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January 21, 2009

Filed under: music»tools»looping

Requiem for a DL4

The Line 6 DL4 delay modeler occupies an enviable position on the pedalboards of a lot of musicians, both famous and not. It's considered by many to be not just one of the best delays available, but also one of the best loopers for the money. At the same time, it has a well-deserved reputation for unreliability. Even though it's been a decade since the pedal was first introduced, Line 6 hasn't ever produced a hardware revision to fix those issues. Why would they? The combination of the two means that a lot of people have bought DL4s several times over.

I am, in general, a fan of Line 6. Their modeling tech is impressive, and it's relatively affordable. More importantly, I usually find that they understand the ways musicians will use their products more than a lot of prosumer music companies do. So it's frustrating when an issue like the DL4's fragility goes uncorrected for so long--and it's really frustrating when the flaky DL4 in question is mine. I'm not keen on paying for another faulty unit (or driving it up to Maryland for a costly repair job), but it's hard to find anything that can replace the Line 6 at anywhere close to the same price.

Don't get me wrong, there are lots of loopers under $500 that beat the DL4 on a spec sheet. Boss has a whole set of pedals with longer sample lengths, or the ability to save samples for later. Its largest model can play multiple loops simultaneiously. Digitech's JamMan even records its audio to CompactFlash cards, so you can load arbitrary sounds from a computer. Either one of these, just to take two examples, is also much more durable than the DL4.

But neither of them is anywhere as usable for real-time looping. On both, for example, there's no way to go directly from the first recording pass into an overdub pass. The JamMan requires a double-stomp to enter overdub mode, which is much harder to time correctly, and the Boss pedals typically require playback to begin before overdubbing (the exception is the RC-50 pedal, which is A) enormous and B) only allows users to replace the default behavior, not make a decision at runtime). The great thing about the DL4 is that you don't have to make a choice--you can start with an overdub right away, or you can go straight into loop playback.

The same kind of thinking--or lack thereof, for other pedals--shows up in the one-shot feature. On other loopers, the option to play a loop just once (like a triggered sample) is usually set via a small switch on the actual unit, and it can't be changed while the loop is playing. In contrast, the DL4 has a switch set aside just for arming "Play Once" mode. At first, this seems like a waste of hardware, but don't be fooled: that switch is tremendously useful. It's not just good for the obvious applications of trigger-and-forget sample playback. You can also use it to tell the looper that you want playback halted automatically, freeing up your feet to make changes on other pedals at the same time. And if you're really good, you can use its ability to restart loop playback for compositional tricks that aren't otherwise available.

In other words, what you get with the DL4 isn't the most fully-featured pedal. But the designers have made sure that every feature it does have is accessible with one step at all times. If you're playing a looper as an instrument, and not as a foot-mounted samplebank, that's arguably far more important. Any time you have to reach down to adjust an option, or let a sample play through in order to get a proper overdub going, you've broken the performance.

When the DL4 started acting up, losing power or freezing onstage, I went looking for alternatives. The only one I've found with any potential is the Eventide TimeFactor, which offers overdub/play flexibility out of the box. Even then, to get a decent one-shot mode you have to add an external switch--but it at least gives the impression that it was designed with real-world users in mind. This may seem like faint praise (and it is), but it's something I think you can only rarely say about effects boxes. Why this is, I'm not sure. I suspect that the margins are too thin, and the consumer data too sparse, to provide proper incentives at the lower end of the market.

Frankly, when it comes to pedals gone bad, there are few good choices. Repair in our throwaway culture is expensive and hard to find. Reuse is a nice idea, but unlikely in practice, since I don't really have the technical background or patience for DIY pedal-making. And with commodities this expensive, the cost for a replacement is a bitter pill even in the best of times. In a rough economy like this, it's even harder to swallow.

