Phil O'Keefe

It's amazing how much a pick can influence your tone by Phil O'Keefe One of the least expensive, and yet most often accessed items in my studio is a small tin box that I keep stocked with a wide variety of different plectrums. Unclaimed picks show up around here all the time, despite my best efforts to find their rightful owners. When they do, I put them in the box. I use an empty cigarette tin, but a empty gum, cookie, biscuit or candy tin or box will work just as well. I've managed to collect quite an assortment of picks in different sizes, shapes, and thicknesses. The types of materials they are made from is equally diverse: nylon, metal, felt, stone, simulated tortoise shell and a dizzying variety of different types of plastic. That little tin box has come in handy not only for the inevitable times when someone shouts out that they "can't find their pick," but also for when I am seeking a different sound or texture on a recording. For example, a thin pick can accentuate the wispiness of a strummed part, and a thick pick can accentuate your note attacks. Different materials can also have noticeably different sonic attributes, and different pick sizes and shapes can change the way you hold the pick, which can also affect the sound. Guitarists and bass players are often creatures of habit, and once we find a pick we like, we tend to stick with it. But don't overlook the variety of new sounds you can get from something other than what you normally use. If you're not in a situation where you "find" picks on a regular basis, try trading picks with some of your friends, or go to your favorite local or online store and spend five or ten dollars on a variety of different picks. Make sure you "mix it up" and get a diverse collection. Then spend a little time trying them out, and take note of how each one sounds and feels. You are bound to find some new sounds; and may even find a new favorite pick in the process. Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Associate Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

Recently revamped, Fender's vintage reissue P-Bass is now even more historically authentic By Phil O'Keefe The Fender Precision Bass, first released back in 1951, was the first commercially successful solid body electric bass. It's the model that really started it all for modern bass players. By 1957, the model had morphed into the modern P-Bass as we know it today, with its famous hum-canceling "split" coil pickup and comfortable tummy cutout and forearm body contours. Further changes occurred over the next couple of years - most notably, the change from an all-maple neck and fingerboard to a maple neck capped with a rosewood "slab" fingerboard in 1959. By 1963, more changes had been made, and further changes were made in the years following the sale of Fender to CBS in 1965, but it's those early basses from the "pre-CBS" era that remain the holy grail for a lot of players. Unfortunately, vintage basses are getting harder to find, and more expensive every year. To address this, Fender started making vintage reissues of some of their most iconic models in 1982. Fender recently revamped their entire American made vintage reissue series, with an eye towards making them even more historically accurate. They've retooled, and where possible, returned to the vintage tooling that was used on the original instruments back in the 1950s and 1960s. This attention to detail has resulted in their most authentic reproductions of those legendary guitars and basses yet. What You Need To Know Fender sought out exemplary "golden" vintage examples of the original '63 P-Bass, and examined and measured them in great detail to serve as the benchmarks for their new American Vintage series models.The pickguard is mint colored, which looks great paired with the Olympic White nitrocellulose lacquer finish on the lightweight alder body. Other colors are also available, including Faded Sonic Blue, Seminole Red and three-color sunburst. The sunburst models come with a 4-ply tortoiseshell pickguard instead of the mint pickguards that are used with the other colors. The nitrocellulose lacquer-finished maple neck is evenly and subtly flamed, and it looks gorgeous. 1963 marked the introduction of the laminated rosewood fingerboard on Fender basses, which replaced the earlier slab style fingerboards. True to the vintage originals from 1963, the American Vintage '63 Precision Bass features the round, laminated rosewood board, which is actually more labor intensive than the slab style board to make. The fretboard radius is the vintage correct 7.25", while the 20 frets are also the smaller vintage-approved style and size.Fender used to offer four different optional neck widths for various models. These were called A, B, C, and D, and measured roughly 1.5" (A-width), 1.625" (B-width; probably the most common), 1.75" (C-width), and 1.875" (D-width) at the nut. These neck widths should not be confused with the neck profile - the way the back of the neck is curved or shaped - which is typically described as V-shaped, D-shaped, U-shaped, or C-shaped. The American Vintage '63 P-Bass has a C-shaped neck profile. The width of the American Vintage '63 Precision Bass most closely matches the traditional C-width.At 1.740" wide (measured at the bone nut), the new '63 P-Bass has a slightly wider neck, but it's not unbearably thick - even with my fairly short and stubby fingers, it felt fairly comfortable, especially by P-Bass standards. I'll admit I'm generally more of a Jazz Bass kind of guy, but I must say that the neck on this bass, while by no means small, is very "playable." It has a very comfortable feeling profile. One of the more glaring cosmetic flaws in the earlier reissues was the look of the material used for the neck's dot inlays. Typically, they were stark white, which is an instant visual cue that you're looking at a reissue and not an original. Thankfully, Fender has addressed this with the revamped reissues, making them look a lot closer to the vintage "clay dots." The pickup is a re-voiced classic hum-cancelling split single-coil unit. Volume and tone controls with chrome flat-top knobs and a pickguard mounted 1/4" output jack round out the electronics. The tone is classic vintage Fender P-Bass, with a deep, powerful and clear fundamental tone, and a throaty growl when pushed. In a nutshell, this bass sounds great! The vintage style four saddle bridge features threaded steel "barrel" saddles. The stock strings are roundwounds. The strings are "top mounted" and anchor at the back of the bridge, as opposed to going through the body.Nickel / Chrome bridge and pickup covers come pre-mounted on the Vintage '63 Precision Bass. Back in the day, we used to call them "ashtrays" since they were typically removed from the bass and somewhat resembled them when they were stored in the case, and could serve as one in a pinch. While a lot of players will probably still decide to remove them, it's cool that Fender is including them with the American Vintage models. The accessory pack on this bass is as good as anything I've seen in quite a while. Not only do you get the usual assortment of hang tags, you also get vintage style recreations of the original manuals, a vintage style strap, and lots of other cool "case candy." Fender even thoughtfully includes a 1/4" cable. The vintage-style orange plush lined G&G case is quite nice. It fits the bass very well, and the cream Tolex exterior with contrasting black ends looks really sharp. A small Fender logo near the handle is the only visual clue to what lies inside.The tuning machines on this bass are the vintage correct "reverse" open gear design. According to Fender, their "flash coat" finishing uses a 100\% nitrocellulose lacquer "flash" topcoat over the 100\% nitro sealer and color coats, and this final flash topcoat gives the guitar "a more authentically vintage appearance." The nitrocellulose lacquer finish on the American Vintage '63 Precision Bass is admirably thin, and does indeed have a great vintage-approved visual vibe to it. Limitations It's a P-Bass, and that means the neck is probably not the first choice for players with smaller hands. While it's still comfortably playable, a thinner Jazz Bass neck may be more appropriate for those who have smaller hands. The reverse of that is also true. If you have average to larger-sized hands, you're probably really going to love the neck on this bass.Nitrocellulose lacquer finishes require a bit more care than urethane or polyester finishes. You'll need to use caution when taking the Fender American Vintage '63 Precision Bass out of the case whenever it has been stored in a cold car trunk or similar location, or you'll risk causing small cracks in the finish called "checking." A large red tag on the outside of the case cautions you about the dangers of exposing the bass to temperature extremes. The American Vintage '63 Precision Bass has a thin 100\% nitrocellulose lacquer sealer coat. This provides a less "smooth" base for the color and clear top coats than some other sealers that have been used in the past, and very fine imperfections due to the texture of the underlying wood are more likely to be seen in the final finish. The finish is very nice on this bass, but don't expect the same level of smoothness that you sometimes see with thicker polyester and urethane finishes.Because of the way it completely covers the anchor point for the top-mounted strings, the bridge cover will need to be removed to change them. It's not a huge deal, and many players will opt to leave the cover off all the time anyway, thus making the point moot, but if you like the looks with it left on (and it does look cool), just know that string changes will take a little longer to accomplish. Conclusions Fender has really stepped things up a couple notches with their latest American Vintage series. The Fender American Vintage '63 Precision Bass is far more "vintage correct" than previous models, and it plays and sounds fantastic. All the vintage inspired goodies (such as the case candy and the retro case itself) further add to the illusion of having a modern day version of a vintage classic. Fender says "they did the research, and it shows." They're not lying. Fender has really nailed that elusive vintage vibe with this bass - the sound, the look, and the feel - and you can get it for a fraction of the price of what the vintage originals go for. If you're in the market for a classic P-Bass that looks and sounds like the ones that were made 50 years ago, you really need to check it out. It's a fantastic throwback to one of Fender's golden eras. Resources Musician's Friend Fender American Vintage '63 Precision Bass online catalog page (MSRP $2,499.99, $1,999.99 "street") Fender's American Vintage '63 Precision Bass web page. Harmony Central Review Preview video Additional Photos Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Senior Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

