You’re going to see one of two pictures of this thing if you search for it on the internet.  There’s this version (and its variants):

And then there’s this version and (its variants):

I wish that the Icicle felt as solid and sturdy as that first photo.  I mean, damn, look at that thing.  The chassis is brushed metal with rubber along certain edges for protection.  There’s even a bit of ribbed rubber where the USB cord meets the main housing to keep the wire from kinking.  Unfortunately, the second, plastic Icicle is the true Icicle.  Plastic except for the XLR contacts.   If you were to put this in your back pocket, forget about it, and sit down, the chassis would break or crack.

To its benefit, I’ve hauled this device in a backpack along with a lot of other equipment for months and it held together.  But I fear that, someday, it will get jostled, get pinched or caught on something, and break.  Get a case for it, or put it in its own padded pocket.  It will not take a beating.

Function

Getting the Icicle to work on your computer is a snap.  No drivers necessary, just plug it in and go.  How much gain you apply via the gain knob on the side of the housing is your only option on the entire device.  That’s all you really need, and I appreciate that kind of simplicity.  But underneath that simplicity lie three major drawbacks.

First, it does not work well with dynamic microphones (a microphone that does not make use of phantom power) even though Blue insists that it works just fine in all of their promotional material:

The Icicle works with both dynamic and condenser microphones, providing high quality and hassle-free connectivity with Mac or PC.

- bluemic.com

No. No, it does not.

I tried the Icicle with my Shure SM58 and my Shure SM57 on my Dell desktop, my Gateway laptop, and my Dell laptop and I could clearly hear little blips, bloops, and whines in the background.  Electronic noise generated by the hard drive and other components in the computer was leaking onto the recording.  I consider this a functional failure; the noise is loud enough to be a dealbreaker.

Perhaps a dynamic mic that’s not an SM57 or SM58 wouldn’t cause this error, but when is the last time you used a dynamic mic that wasn’t an SM57 or SM58?  Condenser mics, or any other mic that requires phantom power, did not produce this noise.

Second, monitored input lags behind playback if you play and record at the same time.  When plugged into a PC the Icicle makes use of DirectX drivers which are not cabable of low latencies.  If you’re just recording a spoken podcast, that kind of latency might not be a problem; keep an eye on your signal to be sure you aren’t clipping and you’ll be fine.  But, if you need to record some vocals or guitar over the drum parts you’ve already recorded, the lag is going to make it impossible.  Well, not impossible. . . You could, I suppose, turn off any monitoring from your Icicle, sing/play against your pre-recorded tracks, and then sync your recorded vocal or guitar manually by dragging it around in your DAW.  That’s going to get really tedious really fast.  God help you if you need to punch in/punch out.

Some of you may be thinking “Hey, why not ditch those DX drivers and use ASIO4ALL?”  Normally, that would be an awesome idea.  But the Icicle makes use of your computer’s output instead of housing its own.  ASIO4ALL can’t use two devices (your soundcard and the Icicle) at the same time*.  You can try it, but you’ll probably get a BSOD.

NOTE: Mac users, you may not have to put up with any of this latency nonsense.  I don’t know for sure, but from what I understand, Apple Core Audio can build a single, low-latency device out of the icicle and your onboard soundcard.  If you’ve had experience with this, please comment.

Third, and this is a minor gripe, the usb connection is a ‘USB mini B’ rather than the standard ‘USB B’ connection.  Not a big deal, but if your cord goes bad or goes missing, you won’t be able to replace it with whatever you’ve got lying around.

Sound

Provided you aren’t using a dynamic microphone, the Icicle is on par with any other entry level preamp.

Features

Other than the connectivity between a usb port and your (phantom powered) microphone the only feature is the bluish glow it gives off when you plug it in.

That’s nice, I guess.

Verdict

I used this device as a workaround for recording the voice of someone I was filming.  The camera’s on-board mic/preamplifier was no good and I did not want to drag my heavier interface back and forth to the shoot.  The Icicle was perfect because I did not need to monitor or sync their delivery against a prior performance.  It would also be useful to record a podcast or something analogous.  But, if you’re trying to make music, the latency issues will be a major hindrance the first time you try play and record at the same time.  Low durability may be an issue regardless of how you use it.  And if you’re using a dynamic mic, the noise the Icicle generates is unacceptable.

I advise most to pass on this device.  The Icicle is only suitable for certain specific tasks.

