Analyzing Joost Assink’s SRV Little Wing Patch

This is a brief analysis of the Ultimate Fender John Mayer Clean patch created by Joost Assink (JazzInc) and posted in the Line 6 Helix forum. I’d like to thank Joost for posting this patch and the included speaker impulse responses, and giving us the opportunity to analyze an interesting and different patch. Hopefully he’ll correct any mistakes I made in reconstructing the patch design and any of the included details.

There are three signal paths, two for parallel guitar amps and one for stereo playback through Return 1/2.

Path 1: Guitar with clean amp, delay and chorus

Path 1 starts with multi-input for guitar directly into a volume Pedal block. In this case, the volume pedal controls the drive into the amplifiers and can be pulled back just a little to eliminate the small amount of breakup you hear at full volume (especially with double coil pickups).

The Looper is next in the signal path meaning the raw guitar (after the volume control) is what is recorded in the loop. When overdubbing and playing along with the loop, all overdubs and the live guitar are going into the same amp and effects. Putting the looper at the end of the signal chain allows you to have different effects in different loop overdubs. In this configuration, its important to keep the amp pretty clean as any distortion created by the louder signal, either the loop or guitar, will tend to “duck” the other when the amp distorts. Note that the Looper is stereo, but should probably be mono since it is followed by mono blocks. This will increase the potential length of the loops.

There is a split following the looper to create two parallel paths, 1A and 1B. Path 1A has the Transistor Tape delay while Path 1B has 70s Chorus. Using this parallel path separates the delay and chorus so that the delay repeats are not modulated and retain their clarity.

The 1A and 1B paths come back together into the US Double Nrm amp model, the Normal channel of a Fender Twin Reverb. This is a pretty low gain amplifier and that helps keep it clean. The power amp is set to be as clean as possible, with Master on 10 to minimize preamp distortion from the Drive control. The Bias is set high and Bias X set low to make the amp even cleaner. Sag is set low to maintain articulation and tight response. Drive is set so the amp just barely starts to breakup when the guitar is played hard.

The ’63 Spring Reverb is chosen and placed between the amp and the speaker to better simulate the Fender Twin Reverb. Since the reverb is going into the speaker IR model, the brightness of the springs will be tamed by the speaker. This isn’t exactly the same as a Twin Reverb since the reverb unit is after, not before the power amp. However, this won’t make that much difference if the amp is mostly run clean. A Hall reverb would likely sound better after the speaker, to better simulate the speaker in a room.

Next is an Impulse Response block containing a speaker IR. I’m guessing what’s in the IR based on its name, and the file names in the Redwirez IRs. It looks like a G12M with a U67 condenser mic at CapEdge set at 50% blended with a Weber-Blue 12 with a Royer R121 ribbon mic at Cap set at 50%. This will be a pretty warm sounding speaker. Joost likes to close-mic the speakers, so these might be set at 0″.

The speaker is followed by a Studio Tube Pre Preamp set to be transparent. This is likely acting somewhat as a limiter using the Drive control to add some distortion to clip off the overly bright clean peaks.

The last thing in the signal chain is the LA Studio Comp compressor. I’m not sure exactly how the compressor is being used. The peak reduction is set to 0, so the compressor should not be applying any compression. The Mix is set to 54% meaning the compressor is being run essentially in parallel with the dry signal. This would provide some sustain as the compressor releases, by preserve pick attack which can get chopped of by a fast attack time, which the LA-2A has. I’m guessing the compressor is playing a similar role to the tube preamp, it just tames some of the harsh peaks that can come from a very clean guitar amp played loudly, but do it without adding any distortion.

Path 2: Guitar with slightly dirtier amp and distortion pedal

Path 2 receives the same multi-input and also starts with a Volume Pedal block. This block, and the one in Path 1 are both controlled by EXP Pedal 2 so that both amps are controlled together. Again, the volume pedal block is in front of the amp so it controls amp drive, similar to the volume control on a guitar, but without the high end loss from cable capacitance.

Next is a distortion block using the Scream 808 model, set reasonably hot and bright. Turning this pedal on provides distortion, but only in Path 2. Path 1 still has the clean sound, blending the clean and distorted sounds together. The Tube Screamer this model is based on also blends direct signal with the distorted signal, but this configuration provides greater control of the blending. Blending distorted and clean tones together can give the best of both. The distortion adds sustain and rich harmonics while the clean tone provides solid pick attack, articulationn and chord clarity.

The amp model in Path 2 is a US Deluxe Vib which will break up sooner than the US Double Nrm in Path 1. The amount of breakup is reduced a little by turning down the Master a little, and setting the Bias all the way up and Bias X all the way down. I’m not sure, but I think the US Deluxe Vib model has the bright switch on while the Nrm models don’t.

Again in Path 2 the reverb is between the amp and the cabinet IR. In this case a Plate reverb is used to provide a warmer, or different color reverb.

Next is the Impulse Response block for the speaker. Again I’m guessing, but it looks like a G12M with a U67 condenser mic set at 50% blended with a Weber-Blue 12 with a Royer R121 ribbon mic set at 40% followed by an EQ high-pass filter at 50 Hz and set at 40%. This will also be a pretty warm sounding speaker, but a little brighter than the other IR in Path 1.

A footswitch is configured for a minimum at 80 Hz and a maximum at 160 Hz. The same footswitch controls a 75 Hz and 180 Hz Low Cut on the IR in Path 1. These are probably to remove mud from the distorted tone. I prefer to use an EQ block to place the low cut before the distortion so that the low frequencies don’t create a lot of intermodulation artifacts with the mids. This clean up the mud and also makes the distortion smoother and sweeter. Both IRs also have High Cut set to 8 kHz to remove any fizz from the distortion without having much impact on the clean tone.

Path 2 has the same Studio Tube Pre block and LA Studio Comp after the speaker IR. The compressor has a small amount of peak reduction, and the Mix is set to 100% so there is no blending with the dry signal. This is likely to complement the compression in Path 1 and reflects the different role the amplifier plays in this path.

Combined Amps

Both amps are mono and centered on the output so that their tones are blended into a single cohesive whole. This is important since the effects in each amp are quite different. This is a great example of using different amplifier and effects for different purposes, and blending them together to combine the effects. This makes it easy to optimize each path for its intended purpose or contribution, then to blend them together, using their output volumes and panning to control the mix. This is an interesting patch that can help us learn more about how to use the Helix. Thanks Joost for the excellent contribution.

Here’s some final thoughts from Joost:

  • There is also a mid boost, which simulates the mid-cap bypasses you see on such amps. Can be handy for cutting through a dense mix.
  • The 35 impulse response adds some room sound and indeed is a little brighter on the darker amp
  • The LA2As just add a little something even when they’re not compressing, and they do take the edge off. Same goes for the tube preamp. Basically, I do what I would to in the studio: get some great speakers, great mics, a good preamp and a good compressor to capture the sound of the chain before it. I also really like the Emphasis control on the LA2A to tweak response. (same goes for the power amp characteristics on both amps to tweak feel and response.)
  • The double amp setup gives a lot of options. For instance, the treble on the Twin is a bit too much for me, so I like to tone that down and bring in the thicker, sweeter treble on the DeLuxe. Vice Versa for the Presence. For different applications these two amps can also be used to their strengths together.
  • I really feel you need very good IRs to get the best out of the Helix.
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Creating a Helix Electric Guitar Patch (updated)

Introduction

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.

This update incorporates 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.

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 ambience, especially in a larger room, to avoid washing out the tone and to fit better in the overall mix.
  • Simple ambience 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-capaticence 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 ambience 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 ambience, 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:

FS1

Delay

FS2

Chorus

FS3

Tremelo

FS4

Uni-Vibe

FS5

Phasor

FS7

Looper

FS8

Distortion

FS9

Overdrive

FS10

Drive

FS11

Compressor

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 lighter 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.

IMG_1643.JPG

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.

In this configuration, Path 1 has most of the mono blocks including before the amp effects, the amp and the speaker IR block. Path 2A is mono for 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 122 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, and MIDI CC messages do 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.

Guitar In

For this patch I have the Noise Gate on the input turned off since I generally use the patch with blues or clean tones. But its probably a good idea to leave the Noise Gate on with a minimum threshold in order to eliminate noise while still retaining the subtle dynamics of your guitar.

Wah: Fassel

The first effect in the signal chain is the Fassel Wah. Of all the Wah Wah pedals in Helix, this one sounds the most musical to me. I liked it in the HD500X too. Its before the compressor to deal with any odd peaks when using the Wah with a clean tone.