How to compare, and choose the right recording chain components for the job at hand By Phil O'Keefe The relatively low cost of modern recording tools means that even modest home studios often have, or can afford a variety of different equipment. It's not unusual to see recording rigs with multiple types of mic preamps on hand. These might include preamps built into the computer audio interface, plus a few more built into a small mixing console, and maybe even a few outboard standalone preamps. It's also very common to see multiple microphones in a home recordist's arsenal. This variety of gear potentially gives you multiple sonic options to choose from, especially when you start to combine the two - microphones and preamps - in various different ways. Unfortunately, you'll often run into the situation where you don't have enough of each to do direct side by side comparisons of the different possible combinations - at least not quickly, and without having to re-patch things. Picking the right mic and preamp combination for the task at hand can be a bit complicated. Whenever it requires unplugging one thing and plugging in something else, or anything that takes time to configure, you've lost the advantage of hearing a direct, instant comparison between the two. Being able to make an instant comparison, in the context of the music, is an extremely valuable tool that makes it much easier to hear and directly compare the sonic differences; allowing you to make more informed decisions regarding what sounds and works best for that specific situation, whether it be which mic works best for a particular singer, or which mic and preamp combination gives you the best sound for the song when tracking a guitar amp or snare drum. GETTING AROUND THE PROBLEM Getting around the problem often requires a bit of ingenuity, and some careful patching, and occasionally some willingness to compromise a bit. For example, depending on the specific gear you have available, you may have to get creative with the routing in order to come up with a way to quickly compare multiple combinations. If you have a a multi-channel mic preamp such as the API 3124+, or a audio interface with multiple onboard mic preamps, or a mixing console with multiple mic input channels and you don't mind using those preamps, setting up comparisons between multiple microphones is relatively easy. Just plug each mic into one of the multiple identical mic preamps, set the mics up in the desired location relative to the sound source (and with their diaphragms as close as possible to each other), and run the direct outputs from the multi-channel preamp or mixing console into the line inputs on your audio interface and assign each one to a separate track in your DAW software. Don't forget to match up the levels (some mics are bound to be more sensitive than others) so you can avoid the "louder is better" bias influencing your decisions. If your DAW allows you to set up mute groups, you can set the mutes up so that when one channel is unmuted, the others automatically mute, making fast, direct comparisons easy. (Fig. 1) Alternatively, you can use a hardware DAW control surface to quickly mute and unmute the individual DAW channels as needed. This method can also be used with a hardware mixing console when using it for comparisons - just use its built-in channel on/mute buttons. Figure 3: The Radial Cherry Picker allows you to connect one mic to four mic preamps for instant comparisons Conceptually, the Cherry Picker does the same thing as the Gold Digger, only in reverse - allowing you to run a single input (such as a mic or line input) to any of four outputs, and switch between the four at will. There is switchable 48V phantom available on the input if you need it. Both units use relays for the switching, have totally passive signal paths, and don't add any noise or degrade the signal in any way. The Radial Engineering Gold Digger and the Radial Engineering Cherry Picker "street" for about $350 each. Combining the two units (Cherry Picker and Gold Digger) can give you even more flexibility in terms of auditioning what are arguably the two most important equipment elements of the recording chain / signal path - the mic and the preamp. Some microphones sound better on some some sound sources when used in partnership with specific preamps, and the best way to decide what works best is to hear it in context. Again, being able to make fast comparisons between different combinations of microphones and preamps is the ideal that you want to shoot for. By connecting multiple microphones and preamps together with the Gold Digger and Cherry Picker, you can easily and quickly compare various combinations of mics and preamps in context. By patching the output of the Gold Digger into the input of the Cherry Picker, you can use the Gold Digger to select between up to four microphones (it also works great for auditioning multiple line input sources and different direct boxes too) while using the Cherry Picker to select from up to four different mic preamps. Just click a button on the Gold Digger to select the mic you want, and then another button on the Cherry Picker to pick the mic preamp you want to use with it - you can even hit the two switches simultaneously, and swap between different mic and mic pre combinations very quickly, allowing you to hear which pairings work best for a specific instrument or vocalist in the context of the song. Once you've experienced the ability to quickly combine microphones and preamps in various combinations and instantly hear the results side by side, you won't want to go back to doing it the old way. The Radial Gold Digger and Cherry Picker are great tools that allow you to make better decisions in terms of how you use the mic and mic preamp tools you already have; giving you the ability to make informed decisions based on comparative listening - and that can really help you make better recordings. Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Senior Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

Amp sounding weak? Is it making funny noises? Maybe it's time for new tubes By Phil O'Keefe It happens gradually over time as the amp is used, but the sound of even the best tube eventually deteriorates. The next time you're struggling with your tone and wondering why it's just not quite "right" anymore, or why it doesn't seem as cool as you remember from days gone by, ask yourself this - when was the last time I changed my tubes? THREE TYPES OF TUBES There are three main types of tubes. Not all "tube" amps use all three types; often, solid state components are substituted for one or two of them, depending on the amplifier design. True tube amps generally have both tube preamp and tube power amp sections, and may have a tube power rectifier, or a solid state rectifier. Hybrid amplifiers utilize a tube or tubes in either the preamp or power amp section, with the other section of the amp being solid state. The vast majority of hybrid amps use a solid state rectifier instead of a tube. Let's take a closer look at each tube type, and some of the symptoms to watch out for. Preamp Tubes These are found in the vast majority of tube amps. Even the majority of "hybrid" amps use a preamp tube, although occasionally you'll find a hybrid amp that flips that paradigm and uses a solid state preamp and tubes in the output or power amp section of the amplifier. Examples include some of the old Music Man amplifiers from the 1970s and early 1980s, as well as the Fender Super Champ X2. Preamp tubes are often used for other functions within the amp, such as for the tremolo (mislabeled as "vibrato" on many Fender amps), and sometimes for the reverb driver and recovery circuits too. If the reverb or tremolo on your Fender amp is dead, the repair might be as simple as a quick tube replacement. Occasionally a preamp tube will be used as a phase splitter (often called a phase inverter), which splits the signal into two, with one positive and one negative polarity to feed the two halves of a push/pull (Class AB) tube power amp. Some common preamp tubes include the 12AX7 / ECC83, 7025, EF86 / 6267, 12AY7 / 6072A, 12AU7, 12AT7 and 5751. A few examples are shown below. (Fig. 1) Preamp tubes are occasionally covered with metal shields. They help protect the tubes and provide electronic shielding that helps reduce noise, so if your amp is so equipped, make sure you replace the shields after you test or replace the tubes. Figure 1: A few examples of preamp tubes, including the 12AX7, 6072 / 12AY7, and EF86 Power Tubes These tubes are generally larger than preamp tubes; taller and (with the exception of the EL84) fatter in diameter than their preamp tube cousins. They can be found in the power amp section of the amplifier, and along with the output transformer, they provide the final amplification "oomph" that drives your speakers. Power amp tubes wear gradually. Unlike preamp tubes, if you've been using the same set of power amp tubes for quite a while, you could very easily notice a dramatic improvement in tone by replacing them - especially if you've been driving the amp hard on a regular basis. Unlike preamp tubes, which can usually be replaced without having to consult a tech, some amps need to be biased after replacing the power amp tubes for best results. When in doubt, check your amp's manual, or ask on the Harmony Central forums. Common power tubes include the 6L6, 6V6, 6550, EL34, and EL84. (Fig. 2) Unlike preamp tubes, they are almost never covered with metal shields, but they may have retainers to help hold them in place that you may need to remove before taking the tube out of its socket. Figure 2: Common power amp (or output) tubes include the EL84, 6V6, 6550, and EL34 Rectifier Tube This tube converts the incoming AC mains power from the wall outlet into the DC current that the amp needs to run. Without a rectifier, the amp won't even turn on. No sound, no pilot light - nada. If your tube amp fails to power up, check the fuse and the rectifier tube, if it has one. In most cases, one or the other is blown and needs replacement. The rectifier is a large tube, similar in size to many power amp tubes. When looking at the back of the amp, if it has a rectifier tube, it is generally on the far left hand side of the amp, right next to the power amp tubes. A tube rectifier is more commonly found on lower-wattage amps (under 50W), while high-power amps tend to use a solid state rectifier instead of a tube. A tube rectifier doesn't make as big a difference to the sound of the amp as much as it does to the feel of it. With a tube rectifier, note attacks can "sag" a bit, and they don't punch out as immediately or as forcefully as you'll normally get with a solid state rectifier - the attack transients of notes are a bit more compressed, especially when you really dig in and hit a note forcefully, and when the amp is being run loud and hard. As long as the amp powers up, the rectifier tube is doing its job, and normally there is no real benefit to be gained from replacing it prior to it failing. Some common rectifier tubes include the 5U4 / GZ32, 5Y3 / GZ30, 6CA4 / EZ-81, and 5AR4 / GZ34. (Fig. 3) Since rectifier tubes generally run until they fail, it's normally not something you need to worry about, but if you gig or tour frequently, it wouldn't hurt to keep a spare on hand - just in case. Figure 3: Common rectifier tubes include the 5AR4, 5Y3, and 6CA4 DIAGNOSING TUBE PROBLEMS Rectifiers are easy. When they fail, it's usually pretty obvious, but what about preamp tubes and power amp tubes? Both can show signs of wear or have problems that can affect your tone even before they fail completely. On preamp tubes, you can generally use them until they start to cause noise (such as crackling, hissing, or hum) or other audible issues, or barring that, until they fail completely. Swapping out a suspect tube for a known good tube of the same type is a time-honored way of troubleshooting preamp tube issues. Just make sure you power off the amp and unplug it before changing any tubes, and remember that tubes get HOT! Always wait until the amp has cooled, or use a oven mitt to grasp the tubes with. Another way to test preamp tubes is to power up the amp, turn it up to a moderate level, and then gently tap on the preamp tubes, one at a time, with the eraser end of a pencil. If the tube is microphonic - if it makes a hollow sound that you can hear through the amp's speaker, or if it pings, or makes any kind of objectionable noise, it's probably time to replace it. Individual preamp tubes can be replaced one by one - there's usually no need to replace them all at once. Like tires on your car or strings on your guitar, power amp tubes start wearing the moment you put them in your amp and fire it up. The more you use the amp, and the harder you push it, the less time you can expect them to last. How long will they last? It's impossible to say. I've had old vintage amps that still sounded fine, even though they had the factory-installed original tubes in them. I've had other amps that needed to have the power amp tubes replaced after a year of heavy use. If your amp seems listless, lacks punch or feels flabby or weak in the bass, then it's probably time for a fresh set of power amp tubes. Since they wear out gradually over time, you might not notice that their sound has deteriorated, but when you replace a worn set of power amp tubes with a fresh set, the dramatic difference in tone can often be easily heard. THE BEST INSURANCE If you play a lot, you might want to consider buying a full replacement set for your amp. Test them when you first get them to make sure they're all working properly, then store them somewhere safe. A padded tube carrier / case is a good investment, especially if you tour. If you do gig, make sure you bring those spare tubes along with you. And since it's not at all uncommon for the fuse to blow when a tube fails, make sure you also carry a few spares of the ones used in your amp - you never know when they might save the day! Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Senior Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazine

