect block is configured. Of course the tone that works for my playing style, guitar and FRFR amp may not be even close to what you are looking for. But the thought process might be useful in helping you come up with your own tone.

The previous update incorporated some of the things I learned Analyzing Joost Assink’s SRV Little Wing Patch into my Electric Guitar patch:

  • Use a Studio Tube Pre instead of a Low and High Cut EQ for controlling drive and as a mid-focus EQ. This not only provides a warmer tone, but adds the flexibility of getting some distortion from the Studio Tube Pre that isn’t possible with the Low and High Cut EQ.
  • Moves the Amp and Impulse Response blocks to path 1B so that most of the mono blocks are in Path 1, and to make room for more stereo effects in Path 2.
  • Added another Studio Tube Pre after the Amp and (speaker) Impulse Response blocks to warm the tone a bit.
  • Added the LA Studio Comp at the end of the chain, just before the Looper to further warm the tone and add a tiny bit of compression on the final result.

This updated version makes the following changes:

  • Use of Line 6 Litigator instead of the US Deluxe Vib amp model
  • Changed the IR to one I made myself
  • Use of tube preamps before and after the amp/IR blocks for drive voicing control
  • Red Comp in stead of Deluxe Comp compressor because it provides compression more suited to guitar
  • Teemah! and Minotaur for overdrives
  • Parallel path for Leslie
  • Using snapshots for open tunings, acoustic and Leslie in the same path

Hope you find the update useful.

Approaches to Patch Design

Most guitar players use a number of different guitars, pickup combinations, tones and effects in different songs or even in different parts of the same song. This adds interest and color to your playing that helps maintain the audience’s attention. You can do a lot with just the right pickup selection, and using the guitar volume and tone controls. But Helix gives a wealth of other choices for distortion levels, effects, amp models, and synth effects. How do we setup patches to organize all these capabilities so they are ready and easy to use in a live setting?

There are two broad approaches to designing patches: Stomp mode: get the most out of each patch (includes snapshots), or Preset Mode: make each patch for a specific purpose. These two approaches correspond to the Helix Stomp Footswitch Mode and Preset Footswitch Mode respectively. In stomp footswitch mode the footswitches are used to control 8 or 10 (Stomp Mode Switches global setting) effect settings while in preset footswitch mode the footswitches are used to select between 8 different patches. The Preset Mode Switches global setting can be used to provide a combination of both with one row of stomp switches and another row of four presets.

Stomp Mode

Stomp mode minimizes the number of patches and reduces patch switching within and between songs. The idea is to design the patch to reflect your playing style and the range of tones you need. Then you use just one patch, getting different tones by turning effects blocks on and off within the patch. This is very similar to how you would setup a typical guitar amp and pedal board. You might have two speaker cabinets or two amps to get a range of tones, but that’s it. You would typically have one pedalboard that contains all your effects in a fixed order. Then you change tones primarily through bypassing or turning on effects in the pedalboard.

Helix can be used this way too. But there are some challenges to address:

  1. Although each main path has its own independent DSP, and can contain up to 16 blocks (not counting inputs, outputs, splits or merge blocks), its still pretty easy to run out of DSP in one of the main paths.
  2. There are at most only 10 stomp footswitches available within a patch. If you have more than 10 effect blocks in the patch, you’ll need some external MIDI controller to control bypass on some of the blocks
  3. Mono blocks sum their stereo inputs. So any stereo effect block before a mono block is lost and just wastes DSP

Helix now supports snapshots within a patch to support changing configurations within a patch. Snapshots are a special case of Stomp Mode where a single footswitch can change up to 64 parameters in the block. Snapshots can’t change or reorder blocks, they can only be used to change parameters in a preset. This allows you to configure the preset for very different purposes within the patch, and switch to them immediately without any pause, and without loosing reverb and delay tails. So snapshots extend Stomp Mode with the ability to change and store a large number of parameter changes for later use.

Use stomp mode if you have a signature tone that just uses different distortion levels and a fixed set of effects. Don’t use stomp mode if you’re playing a wide range of styles in a cover band, it will be too hard to make one patch do everything.

Preset Mode

Preset mode minimizes the number of blocks in each patch, and uses different patches to create different tones. Each patch is designed for a specific purpose either for a section within a song, or for different songs. Patches are often ordered in setlist and banks to allow fast switching from one tone to another. Once a patch has been selected, the Mode switch (FS12) can be used to temporarily switch to stomp footswitch mode to control the blocks within the patch. In this case you’re much less likely to run out of footswitches to control the blocks since the patch is designed for a specific purpose. You can also use the Preset Mode Switches global setting to have a row of four stomp switches and a row of four patches. This may be very convenient for Preset Mode since you can quickly switch between four presets in a bank for different sections of a song, control up to four effects blocks within the patch, and use the bank switches to select the next song in the setlist.