  • FcLow: 455 Hx
  • FcHigh: 2.2 kHz
  • Mix 100%
  • Level: 0.0dB
  • Controller: EXP Pedal 1
  • Footswitch: EXP Toe

Dynamics: Deluxe Comp

I like this compressor because it gives a lot of control that can be used to reproduce other compressors as needed. 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.

  • Threshold: -40.0dB
  • Ratio: 6:1
  • Attack: 38 ms
  • Release: 200 ms
  • Mix: 100%
  • Level: +7.0dB
  • Knee: +6.0dB

Preamp: Studio Tube Pre

The Studio Tube Pre is designed to come before any distortion to provide low cut to control bass mud and high cut to control treble ice-pick. This block is tied to the Drive footswitch (along with the Amp Drive control).

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 pretty high because the block doesn’t add that much distortion and I want to preserve the guitar high frequency response then the amp is just breaking up. There’s just enough high cut to keep the drive-level distortion from getting fizzy. See the Amp block for more details.

  • Drive: 7.5
  • Polarity: Normal
  • Low Cut: 120 Hz
  • High Cut: 7.8 kHz
  • Level: 4.9dB
  • Sensitivity: Line

Distortion: Valve Driver

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 at 10.0 (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 4.2 to 6.0. Recall that when the drive is at 6.0, the Low Cut is increased to 160 Hz to reduce the bass going into the distorted amps. Fender 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.
  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). Using the distortion models gives more control of both the distortion and the voicing, so that works best when using patch mode.

Valve Driver 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 and Treble are used to provide additional bass and treble cuts for higher gain distortion voicing.

  • Gain: 3.4
  • Bass: 5.5
  • Treble: 2.2
  • Level: 5.5

Distortion: Compulsive Drive

The Compulsive Drive distortion model is used to create the Distortion tone, and is 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.

  • Gain: 3.1
  • Tone: 3.4
  • Peak Type: High
  • Version: V4
  • Level: 6

Scream 808 (Ibanez TS808 Sube 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 phasor 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 seem 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 2 with the min and max values set to mimic the typical speeds of a Leslie speaker. Lamp bias controls how the effect ramps up and down.

  • Rate: 0.7 – 7.6 (Controlled by EXP Pedal 2)
  • 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

Amp: US Deluxe Vib

I’ve been using Fender amps for many years and at one time owned a Deluxe Reverb. I should never have sold it, but there you go. This amp model has the bright switch on, and has a little extra gain compared to the normal channel of the same amp. 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 Redwirex, 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

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 almost all the way up 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, support the Clean and Drive gain stages as described above. Recall that the Drive footswitch also controls the Studio Tube Pre to add some additional distortion and increase the 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 distortion model 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 lower the Bias X to provide a good clean tone. Reduce Sag to get a tighter tone.

  • Drive: 5.0 (Drive footswitch off), 6.0 (Drive on)
  • Bass: 4.7
  • Mid: 6.8
  • Treble: 5.3
  • Presence: 2.1
  • Ch Vol: 7.2
  • Master: 9.0
  • Sag: 4.0
  • Hum: 5.0
  • Ripple: 5.0
  • Bias: 8.3
  • Bias X: 3.2

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 taylor the sound.

After trying a lot of Helix Cab models, and a number of my own Redwirez and Rosen Digital IRs, I finally settled on 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.

  • IR Select: 33 (or the index were you loaded that IR)
  • Low Cut: off (the low cut is already included in the IR)
  • High Cut: off
  • 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 the JOOSTALNICO-G12M-R121-U67 IR.

Path 2

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

Preamp: Studio Tube Pre

This Studio Tube Pre is designed to come after the amp 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: 4.5
  • Polarity: Normal
  • Low Cut: 60 Hz
  • High Cut: 12.0 kHz
  • Level: 7.7dB
  • Sensitivity: Line

Modulation: 60s Bias Trem

All the effects from here on to the output are stereo. The effect order is modulation, delay and then reverb. Tremelo is a nice vintage effect, and one that’s included in the old Fender amps. So I included it this patch and assigned it to the Tremelo footswitch. The settings use a moderate intensity so that the signal doesn’t pulse too much. Spread is set to provide a small amount of ping-pong effect into stereo speakers. Set Spread to 0 for mono or no ping-pong, set to 10 for full left right ping-pong.

  • Speed: 3.2
  • Intensity: 6.4
  • Mode: Tremelo
  • Spread: 1.1

Modulation: Chorus

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: Triangle
  • Tone: 5.0
  • Spread: 10.0
  • Mix: 50%
  • Level: 0.0dB

Delay: Simple Delay

This is the first of two delays. The Simple Delay model is used to create a slap-back delay to create ambience without loosing clarity and articulation that can sometimes happen with reverb. This effect block is on all the time and therefore isn’t assigned to a footswitch. The mix is set so that the delay is barely noticeable when it is turned on. Scale is set at 76% so the slap-back comes more out of the right speaker, giving better ambience in a stereo FRFR amp. Scale at 0% puts the delayed signal entirely in the left channel, 50% puts the delay equally in both channels, and 100% puts the delay entirely in the right channel. Trails can be off since there are no repeats for this delay.

  • Time: 125 ms
  • Feedback: 0%
  • Mix: 18%
  • Level: 0.0dB
  • Scale: 76%
  • Trails: off

Delay: Mod/Chorus Echo

This delay adds an obvious 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 ambience 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 Mod/Chorus Echo provides some modulation of the delays, giving a wider, richer overall tone without creating a wooshy chorus on the main tone. Low Cut and High Cut are set to push the delay into the background where it won’t conflict with the main signal.

The Scale and Spread controls can be confusing, especially since they are not documented in the Helix manual. The Mod/Chorus Echo, like the 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 time offset for the right channel delay line, as a percentage of the left channel’s delay. Scale at 0% puts the delayed signal in the left channel. As the Scale is turned up, the delay is introduced into the right channel with a time offset. As you get closer and closer to 50%, the offset changes until at 50% the delay is ping-pong and even in both sides. As you continue to turn the Scale up towards 100%, the offset is re-introduced, but on the opposite side. The offset gets shorter and closer and closer to zero until at 100% the delay is equal and at the same time in both channels (i.e., mono). Spread appears to control the stereo spread of the modulation, and has no effect on the position of the delay repeats which are controlled by the Scale parameter. This is different with the PingPong delay where the spread determines the stereo spread of the ping-pong, from mono to full left/right. With Spread at 0, the modulation effect (chorus or vibrato) appears to be mono. With Spread at 10, the modulation effect bounces between channels and is in stereo.

I have set Scale high so there is just a little delay offset between the left and right channels. Mod Mode is set to Chorus in order to add a chorus on the delayed signal. Speed is set slow and Depth low to avoid over processing the delays so they appear to decay naturally. Spread is set to 10.0 so that the modulation on the delays is in stereo. Trails are on since there are repeats that fade out when the effect is bypassed.

  • Time: 500 ms
  • Feedback: 29%
  • Low Cut: 155 Hz
  • High Cut: 10.5 kHz
  • Mix: 18%
  • Level: 0.0dB
  • Scale: 94%
  • Mod Mode: Chorus
  • Speed: 1.0
  • Depth: 13%
  • Spread: 10.0
  • Trails: On

Reverb: Hall

Helix has lots of really nice reverbs. I personally like a very small amount of very natural reverb. So I choose the Hall 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.

  • Delay: 5.3
  • Predelay: 33 ms
  • Low Cut: 220 Hz
  • High Cut: 6.5 kHz
  • Mix: 27%
  • 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. 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.

Notice the mix is set at 50%. This provide parallel compression where the compressed signal is mixed with the try signal in order to get the advantages of compression while retaining the clarity and articulation of the dry signal.

  • PeakReduc: 0.9
  • Gain: 6.2
  • Type: Compress
  • Emphasis: 4.0
  • Mix: 50%
  • Level: 0dB;

Looper

The Looper is 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

Output

The output is set to Multi to feed the 1/4″, XLR, Digital (S/PDIF), and USB 1/2 outputs simultaneously.

Wrap-up

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.

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Creating an Acoustic Guitar Impulse Response for Line6 Helix

This post describes how to extract an Impulse Response (IR) from a Fishman Aura Spectrum (Aura) acoustic guitar body image that can be loaded into an IR block in the new Line6 Helix amp modeler. There are some free IRs on the Web that you can try with your guitar to see if this is something that might work for you. If you don’t have a Helix, you can still use the IRs by using a computer and programs like Apple MainStage and SpaceDesigner that support loading any IR in track plugins.