Still, I'm trying to become less prone to fix problems through additional consumption. So if the choices are bad, perhaps the point is to find new ones. I still have looping tools available to me--the Loop Junky or a Mobius-based laptop rig. And I'm hoping this will serve as a reminder of why I started playing with loops in the first place: not just that I'm a misanthrope, but as a set of constraints for channeling my creativity. I like the DL4, but I'd gotten comfortable with it. Its constraints weren't really limitations anymore. It's probably time to find new ones.

December 31, 2008

Filed under: music»recording»production

Don't Fear the ReaFIR

I don't really know how to say this, but: I forgot to get you anything for the holidays. I feel terrible, honestly. And after you got me such a lovely sweater.

(Belle! Take the sweater off the cat! She has some dignity to preserve!)

I'm sorry. Let me make it up to you. Here, have a peppermint stick and a quick tutorial on cheap noise reduction.

There are two cardinal sins of audio that I've committed, and which I've noticed in work by others, since it became easy to produce digital audio and video--by cardinal sins, I mean errors that make it instantly evident that this is not a professional production. The first is bad mike technique--having the microphone too far back, or too close, or using the wrong kind of microphone for the task at hand. The second is noise--noise from preamps, noise from wind and AC systems, or just the hum of a bad ambient environment.

The thing is, mike technique is hard. And you don't always have the option of great equipment, or the time to perfectly position it. You can't fix mike technique for free. And noise is also hard--I have noisy recordings all the time, because I use relatively dirty preamps with very quiet microphones, and I record in locations that aren't soundproofed (it is also likely that I'm simply not as good at this as I think I am). But constant and regular noise (such as that caused by a cheap preamp or a climate-control system) can be cleaned up (or at least, minimized), for free, after recording. And it gives us a chance to learn about DSP! Who doesn't love that?

Before going into the details of our signal processing, though, a disclaimer: sometimes simpler ways of dealing with noise are better. For example, rather than worry about filtering you could always just mask the noise with background music. Or you could use a noise gate, which would dip the volume when a person isn't talking. But I find that without music or something else to fill the spectrum, a gate can even make noise more noticable when the voice "pops" in from the silence. Besides, there are plenty of times when background music just doesn't match the desired mood, or when it's distracting. In this case, a slight amount of filtering combined with a gentle gate has produced very good results for me.

So let's say that we've got an interview recorded in a room with the AC fans running in the background, and on playback it doesn't sound great. What we're going to use to strip the white noise out of this audio clip is a Finite Impulse Response (FIR) filter. As might be obvious, this kind of filter is in contrast to an Infinite Impulse Response. Both work using the same basic principles, FIR just limits its scope a bit. Although the math for these filters quickly becomes complex, at its heart they rely on a very simple principle of weighted averages.

Remember that digital audio is represented as a series of numbers, each of which represents the value of a sample at a specific point of time. From sample to sample, sounds with high frequency content will show more change than those with little high frequency content, simply because the innate property of a high-frequency wave is its rapid change over time. So to filter out high frequencies, the easy approach is simply to generate a new wave, where each sample is the average of itself and the samples around it. That "smoothes out" the high frequency sounds, but leaves the low frequencies--which, after all, change much less from sample to sample--basically unaltered. Other kinds of EQ filters can be generated by altering the weights for each sample in the average.

What's really interesting about FIR is that you can combine it with a Fast Fourier Transform (also known as a FFT, which is a fascinating process for doing spectral analysis using math I don't completely understand) to determine the weighting for a desired filter curve. This is what the plugin we'll be using, ReaFIR, does to perform its noise reduction. Using the FFT analysis window, it takes a fingerprint of the noise we want to remove, and then sets up an filter to subtract that from the audio stream.