Like stomp mode, preset mode also has some challenges to address:

  1. It takes some time to switch patches, this has to be done carefully within a song
  2. Helix doesn’t support effects trails between patches, so synth, reverb, delay, and other effects might be cutoff abruptly depending on when you switch the patch

Use preset mode if you are playing in a cover band and have to reproduce very different guitar tones, possibly with a Variax, need to switch patches within or between songs, don’t need too many effects in each patch, and can deal with the patch switching delay.

The rest of this post explores a patch built using the stomp mode. This works well for me because I play three different instruments in my acoustic band: mandolin, acoustic guitar and electric guitar, and use a Variax JTV-69S in my rock band. I use a patch for each instrument and only change patches when changing instruments. Helix is great for this because of the I/O flexibility – I can leave all three instruments plugged in at the same time with only one instrument active in each patch.

Signal Path

Although there are no rules for establishing the order of amp and effect blocks in your signal chain, there are some guidelines that work well in practice. Here’s a few best practices that may be useful in guiding your tone setup:

  • The larger the room, and the louder you play, the less effects, especially reverb or delay you need.
  • Use delay instead of reverb for ambiance, especially in a larger room, to avoid washing out the tone and to fit better in the overall mix.
  • Simple ambiance can be achieved with a slap-back delay of 125 to 175 ms with no repeats. Blend to taste. Shorter slapbacks can often be left on all the time.
  • Smooth out the guitar sound, and blend into the mix better using 500 ms delay with a few repeats. Especially useful in a three-piece situation.
  • Most guitar players play in mono – but that’s changing with digital amplifiers. Before the amp effects are almost always in mono while after the amp effects (often in the recorded track) are usually stereo.
  • Overdrive, reverb and delay are timeless while chorus has an 80’s feel. Use sparingly and with caution.
  • Minimize cable length and use low-capacitance cables to get the most out of your guitar.
  • Use the minimum number of effects in the signal path that you need at any point in time to avoid killing tone.
  • Use the minimum amount of distortion needed for the song. Too much just washes out the guitar and has no articulation.
  • It can be useful to stack distortion blocks to increase sustain. However, distorting an already distorted sound can loose articulation and make the guitar tone less distinct. Try to get the distortion you need from a single source if you can: different distortion blocks, amp gain or poweramp distortion.
  • Use EQ before distortion to cut bass to reduce mud, and another EQ after distortion to cut treble to reduce ice-pick. Increase bass and treble cut with increased distortion.

A typical effect chain starts with tone shaping effects and ends with ambiance effects.

Static Tone Shaping: Tone shaping comes first, including guitar tone, volume and pickup selection. This is followed by compression to control pick attack and sustain.

Dynamic Tone Shaping: Next comes variable tone shaping devices like Wah Wah, phase shifter, or Uni-vibe, possibly Flanger too. These are modulation devices, but modulate phase or tone more than frequency, and therefore can go in front of distortion. Of course in the old days, all effects were at the front of the amp, so we’re use to hearing them this way too.

Distortion and Overdrive: Overdrive, gain staged for different boost/distortion and voicing levels. One should be for controlling metal lead distortion, and another for creating the overall amp sound. The second should clean up well when turning down the guitar volume. This section can also be handled completely by the amp if it has sufficient gain staging options. Cartographer is a good amp model for this because it has two Drive controls and two Bright switches to control the gain and distortion voicing. Use it with snapshots to setup different gain staging configurations that could eliminate the need for distortion pedals. Using overdrive pedals however can give more control over the amount of distortion and overdrive, as well as the tone shaping or voicing. Use a tube preamp and/or EQ before distortion to control the distortion tone. Use a tube preamp and/or EQ after distortion into a clean amp to do a simple volume boost for clean or distorted tones.

Amplifier: The guitar amplifier would typically come next, and usually includes the speaker cabinet and mic. This allows all the modulation and ambient effects after the amp to be “in the air” and not overly impacted by the amp itself.

Modulation Effects: Mod effects like flanger and chorus come next. These effects modulate frequency and usually work best after distortion. More classic tones came from pedals before the amp which provided most of the overdrive. This can result in a less articulate tone, and reduces the impact of the effect. In some cases, these effects were produced in the studio after the recording, especially flanger for a more pronounced effect that is operating on the distorted signal rather than being distorted by the overdrive.