*Technically, you can use two devices at the same time with ASIO, but not with the same program.  You’d need two hosts or DAWs, each using their own device.  This isn’t a workaround because your recorded audio would be in another program, and wouldn’t be synced anyway.  Applications for that kind of thing are limited.

Metal body with sturdy plastic knobs.  If you were to drop it from over your head, it would probably survive the fall and work fine.

Function

Idiot proof.  If the connection fits, that’s probably where it goes.  Turn the knob to the right for more signal.  If you need phantom power, press the button marked phantom power.  The market is saturated with devices that depend on software or a menu of some kind to gain access to hundreds of parameters, most of which you’ll never use.   But the AudioBuddy will not subject you to that kind of thing.  It’s just a two channel pre-amp, and it’s proud to serve it’s purpose.

Sound

I used the AudioBuddy to make up for a couple failing pre-amps on my mixer (Alesis MultiMix12).  I was wary of the purchase because I’d read some fairly nasty reviews about the sound quality.  If you’ve heard the same, feel free to disregard that opinion: this device is as clean as any entry level pre-amp and is free of any noise or ground loops.  Everything I recorded through the AudioBuddy sounded just fine.

Features

So, this thing says “Dual Mic Preamp /Direct Box” right on the chassis, and rightly so.  It can take an unbalanced, high-impedance signal (like a one directly from an electric guitar) and convert it to a low-impedance, balanced signal (like one that your mixer might want).  If you need a direct box in order to attach an electric guitar or bass directly to your mixer, then this will work for you as a direct box.  If, on the other hand, you need a direct box to take a line level signal (like the 1/4-line outs on your mixer or keyboard) and and attenuate it to be reamplified or to connect it to an XLR input on another mixer, this device will not do that.  This is not the direct box you are looking for.  That’s not a dig at this device, I’m only acknowledging that many people are looking for a direct box to take an unbalanced line level signal and attenuate it (reduce the decibel level) for re-amplification in addition to converting it to a balanced signal.  Again, this device won’t do that.  Snag yourself a couple of these for that purpose.

Verdict

This is a solid device that serves it’s purpose and will not let you down.  Buy one.

And all this stuff, as wonderful as it is, is causing a crisis of conscience:  Someone, somewhere told you that you should never mix anything from the stage.  ’You can’t hear things properly from that perspective’ they said.  ’Let the pro do their job,’ they said.

But as you look at your plethora of outputs you can’t help but feel as though you’re asking a lot, maybe too much, of the man or woman behind the mixing desk.  So, should you stay true to the ideal of using and objective audio engineer and let them mix everything from their perspective off the stage OR do you solve what you’re fairly sure will be a logistical problem of too many inputs by bringing your own mixer, thereby committing the sin of mixing from the stage.

Short Answer: If you’ve got two or more electronic instruments,  definitely bring something to mix thems onstage and feed the soundguy a single ‘keys’ stereo signal to mix along with the vox/bass/drums/guitar. Don’t try to run your whole band through that mixer, just the electronics the soundguy probably doesn’t have room for.

Long Answer: Ideally, the audio engineer working for the venue has both the knowledge and the equipment to take whatever you give them and mix it from an ideal listening position. So why attempt to mix it for them from the stage (where it is difficult to get an idea of how things really sound)?

1. They do not have the knowledge.

I’ve met a lot of amazing audio engineers who were nothing but connsumate professionals.  But the person mixing your band may not be one of these shining beacons of professionalism and experience.  The person mixing your band is the promoter’s friend who agreed to do it.  They are the bar’s assistant manager who is the only one remotely familiar with the equipment.  The venue may or may not be paying this individual at all.

Why would they pass off something so important to someone so inexperienced?  Sadly, the venue does not really know or care about how good things sound. As long as there are people paying admission and buying drinks, they’d never dream of forking over any extra money to a soundperson that might make things sound amazing, or fire one that fails to do a stellar job.

2. They do not have the equipment.

Unless your venue is the Hammerstein Ballroom, or Webster Hall,* you are playing in a small club with a small sound system. Probably built by the manager’s friend, or that staff member who happens to be in a band and is only marginally familiar with what’s needed. Even if you’ve got a real pro behind the mixer, they only have a 12 or 24 channels and many of those channels are already occupied by previous setups: the mic’ed drumkit, the radio they pipe in, an iPod they keep connected during the daytime, the TV they connect during the daytime, etc, etc. While your venue is probably prepared for a basic setup of vox-guitar-bass-drums, they might not have any extra room.