A control surface gives you hands-on control and makes mixing easier. Here's how to set one up for use in Pro Tools By Phil O'Keefe Using a mouse to mix with can be a real time drain, and not a lot of fun. While the mouse and keyboard are essential mixing tools, using the mouse to manipulate control values is not very intuitive, nor is it very fast. It's great for precise, detail oriented tasks, but music is as much about feel and emotion as it is about precision, and those who like to "play" the mixing console like an instrument will find that using a control surface of some kind adds considerably to their mixing enjoyment, and may even result in more real-time input from the person mixing the project, which can translate to better sounding and more human feeling mixes. Additionally, while a mouse can be used to adjust only one parameter at a time, a good control surface will allow you to adjust multiple things at once, which can further speed up the mixing process. HARDWARE OPTIONS There are a lot of different choices available in terms of control surface hardware. Control surfaces range from fairly simple devices that allow you to control transport functions and fader, pan and other functions for one channel at a time, such as the PreSonus Faderport (Fig. 1), to multi-channel, expandable control surfaces with touch screens and other high-tech features, such as Avid's Euphonix MC Control. (Fig. 2) Figure 1: Even affordable controllers like this PreSonus Faderport give you hands-on control of your mix in ways a mouse can't match Figure 2: Larger controllers, such as this Euphonix MC Control allow you to control multiple channels simultaneously Speaking of touch screens, for iPad and Android tablet owners, there are apps such as Saitara Software's Ac-7 Core HD for iPad (Fig. 3) and Humantic's TouchDAW for Android that will turn your device into a multitouch control surface. These usually require a wifi network that both the Pro Tools host computer and the tablet must be able to access in order for them to communicate with each other, but in most other respects function similarly to the other controllers mentioned in this article, except with virtual controls on the touchscreen instead of physical ones. Such controllers also have the distinct advantage of being wireless and mobile, which can come in very handy when you're trying to record in one room with the computer and the rest of your gear in a second room. Figure 3: With the right app and a wifi network, your tablet can work as a wireless multitouch control surface for your DAW HOOKING UP THE HARDWARE Connecting the hardware is generally fairly straightforward. Most controllers use either USB, Firewire, or Ethernet to connect the hardware to your computer, so a lot of times it's as simple as connecting the power supply to a wall outlet (some units are bus-powered) and a single interface cable between your DAW computer and the control surface. You should always check the manual for both the DAW software and the hardware unit to make sure you follow all installation instructions for your particular products since occasionally you'll need to connect the hardware before installing the software, or vice versa. SETTING UP THE SOFTWARE Regardless of which control surface you decide on, you'll need to configure Pro Tools so it can "see" the control surface and respond to its commands. This is also relatively straightforward for hardware control surfaces, and only slightly more complex for iPad apps. Let's start with what needs to be done for both controller types, and then we'll get into the extra steps you need to take to get your tablet-based controller up and running. First, launch Pro Tools. In the Setup menu, select "Peripherals." In the box that opens up (Fig. 4) you'll need to select the Ethernet Controllers tab if you're setting up an ethernet controller such as the Euphonix MC Control. For a controller that connects with a USB cable, Firewire cable, or over wifi, select the "MIDI Controllers" tab instead. Figure 4: The Peripherals dialog box allows you to configure Pro Tools to work with your control surface You'll notice that there are multiple slots available. These let you assign multiple control surfaces, which can then be used simultaneously to provide more channels. For example, I often use the control surface features built in to my Yamaha 01V96 digital mixer along with my iPad as a wireless control surface, giving me a total of 24 faders between the two. Under "Type", select the appropriate control surface. If you have a Command 8, M-Audio Keyboard, or a Motor Mix, you would want to select that from the list. (Fig. 5) Many control surfaces use the Mackie HUI protocol, and if that's the case with your device, you'll want to select HUI as the controller type. This is also what you'll need to select for most tablet based control surface apps. Under "Receive From" and "Send To", you should see your controller listed. It's important that you select the same device for both. If you're using a hardware controller, that should be all you need to do. Click on OK, and the software and hardware should automatically lock up. Figure 5: Many control surfaces use the HUI protocol to communicate with Pro Tools Tablet users will need to select "Network, Session 1" from the drop down list for both "Receive From" and "Send To". This tells Pro Tools to look to the wifi network connection for the controller data… but before you do that, there are a couple of extra steps tablet users need to take first. Start by making sure the tablet and the DAW computer are both connected to the same wifi network. Launch the control surface app on your tablet and configure it per the app manual's instructions. Mac users should then launch the Mac's Audio MIDI Setup utility (it's in the Applications / Utilities folder). Click on "Window" in the menu bar, and in the drop down menu, select "Show MIDI Devices". In the MIDI Studio window that opens up, double click on the Network icon. This opens up the MIDI Network Setup window. Create a new session by clicking the "+" button under My Sessions and then click on the checkbox to select it. The default name is Session 1. In the Directory list in the MIDI Network Setup box, you should see your tablet listed. Click on it to select it, then click on "Connect". Once it is connected, it will be listed in the "Participants" area. (Fig. 6) Figure 6: Audio MIDI setup's MIDI Network Setup window allows you to configure your tablet to control your DAW wirelessly Windows XP through Windows 8 PC users will need to download and run Tobias Erichsen's rtpMIDI app, which provides the same MIDI networking over wifi functions for their systems. The setup procedure is otherwise nearly identical to the one for Macs. While you're there, don't forget to also check out the well-written rtpMIDI tutorial too. BENEFITS In addition to turning your tablet into a wireless remote control for your DAW, there are lots of other benefits to using a control surface. One of the biggest is that they allow you to control more than one thing at a time. For example, crossfades where one thing is fading in and another part is fading out, or an instrument that simultaneously fades in while panning across the stereo sound field. As I alluded to in the opening paragraph of this article, control surfaces are perfect for "playing" the mix like an instrument, in a manner similar to how old school analog consoles were used - working the controller's mute buttons, faders, and knobs to perform automation "mix moves" and accentuate things as you mix in a natural way as the music is playing, as opposed to endlessly clicking on a vector display to adjust things with your mouse. Think it doesn't make a difference? Try this: take two passes at a mix, using just the mouse for one, and augmenting it with a control surface on the second mix. See which one takes longer to finish, and which one "feels" better to you once they're done. Chances are you'll prefer the mix that used the control surface, and it will probably be the one you have the most fun doing too. Once you try mixing with a control surface, you'll never want to go back to mixing without one. Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Senior Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

There's nothing worse than having your instrument fall off its strap. Here's how to keep that from happening By Phil O'Keefe If your instruments still have the stock strap buttons installed, and you use normal guitar straps and no locks, then sooner or later, you're going to have the uncomfortable and possibly expensive experience of having your strap working its way off the strap buttons, and your treasured and expensive instrument crashing to the floor as you watch helplessly. This can happen at any time, and it's almost always when you're least expecting it, which means that you're not always going to catch it in time. It's a sickening feeling as you pick the instrument up off the floor. Did it get a big ding in it, or even worse, did something get broken? I've seen headstocks break simply because a guitar fell a few feet from the strap to the floor and landed the wrong way, and that's an expensive repair… and one that can easily be avoided. LOTS OF AFFORDABLE OPTIONS There are all sorts of clever ways that people have come up with to keep straps firmly attached to the instrument, and most are under $20. Let's take a look at some of them. Want a really cheap option? If you drink Grolsch beer, then the rubber washers that come with the bottles can be used as a strap retainer washer. You'll need to remove the strap button and attachment screw from the instrument completely, and attach first the rubber washer from the Grolsch bottle, then the strap. Each needs to be put on from the "inside" of the button (the part that's closest to the body of the guitar), then just screw the button back on to the guitar with the washer and strap in place. You won't be able to remove the strap from the guitar without reversing this procedure, so it's best for guitars where you don't mind having the strap semi-permanently attached, but it does have the advantage of being practically "free" - again, if you happen to know of someone who drinks that particular brand of bottled beer. The washers can sometimes be found in bulk separately (without the beer) and inexpensively on Ebay, but if you're going to go that route, you might want to consider Fender's Strap Blocks. These are similar rubber washers, and a set of four (enough for two instruments) can be had for well under $5. Other systems are designed to use replacement strap buttons and a locking device that you attach to a standard guitar strap. These systems lock the strap to the guitar, but can be easily removed in seconds for storage. I've been using Schaller Strap Locks (Fig. 1) for decades on several of my guitars and straps, and have never had any issues with them. They make it easy to disconnect the strap from the guitar, but hold it securely unless you intentionally disconnect it. They're available in gold, black, and nickel to match the hardware of your guitar, and starting at under $20 for a set, they're reasonably priced. Figure 1: Schaller's popular Strap Locks work great, are inexpensive, and allow you to remove the strap quickly I've also had a few guitars with the Jim Dunlop Dual-Design Straplok System (Fig 2) installed, and have had good results with them. They use a push button to unlock the strap from the strap button so you can disconnect it, and while they're not directly compatible with the Schaller system, they're conceptually similar and work equally well in my experience. They are also available in a variety of finishes. Figure 2: Dunlop's Straploks are equally effective and affordable Ernie Ball has a product called Super Locks. Again, these are available in black, nickel or gold. As with the Schaller and Dunlop locks, they use a proprietary design with a replacement strap button and locking mechanism that attaches to any standard guitar strap. A somewhat different approach is taken with replacement strap buttons that are designed to more firmly anchor the strap. Some vintage Ibanez guitars came with V shaped strap buttons that hold a strap more securely than the typical round strap buttons do. For those who want something similar, there are the Planet Waves Elliptical End Pin strap buttons. They are wider than standard buttons, which requires the strap to be placed at a sideways angle for attachment and removal - a position the strap isn't likely to be in unless you're intentionally trying to take the strap off. Jim Dunlop makes a couple of plastic strap locks that can be used with standard strap buttons and straps. The Dunlop 7007 and 7036 Lok Strap Systems (Fig 3) both fit around the strap button and over the strap, and a simple turn locks them in place, preventing the strap from coming off of the button. They're similar in concept to the Grolsch washers and Fender Strap Blocks, but with the advantage of being easily removable without having to unscrew the strap button from the guitar. Figure 3: Requiring no modification of the instrument, Dunlop's Lok Strap systems are inexpensive and removable Another approach is to build the lock into the strap itself. One such strap with a built-in locking mechanism is the Planet Waves Planet Lock guitar strap. (Fig. 4) As with the plastic Dunlop locks, they require absolutely zero modification to most guitars - if it uses standard strap buttons, this strap should work with it just fine. Figure 4: Putting the lock into the strap as Planet Waves did is another approach that can work well with stock strap buttons CHEAP INSURANCE The important thing isn't which of the methods you decide to use, but that you do something to insure that your instrument remains firmly attached to its strap unless you intentionally disconnect it. A few dollars and a little time spent now can protect your valuable instrument from damage, and help you avoid expensive and time-consuming repairs. It's cheap insurance that every guitarist and bassist should invest in! Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Associate Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