Flanger might go before or after distortion depending on how pronounced the effect should be. Chorus would generally be after distortion in order to simulate doubling or Leslie effects.

Ambient effects: Delay and reverb effects go last. Usually Delay comes before reverb. Use a slap-back delay for clean ambiance, and a longer delay with repeats to smooth out the overall tone.

Assigning Footswitches

Its a good idea if you are using multiple patches to organize the stomp footswitches as consistently as possible between patches. This makes it easier for you to remember where each effect footswitch is located. Helix has the scribble strips, which certainly help identify what a footswitch does. But you don’t want to have to look down at the pedalboard to find an effect switch in a live situation. Here’s a few guidelines:

  1. Put the footswitches in signal chain order from right to left. This corresponds to how many people organize their analog pedalboards, with the Wah at the far right. Reverse this if you are left handed or prefer to use you left foot to control the Wah
  2. Use consistent footswitch assignments between patches to make it easy to find the right footswitch
  3. Name the footswitches with generic effect names, not the specific default Helix effect model names. Again this is to provide consistency between patches and make it easier to recognize the effect from the scribble script
  4. Put effects you change most often in the lower row, they’re easier to get to in a live situation
  5. If you use the Looper, put it on FS7 so its right next to the Record/Overdub footswitch after you switch to Looper mode.

Here’s my typical footswitch layout:


I use this same layout for mandolin and acoustic guitar, although the Overdrive and Distortion effects are very different.

Electric Guitar Patch

With the preliminaries finally out of the way, we can now get down to the actual patch details. This is my goto electric guitar patch. It designed primarily for Americana, Blues and Rock styles, and using a Stratocaster (or single coil pickups). Its based on a Fender style amplifier, but takes liberties with the speaker model to get the desired warmth.

The intent of this patch isn’t to copy any particular artist or song tone, although it is certainly inspired and informed by many great players, in particular Matt Schofield. Rather it is my own preferred tone, with enough variation in distortion and effects to cover a wide range of songs. I play in a typical club cover band, around two to three times a month. I don’t try to recreate the exact tones, or necessarily play the exact solos of the songs we’re covering. Rather I take some liberties and interpret the songs in my own tone and style, making sure to focus on the hooks. This makes it more fun for me while giving the audience the spirit of the song with enough variation to make it interesting. I deeply respect people like Richie Castellano who can accurately recreate the tones, effects and exact leads of so many songs. I can’t do that, and I don’t necessarily want to.


Path 1

Because of dynamic DSP limitations, and the number of effects in this patch, I have put the “before the amp effects”  and amp on Path 1 and the “after the amp effects” on Path 2. The output of Path 1A is sent to Path 2A which has no other input. The output of Path 2A is the Multi output, so the 1/4″, XLR, Digital, and USB 1/2 outputs are all active simultaneously. Note the pictures are from Helix Native because its more convenient to capture them.

In this configuration, Path 1 has most of the mono blocks including before the amp effects, and a couple of mono effects that go after distortion, but before the cabinet. Path 2A is mono for the Cali Q Graphic EQ, the IR block, and Studio Tube Pre, then stereo after that. This balances the DSP load between path 1 and 2, and provides extra DSP room on Path 2 for other expensive stereo effects like the 145 Rotary or 3 OSC Synth. The only issue is that there aren’t enough footswitches to control all the effect blocks in this patch. As far as I can tell, Patch Edit Mode, does not currently support block bypass. I have raised this issue with Line 6. If Bypass was available as a mappable parameter, then you could use Patch Edit Mode to control seldom used blocks that aren’t assigned to a footswitch. Another alternative is to use a MIDI controller such as the FCB1010 with the Eureka Prom to provide extended footswitch controls.

Guitar In

For this patch I have the Noise Gate on the input turned on with a minimum threshold in order to eliminate noise while still retaining the subtle dynamics of your guitar. I mostly play the JTV-69S, and use the Daytona magnetic picks a lot. So they can generate some noise. Generally I don’t worry that much about a bit of noise, but the noise gate tames it pretty well between songs. If its real bad, I can always use one of the models to get super quiet tones.

Wah: Chrome Custom

The first effect in the signal chain is the Chrome Custom Wah. Of all the Wah Wah pedals in Helix, this one sounds the most musical to me. Its before the compressor to deal with any odd peaks when using the Wah with a clean tone.

