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  1. A Joint Venture In Value by Rick Van Horn KEY NOTES Pro 3 bags feature excellent design and construction quality All bags offer outstanding affordability Pro 3 hardware bags need internal stiffening Pro 4 bags have limited applications Fred Beato's name has been synonymous with high-quality, innovative drum bags for many years. Tom Shelley's Universal Percussion is equally well known for the design and distribution of quality products that are manufactured offshore for exceptional affordability, such as Wuhan cymbals, Attack drumheads, and Cannon hardware. Not long ago, Fred and Tom got together in a joint venture. Fred would design a line of drum bags that would meet the needs of contemporary drummers. Tom would arrange to have Fred's designs manufactured in China, where low material and labor costs could help make the resulting bags more affordable in the US market. The bags resulting from this joint venture have been dubbed the Beato Pro 3 and Pro 4 series. Between them, the two lines offer a wide range of sizes and features. Pro 4 Series Although it's out of numerical sequence to do so, I want to start with the Pro 4 series. These are economy bags designed to provide minimal protection at minimal prices. They're made of a strong, water-repellant nylon material, using extra-heavy thread. The insides feature a soft, scratch-resistant surface, but no padding. Each bag features heavy plastic piping at the edges, as well as a heavy-duty zipper. For what they are, they're well designed and well constructed. However, the only way I'd recommend these bags for serious gigging purposes is if you handled your drums yourself, and then only if you loaded them in and out of a car from the trunk and seats. I wouldn't feel comfortable loading them into a van or pickup truck, where the loaded bags would likely slide on the metal floor of the vehicle and bump into one another. I don't think the fabric would wear well against such sliding, and the drums would not be protected from any impact. On the other hand, if you tend to leave your drums in a storage area or a rehearsal space, the Pro 4 bags would make excellent "dust cover" containers. This might make them especially appealing to student drummers who keep their drums at home. The bags are also inexpensive enough to be used as "case liners" within ATA-style road cases, making the drums easy to pull out of the road boxes. Pro 4 bags are available in five-piece Standard and Fusion sets, for drums only. No cymbal or hardware bags are offered in this series. Pro 3 Series The Pro 3 series is, in a word, impressive. These heavy-duty Cordura bags are well padded to protect their contents. Each drum bag comes with a shoulder strap, heavy-duty zippers, and a separate circular compartment for accessories (perfect for spare heads). In a nice design touch, two side handles and a third handle perpendicular to the others provide carrying options. Heavy-duty webbing is used to reinforce the bags where the handles are attached. Drum bags are sold individually or in Standard and Fusion drumkit sets. Pro 3 hardware bags are extremely well designed. I particularly liked the way the zippers open around the top perimeter of the bag, allowing the top flap to be pulled back out of the way, exposing the bag's contents in a "trunk-like" manner. Top carrying straps have a padded grip; a handle at one end is covered with rubber for comfort. At the other end of each hardware bag is a set of built-in wheels, making transport very easy. My only complaint with the bags is that the bottoms, though stiffened, are not rigid. So if you put a load in the bag and start to wheel it away, the bag sags in the middle. This problem can be reduced somewhat by keeping the hardware within the bag extended enough to run from top to bottom, providing internal support. A solid bottom would be even better, but that might increase the cost factor beyond the "affordable" range that the bags now enjoy. Pro 3 cymbal bags come in two versions. The basic model is a no-frills padded bag with one interior compartment, one smaller compartment, strong web handles, and a shoulder strap. The Deluxe model offers five internally divided spaces, one outside cymbal compartment, and a large outside pouch that also contains four drumstick sleeves and a zippered accessories compartment. The bottom of the bag is reinforced with a wide base and plastic feet. This is a really nice cymbal bag! Pro 3 stick bags are available as one- and two-pair marching stick quivers, and also as six-, six-plus-, and twelve-pair traditional bags. Each is made in the same manner as the drum, hardware, and cymbal bags, with excellent construction and a reasonable amount of padding. A Word About Value For as long as I've been writing columns and reviews for MD, I've preached that an investment in drum equipment should be protected by an investment in cases or bags. The problem comes when the first investment exhausts the drummer's budget, eliminating the possibility of the second investment—at least right away. The new Beato bags should help solve that problem. Now, I don't believe that anything is a value simply because it's inexpensive. It must also have the quality and functionality required to do the job it's designed to do. By that definition, the Beato Pro 4 bags would be valuable as dust covers or extremely light-duty carrying bags for drums. But they would not serve well for serious gigging drummers. On the other hand, the Pro 3 series offers drum, cymbal, hardware, and stick bags that should more than meet the needs of any drummer playing out locally. And when that functionality is combined with the bags' impressively low prices, now we're talkin' value. THE NUMBERS Pro 4 Series Bags Pro 4 five-piece Fusion set $48.95 Pro 4 five-piece Standard set $49.95 Pro 3 Series Bags 9x10 drum bag $31.95 11x12 drum bag $33.95 12x13 drum bag $35.95 5x14 drum bag $33.95 6x14 drum bag $34.95 12x14 drum bag $44.95 16x16 drum bag $47.95 16x22 drum bag $71.95 18x22 drum bag $73.95 Five-piece Fusion set $179.95 Five-piece Standard set $185.00 Cymbal bag $28.95 Deluxe cymbal bag $54.95 Marching Stick Quiver (one pair) $9.00 Marching Stick Quiver (two pairs) $12.95 Stick bag (6 pairs) $9.00 Stick bag (6+ pairs) $10.50 Stick bag (12 pairs) $12.00 25" hardware bag with wheels $65.25 36" hardware bag with wheels $72.25 47" hardware bag with wheels $79.75 (800) 282-0110, www.universalpercussion.com © 2006 MODERN DRUMMER Publications, Inc. All rights reserved. Reproduction without the permission of the publisher is prohibited.
  2. I've never understood drummers who like using black or dark brown drumsticks. Boring. I want the audience to be able to appreciate the impressive way my sticks move. (Ha!) But to be able to do that, the sticks have to be seen. If you feel the same way, then chances are you're going to dig Vater's Color Wrap series. And what colors they are! The high-density wrap material used for this new series looks amazing under lights. We're talking a deep, penetrating finish that is visible for miles. (Okay, I exaggerate a little here.) The Sparkle wraps are tremendous, and the Optics are more dazzling than a Pink Floyd laser show at your local planetarium. (Okay, I exaggerate a little more here.) I wish I had a drumkit wrapped in this stuff. The wrap material is also surprisingly durable. I've been using the sticks on my gigs for a few weeks, and they aren't very nicked up, even after I've bruised up my kit with 'em. Playing heavy rimshots, riding on the rims, and even smacking the edge of a big ride cymbal didn't tear them up like you might expect. Okay, so the wrap looks good and is durable. But how does it feel? Well, while I've always been impressed by the quality of Vater sticks, frankly, I've found them to be very smooth and a little slippery for my grip. But with the Color Wraps, there's a good grip surface and an added feeling of security. I could hold the sticks loosely without feeling like I was going to drop them. So Vater's Color Wrap sticks feel great in my hands, seem very durable, and look amazing. But all is not perfect. My only gripe is, as of now, they're only available in 5A, 5B, and Power 5B models. I'd sure like to try the Color Wraps in some cool Vater sizes, like their Manhattan 7A or Fusion models, or from the other end of the spectrum, their Power House model. Last but not least, just what colors are available? Color Wraps are offered in blue, gold, and red sparkle versions, as well as black, silver, and purple Optic finishes. (Check out Vater's Web site to give them a closer look.) All models are available with wood tips only and list for $13.95 per pair. Your playing will look mahvelous. (781) 767-1877, www.vater.com © 2006 MODERN DRUMMER Publications, Inc. All rights reserved. Reproduction without the permission of the publisher is prohibited.
  3. Special Delivery by Billy Amendola KEY NOTES Mesh heads provide accurate and realistic playing surface Much improved V-11 V-Hat 22" ride is “odd man out” in this family Larger snare pad Roland has addressed the pricing issue somewhat by upgrading their mid-priced TD-8 V-Stage kit with a lot of the same features offered in the high-end TD-20, and thus creating the new TD-12. Okay, let’s keep it realistic: For half the size and cost, you can’t get everything you want. But with the TD-12, you do get your money’s worth—and then some. To Have And Have Not One of the features not brought to the TD-12 from its big brother is separate outputs, which are essential if you want control over every drum individually for recording. Instead, the TD-12 offers two stereo mixes and two individual outputs. This means you’ll have to spend a little time inside the module working out the levels before recording. As an example, you could internally balance the three toms, then group them all on one output fader. The TD-12 provides 560 drum instruments and 262 backing instruments for play-along sequences, all in a smaller, more affordable package than the TD-20. (To hear some of those sounds, go to www.moderndrummer.com.) I personally found some of the TD-12’s upgraded sounds cooler and more usable than those found in the TD-20. The usual V-Drum sonic package is all here, from authentic acoustic sets to ultra-modern electronic kits. You can customize your snare drum, toms, bass drum, and cymbals with the modeling-based V-Editing function. On the kick and snare you can change shell depth, head type, tuning, muffling, shell material, and strainer adjustment. On the toms you can also edit shell depth, head type, head tuning, and muffling. And for that “ultra-realistic” quality, you can even add snare buzz. When it comes to cymbal sounds, you can alter size, sizzle type, and sustain. The TD-12 is also brush-compatible, with sounds and sensitivity that make such a feature practical. Plus it includes what Roland calls the Expressive Interval Control feature for accurate snare and ride/crash cymbal playing. This function varies the sound in natural ways, based on the speed of stroke repetition. As with previous V models, there are ambience parameters including room size, type, and shape, as well as mic’ position. What You Hit The drum pads on the TD-12 come equipped with Remo’s now-legendary mesh heads. These pads have kept Roland at the top of the field since they first introduced V-Drums. They’re comfortable to play on, virtually silent, and as realistic as can be with every stroke. The VH-11 hi-hat is a major improvement on the TD-12. And the best news is: It’s compatible with the TD-10, TD-20, TD-8, and TD-6. The new hat now comes with only one pad (a top-moveable playing surface). Underneath is a motion sensor unit that stays attached to your traditional hi-hat stand. When I checked it out, it was accurate on every beat. As far as I’m concerned it’s the closest Roland has come to accurately simulating an acoustic hi-hat. If you’re accustomed to playing the TD-8, you’re sure to love the upgrade features that have been incorporated into the TD-12—especially the new, larger 10" PD-105 snare pad. On the other hand, if you’re used to playing on the TD-10 or TD-20, you may need to adapt to the TD-12’s smaller 8" tom pads. Speaking of pads, I’d recommend eventually adding two more pads to the TD-12, one for an additional crash and one to trigger your sequencer. As it’s designed now, the size of the TD-12 is ideal for small gigs or recording situations. It takes no time at all to pack it up, get it in the car, and re-set it up at a club or studio. Bottom Line Electronic kits in general are great for songwriting, for home recording, and especially for practice situations. With this in mind, it seems like there’s a Roland kit for everybody. The new TD-12, in particular, delivers lots of high-end features without an unattainably high-end price tag. THE NUMBERS TD-12 Electronic Drumkit $4,299 Includes TD-12 sound module, one 10" PD-105 dual-trigger V-Pad, three 8" PD-85 dual-trigger V-Pads, one 11" VH-11 V-Hi-Hat, two 12" CY-12R/C dual/three-way trigger V-Cymbal Ride/Crash units, one KD-85 V-Kick trigger pad, and an MDS-12 V-Stage series stand with cables. (Bass drum pedal, snare stand, and hi-hat stand not included.) (800) 386-7575, www.rolandus.com © 2006 MODERN DRUMMER Publications, Inc. All rights reserved. Reproduction without the permission of the publisher is prohibited.