Background

I’ve been trying for years to get a good live acoustic guitar tone. I started with K&K pickups that combined a piezo under the body pickup with a pencil condenser mic. That sounded OK. The piezo wasn’t as bright or quacky as typical under the saddle piezo pickup setups, and the mic sounded good. But it required an external preamp to power the mic and mix it with the piezo, and teneded to feedback a lot.

After seeing James Taylor use a Line6 Variax 700 Acoustic, I got one and used it for years. It sounded better than most under the saddle piezo pickups, had lots of useful instrument models, played nice, and was very convenient, predictable and consistent for live gigs. But it always seemed to have a somewhat “digital” tone that was a bit harsh. I tried to EQ that away, and that helped. But I could never get a sound that made me go wow, and found I was reluctant to play some songs that featured the guitar because the tone just wasn’t that satisfying.

Then I saw that James Taylor used an Aura to get his fantastic live acoustic guitar sound. So I got one and it changed everything. I could now use my small, light Martin 00C-15AE guitar and a nice Taylor 314cd – Neumann U87 body image to finally get the tone I was looking for. In summary, the Variax 700 Acoustic sounded better than the 00C-15AE with just the Fishman under the saddle pickup. But through the Aura, the 00C-15AE sounded a lot better than the Variax. And that little Martin guitar has a magic feel that’s hard to describe.

I play three different instruments in most of our live gigs: mandolin, acoustic guitar, and electric guitar. I use a Collings mandolin with a K&K pickup, the Martin 00C-15AE, and a Stratocaster Deluxe with Tom Anderson pickups for electric guitar. That’s a complicated set of instruments with very different amplification requirements. I’ve been using Line6 amp modelers for years to provide a single device that supports patches for all three instruments direct into the PA. I recently upgraded from a POD HD500X to the new Helix and it really improved the usability and tone of my setup.

The Helix has support for Impulse Response (IR) blocks. So it seemed like it should be possible to use an acoustic guitar body IR block to free up the Aura for other uses, and further simplify my live rig. The rest of this post describes how to extract a body image from the Aura that can be used in a Helix IR block that reproduces the body tone.

Picking the Fishman Aura Spectrum Image

The first step in the process is to pick the image you want to extract. Picking the image requires auditioning the available images to pick the ones that work best for you, your particular guitar and playing style. You could send your guitar to Fishman and get a custom image created, but that’s expensive and probably unnecessary.

Here’s some guidelines for picking the image:

  1. Start with a properly setup guitar with new strings
  2. Use the pick you will use most of the time – using thicker, rounder picks for warmer tone
  3. Either use your computer or Helix with the Aura in an effects loop so you can record a short acoustic guitar loop to simplify the auditioning process. This lets you listen to only the processed sound, and leaves your hands free to select different images.
  4. Take notes on the images you like so that you can narrow down a small set that meet your needs.
  5. Be sure to audition through the speaker system you’ll be using live. What sounds good through headphones might not sound good at all through a speaker system. For example, I really liked the Aura Martin body images through headphones. But through speakers they had a ring and tended to feedback very easily. I suspect this is because of an emphasis around 200Hz that’s what makes Martin guitars sound they way the do and cut through. They might work for you, but I found the Taylor images to more suitable for my needs.
  6. Audition with the guitar still plugged in (if you are using a looper) so that any feedback issues will still be detected. Verify your final selections playing live to make a final check for any feedback issues.
  7. Make sure the Aura has the tone controls set flat, has the blend control set to full body (all the way up), and the compressor is turned off. You want to only hear and capture the body sound. Tone controls, blending and compression will be done outside the IR block using other Helix blocks.
  8. Start with body models that match your guitar directly or at least the body style and wood. You may find that other body images sound better. Don’t get stuck on what’s suppose to be “right”. Experiment with other body styles and woods, and pick the one that sounds best to you, for your style application, and amplification system.
  9. Use the Aura Gallery III app to load additional images into the Aura User patches if needed.

Once you have a set of body images selected, you’re ready to capture and convert them to IRs.

Setup the Audio interface

Helix supports nearly all IRs in .WAV format. However, the Helix application may
automatically change its attributes before sending it to the Helix hardware:

  • Converts all .WAV IRs to 48kHz mono and 16bit.
  • On Mac OS X, as of version 1.04.3, Helix will not load or convert WAF files with sample size of 24 bit. These files need to be converted to 16 bit before loading into Helix.
  • When loading a stereo file, Helix Edit uses only the left side.
  • Shortens (or lengthens) the IR to 2,048 samples (about 43 msec).
  • The user may choose a 1,024-sample version to save DSP. This option simply fades out the IR halfway through.

Helix’s IR blocks only point to a specific IR number (1-128); they don’t include the IR
files themselves. People using your Helix patches will need the same IR file assigned to the same IR index number to perfectly recreate the preset. Otherwise they will need to edit the patch and change the IR index number to where they loaded the IR.

Set your audio interface sample rate to 48000Hz and set the sample size to 16 bits. If your audio interface won’t support this setup, you will need to use an audio utility app to convert the sample rate or sample size before loading into Helix.

Capture an IR for the selected Aura Image

To capture an IR from the Arua, you need to connect the Aura to you computer using your audio interface. Connect an output of your audio interface to the Aura input, and connect the Aura output to an input of your audio interface. You can of course use Helix as your audio interface, but use a large buffer size and long latency to be sure to avoid any clicks and pops. In the examples below, I used a Focusrite Saffire Pro 40, connecting Line Out 3 to the Aura input, and connecting the Aura output to Input 7.

Make sure the Aura is set as follows:

  • EQ on the pickup only and/or EQ controls are set flat
  • Set the blend at 100% (all body)
  • Turn the compressor all the way down
  • Adjust the Volume for good signal level and no clipping (-18dB is a good target level).

To capture the IR:

  1. Open Logic Pro X Impulse Response Utility
  2. Create a project with these settings:
    • Mono project
    • Sweep Channel – the audio interface output connected to the Aura input (3: Line 3)
    • Set the Input to the audio interface input connected to the Aura output (7: IP 7)
    • Set Sweep Length to 10 s.
    • Reverb time is not important in this case because there’s no reverb to capture after the sweep. So set it as low as possible.
    • Save the project using the same name as the Aura image you are capturing
  3. Click the Record button and then click Sweep
  4. Check the levels and make sure there is no clipping. The levels can be adjusted in the IR utility, using the Aura Volume control, or the audio interface input channel gain control
  5. Press Deconvolve
  6. Trim the beginning if the IR to remove any empty content – use the scroll bar to zoom in order to see the beginning or end clearly.
  7. Crop the IR to about 43 milliseconds. If you drag from the beginning, you can see the actual length in samples in the upper right corner. Drag so the length is 2048 and then press Crop to get just that sample.
  8. Add a short fade at the end to avoid any potential clicks.
  9. You can audition the IR using a .aif file that captures a recording of you guitar’s piezo pickup.
  10. Click Create Space Designer Setting… – use the project name as the IR name. This creates a file: ~/Music/Audio Music Apps/Impulse Responses/<project name>.SDIR that is the IR.

Convert the IR file as needed.

You may need to do some conversions of the IR before loading them into Helix. The Saffire Pro 40 cannot be set to 16bit sample depth (neither can Helix), so I had to convert the .wav file. I use felt tip Sound Studio for these conversions, but there are many audio utilities that will work.

  • Rename the .SDIR file to .wav so that it can be loaded into Helix
  • The file should have Sample rate 48,000 Hz and, Bits per sample 16.
    • If the sample size is 24 bits, use Sound Studio to load the IR and re-save it with sample size set to 16.

Load the IR into Helix

Use the Helix app to drag and drop the IR .wav file into an open slot in the IMPULSES pane.

Then add an IR block to your acoustic guitar patch and select the IR index number to the desired guitar body image. Add EQ, effects and preamp models as needed to get the range of tones you need. I’ll have another blog posting on my Helix acoustic guitar patch soon.

If you reconnect your Aura through a Helix effects loop and select the same body image in the Aura, you can compare the sound of your captured IR block with the sound through the effects loop. If everything was done correctly, you should not be able to hear any difference between the two.