  • Position: EXP Pedal 2
  • Fc Low: 300 Hz
  • Fc High: 2.0 kHz
  • Mix 100%
  • Level: 0.0dB
  • Footswitch: EXP Toe

Compressor: Red Squeeze

I use to use the Deluxe Comp because it gives a lot of control that can be used to reproduce other compressors as needed. But I found the Red Squeeze works a bit better for guitar and is simpler to use. The Red Squeeze models the MXR Dyna Comp compressor which has 36dB of compression, very fast attack (5 msec), very slow release (1 sec), and high compression ratio (10:1 or more). This is well suited for guitar and provides very nice sustain as well as the funky attack. Kinky Comp is another good choice (and uses less DSP).

The compressor is mostly used on very clean tones just to even out the guitar dynamics a bit, and make clean tones stand out a bit more for solos. It’s placed before any EQ or distortion effects in the signal chain so it sees the dynamics of the guitar itself, not the output of most effect blocks. The compression ratio is set very high, which seems to work well on electric guitar. The Level is set for makeup gain and a tiny boost for clean leads. I keep the Mix at 50% so I can get the sustain from the compressor while retaining some of the guitar’s dynamics and making the pick attack sound more natural.

  • Sensitivity: 5.5
  • Mix: 50%
  • Level: +0.7dB

Drive: Studio Tube Pre

I use two Studio Tub Pre’s, one before distortion and another after distortion to control distortion voicing. The Studio Tube Pre before distortion to provides some low cut to control bass mud, while the one after distortion provides some high cut after the IR to control treble ice-pick. This block is tied to the Drive footswitch (along with the Amp Drive control) and is normally off.

The Studio Tube Pre sounds good and is a flexible means of adding some early distortion through its Drive control, and a mid-focus EQ using a combination of the Low Cut and High Cut parameters. By adjusting these two parameters, you can control the width of the mid-focus EQ and where it is positioned in the frequency spectrum.

In this case the high cut is kept off because the block doesn’t add any distortion and I want to preserve the guitar high frequency response when the amp is just starting to break up. See the Amp block for more details.

  • Gain: 4.9
  • Polarity: Normal
  • Low Cut: 90 Hz
  • High Cut: Off
  • Level: 6.8dB
  • Sensitivity: Line

Overdrive: Teemah!

Before going into the details of this block, we have to consider gain staging. Since this patch is based on patch mode, and we want to get a wide range of tones out of the same patch, we use gain staging to control different levels of distortion. I like to have four gain levels in a patch like this one: Clean, Drive, Overdrive, and Distortion. Each of these gain levels increases distortion and uses various tone controls to control the distortion voicing.

  1. Clean: the amp master volume is set pretty high (or all the way up) so that any initial distortion comes from the power amp section, not the preamp. For the Clean tone, the Amp Drive control is set just below any noticeable distortion
  2. Drive: this adds enough Amp Drive to just get the amp clipping. Its for typical Blues tones where the distortion is coming from the power amp and the sound is warm, full, expressive, and reacts dynamically to how hard you pick. Clean and Drive are controlled by the Drive footswitch where the Amp Drive switches from 3.6 to 4.2. Recall that when the Drive switch is on, the Studio Tube Pre Low Cut is increased to 90 Hz to reduce the bass going into the distorted amps. Fender-style amps really seem to need this base cut. Without it, the distortion gets muddy and a little nasty sounding.
  3. Overdrive: This adds the next level of distortion, usually for heavy blues leads. A distortion model is used for this additional distortion in order to control the voicing. Some treble cut will be needed at this distortion level to keep the tone aggressive, but still reasonably warm. This gain stage should clean up well when rolling back the guitar volume control.
  4. Distortion: This is the most distorted tone in the patch and is used for heavier, closer to Metal leads. Again it uses a distortion model to control the distortion voicing.
  5. Insane: You can also combine any of the three Drive, Overdrive and Distortion tones to get increase distortion with different voicings. This is a lot of flexibility from three footswitches and one amp.

Some amp models (e.g., Soldano SLO-100 or the Solo Lead model) have clean, crunch and overdrive channels that support gain staging, distortion levels and voicings. However, these channels can’t be changed within a patch (no scenes in Helix). Other amp models like Cartographer have multiple Drive controls that along with the Master volume provide a wide range of gain staging options. Using the distortion models gives more control of both the distortion and the voicing, so that works best when using patch mode.

Teemah! is used to create the Overdrive tone, and is controlled by the Overdrive footswitch. Gain is set to provide additional distortion for blues leads while Bass Cut and TrebleCut are used to provide additional bass and treble cuts for higher gain distortion voicing.