Of course, they could, in theory, just disconnect these things and plug you in, but consider that they won’t be comfortable doing so. To those of us with experience in running a mixer, plugging and unplugging superfluous equipment is no big deal, but for most human beings, a mixer is something you set up once (hopefully you get your friend to do it) and then NEVER TOUCH AGAIN, lest you anger the fader gods and, lo, bring much clipping and feedback upon the ears of he who would commit such sacrilege.

Even if the person behind the mixing desk has extra channels and the will to hook them up, he may not have enough cables to do so . . . if he has the cables, he may not have the DI’s necessary to carry the signal from the stage to his desk without ruining it.

And EVEN IF your venue has all the equipment . . . .

3. They do not have the time.

Whatever he disconnects, he will need to reconnect.  Whatever he connects, he will need to disconnect.  That takes time, and you’ve probably only got a few minutes.  The venue will not undo any of their static connections (the iPod, the TV, etc) if it means they’ll be there 45 minutes after the show reconnecting them.

4. YOU do not have the time.

Also, you will probably not get much of a soundcheck. You will load in your gear an hour early, set up your gear as the band before you tears theirs down, and you’ll get 3 min or so of a soundcheck before you actually perform. The venue may be willing to humor any request, but they also want the next band to go on at 10pm sharp.  Any time you waste will take time away from your set.

Thus . . . it’s probably a good idea to bring a small but workable mixer for your electronics (and a couple DI’s) and hand the soundguy a stereo mix. It’s true that you won’t be able to monitor exactly how things sound on the floor, but audio isn’t magic. You can walk out once, check the sound, walk back, and do your best to keep things consistent with what you know about your equipment. It may not be perfect, but it will be in the ballpark, and since in the ballpark is all that’s really available to you, well, you’ll have to learn how to deal with it. And BTW, the venue will certainly not be upset that you’ve decided to make their job easier by summing your multiple electronic inputs with a mixer.  They’ll simply recognize that you’re proactively solving what you know might be a problem.  They’ll do their best.

Submixing isn’t ideal, it’s the only practical solution for 95% of bands in 95% of venues.

*If you find that you are, in fact, playing Webster Hall, or a stadium, or whatever, and you have access to adequate equipment and a competent professional and the time to take advantage of both . . . by all means . . . hand the pro every input and he’ll mix it like a champ. But for those of us who are playing small venues, venues where we must make a few compromises, don’t freak out about using a mixer to sum up your drum-machine, laptop, and keys 1 and 2.

1. bits of your subject around the edges get keyed out (you don’t want that)
2. bits of your green background around the edges of your subject are left in (you don’t want that either)

You sought help in various video edtiting and post production forums.  The advice you got probably centered on your lighting.  ”You have to have perfect lighting, and a perfectly smooth green background with no variation in shading or color whatsoever,” they told you.

Don’t worry, you don’t need to reshoot (with a different camera or different lights or a different green screen).  You just need to use the right tools. You’ve probably been keying in Premiere with Chroma Key or Color Key.  Maybe someone told you to try keying in After Effects and, upon trying, you discovered that their keying plugins were just as useless.

Unfortunately, those methods are always going to result in an awkward and unusable key.  I don’t really understand why Adobe insists on including them in their programs; Keylight is the plugin you want.   They openly admit that their native plugins aren’t very good compared to Keylight.  Seriously:

Though the color keying effects built into After Effects can be useful for some purposes, you should try keying with Keylight before attempting to use these built-in keying effects. Some keying effects—such as the Color Key effect and the Luma Key effect—have been superseded by more modern effects like Keylight.

- Adobe Online Help Manual

Adobe includes Keylight with every purchase of their creative suite, but does not automatically install the plugin.  You’ll have to dig up your installation disc and find the ‘third party content’ either on its own disc or as its own option in the installation menu.  Tell the Adobe installation dialog to install only Keylight, reboot your machine, and you’ll be in business.

Open After Effects and begin a new compositon with parameters that match the footage you’d like to key.  Find Keylight in the effects window and drag it to the footage in the monitor tab or to the footage timeline in the timeline tab.  Select the effects tab to see Keylight’s options.

On the surface, Keylight works the same way any other keying plugin works: you tell it which color it is supposed to make transparent and it does it’s best to remove that color from your footage.  So first, tell Keylight which color you’d like to extract. Select the dropper tool and find and click on the green (or blue) area behind the subject.  Most of the background should immediately disappear, but the edges of the subject won’t be quite right (that’s normal).

