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  1. Understanding Those Magic Boxes of Doom By Jon Chappell Most outboard effects behave in predictable ways as you move from manufacturer to manufacturer. For example, you can pick up any brand of digital delay, set at the delay time to 125 ms, the feedback to one repeat and the level to 50\%, and get essentially the same, expected sound. Quality issues aside, you can also get predictable results from an EQ. This is a good thing, as it helps you set up the sound you hear in your head on different rigs. But guitar distortion pedals are the “black boxes” of the effects world; they are all unique, inscrutable, and adhere to no known standards for parameter definition. You don’t know how the Tone control is voiced, which harmonics are emphasized as the Distortion knob is cranked, or even what effect the Level control has (such as whether it works dynamically with the other controls or just boosts the existing signal to a louder level). Often the manuals are no help either, preferring not to reveal the mystery of what goes on inside their magic boxes. So the bad news is, it’s virtually impossible to tell what the distortion pedal sounds like without auditioning it personally. There are no shortcuts, like reading reviews or scanning spec sheets. You just gotta drag yourself down to the local emporium and plug in. The good news is, it gives you an excuse to go shopping! And you can evaluate these disparate mystical contraptions—and even compare and contrast them— by using some basic common sense. What’s in a Name Fig. 1: Behringer’s Blues Overdrive BO100 and the Boss Super OverDrive SD-1. Both feature similarly named controls. The key to t he pedals’ tonal character lies in their model names. If you’re seeking a warm bluesy overdrive, you can pretty much a eliminate anything with the word “metal” in the title. Conversely, if you’re trying to make Slipknot’s James Root look like a tone wimp, don’t limit yourself to mere “overdrive” pedals or effects with the word blue or tube in them. Often the best clue to pedals’ sounds are in their names, even if they feature controls that are similarly named, as shown in Fig. 1. You’ll find it’s tough to get any hard information from ads, because companies try to outdo each other with descriptive superlatives. Also compounding the confusion is that some companies name their controls in a completely nonscientific way. Witness one company that released a pedal with controls called “Butt” and “Face.” The Ibanez Tube King’s Void control (Fig. 2) is perhaps not as flip, but it’s equally mystifying. Remember though, as odd as these names might strike you, it doesn’t mean the sound is necessarily worse (or better) than a pedal with more conventional named controls. Again, you can’t determine the quality of a pedal sound by looking at it, but you can get clues to its category. Overdrive, Distortion, and Fuzz Fig. 2. It’s pretty clear what you’d use the Ibanez Tube King for. And many of the controls are intuitive. But what’s “Void”? These terms are all used loosely to describe distortion, but discriminating tone freaks will make distinctions between them. Overdrive is what happens as a signal is pushed beyond the limits of the circuitry’s ability to reproduce the sound faithfully. A slight overdrive has a warm, smooth, and somewhat dull sound that is musically pleasing on a guitar signal. The next category, distortion, covers the widest tonal area, because it includes everything from “just beyond tube warmth” to the brittle buzz saw effect of System of a Down. Within this range is all manner of metal, alternative, hardcore, and industrial timbres. Fuzz is almost a caricature distortion effect, producing a nasally, buzz saw tone that is generally useful only on single node lead lines. It was made famous by Jimi Hendrix (who played a Fuzz Face), and is now called for by name, though many producers use the word generically described any distortion beyond the warm blues furriness. There’s no accepted category for “beyond fuzz,” but certain hardcore adjectives are making their way into the nomenclature, notable metal (see Fig. 3). Fig. 3. Though the DigiTech Death Metal has conventionally named controls—Level, Low, Mid, High—the name (not to mention the color scheme!) tells you its tone will be anything but conventional. Once you have narrowed the sonic field to blues, metal, or subterranean death knell, and are fairly clear on the naming conventions of distortion’s controls, it’s time to check out some pedals. You can do this by going to friends’ houses or visiting the local music shop. But you must be systematic when using your “testing instruments.” You can’t go into a store and try Seymour Duncan’s Lava Box with a Les Paul through the Marshall and expect the unit to behave the same way when you try it in Store B’s Strat and Fender Vibrolux. The best way would be to bring your own rig to the shop. But if that’s not practical, work to get the closest setup at the store to the one you have at home. Hint: approach the store during their downtime so the salespeople will be in a better position to set you up. Bench Test Once you have the prospective pedal set up and under your feet, the first thing you should do is establish the pedal’s unity gain setting. That’s the point that produces the same volume level when the pedal is active as when it’s bypassed. This will let you hear the pedal’s effect without the influence of psychoacoustics—that is, the ear responding differently to the same frequencies at varying loudness levels. Typically, your starting levels will look like those in Fig. 4. Kick the panel on and off a couple of times to hear what the pedal does to your tone in its most neutral state. Then slowly crank the distortion control from leftmost to rightmost position, noting not only the differences but how the unit is calibrated—how drastically the unit changes from low to high. Make sure to play real-world examples: lead lines, rhythm figures, arpeggios, percussive, and sustained passages. Then try touching up the sound with the pedal’s tone control. That’s part of the pedal’s character too—how its E.Q interacts with the distortion. Using EQ with distortion is an important element in tailoring your sound. Generally, the higher the distortion setting, the more treble you’ll need to add. The reason is that the more distorted your signal, the more compressed it becomes, and compression rolls off high frequencies. But should you use the pedal’s EQ or your amp’s? Or an outboard EQ? And if you use an outboard EQ, should it come before or after the distortion? Only your ear can decide. Just remember this paraphrase from Woody Allen: If it’s not done dirty, it’s not done right. He was talking about guitar tone, right? Jon Chappell is a guitarist and the Senior Editor of Harmony Central. He has contributed numerous musical pieces to film and TV, including Northern Exposure, Walker, Texas Ranger, All My Children, and the feature film Bleeding Hearts, directed by actor-dancer Gregory Hines. He is the author of The Recording Guitarist: A Guide for Home and Studio (Hal Leonard), Essential Scales & Modes (Backbeat Books), and Build Your Own PC Recording Studio (McGraw-Hill), and has written six books in the popular For Dummies series (Wiley Publishing).
  2. Using Reverse Polarity Adaptors on Your Mix By Jon Chappell A phase switch on a mixer is actually a polarity switch: it changes the positive voltage to negative and the negative voltage to positive. The reason it's called a phase switch is that that's how people use it: to eliminate phase cancellations in signals coming into the mixer. Reversing the polarity of a signal (and the waveform produced by that signal) is the same thing as having the signal 180 degrees--the maximum--out of phase. That's why in this one situation you can use the terms "out of phase" and "reverse polarity" interchangeably, even though it's not quite correct: because it produces the same results. To Phase or Not Switching the phase--er, polarity--of a channel is often a good test to perform on elements in a mix, especially where a lot of mics are used, and where more than one mic is used to capture a source. For example, if your guitar or bass blends a mic with a pickup (as is often the case with a plugged-in acoustic guitars), you may want to do a quick polarity check to make sure the mic and pickup are in phase. And if not, reversing the polarity of one of those signals may improve things. How do you know if you've improved it? If it gets louder or fuller. If the sound becomes hollower, boxier, or softer when you switch, chances are it was okay as is. You may even want to go with the "incorrect" configuration if it helps eliminate feedback, helps to balance a boomy instrument, or otherwise offers a more desirable sonic character. In the final analysis, you must use your ears to decide. But what if your little stage mixer doesn't sport one of these handy switches? Well, you can create a simple XLR adaptor, consisting of: a male-to-female connector, where pins 2 and 3 are switched at one end. This goes inline between the mic cable and the mixer. You'll also hear this device referred to as a "polarity transposition lead," but it describes the same thing--a short inline (male to female) XLR adaptor cable with the positive and negative wires crossed at one end only. You can make your own, if you're handy with a soldering iron, or you can buy one of the many available pre-made ones online or at your favorite brick-and-mortar emporium. Home Brewed vs. Store Bought To make your own, you just swap the wires going to pins 2 and 3 on one end only of a male-to-female XLR cable. Figure 1 shows the schematic. Fig. 1: In making your own polarity reverse adaptor, just switch the leads of pins 2 and 3 at one end (leave pin 1, the ground, connected). Note that since this is a three-wire system (positive, negative, and ground), you do need to know which wire is the ground--and to leave that one alone. If you don't want to bother with disassembling a connector and breaking out the soldering gun, just buy a ready-made connector that does the job for you. I found several on the web, including models from Shure, Hosa, Whirlwind, and Pro Co, but the Hosa gets a mention here because it is the least expensive (see Fig. 2). Fig. 2: Hosa has a no-frills adaptor (the GXX-195) for only about $8. You could buy two at that price for less than one from many other brands. Whirlwind has one called the IMPHR ($16, street), which has the letters boldly displayed on the barrel housing, making it easy to identify when rummaging through your toolbox (see Fig. 3). Fig. 3: Whirlwind's Phase Reverse is another way to deal with a miswired mic or cable. Guerrilla Tip I don't advocate doing this except in an emergency, but you can actually make an inline splice, which doesn't require soldering tools or even disassembling the connectors. If you cut a mic cable with scissors or a carpet knife, you can more easily strip the insulation of off the wires coming from pins 2 and 3 and reverse them than if your were working with the lugs at the connectors. Use a volt-ohm meter to check the continuity of the pins and the corresponding wires to make sure you're making the right hook-ups. You can re-connect the cut wires by simply twisting them together and knotting or taping the cord such that there's no strain on the connections themselves. It ruins a cable, of course, and it's not the most secure way to make an electrical connection, but in an emergency it can be done (and I've done it). And here's another tip: if you do go this route, make the cut close to the connector so that you lose only a couple of inches of cable when you have time to do the connection the right way (i.e., with a soldering iron and onto the lugs of the connector). Alligator clips will also help hold the wire-to-wire connection together, but be careful not to bump them. Jon Chappell is a guitarist and Associate Editor at Harmony Central. He has contributed numerous musical pieces to film and TV, including Northern Exposure, Walker, Texas Ranger, All My Children, and the feature film Bleeding Hearts, directed by actor-dancer Gregory Hines. He is the author of The Recording Guitarist: A Guide for Home and Studio (Hal Leonard), Essential Scales & Modes (Backbeat Books), and Build Your Own PC Recording Studio (McGraw-Hill), and has written six books in the popular For Dummies series (Wiley Publishing).