  • Gain: 4.0
  • Bass Cut: 2.1
  • TrebleCut: 5.7
  • Clipping: Center
  • Level: 5.2

Distortion: Minotaur

I use to use the Compulsive Drive distortion model is used to create the Distortion tone, controlled by the Distortion footswitch. Compulsive Drive is based on the Fulltone OCD. This is a very nice, and very flexible boutique distortion pedal that is a real Helix gem. This patch uses Compulsive Drive to get a nice creamy distortion that just sings. Combine it with the Drive footswitch to increase amp drive and low cut to get a bit more distortion with a slightly different voicing. However, I’ve recently switched to using Minotaur. The motivation for the switch is I’m tending to use less distortion, and Minotaur is a bit more mid-focused and cuts through better without having to have so much drive. I’m also using a different Hi-Gain patch that use Cartographer for more saturated distortion tones, so I don’t need them in this patch.

  • Gain: 7.9
  • Tone: 4.8
  • Level: 6.5

Scream 808 (Ibanez TS808 Tube Screamer), and Vermin Dist (Pro Co RAT) are also very good choices for this block. These have different distortion characteristics, and voicings.

Modulation: Script Mod Phase

Next in the signal path are modulation effects that change tone or phase of the signal. These can be placed before or after distortion. Their effect is a bit more pronounced after distortion, so I’ve placed them here, between the distortion pedals and the distortion created by the amp. That’s a compromise that attempts to get the benefits of both approaches. I keep the rate slow and the mix down to keep the phaser effect subtle. This make the effect usable in a wider range of situations.

  • Rate: 1.9
  • Mix: 39%
  • Level +1.0dB

Modulation: Ubiquitous Vibe

I use to own a UniVibe and loved the effect. Previous models in earlier Line 6 products weren’t that great, but the Helix Ubiquitous Vibe model seems dead on. This is just one of those effects you might need sometimes, especially for Hendrix tones. Its also useful when you want some tone modulation, but chorus is too much. The rate is controlled by EXP Pedal 1 (when the following Volume block is off) with the min and max values set to mimic the typical speeds of a Leslie speaker. Note that the min and max are reversed (min is 8.0, max is 1.2) so that then the effect is turned on, and the EXP 1 pedal is all the way down, the rate is slow. Lamp bias controls how the effect ramps up and down.

  • Rate: 1.2 – 8.0 (Controlled by EXP Pedal 1)
  • Intensity: 6
  • Mode: Chorus
  • Lamp Bias: 2.7
  • Mix: 50%
  • Level: 0.0dB

Distortion: Tycoctavia Fuzz

This is the odd effect that you might need for Hendrix tones. I don’t currently have this assigned to a footswitch, so it has to be controlled by selecting the block and pressing the Bypass switch. See for a great demonstration of a UniVibe and Octavia. You might also be interested in his Guitar Effects Survival Guide course. I found it very useful.

  • Fuzz: 7.5
  • Level: 6.7

Volume: Volume Pedal

I added a Volume Pedal right before the amp block in order to provide some foot control of the amp drive. The volume range is limited to 38% to 100%. This way I can turn the volume down a little bit with the EXP 1 pedal, but don’t risk turning it all the way down if it gets stepped on by accident.

Amp: Line 6 Litigator

I’ve been using Fender amps for many years and at one time owned a Deluxe Reverb and a Super Reverb. I should never have sold them, but there you go. I never thought anything could displace the US Deluxe Vib model, but Line 6 Litigator has done it. This amp model has the additional gain and voicing for distortion that are missing from the US Deluxe Vib. It breaks up well at that critical junction where the power amp is just starting to clip.

There are a lot of choices on how to configure an amp and speaker model:

  1. Amp+Cab: automatically loads the matching cabinet for an amp, but allows the cabinet to be change. The lowest DSP load for an amp and a cabinet.
  2. Separate Amp and Cab models: allows the placement of effects between the power amp and cabinet, supports two cabinets in stereo. Uses more DSP.
  3. Amp and IR: lets you choose other cabinet models. Those from Redwirez, OwnHammer and Rosen Digital are very good and there are a lot of free cabinet IRs on the Web.
  4. Preamp: useful for input directly into a power amp connected to a guitar speaker cabinet, or in some cases as a very flexible distortion block to use instead of a pedal.

In this patch, I use the Amp model and no Cab model because I’m going to use an IR block for the speaker model.