Now select the ‘Status’ view.  Your footage will become a black and white representation of what is or is not transparent based on the key color you’ve selected.  The color white represents parts of the picture that will remain, black represents parts of the picture that will be taken out, and grey represents parts of the picture that will be partially transparent.  This is called an alpha layer.

Based on the status view above, you can tell that the footage won’t look good.  The gray in the background tells you that parts of your green screen will still be visible, while the grey in the foreground tells you that the subject will be partially transparent.  Your goal is a white subject and a black background.  Increasing the ‘Screen Gain’ will simulate a more saturated picture, and so bring more of the background into the keyable range.  A small increase, around 110, is best.  Too much and you will distort your subject.

Increase ‘Clip Black’ until black covers the entire background.  Too little and some of the background will remain; too much and it will remove some of your subject.   Decrease ‘Clip White’ until your status view shows a completely white subject.  The values you end up with will depend entirely on the nature of your footage.  The only guideline I can offer is that your clip black value can’t be greater than your clip white value (otherwise you will remove your subject and leave in your green screen).  You’ll need to adjust both until the border between white and black matches the border between your background and subject.  Your goal is a white subject against a black background, with a tiny amount of gray at edge.

These alpha layer adjustments are why Keylight works so well. Most keying plugins focus only on the key color and colors similar to it.  This is a problem because a plugin that uses this method will always remove bits of green on your subject near the edges of the key, and will always fail to remove bits of your background that aren’t similar enough to your specified color.  Keylight, on the other hand, only uses the color you’ve selected as a baseline to create an adjustable alpha layer.  By using the alpha layer, Keylight ignores these problem areas and predicts and edge based on similar and dissimilar colors.  Because it is only a prediction the alpha layer will leave in the bits of green that are obviously part of the subject, and remove the non-green bits that are obviously part of the background.

If you find the border between the white and black areas of the status screen are a little unnatural, adjust the ‘Screen Pre-blur’ to around 0.1 or more.  ’Pre-blur’ will not blur your footage, but it will make keylight behave as though it was.  This will smooth out your key along the edges.

Some of the light reflected off the green screen will spill over onto the subject.  Keylight intelligently detects this spillage and replaces the green with a neutral gray (or whatever color you specify . . . but grey will work most of the time).  If this color replacement creates a dark shadow near the edge of the subject (and it likely will), change the ‘Replace Method’ to ‘Hard Colour’.

Switch the view back to ‘Final Result’ to check that you have a solid key with clean edges.

To be absolutely certain of your key, find your background and place it in a timeline below the footage.

Keylight is capable of a lot more, but as long as your footage is marginally clean you may not need those additional features.

Now that the praise is out of the way . . .

A recent post of his – Rehearsing is for Cowards – got me thinking that, no, rehearsing is most decidedly not for cowards.

I think I understand what Seth is saying when he says you shouldn’t ‘rehearse,’ you should ‘explore’.  Well, sure.  Whenever you’re practicing your mind should be open to change and new ideas. During a live performance the spontaneous energy of your audience and the performers can push your show in a different but more engaging direction than you had originally intended.  Insisting that your performance and the performances of those around you be exactly the same every time is doing yourself a disservice.  But there’s somthing about his language (“repetition,” “regurgitation”) that I feel casts good old woodshedding in a negative way (which it doesn’t deserve).

Consider the following:

You went on that stage to play Rock Your Face Off, Pt. II: The Reckoning, but something a little different found it’s way out of your instrument.

So were you exploring, or did you just plain screw up?

The truth is that you and the bassist were totally out of sync, and the drummer had to change tempos to adjust.  The result was a rendition of RYFO2:TR with an extra beat in the second chorus.  Who enjoyed that?

Did you? No, you were freaking out, and the euphoric rock star you become on stage took a temporary back seat to the confused musician who knows it’s not going well and was trying to find a way back to the groove.

Did the Audience?  No, they know how RYFO2:TR goes, and they know there isn’t an extra beat in the second chorus.  For a while they were rocking out to the steady waves of energy emanating from your performance.  But your little exploration disturbed those waves enough that everyone’s dance steps and head-bobs became erratic and stalled.  Rocking out recommenced only after you and the band were back in familiar territory.