  3. Overcoming New Project Inertia Through Templates By Jon Chappell I had an experience recently that I’m sure many of you can relate to. I heard a great acoustic guitar patch in Ableton Live Suite, and so wanted to use that in a project I was working on. But I’m not a Live user (not yet, though I plan to be). So when I installed and launched Live 7, I was completely lost. Staring at a blank project window—which looked nothing like Cubase and Pro Tools, the DAWs I’m used to seeing—I couldn’t even figure out how to get my MIDI keyboard to make a sound (despite the fact that I was transmitting MIDI just fine). I wasn’t going to be able to jump in and start working without (oh, the horror!) cracking the manual. And Live is known for being easy to use. It’s just that it was brand new to me and I was under deadline pressure. But wait. Instead of doing the unthinkable (reading TFM), I simply opened up a couple of Live projects I found in other places (online, on demo discs, sent from friends). Once I opened a few existing projects, I found one that was almost exact in setup and approach to what I needed. Then I went to work doing something I’m really good at: reverse engineering. Reverse engineering is for people like me who aren’t quite smart enough to create something new from a clean sheet, but who are savvy at editing—seeing what was done and improving it, or simply changing it to suit the situation at hand. I’ve been doing it my whole professional life, from rewriting articles to “designing” (ha!) my own web pages. Usually, you’re reverse engineering another’s work. But sometimes you can turn that inward so that you’re doing it to yourself. Templates on Parade You perform reverse engineering every time you work from a template. A template is a file that resembles closely what you’re going to end up with in structure and setup, but where the actual content is completely original. Commonly provided as templates are such files as web pages, newsletters, and résumés. If you’ve ever used a sample of one of those as a starting point, you know how a template—or reverse engineering—works. Simply dump the guts but leave the skeleton intact and put your own stuff in. So I went back to the audio programs that I use a lot, and thought hard about making templates. I’ve been using Pro Tools lately, so I’ll use that as an example, but these techniques her apply to all DAWs. It’s just that Pro Tools is good for explaining template concepts, because its interface is fairly intuitive. Even if your DAW does things differently, you can readily see what’s going on by seeing how PT handles things. First rule of templates: opening a blank project to begin anything is a huge waste of time. Better to have 10 templates to choose from than one blank project window. In my own case I know I’m almost always going to immediately set up 90 percent of my projects this way: four audio channels for a principal guitar and three overdubs, five MIDI instruments (piano, bass, drums, percussion, pads), and at least two click tracks (one with a pulse according to the meter and one based on subdivisions of that meter for slow recording). This is to say nothing of the additional setups I always use: buses for shared effects (which PT handles as Aux Input tracks), insert effect plug-ins, virtual instrument plug-ins, and the look of the environment itself (windows I like to have opened and positioned in various places on my dual monitor setup). It doesn’t matter if I’m doing hip-hop or folk music: I always like to have things look a certain way to begin. And that way is certainly not a blank project window. So my template looks like Figure 1. Fig. 1. The mixer window of a basic template, including audio tracks, a MIDI rhythm section, some various click tracks, and an aux and master bus. It’s not fancy or complicated, but remember, there’s some work done under the hood as well, such as the tweaking of plug-ins, and such. The point is, you don’t have to do much to create a template that’s personalized and gives you a little jump in overcoming the inertia of starting a new project from scratch. A Template by Any Other Name If you’re still foggy on what makes a template vs. an actual file, it’s not your fault. Microsoft Word and other programs make the concept of a template needlessly complicated by introducing a new file format. When you open a blank doc in Word, start typing, and realize (quickly, I hope) that you should save your work, you’re faced with saving the file as its own document (.doc) or a template (.dot). Saving as a template means you can never change it, because every time you open it and choose Save, it’s really as Save As. You can’t actually change the template document itself by invoking Save. This is usually a good thing for templates, but if you’re careful enough, you don’t need a separate file format. Simply open a document, and do a Save As as your first act. This ensures that you don’t start working on the template and reflexively (as I do) hit CTL-S as you work. By doing a Save As first, you now are working on the specific project, and the original template is safe. The advantage of being able to actually save to the template (which Word doesn’t let you do, except through convoluted means) is that when you discover something that should go into your current project as well as the template, you can perform that easily. You should always have the template in the back of your mind when you’re working on a separate project. In fact, I’m always aware if I’m making a change that could apply to the template. I don’t necessarily close the current project and open the template, but I do jot myself a note and do it at a good break point. I’m especially on the lookout for this when I do something “unusual but global.” For example, when I invoke a seldom-used window (like the MIDI event list editor), I’ll first size the window to my liking and then shuttle it off to my secondary monitor to get it out of the way. But when I open the window again, the program remembers the size and position. That’s how it should be in every file I work on, so that’s one for the template. It’s a little thing, but little things add up! Transport Tricks Let’s talk about some specific template fodder. The look and feel of the transport is a great candidate for template work. I almost always use bars and beats, so my primary counter is set in musical units, while my alternate counter (called a subcounter in PT) is in minutes and seconds. I also use MIDI a lot, so I like to have PT’s transport in expanded mode, which offers MIDI options as well. But some people don’t like the transport window cluttering up their project at all; they would just as soon minimize it or dispense with it altogether, using keystroke equivalents for any transport navigating. Figure 2 shows two ways to set up a transport: the deluxe way, which is what I use, and the minimized way, for less clutter. Note, however, that the movable transport window in the unexpanded mode resembles exactly the permanent transport to the right of the counter up top. If you’re going to minimize your transport to this level, you should just hide it completely. Note also that in my version, it’s bars|beats|ticks as the primary counter and hours|minutes|seconds in the subcounter. In the unexpanded counter, it’s hours|minutes|seconds in the primary, with samples underneath. You can see that the counter is parked at 2 seconds, which equates to 88,200 samples. This means I’m working in a session whose sample rate is 44.1kHz. Your template’s counter should reflect how you work. Fig. 2. The transport and counter are configurable is several ways, and should be the first order of business in setting up a template. Marker Madness Speaking of transports, one of the things that bothered me about PT was that it wasn’t easy to go to, say, bar 33, in a jiffy—at least not on using keyboard shortcuts. But PT does allow you to go quickly between markers by typing period-marker number-period (on the number pad, it’s .#.). This, I realized, was a template opportunity! So I set up my template by placing a marker on every single bar for 128 bars. This is a chore, yes, but it’s a one-time chore, and it’s easy to do. So my template looks like Figure 3 with respect to markers. Fig. 3. Pro Tools allows you to go quickly between markers (if not bar numbers) using keystrokes, so just set up a marker for every bar. It’s a one-time chore, which means it’s perfect template material. The Marriage of True Mids You can set up not just one thing in a template, but a combination of elements as a starting point in a template. For example, I often group a virtual instrument and a plug-in effect. Let’s say I find I use one instrument a lot, but it needs the same EQ (what instrument is ever perfect “out of the box”?). So I keep my EQ plug-in married to the instrument as part of the template, as Figure 4 shows for the bass. Fig. 4. In addition to using staple instruments, such as bass, I often couple them with an EQ curve in my template. The virtual instrument and effect are treated as a single unit as far as the template. Fix the Mix You can even set up dummy tracks or things that you won’t necessarily use every time but are there as place holders should you need them. A good example is the use of plug-ins. You can’t load up a template with a hundred active plug-ins because they’re a drain on the CPU. But you can set up a lot of plug-ins, disable them (or the channels they’re assigned to), and save the template in that state. Then when you open the template, in its default state, it won’t be sucking on the CPU from the outset. In Pro Tools, you choose “Make inactive,” which allows you to see the channel (with all the labels appearing in italics, as shown in Figure 5) without the plug-ins (or any program material) loading down your CPU. Fig. 5. Note that the four guitars on audio tracks each have an insert reverb (D-Verb) and EQ. The tracks have been made inactive here, so that I can still see them but where their plug-ins won’t be a drain on the CPU (something Bypass doesn’t accomplish). Template Temperament After you get in the habit of setting up templates, you realize it’s like your music creation wardrobe: you’ll have something for every occasion. Stuff your templates full of material and then delete what you don’t need as soon as you open them. But having a loaded template provides you with a system check. For example, I always have a bit of audio and MIDI, and effects buses (if they’re separate tracks, as they are in PT) loaded in my templates to make sure the system is “firing on all cylinders”—in other words, whether the project loads and passes audio correctly, including the plug-ins. You should have something in different types of tracks, preferable pleasing to the ear, even if you’ll dump the stuff as a first step. If you’re not hearing the audio track and its associated effects as soon as the template finishes loading, you won’t have to wait until you’re in mid project to troubleshoot the problem. So if you want to really impress someone, don’t just tell them your favorite DAW tips. Show them your template!
  4. Use Your DAW to Create Arpeggios, Tremolo, and Other Delay-Based Effects More Effectively Than Your DDL By Jon Chappell There are tons of ways guitarists and keyboardists can use timed delays to enhance their recorded parts. From arpeggios to rapid-fire drones to tremolo, timed delays can add a new rhythmic dimension to a percussive or staccato figure. But setting up an outboard delay (or using the plug-in variety) is often clunky, and it won’t move with the music if the tempo changes (between sections, with intentional accelerandos and ritards, or just naturally shifting over time). Tap tempo addresses some of those problems, but that’s yet another thing to do with your body. And for the repeated sounds, an outboard DDL’s feedback parameter offers very poor control. So instead of trying to sync up your delay to create rhythmic subdivisions, you can use your DAW and copy tracks to perform the same function. Any timed-delay operation works best on staccato or steady-stream rhythms, such as a continuous line of eighth notes. Record your part on Track 1, and then simply “clone it” by copying and pasting it to an adjacent track. (A better way is to use the shortcut keystroke by holding down a modifier key and dragging the part; on Pro Tools, it’s the shift key.) Make sure you’ve done all your trimming first, so that you can more easily align and offset clips by having the event start point line up with the note’s initial attack. Repeat this cloning process as often as you need discrete repetitions of your part. More track copies means more repeats, which is analogous to a higher feedback setting on your DDL. The longer you space the tracks apart in time (or the greater the horizontal distance on the timeline between start points), the higher the delay-time setting. Then you have to snap your cloned tracks to predetermined time intervals. Obviously, you should know the rhythmic scheme of your part before setting out. Know your tempo, have a calculator handy, and set up the grid to show minutes and seconds rather than bars and beats. The red circles in Fig. 1 shows Pro Tools set up to show the grid and counter in minutes/seconds, the tempo, and the start time of the clone track (at 200ms). DELAY BY THE NUMBERS For easy math, let’s say I’m doing a slow tune at q = 60. That equates to one beat per second or one quarter note every 1,000ms. To do subdivisions of the quarter note from there, I simply take a fraction of 1,000ms. For example, eighth notes will come at 500ms intervals, 16th notes at 250ms intervals, and eighth-note triplets every 333ms. Once you figure out what your subdivisions are, the next step is to drop markers at the appointed intervals. Use the snap-to-grid function to get your cursor to quickly go to the right grid point. Then when you drag-copy the part from the original to the new track, the audio segment should snap right to the pre-set markers. POOR MAN’S TREMOLO Fig. 2: Click for full image and description. In Fig. 2, I have set up a five-note subdivision for my quarter note, which will yield what I call “poor man’s tremolo.” This means that I am either too lazy or too unskilled (or both) to play a proper picked tremolo for any length of time. If you’ve ever had to play Italian-style mandolin or plectrum banjo (as I have, as a pit guitarist), you can feel my pain. And have you ever tried to create a tremolo with a delay unit? It’s exceedingly difficult, even with a multitap delay. But on a DAW, you can create rock-solid tremolos by simply copying your principal track and spacing it evenly, according to the time-converted subdivision of the tempo. So in our example of q = 60, you can create a five-note-per-beat tremolo by simply copying the original track every 200ms (principal note = 1:000 ; 2nd note = 1:200; 3rd note = 1:400; 4th note = 1:600; 5th note = 1:800; new downbeat = 2:000). Too bad I can’t transport this technique into the orchestra pit, but it did get me thinking about “manufactured tremolo.” Five notes seems to work well here, as four notes per beat are 16th notes (which is not quite fast enough for a tremolo) and six notes is 16th-note triplets, which sounds a little “virtuosic”—especially when repeated with the machine precision of a DAW and not a human picking hand. Speaking of humanization, one trick I’ve learned is to vary the volume of the subsequent, delayed tracks. The highest track (Track 1) should be the loudest, and you can simply lower the volume of the other tracks to make the principal note speak better. I often soften the very last track (or trem note) before the downbeat to set up the new principal note. POWER MOD And you say you miss the modulation function of your DDL? No problem. Just introduce chorus, flanger, or reverb into the repeated tracks with a plug-in. The possibilities are endless here, because you can apply a different effect (or different amount of the same effect) to each repetition. This is a lot of processing power to be gobbled up by a single part, so watch your CPU load. Or bounce to disk and re-import when you think you’ve got it. Using track copy this way—to add a post-production delay—makes a lot of sense. You shouldn’t record a time-based effect in line, anyway, the way you would, say, distortion, and wah. Using a separate track in a DAW for the time-delay effect works just like a parallel effects loop: it leaves the straight signal un-modulated and blends in only the effected sound. Plus, you get really good at converting musical time (tempo) to real time (seconds), and you understand better how subdivisions and small-unit rhythms work. And if you’re really, really good, you can re-create these live, on the fly. But I’m still working on that!