DIY crossovers and multiband processing in your DAW By Phil O'Keefe Crossovers are not normally something people think of when it comes to DAW software. Many musicians will be familiar with them as part of a home or car stereo system, bass amp and speaker rig, PA / DJ system, or keyboard setup. We'll get to how you can use them to do some useful tasks in your DAW in a moment, but for those who need a refresher… CROSSOVER BASICS A crossover is a passive or active device that uses equalization filters to divide the frequency spectrum into multiple bands, and then allows you to patch and route each band individually. A two band crossover "splits" the signal into separate low and high frequency ranges. The point where the signal "splits" is called the crossover frequency. How quickly it transitions or "rolls off" frequencies beyond the crossover point is called the filter slope, and is usually measured in decibels per octave. A 12 dB per octave filter will reduce the level by 12 decibels for every octave past the crossover cutoff frequency, while a 24 dB per octave filter will be twice as steep, and signals one octave past the cutoff frequency will be attenuated 24 decibels. Crossovers are commonly used in things like bass amplification and PA systems; allowing different parts of the frequency spectrum to be routed to separate power amplifiers, along with speakers that are optimized for each band. Multiband (also called bi-amped, tri-amped, two-way, three-way, etc.) setups generally offer better fidelity than a full-range speaker system, since speakers can be optimized to work in narrower frequency ranges instead of trying to reproduce the entire audio range. The amount of amplification can also be optimized for each frequency band and the connected speakers. CROSSOVERS IN YOUR DAW However, in recording, there are also times when you might want to divide the frequency spectrum of a sound into different components. I recorded some basic audio clips and took lots of screen shots to help demonstrate the sound of multiband processing and show you how to set it up. A basic mono bass part (Fig. 1 and audio Clip 1 - clips are at the bottom of the article) is our test subject, although the principles could be just as easily applied to other types of sounds. It was tracked in Pro Tools over a simple drum beat generated with FXPansion's BFD Eco. From there, it was manipulated in a few different ways, which I'll describe as we go along. Figure 1: A basic mixer layout with a mono bass track Let's say you'd like to use the mono bass recording from example 1 and add some distortion to it, but when you insert a distortion as an inline plugin (Fig. 2), the bottom of the mix gets muddy, or the fuzz plugin causes the lows to practically disappear completely as can be heard on Clip 2. Wouldn't it be nice to be able to add the "dirt" just to the upper frequencies of the bass signal, without having to process and mess up the bottom end? Figure 3: A phaser-vibrato plugin running inline as a channel insert will modulate all frequencies Sure, there are some cool multiband distortion and compression plugins out there, but unfortunately, there are not many "crossover" plugins that allow you set the crossover frequencies and the type of filters and their cutoff slopes, and also set the number of bands and assign each to a separate mixer output channel so you can process them individually. However, there are a couple of easy ways you can set up similar processing with the tools your already have built into your DAW. What if we could split a sound -- any sound in your DAW -- into multiple channels, each covering a different frequency range? Then we could process each "slice" or range of frequencies independently before mixing everything back together again. That would be pretty cool, huh? By splitting a signal into multiple bands, you can do all sorts of things to it from there, such as widen the apparent stereo width or subtly chorus it in one frequency range while distorting it or using a sub bass synth plugin in another. It also allows you to EQ and compress each frequency range independently. METHOD ONE - CLONE AND FILTER One way to do it, assuming you have a fast enough hard drive to handle the extra streaming, is to just clone the track you want to "split up" and process the original and the clone tracks differently with various EQ filters. The advantage here is that it's relatively easy to set up; just select a track and use your DAW's clone function to create copies of it. The downside is that this approach at least doubles the amount of audio data that the "track" needs to stream from the hard drive. If you want to use a 3-band crossover, it will triple it. Figure 4 (and Clip 4) show our original bass track along with a cloned version of it, and Figure 5 and Clip 5 add in a distortion plugin on the high frequency channel. The original bass track has a low pass filter inserted with a 200Hz cutoff frequency; this becomes the "low frequency" channel. The clone of our bass track has a high-pass EQ filter (bottom EQ plugin in the image), with the cutoff also set to 200Hz. This essentially filters out all of the low frequencies, and turns this channel into the "high frequency" channel for processing and mix purposes. You can hear the sound of this "crossover" in Clip 4. Compare it to to original mono bass track (Clip 1) and see if you can notice any differences in the bass sound near the 200 Hz crossover frequency. Figure 4: Low Pass EQ filtering on the original track, and high pass filtering on a clone of that track basically divides the track into two in a manner similar to a crossover Figure 5: Inserting a distortion on the channel with the high-pass filter allows you to affect only the higher frequencies of the bass, while leaving the bottom alone METHOD TWO - DIVIDE & CONQUER An arguably better way to go, assuming you have a few aux sends and returns to spare (and most modern DAWs have plenty) is to set up two (or more) pre-fader aux sends to route the signal off of the original source track and bring it back into the mix on multiple but separate aux return channels. If you just want to separate the signal into two bands; high and low frequencies, two aux sends and two returns is all you need (Fig. 6 / Clip 6). If you want a three band "crossover", you would need three aux sends and returns, with a high-pass and low-pass filter for the high and low frequency bands, and a band-pass filter for the midrange channel. If you don't have a "bandpass" setting on your favorite EQ, you can use a lowpass and a highpass filter together to achieve the same thing; rolling off the highs and lows and leaving only the midrange. In Figure 7 (and Clip 7) a distortion plugin has been added to the aux return channel with the high pass filtering; again, this results in the bass signals above 200Hz being distorted, while those below that cutoff / crossover frequency are left unprocessed and without distortion, resulting in a much fuller and more solid sounding bottom, as can be heard in Clip 7. Figure 8: Don't be afraid to try different crossover frequencies. In this example, the crossover point has been moved up from 200 Hz to 500 Hz, which gave a more subtle vibrato effect GENERAL TIPS In general, I recommend picking a filter type and setting and sticking with it throughout the project - always using 18dB, or sticking with only 24db per octave slopes for the entire project and in general, I think the steeper the better -- but remember -- there's a lot of debate and preferences over different types of filters and different slopes in crossovers and filtering in general, so feel free to experiment with different settings and different EQ plugins until you find what you like the sound of. There is a potential for some "weirdness" at the crossover frequencies. Sometimes you can get a bit of a volume boost or dip at the crossover frequency, or a bit of phase shift. Crossovers are always a compromise, but the creative and problem solving possibilities of multiband processing may be worth the slight tradeoff in absolute fidelity. Only your ears can answer that for you. Don't be afraid to mix one band louder or softer than the others to get the sound you want. Things like distortion can change the harmonics and apparent loudness of a band, and sometimes you may need to attenuate that band a bit to compensate. A mono to stereo plugin can be great to add some stereo width to a sound in one band, while leaving the other bands in mono. If you experiment with stereo effects processing on one or more band, make sure you check your mix for mono compatibility to insure that there is no phase cancellation happening. Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Senior Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

Never upgrade right before a session or gig… and when you do upgrade, these tips will help you do it faster, and much more safely By Phil O'Keefe Upgrading anything in a stable system is risky; a large percentage of the equipment related problems I've ever encountered in the studio were due to upgrading or changing something. Upgrading something as major as your DAW program or operating system right before a scheduled session is a recipe for disaster. I always like to test out the gear I'll be using the night before the session to make sure everything is good to go. When doing this, the rule I live by is "if it's working properly, this isn't the time to mess with it!" In fact, it can be a good approach to system upgrades in general, but sooner or later, you're going to need to upgrade software. Here's how to make the task easier and less risky. IT ALWAYS TAKES LONGER THAN EXPECTED It's easy to get tempted by the promise of new features and capabilities, and if you really need the new functions offered in the latest version of your DAW software or operating system, that's a legitimate reason to upgrade, but always make sure to leave yourself plenty of time to not only do the upgrade, but to troubleshoot and fix any potential problems that may arise as a result of it. Those unintended consequences can be sneaky. Upgrade your DAW version, and some of your plugins might not work afterwards. Upgrade your system software, and your DAW program might not work, or you may find that your audio interface needs a new, as yet to be released driver to work properly with the new software. Check on compatibility in advance by reading the system requirements and compatibility information on the manufacturer's websites, but even if you do that, you should be prepared for the unexpected, and give yourself plenty of time to deal with it. BE SMART - COVER YOUR ASSETS Before upgrading anything on your computer, running a full system backup of all your hard drives should be the very first thing you do. I've never regretted having a backup of everything on hand, but if you need it and don't have it, you'll definitely regret not taking this step. There are two other things I've found to be invaluable whenever I need to change or upgrade anything on my computer. The first is a good disk cloning or disk imaging program. This can be used to make a clone of your existing system disk that can be used to restore it to its original, pre-upgrade condition in the event that something goes wrong and you want to revert back to the earlier configuration and software version. The second is a new hard drive. (Fig. 1) After I've run my backup and made a clone of the system drive, I then install that clone disk image on to the new disk drive and proceed to install the software upgrade to that. This allows me to set the original drive aside without making any modifications to it. If I need to revert, putting the old drive back in is often the fastest way to do it. It also saves me the hassle of having to install everything from the operating system on up on to the new drive one program and driver at a time. Figure 1: Replacement hard drives are relatively affordable For those who use desktop or tower based computers, swapping drives can be very fast, especially if you install a hot swap drive bay. (Fig 2) These bays are usually quite inexpensive - typically $30 or less. Look for a trayless design that allows you to insert the drive into the bay without having to mount it into its own separate caddy or tray first. FIgure 2: A hot swappable drive bay, such as this one by StarTech, can make changing your system disk a snap Drive swaps don't make as much sense if you're using a computer that has difficult access to the hard drive, as is the case with many laptop computers and some Mac systems, but if you can get to the drive fairly easily, it's the fastest way to restore the system. For laptops or systems with difficult to access internal hard drives, doing the system backup, creating a disk image / clone with Time Machine (Mac) or Symantec System Recovery 2013 Desktop Edition (PC), and then installing the new software or operating system upgrade is your safest bet. That way, you can revert back to the disk image clone of your old system drive as it was prior to the upgrade in the event that something does go wrong, or if you find that the new software isn't working out for you. Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Associate Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

Music is an auditory art, and our ears are crucial to us. They are also easily damaged by overexposure to loud sounds, and once you lose part of your hearing, it is gone for good - which makes protecting our hearing absolutely essential. There are several options. (Fig. 1) Foam plugs are inexpensive and effective, but many musicians dislike the muffled high frequency sound quality. Muffs suffer from similar sonic gremlins. They are also more expensive, and look funky on stage, although for other noisy environments, such as mowing the lawn, they offer excellent protection. Fig. 1: A variety of hearing protection products, including muffs, foam plugs, and high-fidelity ear plugs Custom fitted plugs that are made from impressions of your ears are the most comfortable. They can be fitted with drivers so that they can serve double duty as in-ear monitors, but they tend to cost in the hundreds. Companies such as Etymotic and Hearos make high-fidelity, reusable earplugs that attenuate all frequencies by approximately 20dB, and provide much more natural sound quality than foam plugs. Best of all, they cost about the same as a pack or two of guitar strings. When things are loud, many people find it's actually easier to pick out individual sounds, such as their own instrument, while wearing such plugs, as opposed to when wearing no protection at all. There are even "designer" hearing protectors, like the ones by V-Moda that provide significant protection yet sit unobstrusively in your ears, and look like you're wearing regular earbuds (Fig. 2). Fig. 2: V-Moda's "Faders" are tuned earplugs with a detachable cord and carrying case. Regardless of which option you pick, it is important to always wear protection when you are exposed to loud sounds. Hammering nails, mowing the lawn, high-volume gigs and practice - whenever it's loud, protect yourself. Your ears will thank you for it! Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Associate Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