It was an accident, no one enjoyed it (not even you), and it degraded your performance.  In this situation I think Seth’s advice is a little dangerous.  It would be easy to tell yourself that you and the band were just ‘seeking the hitches’ and that extra beat was a ‘chasm that helped you leap forward’, but I don’t think any of those phrases would be accurate.

I think it’s fair to say you screwed up.

I’m not sure if Mr. Godin has ever played an instrument or performed in any musical capacity, but if he had, I don’t think he would look so far down at “the repetition of doing it before you do it.”  That’s exactly what we musicians require.  We musicians have to deal with muscle memory.  On top of having new ideas, we need our muscles to execute those new ideas.  But sometimes, our muscles just don’t like new ideas.  When that happens, the only option for you is repetition.  Otherwise, your new idea stays entirely cerebral and never becomes a physical reality.

Music is Not Magic

In the music community, there exists the idea that musical ability is, to some degree, magic.  The idea that mastery of your instrument depends not on study or technical skill but getting in touch with ‘The Music”.  Only those that can attain some spiritual link with ‘The Music’ can succeed.  Can’t get through that tough 16th note passage?  Well, you’re not in touch with the magic.  You must not love it enough.

Sorry, but that’s not how it works.  Sometimes there is nothing to do but grind through something – again and again – in exactly the same way  - until it feels natural.  Then exploration will make sense, will feel right.  Improvising around something you don’t have a handle on won’t get you anywhere you want to go.

So why do I care?  Well, I think this idea of music as magic is destructive.  I’ve seen a lot of musicians sabotage themselves by trying to ‘get in touch with the magic’ instead of practicing.  I’ve also seen a few teachers urge their students to ‘feel the music’ when they came up against a lack of technical skill in their students.  Maybe they just didn’t know how to put into words what the student needed to do next, or maybe they just lacked the patience to put it into words, but the result is the same: they’ve given the student the impression that they are permanently defective rather than just under-practiced and inexperienced.

I’ve also heard a lot of awful music made from bands that did not bother to do anything other than what was immediately available to them technically.  They mean well, setting out to ‘explore’ and ‘break rules.’  But in failing to create a structure or idea that would force them into new territory they just keep churning out what’s comfortable.  Paradoxically, their attempt to challenge themselves and their audience results in music that is safe and boring.

Of course, there are some things that can’t be rehearsed.  Releasing your band’s first EP or starting your first independent theater company are things you’ll never feel completely prepared for.  You can try to run the gauntlet ahead of time in your head, but you’re wasting your time.  You’ll reinforce what you imagine might happen, leaving you relatively unprepared for – and stealing time away from dealing with – what actually happens.  In these situations, you’re probably just treading water out of fear.

Exploration and experimentation should be part of every creative process.  But sometimes, you just have to grind it out.  Call it ‘repetition’ or ‘regurgitation’ or whatever ulgy name you want.  Just don’t lie to yourself about what your audience wants or what you’ll have to do to deliver.

Also, in case you did not get ahold of it last time, here’s every program you’ll need in one convenient package:
Launch-Mono Starter Kit.zip

Intro to Polygome

First, plug in your Launchpad and launch nonome. Depending on what you were doing last time you used a Max5 patch, nonome might try to access ASIO4ALL, even though it does not need any audio outputs.  Why? Max5 Runtime will try to use the same audio device it used last time you opened a Max patch, so even if your current device (nonome) does not require access to the soundcard, Max5 will claim it anyway. So check your icons in the lower righthand corner of your screen.

If you see the ASIO4ALL play or pause symbols, Max5 is trying to claim your soundcard. There’s no way to make it stop from the nonome or Max5 menu, but if you open a program that includes dsp settings you can tell Max5 to stop trying to use your soundcard. Click on Launch-Mono Starter Kit\mlr_Aes mdb 0.42\_mlr.maxpat to open mlr. Go to options -> dsp_options and in the driver field select None.

Close mlr and close nonome. When you reopen nonome, you shouldn’t see any ASIO4ALL icons in your dock. If you do, review the above and try again.

Once you’ve sorted out your audio drivers, dig into your the Starter Kit and run Launch-Mono Starter Kit\polygome64v98\_polygome64.maxpat. When the app pops up click on the sys\prefix\gome button.

This sets your nonome to the \gome prefix and ensures that these two applications talk to each other. The OSC prefix in nonome should now read /gome.

This is typical of almost every monome app you’ll use. Every app has its prefix, and setting nonome’s OSC prefix to match enables communication between the two. Many apps have a button like polygome that sets the prefix automatically, but some require that you edit the OSC prefix in nonome yourself.