Some final thoughts

Using body images as IRs in Helix really opens up the possibilities for using Helix as an effects processor for many instruments besides electric guitar and bass. Body images such as those in the Fishman Aura Spectrum really do turn a quacky piezo sound into the wonderful, complex fullness of a good acoustic guitar.

The Aura images are more than just the body IR. There are complex algorithms used to create them based on the difference between the pickup and body sounds along with phase corrections. How to create an acoustic guitar impulse response shows how to capture an IR for your guitar. Bodilizer is used to analyze the difference between the body and piezo impulse responses. Unfortunately, Bodilizer doesn’t currently provide any means of exporting an IR that can be loaded into Helix. And Bodilizer development seems to be stalled. There are a few acoustic guitar IRs available on the Web, and hopefully Line6 will decide to support some in future Helix updates. But for now, capturing them from a Fishman Aura is a very good option.

You can try blending the body response with the guitar pickup by adjusting the Mix control of the IR block. Adding a little piezo and add definition and attack to the tone without adding too much quack. Some compression can also help tame any piezo attack emphasis and make the guitar stand out better in the mix. You can also use Logic Pro X Match EQ to attempt to match the piezo pickup output to the sound of your mic’d guitar. That will create an EQ curve that you could replicate in a Helix EQ block to better blend between the pickup and body tones.

Good luck, and I hope this helps get even more out of your Helix investment.

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Creating POD HD500X Electric Guitar Patches

Amp Tones and Effects
The POD HD500X is a complex device capable of a wide range of tones suitable to any musical genre. However, it can be a tweaker’s delight or tweaker’s nightmare. What follows are some notes and general advice I’ve collected over the last few months on how to setup good patches in a POD HD500X.
Try to get the most out of a each patch, and avoid switching patches too often. Setup four patches, each configured with the same effects, but using different amp and cabinet models for the specific purpose. The patches are all in the same bank, and progress from clean and mellow to distorted and aggressive.
  1. Clean – Blackface Dbl Vib, 2×12 Blackface Dbl
  2. Blues – Tweed B-Man Brt, 1×12 Blackface ‘Lux
  3. Crunch – Class A-30 TB, 2×12 Silver Bell
  4. Metal – Solo-100 Overdirve, 4×12 Blackback 30 (or some Marshall-style amp)
Get the desired amp tone first, then incrementally add effects. Use the Looper in the Pre position and record a dry guitar loop so you can tweet hands free. Check periodically at higher volumes and with a backing track.
Limit the number and wetness or depth of effects. Most of the effect blocks are taken up to support gain staging allowing a single amp to have a range of distortion tones and voicings. This results in a little overlap between the patches. But since the amps and cabinets are all different, the overlap is more in how the patch could be used rather than the actual tone. This overlap is actually useful because it allows each amp or patch to be able to be used for a wide range of tones using just a couple of foot switches.
Front of the amp effects:
  • Tube Comp – I use this to smooth things out a bit and provide a little extra boost in front of the first two patches since Fender amp models don’t have that much gain. I did some experimenting with a spectrum analyzer (my iPad) and found the Tube Comp was the only compressor that didn’t have high-end rolloff.
  • Fassel Wah
  • Tube Drive – This provides the most aggressive tone in the patch. Tube Drive is pretty transparent with Bass: 75%, Mid 50%, Treble 75%. Use the tone controls for preamp voicing control, roll of a little bass to reduce the mud, and roll of a little treble to manage any remaining fizz/ice pick.
  • Vintage Pre – this is the secret ingredient. I use the Vintage Pre essentially as another tube stage in the amp. It provides the first level of gain boost using the HPF and LPF as simple voicing controls in that gain stage to focus on the mids. Again, the Vintage Pre doesn’t do any other strange things to the tone and its simple, has its own drive control to add a little of its own color.
This allows each amp to have a progression of distortion levels and voicings. As the distortion increases, the voicing has to change, usually additional high and low cuts
  1. Tube Comp, Tube Drive and Vintage Pre off – the straight amp with the Gain control set to the minimum distortion level for that patch (not necessarily clean)
  2. Tube Comp on – just a bit more gain to push the front of the amp a little harder and provide some compression – this is often left on all the time
  3. Vintage Pre on – the first increase in distortion and subtle voicing change for warm bluesy leads
  4. Vintage Pre off, Tube Drive on – the next increase in distortion with a warmer voicing to mellow out the increased distortion
  5. Vintage Pre on, Tube Drive on – the greatest level of distortion with the combined voicings to reduce bass and treble.
Use the volume and tone controls on your guitar to fine tune the tone based on the song needs. Turning the tone control down about half way when using the bridge pickup can help take the edge off the tone and warm it up. Also, setup your tones using the neck pickup on your guitar, not the bridge pickup. This will provide the best compromise for overall guitar tone when you switch pickups, controlling the mud by setting the bass for the neck pickup and controlling the ice-pick using the bridge pickup tone control.
After the amp effects:
  • Analog Chorus
  • Digital Delay
  • Chamber reverb
  • FX Loop – mostly used for a Boss JamMan for longer loops.
Keep the delay and reverb on all the time for ambience, but pretty dry. Chorus comes and goes as needed.
Set the Input 1 to Guitar and Input 2 to “Same” for most patches – unless there are different instruments on path A and B. This ensures any mono effects in the Pre path (between the input and the path A/path B split) will have unity gain. 
Set the mixer to path A full Left and path B full Right – unless there are different instruments on path A and B or path B is muted. In those cases, Path A and B should be centered. Use the mixer gain controls to adjust the balance between path A and B when using two paths.
Run a bit of sag to provide some power amp compression and keep the Cab Resonance to 0 to avoid over-hyping the bass and treble. This keeps the tone warmer, more natural, less fatiguing and puts the control back into the voicings at the front of the amp. When the global EQ is available, use this for after the amp/effects final output voicing. This will be used to tweak the tone for different volume levels. Quieter settings might require some additional bass and treble.
General Guidelines
 