  5. No More Math! Use This Excellent Excel Solution for All Your Impedance Calculations By Jon Chappell One of the great things about doing research on the Web is that you often find out stuff you weren’t necessarily looking for in the first place, but is valuable in another application. Case in point: I was researching the best way to hook up some lithium-polymer batteries to drive a high-powered motor for my radio-controlled airplane (a 60" P-51 Mustang), and I happened to stumble across a familiar, but out-of-context website: www.duncanamps.com. This was linked from one of the aeromodeling forums I frequent, and touted as “one of the best sources for software-based electrical-circuit calculators”— little apps that people make on Excel that provide easy answers to life’s little calculation problems. Wait a minute. duncanamps.com for electric motors? Of course, the site in question was for amp info! They wouldn’t be talking about batteries, I thought. And sure enough, they weren’t. But what the site was talking about was the way to calculate the resistance in series and parallel circuits. You see, combining the source of power (batteries) follows the same principles as the receivers of power (the speakers); it’s just from the other end. And that’s how I discovered this fantastic app. for calculating speaker hookup impedances. Since we guitarists never have to think about the power side (smart guys like Duncan Munro do that for us), we’re on the load side of the equation—usually in the form of hooking up speakers. The app. that Duncan amps offers is the most useful calculator (in the form of an Excel spreadsheet that you can download for free) for calculating power and impedance ratings for common amp hookups. Let’s look at the theory, and then you can forget all that and bookmark www.duncanamps.com for the easy way! Ways to Hook Speakers Up There are two ways to hook up speakers: in series and in parallel. At the most basic level, with two speakers, equal impedances in series doubles the total. Equal impedances in parallel halves the total. So when you have two speakers at 8 ohms apiece (guitar speakers are usually 4 or 8 ohms), hooking them in series (the positive terminal of the first speaker after the amp goes to the negative terminal of the second speaker) creates a circuit with 16 ohms. Hooking them in parallel (with their positive terminals all connected to each other and their negative terminals all connected to each other) creates a 4-ohm impedance. The problems come with more than two speakers, or when the impedances are mismatched, or both. Then you need some math chops. So let’s look again at the nature of series and parallel. Get Serial Series is easier, from both a conceptual standpoint as well as a mathematical one. A series circuit is where everything is in a daisy chain, like the battery configuration in a long-handled flashlight. All the terminals are wired end to end, so that you have a positive connecting to a negative connecting to a positive, etc. (By the way, most small electronic devices have their batteries connected in series, so even if you see an arrangement of eight double-AAs in two rows of four each, they’re configured electrically like in our long-handled flashlight example.) Schematically, here’s how a series circuit looks, with simple flashlight batteries, and with two speakers. Fig. 1. A series circuit for batteries and speakers. Each has a positive and negative terminal, which can be wired together. When working with speakers, you hook up the positive and negative terminals the same way as batteries. Don’t be confused by the geometry. Electrically, the flashlight and the speaker cabs are wired the same—in series. This is similar to the way you daisy-chain pedals, with the output of one pedal feeding the input of the next one in line. And just as in a pedalboard, the outer most connections, are the ones that go somewhere else. In an amp, it’s a circuit, like “circle,” so the first-in-line negative terminal and the last-in-line positive terminal are both connected to the speaker output jack on the amp’s back panel. When you want to compute the total resistance of the speakers in your chain (whether that’s a cabinet with its speaker terminals wired in series or a succession of separate speaker cabinets in a daisy chain), you simply add up the resistance, presented as ohms, and (usually) stamped on the back of the speaker. So if you have three 8-ohm speakers, your total resistance is 24 ohms. Mathematically that’s represented by this formula: Series R total = R1 + R2 + R3 + … Parallel Paroxysms Parallel connections hook all the positive terminals together and all the negative ones to each other, except for the first negative and last positive ones, which go to the source. Fig. 2 shows what this looks like with batteries and speakers. Fig. 2. In a parallel circuit, all the like terminals are ganged to each other. You can think of parallel as the “opposite” of series in one sense: the resistance decreases as you add more components. But it’s not simple subtraction (which would be the true opposite). In fact the formula is kind of complicated: Parallel R total = 1/(1/R1 + 1/R2 + 1/R3 + …) So let’s take the normal ohm values for speakers, which are 4 and 8. If you tackle the parentheses first, you realize right away that you can’t add denominators that aren’t the same, like 1/4 + 1/8. You have to convert them to the lowest common denominator (2/8 + 1/8 = 3/8). Okay so, we solve for the parentheses, and then we invert the results, creating the reciprocal, because that’s what the front of the equation, 1/(…), tells us to do. We’re left with 8/3. That isn’t just a fraction, it’s a term that has to be solved. So we divide 8 by 3 and we get 2.67 ohms. That’s the impedance of a 4-ohm and 8-ohm speaker hooked up in parallel. And it’s not easy to intuit that number. Whew. That’s a lot of work. Is there an easier way than the equation above? Yes! Easier, but not as elegant, as it requires more steps. The way to re-jigger that complex, fraction-laden formula is to do this: Parallel R1*R2/R1+R2 So plugging in the numbers 8 and 4, we get 32/12. That equals 2.67. You still have to divide as your last step, but most people find this: Parallel XY/X+Y is easier to deal with than 1/(1/X + 1/Y) Deeper into the Parallel Universe Things get even weirder with parallel when we combine speakers of differing impedances. And that is when, dear readers, we stop thinking and start surfing! Though the math still holds true, you can go to this fantastic calculator at www.duncanamps.com/technical/impedance.html You can download and then use this calculator for several different speaker configurations, including series-parallel, which will give you different impedance values according to the way you hook up more than two speakers. Check out the following four screen shots, taken from the separate sheets of the Duncan amps Excel application, and you’ll see why this is a cool program—simple but clever. Fig. 3: Here is how a simple two-speaker hookup looks in series in the Excel program offered at www.duncanamps.com. Notice that I’ve highlighted the cell located at E8 (Column E, Row 8). Whenever you select a cell in Excel, it shows you the formula used in that cell in the formula bar (also highlighted). Here, no surprise, it just adds the values of the cells (indicated by the column/row locations E6 and E7). The great thing about this spreadsheet is that it will show you the power distribution to each speaker (see E10 and E11) in addition to the load the speaker produces on the circuit. This can come in handy later on with more complex arrangements. Fig. 4. This is a simple parallel circuit, again, with the key cell selected, which shows the formula (Excel-style) in the formula bar. This is where the calculator starts to earn its money, if you’re plugging in numbers that are not the same values for the speakers. The trick of taking half of the value of two speakers of equal impedance works only with that scenario. The more complex formula for parallel will work for all parallel scenarios (different numbers of speakers, differing impedances among speakers). Fig. 5. Here’s the spreadsheet configured with three speakers in parallel. For easy math, I’ve left the speakers with the same impedance (all 8 ohms). But if I changed one of the speakers to 4 ohms, the value would be 2 ohms. Change two of the speakers to 4 ohms (leaving one at 8) and the value would be 1.6 ohms. Complex Arrangements The next two screenshots show complex arrangements: more than two speakers and in the arrangement called series/parallel, which combines aspects of both serial and parallel circuits. Note that in these next two examples, we still have not used speakers of differing impedances. This not only keeps the math more intuitive, but it’s the most likely scenario you find yourself working with speakers of the same values. Fig. 6. A series-parallel hook-up with Speaker A in parallel with B, and AB in series with C. The arrangement above mixes serial and parallel by putting the A speaker in parallel, while leaving the B and C speakers in series. The result in the above series/parallel configuration is 12 ohms, which is less than 24 and closer to 16, or a two-speaker cabinet in series. Fig. 7. A series parallel configuration with Speaker B in series with A, and AB in parallel with C. The above arrangement is a slightly different series-parallel configuration than the one shown in Fig. 6. Here, the B speaker is placed in parallel, and the A and C speakers are in series. This produces a total impedance that is closer to the all-parallel hookup of 2.67 ohms, but higher than that dangerously low rating, and closer to the 8-ohm rating of a single-speaker cabinet or the 4-ohm rating of two speakers in parallel. In both arrangement of the series/parallel configurations above, the range is closer to the original 8 ohms of the speakers used than if the circuit were entirely series (24 ohms) or entirely parallel (2.67 ohms). This gives us much more flexibility when trying to maximize the power of the amp, and run it safely. Better Living Through Excel The calculator takes you up to four speakers in series, parallel, and series/parallel with two configurations each. This should give you plenty of options to think about, and be sure to run numbers of different impedances to see what the results will bear. You really shouldn’t be running more than four speakers on one amp, because it creates power-management problems, but with this handy calculator, you can “run the numbers” of any likely scenario you’re likely to encounter—and learn about speaker hookups in the process. \_\_ Jon Chappell is a guitarist and the Senior Editor of Harmony Central. He has contributed numerous musical pieces to film and TV, including Northern Exposure, Walker, Texas Ranger, All My Children, and the feature film Bleeding Hearts, directed by actor-dancer Gregory Hines. He is the author of The Recording Guitarist: A Guide for Home and Studio (Hal Leonard), Essential Scales & Modes (Backbeat Books), and Build Your Own PC Recording Studio (McGraw-Hill), and has written six books in the popular Dummies series (Wiley Publishing).
  6. Don't Impede Me! By Jon Chappell P.A. systems and guitar heads and cabs are often modular affairs, with the speaker cabinet being separate from the power amplifier (unless your P.A. have powered speakers). Having separate speakers provides more versatility, as you can often mix and match speaker systems to suit the job. But before you start using your Crown power amp with your guitar player’s Celestion cabinets, or attaching a 4x12 cab to your Marshall head where you’d been using 2x10, you have to know that you can safely match up those particular components. If you get it wrong, you could permanently damage your amplifier, speakers, or both. (Of course, a speaker system designed for guitar amps might not be the best choice for a P.A., but that’s a separate question.) So we need to deal with, understand, and read the labels regarding power and impedance. These specs are related, so we’ll discuss them together, and give you enough to assemble a properly matched system. Keep in mind that a given power amplifier is designed to deliver a specific amount of AC power, and a given speaker system is designed to handle a specified maximum amount of AC power. If your amp delivers too much or too little juice to the speaker, the amplifier, the speaker, or both can be damaged. The amplifier and speaker are a team, and they need to be properly matched. So let’s see how to make a good match. CURRENT AFFAIRS Let’s start with the obvious: a power amplifier increases the level of an electrical signal to the point that it can cause an electromagnet-powered speaker coil to move in and out in the same pattern as the electrical waveform, which in turn causes compression waves in the air that we perceive as sound. The whole thing is predicated on the electrical signal, the speaker cone, and the resulting sound wave moving back and forth identically (to the extent possible) between positive and negative values. Alternating current (AC) is so named because it alternates between positive and negative values, which are what we need here, so a power amp always sends AC power to the speakers. DC power is deadly for speakers because it continually drives them in one direction (positive or negative). THE HIGHS AND LOWS OF IMPEDANCE You undoubtedly have heard other musicians talk about high- and low impedance-inputs on mixers, power amps, speakers, and guitar amps. Indeed, impedance is an important factor in matching parts of the signal chain, and it pays to understand at least a little bit about it. For practical purposes, an electrical current always meets some opposition when flowing through a circuit, even if that circuit is merely a straight copper wire. Therefore, a speaker’s electronics always present a certain amount of opposition to the AC electrical flow coming from the power amplifier. The opposition to alternating current is called impedance because it hinders—that is, impedes—the flow of electrons. Impedance is represented by the letter Z in equations, and it’s measured in ohms, which is symbolized by the Greek letter omega (Ω). (Impedance is closely related to resistance, but the two are not identical.) Every speaker system has an impedance rating that indicates how much opposition its circuitry presents to the signal coming from the power amplifier. Common impedance ratings for speakers are 2, 4, 8, and 16Ω. If the speaker impedance is high, it won’t let much current flow, so it doesn’t demand much from the voltage source (the power amp). In such a case, the speaker is said to present a small load to the power amplifier. On the other hand, if the impedance is low, a lot of current can flow, and that puts a high demand on the power amp—that is, it presents a large load. Remember: high impedance, small load; low impedance, large load. Got it? Good, let’s talk about power. WATTS THAT? Electrical power measures how much work a given voltage and current can perform when presented with a specific impedance load. This is why we said earlier that the concepts of electrical power and impedance are related. The unit of measure for electrical power is watts, the symbol for which is the letter W. When you check the power-output rating of an amplifier, therefore, you will see that power rating expressed as a number of watts of power into a number of ohms of impedance. On the back of the power amp, you’ll often see a notation that reads 250W/channel @ 4Ω. This means that each channel outputs 250 watts, assuming the amp channel is connected to a speaker cabinet providing a 4Ω load. Matching the impedance of the load (the speakers) to the output of the amplifier is crucial for achieving maximum efficiency in a system. Efficiency means that all of the power is being used to drive the speaker, and as little as possible is being wasted as heat. If an amp expecting a resistance of 4 ohms encounters a lower impedance (a larger load), such as 2 ohms, it will work harder and harder to deliver current to keep up with the current-sucking load. Eventually, it will heat up and burn out. On the other hand, if the amp encounters a higher impedance (smaller load)—say, 8 ohms—it simply will deliver half as much power (in theory), which is wasteful but generally not dangerous in and of itself. However, you can still have problems if the amp’s power drops so low that it can’t properly drive the speakers. In that event, the amp can start distorting (clipping) the signal, and distortion can rip speakers apart. In fact, assuming we aren’t talking about extremes, you are more likely to blow out a speaker by using an amp that is not powerful enough than by using one that is too powerful. So if you’re faced with a mismatch, remember, assuming you have sufficient power, “four into two won’t do; four into eight is great.” The same applies to other impedance ratings: an amp that is designed to work with 8Ω speakers might be fine with a 16Ω speaker system but not with a 4Ω or 2Ω speaker system. It is important to note that in the real world, cutting the speaker impedance in half does not necessarily cause the amplifier to deliver exactly twice the power. There are many places in a circuit where power is lost, including the speaker wire. Higher current can cause greater losses in transistors and the power supply, as well as in the wire. Heat, that mortal enemy of electronic equipment, also adversely affects an amp’s performance. Eventually, if you abuse the amp, its protective circuits should kick in—but some amps don’t use a lot of protective circuitry and expect you to behave yourself, so if you abuse them, and they’ll simply blow up. Having put out power-amp fires onstage in mid-show, I can tell you this is not a fun addition to your light show. Finally, keep in mind that there are several ways to rate power in an amplifier, including continuous power (the long-term average heating power with typical program material), program power (maximum average levels over the medium term, typically up to a minute), and peak power (calculated for short-term peaks, usually about a tenth of a second). For P.A.s, you generally want the program-power rating, preferably rated in watts RMS. WHEN ONE IS NOT ENOUGH Fig. 1. Series wiring, where the impedances are added together, makes the total impedance larger than any one of the speakers. When you put two speakers in a cabinet or wire two cabinets to the same power-amp channel, how they’re wired affects the total impedance presented to the amplifier. For example, a cabinet housing two 8Ω speakers can have a total rating of 4Ω, 8Ω, or 16Ω, and wiring multiple cabinets or speakers can get a mite complicated. Let’s take the fear out of dealing with a mishmash of speaker setups. If you have two speakers that you want to install into a cabinet, or you want to change the existing wiring, you can hook them up in parallel, series, or series-parallel. (The same rules apply when driving more than one speaker cabinet with one power-amp channel.) Figure 1 shows how you connect up the terminals of the speakers to produce series, parallel, or series-parallel configurations. Each scheme yields a different total impedance. Let’s look at each. Series. In a series setup, you merely add the impedances together. I know we said we’d keep the math to a minimum; this is easier than it looks. Let’s assume we have an 8Ω speaker and two 4Ω speakers. For a series setup, the equation is simple: Z1 + Z2 + Z3 = Z So in our example: 8Ω + 4Ω + 4Ω = 16Ω Thus, if we wire the speakers in series (see Fig. 1a), our power amp will be dealing with a 16Ω system. Parallel. If we wire the speakers in parallel, and all three speakers have the same impedance, the formula is easy: the impedance of one speaker divided by the number of speakers. So if we had three 8Ω speakers wired in parallel, the equation is 8 ÷ 3 = 2.667Ω. If our power amp can handle 2Ω loads, that should work fine, but if the amp is looking for a 4Ω load, this is going to make our amp work awfully hard. With an amp designed for an 8Ω load, this system is going to be bad news. If the speakers are of varying impedances, things get more complicated. This looks scary but it is easier than it looks: Z=1/ 1/Z1 + 1/Z2 + 1/Z3 … In our three-speaker example, that gives us this: Z= 1/ 1/8 + 1/4 + 1/4 = 1/ 5/8 = 8/5 = 1.6Ω Fig. 2. Parallel wiring, where the total impedance will be less than any one speaker. Fig. 3. Series-parallel wiring, where the impedance will vary, according to which speakers are in series and which are in parallel. Series-Parallel. Calculating the impedance for a combination of series and parallel wiring is just a matter of applying each equation as needed. Since we have three speakers in our example, we can do this two different ways. Let’s see what happens if we wire our two 4Ω speakers in parallel, and that combination is then wired in series with the 8Ω speaker. First, we calculate the combined value of the parallel speakers. Since we have two identical values (4Ω) in parallel, we can take the easy way out: 4 ÷ 2 = 2Ω Now, let’s put that 2Ω system in series with our 8Ω speaker, which calls for simple addition: 2Ω + 8Ω = 10Ω total If our power amp is designed to work with 8Ω or lower impedances, 10Ω is a low load, though not out of the question. Our system won’t be terribly efficient, but it probably will work. If the amp is designed for 4Ω or less, we’re going to waste a lot of power with this system. Instead, let’s wire one 4Ω and one 8Ω speaker in parallel, then wire that combination in series with the other 4Ω speaker: Zparallel = 1/ 1/4 + 1/8 = 1/ 3/8 = 8/3 = 2.67Ω and Ztotal = 2.67 + 4 = 6.67Ω That will probably be close enough for an amp that can handle 8Ω loads, and it’s fine for amps that are designed for 4Ω loads. One final tip: If you’re in doubt, wiring in series always results in a greater impedance, and while it might be less efficient, it’s the safest way to go.