Let's see it in action, step by step:

click to view larger

  1. Add a ReaFIR instance to the track on which you want to perform noise reduction. Set the Mode pulldown to Subtract.
  2. Find a nice, long (1-4 seconds) of relative silence. We're going to use this to build the reduction fingerprint, so you want as pure a sample of noise by itself as possible. If there are any other sounds, they'll be incorporated into the fingerprint, and you may find yourself filtering out parts of the sound that you didn't want. This sounds really weird, and not usually in a good way.
  3. Check the "Automatically build noise profile" checkbox, and then using the DAW transport, play the clip you've picked for training. You should see yellow lines representing the frequency domain of the noise jumping across the display, with the red line (which represents the filter) fitting itself with the average of the yellow. Be sure to stop playback before you hit any voice or non-noise content. I often cut the noise out and move it to an isolated section at the end of the track, just in case I let it run too long by mistake.
  4. Now uncheck the "build noise profile" checkbox, and your filter is all set! If you play the track now, the noise should be magically gone, even during other sounds. You'll also probably hear a few artifacts, the most common of which is a slight whistling in the high frequencies due to resonance in the filter bands. I usually find that you can apply a gentle lowpass filter and tame this until it's unnoticeable.
This is really just the simplest trick that you can pull with ReaFIR, although it's the function I use most often. Another neat feature is to apply it as a mastering EQ (making sure to switch the mode from "subtract" to "EQ) after using the FFT to grab a fingerprint from a CD or a piece of music--it'll "clone" the sound of that track for your own, which works well if they're in the same style. An analysis EQ like this is a very useful tool to have around.

Well, I'm glad we got this sorted out. I'm sure you'll agree it's much better than a fruitcake, which was my backup gift. And just think: now that you've got this under control, we can celebrate the next holiday with an in-depth discussion of convolution reverb, which is based on many of the same principles. Why, maybe we could even start now...

Oh, you have to go? So soon? Ah, that's a shame, but if you must...? Then you must. I understand. Have a safe trip, then. And happy new year!

December 22, 2008

Filed under: music»tools»digital

Review: Edirol R-09 Audio Recorder

A $300 handheld audio recorder is a hard sell to someone who isn't pretty crazy about sound gear. And frankly, for people who are audiophiles, it could still be a hard sell--why bother spending that kind of money on a single unit with built-in electret mikes, when it could buy a decent condenser and an SM57?

I bought the R-09 when I started at CQ, thinking that it would be useful for loaning to reporters. It's worked well for that--most journalists, when they buy a recorder, pick up one of the cheap voice/memo units, which sound awful and aren't terribly sensitive. For close-up work, that's fine, but it quickly becomes unusable in, say, a committee chamber. Reporters who borrowed the Edirol really appreciated its ability to capture quiet voices in poor acoustic spaces. One even bought his own after he found himself borrowing mine on a regular basis. It's also an easy device to use, which is important for non-technical people like the average journalist, and is sadly not true for all portable recorders these days.

Of course, the main goal for a unit like this--and the reason that you'd buy it over a set of separate microphones--is to use as a field recorder for quickly capturing sound without dragging a ton of equipment around. As such, a slight lack of fidelity is acceptable, since it's not like you'd be using a U47 to get a sample of street ambience anyway. That said, I don't have any complaints about the quality of the R-09's recordings. They seem relatively flat (EQ-wise) to me, the gain is adequate, there's not much self-noise or handling noise, and the stereo separation is surprisingly good. Recordings I did of union protesters on K Street a while back were clear and offered a great sense of space. More importantly, it's fast enough that I could capture something like a protest if I just ran into it--nothing to hook up, unless I want to plug a pair of headphones in to double-check the sound. Just pull it out of my bag, turn it on, hit record once to set the levels, then press it again to start recording.

That kind of convenience actually means that I've started using the R-09 for jobs where it probably wouldn't have been my first choice before. At CQ, unlike at the World Bank, I don't have the luxury of a studio with a selection of vocal mikes all set up and ready to go whenever I need them. While working on the debt explainer, I recorded Kerry with an ElectroVoice RE-20, which is a fantastic vocal mike, and myself with a Sony lapel mike. She sounded fine, I sounded awful, but I didn't want to drag all my recording gear down to the quiet room again to get a punch-in take. So instead, I carried the R-09 down to record a few lines, brought it back after each take, and imported the WAV files directly into Cubase. Because of the omni pattern, the resulting takes have a bit more room noise than Kerry does, but it holds up surprisingly well against the more expensive microphone, and the workflow was much more efficient and flexible.