  4. As Unique As "El Negro" Himself by Norman Arnold KEY NOTES Swivel mounting lugs Wide range of pitches Horacio "El Negro" Hernandez is known for his stellar ability to play left-foot clave patterns using a cowbell mounted on a foot-pedal bracket. Horacio also incorporates bells into his complex and fluid Latin sticking patterns. So there's a certain logic to his development of a signature line of cowbells in conjunction with Pearl's percussion division. The Horacio Hernandez (HH) series features five different copper-finished bells. Each has one curved surface and one flat surface. The curved surface is easy to strike when the bell is mounted perpendicular to the player, while the flat side is the better target when the bell is facing the player. Each bell comes with a rubber muting band that can be attached to the bell to mute the overtones. One particularly cool feature on all of the bells is an innovative mounting bracket that can rotate 90°. This allows stacking of multiple bells on one post without the tightening bolts being on top of one another. It's a great idea and an excellent design. The Bells Are Ringing... In terms of sonic performance, all of the bells have a great fundamental tone and very prominent overtones, whether played mounted on a stand or held in the hand. For hand playing in particular, the flat and curved sides make it possible to get a much wider range of natural tones than are possible with a more traditional bell design. The small BELLa bell is the highest pitched of the bunch. It has a piercing tone that would cut through just about anything. The ClaBELL and ChaBELLa are both mid-pitched bells. They're great for foot-pedal mounting, and would give you that constant quarter-note part when playing a cha cha. Both bells are also very resonant, with the ChaBELLa sounding like a classic cha cha bell. The IsaBELL is a mambo bell with a great low sound. There's a noticeable difference to the sound when you hold the curved or flat side in your hand. Playing with the flat side resting in your hand lets the curved surface ring out, generating a series of cool overtones. Holding the curved side in your hand reduces the overtones, creating a more muted sound. The same holds true for the lowest-pitch MaryBELL. A Bell Choir Pearl's Horacio Hernandez cowbells are instant classics. The line offers enough sonic variety to justify having more than one bell, and mixing and matching them would be a breeze. Mounting the ClaBELL on a pedal under your congas would be a fantastic place to start. The IsaBELL and ChaBELLa would make a great timbale setup. In fact, considering that none of the HH cowbells are particularly expensive, it would be cool to get all five. Add in Pearl's stable and well-constructed foot pedal bracket and their percussion rack with four mounting brackets and poles, and you'd be set for any combination of bells—and virtually any musical situation. THE NUMBERS HH-1 BELLa (high-pitched cha cha bell) $39 HH-2 ClaBELL (foot clave bell) $45 HH-3 ChaBELLa (low-pitched cha cha bell) $47 HH-4 IsaBELL (mambo bell) $50 HH-5 MaryBELL (timbale bell) $55 (615) 833-4477, www.pearldrum.com © 2006 MODERN DRUMMER Publications, Inc. All rights reserved. Reproduction without the permission of the publisher is prohibited.
  5. When Trash And Crash Are Not Enough by T. Bruce Wittet KEY NOTES Good blend of exotic wash and clean articulation Wide stylistic applications,including acoustic jazz 22" ride is “odd man out” in this family Hats lack chick sound when played with foot Catch Stanton Moore behind the drums and you'll notice two things. The first is the raucous and loose New Orleans feel, which Stanton comes by honestly, having cut his musical teeth in the Crescent City. The second thing that's apparent is contrast. You're apt to hear an 18" kick alternating with a 26" vintage bomber, both barking from the same drumset. You'll also hear cymbal sounds that vary from articulate to splashy. Enter the new Bosphorus Stanton Moore line, a series designed in extensive collaboration with Stanton. The series consists of five different models that have spent several years in the prototype stage. Bosphorus is a bit of an oddity. It's a Turkish company making cymbals by hand, but driven by American drummers (including jazz great Jeff Hamilton). When the Old World pairs with the New, possibilities abound. That was our expectation when Bosphorus sent a pair of 14" Fat Hats, 20" and 22" Wide Rides, a 20" Pang Thang, and a 20" Trash crash for review. Perhaps more to the point, the cymbals ought to fit into Stanton's dynamic, slap-happy style of drumming—which owes much to retro sounds—yet hold their place through a big PA rig. First Glancing Blows At first glance, the Stanton Moore cymbals appeared quite faithful to their forefathers. The basic hammering on the cymbals matched up perfectly with that of a mid-1970s Turkish K in my possession, as did the lathing—at least the top lathing. Here we find reasonably tight grooves that encircle the cymbal and follow slightly irregular paths. The bottom of each cymbal, however, seems to have been lathed just enough to remove excess material, leaving narrow, uneven bands of raw metal showing through. I checked this with Bosphorus CEO Michael Vosbein, who responded that a wider chisel is used for the bottom. At any rate, the idea is that the top of the cymbal is richer in harmonics than the bottom, which exercises a muting or gating effect. 14" Fat Hats The Fat Hats were thin and flexible (top and bottom), with the bottom cymbal barely a third above the top in pitch. They mated well and produced a nice, sustaining sizzle when the two plates were barely touching. In my mind's eye I could see Stanton wailing on these. When I took the hats out of the testing room and onto a gig, I was concerned that the chick sound was not pronounced enough (which is something to be expected of thin cymbals). But taken in the context of Stanton's style, which is more about stick work and open/closed effects, the Fat Hats performed exceptionally well. Maybe the chick sound was a little wanting, but I could be happy using these in a range of gigs from loud to very soft. Wide Rides The 20" Wide Ride was a killer. Stanton originally intended it as a left-side crash/ride, but he's recently been positioning it on his right. Whatever the configuration, this cymbal produced an articulate sound from a range of sticks. With a 5B, the cymbal was awash in rich overtones but not overshadowed by them. I had to work a little to control the swell, but I didn't mind because the frequencies coming up were so balanced and warm. On a gig, these were swallowed up; in the test room, I "cheated" by affixing a two-inch piece of duct tape to the underside. I'd use this cymbal as my first ride in an acoustic situation. The 22" Wide Ride surprised me. I guess I expected a larger version of the 20". (After all, it shares the same "wide" designation.) But the bell displayed less hammering, the profile was lower, and the playing response was more direct. Even the feel was a little stiffer. The reasons for these disparities dawned on me after playing the cymbal for a while. Stanton's thing is about contrast, right? Here we have an extremely articulate ride cymbal that can project, but that's not overly heavy. Owing to its low profile, it generates low-frequency undertones that tie it nicely into the ensemble. "Push crashes" (short accents played with the shank of the stick) jump out and then rear back. When I played this ride with mallets, it resisted crashing ever so slightly. Perhaps this was due to the muting effect of the underside lathing. But with a little playing effort the sound grew into a hearty roar. 20" Pang Thang The 20" Pang Thang was another favorite. It resembled traditional pang-style cymbals, owing to its thin overall design and 2" flattened edge. But it was more full-bodied than most China-type cymbals. I daresay that some drummers out there would find this the ideal jazz ride: The high profile made the stick work audible in frequency and crystal clear in impact, while stick-shank accents across the bow were swallowed up almost instantaneously. Finally, the flat edges presented another brash playing surface. 20" Trash Crash The 20" Trash Crash was a party animal. Ostensibly another high-profile, China-style cymbal, the Trash Crash features ten hammer clusters—like craters on the moon—evenly spaced around the circumference. Each cluster consists of roughly fifteen separate hammer peen marks, made after the basic hammering was completed. Those clusters mess with the sonic integrity of the cymbal something fierce! While traveling across those peaks and into those valleys, I located all sorts of disparate overtones and sweet spots for riding. (Curiously, this crash cymbal was marked "ride" on the bottom.) A strong, crashing blow generated a fat, throaty roar. Like the Pang Thang, the Trash Crash is emphatically not your basic "gunshot" China punctuation cymbal. Mount it right side up! Conclusion The Stanton Moore series is a versatile group of cymbals that lean toward the trashy end of the spectrum. The old jazz drummers' adage that "Every cymbal should be a ride and a crash" applies to each of these babies. Their sonic potential is limited only by your imagination, stick placement, and the inescapable exotic timbres. If you need ultra-distinct 8th-note rides and shrill crashes, then these aren't for you. But if you don't mind breaking a few rules and muddying the waters, you'll come up with some real dirt here—grade-A topsoil, if you ask me. THE NUMBERS 14" Fat Hats $495 20" Wide Ride $495 22" Wide Ride $545 20" Pang Thang $575 20" Trash Crash $575 (770) 205-0552, www.bosphoruscymbals.com © 2006 MODERN DRUMMER Publications, Inc. All rights reserved. Reproduction without the permission of the publisher is prohibited.