There are a number of potential reasons for getting poor tone from the POD HD500X:
  • Too many things in the signal chain making it confusing and difficult to manage
  • Too many EQs potentially overlapping or fighting with each other
  • Poor signal chaining or gain staging resulting in undesirable or uncontrolled distortion in the effects blocks
  • Too much gain creating an overly distorted, tone with limited dynamics
  • Wrong amp model for the sound your after
  • Wrong speaker cabinet or mic for the sound your after
  • Too much cabinet resonance
  • Too much Sag, Bias an dBias Excursion Settings
There are many reasons why tweaked tones don’t sound good live. Most are all based on tweaking out of context – headphones instead of speakers, low volume instead of gig-level volumes, by yourself instead of with the whole band, controlled situation instead of the chaos of playing live, long tweak times instead of having to do everything in a hurry. All these accumulate to fool us into thinking that something that sounds big, fat, wide, swirly, whooshie, big ambience, lots of sustain, etc. by yourself at low volume will be even better turned up and in the mix with the rest of the band. But the opposite is often true. Less is more when you’re playing live with a whole band.
  1. Tone starts with the song and the context. Listen to what’s going on around you and see what’s required to fit in, complement and add to the sound, not pull it in a different or odd direction. Think about dynamics and motion.
  2. Next comes the instrument itself. Make sure it is a good instrument, is properly setup, has a good set of strings,  and is well maintained.
    1. Pick the right pickup for the song, see above
    2. Its really difficult to set a single amp tone that works for all pickup combinations. Set the tone for the neck pickup, then use the tone control on the guitar to adjust the tone for the bridge pickup to avoid having it be too bright or uncomfortable
    3. Use the heaviest strings you can get away with. Heavy strings last longer, don’t break as easily, are easier to control when bending to get the right pitch and have better tone.
    4. Make sure you’re in tune – tune up, not down, stretch the strings to get out the slack so they don’t go out of tune. Make sure the instrument is at room temperature for final tuning.
  3. Believe it or not, this one is one of the most important parts of tone – pick the right pick. Thin picks make it easy to pick fast, but they’re too flexible to control string dynamics. Heavy picks have a wider dynamic range and sound warmer. Picks shouldn’t be too sharp. The rounder the point, the warmer the sound. Find picks that have beveled sides so they glide over the strings better. Use a good pick material. Wegen picks are very good. Try turning picks on their side to get different tones from the same pick. Use hybrid picking and picking with your fingers for more tone options. Don’t pick too hard, turn the amp up and pick softer, you’ll have more control, better tone, and won’t introduce a lot of fret buzz.
  4. start simple focusing on the amp, speakers and mic before adding anything else
  5. Keep the cabinet resonance low or at 0. Pick the microphone carefully. Mics that sound bright and full by yourself can sound fizzy and icy when turned up and with the rest of the band.
  6. do all the tweaking with a backing track, hopefully made from your band, but something similar will do
  7. start the adjustments at low volume so you don’t tire/kill your ears, but check periodically at the volume closest to gig level you can get
  8. adjust patches on the same FRFR amp you’re using for live playing. Headphones or studio monitors are likely to sound a lot different then the PA
  9. If you have to use a guitar amp with the POD, try to use a power amp input or effects return and turn off cabinet emulation in the POD. Going into the front of a guitar amp will be challenging, and patches will sound entirely different than in your headphones.
  10. when adding effects, be conservative, keep mixes pretty dry. Effects can add a lot of mud into a mix making your guitar become indistinct.
  11. Don’t overdo the gain/distortion. These can kill definition and the tone of your actual guitar. Use the minimum gain/distortion required to do the job.
  12. Check your gain staging between effects. Bypass all effects and add them back in one at a time making sure no effect is so hot its overdriving the input of the next effect in the chain. I like to keep effects at mostly unity gain – they’re the same volume on or off. There are exceptions, when I use something like the compressor or vintage pre to intentionally drive the amp harder. This works good for Fender amps which are naturally lower gain, but should be unnecessary for high gain amps. Use the drive control instead
  13. Use a dB meeter (or audio tools on your phone) to set the channel volume so that you get the intended volume differences between patches.
  14. Once you get some patches you like in a live setting, use them as reference patches for tweaking at home. Once you know what these good patches sound like at home and through headphones, you’ll have a good reference point for building other patches. Switch back and forth between the patch you’re working on and the reference patch and think about the differences and how they translate to the song or what you’re trying to achieve.
  15. Make sure you’re not clipping the input of your FRFR amp or the PA.
Setting levels
There are 7 controls that interact to give distortion and volume. Start from the front and work to the back.
  1. Drive – sets the gain into the amp model preamp and controls preamp distortion
  2. Master Volume – sets the volume into the amp model power amp and controls power amp distortion. Preamp and power amp distortion sound different, so the relative settings of these controls set the amount and color of the overall amp’s distortion. Set these first to get the tone you want. Its generally best to use the smallest amount of distortion needed to fit the song.
  3. Channel Volume – controls the relative volume of each patch. This is the control you adjust with your dB meter to balance the volume between patches. All of the above are saved in the patch. Adjust this after you have the distortion and tone you want for the patches.
  4. Volume – the POD output volume for all patches. Start with this about 3/4 way up. It provides a convenient control for you to adjust the overall volume of all your patches up or down into the PA.
  5. PA or FRFR Input Gain – set this to ensure proper signal levels into the PA, ensuring there is no clipping into the PA even if you turn the POD volume all the way up.
  6. PA channel strip volume – sets the volume of your instrument relative to the rest of the band, usually controlled by the sound man.
  7. PA master volume – sets the overall volume of the band to the audience.

Using a Looper for Solo Gigs

I had a very nice discussion with Tim (didn’t get his last name) at restaurant in Darling Harbour Saturday, Sydney Australia, Oct 12, 2013. He was doing an afternoon solo gig. Tim is a bass player, but seems pretty well rounded. He played a combination of acoustic guitar and a uBass leveraging a looper, along with his vocals. We discussed his approach and best practices for using a looper live which are summarized below. This entry summarizes some of the points of our discussion, and a bit more thought I put into afterwards.

A looper is like adding another instrument, one you are playing along with other instruments. It takes a lot of practice to get good at using a looper. The key skills are starting and stopping accurately, on the beat, and using the loop layers effectively to add to the performance, and not clutter it. I hope you find these best practices helpful. Let me know if I missed some.

1. Using a looper requires careful selection of the songs – they must be amenable to looping as described in some of the other best practices. Not all songs are good candidates for looping.

2. Avoid songs where the only practical thing to loop is the verse, not the chorus. This tends to make the chorus fall flat instead of being a crescendo as is usually the intent.

3. Pick songs that are relatively simple. Complex songs are hard to manage with a looper as there’s too much going on already, and the looper can become a distraction that inhibits the rest of the song.

4. Keep the loops really short, ideally four bars or so. This minimizes the time required to create and/or layer loops.

5. Don’t dedicate too much time in the song creating the loop layers. Five minutes to create the loops for a three minute song doesn’t make sense.

6. Create the loop at the beginning of the song, introducing each instrument in a layer as part of the song introduction.

7. Alternatively, create a verse or chorus loop while doing the first one, while singing the vocal, so the audience never notices the creation of the loop and it adds no time to the song.

8. Don’t run the loop the whole song. Turn the loop on and off to give the song some dynamics and flow. Keeping the loop on too long can become distracting, and makes the songs sound thin when the looper is off or when starting the next song. The sound needs to be relatively consistent within and across songs.

9. Keep the loop layering simple – no more than three layers usually. More takes to long, introduces more chance for errors requiring undo or loop creation restart, and can make the overall sound distracting as it clearly isn’t coming from the performer.

10. Practice starting and stopping the looper to ensure good loop timing.

11. Starting a loop creation directly off a count-in can be tricky. Practice this. But often its better to start the song intro without the looper and then create the loop after the song is in progress, the tempo is set, and you’re in the song groove. This will make it easier to be more accurate with the loop start and stop times.

12. Work out the arrangement of the song ahead of time and lay it out in your SongBook. Don’t try to do the arrangement and loop planning live. Have it worked out ahead of time what will be looped, when and with what content, and when the loop will be on or of.

13. Avoid creating multiple loops in the same song (which requires a loop reset). Its too distracting.

14. Rehearse with the looper, practicing exactly what you planned to perform. A looper is like learning another instrument and takes practice all by itself.

15. Use loops mostly to provide a background instrument for solos. This keeps the song consistent since the loop is the same thing you were playing during the vocal with the solo guitar replacing the vocal.  The song will have a coherent and consistent structure and sound without the loop adding a lot of unexpected and inconsistent content.

16. Be consistent. Your performance is a conversation with your audience. You can move from tension and release within and between songs, and reinforce this with the looper as another instrument. But if you use a looper in one song, use it consistently in similar songs for continuity of the sound. Don’t perform with no looper on one song, followed by five layers of loop on the next similar song.

17. You have to somehow synchronize the start and end of a loop, and anything you add to a loop, either with overdub or multiple loops. If you have a foot pedal, then you can start and stop the loop while keeping you hands free to play. This can take some practice, especially for establishing the tempo for the first loop. But it works best and requires the least amount of loop reparation time. A good looper (like Loopy HD) can even determine the tempo from the first loop, and establish the number of measures in order to support changing the length of subsequent loops.

If you don’t have a foot pedal, then you need some way of getting the loop started and stopped at the right time. This usually requires:

  1. setting the loop tempo
  2. setting the loop length so it can stop automatically
  3. doing a count-in to synchronize your playing with the start of the loop
  4. Using a click or metronome with the loop to keep tempo

That’s a fair amount of setup for the first loop. After that, you have more flexibility on subsequent loops since the existing loop is essentially providing the count-in synchronization, and you’re free to start the overdub anytime that is convenient.

If you’re using multiple loops, and they can have different lengths, then more setup is required between the loops. Subsequent loops are generally whole-number multiples of the initial loop in order to ensure synchronization. So keeping the initial loop very short, even just one measure, makes it easier to add loops of different lengths.

If you’re using the looper in Apple MainStage 3, note the following:

  1. The metronome doesn’t necessarily start on the one. Rather it appears to be running all the time, and you just turn its audio on and off.
  2. Pressing the count-in button in the Looper will count in up to one measure. The count-in starts when you press record and the beat is determined by the metronome. So if the metronome is on the 3rd beat of a measure when you press record, then you’ll only get 1 beat of count-in.
  3. The metronome needs to be on for the count-in to be meaningful, and to provide something to sync with since the looper should be syncing to the beat and stopping at the end of the bar of the last measure.
  4. When sync is off, and there is no number of measures set, the following happens:
    1. pressing record starts record and play
    2. the next press of record sets the end of the loop, but does not turn off record
    3. to set the loop end, turn off record and turn on playback all at once, press the play button while recording
    4. press record again to turn on recording anytime while the loop is playing back to add additional layers. Record can be turned on or off anytime during the loop playback and does not restart the loop. Playback simply continues

5. Creating a Project: Using Project Templates

In our last entry we discussed how to get input into the GarageBand on iOS. With that somewhat complex (and possibly expensive) problem behind us, we’re ready to create a project and start recording a song.