Compared to the other prosumer field recorder I've used, the Marantz PMD-660, the Edirol unit is a lot smaller and a lot simpler. The Marantz was capable of doing rudimentary editing tasks, like tagging and file splitting, that the Edirol just doesn't do. It also boasted full-sized XLR jacks instead of the 3.5mm TRS minijack on the R-09, which was great if you wanted to use a external mike without an adapter, but if you didn't carry a mike, the internal microphones on the Marantz were genuinely terrible. So for certain applications--portable film recording, for example, or crewed radio production--the Marantz makes sense (and from its control layout, I suspect it's more aimed at those markets anyway). There's also a durability factor--the PMD-660 could probably be used to hammer nails between recording gigs. But for $300 less, the R-09 has been a really helpful tool that I'd recommend to anyone who either wants to step up from basic recorders, or needs to be able to capture sound at a moment's notice.

December 17, 2008

Filed under: music»tools»effects

The Other Kind of Loop

I enjoyed writing about the Loop Junky and DOD Envelope pedals from my collection last month. I think I'll keep doing it every now and then. I don't own any really amazing or exotic hardware, but I find that there's a shortage of decent commentary on non-boutique effects. Most of the discussion on them is at places like Harmony Central, a site that illustrates both the best and worst of crowd-sourcing, and could provide months of mockery (sample quote: "Even though i just tried it out in the store i can tell that it's not reliable." Someone get that man an engineering job!).

So if we're going to add to the Internet's collective wisdom on entry-level stompboxes, why not start with the one pedal that makes no noise on its own, that few people would consider, but should be one of the building blocks for any compulsive pedal-purchaser: the Boss LS-2.

The LS-2 is basically a foot-switchable, floor-mounted, battery-powered mixer. It's hard to understand the appeal of such a thing until you've used one--what's the point of a pedal that just connects other pedals? Leave the mixing to the sound guy! But the LS-2 isn't really just a mixer. It's a swiss army knife for musicians. It solves problems. And when you buy effects on a budget, you usually find yourself with problems that need solving.

Say you've got a pedal that's got a fantastic tone, but it creates a dip in volume: leave it running through the LS-2 with the mix gain up, and swap its loop in instead of activating it directly. Alternately, you've got a pedal that sounds great when cranked, but you want it to have that sound at unity gain: the LS-2 can act as a trim for it. Noisy pedals can be isolated in the same way. Your distortion (or other effect) doesn't have a wet/dry mix control: now it does. You want to be able to bring in multiple effects with one step, mixed together at arbitrary volumes: no problem, it can do that too. Multiple instruments into one amp (switched or mixed), selective multi-amp output, muting continous effects without halting them, or just quick volume presets, it's a versatile device. The one thing I haven't been able to figure out so far is how to get it to swap a pedal from one place to another in a signal chain, but it's probably possible somehow.

Of course, you can also use the LS-2 for its original purpose of routing signals from one place to another, and that's a valuable education in its own right. After all, having lots of effects is easy--it just requires too much money. But using them effectively means learning how to order them, and how to bring them in and out of the chain for a musical purpose. That's a lot harder to learn, yet it's a skill that's emphasized over and over again in audio. Unlike video or graphics production, which (in my opinion) stresses the composition of distinct elements as layers, audio is about basic operations processed in series and parallel, resulting in a final mix. Experiment with the LS-2, and you'll have a head start on thinking about production and synthesis.