  6. Most sequencing software doesn’t allow for adding parallel effects — but fortunately, there’s an easy workaround. By Craig Anderton In almost all cases, Digital Audio Workstation software assumes you want to put any track effects in series, one right after another. But what if you want to use parallel effects, where a track splits into various effects, which then mix back together to create a potentially more complex and detailed sound? One option is to use a plug-in like BIAS' Vbox, or TC Electronic's late, great Spark FX Machine. These are essentially "plug-in hosts for plug-ins" that create a matrix of slots where you can insert effects in both series and parallel. Yet while these are convenient, you don't actually need any special kind of plug-in to do parallel effects. SEND IN THE CLONES The trick is to copy (clone) the track to which you want to apply the parallel effects, resulting in several parallel audio tracks. You then apply effects to these tracks as needed. For example, suppose you want to add a parallel effect to a piano track, where a noise gate lets through only the peaks; furthermore, this goes to a reverb that's panned far left. Meanwhile, a second noise gate sends a different set of peaks through a short delay, to a different reverb that's panned far right. You could do this with aux sends, but there's an alternative. For this example, we need three parallel tracks: Straight piano only Straight piano + noise gate + reverb1 (panned left) Straight piano + noise gate + delay + reverb2 (panned right) Copy the straight piano track two times for a total of three piano tracks. The first track is the "straight," unprocessed track. In the second track, insert the noise gate and reverb, then pan the track left. For the third track, insert the noise gate, delay, and second reverb, then pan that track right. (Of course, you could also just slide the third track behind a bit in time to create the delay, but sometimes it's a lot more convenient to just dial in a delay, particularly if you need to sync to tempo). Three tracks are creating the parallel effects setup involving a piano, reverbs, gating, and delay. This example shows Cubase SX, but the same concept applies to virtually any sequencer/hard disk recorder. Because tracks in today's DAWs are aligned with sample accuracy (and assuming the effects paths have delay compensation), you won't hear any flamming, comb filtering, or other undesirable effects when you combine the tracks. "VIRTUAL MICS" WITH PARALLEL EQ Here's a real-world example of using parallel effects to create a wider stereo image. In some ways pianos are fun to record, because they generate sound over a wide area. Stick a couple mics in the right places, and you'll end up with some great stereo imaging. But other instruments, such as classical guitar, accordian, percussion, etc. don't have a wide stereo image if you hear them from more than a few feet away — although up close, it can be a different story. If you're facing a guitarist, your right ear picks up on some of the finger squeaks and string noise from the guitarist's fretting hand. Meanwhile, your left ear picks up some of the body's "bass boom." Although not as directional as the high-frequency finger noise, it still shifts the lower spectra somewhat to the left. Meanwhile, the main guitar sound fills the room, providing the acoustic equivalent of a "center channel." This all became very clear to me when recording a guitar/keyboard duo, where the keyboard had a nice spread but the guitar kept getting shoved to the center of the image. What to do? I tried using two mics on the guitar, but the phasing issues were unacceptable. Then I thought about what made the sound "wider" as you got closer, and a solution suggested itself. I've also used the following technique to stretch a piano and organ's image beyond what I could obtain simply by using two mics; in fact, this basic principle works for most sound sources where the bass doesn't need to be in the middle of the stereo image. The first step in simulating the effect of being close to the guitar was to clone the original guitar track to two more tracks. The first clone provided the "squeak" component by including a highpass filter that cut off the low end starting around 3 kHz, with 11 dB of total cut. This was panned right. The second clone for the "boom" channel used a lowpass filter with a sharp cutoff from 350 Hz on up. An additional shelf added a mild 3 dB of bass boost, kicking in at 125 Hz. This came in a little closer to center than the squeak channel. I also needed to remove some low end from the "straight" channel to make some sonic space for the boom channel. Adding these two tracks to the main track pulled out some of the "finger squeaks" and "boom" components that were in the original sound, and positioned them in a more realistic stereo location. This also stretched the stereo image somewhat. And because these signals were extracted from one mic, there were none of the phasing problems associated with multiple mics. These three EQ curves, when panned as described and mixed for the proper balance, create a much larger image that belies the fact a recording was done with a single mic. As to mixing these three elements, the drastic amounts of high and lowpass filtering on the cloned channels brought their overall levels way down, even without touching the channel fader. If you isolate these tracks, it seems as if their impact would be non-existent due to the low level and restricted frequency range. But if you mix them in with the main channel, the entire sound comes to life…which is definitely a Good Thing.
  7. How do you know which take is the one? by Craig Anderton The essence of recording comes down to this: capturing great performances. All related tools, techniques, and technologies become irrelevant without that crucial element. Granted, some tools help promote better performances, such as the way loop recording allows the performer to get into a "groove." But ultimately, a good performance also has much to do with production skills, and knowing how to get the most out of a performer. However, is that process based solely on gut feelings, or is there some quantifiable element that can help ensure getting the best possible take? THE PERFORMANCE CURVE I've worked with a lot of artists over the years, and found that certain artists tended to reach peaks at a particular time in the process of recording their takes. The following charts describe the different types of performances I've seen in the studio. The double-peak: After observing what happened with my own performances when recording composite vocal, guitar, and keyboard parts, then picking the best sections, I noticed that the quality of my takes follows a definite pattern. The first couple takes are pretty good, then they start to go downhill before taking an upward path again. Eventually they hit a peak that sometimes exceeds the initial one, then past a certain point, deteriorate at a pretty rapid rate. I call this a "double-peak" curve because it has a peak at the beginning, and a peak toward the end. It's uncanny how often this happens. It doesn't even matter which instrument I'm playing. But that's just me; I've also produced quite a few artists over the years, and noticed other distinct patterns. The quick starter: This performer starts strong, has several good takes in a row, then doesn't really improve on the performance over time. Many times, these are musicians who play live a lot. They're conditioned to get things right and "give it their all" because live, you're get only one take. The long ramp-up: In this case, the performer takes a while to "warm up" and get into the groove. This often happens with musicians who compose in the studio. As they feel their way around the part, they become more comfortable with it. After they hit their stride, sometimes you'll get a killer take; sometimes you'll get a series of takes that are all pretty good, and when composited together, produce a definitive performance. The anything goes: This is the kind of performer who goes strictly from the gut. Rather than follow a particular curve over the course of several takes, they hit high and low points within individual takes, as the mood hits them. These are the most time-consuming performances to comp, because you might end up taking different phrases from early, middle, and late takes. Yet the final results can be really good, because there are a series of spontaneous moments that produce multiple high points during a take. However, you want to make sure it "breathes," and has some less intense sections to provide contrast with the moments of high emotion. In any event, these are people where you record everything because you never know... The rock steady: I first encountered this type with classical guitarist Linda Cohen. All her takes are consistently good, so the only real question is whether she can do one that's better. There are other patterns as well, but these are main ones I've recognized. WELL, SO WHAT? This may sound a bit abstract, but there are practical ramifications. For example with Linda, she knows when she's done a good take, at which point she tends to want to move on (classical guitarists don't get huge budgets for studio time, so time is of the essence). I usually agree with her, but once I had heard her do a tune better in rehearsal than how she did it in the studio, even though her take was technically flawless. I wanted to ask her to do another one, but knew she'd think it was superfluous. So, I said into the talkback mic that unfortunately, the record button hadn't been enabled on her previous take. She was kind of bummed but she's a pro, so she did another take. However, my "mistake" sort of shook things up; while her part was again perfect, it had a little more feel (I presume it was because she had gone through these conflicting emotions of thinking she'd done a real good part, only to find it hadn't been recorded). Of course, the previous take had been recorded, so if the newer one wasn't as good, there wouldn't have been any problem. However, you wouldn't try that approach with the "long ramp-up" performer, because it takes them so long to get where they're going that they'd likely strangle you if they thought you'd made such a major mistake. With their type of curve, you're best off saying "the last take was really good, but they keep getting better, so let's try just one more." The "quick starter" is something else. If the takes aren't happening, I prefer to move on to a different song entirely, then return to the one where I want a better take. Coming back to it seems to sort of "reboot" this type, which takes advantage of their "quick starter" mentality. For the "anything goes" style, I usually don't ask for new takes, but tend to go more for punches in specific sections ("That was good, but I think there needs to be a bit more energy when the second verse comes in"). These performers seem to break performances down into smaller pieces rather than thinking in terms of takes, so they fit well with a punch-in oriented approach. With the "double peak" type, as long as the takes keep getting better in the second series of peaks, keep recording. Once you hit two or three takes in a row that don't improve on previous takes, move on. It's unlikely you're going to get anything more that's worth recording. THE VOLUME JUMP TRICK Here's another sneaky trick I was taught by an extremely talented engineer: Just when you think the performer is about to peak, turn up the headphone volume ever so slightly – no more than a dB or so. This raises the person's energy level an extra little bit, and often inspires what ends up being the best take. But you only get one, or possibly two, chances to do this. It's the novelty of the change that makes the difference. So, you have to gauge precisely when that Cartier-Bresson-like "perfect moment" is about to occur. Knowing the performance curve helps you decide. With "anything goes," I wait until I have enough takes in the can that I know it's possible to put together a good part. Then I'll goose the volume a tad and do a few more takes. Sometimes these are ideal for adding that slight extra "edge" on the final verse or chorus, or elsewhere for that matter. The "long ramp-up" type is the most difficult to anticipate. You have to choose the moment that's just before their best take. Some performers have such consistent performance curves you can almost do it by the numbers-- for example, you know that the ninth or tenth take is almost always the best one. In other cases, you just have to trust your feelings about when to do the boost. With a "double peak" performer, it's usually pretty obvious when the second peak is happening. That's when to do the level boost. For the "quick starter," I record a take or two, then bump up the volume a tad to see if I can get that "magic take" just before things start to fade. For the "rock steady," I'll say, "Okay, we have what we need, but let's do one more for luck," and turn up the volume a bit. Hey, if it gets a good performance, anything goes. BREAK TIME The performance curve also influences when the performer should take a short break, which most musicians feel the need to do occasionally during the course of a session. With the "quick starter" type, have the glass of water already set up next to the mic; once you start, you don't want to stop. The "long ramp-up" performer can sometimes benefit from working breaks into the process. This seems to impart a somewhat fresh perspective when the performer returns; they'll proceed in the direction they were going, but with a slightly different "vibe." This may give more options in the final composite performance (e.g., you can drop in the second verse from one of the post-break takes to add a bit more variety). With the "double-peak" type, the best place to work in a break is if the second peak is slow in coming. Sometimes a break will "break the ice" and cause the second peak to shake loose. If it doesn't, then it's probably best to move to a different tune. Sometimes the planets just are not in alignment to do the perfect performance, and part of producing is recognizing when that happens. THAT'S A TAKE Granted, there are a lot of variables, so the above are more guidelines to get you thinking than ironclad rules. Having said that, once you become aware of this phenomenon you might be surprised at how often it is an ironclad rule. Just like some people are night people and some are day people, it seems some people settle naturally into a performance curve that doesn't vary much, if at all. So next time you're recording, see if a performance curve manifests itself. You might really be able to use that knowledge to your advantage. 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.
  8. Going mixerless doesn't mean you have to give up hands-on control By Craig Anderton If you've decided that a mixerless studio (where you do all your mixing "inside the computer") works for you, you may be having second thoughts about giving up the wonderful hands-on, real time control offered by a hardware mixer . . . not to mention the pure joy of "fader slamming." But you can have your cake and eat it too, by adding a hardware control surface to your DAW software. Piloting your DAW with a control surface instead of a mouse can help make sessions flow faster and easier. These are good things. You won't necessarily want to give up your mouse -- much of the time, you'll likely have one hand on the mouse and the other on the control surface. But there's no denying the relief of watching motorized faders snap to attention as you make your automation moves, rather than tweaking levels one fader at a time with the mouse. Two Kinds of Support Most DAWs offer two kinds of control surface support: generic and supported. Generic means that the software doesn't really know what you're using, so you have to describe its characteristics -- how many faders it has, what you want to control with them, and the like. A supported surface will have code within a DAW that takes advantage of a specific feature set. For example, if a control surface has dedicated controls for EQ, the DAW maps its EQ to these controls. Note that support is up to the DAW manufacturer, not the company that makes the control surface. This is why a general purpose control surface may support certain functions with one piece of software, and different functions with another. Furthermore, some implementations are better than others. A control surface that's fully supported with one DAW may cover only a few functions with another; however, as most DAW manufacturers recognize the importance of control surfaces, the trend is to deliver ever-improving implementations. We also need to differentiate between "basic" control surfaces, which are essentially a collection of faders, knobs, and switches, and "interfacing" control surfaces, which may include mic preamps, audio switching, and other interfacing functions. This article's frame of reference is the basic types, but much of this applies to interfacing types as well. Better Living Through Emulation Maybe you've taken a real liking to a certain control surface (we'll call it control surface "X"), only to find that your DAW doesn't support it. Or does it? Some control surfaces can emulate other ones, which may provide a solution. For example, DAW "A" may not support control surface "X," but does support control surface "Y." If control surface "X" can emulate control surface "Y," then bingo -- the DAW and control surface are compatible. Sometimes you have more than one option, because the DAW will support both "X" and "Y." Which to choose? Try them both, and see which offers better support. Generally if the DAW has some type of "native" control surface -- e.g., it's made by the same company that makes the DAW -- you're better off emulating it than using some other mode of operation. Making Connections Although there are differences among units, most install pretty similarly. You'll want to check the unit's manual and "read me" files, but the general procedure involves two main tasks: getting the computer to recognize the control surface, then telling the software to use it. Connect the computer to the control surface via the method it supports (MIDI in/out, USB, FireWire, etc.). If your unit supports more than one protocol, generally MIDI is the least preferred choice. Apply power to the control surface before powering up the computer. Upon booting, you will likely need to install drivers or other accessory software. As with any hardware device, drivers are updated frequently so check the control surface manufacturer's web site for an updated driver. Installation procedures vary from controller to controller, so pay careful attention to any instructions regarding installation. After installing drivers and re-starting the computer, in most cases you're ready to add the control surface to your software. However, with USB or FireWire, it may be necessary to unplug and re-plug the connection while the computer is on for the control surface to be recognized. Open you DAW of choice. There will likely be some menu option for specifying a hardware control surface. This is a situation where you really want to look at the DAW's online help to see if there is any specific information regarding the control hardware you plan to use. You'll have to make sure the program you're using recognizes the ports into which the controller is plugged. If MIDI, make sure MIDI is enabled as an input. If USB, there will be virtual ports that you will be able to choose when setting up the host to accept the controller. Finally, there may be an additional operation inside your host, such as choosing the controller you plan to use from a list of controllers. At this point, your controller's faders should mirror the settings of the onscreen faders. If not, there is a connection or driver problem, or perhaps some aspect of using a control surface is not enabled in the host. If there's any sort of toolbar, menu item, or icon that corresponds to your controller, it may offer further options -- such as operating tips, the ability to switch "pages" on the controller, etc. Fader Away If you've read this far, you're obviously into the concept of control surfaces. So here's the most important thing you need to know: Control surfaces have a learning curve. Yes, they're physical devices, and you need to develop physical dexterity to "play" them correctly. This involves knowing where the controls are, and what parameters they affect (most control surfaces use bank-switching, so a limited number of faders can do the job of many). A control surface becomes beneficial only when using it is second nature -- and it can take weeks for that to happen. But overall, the learning process is well worth it.