The Amp Master volume is set pretty high so that any initial distortion is created by the power amp, not the preamp stages. The Amp Drive control is controlled by the Drive footswitch to, along with the Studio Tube Pre early in the signal chain, and the one following the IR, support the Clean and Drive gain stages as described above. Recall that the Drive footswitch also controls the Studio Tube Pres to add some a bass cut when the Drive is increased. The tone controls are set for the desired clean tone using the Strat neck pickup. That often results in the bridge pickup being a bit too bright, but turning the guitar tone control down just a little fixes that and provides the overall clean tone.

Distortion tone voicings are controlled by the overdrive and distortion block controls and are set to sound good into this clean tone setting. These tones are pretty warm to suit my band’s particular needs. You might want to brighten them up a little. I raise the bias and the Bias X to provide a good clean tone with a little bit of breakup when digging into the guitar a bit. Reduce Sag to get a tighter tone.

  • Drive: 3.6 (Drive footswitch off), 4.2 (Drive on)
  • Bass: 2.9
  • Mid: 7.1
  • Treble: 5.7
  • Presence: 3.6
  • Ch Vol: 7.5
  • Master: 7.2
  • Sag: 6.1
  • Hum: 5.0
  • Ripple: 3.7
  • Bias: 6.5
  • Bias X: 7.5

Tremolo: Tremolo

Similar to Fender amps, I place the tremolo between the amp and the speaker. I find these tremolo settings to be pleasing and natural.

  • Speed: 2.5 Hz
  • intensity: 6.3
  • WhaShape: Sine
  • DutyCycle: 50%
  • Level 0.0dB

You can use a stereo auto-pan after the amp for an interesting effect. But I wanted this to be the traditional mono tremolo.

Slapback Delay: Simple Delay

A quick slapback delay can add depth to guitar tones without standing out or contributing to muddy ambiance. I leave this on all the time, but it is configured to be turned of if the long delay after the amp in path 2 is turned on. Mix is set so the slapback is there, but not that noticable when you play.

  • Time: 160 msec
  • Repeats 0%
  • Mix: 21%
  • Level: 0.0dB
  • Tails: off (since there are no repeats)

Path 2

Path 2A has another Studio Tube Pre followed by all the after the amp stereo effects.

EQ: Cali Q Graphic

I added this traditional guitar-centered EQ between the amp and IR. Its set flat, but is available for tone shaping if needed. I like this EQ for guitar because the five frequency bands are right in the range for an electric guitar: from 80 Hz, the low E string to 6 kHz which is about the high-frequency limit for an electric guitar.

Leslie: 145 Rotary

The Leslie block is placed in path 2B so that it is in parallel with the cabinet model. I use a snapshot to switch the 145 Rotary block on. Speed is controlled by EXP 1, so the Volume block is bypassed in this snapshot. The 2A merge block controls the mix of the amp and IR tone with the Leslie tone. This balance is usually -60dB so there is no Leslie tone. The Leslie snapshot changes the B level to -16.4dB to blend in the Leslie. Again, headroom is high because the effect (designed for guitar input) is being fed by the amp output.

Impulse Response

In an electric guitar setup, the things that touch the air often have a major impact on the overall tone. That starts with the guitar (including pick, strings, and pickups) and ends with the speaker cabinet. Helix provides a lot of cabinet options, including dual cabinet modes. But there are also a wealth of guitar speaker cabinet impulse responses (IRs) on the market and free on the Web that also sound wonderful. Support for IR blocks is one of the distinguishing features of Helix over the POD HD500X. Selecting the right cabinet (open or closed back), speaker, mic and mic position can really tailor the sound.

After trying a lot of Helix Cab models, and a number of my own Redwirez and Rosen Digital IRs, I discovered JOOSTALNICO-G12M-R121-U67 IR.wav from the Helix forum post My Two Rock/Fender clean tone, PRESET+IR by JazzInc. This is a very warm, low-end heavy model that uses a blend of two Redwirez models:

  • Basketweave G12M25s, with a Neumann U67 mic 0″ from the CapEdge
  • Celestion-blue 12, with a Royer R121 ribbon mic 0″ from the Cap

The warmth comes from the proximity effect of the close mic positions, the use of a ribbon mic, and the U67 which has extended low end. This combination of speakers and mics is still crisp and smooth. Distortion tones are thick because of the bass response of the speakers, but not muddy because of the bass cut before distortion. Those two speakers also provide a warm distorted tone since they aren’t overly bright.

Joost created some other IR mixes and I found G12M25s-SM57-Cap-0in-7200c.wav to be particularly good. However, our bass player had an ealry 60’s Tremolux that had a pair of Oxford 10K5’s that just sounded wonderful. So I captured my own IR and have recently started using this.