My favorite routing use for the LS-2 is as pedalboard master controller. The ability to swap entire effects loops in and out basically gives you the instant complexity of multieffects patches combined with the flexibility and accessibility of individual stompboxes. For example, I used to run the MXR Bass DI+ and a chorus on the A loop, and a second distortion into an envelope on the B loop. By turning the pedals inside each loop on and off, and then switching between them, I could get from clean sounds to combinations instantly, or even switch between two very different signal chains, just like changing a patch--and yet, I still had all the knobs right in front of me for on-the-fly adjustments. It's the poor man's M13.

There are two things that are good to know when planning signal chains with the LS-2. First, the gain controls are on the return side of the chain, which is as it should be--otherwise it would be impossible to boost or cut volume-sensitive effects like distortion or envelope. Second, the send and return sides don't technically have to form a circuit: the active loop dictates where the input signal goes and which return is routed to the output, but they only have to connect if you want them to. The cases in which they don't are a relative minority, true, but they're useful edge cases regardless. Just off the top of my head, I can imagine using it to patch in a drum machine that lacks passthrough inputs, or to feed a recorder/house mix at an arbitrary point in the chain (say, before any reverb). Sure, there are better tools for doing that kind of thing, and if you have them handy, you should use them.

But if you don't... well, that's why you have the LS-2. It's the MacGyver of guitar pedals. It solves problems and makes useful stompboxes even more useful. And that's why it's one of the few items in my box o' gear that I would instantly replace if it were unavailable.

December 3, 2008

Filed under: music»performance»band

Riff Vs. Beat

Playing with other musicians for the first time, especially after noodling around solo for so long, has a lot in common with the process of culture shock. At first, there's a general wariness on the part of everyone involved, followed by a stretch of amusement and acclimation to musical quirks and customs, then acceptance, and finally (I often find) weariness and longing for familiar ground.

In other words, ad-hoc practice with a set of metal-heads didn't entirely end well. But it was illuminating.

Ultimately, experiencing other countries proves more of an education about your own, and that's certainly true musically. For example, my own context is rock- and blues-oriented, which means that I unconsciously base my playing and writing around the emphasis of certain beats in 4/4 time. I've gotten comfortable with that. But in a lot of harder metal and prog rock, primacy is given to the riff: a pattern of notes in a rhythm independent of the measure's basic beat, and which may even necessitate odd time signatures, like 5/4 or 9/8.

I don't know if this makes me a musical bigot, but I really hate odd time signatures. Four on the floor, baby.

But here's the other thing: when you come into contact with another musical culture, there are customs that end up being enforced. Just as intercultural communication, your role changes as you negotiate a common ground of understanding. And in my case, it ends up being constricted, which drives me crazy.

See, the job of the bass (or any instrument, really, but particularly bass) in rock music is circumscribed by the other band members. It exists in the space left over. If the guitarists insist on using distortion and heavy chords at all times, it cuts into the bass's ability to add effects or play in the upper register. Or, on the other hand, if the keyboard starts emphasizing left hand lines, now there's competition for the traditional bass role. You can try to fight for position, but I find that bass is at a disadvantage in these situations: it doesn't have the range of keys, or the volume of a cranked guitar rig (remember, higher frequencies require less energy to create at an equivalent loudness--and I hate volume wars anyway).

So the funny thing is, I got invited to jam because of my solo bass work, which crosses into a wide range of sonic territory, only to find myself relegated back to playing root notes on the clean preamp channel. Everything else got lost in the soup of distorted guitar crunch. I admit to being puzzled: this was not an unforeseeable outcome. So why invite me in the first place?

November 20, 2008

Filed under: music»business

Zune In

About three months ago, I bought a Zune. I wanted a way to find new music without buying a ton of CDs--and lately I've started to feel less comfortable with the ecological footprint from physical distribution, anyway. Of all the subscription services, Zune seemed to have the best deal. And while it's annoying that song burning is disabled, so far it's been a pretty good value for the money. I've been trying out all kinds of bands, new and old, and enabled my work computer as well for a kind of suped-up Internet radio. The hardware's not bad, either.