  9. Great Sound Made More Attainable by Martin Patmos KEY NOTES Streamlined cosmetics help lower the cost Black Brass model is versatile enough to be a primary or secondary snare Walnut model's harmonics are rich and earthy Seam on outer ply of Maple drum is obvious Mapex's Black Panther snare drum line has been around long enough to establish a reputation as a quality series of instruments. The five most recently released models are essentially redesigned versions that, as the company states, “focus on playability rather than cosmetic frills.” Furthermore, these drums are designed to bring Black Panther snares to the customer at a more affordable price than before. By combining this friendlier price with great sound, these Black Panther snares have a lot to offer. Our review group consisted of three out of the five new models. They included a 61/2x14 Maple model (51/2x10 and 51/2x13 sizes are also available), a 61/2x14 Walnut model, and a 51/2x13 Brass snare finished in black chrome. So What's Different? The design differences for the new Black Panther models involve a few cosmetic changes, and little else. According to the Mapex Web site, the drums are “handcrafted to the same exacting specifications, and feature maximum sensitivity, remarkable clarity, and incredible resonance.” When it comes to those sound characteristics, I'd have to agree. But we'll get to that. Our review drums feature small, lightweight oval lugs that utilize one-point mounting. Chunky, easy-to-use throw-offs operate smoothly, silently drawing the snare wires into contact with the head. Snare-tension screws on both the throw and butt ends allow for fine adjustment of the 20-strand snares. Nylon washers, brass lug nuts, Remo heads, and 2.3-mm chrome hoops finish things off cleanly. And that's about the only noticeable difference. The hoops and hardware on other, more expensive Black Panther models may be black chrome, die-cast, or something else. Otherwise, these drums are cut from the same cloth, with their shell finishes complemented nicely by the traditional chrome hardware. Black Brass Smooth 1 mm–thick brass finished in black chrome makes up the shell of this snare, which possesses a sound that is simply addicting. The shell's smooth, mirror-like black finish was flawless inside and out, interrupted only by the Black Panther and serial number badges. With eight lugs per side, the drum tuned up easily and was responsive over a nice range of tensions, with the potential to be tightened up to a high crack, or tuned down to provide a deeper tone. At 51/2x13, the black Brass drum's size and tuning range gives it the potential to be played as either a primary or secondary snare drum. Its dark metallic harmonics rang fully near the edge of the batter, while the center region produced a dry crispness that was wonderful for snare chatter, ghost notes, and complex sticking. The response was immediate throughout a good dynamic range. Honestly, I had a lot of fun with this snare, and found it equally effective for chatter against a jazz ride or syncopated figures over a double bass pattern. Addicting indeed. Maple The new 61/2x14 Maple Black Panther proved to be an exceptional contender among maple snares. A natural wax finish enhances the yellow wood tones of the 5.1-mm, 6-ply shell. The shell itself is strong, resonant, and finely crafted. Just tapping around the shell with my finger revealed some nice, resonant sound characteristics. The exterior of the shell was super-smooth, while the sanded, unfinished interior had the faintest variations when running my finger across the grain. The bearing edges were immaculately smooth and perfect all the way around. My only criticism is that the seam of the outer ply, which runs through the serial number badge at an angle, is quite noticeable due to the wood's light color. The ten-lug drum tuned up easily and was responsive at different tensions, offering a range of tonal possibilities. All the classic sound characteristics of a maple drum shone through, including plenty of presence, full harmonics, a woody tone, and great response at different dynamic levels. When tuned up, the drum was crisp, articulate, and warm. Tuning it lower presented a solid fatness. In short, the Maple Black Panther sounds like an outstanding all-around snare. Walnut The Black Panther Walnut snare is a beautiful instrument to play and to look at. Having never played a walnut drum before, I was eager to check this one out. Boy, did I get a thrill. If you've never heard a walnut drum before, imagine a maple drum without the higher harmonic overtones and comparative brightness, and instead with a deeper overall tone, rich lower and middle harmonics, and a darker, earthier presence. With these sound characteristics, the Black Panther Walnut's dark harmonics rang true throughout a range of tensions from low to high. The drum's earthy character was exceptionally dynamic and sensitive, offering both rich brushwork and the most astounding, thunderous rimshot crack I've ever heard. Seriously, the first time I laid one down I was in such awe that I missed a beat in disbelief. The 5.1-mm, 6-ply shell was well crafted, solid, and resonant. The natural wax finish applied to the smooth exterior emphasized the rich dark brown color and the finer lines in the wood grain. Even though walnut is a rougher wood than maple, the bearing edge was still quite smooth, while the finely sanded interior maintained a slight roughness that ought to subtly enhance the sound in the drum's interior. As with the other drums, the Walnut tuned up easily with ten lugs per side. However, I'd recommend experimenting with it at first to get a feel for the drum's harmonics. As to the overall sound of the Walnut, I loved the earthy tone and power it provided. With these characteristics, it blended well with darker-sounding cymbals and didn't harmonize as nicely with really bright ones. Owing to its unique sound, its volume, and its versatility, I loved this drum. Conclusion I found each of the new Black Panther snare drums to be well-made, expressive, and versatile instruments that would work well in a great variety of situations. By cutting costs as the result of a few simple hardware alterations, Mapex has helped to bring the price of these drums within the reach of more drummers. Plus, whether against black metal, dark brown wood tones, or light yellow ones, that shiny chrome hardware goes with everything. Whether you're looking to upgrade your snare or just looking for a new voice, these new Black Panther snares are worth investigating. THE NUMBERS 51/2x13 Brass (BPBR3551C) $509 61/2x14 Maple (BPML4650CWN) $539 61/2x14 Walnut (BPWT4650CWN) $559 (615) 793-2050, www.mapexdrums.com © 2006 MODERN DRUMMER Publications, Inc. All rights reserved. Reproduction without the permission of the publisher is prohibited.