Also, be sure to select Launchpad in the Select Launchpad dropdown menu in nonome.  This is especially important for using Polygome.  If you route nonome to the Microsoft Synthesizer or other device, you may produce unwanted MIDI data.

Switch back to Polygome and set the MIDI Output to Microsoft Synthesizer if it isn’t already.

Now, grab your monome and start pressing buttons. If you’ve done everything correctly you should hear a piano arpeggio. Congratulations! You’ve got Polygome working. Polygome does not make any noise by itself, it only sends MIDI data. If you’ve followed the instrucitons above, you’ve routed that MIDI data to your soundcard’s default MIDI bank. The piano arpeggio is the result of successful communication between the two.

Installing MIDI Yoke

Now that Polygome and your Launchpad are working together, the next step is integrating it into your VST host (Ableton, FL Studio, Cubase, Reaper, etc.).  This is how you will use Polygome to run your favorite softsynth.

MIDI Yoke will create several virtual MIDI channels you can use to route MIDI data from one program to the next:  from the Polygome to your VST host.  Double click MidiYokeSetup.msi in the starter kit and click through the installation dialog.  When the programs asks you to reboot, do so.

Integrating Polygome with your VSTi Host

Upon reboot, fire up nonome and Polygome as per the Intro to Polygome part of this tutorial.  But this time when you select MIDI output on the Polygome you should see MIDI Yoke channels in the dropdown menu.  Instead of the Microsoft Synthesizer, Select Out to MIDI Yoke: 1 – Channel 1.

On top of sending data, Polygome will need to receive data from your host in order to stay synced to your host’s tempo.  To enable this, choose beat clock under source in the clock options.  Uncheck the internal checkbox, and select In From MIDI Yoke 2 in the recieve beat clock from dropdown.  Later, you will set up your VST host to send clock data to this Yoke channel.

Note that if you’ve set up Polygome correctly, you won’t hear anything.  You’ll have to set up your host to make any noise.  This next section will use Reaper as the host, so the exact instructions for the host of your choice might be a little different, but the principles are the same.  You may need to consult your host’s manual to find and adjust for differences your setup procedure.

Open Reaper and start a new project.  Create a new track (Ctrl-T) and put your favorite VSTi in the FX slot.  To gain access to the new MIDI Yoke MIDI devices, you may have to adjust your host’s preferences.  If you’re using Reaper, go to Options -> Preferences and choose MIDI Devices in the lefthand menu.  To allow Reaper to accept data from the Polygome, right-click In From MIDI Yoke 1 and select Enable input.

Next, tell Reaper to transmit it’s tempo to Polygome by right-clicking Out to MIDI Yoke: 2 and select Configure input. . . When the Configure MIDI Output dialog box shows up, check Enable output to this device and Send clock/SPP to this device.

Reaper and Polygome are now comunicating with each other in general.  All that remains is to route Polygome to the track containing the VSTi you’d like to run.  On that track, arm for recording and enable monitoring by clicking the little ar button to the left of the fader and the button with the speaker icon to the right of the meters.

You should now see a dropdown menu appear over the meters.  Left-click on it and select MIDI Input -> In From MIDI Yoke: 1 -> Channel 1.

Everything is now connected and properly routed, but Polygome will not produce MIDI data for your VST until it gets a tempo from Reaper.  Press Play in Reaper to begin transmission of this tempo.

Congratulations! If everything has gone well, you how have Polygome hooked up to your host, and routed to your favorite VSTi.  Like mlr (covered in Part 1), there is a lot more to Polygome than I’ve covered here.  You’ll have to consult the documentation and experiment with it to gain full access to everything it can do, but this should be enough to get you started.

This thing really is worth the $200.

Yes, that’s me, and this is a promotional video I did for Ex Machina.  We had a heck of a time finding a drummer, and I finally said ‘Hey, we’re an industrial band, who needs a drummer with technology like this available.’  Check out the details on exmachinaonline.com

. . . and you are totally underwhelmed.

The clip launching is nice but that’s not what you got it for, is it?  You want to slice clips, make drones, and step sequence monome style.

Well, you can.  And, contrary to the instructions that come with the Launchpad would have you believe, you don’t have to spend a dime on Max4Live, or a full version of Ableton Live, or a full version of Automap.  Everything you need to geek out with your grid is available for free.  And I’ve collected it all in one convenient place.