Today Microsoft turned on a feature that gives subscribers 10 free tracks each month from the big labels, including the large portions of the store in DRM-free MP3 format. As far as I'm aware, that basically makes it the best deal available for digital music on Windows, even if you don't own a Zune (the subscription tracks can still be played through WMA-capable apps, like Windows Media Player, if you don't care for the Zune application). It's got the advantages of an unlimited subscription, plus an album or so to keep each month. I'd still probably recommend Amazon's MP3 store for straight purchases, but if you're trying to branch out in your listening habits, you could do worse with $15 every month.

November 14, 2008

Filed under: music»tools»effects

Ode to the FX25B

While I'm rhapsodizing about unreliable analog effects, let's take a moment to reflect on the humble DOD FX25B Envelope Filter.

(I had to recolor this one to be green in order to match the physical pedal. For some reason, DOD's image is blue. This is not the worst sin committed by DOD's website--among other things, it sorts the pedal lines into categories like "Wrath of DOD" and "DOD is Love." Also, they have a Yngwie J. Malmsteen signature pedal, which I believe is actually considered a dangerous munition for the purposes of export.)

Put it this way: the FX25B lives up to DOD's hard-earned reputation as a builder of cheap, oddly-designed, unsubtle effects pedals*. The "blend" knob (added as a nod to the bassists who mostly bought the original unit, hence the "B" in FX25B) is not really a blend so much as a dry signal cut--keep it at 9 o'clock if you want any bass at all. The sensitivity is, sorry to say, way too sensitive, and usually has only one useful position, which (for me) is about 98% of the pot's available throw, and has to be precisely tweaked. Oh, and best of all, activating the pedal causes a volume drop of 20%, give or take.

And yet, with all those caveats, I always come back to this little green monstrosity whenever I want an envelope filter, because I can't get anything that will quack in quite the same touchy, uneven, delightful way. I've tried using Boss autowahs, a Bass crybaby, a regular wah, Digitech's bass synth, every Mu-Tron and generic envelope emulation in the Pod, and a gaggle of squawking VST effects--none of them give me the sound in my head as well as the FX25B. Not bad for a pedal that goes for ~$15 on eBay.

Little effects pedals like this are often called stompboxes, for the simple reason that they're metal boxes that you stomp on to activate them. But I love the word stompbox because it really sounds like something a little garish or unrestrained, and I myself am not a restrained effects consumer. It's like in every multi-FX unit review, the writer invariably calls the presets "showy" and "extreme"--I'm the guy who thinks those presets are awesome, and keeps them around just in case. It goes to show how individual sound choice can be, and why a pedal that can seem as profoundly silly as the DOD Envelope Filter is nonetheless one of my favorite stompboxes from my collection.

Or maybe there's just no accounting for taste.

* A quick key to the range of mainstream pedal manufacturers: if you want uninspired but reliable sounds, buy Boss. If you want incredibly cheap, flimsy, over-the-top crap, get one of those plastic Danelectro units. If you can't make up your mind between the two, you're a potential DOD customer.

November 13, 2008

Filed under: music»tools»looping

Wow and Flutter

These two words, wow and flutter, describe the charming warble created by variations in speed on an analog tape recorder. Believe it or not, those are technical terms. You've probably heard the phenomenon, even if you don't know it--it's the way that cassette players (back when people listened to cassette recorders) or old TV episodes slip slightly off pitch.

Whenever there's a technical flaw in audio equipment like this, it inevitably ends up being stretched in two directions. Obviously, when tracking musicians, engineers try to minimize the amount of wow and flutter so that the recording will be accurate. At the same time, the effects aren't always unpleasing, and so musicians often begin incorporating it into the music itself. So in much the same way that distortion from primitive amplifiers became part of the electric blues 'sound,' the pitch inaccuracy of tape-based delay and loop effects has become an effect in its own right. It's popular enough that digital delays like the Line 6 DL-4 or Boss Giga Delay have tape emulation modes, complete with adjustable, artificial wow and flutter (not to mention thousands of guitar snobs insisting that they prefer the real thing).