  10. …And Mastering Loses. But There Are Ways to Help Even the Score by Craig Anderton One of the main functions of mastering is to make a mix transportable: In other words, it should sound good over any playback system. But what works against this is that playback systems vary wildly. Although we no longer have to worry about all the violence that happens to a signal courtesy of analog playback media (cassettes and vinyl, which offered a nearly infinite number of possibilities to screw up sound), speakers and their associated enclosures are like very mischievous equalizers. For a mix to be truly transportable, it needs to sound "right" when played over: Great audiophile systems with flat response, superb definition, and state-of-the-art speakers. Boomboxes that have the "MegaGigaSuperBassBoost" button pushed in, which hype the speaker's low end to an absurd degree. Boomboxes that don't have the "MegaGigaSuperBassBoost" button pushed in, resulting in a low end that's thinner than Kate Moss on a hunger strike. Earbuds on portable music devices, whose response is basically luck of the draw. Car radios. Let's not even go there, even though we have to. Tabletop radios, which have the same type of issues as boomboxes. What do all these different systems have in common? Speakers (even if they're miniature versions that fit into ear buds). How do we compensate for differences among speakers during the mastering (and for that matter, mixing) process? WELCOME TO FLATLAND Edwin Abbott's Flatland is one of the greatest mathematical fantasy books of all time, albeit in an admittedly uncrowded field. But when it comes to speakers, "flatland" is just that: A fantasy. Look at the response curve for even the finest speakers that money can buy, and you'll see something that approximates a relief map of the Alps. This response (see Fig. 1) undergoes further degradation when interacting with the listener's room, which is seldom acoustically treated; but let's pretend that's not an issue, as it multiplies the variables into the world of Really Really Big Numbers. Fig. 1: Here's a typical frequency response curve for a high-quality, pro audio-oriented two-way speaker with active crossover. Although it's anything but flat, this is actually better than average - yes, this is what you're up against. Differences among speakers exist over their entire range: Lows, mids, and highs. So, over the years, mastering engineers have recognized that the only want to deal with this madness is to create a recording with the flattest, most "average" response possible. That way, it will sound only a little bit "wrong" over every system, rather than okay on some systems andway wrong on others. (The exception is that of the audiophile with the really flat system - who after putting the requisite time, expense, and effort into assembling a great system,should be entitled to the best possible sound.) It's difficult to create a truly average midrange response, because that's just one of the places where speakers exhibit significant differences. (An aside: I always get a kick out of speaker reviewers who breathlessly exclaim that a particular speaker "revealed things I'd never heard before in my favorite recordings!" This isn't surprising, because any speaker will reveal things you've never heard before, as it's basically EQing the recording differently.) High frequencies are a different matter. The energy in real music tends to drop off fairly rapidly above 5kHz, so really, what we want is a "sensation" of brightness. You're not going to get the huge peaks caused by notes piling on one another, because there just isn't that much energy up there. A little bit of a boost in the "air" range above 10kHz will do wonders for making a mix transportable, as the tweeters open up a bit more. As a bonus, most playback systems have a treble control that can be trimmed or boosted according to the listener's preferences, based on their acoustics and how shot their hearing is from going to too many concerts without hearing protection. WORKING TRANSPORTABILITY INTO THE MIX Ideally, your mixes should already be pretty close to perfect before they get to the mastering engineer; and there are some steps you can take while recording and mixing to make the master more transportable. Instruments with lots of low frequency content, like kick drum and bass, will not translate well over systems with poor low frequency response. To allow them to be heard on bass-shy systems, use midrange EQ to bring out pick noise in the case of bass, or the "thwack" of a beater with acoustic kick. With electronic kicks, there are a variety of ways to accentuate a "click" at the beginning of a note. Psycho-acoustically, your brain will tend to "fill in" the sub-harmonics. Speaking of sub-harmonics, some sub-harmonic, sinewave-type basses (as often used in drum 'n' bass) will never make it through small speakers. Try using a waveform with more harmonics, and close down a lowpass filter to get a bassier tone - but which still has some harmonics present. Finally, although it's been said a million times before, let me be number 1,000,001: Acoustical treatment is your friend. You can neither record nor mix sounds properly in a room where the acoustics are adding the equivalent of a random EQ with insanely steep peaks and notches. Although a mastering engineer will try to deal with this, it's not easy and not always successful. Speakers tend to have the hardest time maintaining a flat response below about 80Hz. Here we're up against the laws of physics, as bass frequencies have really long wavelengths and to re-create them, you need to push a lot of air. Speakers that fit in the average living room, or boombox for that matter, simply can't push enough air at really low frequencies. This is one element that separates the big bucks speakers from the pretenders: How low they can go without giving up. This is also why many devices have bass boost switches, although that's not quite the same as having "real" bass. An analogy: A woman puts on great eye makeup that makes her eyes look bigger, but really, they aren't any bigger. To make matters worse, we have two other bass range issues. For one, the bass response of our ears falls off at lower volumes (the infamous "Fletcher-Munson Curve"), so our ears' deficient frequency response interacts with the speakers' deficient frequency response. The second is in the recording itself. Unless the studio has really great acoustics, or all instruments were taken direct, there will likely be frequency response anomalies due to room acoustics that cause peaks and dips in the bass range. So in a worse-case scenario, the bass peak in the recording process doesn't get caught while mastering, and plays back though a speaker system that has a resonance at that frequency, which interacts with the room resonance in which the speaker lives. Nasty. THAT SOUNDS HOPELESS AND DEPRESSING. WHAT'S THE SOLUTION? There is no solution, so you're correct in feeling hopeless and depressed. But you can try to come as close as possible to the ideal. If during mastering you can smooth out the bass response to have no significant peaks and dips (except where you actually want them, like a big peak on a techno kick drum), you'll have gone a long way toward making a transportable master. It's even better if you can take care of some of these issues in the mix (see sidebar). Because our ears' response gets iffy in the bass range, it can really help to have some visual feedback as to what's going on in the bass range. No matter how good an engineer you are, it's really hard to quantify a 1.5dB peak at 72Hz solely by listening. I find a good spectrum analyzer that can display an average response is extremely helpful. Why average? Because there will always be natural response peaks and dips. What we're looking for is a pattern of build-ups and anomalies which, if played back through a system with a peak or dip at that same frequency, will exaggerate the problem even more. My favorite tool these days for fixing this type of problem is the Har-Bal EQ/mastering program. It provides a very unambiguous look at the bass end, and you can use a simple "pencil" tool to draw out peaks and dips, thus smoothing out the low end (Fig. 2). Since doing this, my mastering clients have all - without any prompting - commented that the mixes are more transportable. Granted there are more factors that just bass response that contribute to making a mix transportable, but bass is an important factor. Fig. 2: Here, the Har-Bal program is smoothing out a song's bass response; for clarity, the window has been trimmed to show only the region below about 500Hz. The upper gray line indicates the original peak response, and the lower gray line, the average response. The yellow line indicates the peak response after "drawing" a smoother bass response, while the green line indicates the average response after smoothing. OTHER TOOLS Another great mastering tool that helps compensate for speaker anomalies is a good multiband compressor (Fig. 3). Some subtle midrange compression can bring down peaks and raise valleys that might otherwise be emphasized or de-emphasized by a speaker. In the high end, you can add no significant compression, but just boost the overall level a bit. Meanwhile, in the 300-400Hz range, you can lower the level a bit (without compression) to "tighten up" the sound somewhat, as that's often where you'll find a bit of "mud." Meanwhile, adding compression in the bass range helps even out the response. Fig. 3: The Multiband Dynamics Processor in iZotope's Ozone has been set up to add some mild compression in the low end; this smooths out the bass response a bit, which helps make the mastered recording more "transportable." Eventually, through proper use of equalization and dynamics control, it's possible to create a master that keeps unwanted peaks and dips under control. And when you've accomplished that, you're in good shape. As a final reality check, it's worth playing your master over varying systems just to make sure you've come as close as possible to the ideal. If your mix sounds a little fatter than usual on systems that normally sound a bit on the thin side, thinner than usual on systems that normally sound annoyingly muddy, and perfect on really good systems, great. No matter what speakers the recording plays back over, you've probably done about as good a job as you can do. Craig Anderton is Editor in Chief of Harmony Central and Executive Editor of Electronic Musician magazine. 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. They used to just record audio, but today's all-in-one studios can also do mixing, CD burning, and even mastering. By Craig Anderton Back in the analog age, a "studio in a box" simply had a mixer and multitrack recorder. But in the digital age, you're likely to find effects, automation (snapshot, and on more expensive boxes, even moving faders), connections for external storage, and now, even CD-R or CD-RW drives so you can take a project from plugging in your first instrument to burning a CD for the band members. As equipment has gotten better, though, so have the technical standards by which music is judged -- when you burn a CD, it had better stack up to commercially-available ones. Unless you have mixing chops of the gods, the only way to get that kind of pro veneer is to master your final mix. As a result, some all-in-one studio products -- especially those with recordable CD drives -- now include effects algorithms designed specifically for mastering. As one example, let's look at Korg's D1600, which includes several tools to help you create the best possible-sounding final product. In addition to traditional insert effects, the D1600 also includes a Final Effect slot where you can insert effects that shape the overall mix. The Two Crucial Mastering Effects For starters, the D1600 includes the two most needed mastering effects: multiband parametric equalization to correct for frequency response problems (e.g. increasing articulation through a slight upper midrange boost, or reducing "muddiness" by trimming back the bass), and dynamics control. Dynamics is the key to obtaining today's controversial "hot" sound, which places great importance on creating as loud a mix as possible. Using EQ on your entire mix may be unnecessary if you properly equalize individual tracks first. In addition to the channel EQ, which is good for general tone-shaping, Insert Effect #I027 (4-band parametric EQ) can provide very sophisticated EQ control for individual channels. To process the overall tonal balance, you can insert #F010 (parametric EQ) as the Final Effect. However, as the D1600's mastering-oriented dynamics processors also include basic EQ options, you'll probably want to use the Final Effect slot for dynamics processing so you can control both dynamics and EQ simultaneously. Dynamic Controls Any mix's maximum level depends on the available headroom -- peaks that exceed it create distortion. Dynamics control reduces peaks, thus opening up more headroom. You can then increase the overall level until once again, the peaks hit the maximum possible level. The D1600 offers several dynamics effect presets (#F001 - #F008) for the Final Effect slot. The first four are based on stereo compression (Effect DY1). Compression causes a change in input level to create a smaller change in output level. With stereo mixes, compression gives a punchy, "pop music" sound. The other four presets are based on stereo limiting (Effect DY2). A limiter limits peaks, but doesn't affect the dynamics of lower-level signals. Subtle settings give a more "natural" sound than compression, so limiting is a favorite for acoustic music (jazz, classical, etc.). Limiting also acts a "safety valve" to make sure signals don't exceed the maximum available headroom. Extreme amounts of limiting or compression produce a very "hot," loud master. This is popular with dance music, because DJs want the minimize track-to-track level variations as they segue from one tune to another. However, you have to be careful. Dynamics processors can't repeal the laws of physics; push them too hard, and you'll end up with distortion. Salvage Jobs Sometimes, mastering engineers are called upon to do more of a salvage job than a simple enhancement. Live recordings, for example, may need sophisticated EQ anddynamics, which is more than you can fit into the Final Effect slot. The solution is use the D1600 more as a signal processor than a recording device. For example, run your two-track master into two inputs of the D1600, and use parametric EQ for the insert effect slot. Multiband limiting (DY3) might also be appropriate as an insert effect if the dynamics vary too much. Next, for the Final Effect, insert dynamics processing to produce a smooth, hot-sounding master...then burn your CD. It's amazing how much some judicious processing can enhance material you thought might not be even useable. Be aware, though, that mastering is a subtle process. A few dB or EQ or dynamics control is usually all you need. Higher amounts will unbalance the overall sound -- if you boost the bass a lot, for example, then the treble will sound thin by comparison. This is why mastering is considered such an art: it's all about subtle changes adding up to a major improvement in the sound. There's no guarantee that using mastering tools is going to produce a great master recording, any more than buying a guitar is going to make you a great guitarist. In either case, practice makes perfect. But in the case of all-in-one studios, it's at least nice to know that the tools are there for you to use.