Projects are where you do the work of arranging the song, choosing instruments, recording tracks, editing and mixing. That’s basically all the rest of the things you need to do to produce a song. The big things that are left are mastering to produce a final product that includes content from multiple projects or songs, and distributing the results.

In this blog entry we’ll look at various approaches to creating a project, including the use of project templates to speed things up and save work. We’ll look at song layout and different approaches to recording songs: either a section at a time, or by laying down more complete or even free-form tracks.

Creating the Project

Each song is created in its own project, what GarageBand calls a Song. Most DAW’s use the concept of a Project since there are many things you might record besides songs. So we’ll use the terms interchangeably when referring to GarageBand.

When GarageBand starts up, it usually picks up where you left off. That’s very convenient, but can sometimes be confusing to new users. To create a project, you need to be viewing the song list. If you’re in a song, you get back to the song list by pressing the My Songs button.

To create a new song, press the + button. You are presented with two options, New Song or Duplicate Song. Press New Song for now, we’ll look at using Duplicate Song later when discussing templates.

GarageBand immediately presents you with the Instruments selection page where you select the instrument for the first track. Let’s pick Smart Drums as that’s a great way to lay down a quick drum track we can use to set the tempo and guide the rest of the tracks.
Next you will want to press the puzzle piece button to edit the song sections. We’ll cover this more below when discussing different ways of laying out and recording songs. So for now, just use the default Section A with 8 bars. Next select the Drum Machine you want to use. I like the Vintage kit or the Classic Studio Kit to use for a drum backing track as the drums are fairly simple and clean, making the track easier to follow. Select a few drums and arrange them for intensity (up/down) and complexity (left/right) until you get a rhythm that seems to fit the song, and provides a good solid rhythm track. Keep this simple as simple tracks are easier to follow when recording other live tracks.

Make sure you set the tempo and time signature correctly before going any further. This is another good use of a simple drum rhythm track. It gives you something to play and sing along with to be sure the tempo is right before recording any other tracks. It is very hard to change tempo once analog instrument tracks have been recorded. Some DAW’s do support temp changes and will stretch the audio to fit the new tempo without changing the pitch. GarageBand on iOS doesn’t support this. And it can only be done to a very limited extent without starting to introduce distortion. So its best to get the tempo right before recording anything.
Now press the rewind button to make sure the play cursor is at the beginning of the section and press record. GarageBand will record the drum loop and stop recording and switch to playback when it gets to the end of the 8 bar section A.
This might be a little surprising. If you forgot to set the section length to the right number of measures, or you didn’t have Automatic section length turned on, then recording will stop when the cursor reaches the end of the section. This may not be what you intended. But it is convenient for getting the initial drum section recorded as we can easily loop the drum track in the other sections.

Now you can go to the tracks view to see what got recorded, and get ready to create and record the rest of the tracks, using the drum track as a guide for different song sections, tempo and when the song ends.

Using Project Templates

GarageBand for iOS does a great job keeping the song creation process simple and easy to do. But there can be a lot of repetitive activities that are the same from song to song, especially for the same band. Most DAW’s support project, track, instrument and effects templates that you can setup for common recording needs. Templates have everything already setup so you are ready to pick a track and start recording. They can save a tremendous amount of time in setup and management of projects.

GarageBand doesn’t support templates directly, but you can simulate them by creating what we’ll call “template projects”. Essentially these are projects you create that have a drum track, and all the other tracks added with instruments selected for a particular recording situation. Say you have a band, and most of your songs will use the same instrument and vocal tracks. Why not create those tracks once, and then reuse them for each new song?

To do this, start with the simple drum track you created above. Now add tracks for each instrument you want to record, but don’t actually record anything in any of the tracks. GarageBand will let you switch to the track view from a selected instrument as long as there’s at least one track in the song. That’s our drum track. The label for the track depicts the instrument you chose for the track. You can’t change this label. But once you’ve recorded something in the track, you can relabel the region itself. You can use the region names give the tracks more reasonable names, indicating what actual instrument is being played and/or by whom. My project templates have mostly audio input tracks that all look like microphones and are labeled “Audio Recorder”. So I can’t tell the vocal tracks from the acoustic guitar tracks. And some of these tracks may be configured specifically to what is being recorded. For example, some might be configured as mono and others stereo. Note that with Audio Recorder set to stereo, and careful separation of the inputs, you can record two things at a time in GarageBand, and use only one track. You can use the Pan control to somewhat control their relative volume. But these will be essentially hard-panned left and right, so you’ll need to choose instruments appropriate for that positioning in the sound stage.

Now save the song and change its name to My Template or something that indicates what the template is for.
To use the template, go to the My Songs list, scroll to the desired template and long press the the song project template until it starts wiggling. Then press the duplicate button (+ in a square) to create a copy of the template. Next, long press the name of the template copy to set the name of the new song project. You now have a new song with an initial drum loop and all the tracks setup that are appropriate for that song. This can be quite a time saver. Now just set the tempo, adjust the song layout, and you’re ready to start recording tracks. Depending on the song, and the drum track in the template, you may have to redo the drum track to something more appropriate. That’s part of the reason for creating very simple drum tracks in the template – they’re more reusable in other songs as a simple rhythm backing track to augment the metronome while tracking.

Workflow: Section at a time or track at a time

Before recording much beyond the initial drum loop like we did above, you need to plan the layout of the song. Typical songs have an intro, some number of verses and choruses, perhaps a bridge between different parts, and an outro or ending. There are three different approaches to laying out a song. Each has advantages and disadvantages and one may be more appropriate for a particular song than another.

1. Free Form Layout

In free form layout, you will often turn off or delete the drum layout track, and turn off the metronome. Then you create a single section for the song an set the section length to Automatic (on). Finally, you pick an instrument like an acoustic guitar or piano and play the whole song start to finish. The length of the section is automatically set to the length of this track.

This is the way analog recording with at tape machine was always done. You may have had an external metronome to keep tempo, but the layout was often done with a “layout track” that established the different song sections and tempos, and is used as a guide for the rest of the song. This track may or may not be included in the final mix, so its not necessary for it to be mistake free – only that it keeps the desired tempo and has the song sections the way you want them.

This can also be very liberating because you’re not bound to a strict tempo or measure boundaries. But this freedom comes at a hight cost when you start adding tracks and editing later on. If the song is not following a metronome and/or rhythm track, then the measure boundaries aren’t meaningful. This makes region selection difficult when you need to edit later on because the tracks don’t lineup on measure boundaries.

Free form is however useful if you’re recording a number of instruments at the same time, and there’s a lot of interaction between the musicians that requires flexible tempo, pauses, etc. In some DAW’s you can record a video of a conductor while the audio recording is being created so the conductor is available to direct subsequent tracks.

2. Track at a time

The next technique is to record a complete track, but following sections you laid out ahead of time, and using fixed tempo with a metronome and/or drum track to ensure the recording is on measure boundaries. Using this approach allows you to plan out ahead of time the different sections of the song, but still play the whole track all they way through with each instrument. After laying out the song sections, make sure you select All Sections in the Song Sections list or recording will stop at the end of the current section.
When you are recording, you can see the section labels in the timeline to help guide the musician. The labels are very dim and hard to see, but they are there. Remember you can also drag down from the timeline in the instrument recording view to see the track you are recording, and adjust its volume, solo or mute.

A GarageBand song can have a maximum of 26 sections (Section A through Z), with a total maximum length of 320 bars. Each section is 8 bars by default. Tap the information button to change the number of bars. The last section of a song has an Automatic button. Automatic means this section will continue to record until you press stop (or reach the 320 bar limit). The length of the section is determined when you stop recording. Turn off Automatic in order to manually set the number of bars for the last section. You can also duplicate and rearrange song sections. This is particularly useful when building a song a section at at time as described next. But its also useful for duplicating the drum sections you may have created for verse and chorus sections.

You can also start with a single Automatic section, record a scratch or layout track, and then go back and create the individual sections based on the layout track. Each new section will slice the layout track at its length. Just select All Sections, and change the length of the layout track to expose the original length so you can create the next section.

Once you have the song sections laid out, you can select each section in the song sections (or arrangement) list, and record different drum patterns in each section. Adjust the drums, volume and complexity in each section to fit the section of the song. This establishes a good starting point for the rest of the tracking.