The LoFi Loop Junky I picked up last month in Portland is not a tape delay at all, but it does, quite intentionally, share a lot of the same sound palette. As the name implies, it's a low-fidelity sampler, with a hard frequency limit of 2.6KHz and a lot of self-noise from the storage mechanism. There are also two knobs for a built-in vibrato, giving it the same kind of pitch waver. I didn't buy the Loop Junky for those wow-and-flutter features, but they're rapidly becoming my favorite reason for using it.

When I first started the pretentious solo project, part of the goal (besides the ability to play music without subordinating my ego to a guitarist's) was to work on songwriting within the constraints of a loop pedal--and those constraints are not inconsiderable. But with that said, the equipment I've used has still been very capable: the DL-4 is not only very high-fidelity, with a long running time and layer-on-layer abilities, but it can also alter playing speed, reverse loops, or play them as one-shot samples. So while structurally it imposes some limits (oh, for an undo function!), sonically and mechanically it's a powerful machine, and it gives users a lot of freedom to explore.

You get none of that with the Loop Junky. It plays loops one way, at one speed, with no automatic stop--and it makes everything you play through it sound a bit like a cheap 8-track, to boot. When I originally heard about the pedal, that didn't matter: its main selling point for me was that it was a boutique pedal with a small footprint, a long recording time, and a battery life measured in centuries. As an occasional open mike player, where setup time is important, the heavy power draw and sheer mass of the DL-4 had become cumbersome. I figured a more agile pedal would be worth some feature set sacrifices.

What I didn't anticipate was the degree to which the Loop Junky, like all Z. Vex pedals, would make up for the loss of flexibility with character. For example, the vibrato is a lot of fun, but it has a kind of low-pass filter on its sound so it's really only audible on mid-range notes, or on chords higher up the fretboard. On low-fret basslines or percussive riffs, the effect is wasted. As a result, with only one layer available, I find myself adapting my approach: instead of looping beats and basslines, it's much more fun to loop the chords and play the rhythm parts by hand. Sometimes this means that a given song just won't work, but the challenge of adapting material to the pedal is markedly different from the Line 6. Instead of simply adapting a song's parts into structural cues, it feels more like creating a new arrangement.

To be honest, it's not surprising that I like the pedal for these reasons. I'm a digital aficionado for recording and production, but when playing out I usually prefer the simplicity and character of analog effects, and I doubt that I'm alone in feeling that way. It's part of what makes the words "wow and flutter" so appropriate--they capture the distortions and flaws that musicians have embraced and incorporated into the genres and forms of their music, not as a burden, but in a sense of joyful play.

September 23, 2008

Filed under: music»recording»production

Unnatural Selections

Buying an MP3 player for the first time has made me think a bit more about the weirdness of contemporary popular music.

I used to rail about MP3, but writing the Audiofile articles for Ars opened my eyes on a lot on the realities of the technology. I've also mellowed out on sound quality when it became obvious that MP3 was a disruptive technology for individual musicians, and as I thought more about the ecological impact of CDs. I'm still not very keen on buying MP3s directly, so I'm trying out the Zune music subscription, and so far I like it quite a bit. I find it's helpful to think about it as a paid replacement for Pandora, one with lots of extra features and offline capability, instead of as a "rental" system.

But as I go through the honeymoon period with the hardware, I'm listening to a lot more music. I'm listening to it a lot more closely, trying to keep my "producer's ear" in practice (as much as it ever was). And when you do that, the surreality of modern recordings is really fascinating.

For example, I was talking to a friend a while back about recording tricks, and I mentioned the standard technique of using a sidechained compressor on drum tracks to make the snare "pop" more or tame boominess. Most people are aware of compression in general terms, as part of the mastering step--the prevalence of Loudness Wars articles makes sure of that. But I don't think most listeners are aware that individual tracks are also compressed, and that the compression can be triggered by other, separate tracks--or that this is, in fact, a special effect that's part of the modern rock sound.