The theory and practice behind this stereo miking technique By Phil O'Keefe Stereo mic techniques are great for capturing and creating a natural, or even a hyper-natural sense of width and space. Probably the two most common techniques are the XY coincident arrangement, and a spaced pair. But the lesser-known Mid-Side (“M-S”) mic technique has its advantages too . . . let’s investigate. THEORY A lot of people are hesitant to try M-S recording, maybe because they read about decoders and math formulas and feel overwhelmed. Don’t worry — we’ll keep it simple. You’ll need two mics: One cardioid, and one bi-directional (“figure 8”). Ideally you want two similar mics, but this isn't essential; experiment with whatever mics you have that meet the polar pattern requirements. M-S uses the center or “mid” mic in combination with the bi-directional mic to achieve stereo. As the cardioid mic points right at the sound source, it picks up the direct sound, while the off-axis bi-directional mic picks up the room ambience and reflected sound. M-S stereo “sum and difference” is just the center mic + the side mic for one channel, and center mic - the side mic for the second stereo channel, with the center mic being positive polarity and common to both sides. As the left and right sides originate from the same mic, but with the phase inverted, collapsing an M-S recording to mono cancels out the left and right sides from the bi-directional mic, leaving only the positive polarity signal from the center (cardioid) microphone. This significant advantage of M-S recordings insures perfect mono compatibility without any phase issues. SETTING UP THE MICS Aim the cardioid mic directly at the sound source. As with normal cardioid mic placement, adjust the “aim” to taste; but if you're a fan of close miking, try moving back a bit further from the source for M-S recordings. Next, place the figure-8 mic so that the two lobes of the pattern are set 90° relative to the cardioid microphone. M-S is a coincident microphone technique, so you want to get the diaphragms of the two mics as close together as you can. Fig. 1 shows a Soundelux (also known as "Bock Audio") E250 (bottom) and ELUX 251 (top) set up as a M-S pair. The cardioid E250 is pointed at the sound source (in this case, the camera), while the pattern selector on the ELUX is set to bi-directional; it’s picking up to the left and right, and its side null point points directly at the sound source/camera. Fig. 1: A Soundelux E250 (bottom) and ELUX 251 (top) set up as a M-S pair SETTING UP THE BOARD AND RECORDING At your DAW, simply route each mic to its own preamp, and assign the cardioid mic to a single DAW track. You can either record the bi-directional mic to two identical, separate tracks of its own and invert the polarity of one of the two tracks later, or record the bi-directional mic to only one track and use a decoder plug-in, or clone the single bi-directional mic's track later and invert the clone track’s polarity — your choice. I generally record the bi-directional mic to two tracks simultaneously in Pro Tools, labeled “SIDE+” and “SIDE-,” and insert a Trim plug-in on the “SIDE-” track to invert the phase. Fig. 2 shows a basic M-S track arrangement in Pro Tools. Now group the two “SIDE” tracks so that any changes you make to the volume level of one will apply to the other track simultaneously, and pan the tracks hard left and right. As you raise the level of the side mic tracks, the stereo width will increase; lowering them decreases it. Being able to adjust the amount of stereo information in the recording after the fact is one of the big advantages of M-S recordings. Fig. 2: A basic Mid-Side track and panning arrangement in Pro Tools. Note the use of the Trim plugin to reverse the polarity (red arrow) on the "Side -" cloned track FREE DECODER RING INSIDE! So why would you need a decoder or hardware matrix box? The downside of simply cloning the “side” mic track and inverting the polarity is that you can’t hear the final results as you position the mics — you have to record them, then flip the polarity and start playback. By encoding the sum and difference data from the two mics and recording the result, you can hear how the stereo field will sound before you track. PAiA has a hardware M-S matrix box schematic available on their website for you solder jockeys: http://www.paia.com/ProdArticles/msdecwork.htm. For other DAW users, Voxengo’s free MSED M-S decoder plug-in (Mac AU/VST, Win VST) works very well; download it from http://www.voxengo.com/downloads/. M-S might not be used as frequently as some other stereo techniques, but the perfect mono compatibility and ability to adjust the stereo width at mixdown make it handy for broadcast production, live recordings of small ensembles, individual instruments, and small groups of background vocalists on multitrack music sessions. Give it a try! Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Associate Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

Sure, you can do a lot of neat stereo tricks with a panpot—but it's not the only way to conjure up some stereo mojo By Phil O'Keefe I'm a stereo freak. I just can't seem to get enough of it, and I love it when things move, spin and fly around the stereo sound field - that is, as long as it is musically appropriate; but that's a subject for another day. Let's look at stereo placement and various ways to position sounds within the stereo field. WHAT IS "STEREO" ANYWAY? The way the brain interacts with our two ears and processes the sound waves that are received by them allows us to spatially localize sounds and determine the direction they are coming from - with some limitations. Some sounds, such as very low frequency sounds without a lot of overtones - such as distant thunder - are difficult to localize. But with midrange and higher frequency sounds, our ears and brain do a reasonably good job at spatial localization. The ridges of our outer ears - called the pinnae, reflect and direct sound into the ear canal, and help us to locate sounds coming from the front and rear. Our brain also uses arrival time differences to determine location - sounds coming from the left will arrive in our left ear a small fraction of a second before they reach our right ear. But neither ear works in complete isolation - in our daily lives, it’s very uncommon to hear sounds isolated in only one ear without hearing some sound in the other ear. Also, due to the acoustical masking effect of our head and the attenuation of sound levels at greater distances from the sound source, level differences also come into play in how we locate the direction a sound is coming from; although again, lower frequencies are more difficult to determine directionally. Armed with a basic understanding of how we perceive sounds, we can put that to use in our mixes. PAN KNOBS The first thing many people reach for when they want to place something in the stereo sound field is the pan knob. Rotate the pan knob left on a mono track, and the audio from that track moves progressively further to the left in the sound field, with more of that sound coming from the left speaker, and less from the right as you turn the knob further to the left. Sounds simple, right? Actually, it is... but panning isn’t the only stereo positioning tool in the arsenal. STEREO TRACKS Pan knobs are fine for positioning mono tracks, but what about stereo tracks? Some people will take the left and right outputs from a stereo keyboard and pan them hard left and right, and then do the same with the doubled guitar parts, and the drum machine tracks and background vocal tracks... and then they wonder why their mixes lack clarity and why it is difficult to hear individual elements of the mix. It's because they've got everything stacked on top of everything else! Parts that come from the same location in the stereo field will tend to blend together into a composite sound instead of discrete, individual sounds. Panning things to the same spot can be useful when you're trying to blend sounds together, but counterproductive for getting stereo separation. Instead of automatically going wide with every stereo source, I recommend finding a separate location in the sound field for various different musical parts and limit the amount of parts that share the same location in the stereo field. You can use the pan controls on a stereo track to adjust their location in the mix in various ways. Narrowing the width of a stereo track via panning is relatively easy - instead of panning hard left and right, try setting the pan controls to 50 / 50, which will narrow the image but still keep it centered in the stereo field, or 25 / 75, which will be equally wide but will move the image further towards the right. Pan the keyboard 75 / 25, the background vocals 25 / 75, the drum machine 50 / 50 and the guitars hard left / right, and the individual sounds will become much easier to hear than if you just panned them all hard left / right, because now each will be occupying its own individual sonic real estate. THERE'S MORE TO LIFE THAN JUST THE PAN KNOB There are other ways to adjust the positioning of stereo tracks beyond just tweaking their pan controls. One way of manipulating the width of a stereo track that is sometimes overlooked is by adjusting the relative volume levels of the left and right channels. Try gradually lowering the volume of the left channel of a stereo file by 6 dB and listening to what happens to the stereo image - it should move further to the right as you decrease the left channel level. If your software does not have individually adjustable level controls for the left and right sides of a stereo track, you can usually "split" the stereo track into two mono files and proceed from there. When working with Mid-Side (M-S) stereo tracks (Basics Of Mid-Side Recording http://www.harmonycentral.com/docs/DOC-1722), lowering the "side" channels from the bi-directional mike will make the stereo image narrower, until you're finally left with nothing but the mono signal from the center cardioid microphone. You can also adjust stereo positioning by using arrival time differences - using delays to move the signal further left or right. If you insert a short delay plug in on the left channel of a stereo track and adjust it for 100\% wet (full delayed signal) and set the delay time for 10 - 20 ms or so, the image will appear to come more from the right side. The longer the delay time, the wider the image will sound, but beyond a certain point longer delays will start to sound like discrete echoes instead of a stereo image. Short delays of this type are also commonly used to create pseudo-stereo from a mono source, by using either a mono to stereo plug in and applying the delay to only one side, or by using an aux send to route the signal from a mono track to a short delay and then panning the aux return channel to a different location than the original source track's pan position. Remember to check your phase relationships by occasionally listening in mono when using this method, and if you hear the sound get thin or weak, try a slightly different delay time or hitting the phase inverse switch on the aux return channel or the delay plug in. Other useful plug ins for creating pseudo-stereo images from mono sound sources include modulation style processors such as chorusing or stereo flangers, as well as using different EQ filtering on the left and right channels. Try taking a track, making a copy of it, then rolling off the highs on one and the lows on the other and panning them in different locations. Feel free to experiment with combinations of these various techniques. In Fig. 1, I have used 75 / 75 panning on the stereo acoustic guitar (tracks 1 / 2) to narrow the width a bit, along with a short 12 ms delay on the right channel (track 2) to move the image to the left a little more. Figure 1: Examples of various stereo placement techniques Some plug ins, such as Voxengo’s excellent (and free) Stereo Touch (AU / VST) plug in uses a combination of delays, EQ filtering and panning to create pseudo-stereo from mono tracks. You can download a copy at http://www.voxengo.com/product/stereotouch Track three in Fig. 1 is a mono background vocal that has a mono to stereo instance of this plug in inserted, and the pan controls are set to 25 / 100 to put the image more towards the right, and thus into a different location in the sound field than the acoustic guitar. Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Associate Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