  7. Direct boxes aren't just for recording bass - check out these tips for guitar By Jon Chappell Here are two different scenarios involving guitar amps and recording, both with a common problem that can be solved with direct-box technology. We don't normally think of direct boxes for recording guitars (that's an onstage bass thing!), but it does come up, especially in the following two situations. Scenario 1: You have a great vintage guitar and amp, but the amp has no master volume, so you really need to crank it to get the quality you want. The sound is so flippin' loud that it overwhelms your monitoring system, so you decide to put the amp in another room (down the hall, in the basement, etc.) and mic it, running the mic cable back to the recorder (where you are). You use headphones to monitor the miked sound, and the speaker volume doesn't shake you out of your chair. Scenario 2: You're in a proper recording studio, with a separate live and control room, and you'd like to sit in the control room with the amp beyond the glass, because it's easier to run the session and talk to the engineer (you're doing a bunch of guitar overdubs). "No problem," says the engineer, "we'll put some close and ambient mics in the live room, and monitor you through these giant control-room monitors so you'll hear a full sound." In both these situations, the guitar is separated from the amp by a considerable distance, and that means running an inadvisably long patch cord (greater than 30 feet) between the guitar and the amp. This isn't a good idea under any circumstance, but especially if you have passive or low-output pickups (as most vintage models are). A neat way around the limitation of recording from the control room with a combo in the studio is to use Radial Engineering’s SGI (Studio Guitar Interface) system. These relatively inexpensive, no-frills, passive (non-powered) gizmos convert your guitar’s high impedance signal to low impedance and transports it to another box that converts it back to high impedance. Here's how it works. The guitarist plugs a cable from the guitar into the first direct box, the SGI-TX (tx is the symbol for "transmitter"). Then the low-Z (three-pin) out is connected via a mic cable of whatever length (say 40 feet) is necessary to a second direct box. (Radial guarantees tone integrity up to 100 meters or 328 feet.) Because the signal has been converted to low impedance, it can travel a long distance without suffering degradation. The signal meets up and connects with to the low-Z input of the second Radial box, the SGI-RX (receiver). The second box should be placed right near the amp. Fig. 1: Covering long distances with Radial Engineering’s SGI system--essentially two complementary-function direct boxes. The image in the foreground middle is a schematic showing the connections. Then the high-Z, 1/4" output is used — along with a normal patch cord — to connect the second direct box to the amp, so that the amp's input gets the high-Z signal it's looking for. Fig. 1 shows the direct-box schematic in the foreground with a scale-looking illustration of how the setup looks in the recording studio. Many studios have "tie lines" or wired jack plates on the walls that can get you into and out of the room you're in, but if you're at home, or in any other improvised environment where separating the guitar from the amp is advantageous, just the two boxes will do the trick. Jon Chappell is a guitarist and Associate Editor at Harmony Central. He has contributed numerous musical pieces to film and TV, including Northern Exposure, Walker, Texas Ranger, All My Children, and the feature film Bleeding Hearts, directed by actor-dancer Gregory Hines. He is the author of The Recording Guitarist: A Guide for Home and Studio (Hal Leonard), Essential Scales & Modes (Backbeat Books), and Build Your Own PC Recording Studio (McGraw-Hill), and has written six books in the popular For Dummies series (Wiley Publishing).
  8. Where the Natural Meets the Fractional By Jon Chappell Natural harmonics on the guitar—the ones found on open strings by laying a left-hand finger lightly over a fret—are a great weapon in the arsenal of a performing and recording guitarist. Often a well-placed harmonic is just the thing a sustained note needs at the climax of a solo. On overdubs, they make nice punctuation points when applied judiciously, and can add a nice splash of color. But when you’re in the studio, the pressure can be on to hit the harmonic the first time, or to hit it repeatedly, consistently, and with good, ringing tone. For that kind of situation, you can’t leave it to chance or experimentation. You have to know where the harmonic falls exactly so that you can nail it take after take. And if you actually want to be facile with adding harmonics on overdubbed solos on a regular basis, you need to develop a system for identifying and playing them quickly. Transcriber, Help Thyself I originally charted that all the natural harmonics on all six strings, because it used to facilitate transcribing solos when I worked as a professional transcriber and music editor. When figuring out solos, I could hear the note clearly, but since I couldn’t see the guitarist’s hands, I always had the hunt and peck for the right string and the right fret. Calculating harmonics from scratch got very tedious very quickly (“reinventing the wheel” was the analogy that often sprang to mind”). So one day I just took time out and made myself a chart that had all the natural harmonic possibilities for all six open strings. Now when I hear a natural harmonic on a recording—or I want to produce one of my own—I look at my chart. Find Your Fret In addition to making transcribing a whole lot more efficient (especially for Steve Vai and Joe Satriani solos!), I found knowing the location of all the natural harmonics helped my own playing. For example, if I were soloing in A, and I knew a high A would be just the thing to nail at bar seven of my eight-bar solo, I could quickly scan my chart and see where all the available A’s were. Or, if I was playing on a floating-bridge guitar, I could hit a G harmonic and pull the bar up a whole step. Check out the chart in Fig. 1, which shows the all the harmonics on all six open strings. Notes the fractional-fret harmonics. These occur not directly over the fret wire, but at the indicated distance between two frets. For example, fret 1.6 is a little more than halfway between frets 1 and 2. These are harder to play, because their node points are much narrower than the “strong” harmonics on frets 12, 7, 5, and 19, but with practice, you can get a pretty full tone consistently. Fret Out Some harmonics are easier to play than others. Of course, any non-fractional fret will be easier than a fractional one. And the closer to the 12th fret you are, the easier the harmonic is to invoke. So if you want to produce A5, you’ll get better results from the second string on fret 5.8 than you will on the fifth string, fret 2.4. (This is where the chart comes in handy.) The exciting thing is that once you have this “menu of harmonics” in front of you, you’ll create more and more opportunities for harmonics to appear into your music. Whole-Fret Harmonics Fret Octave + Interval Open-String Pitch* 12 1 E2 A2 D3 G3 B3 E4 7/19 1 + 5 B2 E2 A3 D3 F#3 B3 5/24 2 E3 A3 D4 G4 B4 E5 4/9/26 2 + 3 G#3 C#4 F#4 B4 D#5 G#5 3 2 + 5 B3 E4 A4 D5 F#5 B5 6/15/22 2 + b7 D4 G4 C5 F5 A5 D6 17 3 E4 A4 D5 G5 B5 E6 Fractional-Fret Harmonics Fret Octave + Interval Open-String Pitch* 1.6 3 + 5 B4 E5 A5 D6 F#6 B6 1.7 3 + #4 A#4 D#5 G#5 C#6 F#6 A#6 1.8 3 + 3 G#4 C#5 F#5 B5 D#6 G#6 2.0 2 + 3 F#4 B#4 E#5 A5 C#6 F#6 2.4 3 E4 A4 D5 G5 B5 E6 2.6 2 + b7 D4 G4 C5 F5 A5 D6 3.3 2 + 5 B3 E4 A4 D5 F#5 B5 5.8 2 + b7 D4 G4 C5 F5 A5 D6 *C3 = Middle C. The guitar sounds an octave lower than written. For example, the open high-E string sounds E3, the lowest line on the treble clef. Fig. 1. A chart showing the natural harmonics on all six strings of the guitar. The top part shows the whole-number frets where the left-hand finger is placed directly over the fret wire. The bottom section shows the fractional frets, where the finger is placed between two frets.
  9. USB Mics Are for Real, and the C01U Does the Basics By Craig Anderton At first, I thought the idea of a USB mic was a joke. Then when I actually saw the C01U I thought that it couldn't be any good, given the price (list is $234.99, but that seems pretty fictional—average street price hovers around $80). However, after using it for a while, I've gotten to the point where I don't want take my laptop on the road without it. The C01U is a USB version of the C01, but replaces the XLR out with a USB out (it would be nice to have both options, though I'm not sure where you'd put the extra jack). There are a lot of surprises: It feels more substantial than expected and has a real manual; it also comes with a mic holder, pouch, and 10' USB cable. Although it will install into Windows XP and Mac OS X without any extra software, if you go to www.samsontech.com, you can download a nifty driver applet (Mac OS X, Windows XP SP2) for extra functionality. (Note: Support in Windows 98 is limited; it works, but the gain range is restricted). The site also has detailed instructions on installing, deinstalling, and updating drivers. However, the installation instructions are needed only if you want to get an idea of what to expect, as the installation procedure itself takes you through the process in detail. Installation was simple: I plugged the mic in to a USB port. With either USB 1.1 or 2.0 Windows installs the drivers automatically, and once I set the volume properly using Windows' Sound and Audio Devices applet (very important—just because it's plug and play doesn't mean gain-staging doesn't matter), I was set. I tested the C01U with Windows XP in Cakewalk Sonar, Ableton Live, and also with Windows' built-in sound recorder, all using standard Windows sound drivers (MME, DirectX, DirectSound); everything worked perfectly. Suitably impressed, I installed the applet. It too was painless to install. Having the low cut filter was helpful, as was the ability to set gain more precisely…and a flashing clip indicator never hurts, either. Note that the C01U doesn't work with ASIO; also, the WDM drivers don't work with Sonar's ultra-low latency kernel streaming version of WDM. However, the C01U works fine with Sonar's MME drivers, and with "standard" WDM-compatible programs. As to the Mac, all is well as long as you're not using ASIO and you do proper gain-staging. Bottom line is that you shouldn't expect to plug the C01U into your favorite multitrack ASIO host. Otherwise the mic works fine, but of course, overall performance depends on how well the host itself works with compatible drivers. Note that the C01U also supports all common sample rates between 8 and 48kHz, with 16-bit resolution. But the sound was perhaps the biggest surprise: It's far better than I expected. This is no little "here's a mic for your laptop to take dictation," but a serious hypercardioid mic with a 19 mm diaphragm that gives a good account of itself with whatever you throw at it. You need to be a little careful about noise, but this relates to the usual mic issues—get a good level, set gain for the highest possible short of distortion, and noise won't be a significant factor. If I caught the sample of a lifetime on my laptop using this mic, I wouldn't think twice about transferring it over to my desktop and using it. The applet allows for some degree of computer control. THE BOTTOM LINE ASIO issues aside, this mic performs well for mobile audio applications. It's easy to understand intellectually that you don't need an audio interface, but that doesn't really hit home until the first time you plug a USB mic into your laptop, and lay down a scratch vocal in a moment of inspiration without missing a beat. What's more, although you're limited in the cable length you can use, you don't have to worry about balanced lines, low level signals, or other mic-centric issues. And for plug-and-play Podcasting, it's ideal. In a desktop situation, it's likely that if you're doing serious musical work you already have an audio interface with at least a mic preamp or two. Yet the C01U does add another mic to your arsenal, and when you're in a hurry, I can't emphasize enough how easy it is to just plug into a USB port and start recording. If you've been limping by with the (shudder) teeny little mic built into your laptop, or some $30 mic designed to grab sound rather than record it, the C01U will be a major step up. Craig Anderton is Editor Emeritus of Harmony Central. He has played on, mixed, or produced over 20 major label releases (as well as mastered over a hundred tracks for various musicians), and written over a thousand articles for magazines like Guitar Player, Keyboard, Sound on Sound (UK), and Sound + Recording (Germany). He has also lectured on technology and the arts in 38 states, 10 countries, and three languages.