Initial Setup

Download and unzip Launch-Mono Starter Kit.zip

First, you’ll need MaxMSP Runtime.  MaxMSP is a program that allows users to build their own virtual sound devices and integrate them with their equipment.  The Runtime version is the free version that’s limited to opening patches from other people; you won’t be able to build your own.  But a lot of the prebuilt tools are going to do exactly what you want and more, so building your own would be about as much fun as building your own car and make about as much sense.

Double click on Max5_RT.msi and follow the install instructions.

edit: Mac Users – The msi in the zip file is for PCs, but you can get a Mac version of Max Runtime here: http://cycling74.com/downloads/

Now that you’ve got the framework to run max patches, you’ll need nonome.  Nonome does not do anything by itself.  Its only job is to fool other applications into thinking your launchpad is a monome.

Make sure your Launchpad is connected (and that you’ve installed the proper drivers) and find _nonome-1.24a.maxpat in the nonome 1.24a folder and double click  (Mac Users – Go ahead, max patches work on Mac, too).  If Windows tells you it ‘cannot open this file’, click ‘Select this program from a list‘ and choose Max5.  Soon you’ll see nonome pop up.  Set the ‘Select Launchpad‘ dropdown menu to Launchpad.  Press the ‘User 2‘ button on your Launchpad twice.  You should see the arrow keys across the top light up in 4 different colors.  If you don’t, Launchpad and nonome are not communicating properly and you should start over.  If they do light up as described, pick your favorite color of the 4 and press ‘User 2‘ again.

If, you have an audio interface you plan on using, set it up now.  If you don’t have an audio interface, or you don’t know what I’m talking about, get back in the starter kit and install ASIO4ALL (ASIO4ALL_2_9_English.exe).  edit: Mac Users – don’t worry about asio4all, your computer already has Apple Core Audio and that’s capable of the low latencies you need.  When you are instructed to use or select asio4all later in the tutorial, select Apple Core Audio instead.

MLR – The Beat Shuffling Program

Now you’re ready for mlr.  It lets you do this:

With nonome open, find Launch-Mono Starter Kit\mlr_Aes mdb 0.42\_mlr.maxpat and open it.  Click on ‘Options‘ and then ‘dsp options‘ and select ‘ad_asio ASIO4ALL v2.’  Mac Users should select Apple Core Audio.

Now take mlr loop 120 1.wav and drag it to the area marked ‘audio file drop‘.  Then, go to the first file selection drop-down menu and select ‘mlr loop 120 1.wav‘.  Set the tempo to 120 and click on the button labled ‘dac‘.  If everything is set up correctly, the loop should be playing.

Once you’ve got mlr running, the actual beat shuffling is very intuitive.  The lighted button tells you which part of the loop is currently playing.  Press one of the 8 buttons on the second row (pay no attention to the first row for now) to start the loop from that point.  If the loop sounds off pitch or too slow, change the value in the ‘speed‘ box to 1 (both the loop and mlr’s tempo are set to 120, so it should need no adjustment to fit properly).

Before I leave you on your own, we should go over the groups function (the squares to the left of the ‘audio file drop‘).  No audio from the same group can fire at the same time, so starting a loop in group 1 will stop any other loop in that group.  To experience the magic, drag mlr loop 120 2.wav to the audio file drop and select it from the second drop down menu.  Pressing a button in the third row will start it off.  Notice that loop 1 stopped.  Pressing a button in the second row will stop loop 2 and start loop 1 again.

To play both of them at the same time, select the second square in the ‘groups’ category of the second loop.  Pressing buttons in rows two and three now causes a wonderful racket.  You might be wondering how to get it to stop, which brings us back to that first row.

That first row is just a control to switch each of the groups off.  A lit button indicates an activated group, the first for loop 1, the second for loop 2, etc.  Hit the corresponding button to stop any audio in that group.

Mlr is capable of a lot more, and now that you are familiar with the program, understanding the manual and experimenting on your own should be easier.

Next up: Polygome

So, do you need one?

Part I – Technical Aspects

Keyboards, laptops, samplers, and iPods all deliver what’s called a ‘line level’ signal.  This type of signal works fine with most consumer equipment and in most consumer applications.  But there are a couple unique attributes of a line level signal that can cause problems with your live performance.