  12. Mastering is a crucial process, but it's not always all that well understood by the average musician…so let's deal with some of the basic issues. By Craig Anderton Your tunes are done, and you've decided it's time to create a CD — which brings you to the subject of mastering, where all the tunes are assembled and optimized for the best possible sound. You really don't want to make any mistakes at this crucial stage. Indeed. Mastering can make or break a record, so there's a lot of interest in doing it right. Here are the ten most common questions I hear from people who are about to get their work mastered. Q. What's the best piece of gear for giving me a professional, "radio-ready" sound? A. The best piece of gear is a professional mastering engineer who has done this process before for hundreds, if not thousands, of recordings. Q. So do I just send an audio CD with all the cuts, and the engineer masters them? A. That's one option, but certainly not the most desirable. Although you should always check with the engineer for specific requirements, if you recorded your music in high-resolution audio, then it's best to provide those high-resolution mixes, as WAV or AIFF files. The mastering engineer will likely do some processing, and 24-bit files give more "calculational headroom." Q. Wouldn't it be better to send a dithered version of the 24-bit files, as the files are going to end up as 16 bits on a CD anyway? A. No. Dither is always applied as the very last stage of mastering, when the higher resolution signal gets downsized to the 16 bits required by Red Book audio. Steinberg's Wavelab includes excellent dithering options, but don't apply these if you plan to hand off your file to a mastering engineer. Q. I want a couple of cuts to crossfade into each other. Should I do the crossfading myself and send the combined cut? A. Probably not. Fades can be dicey, and again, the mastering engineer will likely have tools that provide the best possible audio characteristics when creating fades. Also, that will insure dithering happens to the combined file — you don't want to dither two files, then crossfade them. Just make sure that you include full documentation on where you want the fade to begin and end for the two cuts. Q. So does the same logic applies to fades in general? A. Yes. Ideally you should send files that don't contain fades, and let them be added during the mastering process for the same reasons described above. Again, include documentation as to where you want the fades to occur, or include a version with the fades so the engineer can hear what you want. Q. I really like being able to see the names of tunes on my CD player. How does that work? A. CD Text is not in the official Red Book Audio CD standard, but it has become a sort of de facto standard and many programs include the ability to add it. If you want to include CD Text, provide the name for each song, and check it over carefully for typos, inconsistent capitalization, etc. Better yet, have two people look it over before sending it, because the engineer will enter exactly what you send. A group once sent me an album for mastering and one of the tunes was listed as "Who's Their?" So that's what I entered; I figured they were trying to be cute. But they meant to say "Who's There?" Proof carefully! When you burn a CD in Magix Samplitude, you can optionally enter CD Text information for the entire album as well as individual tracks. Q. I've heard it's a good idea not to trim off the beginning of tunes before sending them to a mastering engineer. But that doesn't make any sense. If I want a tune to start at a specific place, why shouldn't I trim it to the desired start point? A. As mastering engineer Bob Katz says, "Editing is like whittling soap, so it's best to leave a lot more soap on and let a pro produce the very best beginning and ending for the tune. Avoid the problems caused by an overanxious mixing engineer cutting off valuable sound, which you may regret later. Sometimes it's even better to leave room tone or low level noise between tunes; or keep the singer's intake of breath, or the delicate sound of anticipation before the downbeat. Especially for acoustic music and often for electric-based music, it is disturbing to cut off the air' prior to the downbeat. A well-equipped mastering workstation and experienced mastering engineer working in a controlled acoustic environment will know just the right speed and shape of fadeup/fadedown to use on a piece for the smoothest, most natural transition." Another consideration involves the possible need for noise reduction. Sometimes there may be a slight hiss, hum, or other constant noise at a very low level. If the engineer can obtain a clean "sample" of this sound, it can be loaded into a noise reduction program that mathematically subtracts the noise from the track. Even if this noise is way down in level, removing it can improve the sound in a subtle way by opening up the soundstage and improving stereo separation. Don't try removing the noise yourself prior to sending in the files — as with the other gear, the odds are a pro mastering engineer will have better tools. Q. I'm on a really tight budget. What if I do some "pre-mastering" myself, like adding a little compression or EQ? That way the mastering engineer won't have to spend as much time on it, and I can save some money. A. Don't do anything to the raw mixes. The most difficult mastering jobs I do (and the ones which take the most time) are "salvage jobs" where someone tried to master the files, and I have to figure a way to "undo" some of the damage. A good example is if someone used standard compression instead of multiband compression, and there's pumping or breathing. There's no way to "undo" that (well, not that I know of), and if you add more compression, it will accentuate the problem even more. Q.Well I should at least normalize the tracks, right? A. No, don't do that either. It doesn't matter if there are level fluctuations among the various tunes, as that will be sorted out during the mastering and assembly process. The reason for hiring professional mastering engineers is you want them to do their magic, so give them the space to do so. Furthermore, normalization adds one more stage of potentially degrading DSP. Given that you may also need to adjust levels after the mastering process, it makes no sense to adjust levels before the mastering process. Q. I do some mastering at home, and I'm actually getting pretty good at it. What if I include an example I've done at home to give an idea of the sound I'm looking for? A. There's no harm in doing that at all. You should definitely discuss with the engineer what you expect. For example, I worked with one artist who wanted his CD to be really, really loud, as is the fashion these days. But his recordings were very open, with good dynamics, and I hated to throw that away. Besides, many listeners want to hear dynamics, because having contrast among the various sections makes for a more satisfying listening experience. It's also important to remember that overcompressed recordings sound much worse when passed through the processing done by radio stations, because this compresses the sound even further. So we compromised, and I found a "sweet spot" between making something loud enough to satisfy him, but dynamic enough to sound good over the radio and provide a good listening experience. Acknowledgement: Thanks to mastering engineer Bob Katz of Digital Domain (www.digido.com), and author of the book "Mastering Audio: The Art And the Science" (Focal Press, ISBN 0-240-80545-3), for giving this article a reality check.
  13. If you're going to try your hand at mastering, this is required reading. By Craig Anderton Make very small changes when EQing, because an increase or decrease in one frequency range has repercussions elsewhere. For example, if you boost the treble, the bass becomes less prominent. It's amazing how even a 0.5dB change can make a noticeable difference. Adjust EQ to what sounds right, thenhalve the amount of boost or cut you added. This gives your ear a chance to get acclimated to the change in sound. You can decide later whether you want something more drastic. Always save and back up your original unmastered, 2-track or surround mix before you start mastering, and work on a copy. Duplicators will often reject CDs if the level hits 0 for several samples in a row. Yet these very short overloads may not be objectionable to the listener. To get around this problem, after assembling the entire CD, normalize it to -0.1 dB. This leaves just enough headroom that the CD won't be rejected for "overs." Normalizing an entire CD to something less than 0 can reduce the chance of rejection from a CD duplicator. However, you probably don't want to normalize each track individually, as that could interfere with the song-to-song level cosistency. Don't add song fade-ins or -outs when mixing. Tunes may need a longer or shorter fade than anticipated. If you build a fade into the mix, you can only make it shorter. Don't do any more processing than needed. These operations sometimes round off numbers; if these errors accumulate, there can be an audible "fuzziness." While this was mostly a problem with 16-bit systems -- 32-bit floating resolution has given a lot more operational headroom -- it's still a good idea to keep any processing to a minimum. When mastering with a digital audio editor, save the setup you use (plug-ins, levels, etc.) as a preset. For example, Wavelab has a Master Section Presets option. If the vibe of the CD changes over the course of mastering, you can go back to earlier tunes, recall the preset, and make a few tweaks rather than start over from scratch. Saving often-used combinations of effects can make it easier to get up to speed if you need to make any revisions. If possible, test the album's song order before you start mastering. Use your CD burning program or Apple iTunes-type program to assemble a "playlist" of tunes, and record it to Red Book CD, portable MP3 player, Minidisc, etc. Live with the order for a few days so you're sure everything flows smoothly. Use normalization sparingly. Normalization sounds like a great idea: click a button to amplify your signal so that the peaks just reach the maximum available dynamic range. But music doesn't work like that. A heavily compressed tune may seem much louder than a less compressed tune whose peaks are actually higher. Think high resolution audio at all times. Save your final mastered versions in at least 24-bit resolution, even if the target playback medium is a standard 16-bit CD. Then apply dithering to the high resolution file to create the best-sounding 16-bit file. If you mix to DAT or transfer tunes to DAT prior to sending them to a duplicating house, record a minute or two of "digital black" (silence) at the tape's beginning. This gets past the part of the tape that is most likely to have questionable surface characteristics. You can then transfer the DAT digitally to your computer for editing. Also, eject any digital tape in a space between songs. Should any tape damage occur while threading or unthreading, your song will be spared.
  14. The Masterlink Has Been a Workhorse in Studios - Here's How to Get More Out of It By Dan Tinen and Craig Anderton The Alesis MasterLink was one of the first stand-alone CD burners designed specifically for mastering; it incorporates a host of mastering-oriented DSP, and can record/play back CD-Rs in 24-bit as well as 16-bit resolution. These 10 tips will show you to get even more out of this versatile device. The Multiple-Disc, Multiple-Playlist Mix Because standard CD-Rs can hold a maximum of 650 MB (or 72 minutes of 16/44.1 audio), if you want to prepare an hour-long DVD that uses 24-bit resolution, you'll need to split the "album" into at least two CD-24 format CDs worth of material (24-bit data storage uses up 50% more capacity than 16-bit material). Also prepare a single, 16 bit/44.1kHz Red Book CD to show the DVD mastering lab how you want the final project assembled. You'll end up with three or more CDs to send: CD24 (high resolution) with the DVD's first half CD24 (high resolution) with the DVD's second half Red Book CD with all songs at CD-resolution, in order, with spacing and fades Check with your DVD mastering engineer to find out requirements for received material; note that CD24 disks do not require a MasterLink for playback, as they store AIFF files that can be opened by most digital audio editors. Incidentally, mastering engineers may request that the DVD tracks be submitted to them "raw," with no DSP or fades, so they can do that work on their (very expensive) workstation. If you add the types of DSP and fades you want to the Red Book CD, it can help get across what type of sound you'd like to hear on the final DVD. Fade Curve Options MasterLink includes three fade curve options. Each one is useful with different types of material. Lg1 (concave logarithmic) is best for most music. The logarithmic curve fades abruptly at first, then decays slowly. Lg2 (convex logarithmic) keeps the volume up as long as possible, then does a quick fade. For example, suppose you want to fade out a 30-second commercial, but the narrator goes almost to the end. Add a short (500 ms) fade with the Lg2 curve. Lin (linear) can seem abrupt compared to a logarithmic fade. However, if a song already contains a fade made during the original mix, the Lin curve can shorten the fade but preserve the existing fade's curve. Audio/Metering Mismatches Because of the MasterLink's "look ahead" feature (i.e., it analyzes the signal before applying compression to catch any transients), when auditioning DSP it takes about 500 msec to hear any changes made to the EQ or compression settings. Therefore, the display will change before you hear the sound that corresponds to that change. Using the Compressor's Key Function The Key option can compare either the right, left, or both signals to the specified threshold when determining whether to apply compression. L & R is the most common choice, as linking the left and right channels together allows changes in one channel to affect the other, which preserves stereo imaging. However, suppose the channels are out of balance because the left channel has higher peaks than the right. Set the threshold just above the right channel's highest peaks, and key to left channel only. When the left channel signal exceeds the threshold, compression kicks in on the left channel. As the right channel signal does not exceed the threshold, it will not be compressed. The Importance of Limiter Release With short release times, the limiter tracks every little change in level, producing a poten¬tially uneven or "choppy" effect. However, overly-long release times can result in dynamics that sound unfocused and mushy. If the dynamics are crisp and well-defined, and the transitions as the MasterLink goes in and out of limiting sound smooth, then you've found the optimum release time. Note that the rate at which the compressor's gain reduction meter decays provides useful information about the compressor's release time setting. If the gain reduction meter darts rapidly among segments, the release time is relatively fast. If the meter transitions more slowly between segments, the release time is slower. Don't Add Fade Ins and Outs While Mixing Fades are best left for the MasterLink's mastering process, because with digital fades, the change in level is not continuous, but goes through a series of very tiny steps. Inexpensive digital gear often has coarser steps than the MasterLink, which uses a high-resolution fade algorithm to minimize the gap between steps. This produces smoother, "sweeter" fades. What's with the Multiple Levels There are three principal ways to change levels: The overall level that's set for the entire track Gain applied in the compressor Gain applied by normalization Choosing the correct place to change level is like gain-staging in a mixer, with the three MasterLink controls above corresponding respectively to a mixer's input gain, channel volume, and master volume. The track level shown on the top line of Playlist Edit mode is the first in the chain. Changing this affects what the compressor "sees," which lowers or raises the threshold where the compressor starts to work. The normalize setting is the last in the chain, and makes up for any gains or losses in the compressor, EQ, and limiter before it. Want a Hot Sound? Try the Least Obtrusive DSP First Let's assume a CD doesn't sound "loud" enough, and you want to make it louder with dynamics processing. Try the DSP effects in the following order: Normalize. This doesn't affect dynamics at all, it just raises the overall level to use up the maximum available headroom. If this delivers the results you want, there's no need to go further. Limiter. This affects only the peaks of the signal, leaving the rest pretty much intact. Light amounts of limiting (e.g., a threshold between --6.00 and 0.00) can increase the overall loudness without perceptibly altering the song. Compression. Compression affects the signal the most. Use this if you want a really "hot" CD (e.g., dance music), or if the dynamics need to be restricted more drastically than what you can obtain with limiting. Use a Safety Net If mix from your multitrack directly into the MasterLink hard disk, until you burn a CD, there's no backup for that mix. If the hard drive fails, a power surge takes out your gear, or an errant pet knocks over the MasterLink onto a concrete floor, you may lose that mix forever. To be safe, mix to the MasterLink and back up to CD as soon as the mix is complete. Choosing Sample Rate Rate and Word Length When Mixing Many engineers use MasterLink as a "drop-in" replacement for DAT, and mix directly to it from a mixing console. As MasterLink can record at different resolutions and sampling rates, here are some guidelines on how to set the MasterLink's parameters, depending on the type of source material that feeds it. If your mixing console is analog: Use a 20 bit word length. The noise floor that results from the console itself, microphones, preamps, and all the components connected to it is much higher than the 20-bit encoding of a digital recorder. You can of course use 24-bit, but you're just wasting disc space to store random numbers well below audio thresholds. Use a sampling rate of 88.2 kHz. Recording at 88.2 kHz will give you and future listeners on DVD the chance to hear any high frequencies in the 20-40 kHz range that may have been recorded by analog equipment (although very few microphones actually produce frequencies up there, and even fewer speakers can reproduce them). Professional mastering engineers like to receive 88.2 material, because it allows them to use EQ in the top audible octave (10-20 kHz) without distorting the anti-aliasing filter required for 44.1 kHz operation. An additional advantage is compatibility with today's CDs, as sample rate conversion from 88.2 to 44.1 causes no sonic artifacts. This is because conversion simply involves taking every other sample, as if the source material was originally recorded at the lower rate. However, if saving space is important, then you can always go with 44.1 kHz. If your mixing console is digital: Use a 24 bit word length. While the noise level of analog input signals to the digital console is (at best) at the 18 to 19-bit level, the digital signal processing used by the console itself will generate usable information down to the 21-22-bit level. (When you lower a fader of a digital mixer below unity gain, it is mathematically dividing the numbers and creating significant bits of resolution: a 19-bit signal running through a digital fader set to --12 dB becomes a 21-bit signal.) Use a sampling rate of 44.1 kHz. If your digital console has 88.2 kHz capability, you'll of course use that. But for the 99% of digital consoles that are limited to 44.1 or 48 kHz, you may as well use the CD-compatible 44.1 kHz rate to avoid the sample rate conversion process to go from 48 to 44.1 kHz. Although modern sample rate conversion algorithms don't degrade the sound like earlier algorithms did, it's always good practice not to add unneeded processing.