  10. Not Enough MIDI Ports? Here's Your Solution www.tapcoworld.com by Craig Anderton I've noticed a disturbing trend lately in computer interfaces: No MIDI ports, or at best, one MIDI port (or something on the end of a bizarro world breakout cable). Granted, a lot of what MIDI used to do outside of the computer now occurs inside the computer, thanks to virtual instruments and other plug-ins. Yet there are still a lot of great MIDI hardware synthesizers, pedals, effects, footswitches, fader controllers, and other goodies that aren't happy unless they see that little 5-pin DIN connector. So what are you gonna do if you need more MIDI? Well, you know I wouldn't pose a question like that without an answer, and the answer is Tapco's LINK.midi 4X4 interface. It's been out for a while, but during that time it has become a workhorse in my studio, providing MIDI for Mac (PPC and Intel) and Windows computers. In fact, the main reason I'm reviewing this is for any other readers who suffer from lack-of-MIDIitis, because this little box has proven itself over and over again - and its four ports let it handle up to 64 channels of MIDI data. As usual, we won't dwell on the specs because you can get that from the web. So, let's just dive in to explaining why the LINK.midi 4X4 has filled a need in my studio, and why it might in yours as well. THE USB CONNECTION MIDI data uses a fraction of the bandwidth of digital audio, so it has no problem running 4 discrete ports with a USB 1.1 (or USB 2.0, for that matter) connection. Because it's not card-based, it's transportable among computers - any Mac with a native USB port and runs OS X 10.3 or better (except Macs with G3 or G4 accelerator cards), and any Windows XP machine running SP1 or better, will work with the LINK.midi 4X4. It also works with Vista if you update to the latest firmware; the Tapco site has the full story on updating. What's more, the LINK.midi 4X4 doesn't need a wall wart, because it's bus-powered. And, it's nowhere near as picky as a FireWire peripheral: It has worked with anything I've thrown at it. UPFRONT Looking at the front panel, you have three MIDI inputs and one MIDI output - very convenient for easy, front-panel access to gear (the other MIDI outs and MIDI in are on the rear panel). The front also has eight activity lights - green for input activity, red for output activity - and a MIDI switch thru so you can drive several sound modules from asingle master controller. That's pretty much it, except for two little handles. Given that the thing doesn't weigh much, these aren't for picking the unit up but provide protection from anything that might want to smash into its front - although the front panel is slightly recessed, so it already has a bit of an advantage in that respect. BRINGING UP THE REAR The rear panel is an equally simple affair, with the aforementioned MIDI outs and MIDI in, as well as the USB connector and a Kensington lock. It doesn't get much simpler than that. PHYSICAL CONSTRUCTION At less than 2 inches high and 7 inches square, the LINK.midi 4X4 isn't exactly Lilliputian but it's not going to get in the way, either. However, there are a couple other cool aspects to the packaging. First, the top and bottom have a rubbery, slight raised surface that makes it virtually impossible for this box to scratch your desktop. Second, it can lie horizontally, or you can pull out and swivel a little crossbrace that makes a "leg," allowing for space-saving vertical mounting. PLUG AND PLAY Indeed, it is. With Windows, LINK.midi 4X4 is USB class-compliant so it doesn't need any special drivers. I just ran the wizard and clicked mindlessly on "Next" until it was installed, whereupon it showed up in any program I had that dealt with MIDI. On the Mac, I just plugged it in and it worked (one of those times where the Mac's "It just works" slogan is actually 100\% true). I think the only way the installation process could be easier is if the LINK.midi 4X4 walked up to your computer and plugged itself in. CONCLUSIONS If you don't do MIDI, you don't need this. But if you use MIDI hardware, and your interface thinks you don't, LINK.midi 4X4 is a simple, relatively inexpensive solution that adds that extra MIDI you've been craving to pretty much any modern computer. It has never burped, acted anti-social, crashed, complained, or lost its mind. If only everything I used had the same qualities, life would be much, much simpler. Craig Anderton is Editor Emeritus of Harmony Central. He has played on, mixed, or produced over 20 major label releases (as well as mastered over a hundred tracks for various musicians), and written over a thousand articles for magazines like Guitar Player, Keyboard, Sound on Sound (UK), and Sound + Recording (Germany). He has also lectured on technology and the arts in 38 states, 10 countries, and three languages.
  11. Cross-platform guitar amp/cabinet/effects simulation software www.ikmultimedia.com www.native-instruments.com www.waves.com By Craig Anderton I've basically come to the conclusion that amp simulation software is like amps themselves: They're all different. I use all three of these sims for different reasons, as each one has different sound qualities and functionality. So, I'll be presenting some objective and subjective views on each one, and hopefully, they'll be helpful in deciding which one is right for you. FIRST THINGS FIRST Whenever I boot up an amp sim for the first time, my reaction is "This sounds awful!" That's because I call up a few presets, and there's the first problem: The presets were designed by people who weren't playing my guitar or using my pickups. Furthermore, many times presets are designed to show off what a device can do, rather than be part of an ensemble of instruments. But a little tweaking gets me where I want to go. Usually this involves adjusting any input gain or drive control to match my levels, then experimenting with different pickup settings to determine which one sounds best with the preset. Then I'll often scale back some of the settings, or disable some effects altogether, to clean up the sound. Think of presets as more of a point of departure than a final destination. Another issue is whether an amp model sounds exactly like an amp. Many guitarists report that playing through a sim doesn't give the same experience as playing through an amp, but when recording and playing back, it's very difficult (and in some cases impossible) to determine whether what you're hearing is a "real" amp or a simulation. Furthermore, there's the question of which amp you're simulating. Different production runs of amps had different sounds, and even the same amp could change over time as tubes age. But I'm not even sure if "precise duplication" should be the goal of an amp sim, as what they also offer is the ability to make sounds you can't get in any other way. So with these basics out of the way, let's dig a little deeper. SOUND QUALITY All these sims can make some great sounds, but each has their specialty. For example, in addition to the expected overdrive and crunch sounds, GTR has a sterling repertoire of clean and slightly dirty amp sounds. These are exceptional at giving straight guitar sounds some real character and presence. AmpliTube 2's special talent is that it simulates the transition from "clean" to "breakup" very faithfully. This is extremely difficult to simulate, and while GTR also does a good job, I'd have to give AT2 the edge on this one. Guitar Rig's strength is that all its amp models are very faithful (with a little tweaking, of course!) to the originals; I've heard some clips from Nashville guitarist Jerry McPherson where he recorded a miked amp and the same sound from Guitar Rig, and you really can't tell the difference. (Note that both GR and AT2 have "high resolution" modes that tax your CPU, but give a smoother, sweeter distortion effect.) FLEXIBILITY Guitar Rig is definitely the most flexible of three, as it uses a "rack-based" paradigm where you can throw modules in wherever and whenever you want. It also has unique signal splitting and crossover modules that really open up options for parallel and series/parallel effects routing. If you're a "tweak freak," GR can pretty much do whatever you want it to. Guitar Rig 2 uses a rack-based paradigm. You drag modules from the left side (the different distortion options are shown) into the rack on the right. Note the "Tweedman" amp module's "advanced" parameters along the bottom of that module. AT2 takes a different approach. Its setup is fixed: Stomp boxes always go before amps, and a complement of "studio effects" always follow amps—but within those limitations, there are eight distinct routing options that put the two stomp box "pedalboards" in series, in parallel, routed through amps that can be in series and parallel, etc. You can have one amp go through two cabs, two amps through two cabs in parallel, and the like. This strikes a good compromise for people who want some degree of flexibility, but don't want to be overwhelmed with options. GTR lies between these two. It has separate plug-ins for "pedalboards" and amps, so you can put the effects pre- or post-amp, stack one amp after another, and so on. If you want effects before an amp, followed by more effects, followed by an amp, followed by yet more effects, go right ahead. The one drawback is that with the current version of GTR, you can't save the entire combination as one preset: You need to save the pedalboard setups as presets, and the amp sounds as their own presets. I think the odds are good this will be addressed in a future version, so that you have the option to save complete presets. THE ROSTER OF EFFECTS Guitar Rig has the lead in sheer number, as several updates have occurred since it was first introduced, and each one added more effects. It also has many modulation options, such as a step sequencer, analog sequencer, and envelope follower. Additional effects you won't find in competing products are a JamMan-type looping device, eight different distortion stomp boxes, and a rich roster of dynamics control (two different compressors, limiter, noise gate, and noise reduction). There are 35 effects total, as well as two "tape recorders" that can record your riffs, or play back riffs. Another interesting feature is that most effects have an "advanced" mode, so that the main interface is familiar and non-threatening—but if you want to go deeper, you can do so by revealing additional parameters. This shows the AmpliTube amp window toward the top, and one of the stomp box "pedalboard" windows at the bottom. Each of the two pedalboards can hold a maximum of six effects. A sixth one is being selected here. AT2 has 21 stomp box effects, and these lean more toward vintage simulations. For example, their "EchoMan" nails the E-H Memory Man sound, and their Envelope Filter sounds deliciously vintage. But the "special sauce" here is a bunch of decidedly non-vintage post-amp "rack effects," which provide the higher-quality plug-ins you'd expect to find in a studio. You'll find 11 more effects, including digital delay, reverb, tube compressor, parametric EQ, stereo enhancer, etc. These add a bit of a "finished" sound to patches. GTR has 23 effects—and these are all derived from Waves' roster of effects, so they exhibit that "detailed" sound for which Waves is justifiably famous. They cover all the bases (distortion, dynamics, pitch-shifting, reverb, way, modulation, gating, etc.), and are an interesting combination of vintage meets precision. I find them very "musical" and like GR, being able to put effects pre- or post-amp adds a significant degree of flexibility. USEABILITY GTR is probably the least "user-friendly" in terms of creating sounds fast, because of the need to load separate stomp box and amps. However, that can also turn into an advantage once you've had the program for a while. I've built up a nice collection of stomps for specific purposes (a very useful compressor/EQ combination, for example, and a tempo-synched delay/chorusing effects that sounds just plain beautiful), so you end up with a bit of a "Chinese restaurant" effect: One from column A, one from column B, load up amp—done. This GTR setup shows the amp module in the upper left, and two different stomp box pedalboard setups. Unlike AT2, you can have an unlimited amount of pedalboards, with 2, 4, or 6 effects per pedalboard. For instant gratification, AT2 comes out ahead. It has a bunch of useful presets that show the range of the product, but are also very useful. It's simple to operate and tweak. GR2 is somewhere between the two. It comes with a ton of presets, and while many of them are excellent and useable "right out of the box," there are also quite a few "educational" presets that show off specific techniques. These are ideal for veteran tweakers, as they provide excellent starting points for truly innovative sounds; but you'd be less inclined to just "punch 'em up and go" on a session. THE BOTTOM LINE I'd characterize GTR as having the sweetest, least "computer-like" sound, but it's also the most expensive. AT2 has a similar sound quality, although it exhibits a somewhat more "aggressive" edge (notwithstanding that superb clean-to-breakup transition). GR is without a doubt the most versatile of the three, and while some feel the amp sounds aren't as rich, I think those are probably people who haven't switched to high resolution mode—it makes a huge difference and is worth the CPU hit (AT2 also has oversampling and high-resolution modes that are worth enabling). It's almost like the difference between a tube and solid-state amp. For sonic explorers who want to get guitar sounds no one has gotten before, GR is by far the best option. It can do standard guitar amp sounds, but also delivers some exceptionally esoteric and synthetic sounds. Neither GTR nor AT2 offer that kind of versatility. If sound quality is of paramount concern, GTR has the edge. Whether that edge is worth the extra dollars (and iLok authorization) is something only you can decide, but the clean amps are in a class by themselves—and the dirty ones are almost like "high definition" dirt. If your priority is instant gratification and an authentically vintage sound, then AT2 is probably your best bet. It may not be as flexible as the others in some respects, but IK has done a fine job of deciding which functions are most crucial to make sounds, and made them accessible. So now you see why I use them all! Each has its own set of talents and occupies its own unique niche. While that may make purchasing decisions difficult, the good news is that each one is intelligently designed, and regardless of which one you end up with, you'll be able to make some exceptionally cool sounds. Craig Anderton is Editor Emeritus of Harmony Central. He has played on, mixed, or produced over 20 major label releases (as well as mastered over a hundred tracks for various musicians), and written over a thousand articles for magazines like Guitar Player, Keyboard, Sound on Sound (UK), and Sound + Recording (Germany). He has also lectured on technology and the arts in 38 states, 10 countries, and three languages.