When recording by yourself, these tools will make the job easier, and you'll get better results too By Phil O'Keefe One of the biggest challenges facing a guitarist who works solo on recordings is that they're alone. You simply don't have enough hands for all the things you need to do at once, and you can't be in two places at the same time. When you're working in a professional studio and recording guitar parts, things are a lot easier. You don't have to worry about running the equipment, which lets you concentrate on your playing. Because there's an engineer there to help, you can play while someone else worries about mic positioning and capturing the sound. When working alone, it can be challenging to try to play and simultaneously optimize your mic positioning. And commercial studios usually have the advantage of separate control and tracking rooms, which makes it easier to hear how changes in the mic positioning affect the sound, as well as how the sound of the guitar amp is "sitting" in the mix relative to the other parts. Let's look at a few ways you can get around some of these challenges when recording at home by yourself. FIRST, THE SETUP Maybe you have a great sounding room where your guitar amp really comes to life, but that is in a separate part of your house from your studio gear. In fact, most residences have a variety of acoustical environments available that many recordists never bother to take advantage of, simply because it's too much work to move everything. With the right tools, you can start to change that and take advantage of the variety of acoustical environments at your disposal, and without having to move all your recording equipment into another room. Running long guitar cables from room to room is a recipe for high frequency signal loss and increased noise interference… but recording tracks from the control room allows you to operate the gear easier, and to listen over studio monitors instead of headphones while the guitar speaker(s) remain isolated in the other room. One method is to keep the amp head in the control room, and just run a long speaker cable out to a speaker cabinet in the other room. That can work if the two rooms are relatively close to each other, and it offers the advantage of having the amp there in the control room where you can easily adjust it, but what if you're recording a combo amp? To get around that, I recommend the Radial SGI (Studio Guitar Interface system) (Fig 1) or the Little Labs STD (Signal Transmission Device). Figure 1: The Radial Engineering SGI allows you to run your amp up to 300 feet away without noise or signal loss Both of these take the high impedance output from your guitar and convert it into a low impedance signal. The low impedance signal is then routed over an XLR cable, which is better suited for the long cable run. At the other end of the cable is a box that converts the signal back to the high impedance signal that your amp is designed for. (Fig 2) This allows you to isolate your guitar amp in another room where the acoustics might be better suited to the sound you're after, while you play in the control room where you can operate the equipment and monitor over your nearfield speakers. Figure 2: A typical setup using the Radial SGI LESS RUNNING BACK AND FORTH If you have the amp isolated in an adjoining room, it's going to be difficult to hear how everything sounds over the speakers in real-time as you reposition the mic, and you'll need to go back and forth from the tracking room to your control / equipment room to check on how things sound. If you're working with the amp in the same room as you are, you're forced to wear headphones so that the studio speakers aren't also picked up by the microphone. With a loud guitar amp nearby, it can sometimes be tough to discern the subtle differences that small changes in mic position can give you while listening on headphones. Short of recording "direct", or using a amp simulator, how do you get around these issues? First of all, stick with closed back, circumaural headphones. Closed-back (sealed) circumaural (over and around the ear) headphones with good isolation block much of the sound around you from reaching your ears, allowing you to better hear the sound change as you're dialing in the best mic positions. Headphones with great passive isolation include the Sennheiser HD 280 Pro, Direct Sound EX-29 Extreme Isolation (Fig 3), and KRK's KNS 8400. All of these headphones feature around 30 dB of attenuation of outside sounds. Figure 3: Headphones like the Direct Sound EX-29 Extreme Isolation models shown here help block out external sounds so you can hear what you're doing, even with a loud amp nearby If the amp is really loud, it can still be kind of hard to hear the sonic changes as you adjust the mic positioning, but there are ways around that too. If you find yourself in that situation, try using a pair of noise isolating earbuds or earphones ("ear plug" style headsets), and then put a pair of unplugged headphones with good isolation characteristics on over them. Since the headphones only serve to add additional isolation from outside sounds, and don't provide any of the actual sound you'll be listening to, you can even use "shooter's ear muff" style hearing protection in this application instead of headphones. The best isolation results will be with earbuds that fit well and seal snugly into the ear canal, coupled with the best isolating headphones you have available. The in-ear earphones send the sound straight into your ear canals, and their already excellent isolation (from the way they "seal" into your ear canal) is increased by the external closed-backed headphones. Hearing protection ear muffs are available from companies such as Pro Ears, Peltor, Remington, and 3M, and can be found in most sporting goods stores. Since they lack any drivers or wiring, they typically cost significantly less than a set of headphones with similar isolation characteristics. Earbuds with excellent isolation include the Shure SE215 (Fig 4), and the Etymotic ER-4 - both of which can provide a whopping 35 dB or more of reduction on their own. When paired with a set of closed back headphones such as one of the models I mentioned earlier, outside sounds can be cut by an additional 6 to 10 dB or more. This level of attenuation is similar to what you'll get if you have the amp closed off in one room of your house while you're in the adjoining room with the doors shut. Figure 4: In-ear headsets, such as the Shure SE215 can provide even more acoustic isolation when worn "under" a pair of sealed, circumaural headphones NOT ENOUGH HANDS? Ideally, you'll need to have the sound coming out of the amp while you're trying to position your microphones. In pro studios, it's not uncommon for an assistant to move the mic around as you play, while the engineer listens over the control room monitors and directs; telling the assistant which direction to move the mic until they're satisfied with the mic position and sound. Obviously this is less practical when working alone. Yes, you could ask a friend or family member to help out occasionally, but for those times when you're on your own, there's another solution - use a looper pedal to "play" the guitar while you make adjustments to mic positions and overall sound. Some popular loopers include the TC Electronic Ditto (Fig 5), Boss RC-3, Line 6 JM4 and the DigiTech JamMan Solo XT, but you may already own a looper. Many multi effects pedals include looping functions, such as the Zoom G3, Boss ME-70, and the one I use - the Line 6 M9. Figure 5: Loopers like the TC Electronic Ditto allow you to "play" the guitar while freeing up your hands to adjust the mic positioning Any one of these looping tools will allow you to record a riff or phrase and then play it back over and over, freeing up your hands so you can adjust the mic positioning and fine tune the control settings on your amp for best results. Because you're not stuck in one place, you can even go back and forth between the amp room to adjust the mic position, and then to the control room or recording equipment room to check on how it sounds over your monitors. Ideally you should use the looper to record a snippet of the song and part you're preparing to track, so that you can hear how things work with what you'll actually be recording. As you can see, with a little forethought and the right tools, you can get past many of the limitations that come from recording alone. By using these suggestions, you will be able to hear what you're doing much better, which nearly always translates to better sounding recordings. Now go wax some killer tracks, and when you're done, don't forget to stop by the forums and share them with everyone! Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Associate Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

Getting your speaker geometry correct is easy with the right tools By Phil O'Keefe Many of you will already be familiar with the common recommendation of setting up your near field monitors in an equilateral triangle arrangement. In such a setup, the monitors are placed the same distance from each other (typically, 3-4 feet) as they are from your listening position, so that the two speakers and the listener's head sit at the three points of an equilateral triangle. This is relatively easy to achieve with little more than a tape measure, or a few feet of string. But what about toe-in? Toe-in, or angling the speakers inwards towards the listening position so that the listener is more on-axis with the speakers, is another often-recommended monitor setup technique, but it can be a little tricky making sure the two monitors are angled exactly the same way. You can use a protractor, but most are too small to easily get the angles of each speaker matched up more accurately than "fairly close", and if you have a surround speaker setup, things become even more complicated. MODERN TOOLS TO THE RESCUE Do you have an iOS device (iPhone 4 / iPad 2, or later), or an Android device containing a gyroscope sensor, such as a Nexus or Galaxy tablet, HTC Evo 3 or 4, or Galaxy II or III? If you do, check out Genelec's handy SpeakerAngle app. (Fig 1) This very affordable ($0.99 USD) app simplifies the task of setting up and matching your speaker angles to such a degree (pardon the pun), I now consider it a must-have for anyone interested in recording - or for that matter, anyone who is interested in listening to music played back over speakers, or who wants to optimize their home theater's surround sound speaker setup. Figure 2: SpeakerAngle supports surround sound systems up to 7.1 For home theater owners and those who are lucky enough to have studios with surround sound monitoring, SpeakerAngle can be used to configure 5.1 and 7.1 surround sound setups. The basic steps involved in configuring a surround sound system are similar to the ones used in stereo mode. The application's own in-app instructions are clear and complete, but it really is a simple program, and many users will probably be able to figure it out without ever having to refer to the instructions. Still, the instructions are full of useful information to help you with the process of getting your speakers set up optimally, so I'd recommend having at least a quick glance at them. But even if you never check them out, do check out this app. If you use speakers, it makes the process of setting them up correctly a lot easier. iOS device users can purchase Genelec's SpeakerAngle app through the Apple iTunes store, and Android users can purchase it through Google Play.