  15. Tweak your way to a black belt Sonar experience By Craig Anderton It's time to teach Sonar some new tricks -- so grab your mouse, boot up your computer, and let's go. SNAPPIER GRAPHICS If a Project uses a lot (and I mean a lot) of digital audio Clips, particularly Groove Clips, the program can bog down when moving/editing Clips. To fix this, increase Sonar's picture buffer cache size. With Sonar closed, use Notepad to open the AUD.INI file (normally in theProgram Files > Cakewalk > Sonar directory, or just use the Windows Search function). Locate the section. Underneath it is the PicCacheMB parameter. Set it to a high value, like 500. Save the AUD.INI file. SAMPLE-ACCURATE CLIP LENGTHS You can trim clip lengths with single-sample accuracy -- very handy when using Sonar to create loops that must be an exact number of beats. Right-clicking on any audio file gives you the opportunity to choose the Split Clips command. Right-click on the Clip and choose Split. The Split Clips dialog box allows not only sample-accurate splitting, but repeated splitting. Select Samples in the Time Format field. Enter the length in samples in the Split at Time field. Click on "OK." The Clip splits at the specified point. PREVENT (AUDIO) ENGINE SHUTDOWN If occasional CPU spikes shut down the audio engine, requiring you to click on the Engine icon to wake it up again, try this. Open the AUD.INI file in Notepad (see the first tip). Locate the section. Make sure the StopOnEmptyPlayQueue parameter is set to 0, not 1. REWIRE TWEAK While using ReWire, switching between Sonar and the ReWire-compatible application can mute Sonar's outputs if the two programs' drivers are shared. A workaround is to shift the focus back to Sonar or initiate playback on the ReWire slave or master, but here's an easy fix. Go to the Options > Audio > Advanced tab. Locate the Playback and Recording section. Uncheck Share Drivers with Other Programs. THE LOOP EXPLORER GOES MULTITRACK You can audition multiple loops simultaneously in the Loop Explorer, but note two constraints: The loops must all be in the same folder. Unless the loops are "acidized" or Groove Clip loops, they'll need to be at the same tempo and key to play together. Multiple file auditioning is particularly useful with samples from "construction kit" sample CDs (e.g., those that break loops down into individual parts), as you can hear how the parts work together. Go View > Loop Explorer window. Enable Auto-Preview so as soon as you select a loop you'll hear it play with the others. Click on the first loop to select it. Click on the Loop Explorer's Play button. To add a loop, Ctrl-click on another loop in the Loop Explorer's list of files. To de-select an already-selected loop, Ctrl-click on it again. If you've selected loops you want to use, dragging the entire group over to the track view pane loads them into the project. GIVE EDITING THE SLIP Slip-editing, where you drag a Clip's start or end to change length, is non-destructive. To remove the hidden audio permanently: Select the clip. Go Edit > Apply Trimming. Note: Doing this with a Groove Clip will convert it back to a standard clip.
  16. Sounds Are All Around Us, and They're Yours for the Taking By Craig Anderton They're out there, and they're everywhere: Sounds. And they're just waiting for you...all you need to do is find them, which is maybe why they're called "found sounds." You've probably heard more found sounds in recordings than you realize. Although capturing and warping sounds was a mainstay of classic electronic music, artists as diverse as Pink Floyd, John Cage, the Beatles, and a zillion techno producers have all used found sounds in musical - and not so musical - ways. Whether it's the nature sounds behind a new age recording, sound effects in an audio-for-video production, "quotes" from old movies in a dance floor hit, or even swarms of bees providing a menacing backdrop to the TV show "Cold Case," having a collection of unique samples sitting around can come in very handy. YOU WON'T GET THE SOUND IF YOU DON'T HAVE THE RECORDER Capturing sounds requires a certain attitude that's very much like a good photographer, who never leaves the house without a camera. If you're serious about treating the world as your waveform, you need the audio equivalent of a camera: Something simple, small, and convenient enough to use that you actually use it. Here are some of the options. Fig. 1: Sony's PCM-D1 was one of the first high-end, solid-state field recorders that was also well-suited to musicians. Solid-state recorder. Typical models are made by Roland, TASCAM, Sony, M-Audio, Yamaha, Olympus, Zoom, Marantz, Fostex, and others (Fig. 1). These have no moving parts, because you save audio to a memory cartridge - so they're totally quiet. Hard disk-based recorders. These move up a notch in terms of storage, but go through batteries faster and make some noise (although the subcompact hard drives they use are pretty quiet). Korg's MR-1 is a good example of this kind of device. MP3 player with voice recording. Sure, it has a little tiny mic designed to capture audio notes like "Don't forget to pick up the cat litter." But when all else fails, they work - and you can capture some gloriously lo-fi samples that are the perfect complement to electronically-inclined music. Cell phone. Some cell phones have voice recording options; if not, call yourself and record into your voicemail. Minidisc. I admit to a bias toward these "always a bridesmaid, never a bride" devices because they've served me well for a decade. They sound good, are convenient, get good battery life, and of course, are terminally unhip in the Age of iPod. Video camcorder. The audio recorder inside a tape-or SD cartridge-based camcorder can often record with excellent fidelity. Although clunkier than carrying around a tiny solid-state recorder, you can get good fidelity and video to boot. I hardly go anywhere without at least one of these - just in case. ACCESSORIZE, BABY! If you're just trying to grab a few sounds here and there, a recorder is all you need. But if you're on a serious sonic safari, you'll need accessories. Additional recording media. This will also determine what type of recorder to take with you. If the only way to offload files is via a computer, then you gotta bring a computer. With cartridge-based solid state recorders, you'll need to bring plenty of memory cartridges - unless you've also brought a computer to which you can transfer files. And for Minidisc, bring extra discs. Additional power. You don't want your batteries to die just as you're capturing the sound of a lifetime. That's why I always like devices that have rechargeable batteries, but the option to slip in additional batteries if needed. (For serious battery power, make a box with D cells or lantern batteries, and in most cases you can go through the AC adapter input. Once while in Alaska sampling whales, I had a Casio DAT recorder (really) and because it went through batteries like a glutton through filet mignon, I brought several 6V lantern batteries with me - a pain in the butt, but I got the samples. Mics. The internal mics on many recorders are surprisingly good, but it's worth bringing some external mics. This is particularly important if your recorder of choice has moving parts; for quiet sounds, you'll want the mic as far away from the recorder as possible. Plastic freezer bag. Fold this up, and put it in your pocket. If there's rain, oceanspray, or other environmental nastiness, put the recorder in the bag and seal it up. If you still need to record, feed the mic cable out one corner of the bag; it's almost certain you'll be able to manipulate the buttons and work controls while the recorder is in the bag. Notepad. Keep notes on what you've recorded, as you really don't want to have several gigabytes of data staring you in the face without a clue as to what sounds are where. SEEK OUT THE OPTIMUM RECORDING SPACE Airports are great for found sounds: You get crowd noises, announcements of planes going to exotic destinations, restaurant sounds, cars in the parking garage (including door slams with killer ambience), and of course, the roar of airplane engines and the strangely annoying announcements for shuttle trains and such. But whatever you record, consider the best way to record it. For example, with one project I needed to get some airport announcements in isolation, especially the one in the Atlanta airport warning you to watch your luggage ("Maintain control . . .") which with a little cut and paste, would make the perfect "Big Brother" counterpoint to a hip-hop tune. What to do? Well, they have speakers in the bathrooms, and as it was 2 AM and I was delayed coming back from Europe due to a hurricane, I just camped out in a bathroom, found where the speaker was, stood on a toilet seat to get as close to it as possible, and waited for a stretch of time when no one came in and flushed. (However, I do wonder what one guy thought when he came in and saw me pointing a mic in the general direction of the ceiling, while the announcer said "Report any suspicious behavior...".) When you're looking for found sounds, there's always a sweet spot to record them. For best results, wear earphones that enclose your ear to block out noise, and listen as you test different miking positions. I used to use big honkin' headphones, but the latest generation of in-ear earbuds do a pretty good job of sealing out ambient sounds. They also allow for surreptitious recording, as people will think you're just listening to an iPod or other portable player. A RECORDER'S BEST FRIEND: DIGITAL AUDIO EDITORS It's not just enough to record the sounds - you need to massage them with a good digital audio editor. It's hard to recommend a specific one, because they all have their own cool little DSP processes; I especially like processing a sample by using a convolution reverb that has an impulse loaded other than reverb (see Fig. 2); this can make just about anything sound cool. Fig. 2: Convolving sounds with unusual impulses, such as guitar bodies or even thunder (as shown here with BIAS Peak), can turn garden-variety found sounds into truly otherworldly effects. Excessive pitch transposition is a sure-fire route to uniqueness (just remember to remove DC offset if you're shifting way down), as is vocoding, delay, resonant filtering, ring modulation, and those other cool tools in the sound designer's arsenal. I also recommend noise reduction; if there are unwanted background sounds, taking a "noiseprint" of the offending signal and removing it can do wonders for the overall quality. Also, don't forget gating and enveloping. For example, "radical recording" enthusiast Dr. Walker is a fan of taking vinyl hiss, amplifying it way up, gating it, and using it to replace the hi-hat sample in drum kits. Gated shortwave radio noises with really fast decays can sound like weird percussion, and just about anything mixed in with a snare drum can sound interesting. AND FINALLY... Treat the sounds you've found as a real sample library: Back them up, document them, and keep both the original and processed versions. I've even used some samples made way back in the 80s in music I'm working on now...you never know when a found sound will turn out to be a found treasure.