  12. Gibson Gambles...Let's Check Out the Payoff By Craig Anderton When Gibson announced it would be introducing a digital Les Paul, people didn't quite know what to make of it. Initially, Gibson's "digital guitar" was mostly defined by what it was not: It wasn't going to be a modeling guitar like the Line 6 Variax, or a Roland-style guitar synthesizer. Some people even thought it wasn't that great an idea, that the price was out of line, and that guitarists wouldn't "get it." Despite the naysayers, though, I was intrigued. Sure, the idea of providing a separate audio output for each string wasn't new, but it seemed like an idea whose time had come. Thanks to better technology that could be applied to the pickups, multi-input audio interfaces, computer-based hosts, and plug-ins, putting together a sophisticated hex processing system for guitar could be done with a few judicious mouse clicks - not endless re-patching of costly outboard gear. Being able to apply different processing to different strings, or groups of strings, as well as explore a variety of panning options (including full-blown 5.1 surround; maybe it's not just coincidence that 5.1 accommodates six strings!) seemed pretty cool. I was curious to get my hands on "the Les Paul Digital Guitar" and check it out. So I waited as the release date kept getting pushed back, and pushed back, and pushed back some more. Apparently the task of making something that plays like a Les Paul, looks like a Les Paul, and feels like a Les Paul?but is equally at home in a hybrid analog/digital direction?is not a trivial undertaking. For starters, the hex pickup is very different from a standard pickup; it achieves 90dB of dynamic range, and provides surprisingly good freedom from crosstalk between strings. Furthermore, the performance is highly dependent on the relationship of the pickup to the strings, so Gibson had to add a computer-controlled assembly step to make these fine adjustments. (Interestingly, the string set is also unique, as the gauge and tension are chosen specifically to produce a uniform output level for each string). The HD.6X-PRO looks and feels like a Les Paul, but there's a lot going on "under the hood." And, not only was a guitar being developed, but a digital network to go along with it. And something like this has to be right the first time, because you're not going to get a second chance. In fact, some people said the digital guitar was just too complex, and would never come out . . . but they were wrong. It's here, and it's real. And it works. Really works. Let's get the price out the way first: At $4,999 list, it's expensive. Granted, it looks gorgeous (love that metallic blue look), and the package is classy: Spiffy case with roller wheels, the inclusion of Sonar Producer Edition version (the full version, not a lite or limited variety), a Hosa eight-cable snake to handle the multiple outputs, ethernet cable, and a breakout box. But given the cost, you can pretty much bet that you won't see it being cradled in the hands of the rhythm guitarist in your local bar band at the Holiday Inn. Nonetheless, if you have any interest in guitar, it's well worth delving into what this all about. Like so much technology, it wouldn't surprise me if the underlying concepts work their way into lower-cost models in the years ahead. But here's the real bottom line: I predict that pretty soon, you'll be hearing some outrageous guitar sound on some recording, and go "What the...?!?" And it will be the HD.6X-PRO. IT'S A LES PAUL... If you want, you can ignore the electronics entirely and treat the HD.6X-PRO as a standard Les Paul guitar. (There are a couple differences that help playability, though: The guitar fret board is wider than a stock Les Paul?equivalent to the width of a bound fret board?and the frets are a shade lower.) It has a mono output jack for "Classic Mode," two humbuckers, and the usual complement of controls. This is also good news if your breakout box with the electronics (more on this later) gets run over by a PA stack: The show can still go on. Having this option also brings up the possibility of layering the standard sound with what we'll cover next. ...IT'S NOT A LES PAUL Well it is, but I've never seen a Les Paul with a hex pickup, CAT-5 cable jack, a headphone out, a mic input, or outputs for each string. This is all made possible by the "MAGIC" network, which exists on an ethernet-type cable that patches to the guitar (although it's the same type of cable used for ethernet connections, technically the protocol is not the same as ethernet). The jackplate has (from left to right) a volume control for the headphone jack, headphone jack, MAGIC network port, mic in jack, and standard mono out for "classic mode." Not only does it carry the individual outputs from the strings, but it also carries the output from the guitar's mic jack, and sends monitor signals to the guitar, which ultimately end up at the headphone jack. The idea of having a headphone and mic jack on a Les Paul might be off-putting to purists, but those who use in-ear monitors and headset mics will likely find this addition a real plus. Note that the headphone jack is most emphatically not there to let you listen to your guitar through headphones as you practice; it has a much loftier goal. The cable terminates at BoB, the Breakout Box. This has jacks for the standard mono output, six separate jacks for each string, and two jacks for stereo groups of strings (1-2-3 and 4-5-6; these use the same jacks for the top two strings, so these jacks serve double duty). You'll also find a jack that carries the mic out, and another jack pair to receive the monitor signal of your choice. The Breakout Box takes the signals from the MAGIC network, and breaks it out into various analog outputs. It also has two ins for monitoring, and a mic out. Surprisingly, there's no way to connect BoB to your computer digitally: It's all done via analog connections. This is to be expected for live use, where you may want to feed six separate processors, or go to a mixer for concert hall surround effects. But given the digital nature of computers, it's unfortunate there's no way to bypass the analog electronics. However, a company representative has said that Gibson is working on ways to add a USB interface, which would address the digital interfacing issue. That said, if you want to take full advantage of all the guitar's outs, you need an analog interface with at least seven inputs to handle the "classic" out and six individual outs. Of course, you're not obligated to use all these; you could use two inputs for the two string pairs, or three for the two string pairs and the mono out. If you're expecting perfect isolation between strings, it's simply not possible. At first I thought it was a limitation with the pickups, but eventually realized that guitars are resonant little suckers, and hitting one string will, by nature, make the other ones vibrate a bit. With lots of gain (e.g., distortion plug-ins), you can pick up a bit of crosstalk but in some ways, I like this because it makes for a less "clinical" sound. In any event, the crosstalk is much lower than I would have expected, so kudos to Gibson for that; they clearly put some serious effort into the pickup design. OKAY... SO WHAT? I'll tell you what: Guitarists can now do sonic techniques very similar to what keyboard players do with splits and layering, as well as surround effects. Of course, the first thing I tried to do was pan all the strings across the stereo field from left to right, but it sounded overly-gimmicky. It's when I started thinking things through a bit more than things really started to pop. The hex pickup, placed unobtrusively between the bridge and bridge pickup, provides the six individual outputs. For example, one very cool sound was running the guitar outputs into Sonar and inserting the Waves GTR octave divider on strings 5 and 6. With a little tweaking, I got a bass sound that was very much like that tough, distorted bass sound for which Nine Inch Nails is justly famous. Meanwhile, I distorted the top four strings (but remember, there's hardly any intermodulation distortion), chorused them independently with Waves choruses, and spread them a bit across the stereo field. The combination of the rich, shimmering high end, coupled with the raunchy bass, was dramatic to say the least. When I played this against a drum loop (see later for links to audio examples), it sounded like I really didn't need any other instruments?the guitar just took over the soundstage, and wouldn't let go. Nor did I want it to... I then tried envelope filters on the strings, as well as flangers, amp simulators...you name it. One of the effects that surprised me most was delay. I'm so used to hearing delay on the entire guitar that it was very different to hear it just on the top two strings, with the other strings providing a clear, direct sound. I'd go so far as to say that where the HD.6X-PRO really shines is for rhythm guitar parts (power trios, take note). Lead parts tend to be single-note lines anyway, so breaking those leads into individual strings has less to offer than being able to turn a rhythm guitar into a "symphony" of guitar sounds. The "split" approach makes it easy to play leads on the upper strings, while hitting chords on the lower ones with entirely different tonal characteristics. Then I brought out the "big gun": Multiple instances of Native Instruments' Guitar Rig 2. Once you start applying effects like tempo-synched, step-sequenced filters to the guitar, you cross over from rock into a much more "techno" kind of zone. It's compelling, to say the least, and a fertile field for experimentation. Bottom line is you can do synth tricks, but retain the guitar's characteristic organic sound...think "polyphonic AdrenaLinn" (come to think of it, processing each string through an AdrenaLinn would probably be pretty astonishing). Layering is another "killer app." Take the mono output, feed it into a guitar amp, dial in a great guitar sound, and layer it with the other strings. At this point, it sounds like you have an army of guitar sounds, and you can imagine the possibilities. Surround? Well, I haven't gone there yet, because I don't have my studio set up for surround. But I have heard several surround demos at trade shows, and it's clear this represents a whole other world. (Perhaps this is one of the reason why the HD.6X-PRO includes Sonar Producer Edition, as it has a really good surround implementation.) BUT WHAT WE REALLY NEED ARE AUDIO EXAMPLES! Words ultimately fail in describing this guitar, because you can't imagine what it does if you haven't heard it in action. So take a listen; here's the story on the clips. This is the "NIN bass meets Andy Summers chorus" I alluded to earlier. It's a tough, gritty sound that shows some of the guitar's "split" capabilities. This reminds me a lot of the old Roland GR-300 sound. This was obtained simply by plugging in AudioDamage's FuzzPlus 2.2. into Sonar's FX bin for each string. This is just a pretty sound with chorus on each string, as well as delay. This is variation on the above, with a little more resonance added to the modulation. A little synchronized delay can really add "animation" and interest to the sound. This is a fairly complex patch. All six strings feed into an aux bus that terminates in a Guitar Rig 2 distortion sound. The bottom three strings feed a second aux bus that terminates in another instance of Guitar Rig 2, loaded with a synth-type filter and an "analog," tempo-synched step generator. Finally, a little bit of the straight sound of the bottom three strings is mixed in for a little more definition. This is actually just a simple hex power chord, but processed within Sonar to add enveloping/synched effects. Each string has a resonant filter with slow LFO, and they're all summed together to make a big, moving, animated filter sound. Now is my enthusiasm starting to make sense? (Drum loops courtesy of Discrete Drums.) THE BOTTOM LINE The HD.6X-PRO isn't for everyone. First, you have to afford it. Then, you need a pretty sophisticated setup if you want to exploit the guitar to its fullest potential, regardless of whether you plan to use it with a computer as the centerpiece of your studio, or live as a multichannel guitar. And that's still not enough: You have to work with it enough to adapt its potential to your playing style. It's one thing to slap on a bunch of processors, pan the strings all over the place, and create something pretty garish. It's another thing to tame the beast and make it do your bidding in a musically appropriate way. But when you do, I think you'll discover - as I have - that this is not an instrument to be underestimated. You can go places you've never gone before, and they're fun places; perhaps more importantly, they range from beautiful, tranquil spaces to regions filled with snarling dogs and erupting volcanoes. I was delighted. And if you get a chance to play the HD.6X-PRO into some presets that exploit this instrument to the fullest, I predict you will be too. Craig Anderton is Editor Emeritus of Harmony Central. He has played on, mixed, or produced over 20 major label releases (as well as mastered over a hundred tracks for various musicians), and written over a thousand articles for magazines like Guitar Player, Keyboard, Sound on Sound (UK), and Sound + Recording (Germany). He has also lectured on technology and the arts in 38 states, 10 countries, and three languages.