If find most Line 6 cab models, and most commercial IRs need a fair amount of High Cut when going into my FRFR (a pair of JBL EON10’s). But Joost’s IRs, and my IR for the Oxford 10K5s sound pretty good with almost no low or high cut. I add a little low cut just to remove low frequency stuff that happens when you hit the strings. But these settings have little impact on the IR tone.

  • IR Select: 40 (Oxford 10K5 AT4047 Cap-Edge 1.5 in.wav”)
  • Low Cut: 61 Hz
  • High Cut: 12.7 Hz
  • Mix: 100%
  • Level: -18.0dB

Note that IR blocks are not stored with the patch, only the index to the IR block is stored. If you have the IR block loaded at a different index, then you’ll need to change the IR Select to the index where you loaded theOxford 10K5 AT4047 Cap-Edge 1.5 in.wav IR.

Preamp: Studio Tube Pre

This Studio Tube Pre is designed to come after the amp and cabinet to warm the tone and provide after the amp low and high cut filters as needed. The effect is subtle, but does seem to improve the overall tone of the patch.

The Studio Tube Pre is set pretty flat and clean so that it does not produce any additional distortion. The low cut is set to minimize any sub harmonics created by the amp, while the high cut is used to control fizz and ice-pick from the gain stages and amp distortion.

  • Drive: 5.0
  • Polarity: Normal
  • Low Cut: off
  • High Cut: 8.4 kHz
  • Level: 6.7dB
  • Sensitivity: Line

Modulation: Chorus

All the effects from here on to the output are stereo. The effect order is modulation, delay and then reverb. Line 6 has created a very nice, general purpose chorus model that is very flexible. At one extreme, you can set Speed and Depth to 0 and just get a subtle stereo widening through headphones. At the other extreme you can get a rich 80’s chorus that will carry you away. I use chorus sparingly and with moderate settings. Use Predelay to avoid having the chorus kill pick attack and therefore articulation. Spread is set at 10 to give full stereo chorus.

  • Speed: 1.8
  • Depth: 6.0
  • Predelay: 3.2
  • WaveShape: Sine
  • Tone: 5.0
  • Spread: 10.0
  • Mix: 50%
  • Level: 0.0dB

Delay: Transistor Tape

This delay adds an obvious long delay or echo effect intended to be more noticeable. The delay is longer, 1/2 sec, and there are repeats. This delay can be used to fill in softer, sparse phrases, or provide even more ambiance in situations where there are fewer instruments and you need some fill. This is a delay setting that would often be used to thicken vocals. The Transistor Tape delay provides some modulation of the delays, giving a wider, richer overall tone without creating a wooshy chorus on the main tone.

The Scale and Spread controls can be confusing, especially since they are not documented that well in the Helix manual. The Transistor Tape delay , like the Mod/Chorus Echo and PingPong delay, has two separate channels of delay, with the output of each channel flowing into the other. The delay Time sets the time for the left channel delay. The Scale parameter sets the delay time for the right channel delay line, as a percentage of the left channel’s delay. For example, if the delay Time is set to 500 ms, and Scale is set to 0, the the delay time is 0, and the right side will repeat at the same time the note is played. If the Scale is set at 50%, then the right side will repeat every 250 ms, or twice as fast as the left. If Scale is set to 100%, then the left and right sides repeat at the same time.

The advantage to using scale, rather than just setting two delay times, is that the rhythmic interaction between the left and right sides is always the same. The proportion between the two different times stays the same even though the delay time has been changed.

Spread on a Delay block adjusts how wide the repeats bounce between the left and right sides. At 0, the left and right repeats are both in the center and the delay is effectively mono. As the Spread increases, the left and right repeats are pushed further apart, 10 is full left to right panning on the repeats.

I have set Scale high so there is just a little delay offset between the left and right channels. WowFluttr is use to add a little modulation on the delayed signal. Spread is set to 5.0 so that the modulation on the delays is partially in stereo. Trails are on since there are repeats that fade out when the effect is bypassed.

  • Time: 482 ms
  • Feedback: 17%
  • Wow Fluttr: 2.6
  • Scale: 97%
  • Spread: 5.2
  • Mix: 25%
  • Level: 0.0dB
  • Headroom: +8.0dB (because an effect typically expecting guitar level is being feed the ouput from the amp)
  • Trails: On

Reverb: Plate

Helix has lots of really ok reverbs, we’re hearing HD reverbs are coming. I personally like a very small amount of warm reverb. So I choose the Plate model. I use a short decay to avoid having the reverb make the tone become indistinct. Predelay avoids having the reverb cover up pick attack. Low cut and high cut are adjusted to make sure the reverb doesn’t compete too much with the main dry signal. Mix sets the overall amount of reverb. Trails don’t matter because the reverb is left on all the time, and is not assigned to any footswitch.