  17. Turn Your Editing Machete into a Frequency-Selective Scalpel By Craig Anderton Adobe Audition, and its predecessor Cool Edit Pro, have always had stellar noise reduction tools. But Frequency Space Editing (FSE), an editing option that became available starting with Audition V1.5, is impressive not just for its ability to eliminate noise problems with pinpoint precision, but to allow selective editing on very specific parts of a sound. Do you have a drum loop that you really like except for a wimpy kick drum? Isolate just the kick, and run it through a bit of distortion to beef it up...or eliminate a single triangle hit in the middle of a song. And about the guy who coughed in the middle of your sensitive acoustic guitar moment on that live recording: With a little luck, you can nuke the cough and leave the guitar intact. Although other programs have been able to isolate a band of frequencies over a specific time range and manipulate them, Audition's tools are the most cost-effective implementation yet. But be aware that while the results can seem miraculous, they can't solve every problem. For example, with one drum loop I wanted to get rid of an annoying clave hit, but it it had been put through a ton of reverb. Although it was possible to eliminate the main clave sound, removing the reverb meant taking out a lot of frequencies that needed to be kept. On the other hand, I played on a record many years ago where the drummer hit the hi-hat late coming out of a solo, and it always bugged me. With FSE, I was finally able to cut just the hi-hat from the stereo mix, and place it where it belonged. Now that's pretty amazing. HERE'S LOOKING AT YOU Using FSE requires a different look at audio. Instead of the usual waveform display that shows amplitude over time, FSE uses a spectral display that shows distribution of energy in specific frequencies over time. You can access Audition's Spectral view from the View menu. My one complaint about this view is that the vertical axis (frequency) uses a linear scale rather than a logarithmic one, so all the lower frequencies (where most of the interesting musical sounds lie) are squeezed into the display's lowest part. The workaround is to right-click on the vertical calibration, then select Zoom In so you can focus in on a specific frequency range. For example, if you want to remove a low-frequency sound, zoom in to frequencies below 400-500Hz. However, remember that you'll also need to zoom out to see a wider range of frequencies if you want to remove a sound's transient components, which are generally higher in frequency. KILLING A KICK Here's a real-world example of how to remove the kick drum from a funk-type drum loop (I wanted to substitute a tougher, more electronic sound). The following diagram shows the steps you would take to do this; following are descriptions for each step. Here are the main places on the screen to do Frequency Space Editing. Go View > Spectral View. The Spectral view replaces the default Waveform view. Increased energy in a specific range is brighter, while decreased energy is darker. Select the Marquee tool (the square with the dotted border). You'll probably need to use the Zoom tools, both on the vertical and horizontal axis, to make it easier to identify the area you want to process. Be patient; it takes practice to learn to recognize the various "sonic signatures" of different sounds. There are a few zoom shortcuts for the vertical axis. One is to right-click on the vertical calibrations, then choose Zoom In or Zoom Out. You can also do this with keyboard equivalents (Alt = and Alt - respectively). Or, right-click on the calibrations, and drag over the area to which you want to zoom. Here, the marquee (bordered in white for clarity) has selected the kick drum. Note that when you select in one channel, the other channel defaults to selecting the same area as well. The clue that this is the kick is the high-amplitude burst of energy in the bass range (the higher-frequency bursts to the left and right are the snare). While you're learning to recognize which frequencies are essential to a sound, you can always isolate, cut, then audition (and if it doesn't work, undo) to make sure you've found the right area to edit. Now that you've isolated the area you want to cut, go Edit > Cut. This removes the area defined by the marquee. For the smoothest possible removal, go Favorites > Repair Transient. Providing the area you selected is relatively small, this will "morph" audio over the cut, which is the audio equivalent of putting a flesh-colored band-aid over a cut so you don't see it. SO YOU THINK YOU'RE DONE... Well, not quite; in this case, there's a transient at the beginning of the kick that also requires removal. Let's describe the process for removing the entire kick, using four "frames" of a sort of "how-to comic strip." A marquee (shown surrounded with a white border for clarity) selects the main kick drum hit. The object here is to define the kick drum's "body" so it can be removed. Deleting the kick produces the spectral view in this frame. However, note that the kick's attack transient remains. A marquee is drawn around the kick's attack transient (again bordered in white for clarity). Deleting the transient produces the final result. Note how the kick drum is completely gone – it's as if it had never existed. Dealing with signals that include a lot of harmonics is correspondingly more complex. Fortunately, though, these harmonics tend to be fairly thin "slices" that you can remove without altering the rest of the signal. The top half of the above diagram shows a triangle hit's harmonics (each one is bordered in white). The bottom half shows what happens after they've been deleted (note the black spaces) and gone through the transient repair process. On playback, you hear no triangle at all. SO WHAT'S THE CATCH? This technique's "jaw drop factor" depends on what you're trying to process. With a dry, fairly simple drum loop, you can remove individual drums and never even know they were there. On the other hand, as alluded to earlier, if there's reverb it's almost impossible to remove a specific sound because the reverb extends the sound's duration and the amount of bandwidth it takes up. Also, if you're trying to delete a sound with lots of harmonics, remember that the more of the spectrum you remove, the more likely this will affect the sound of other instruments. In the example given above of removing a triangle hit, I had to be very careful to remove the minimum amount of signal possible. Otherwise, other sounds with high frequencies (high hats, cymbals) were affected. This is why it's important to zoom in and remove no more than is absolutely necessary. So much for cautions, here's something very cool you also need to remember: You can apply any editing operation to a frequency space, not just cut or delete. It one particularly tedious example, I had a song where only the open hi-hat was too loud all the way through the song. I isolated each open hi-hat hit, and reduced the level for each one by 4dB. Miraculously, the open hi-hat fell right into the mix, and because it was being attenuated rather than completely eliminated, there were no ill effects on the rest of the tune. Another one of my favorites is adding a bit of processing from PSP's Vintage Warmer to kick drums and toms...yum. Frequency Space Editing is pretty amazing. If you have Adobe Audition 1.5 or higher and haven't checked out this feature, you're missing out on a tool of exceptional potential. 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.
  18. Speed up conversion and editing time by automatically applying processes to multiple files By Craig Anderton There's nothing like batch processing when you need to process a large number of files, as I found out when having to prepare hundreds of samples for my "AdrenaLinn Guitars" sample CD. But you don't have to create sample CDs to need batch processing. Batch processing is a great way to convert libraries of files to MP3 format, or from 24 bit resolution to 16 bits. If you use sample CDs, batch processing can help compensate for problems like inconsistent levels, or differences in loudness maximization. Or consider what happens when you receive a bunch of tracks that all have the same response anomaly, and you need to fix this on all the tracks. Why sit there and do each file by hand when all you need to do is set one up as desired, then have your computer apply those characteristics to all the files? Basic Batching The concept behind batch processing is pretty simple. Although this article focuses on Wavelab, I've also used batch processing with Syntrillium's Cool Edit and Sonic Foundry's Sound Forge, and while the details are different, the overall process is similar: Select the files to batch process. Set up the processes you want to apply (normalization, EQ, etc.). Set up characteristics of the files to be processed (what directory they go in, whether you want to overwrite existing files or create new ones, etc.). Click on "Run," and let the compute do its thing. Wavelab's Batching Here's a typical example: you have several files open on screen, and want to normalize them. The procedure is... 1. Go Tools > Batch Process. Clicking on Add All will add all currently open files to the batch processor. 2. Add the files you want to process. Wavelab has a great shortcut: click on the "Add File Already Open In Wavelab" icon (the one to the right of the "Add all Files From Folder" icon; hold your mouse over an icon to see its name). This brings up a list of all currently open files. Select the ones you want, or click on "Add All." If you want to convert to MP3, a window shows up that lets you set the data compression parameters. 3. The screen now shows the list of files. Click on the Output tab and choose the destination folder, whether to add a prefix or suffix to the processed files, output format (here's where you would convert to MP3 or change bit resolution, etc. Click here to open up the Processor List window. 4. Set up the batch plug-ins by going to the Input tab, and clicking on the "Edit Batch Plugins" icon. This window is where you choose the processors you want to apply to the batch of files. 5. Open up the Plug-Ins folder, then drag the plug-ins you want to apply over to the left "Sequence" pane. You can modify the order of plug-ins by drag-and-drop, and check or uncheck plug-ins. 6. Note that there is a Normalizer plug-in in the Plug-Ins folder (not in DirectX, VST, etc.) which is not listed under VST or Wavelab plug-ins but is only accessible from this list of plug-ins. To normalize all the files you've chosen, drag this into the left pane, then click on "OK." 7. When you're back on the Input page, click on "Run," and let your computer do all the work. There is one caveat: if you want to undo the operation on the open windows, there is no batch undo; you have to select each file and undo individually. Extra Tips That's the basic idea behind batch processing, but let's drill down one more level with some tips. If you save to the folder from which the files originated, as with any other editing process, the new versions aren't saved permanently until you either save them from the file menu, or close the file and click on "yes" when you're asked if you want to save. Concerned about overwriting a critical file by mistake? Check the "Create Backups" box under Options (in the Output menu). Checking the box "Delete Files After Process" deletes the original files after processing. I strongly recommend you never check this box, because there is no undo if you do something like choose the wrong batch process. Remember, when you batch process, you're affecting a lot of files. If you screw up, you screw up big time. If you add a file name suffix or prefix, you don't have to worry about overwriting original files no matter what you do. The Presets option can be very handy if you do a task repetitively, like convert to MP3 from WAV, or convert WAV to AIFF. However, using presets isn't very intuitive, so here's how the process works: Type a name for the preset in the field above the Load, Add, Delete, and Update boxes. Click "Add" after entering the process name, which adds the name to the list of presets in the column on the left. If at some point you want to change the preset, adjust your parameters as desired, then click on the preset name from the left-hand column. This does not load the older version of the preset unless you click on Load; instead, click on Update, and the new preset version will overwrite the old one. In addition to dragging over individual plug-ins to determine how the batched files will be processed, you can also choose any master section preset and just drag that over. Thus, you have two ways to batch process using presets: choose one from the master section, or create your own preset within the batch process function. The "Extra" drop-down menu on the Input page has a number of useful functions, like allowing you to sort the batched files by size, bit resolution, number of channels, etc. Batch processing may not be the most glamorous signal processing option in the world, but when you need to process a lot of files in an identical way, the amount of time it takes to set up the batch process is negligible compared to how much time you would have to spend applying each process individually. Once you become familiar with the batch processing procedure, you might be surprised by how many times it comes in handy. As they say, "Life's a batch!" 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.
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