To the average person, this kind of production is transparent, because it sounds "natural" to us now. We think of that as the way music would sound--under great conditions, granted, but still plausible. But when you start to break apart the processing that's done on even stripped-down productions, and you consider how that compares to, say, a person standing in a room with a band, it starts to form a bizarre picture. Take the following list:

  • The guitars and half the drums may be tied together in one "room" or acoustic space by a reverb.
  • Bass and kick-drum usually don't get reverb because it muddies the mix, so they're in another "room," one that's acoustically dead.
  • Vocals get yet another reverb setting, usually, depending on the effect the engineer's looking for.
  • Drum levels are compressed, often separately, in a way that sometimes--but not always--mimics the response of the human ear to loud sounds. Other tracks, however, are not compressed with the same psychoacoustic triggers. It's like some things are "loud" without actually being higher in the mix.
  • Even simple guitar parts are often double- or triple-tracked, and they're recorded with mikes right up next to the cabinet, as if the listener had their ear right in front of the speakers.
  • Simultaneously, the listener is also directly "in front" of the vocalist, who is also standing (in the stereo field) probably in front of the drums.
  • None of these elements cast any kind of acoustic shadow, or block any of the others from being heard.
It's a profoundly unreal set of manipulations, perversely designed to make music that sounds more real to the listener. It's so good, in fact, that it sounds more real than the real thing. Audio pundits often complain about the glossy perfection of music production, but there's another way to think about it, and that is that all of this production is intended to flatter the listener with the powers of omniscience. The reason producers work so hard to eradicate mistakes is that the audience will be able to hear everything in a way that no physical person ever could.

August 26, 2008

Filed under: music»tools»effects


CDM has a preview today of the Openstomp pedal, which allows custom audio effects built in a visual programming environment, then loaded onto a durable stompbox with two switches and four knobs. It's in limited production now for $349. Not a bad price, considering.

Having messed around with computer-based live effects for a while, then returned to a mostly analog signal chain, I'm torn on this. I love playing with sound design, and would be thrilled to have a customized multi-FX pedal in my hands. But I've also found that pedalboard obsession comes at the cost of musical productivity, and despite months of tweaking, I never did create a distortion better than my MXR Bass DI, or an auto-wah that I liked better than my queasy-sounding DOD FX25.

In the end, I'm not an electrical or DSP engineer. I'm a musician, and that's where my strengths lie. The idea of the Openstomp is something I find powerfully attractive. I may pick one up. But musical utility is not, to me, what it has to offer.

It's also worth noting, I think, that calling it the "open" stompbox is a bit of a misnomer, given that other effects pedals are hardly "closed." Anyone with a soldering iron, a few bucks, and an Internet connection can easily find detailed plans and explanations of how to create (or recreate) their own homebrew guitar pedals. And it's unlikely that the visual DSP in the Openstomp programming kit is any easier to understand than a wiring diagram, given my experiences trying to hack something together in Pd.

Where the Openstomp competes is with proprietary multi-effects boxes like those from Digitech and Line 6. But even there, they have very different goals: the proprietary gear is really all about putting many traditional, analog-style sounds into a single box. Tweaking it beyond the manufacturer's specs is not possible by design. People don't buy a Pod or a GNX-4 because they want to make something crazy. They buy it because they want a pedalboard's worth of familiar sounds at the touch of a button. Being "closed" or "open" is irrelevant to them. Indeed, being limited in very specific ways is actually a feature of these devices, not a bug.

This is not the first time that people have tried something along these lines. The KVR Receptor puts VST plugins in a rack, and Justin Frankel designed his Jesusonic pedal before taking on DAW software with REAPER. The latter never really got off the ground. The former has become mostly popular with synth players who don't want to tour with laptops. If the Openstomp has a future, it most likely rests in its ability to provide one-shot studio-style sound tricks or music visualizations.

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