Line level signals are not shielded from random electrical noise and interference.  If you’re just plugging in some headphones, that’s not a problem.  The length of the cable is, likely, not long enough to actually pick up any significant interference.  But running across a barroom floor or friend’s garage is a different story.  If you’re far away from your mixer, unwanted electrical interference can leak into the poorly shielded signal coming out of the line level device.

To prevent this, a direct box converts that poorly protected method of connection (an ‘unbalanced’ signal with only two conductors per channel) to a well protected method of connection (an XLR or other ‘balanced’ cable’ with 3 conductors ).  That third conductor is what’s keeping out the unwanted noise.

Even if distance isn’t an issue, if the mixer you are connecting to does not have any available ‘line-ins’ you still may need a direct box. You could, in theory, use an adapter to split your 1/8″ headphone or line out directly to two XLR (or microphone) connections. The problem with this is that you’ll probably be giving the XLR inputs on the mixer a signal that is far to hot to handle. Most XLR inputs on a mixer are routed directly to preamps that amplify the relatively weak signal from a microphone.  When you ram an already amplified signal through it, it will clip and distort. To solve this problem, a direct box (in addition to the protecting the signal from noise) will also convert a loud signal to a quiet one ready for amplification.

So, for rehearsing with your friends – no direct box is likely necessary. You’re probably not far from the mixer, and that mixer probably has a line in ready for the nice hot output from your laptop.

But for a live gig where you’re connecting to someone else’s (a bar’s) sound equipment, none of those things may be true. You might be some distance from the mixing board, or that mixing board may not have enough or any line ins. . . . .

Part II – Practical Application [A Question and Answer Session]

Q: “But Phil, the mixer’s not that far away. Plus, I looked right at that mixer saw that it definitely has an available line in. Several of them, in fact. And it doesn’t make sense, how can a club or bar have a mixer with no line ins? I’ve looked at hundreds of mixers and they all had, at least, a couple. Is it even possible to buy a mixer without that capability?

A: Indeed, the line ins you seek are ubiquitous and robust.

But the sound guy has to know what they are and want to use them.   Perhaps your sound guy predates the widespread use of electronic instruments that might make use of a line input, and, thus, does not really know what they are for. Or the club itself might not have an unbalanced cable of any length whatsoever. Or both. Either way, that without an XLR connection just won’t get connected.

Q: “OK, in that case, GREAT NEWS, I called the venue and they said that they have some Direct Boxes I could use for just such an occasion.

A:  . . . . they LIE!

When they say they have Direct Boxes, they likely mean that they have one mono direct box (whereas you will need two). In the event that they have two, they will not be of the same make and manufacture, and they will have wild discrepancies in the way that they attenuate either channel of the stereo signal.  When the venue tells you they have a guitar amp you can use, you can probably count on the fact that it works because it gets used every time they have a show.  If something goes wrong, or someone manages to steal it, they notice right away and they do something about it.  Direct Boxes, on the other hand, are not something that get used often. So when they go missing they aren’t missed (and then replaced), and when they stop working or break they aren’t fixed.  It’s best to rely on your own.

In short:

Starting out: probably no need for DIs.

Gigging regularly: definitely find a couple inexpensive DIs.  You don’t need to spend $200 to get the benefit of a DI.  $30 per channel should do the trick.

I have too many plugins.

A couple years ago, when freeware plugin creation became very popular, I developed a very bad habit.  When I saw a plugin that I liked, or thought might work in a new and interesting way, I would immediately download it.  Then I would usually find every other plugin by that creator and download it, also.  Then I would separate the effects from the instruments and organize them by creator.  And then I told myself that I would use it later.

Eventually, later would arrive when I needed a taste of something new.  But when I reached into my stockpile, I would see this:

. . . and there’s no way to stay informed about that many plugins.  So when I choose, I choose at random.

That might not seem like a problem until you consider that softsynths aren’t magic programs that can be operated on 100% intuition; they all have a learning curve.  So, in effect, whenever I wanted something creatively new or different, I would have to bring myself up to speed on a brand new plugin, but (because there were always more to choose from) I never spent enough time with any one of them to get completely familiar.

Well, that’s an enormous damper on my creative energy, and I’m not going to take it anymore, so last week I purged my library to only that which I’ve used more than a few times.  When I finally hit delete, I felt an enormous sense of relief.

I write this as a warning for those who’ve yet to develop a plugin addiction and a wake-up call to those who have:  Free things are great, but these aren’t consumables, they’re tools, and if you have too many you won’t be able to use them effectively.