3. Section at a time

In this approach, you record each section of a song independently, then you can duplicate and rearrange sections to change the layout after the recording has been completed. DAW’s like RiffWorks make great use of this technique to very quickly create songs. The benefit is that you only have to focus on getting one section at a time right. And then you can duplicate that section as many times as you need it to complete the song. Use this on instrument tracks that repeat in different sections of the song to quickly build up the song. Then the final vocal or lead instrument parts can be done a track at a time as described in the previous section.

This approach is fast and easy, but has some drawbacks.

  • It may be hard to get a sense of the whole song doing a part at a time. You can minimize this effect by combining approaches 2 and 3. Use section at a time to get instruments recorded that are background or rhythm section instruments that don’t change much in similar sections (verse, chorus, etc.). Then use track at a time to record the vocals and solo instruments that need more flexibility within similar sections.
  • The sections can all sound the same. Address this by using the same technique as the previous item, record the main instruments and vocals using track at a time and the less important instruments section at a time.
  • Transitions from one section to another may be more noticeable since they were recorded at different times, in a different position, with slightly different timing, different mic placement and settings, etc.
  • Counting out the measures can be a pain – sing the whole song agains the drum track to be sure you got the layout right before proceeding.
  • You have less content to choose from when editing. Its often easy to fix an error in one section by copying content from another section that doesn’t have the error.
  • You hear the same mistakes over and over in the song.

Section at a time is a very good way to get a song laid out and the initial tracks recorded very quickly, especially smart drum sections. You aren’t spending a lot of time performing the same content over and over. It provides a good foundation for the more important instrument and vocal tracks that often come later.

The important thing is to make sure the tempo, time signature and layout sections are right before recording anything beyond the initial drum sections. Recovering from incorrect song tempo or layout can result in a lot of very difficult editing and a poor result. Planning ahead of time here can save a lot of problems, disappointments and poor compromises later on.

4. IO Devices: Getting sound into the iPad

IO devices for iPhone and iPad have substantially improved in the last year. The introduction of iPhone4 and iPad set mobile recording back somewhat because of the elimination of the stereo analog inputs on the dock connector. Then the iPhone5 changed things again with the Lightening connector. A lot of microphones created for the iPhone, like the wonderful Blue Mikey 2.0, are not compatible with iPhone4 and beyond, or iPad. Its taken a while, but there is now a very good selection of devices that support multiple inputs, different devices, have very good quality and are quite affordable. Most also work with any laptop computer too. Choosing a device depends on what sound sources you are going to be recording, and where you will be capturing the tracks.

The iPad and iPhone have introduced a whole new approach to mobile multi-track recording, making it practical to use mobile platforms for nearly complete music production and distribution. This is fantastic especially for hobbyist and traveling musicians. It is now possible to record tracks at the moment and in context almost anywhere, anytime, enabling the capture of musical ideas that would often have been lost in the past.

But on to IO devices. An important consideration is to maintain portability. You likely have your phone with you all the time. Recording is a bit like photography. The best camera is the one you have with you when you need to take that fantastic picture. So the initial focus on iOS IO devices should be to maintain the mobility of the platform. Otherwise you’re better off using a laptop or desktop DAW system with a multi-channel IO device. They don’t cost that much more, most of us already have one or more computers, DAW software like Reaper is getting much less expensive and more stable and capable, GrageBand for Mac is now free, and there’s a wide range of inexpensive USB and FireWire devices to choose from. But these platforms aren’t that mobile, or easy to use when you are both the performer and the recording engineer.

GarageBand on iOS only allows recording one track at a time in either mono or stereo. This has lots of implications for how you do tracking that will be covered in later posts. But it also has an impact on what IO device you choose. Auria, Cubase and MultiTrack do support recording as many as 24 separate inputs simultaneously using a Tascam US-800 or similar USB audio device, and the camera connection kit. So there are other options. But we’ll focus on the simpler, more mobile devices that are consistent with recording one track at a time.

Most of us will be recording sources such as:

  • Acoustic instruments in either mono or stereo
  • Vocals
  • Electric guitar
  • Bass guitar
  • MIDI keyboards or other tone generator devices

This requires an IO device that can support:

  • Mono and stereo microphone input
  • High-impedance 1/4” guitar input
  • MIDI input
  • Stereo line input

There are devices that support all three of these requirements. But you may find that for mobile recording, it may be useful to use a different input device for microphone, guitar and MIDI in order to keep the devices small, portable, low-power and battery friendly. I’ll start with the simplest, most portable devices, and work up to the more full-featured, covering the pros and cons of each.

Microphone Input

Tuscam provides the iM2 and iM2X stereo mics for iPhone and iPad. This is a really great choice. Both provide mono or stereo recording, the ability to position the microphones to the front or back, have a built-in limiter, support an external gain control, support USB charing while recording, and are very light but with good construction quality. I did some testing of the iM2 mic in comparison with others, including an AT-4047 LDC mic and the iM2 compared very well. Its missing a bit in the low end, but not bad, and that’s not always a bad thing for vocals or acoustic guitar.

However, there are some problems. The first is that both use the 30 pin dock connector and therefore require the 30 pin connector to Lightening adapter to use on iPhone5 and beyond. This isn’t really a problem as it moves the mic a bit further from the device providing more convenient use. Second, the iM2 does not have a headphone monitor output, rather it relies on the headphone jack already on the iOS device. This is a good choice since the jack is already there and we use it constantly. But GarageBand may sometimes get confused about what output device to use. If the headphones are plugged in and you plug in the iM2, the headphones are deactivated. The work around is to plug the iM2 in first, then the headphones, or just unplug the headphones and plug them back in to reset the audio output.  By the way, use caution when recording with earbuds, they tend to leak into the microphone. Its better to use noise isolating, closed back headphones. And don’t use earbuds or headphones with an in-line microphone – this will take priority over any mic plugged into the Lightening connector.

One more thing to watch out for, the current Tascam PCMRecorder app stops recording when the iOS device screen goes to sleep. So you’ll miss content unless you keep the screen alive. These apps should all have options to suspend screen sleep while recording so you can watch input levels and have immediate access to the stop button when recording is completed.

Another option is the Mikey Digital from Blue Microphones. This mic supports automatic gain control has a clipping indicator, and a 3.5mm aux-in jack for recording from stereo devices. Both the iM2 and Mikey Digital have a USB pass-through for charging while recording. This is useful for iPhone since recording with a mic results in high power demand and short battery life. A somewhat more expensive option is the Rode iXY. The iXY supports 24 bit recording which is important for increased dynamic range and typically lower noise. However, GarageBand currently only supports 16 bit recording so you won’t get that advantage unless you use something like Auria.

All of these microphones are small enough to have with you all the time for jam sessions, practice sessions, or those times when you go sit by the water playing guitar and get that great new idea.

Next up are the somewhat larger table-top microphones such as the Apogee Mic or Samson Meteor. These microphones generally have a larger diameter sensor and therefore have better bass response. I have both and they also sound great and compared well with the AT-4047. Both have the advantage that you can plug it directly into the iPad camera connection kit, for a simple mobile platform. Unlike the iM2, the Apogee Mic or Meteor is not physically connected to the iPad or iPhone so there’s more flexibility for mic and iPad positioning while recording. Another benefit is the these mics also work with any computer having a USB input.

The Meteor is mono only, and unfortunately, unlike the Apogee Mic, doesn’t have an external gain control. Until iOS 5, and the GarageBand 1.1 update, the mic was too hot and there was no way to control the mic gain using iOS. The only way to avoid clipping was to move the mic further away which introduces noise into the track and more room effects. GarageBand 1.0, and at this time, all other iOS recording applications have an input gain control, but it only controls the input gain of the iOS device, not the gain in the Meteor mic. GarageBand 1.1 and beyond fixes this. The gain control on the mic input recognizes that the Meteor USB mic has internal gain control and actually adjusts the gain in the mic, not just the gain in the iOS device. You can see the difference by looking at the clipping light on the Meteor mic. With other applications like MultiTrack or Meteor recorder (not related to the Samson Meteor mic), you can turn down the input, but the mic still clips and the sound is still distorted. With GarageBand 1.1, when you turn down the mic input, the gain in the Meteor mic is actually reduced and the mic will no longer clip, same as with Mac OS X. This eliminates the need to have an external gain control on the mic, but requires the apps to support USB mic drivers. Fortunately, since iOS7 these mics now also work with the iPhone since the camera connection kit is now supported by iPhone.