Tips and tricks for getting different tones when recording multiple guitarists, or overdubbing multiple guitar parts By Phil O'Keefe I went over the basics of guitar amp miking in my Guitar Amp Miking 101 article, so if you need a refresher on mic techniques, please have a quick look. In this article, we'll be going into greater depth regarding multiple-guitar parts in recordings. It's not uncommon for recording engineers to face situations where the band being recorded has two guitarists, or when a single guitarist (maybe you) wants to track multiple guitar parts for a single song. Doing this can help add interest and variety to the mix, but in some situations, it can lead to trouble; if you don't watch what you're doing, you could end up with cluttered and muddy mixes, difficulty in differentiating the parts, or even overly-complex arrangements that confuse the listener. Here are some tips and tricks to help you get around some of these common issues, and help you give each guitar its own distinctive identity when using multiple guitar parts. TAKE STOCK Have a look at the rigs you'll be working with and make some notes. If possible, do this well in advance of your tracking session. This will help you spot any potential issues in advance, while there's still time to correct them -- such as instruments with poor intonation, or bad action and buzzes. And of course, have any problems repaired before the session. Also remember that amplifiers are essential parts of the instrument we call "electric guitar," and make sure they're in excellent health too. Any bad speakers, ground issues or weak or microphonic tubes should also be repaired or replaced. Don't forget to ask about any alternative resources they might have available. Just because something isn't a part of their usual "live rig" doesn't mean it won't be useful for a recording session. Practice amps, amp sim pedals, different guitars and effects can all provide alternative sounds for different parts on the recording. However, be careful not to go into the studio with a pile of unfamiliar gear -- while a new toy or two (or even three) can be inspiring on a recording session, too much unfamiliar gear can lead to option overload and anxiety; it's better to have a good idea of what tones you want (and how to get them) before you enter the studio, rather than being overwhelmed by an endless sea of "almost but not quite" and "that's close, but maybe I can find something I like better" options and decisions in the studio. Trust me, you'll face enough decisions in the studio already, so save yourself the trouble and make as many in advance as you can. HAVE A PLAN Based on the resources at your disposal and everyone's goals, have a plan for the tracking session. It doesn't have to be super-specific, with every detail and moment of time spelled out, but a good general idea of what you're looking for is important. Compare your plan with the available gear list. If your plan calls for gear that isn't already available, make arrangements to rent or borrow what you'll need for the tracking session. The arrangement is absolutely crucial to the success of any recording project. This is where pre-production can be a real lifesaver. The band should have a good idea of the parts and the arrangements they want to use on their recordings, and just as importantly, so should the producer and engineer. Attend some pre-recording rehearsal sessions to get an idea of the arrangements, and work through any potential problems well in advance. Home recording rigs can really come in handy for pre-production arrangement ideas. The goal isn't to track the ultimate parts at home, but to work out arrangements in advance. This allows you to figure what does and doesn't work for the song, and do it when the clock isn't ticking and the pressure is far less intense. It also allows you to work on your overdub ideas and have them well rehearsed in advance of the session - all of which will save you time and money when you go in to "do it for real." Make sure you don't try to over-extend yourself. It's not uncommon to have players struggle to lay down a "killer part" that is just beyond their technical abilities, but that lives up to some nebulous ideal they have in their mind… then really struggle when they try to double-track it. In those situations, artificial doubling methods may be in order. An even better option is to rehearse in advance; being realistic about your abilities and adjusting your parts accordingly. Ideally you should be confident and capable of nailing any part on the recording within a couple of takes. If you need more than three, you may be over-reaching. Simplify the part or plan on using another way of doubling it. SPECIFIC TIPS Sometimes you want to have two tracks blend into one composite sound, which is relatively easy to do by playing the two parts as identically as possible, on similar rigs and panning them to the same location in the stereo field. It's far more likely you'll run into situations where you want to differentiate the two guitars, and allow each to be plainly heard. Sometimes this is as simple as panning each to a different spot in the stereo sound field, but there's a lot more options available to you beyond mere panning. Here are some specific suggestions that can help when tracking multiple guitar parts. CHANGE THE GUITAR Doubling the part with the exact same rig can often lead to muddy parts. Instead, try switching instruments. Use a baritone for the doubled part, or even an acoustic guitar. Something as simple as substituting a Strat on the overdub instead of re-using the ES-335 can often be enough to give each guitar its own identity, and can add extra thickness and dimension compared to doubling with the same setup. Figure 1: Sometimes getting a different guitar sound is as simple as using a different guitar. Try using a Tele for the doubled part instead of re-using the Les Paul CHANGE YOUR APPROACH TO PLAYING THE GUITAR Even if you don't have access to multiple guitars, all is not lost. By arranging the parts differently, you can still give each their own voice and identity. By using different intervals and chord inversions, or capoed and uncapoed parts, you can move each part into its own frequency range and sonic space - even if they're playing rhythmically identical parts. Using different picks, tunings and alternate playing techniques such as slide and e-bow can provide additional layering options to explore. CHANGE THE AMP You can get a nice change of tonality by simply using a different amplifier for the overdubbed part. Try a Marshall instead of your usual Fender. If you're not sure what to look for in terms of different amp sounds, look for something with a different type of tubes. If your amp uses EL34 or EL84 tubes, try to borrow one with 6L6 or 6V6 tubes. If you use tube amps, try a solid state amp as a tonal alternative to your usual setup. Don't forget speaker cabinets! Different speakers can make a big difference to the sound of a guitar amplifier, as can closed-back cabinets vs.. open-backed models. In addition to amps and speakers, don't overlook the wide range of alternatives that are available to you with amp simulator pedals, desktop units and plugin software. While these may or may not provide you with your normal "ideal tone", they can provide a staggering range of alternative tones that can be great for layering. Figure 2: If you used an EL84 based amp like the Marshall Class 5, try using a 6V6 based amp like a Fender Princeton for the overdubs CHANGE THE RECORDING SIGNAL PATH Did you use a ribbon mic for the basic tracks? Then try a condenser to bring out the shimmer and treble for that arpeggiated overdub part. Use a dynamic mic on one amp for the first part, and then a ribbon or condenser on a different amp when "doubling" that part. Don't forget that different microphone and preamp pairings can provide different tonalities too. CHANGE THE ACOUSTICAL ENVIRONMENT When doing an overdub, consider putting the amp into a different room, or moving it to a different location within the same room in order to give it its own acoustical "space." You can also vary your microphone placement in order to change the ratio of dry, direct from the amp sound vs. the room ambiance; moving the microphone closer for a drier and more immediate sound or further back for a more distant and spacious sound. Remember - the further from the sound source you move the mic, the more "distant" the sound will tend to be in the mix. Blending close and distant microphones on doubled guitar parts can provide you with a huge sense of depth. CHANGE THE SETTINGS If you don't have multiple guitars and different amps, it may be more challenging, but all is not lost! You can always try different control settings on the guitar and amp that you do have to get different tones on the overdubs and layers. If you are limited to one guitar, try a different pickup setting for the overdub. If you only have one amp, try different gain and tone settings. Different effects pedals can be especially helpful here -- especially overdrive and distortion pedals with "amp-like" tonalities. Fulltone, Catalinbread, ZVex, Tech 21 and many other companies make pedals that are designed to make your Fender amp sound more like a Marshall, or a Vox, or a Hiwatt. While not always identical to the "real thing", they can provide some very cool alternative tones of their own, and a taste of those other models, and at a much lower cost than a large amp collection. CHANGE THE EFFECTS As a general rule of thumb (and when it comes to recording, remember that rules are made to be broken whenever doing so makes things sound better), I feel that busier parts and more complex chords should usually be drier and have less effects and reverb overall. Leads and other single note lines can usually have more effects on them, depending on the part. Long, sustained parts and ringing arpeggios can go either way. The point is, by adjusting the level and type of effects, you can give each part its own sound and sonic "space," or provide extra texture and thickness to layers and stacked parts. For example, try recording the first pass completely dry, and then kick on a chorus pedal for the overdub. Speaking of effects, if you have a great tone happening, by all means, track it. However, be aware of the amount of effects you're using and don't go overboard - you can't get rid of it after the fact. It's for that very reason that many engineers wait to apply a lot of effects until the mix - things like reverb and tempo based effects can be added later, and often the plugins and rack units will have better sound than the pedals for these sorts of effects anyway. They're certainly easier to "tempo sync" than most pedal based effects. IN THE MIX FIXES It's far too tempting to rely on the old idea of "we'll fix it in the mix", and while tracking it right to begin with nearly always beats trying to fix things later, sometimes you have no choice. If you can't re-track the parts, then you can use some of the other methods to put each one into its own space in the mix. Panning them to different locations in the stereo field can often help. If the parts were recorded with close-mikes (and are relatively "dry"), you can easily add different reverb and delay treatments to each one. Don't forget EQ - you can often cut one frequency range back on one guitar while leaving it alone on the other track. Careful, multi-band EQ adjustments can accentuate certain frequencies of one part while accentuating different frequencies in the second part, giving each their own individual tonality. Another useful tool is the mute button -- just because you have a doubled guitar part recorded for the entire song doesn't mean it's always in the best interests of the song and arrangement to let it run non-stop. Instead, try muting the doubled part on the verses and bring it back in only on the choruses, or on the bridge to give it added impact. Remember that good arrangements should have some flow and variation to them, and shouldn't just sit there without changing… but the time to consider the arrangement is well in advance of the actual recording session. Give some thought and rehearsal to the arrangements before the session and your doubling sessions will be far more productive. Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Associate Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

Let your tracks live in a parallel universe By Phil O'Keefe One of the really cool features of most DAW software is a customizable virtual mixer, where you can create layouts and templates for recall at a later date. But in terms of customization, I’m also a huge fan of being able to add auxiliary channels to the mixer for functions such as effects sends and returns. Usually with an affordable hardware mixer you are limited to a just few aux buses, so those tend to get used up pretty quickly for reverb, headphone monitoring sends, etc. But in software, adding an almost unlimited number of sends is usually just a few mouse clicks away. It certainly beats having to sell your hardware board and buy a new model just to get some more aux sends! Having lots of available aux sends opens up the possibility of using some of them for “bus compression” or parallel processing. By parallel processing I don’t mean dual CPUs in your computer, but parallel audio paths for the direct signal and the processed signal, with each on a separate fader. ENTERING THE PARALLEL UNIVERSE Suppose you have a set of drum tracks, or several background vocal tracks, or multiple acoustic guitar tracks, and you’d like to compress all of them a bit — either to smooth them out, or add a bit of punch. You could insert an individual compressor inline on each track by using your software mixer’s channel insert points, and sometimes that may indeed be your best option. It will certainly allow you to use different attack and delay times — or a completely different compressor — on each track if desired. But it takes more CPU power to run all those compressor plug-ins simultaneously, unless you call on some external hardware assistance (e.g., SSL Duende, FocusRite Liquid Mix). Besides, sometimes you just want a bit of compression across all the drum tracks, or background vocal tracks, using the same type of compressor. This is where routing aux sends to a compressor inserted on an aux return channel can come in handy. The concept is similar to using an aux bus and aux return as a reverb send and return, instead of inserting a separate reverb plug-ins for each track. SETTING IT UP Add a stereo aux return to your software mixer layout. In Pro Tools, do this by hitting Ctrl-Shift-N, and selecting “1, Stereo, Aux Return” in the ensuing dialog box. Upon creating your new stereo aux return, assign the input of that aux return channel to an aux send. In Fig. 1, the input for the aux return channel is set to aux (“bus”) 1-2. Add an aux send (again, using aux/bus 1-2) to each track you want to send to the processor. If you have “Send levels follow groups“ enabled in your Pro Tools preferences (under the “automation“ tab), you can then raise or lower all the aux sends on any grouped set of tracks by simply raising any one of the aux sends — and all of the aux sends assigned to the same bus will follow suit for every channel of the group. Of course, you can disable the group and adjust the send levels for any tracks individually as well. Adjust each send’s panning individually. With drums, I normally use the same panning for the aux sends as for the stereo mix. After setting the levels and panning, tap the signals off of the original tracks, and bus them to your aux return channel; this is where any inserted effects or processor plug-ins will process the signal before sending it to the stereo mix bus via the aux return channel fader. This allows you to bring the compressed signal up on a separate fader, and blend it in with the original, unprocessed tracks. Figure 1: Drum tracks 1-6 are all sending some of their signal to Bus 1-2. This goes to a Compressor/Limiter, the output of which returns into the virtual mixer via track 7. This technique makes it easy to fine-tune the amount of compression you want, and because it’s working in addition to the original source tracks, you get a more balanced and natural final result, with less “squash.” Just raise the aux sends until you’re hitting the compressor as hard as you want, then slowly raise the aux return fader until you hear the desired mix of unprocessed and processed signals. Of course, you can insert other types of plug-ins either before or after the compressor on the aux return channel. Placing an EQ plug-in before the compressor will tend to change the way the compressor responds, while placing it immediately after the compressor will tend to shape the overall sound and timbre of the compressor’s output. Neither approach is “right or wrong” and both can be useful, so experiment with both approaches and decide which one sounds best. Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Associate Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