  13. Vintage sounds and modern hardware hit the stage 73-key MSRP $2,700, street $1,999; 88-key MSRP $3,000, street $2,199 www.korg.com By Craig Anderton I’m glad the days of the “race to the bottom” in keyboards is over. Companies like Korg, Yamaha, Roland, Nord, etc. are building serious, stage/recording-worthy keyboards with excellent build quality and generous feature sets. They’re priced accordingly, but these are products with “legs” that aren’t designed to be flipped every six months for some shiny new plastic thingy. Which brings us to the SV-1 Vintage Stage piano. It’s almost anti-climactic to do a review because if you were at AES, it was obvious that everyone playing the SV-1 loved it for one reason or another. Having now had a chance to use it in my own studio, I can confirm that those initial impressions were on the mark. Still, it’s worth covering the SV-1 in depth because with all the keyboards out there, it’s important to know whether the SV-1 is for you or not. THE BASIC CONCEPT Main view of the keyboard. It's not as heavy as it looks! The SV-1 is all about future retro. Retro, in the sense that it re-creates the sounds of piano, tine pianos, organ, clav, string synths, etc. – 36 sounds altogether, with additional variations. Future, because it uses today’s technology to the fullest, adding amp modeling, reverb that’s way better than that funky Danelectro spring unit I used back in the day, programmability, and a keybed with Korg’s excellent, weighted RH3 action (with 8 velocity curves). Both 73- and 88-key models are available, and while not exactly lightweight, they aren’t too hard to carry around at 38.5 and 45.3 lbs., respectively. The 73-key model has a sort of muted red color scheme with black trim, while the 88-key model is black with copper trim. The front panel has a logical layout, with different functions separated into obvious blocks of controls. I get the impression that Korg’s engineers were forced to design the interface while in a darkened club with bad stage lighting, because you can be semi-conscious and still work the controls. Operationally, the Big Deal here is that there are very few hidden functions; the front panel is always “live” and available for tweaking. (There’s also a cross-platform editor/librarian that reveals many more parameters, and provides more details on certain parameters; we’ll get into that later.) The few functions that are less intuitive to access involve setup-oriented aspects, like choosing velocity curves or tunings. The knobs have LEDs around them, but these aren’t high-resolution “rings” like you see on some gear. Rather, they indicate various switched positions. When you call up new sounds, all the LEDs indicate current control settings, which you can of course change. However, even though only a subset of settings is shown in the front panel, several of these spread values across a full 128 value range, which you can access with the editing software. CHECKING OUT THE SOUNDS Before getting into the details, I’m sure you want to hear what it sounds like. Normally I’d record some audio examples, but Korg has been very diligent about posting audio examples on their web site. Several of them are played by Greg Phillinganes, who’s a - uh - slightly better keyboard player than I am; if you want to hear the sounds, he does a fine job of presenting them. There are another 36 examples from the built-in demos (one for each sound), and Korg has indicated they plan to add more examples in the future. SOUND SELECTION Choosing among the 36 basic sounds is easy: You select one of six main sounds, and one of six variations on that main sound. There are six main sound categories – two electric pianos, clav, piano, organ, and “other,” each of which offers six sounds within that category. For example, one electric piano variation might include modulation, another one have strings layered along, etc. The “Other” category is one of my faves: There are three string synth sounds, choir with subtle strings, analog brass, and the analog type of synth sound that powered Van Halen’s “Jump” (with the latter two, it sounds like the SV-1 captured the soul of an Oberhim OB-8). But what’s most important is that the sounds are gorgeous: Clean, well-defined, rich, and lush. I don’t think anyone could find fault with them. Korg definitely got it right (which probably isn’t too surprising to anyone who’s checked out their latest instruments). As one friend said when I clamped headphones on her, “It’s so clear! Also in the sound selection area, you can transpose plus or minus 12 semitones by engaging the transpose function and hitting a key that corresponds to the amount of transposition you want, referenced to C4. Simple. This is also where you can choose from one of 8 tunings (including stretch tunings), one of 8 velocity curves, and whether to turn Local Off if you want to drive an external MIDI tone module from the keyboard. THE EQUALIZER The EQ offers three bands of boost/cut. The equalizer is not about surgical sound-sculpting: There are three bands (bass, mid, treble) that provide gentle (low resonance) tone-shaping, which given the sound quality is really all you need. If you do need more, the editor software provides access to the Mid frequency. Like all other processing sections, there’s an on/off switch to enable/disable the effect. PRE FX The Pre FX section adds effects prior to the amp section. This section includes six effects that Korg considers as “stompbox” effects: Compression, boost, “Univibe,” vibrato, tremolo, and Vox Wah (which can follow the amplitude envelope, or a pedal plugged into one of the rear panel pedal jacks). Boost is a lot of fun; it’s not just a gain boost, but adds a sort of “hardness” to the signal. All PreFX are mono, except for the Tremolo; however, you can often restore stereo image with the Modulation and Reverb effects, which are “downstream” in the signal path). I feel the Wah works best with the pedal, because there’s not enough release time on the decay for my taste. As a result, if you hit complex chords, the wah effect “flutters” as it decays rather than having a smooth decay. Having designed envelope-controlled filters, I know this is a design choice where you have to choose the “sweet spot” between having too fast a decay (where there’s excessive fluttering), and too slow a decay, where there’s a sort of lag and lack of “tightness” to the sound. That said, I would have lengthened the release time a bit. This doesn’t apply to when you’re using the pedal. These effects all have speed and intensity controls, but they’re set-and-forget – there’s no mod wheel for changing the intensity of the vibrato or tremolo in real time. I’d like to see a future software update that allows tying these in with the pedal. AMP MODEL When you want to grittify your sound, the amp model section is glad to oblige. This section has six amp models, and is where the Real Tube comes into play (aha! so that’s why they sound like tube amps!). You can see the tube, glowing away to the left of the Master Volume control. Controls are minimal: choice of amps, and Drive. Drive is the one place where I wish the control had more than 10 front panel steps of resolution, but it’s not a major drawback at all, as between the amp and Drive options you can pretty much dial in any degree of grit you want. And if you can’t, there’s always the software editor (I detect a pattern here…). FAVORITES Store custom variations on the included presets in the 8 memory slots. The Favorites buttons are all about saving variations on sounds that are easy to punch up on stage, or storing some of the 36 sounds in a particular order for fastest access. The buttons are big, with big number labels (come to think of it, the whole front panel is quite readable), once again emphasizing that Korg expects you to be in a club with bad lighting when you rev up this puppy. MODULATION EFFECTS The Modulation FX section adds those swirling, liquid sounds that are such a good match to so many vintage keyboards. This section has two phasers, two choruses, flanger, and rotary. The flanger doesn’t go “through zero” like real tape flanging, but has a good modulation curve that gives a smooth flanging effect. The Rotary Speaker includes a slow/fast switch that speeds up and slows down when you change positions, just like the real thing. As with the Pre FX, you have Speed and Intensity controls but as mentioned previously, no mod wheel to which you can tie Intensity. The two choruses and rotary effects are stereo; the others are mono. REVERB/DELAY Add some final ambience to the sound with the reverb or delay. There are four reverb choices (Hall, Plate, Room, and Spring), along with Tape Echo and Stereo Delay. The natural reverbs are excellent, so much so you won’t be wishing for an outboard reverb; the Spring is little too periodic for my taste out of the box (springs weren’t always that periodic), but there’s always the editor. The Stereo Delay has a tap tempo option, but curiously, it doesn’t affect the Tape Echo. I’d vote for Korg to make the tap tempo apply to whichever delay is selected. The only front panel control here is Depth, which varies the mix of the processed sound with the straight sound. INS AND OUTS USB, pedal connections, and physical MIDI connectors are all available. The I/O is pretty comprehensive. There’s a USB connection for interfacing with the editing software that also provides a virtual MIDI interface (e.g., for driving the SV-1 from a sequencer), and jacks for three pedals. A damper pedal is included with the SV-1; the other two jacks handle a footswitch and expression pedal that can work either the volume or wah. The footswitch and pedals are optional at extra cost. Output options are balanced XLR or unbalanced 1/4”. As for audio, there are stereo inputs designed for feeding in the output from a CD or MP3 player for jamming or accompaniment. There’s no volume control for this on the SV-1, so you’ll need to use controls on the devices themselves. This signal is fed in post-effects, so it passes unmodified to the output. There are two unbalanced 1/4” outs (use one for mono), and two parallel, balanced XLR outputs. The XLR outs aren’t combo jacks, which makes sense – you could send the balanced outs to a mixer, and the unbalanced to something like a miked amp for extra “character.” SV-1 EDITOR The SV-1 Editor software not only performs backup and similar functions, but can access a wealth of parameters that aren’t available from the front panel. The SV-1 Editor is a really useful add-on that works on both Mac (PPC or Intel) and Windows XP or Vista. It looks very cool, and is just as easy to figure out as the SV-1 itself. It also allows modifying the 36 presets and Favorites, which you can then save to build up a collection of patches, or to swap patches with other SV-1 users. Perhaps the most important element of the software is that you can access several parameters that you can’t tweak from the front panel. As you can see from the screen shot, the Compressor reveals Attack, Level, and Sensitivity controls, and the Reverb offers four additional parameters. Furthermore, amps and cabs can be chosen independently, and each has more descriptive names so you can get a better idea of what Korg chose as the source for their models. Another very useful feature is that there are controls for taming any “noise” elements included in a patch, like the noise found with string synthesizers, or the key clicks of an organ. This gives you the option of going for the most authentic sound, or cleaning it up somewhat. If you’re not a tweakhead, you don’t really need to use the SV-1 editor; but those who want to get the most out of the SV-1 will find the editor not only very useful, but exceptionally easy to navigate. CONCLUSIONS The SV-1 is a truly solid performer. I would like to have a mod wheel, and some of the knobs wobble a bit so I was careful not to abuse them, but those are relatively minor complaints. Some people will want more than 8 favorites (it would be nice to double that by using a “shift” key, like holding down Tap Tempo while selecting to get another eight sounds), although the 36 included sounds pretty much hit the mark as is, and are easy to dial up. However, remember that the SV-1 is not targeting synth nerds, but gigging musicians who want really great sound quality, portability, ease of use, piano-type keyboard action, a transparent user interface, and the ability to re-create the sounds of the vintage keyboards of yesteryear – albeit with a more delicious, polished sound. The detail of these sounds is really quite remarkable; for example, it’s always difficult to get the dynamics right on electric piano sounds, but the SV-1 pulls it off whether doing tine, reed, or electronic-based types. The editor/librarian software is a nice touch, as it lets you create a pool of patches that you can blast over as favorites (for example, you might load in a different set of sounds when you’re doing that wedding gig than you would if you’re opening for Herbie Hancock). If I had to choose one word to describe the SV-1, it would be “satisfying.” The look, feel, and sound all add up to a playing situation where you really feel connected with the keyboard – something that was common in the days of vintage instruments, but which sometimes gets lost in instruments with tons of menus and options. Playing it compares to having a really great meal at a restaurant as opposed to just whipping up a frozen dinner at home, even though both fall under the category of “food.” There’s an experience of playing the SV-1 that goes beyond just hitting keys to trigger sounds. The SV-1 doesn’t replace a digital audio workstation/synth like the Korg M3; this is all about a real-time, stage-worthy instrument that’s fun to play and hear. In terms of accomplishing that goal, the SV-1 hits a bulls-eye. Craig Anderton is Editor Emeritus of Harmony Central. He has played on, mixed, or produced over 20 major label releases (as well as mastered over a hundred tracks for various musicians), and written over a thousand articles for magazines like Guitar Player, Keyboard, Sound on Sound (UK), and Sound + Recording (Germany). He has also lectured on technology and the arts in 38 states, 10 countries, and three languages.