A great option might be the Apogee Mic or the Spark Digital from Blue Microphones. These both have external gain controls which are easier to use than changing the gain in the GarageBand input dialog anyway. Generally the clip light on the mic corresponds to clipping in the app. And the Apogee Mic just sounds fantastic.

Another option, especially useful if you already have good condenser mics is to use something like iRig Pro which supports XLR input with phantom power, as well as a 1/4″ hi-Z input for guitar and a MIDI input. This is a good all-around device that has good quality construction and is very convenient.

One last thing about microphones, for vocals, you need to use a pop filter. The mics don’t generally come with one, so you need to get something separate. Any music store will have something that will fit over the mic and provide some protection. Be sure to use one for all vocals.

Guitar Input

There are two ways to get electric guitar into an iOS device. The simplest and least expensive is through the device’s headphone/microphone jack. Devices like the AmpliTube iRig from IK Multimedia fit into this category. These devices are very susceptible to feedback, are generally quite noisy, and suffer from the 200Hz high-pass filter built into the microphone input jack. I don’t find these devices useful, especially for bass guitar, so I won’t cover them further.

The other class of devices are those that plug into the iOS dock or Lightening connector. Devices like the Apogee Jam and iRig Pro fit into this category. These devices have to have their own analog-to-digital converters and a USB interface, so they require power and can be relatively expensive. But the Apogee Jam works perfectly and is highly recommended. Jam also comes with a separate cable that works with any computer having a USB input. All the popular guitar amp simulator applications including GarageBand, AmpKit+ and JamUp work  with the Apogee Jam. These are all good apps, especially JamUp and the companion app, Bias.

I like the guitar amp simulators that are in GarageBand, but they’re not the best. JamUp is great too. You can easily use JamUp with GarageBand using AudioBus. Hopefully JamUp will support IAA soon.

I also sometimes use a Line6 Pocket POD to get guitar into iOS. You can use this with the Apogee Jam, or any other device that supports guitar or stereo line input. The Pocket POD is based on the original Line6 POD 2.0 tones. These don’t compare with the latest Line6 simulators like the POD HD series, but they do sound better than the amp simulators built into GarageBand. So that’s an option if you have a Pocket POD. The Pocket POD also has a “CD input” so you can plug the output of your phone directly into the POD to play along with tracks from iTunes, or jam against backing tracks using OnSong or SongBook. This is very useful and sounds great.

GarageBand doesn’t currently have a bass amp simulator, so you need to use something external or record direct. It is not uncommon to record bass direct. But unfortunately, GarageBand for iOS also doesn’t yet have track or master EQ, so there’s not much opportunity to adjust the bass tone. You can use an external preamp like a bass SansAmp, but that tends to make the platform more complicated and less mobile. I use a Carvin 5-string electric bass or a Dean fretless acoustic bass guitar (ABG) for bass. Both have active electronics and provide a great sound direct into any device. I also use the bass amps in JamUp if I need one, through AudioBus.

MIDI Input

iOS has introduce CoreMIDI support and there are a number of devices that support it including iRig Pro, iRIG MIDI and Line6 MIDI Mobilizer II. Note that the original MIDI Mobilizer does not support Core MIDI and won’t work with GarageBand. These devices are straight forward, and easy to use. iRIG MIDI has support for MIDI Thru. Both support MIDI 5-pin input and output.

Many modern keyboards support USB output, and these will work directly with the iPhone and iPad using the camera connection kit. For example, the Akai LPK26 USB keyboard plugs directly into the camera connection kit connected to an iPad and requires no external power source. Of course that’s a pretty small keyboard so it may not be adequate for all applications.

Other USB keyboards may require a power source or a USB hub. You can provide such a power source easily by using Velcro to attach a simple powered USB hub to the keyboard, and then connecting the output of the USB hub to the camera connection kit. This has the advantage of supporting more than one device such as the Blue Yeti microphone and the M-Audio KeyRig 49. So I highly recommend getting a keyboard controller that support both USB and MIDI outputs for more flexible connections.

For other sound sources such as tone generators, you’ll need a stereo analog input to a stereo track. Unfortunately there aren’t too many highly mobile options for analog stereo input into iOS devices at this time. The Mikey Digital might be a good option.

All-in-One Solutions

The above solutions are all separate devices, and most are mono. Another approach is the more traditional one or two-channel USB audio devices. There’s a lot to choose from here since the USB audio devices that work with most computers will also work through the camera connection kit. Many require separate power, especially when used with the iPhone. I’ll only cover two that have some unique characteristics: the ART USB Dual Pre, and the Alesis iO Dock. Both of these devices maintain mobility while allowing more flexible input/output options and the ability to use your existing microphones. They are less portable than the options described above, and is boarding on a solution that’s not that different than using a laptop and DAW. But this might be a good intermediate solution especially if you want to use your existing microphones or need stereo input.

The ART USB Dual Pre is a typical 2-channel USB audio device. It has everything you need for mic, guitar and stereo inputs and outputs for iPad and it can have external power, but doesn’t require it. The Dual Pre can be powered directly from the iPad camera connection kit, making it a reasonably mobile platform. The Dual Pre also has a separate preamp built-in that can be powered by an internal 9-volt battery if needed. The Dual Pre has no MIDI input though, so you’ll need to have one of the other solutions describe above if you need MIDI.

The Alesis iO Dock is an a good option for the iPad. It provides all the capabilities you’d expect from a two-channel USB audio/MIDI device in a form factor designed for iPad. I won’t go into all the features and benefits because they are well documented on the Alesis web site and in many great reviews. I have had great success with this device, but it hans’t been trouble free. I find the iO dock sometimes suffers from significant digital noise and has to have its power cycled to clean up the sound. This has resulted in some lost recordings – see the next section on direct monitoring. Another problem is that the iO dock only supports 5-pin MIDI, not USB MIDI. So my small, light-weight USB-only keyboard controllers won’t work with the iO dock.
The iO dock might be ideal if it:

  • Supported battery operation. It has to be connected to an AC power source significantly limiting its mobile capabilities.
  • Provided audio input/output from the USB connector so the same device could be use with a laptop computer for USB audio IO.
  • Supported USB MIDI input, not just 5-pin MIDI connectors
  • Had a mono switch for better direct monitoring during tracking, currently direct monitoring puts the source on only one side.
  • Supported Lightening connector and iPad Air

The Dual Pre works well with GarageBand since it is also limited to 24 bit. Some research may be required to determine what’s best for you. The Roland Duo-Capture EX looks good, but check the reviews before purchasing.

Direct Monitoring

Direct monitoring is the ability to monitor the recording input source directly instead of having to go into GarageBand and back out. Direct monitoring has the advantage of no latency. Latency is the delay you hear between the original sound source, and what is played back in your headphones through the IO device, iPad and software. Latency isn’t as much of an issue as it use to be since IO devices, digital input channels such as USB 2.0, computers and mobile devices are much faster than they use to be. But it can still be an issue, depending on the application.

Of course you can’t use direct monitoring if you need to hear the input as processed by the application, say an electric guitar amp simulator or a MIDI virtual instrument. But you can use direct monitoring for acoustic instruments and vocals.

Direct monitoring is generally preferred because it has no latency, but not always. I have had quite a bit of trouble recording with the Alesis iO dock and the Meteor recorder app. The input often gets distorted or filled with digital noise. To be safe, it may be better to avoid direct monitoring so that you are hearing what’s actually being recorded, not what you hope is being recorded. I had to redo a number of tracks because I used direct monitoring and they didn’t get recorded properly.

Another problem with direct monitoring, at least on the iO dock, is that source is only in one ear. I find that tracking is best done with mono monitoring so you hear the same thing in both ears, and you can more easily notice phasing problems.

Some devices such as the Samson Meteor mic can only support direct monitoring as there is no way to turn off the mic. That’s generally not a problem since you’d expect to always want to do direct monitoring with a mic. But note the potential issue with recording problems noted above. I’ve never see those problems with the Meteor mic though. So maybe its only a problem with the iO dock.

Conclusion

This was an long post, and there are many more options I didn’t cover. But this should be enough to get you started, and to avoid some of the pitfalls I’ve experienced in getting sound into iOS devices. It is truly an exciting time when we can have a full-featured multi-track recording device in our mobile phone.

Now that we can get sound into GarageBand, the next post will cover creating projects and doing the song layout in preparation for recoding that sound.