  • Decay: 4.5
  • Predelay: 41 ms
  • Low Cut: 90 Hz
  • High Cut: 7.8 kHz
  • Mix: 28%
  • Level: 0.0dB
  • Trails: Off

Dynamics: LA Studio Comp

A LA Studio Comp compressor is placed at the end of the signal chain to take advantage of its unique contribution to the tone, even when its not compressing that much. This helps glue the effects together and provides a good controlled signal into the FRFR amp. Again, the effect is subtle, but does contribute to the overall tone. The use of the LA Studio Comp, and the Studio Tube Pre after the amp are intended to duplicate what would be typically be done in a studio when setting up for an electric guitar track.

  • PeakReduc: 5.0
  • Gain: 5.2
  • Type: Compress
  • Emphasis: 10.0
  • Mix: 100%
  • Level: 0dB;

EQ: Parametric

The last thing in the signal chain before the output is a Parametric EQ. This provides the final tone shaping of the total signal path including the effects. It plays the same role an EQ in a recording track would play for final tone shaping to fit into the mix. I also use this EQ in snapshots to provide for an acoustic tone from the JTV-69S with the amp and IR blocks turned off. Like the Cali Q Graphic EQ, the frequencies are chosen to be useful for guitar tones.

  • Low Freq: 110 Hz
  • Low Q: 0.7
  • Low Gain: 0.0dB (may be different in the Acoustic snapshot)
  • Mid Freq: 993 Hz
  • Mid Q: 0.7
  • Mid Gain: 0.0dB (-2.4dB in the Acoustic snapshot)
  • High Freq: 8.0 kHz
  • High Q: 0.7
  • High Gain: 0.0dB (+2.5dB in the Acoustic snapshot)
  • Low Cut: off
  • High Cut off
  • Level 0.0dB (+3.0dB in the Acoustic snapshot to level the volume)


The Looper can be placed at the end of the signal chain so that any effects that were on when the loop was recorded are include in the loop. Playback and Overdub are adjusted so that as overdubs are added, they are reduced in level, leaving headroom to play on top of the loop. If you don’t turn Playback and Overdub down, the loop will become saturated after a small number of overlaps, and won’t leave any room left to hear what you’re playing on top of the loop. See Using a Looper for Solo Gigs for some ideas on how best to use a Looper.

A note on the Helix Looper: the 1/2 FULL speed switch appears to be global. It is not saved with the patch, and remains at its last setting when switching patches. This can be quite surprising since a FULL loop in stereo is only 30 sec long. This may be shorter than most of your loops if they are a full verse or chorus of a song. So glance down when you first use the looper in a patch and make sure the looper is set to be able to accommodate the length of the loop. In 1/2 mode, the looper is twice as long, 60 sec for a stereo loop. This is often long enough for a verse or chorus of a song. But the fidelity of the tone is diminished in this mode. This often doesn’t matter that much because the loops are intended to be background and have their levels reduced anyway.

  • Playback: -2.6dB
  • Overdub: -4.0dB
  • Low Cut: 20 Hz
  • High Cut: 20.0 kHz


The output is set to Multi to feed the 1/4″, XLR, Digital (S/PDIF), and USB 1/2 outputs simultaneously. The output Level is set to -1.5dB and is controlled by the Drive switch (FS10). When the Drive switch is pressed, this cut is removed to provide a small lead boost.


I use four snapshots in this patch:

  1. Standard: puts the Variax input block in standard tuning
  2. Open G: puts the Variax input block in Open G tuning and switches to T-Model 1 (for Stones tunes)
  3. Acoustic: sets the Variax model to Acoustic-1,  turns off the amp and IR blocks and re-voices the Parameteric EQ for acoustic guitar.
  4. Leslie:


This has been a long post to produce a pretty specific patch. This tone may be useful to you directly, or as a starting point for tweaking your own variant. Or it may not be useful at all. But hopefully the thought process for how the blocks were selected, configured and positioned in the signal chain will be useful. Its like the Scientific Method – its not so much what we discover and learn from the method that is important, after all, things change. What’s important is the process through which we explore and discover those new things. There’s always more to learn. Have fun with Helix, and I hope this helps create great tones for you.

The Mid-Gain patch and Oxford 10K5 IR are available in my Dropbox.

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