  14. Want to get into MIDI guitar? Now you can do it without breaking your budget www.sonuus.com By Craig Anderton The G2M is compact and battery-powered. Back in the 80s, MIDI guitar was supposed to be the Next Big Thing: What guitar player wouldn’t want to be able to play anything from pianos to trumpets to ambient pads from that familiar six-string interface? Unfortunately, the question that wasn’t asked was “What guitar player wants to give up the expressiveness of a guitar and modify their technique in order to hear sounds with delays and glitching?” The answer was “Not as many as the industry hoped, by a long shot.” However, lots of guitarists did embrace the options MIDI gave them; and while MIDI guitar never broke through on a huge level, Roland kept the flame alive with a series of products that made continuous, incremental improvements in tracking, sound quality, and flexibility. Quite a few studio musicians also took advantage of MIDI so that when the producer said “Sure wish I could put a cello part here,” they’d whip out their MIDI guitar and synth, and deliver the goods. MIDI guitar isn’t about replacing guitar, but supplementing it with new choices. And now Sonuus, a small company out of the UK, has delivered MIDI for the masses: The G2M costs under $100 street, but provides true guitar-to-MIDI conversion. The catch? It’s monophonic, not polyphonic, so forget about chords. However, many would argue that where guitar excels as a MIDI controller is indeed single-note lines. Trying to play piano from a guitar can be awkward; playing a sax sound from a solo line makes a lot more sense. Just remember that MIDI is constantly butting up against the laws of physics. Keyboards are controlled by switches, and a guitar string is anything but a switch – trying to convince it to be one is not easy. Yes, you do have to play cleanly; yes, you may have to modify your technique (although there are MIDI guitarists who say that working with MIDI guitar has made them better, more accurate guitar players). But can a box smaller than a pack of cigarettes give a satisfying guitar-meets-synth experience? Let’s find out. THE BASICS You can get the specs from the web, so as usual, we’ll concentrate more on what it’s like to actually use the G2M. The first thing you’ll notice on the front panel (Fig. 1) is that the input is a standard, 1/4” phone jack - you don’t need any kind of special pickup or cable. A standard 9V battery powers the G2M, which is switched on when you plug into the input (so remember to unplug it when not in use). Battery consumption is about 10mA – on a par with medium-power guitar effects. The company quotes an estimated 70 hour battery life, which is a good thing because there are no provisions for adding an AC adapter. There’s also a front panel boost switch, but in practice, I found that that I didn’t need it. Fig. 1: The front panel is pretty minimalist: Input jack, and a boost switch to accommodate low-level pickups. Fig. 2: The rear panel has a 5-pin MIDI out connector and a Thru 1/4” phone jack that carries the guitar’s audio signal. Looking at the rear panel (Fig. 2), there’s a MIDI DIN connector and a Thru jack. The latter is helpful, as one of the common uses for MIDI guitar is to layer standard guitar sounds with synthesized sounds. This does two things: Gives a bigger sound, but also, the “real” guitar signal can help mask any glitching that may be in the MIDI synth sound. However, you may need to reduce the guitar’s level and pull back on the tone control, which limits the Thru’s usefulness; more on this later. The MIDI out works with standard synths and any interface with a 5-pin MIDI jack input, but bear in mind that some interfaces don’t have physical MIDI in jacks any more, as they assume you’ll hook up USB-based MIDI devices. In fact, I wouldn’t be surprised if Sonuus’s next product turns out to be G2M USB, where the MIDI connector gets replaced with a USB port. That would also allow the unit to be powered by USB so you could ditch the battery. On the top of the unit, there are several LEDs (blinky lights are always welcome!). These indicate: Power on and tuner function Low battery indicator Clip indicator MIDI activity indicator HOOKING IT UP I booted up Sonar 8, enabled the MIDI in on the V-Studio I/O interface, patched it to the G2M MIDI out, and plugged a Peavey Milano guitar into the G2M. First step is tuning, but you can use the G2M to help in the process: Play an open string, and the Tuner LED pulses. The slower it pulses, the closer you are to being in tune. Of course there’s no kind of sharp/flat display and the tuner is very basic; it’s not as easy to parse the display compared to a dedicated tuner, but it does the job. Next step is setting up a synth. There’s one crucial point: The G2M recognizes pitch bending, but the synth needs to have its pitch bend range set to plus/minus two semitones for the bending to work accurately. This range is pretty much standard, although some synths default to plus/minus one octave, and some vary it per preset. In any event, most synths let you save the pitch bend range as part of a preset, so you can edit it as needed. I chose Arturia’s Minimoog V as the target soft synth because I could set it for a mono (single-note) response, as well as turn off pitch-bending – if the part you’re playing doesn’t need pitch bend, you’ll get more accurate tracking if you turn it off. I chose mono response for two reasons: It allows getting glide effects with the Minimoog V, and as the G2M produces only one note at a time, I figured it was a better match. PLAY TIME! I recorded several parts into Sonar: Bass lines, leads, percussion, etc. Of these, I thought the lead was where the G2M was most appropriate, so let’s look at that in more detail. Fig. 3: The MIDI note was recorded about 20ms after plucking the note itself. The screen shot in Fig. 3 shows the delay between playing the guitar into a track and the MIDI note it produces. The latency is about 20ms, which is actually pretty good. Interestingly, this seemed relatively constant over the guitar’s note range; I expected it to get longer on low strings, and shorter on high strings. The fact that it didn’t likely indicates that Sonuus is using a different type of pitch detection technology than other MIDI guitars. Fig. 4: Screen shot of the original solo, as played, without any editing. To work with a “real-world” situation, I tried to play cleanly but not ridiculously so. Fig. 4 shows the results: Most of the part is relatively clean, but there are some little glitchy notes. Listen to the audio example “Original Lead” below to hear the part as recorded. Fig. 5: Screen shot of the original solo after editing. Audio Example Here's the way a lead guitar part played back originally from the MIDI data, prior to editing it and cleaning up the glitches. OriginalLead.mp3 If there’s one thing I’ve learned about MIDI guitar over the years, it’s that you better get ready to do some MIDI editing. I didn’t want to go overboard, but Fig. 5 shows the result of spending a few minutes editing the part. You’ll note it’s cleaner, and if you listen to the audio example “Edited Lead,” you’ll hear that the part is considerably tighter and less glitchy. Audio Example Here's what it sounded like after doing some MIDI editing. EditedLead.mp3 CONCLUSIONS MIDI guitar takes effort, no doubt about it. But don’t make it any harder than it needs to be: Use the neck pickup and pull back the tone control to reduce highs, as this emphasizes the fundamental and makes tracking easier. Be careful that you’re not standing too close to transformers and other sources of interference that can get into your pickups, as this can confuse the tracking – even if you don’t use the audio Thru out, use it to audition the guitar sound and make sure all is well. Also, some synth patches work better than others; it’s almost impossible get pads to screw up, but highly percussive patches are quite sensitive to little glitches. Finally, I found that resting the heel of my hand against the bridge to deaden the strings somewhat helped tracking considerably. When I get a chance, I’m going to try using flatwound strings to see if that helps even further. Speaking of the audio Thru, I didn’t really find it that useful because I needed to turn the level down on the guitar to avoid clipping, and the bassy tone wasn’t always what I wanted anyway. I got very good results by splitting the guitar signal with one split going to an amp, and the other to a DigiTech RP250, whose output went to the G2M. I created an RP250 preset that pulled back the highs and added a tiny bit of compression, which helped the tracking. I don’t want to paint a rosier picture than is justified; every MIDI guitar I’ve played takes some effort to get good results, and the G2M is no exception. In fact, it’s a little tougher than, say, a typical Roland GR-series device because it’s monophonic. You need to play carefully and deliberately; there will likely be some mistracking, and if you try to do some fast shredding, the G2M probably won’t be able to keep up. On the other hand, we’re dealing with a true guitar-to-MIDI converter that does a credible job of opening up the synth world – and at a bargain price. As long as you don’t expect to play with the same level of abandon as a standard guitar, you’ll be fine. Best of all, you’ll be able to add textures to your music that would normally require having keyboard technique. If you’ve ever wanted to try out MIDI guitar, the G2M is a fine place to start. Craig Anderton is Editor Emeritus of Harmony Central. He has played on, mixed, or produced over 20 major label releases (as well as mastered over a hundred tracks for various musicians), and written over a thousand articles for magazines like Guitar Player, Keyboard, Sound on Sound (UK), and Sound + Recording (Germany). He has also lectured on technology and the arts in 38 states, 10 countries, and three languages.
  15. A Space-Age Approach To An Age-Old Need by Rick Van Horn KEY NOTES Innovative design and functional features Extremely durable construction Cases are heavy Hardware case had fit problems A lot of thought has been given to the design of Stagg's new Advanced Concept molded hard-shell plastic drum cases. You need only glance at their unusual shape and distinctive molded contours to get the impression that somebody planned these cases as more than just "containers." They're meant to be functional pieces of equipment in their own right. The Shape Most drum cases are essentially cylindrical, in order to conform to the cylindrical shape of the drums inside them. There's generally a flat section on the side that allows the case to be placed edgewise on the floor—"standing up," as it were. Stagg cases are much more triangular, combining the cylindrical portion with two extended "corners." These corners provide the flatted edge for standing, and also reinforce the overall structural integrity of the case. That's the upside of the cases' design. The downside is that it makes the cases larger and bulkier than traditional models, which may become an issue in the trunk of a Corolla. The Window Sticker A list of the features offered by the Stagg cases reads like the options shown on the window sticker of a high-tech car. Let's take a look. "Eminently stackable and stable due to Stagg's X-centric design." All Stagg drum cases, no matter what their size, have molded circular protuberances that mate with recesses on all other cases, locking the cases together when they're stacked. Also, instead of stacking each case in the center of the one below it, the cases stack in such a way that their "bottoms" (the more or less flat area opposite the handles when the cases are carried) are all flush. This puts the weight of the stack over the strongest part of each case, and also allows you to back the entire stack of cases up against a wall for further support. Cool idea. "All drum cases are lined top and bottom. The Basic Snare case is fully lined." The lining is a sheet of fabric over a slight amount of padding. This is a nice feature, as far as it goes. My problem is that the cases are of the "telescoping" variety, meaning that each case of a given diameter can expand to accommodate drums of different depths. With any drum that's deeper than the completely compressed case, the lid of the case rests on the top of the drumshell. That shell is actually providing the structural support for the case. I'm not a fan of this design, even though it does help keep manufacturing costs—and purchase price—down, since the manufacturer only has to create one model for each drum diameter. But if a case is going to be a telescoping model, I'd like to see a lot more padding or other protection against top and bottom impact for the drum inside. "Convex-shaped top shell to protect against compressive force from above." Most hard-shell cases simply stretch some material across the top of the drum in a flat fashion to create the lid. Any impact immediately bows the lid down, risking damage to the head of the drum inside. The Stagg cases can't totally eliminate that risk, but the convex shape does "dome" the lids a little to absorb impact before the lid would come in contact with the head below. "Shock-absorbent support zones to disperse external shock to the case and drums." This is a more important structural feature—and a nice one. The shape and molded contours of the case help "spread" any impact around the drum, instead of allowing a localized blow. This should provide much more side-impact protection than that afforded by a traditional cylindrical case. "Case tops feature Water Transport Channels to avoid liquid pooling on the top." You may have seen the recent MD ad with a photo depicting rain beating down on—and pouring off of—a Stagg case. The way I see it, hard-shell plastic cases should be impervious to rain anyway. But at least there won't be a pool of water on the top of the case to dump on your shoes when you pick the case up. "Adjustable large format buckles at the end of each strap for flexibility of shell depth." These molded composite squeeze-type buckles are easy to operate, but it'll take you some time to get used to having to open two buckles on every strap in order to raise the lid on the case. This is another aspect of the "telescoping" design. "All drum cases feature sturdy "D" rings that allow the owner to pass a cable through for locking with a padlock, or to secure the cases to the inside of a truck for transport." This seems like a minor feature, but it could mean the difference between going home with or without your drums after a gig. The D-rings are riveted to each case on a nylon strap, and would not pose much of an obstacle to a determined and prepared thief. But they certainly would help prevent an impulsive "grab and go" heist. "Bass drum and large floor tom cases are fitted with recessed transport wheels." Wheels are great for pulling a case on level ground, going up a ramp, or moving from a dock-level truck into the backstage area of a major venue. But they don't help when it comes time to lift that puppy into the bed of a pickup or the trunk of a car after a gig at the local Elk's Club. With that in mind, this may be a good time to mention the fact that the durability of a thick molded plastic case comes with a downside: weight. For example, the 10" Stagg tom case weighed about 71/2 pounds—exactly the weight of the 8x10 maple tom we carried in it. The 22" bass drum case weighed 221/2 pounds—only a few pounds less than the drum we put in it. By contrast, most drum bags add very little weight to the drums they contain. But, of course, neither do they offer the protection factor of the hard case. I'm just saying that you should make an informed choice. "Hardware cases feature recessed wheels, and are designed with I-beam and triangle construction for maximum horizontal and vertical strength." In general, I liked the 40" hardware case that we tested. It was a very practical, open-topped rectangular container that easily accommodated a fair amount of stuff. The metal reinforcement inside gave it the structural integrity necessary to remain secure and rigid while being rolled or carried. "Height Extension and Stacking Tray Extensions are available for the 40" hardware case." The idea here is that if you need more vertical space for additional hardware, or you want a shelf section for cables, mic' stands, and so forth, you can add them to the basic case. The various sections lock together with rotating ATA-style clamps. And this was where I encountered some problems. To begin with, the clamps didn't seem to lock down very securely. In some cases, the upper and lower halves of the clamps didn't align properly for optimum grip. In others, the amount of "play" that remained after the clamps were closed allowed the various case sections to slide a bit. I never got the sense that the case was going to come apart, but neither did it feel like one solid unit. Also, while I was able to fit the vertical-space section onto the base unit easily, I was completely unable to do the same with the shelf unit. The molded countour of its plastic body did not conform to the contour of the base unit, so there was no way to mate the two. These two problems reflect a need for more attention to quality control and proper fit between the components of the hardware case system. "Address tag and STAGGCase model number designed into the fastening strap." This is another thoughtful touch—although the address tag is a little hard to get out of its holder. Also, the tag is completely removable. So in terms of true security, it doesn't replace the age-old practice of painting one's name on the side of the case. "The air is free." Stagg says that their cases are designed with just enough space between the top and bottom sections to provide the airflow necessary for opening or closing the cases quickly. I've used some drum cases in which a certain amount of "vacuum hold" between the sections was, indeed, a minor inconvenience, and it's thoughtful of Stagg to take this problem into consideration. However, the spacing between case sections in our test group became more pronounced as the sizes increased, to the point where the 16" floor tom and 22" bass drum case lids literally "rattled around" on top of the lower sections. A slight reduction of the space would provide a more solid-feeling case assembly in these instances. Not On The List A feature of Stagg cases not mentioned in their promotional material is their strap-type handles with rubber comfort-grips. Each case has a pair of these—one attached to the bottom section and one attached to the lid section. (There's a single additional handle between the wheels on the bass drum and large floor tom cases, for two-person carrying.) The two straps come together closely when a case is fully compressed. But they'd be spread further apart if the case contained a deep-shelled drum, which might cause a carrying problem. A different carrying problem was created (at least for me) by the rubber grips on the straps. They're fairly large, ostensibly for gripping comfort. However, I have a small hand, and I found it difficult to comfortably grasp both of the grips in such a way as to balance the weight of the case evenly between the straps. I have drum bags with strap handles that fit within a wrap-around cover, thus creating a single grip position. I'd suggest this sort of approach for the Stagg grips. Case Closed I'm impressed with Stagg's innovative approach to the design of what have, heretofore, been pretty unromantic pieces of drum gear. True, some of the positive design aspects create corresponding negative ones that must be considered. And there are some significant quality-control issues that need to be addressed (particularly on the hardware case). But based on the effort that obviously went into the initial design process, I'm confident that those issues will be dealt with in short order. All in all, I'd say that Stagg's foray into the professional drum-case market is a pretty auspicious debut. THE NUMBERS 8" tom case $71.99 10" snare case $79.99 10" tom case $99.99 12" snare case $89.99 12" tom case $99.99 13" piccolo snare case $91.99 13" snare case $99.99 13" tom case $119.99 14" regular and free-floating snare case $99.99 14" piccolo snare case $92.49 14" snare case $99.99 14" tom case $129.99 16" floor tom case $169.99 18" floor tom case $219.99 18" bass drum case with wheels $219.99 20" bass drum case with wheels $269.99 22" bass drum case with wheels $289.99 40" hardware case with wheels $259.99 40" height extension module $119.99 40" level extension module (shelf) $129.99 48" hardware case with wheels $269.99 (877) 231-6653, www.staggmusic.com
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