Wattage, Speaker Efficiency and Amplifier "Loudness" When it comes to volume, wattage is only part of the equation By Phil O'Keefe There seems to be some confusion when it comes to how "loud" an amplifier can get. When it comes to "volume", many musicians only consider the amplifier's power or wattage rating, and in general, more watts does mean "louder". But while wattage is an important consideration, the efficiency of the speaker(s) that are connected to the amplifier are also an important factor in the loudness equation. DECIBELS AND LEVELS Decibels (abbreviated "dB") are a logarithmic unit of measurement that pertain to a ratio between two numbers. Okay, I can see eyes rolling and glazing over, so I'll simplify things, and attempt to keep the "math" to an absolute minimum. With a logarithmic scale, you can't just add numbers in the usual way - a doubled number isn't "twice as much", but rather, many times more. For example, 100dB is many times greater than 50dB, not just "twice as much". When it comes to "loudness", which is measured in Sound Pressure Level, (or SPL), a 10dB increase in level is roughly equivalent to a "doubling" of perceived loudness. In other words, if one amp is generating 90dB SPL and another amp is hitting 100dB SPL, the second amp will generally be perceived to sound about twice as "loud" to the typical listener. WATTAGE, POWER AND SPL So how many watts does it take to get twice as loud? Let's imagine two amps - one of ten watts, and a second of twenty watts. The twenty watt amp is double the power of the ten watt amp, but doubling the power only translates to an increase of 3dB SPL. Remember, in order to sound "twice as loud", you need an increase of 10dB, so while a twenty watt amplifier will sound noticeably louder than a ten watt amp, it will not sound twice as loud. The same thing holds true at higher wattages - a 100W amp is not going to sound twice as loud as a 50W amp; assuming identical speakers, it will only be 3dB louder, which is noticeable, but definitely not a doubling of perceived loudness. SPEAKER SENSITIVITY RATINGS Speakers have specifications in terms of their sensitivity and efficiency - their ability to convert the incoming electrical energy into acoustical energy. Dynamic, moving coil speakers (the type found in most guitar and bass amps) are notoriously inefficient, and most of the incoming power is actually converted into heat, and not sound. Normally, speaker sensitivity is measured in a anechoic chamber (non-reflective, soundproof room) and expressed something like this: 90dB @ 1W / 1m Translated into English, that means "ninety decibels (SPL) with one watt of power, and measured at a distance of one meter from the speaker." A more efficient speaker will have a higher number, and a less efficient speaker will have a lower number. All other things being equal, a more efficient speaker will make your amp sound louder than if it has a less efficient one installed. (Fig. 1) Figure 1: While the two amps pictured are nearly identical in power (20W vs 18W), the one on the left is significantly louder due to its much more efficient speaker PUTTING IT ALL TOGETHER So let's assume we have a speaker with a sensitivity of 90dB @ 1W / 1m and a power handling capacity of up to 100W. If we power that speaker with 1W of power, it will generate 90dB when measured at a distance of 1 meter. If we double that power to 2W, the SPL measurement will increase to 93dB. If we increase the power to 10W, then the SPL measurement will increase to 100dB, which is "twice the perceived loudness" when compared to 1W. So it actually takes ten times more power to give us a perceived doubling of volume level. Since this imaginary speaker is rated to safely handle up to 100W, we could double that volume level yet again, and in theory, hit up to 110dB SPL by increasing the power all the way up to 100W. One watt = 90dB. One hundred watts, or 100X more power = 110dB. That's a huge increase in power but only a "doubled double" (4X) increase in terms of perceived volume levels! As you can see, it takes considerable increases in power - in the wattage of the amplifier - to "double" the perceived "volume". This is where speaker sensitivity / efficiency comes into the equation. If we replace that 90dB @ 1W / 1m speaker with a model that has a sensitivity of 100dB @ 1W / 1m, the numbers change dramatically. For starters, 1W of input power will give us 100dB SPL. Remember, the first speaker required 10W to achieve that same volume level! So by installing a more efficient speaker, we can get the same perceived volume level from a 1W amp as we could from a 10W amp that is coupled to a less efficient speaker. Again, this applies all the way up to the maximum power handling capacity of the speaker. Assuming our 100dB @ 1W / 1m speaker can also handle up to 100W, it can give us up to 120dB SPL; again, that's double the perceived "volume level" of the 90dB @ 1W / 1m 100W speaker's maximum level of 110dB SPL. AMPLIFIER POWER PLUS SPEAKER EFFICIENCY AND POWER HANDLING = MAXIMUM VOLUME So remember, while increasing the amplifier power can make you louder, increasing the speaker sensitivity will make more efficient use of the available power from any amplifier. This means it's impossible to make generalizations about the "loudness" of any amplifier based solely on its wattage. You simply must factor in the power handling capacity and sensitivity of the speakers in order to know "how loud" it will be capable of getting. If your 15W amp has a relatively inefficient speaker installed, but is still "almost" loud enough for your needs, you may not need a higher wattage amp - simply installing a more efficient speaker, such as the Electro-Voice EVM 12L in Fig. 2, may give you all the increase in volume you seek, without having to replace the entire amplifier. Similarly, you may not need a 100W amp; replacing the stock 95dB @ 1W / 1m speakers in your 50W amp with new speakers that are rated at 101dB @ 1W / 1m will more than make up the difference in terms of the "volume levels" you will be able to generate... it will actually be capable of "sounding louder" than that 100W amp will when it is running into the less efficient speakers. Figure 2: Replacing inefficient speakers with a highly efficient speaker model, such as this E/V EVM 12L, will make any amp sound louder Of course, if you really want to get loud, then the answer is to couple a high power amplifier with high efficiency speakers that are rated to handle the power... -HC- __________________________________________________ Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Senior Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.

By Phil O'Keefe Posting images in the Lithium software is a bit different than it was in previous HC software. Here's how it's done. First of all, either create a new topic, or reply to an existing one. A Post Message box opens up, with the Rich Text tab selected as the default. Immediately below the rich text tab you'll find a line with an assortment of tools and icons, including B (bold text) I (italic), U (underscore), strikethrough, the icon for the spoiler tag, Insert Code icon, Paste from Word icon, insert emoticon icon, insert link icon, insert image icon, insert video icon, and two list icons for ordered and unordered list. The insert image icon has a graphic image of a little tree on it. (Fig 1) To insert an image the steps are fairly straightforward, but differ slightly, depending on whether your image resides on your local computer, or is online in an image hosting site such as Photobucket. Let's look at each separately. Figure 5: Previously uploaded images can be easily re-used without having to re-load them

Aguilar Announces "Soapbar"-Shape Bass Pickups Posted by Harmony Central News on Jan 13, 2013 10:47:25 AM New York, NY — Aguilar, the leading edge USA manufacturer of high quality bass pickups, is pleased to announce the expansion of their pickup line with two series of soapbar bass pickups. These pickups will be available in a variety of 4, 5 and 6-string standard soapbar sizes. The first of these, the Super Singles™, give you massive single coil sound in a soapbar size. Players can expect all the articulation and nuance of a single coil pickup combined with great sustain and harmonic development. These pickups will give you excellent string-to-string response and even tone in all registers. The DCB™ pickups use dual ceramic bar magnets that offer a dynamic and responsive attack that works for all playing styles. These pickups offer well-developed overtones and singing sustain in the body and decay of the notes. The DCB pickups sound great installed in a passive bass or in conjunction with an onboard preamp. Like all Aguilar pickups, the Super Singles and DCB’s are wound in Aguilar’s NYC factory. They will be available starting early March 2013. The street prices for both pickups are TBA. Since their release in 2010, Aguilar pickups have become the new standard for exceptional bass guitar pickups. Providing the rich, musical tone that Aguilar gear is known for, Aguilar pickups are used by top luthiers and bass players the world over. Whether you are looking for period-correct vintage flavors or the most modern tones; Aguilar has the pickup for your bass. Your bass has a voice, let it be heard. Aguilar pickups are used by many famous bass players including Justin Meldal-Johnsen (Beck, NIN, and many others), Anthony Wellington (Victor Wooten), Jeph Howard (The Used), and Nathan Thomas (Scotty McCreery). For more information, please visit: www.aguilaramp.com.

Steinberg reveals compact, ultra-rugged USB 2.0 audio and MIDI interface, boasting audio quality of up to 192 kHz HAMBURG, Germany — January 14, 2013 — Steinberg Media Technologies GmbH today announced the upcoming release of their latest audio interface. The UR22 features 192 kHz audio quality, USB 2.0 connectivity, MIDI input/output alongside two combo inputs with D-PRE microphone preamps and line outputs. Its portable size and solid build make it the perfect choice for mobile setups and small-spaced recording studios. “Together with Yamaha, we aimed at designing a compact audio interface with sufficient connectivity plus a build and audio quality that will simply blow you away. The UR22 is the result of this development. And by combining the UR22 with the Cubase AI music production software you get an eminently professional recording environment at an affordable price,” said Stefan Schreiber, hardware product marketing manager at Steinberg. On the front side of the UR22 interface, two Neutrik combo connectors provide microphone and line input, each with gain control and peak indicators plus an additional high-impedance switch on the second input. The panel also includes a headphones output with a dedicated volume control, a master output volume control and a mix knob for hardware-based zero-latency monitoring. The rear panel features two balanced line outputs, MIDI input/output connectors, a phantom power switch enabling +48 Volt supply on both channels and the USB 2.0 port for high-speed connectivity to PCs and Mac computers. The UR22’s full-metal chassis houses studio-grade converters, delivering pristine sonic fidelity with a maximum sample rate of 192 kHz and a 24-bit resolution. Yamaha’s discrete Class-A analog D-PRE preamps ensure unprecedented sound quality on top of the stable gain required by microphone signals. Freely available as download to all UR22 customers, Cubase AI provides a comprehensive set of tools for recording, mixing and editing audio. The UR22 is compatible with other major recording software applications and includes drivers for Windows and OS X systems. Availability and pricing The UR22 interface will be available from resellers and through the Steinberg Online Shop beginning in February. The MSRP for the UR22 is $199.99. Features at a glance 24-bit/192 kHz USB 2.0 audio interface 2 Class-A D-PRE mic preamps supporting +48V phantom power

You used to need a separate multitrack and mixdown machine, but todays's DAWs can combine both - to good advantage By Phil O'Keefe Back when I first became interested in recording, most projects required two tape machines - a multitrack deck for capturing the performances and production, and a second two track recorder for capturing the stereo mix. Even well into the digital era, a second machine has remained popular with many engineers. However, most modern DAW programs can render a stereo mix from the internal tracks and mixer. In Pro Tools, this is most commonly done with the "bounce to disk" function (File / Bounce To / Disk) You can then select the format of the bounce in terms of sample rate and bit resolution, as well as whether it will save two mono files (one for left, one for right) or a generally more manageable single stereo interleaved file. While the bounce to disk function has some advantages in terms of the file formats it outputs, it's not without its problems. First and foremost is that it is a real-time process, and the entire song has to play back, in real time, in order for Pro Tools to render and write the stereo mixdown file. If you're working on a very long song or other audio project such as a soundtrack for a short film, this can translate to a lot of waiting around every time you want to run off the latest version of the mix. Wouldn't it be nice to be able to make that "one small correction" to the mix, and punch in only that one spot to correct it? By mixing to a separate stereo mixdown track within the DAW, you can. Even better, once you've recorded the entire mix the first time, you can punch in on it at will to make any changes or adjustments, then consolidate it back to a single file and quickly export the mix file as a stereo interleaved WAV or AIF file without having to wait for the entire song to play back as you would if using the bounce to disk function. SETTING UP - THE ANALOG PATCHING APPROACH The object is to get all of the various tracks of your DAW to record to a stereo file. Often called a layback or internal bounce, the basic process is the same whether you're doing a submix of multiple background vocals to a stereo pair, or creating stereo stems for an outside engineer to mix, or bouncing down all the tracks in your session to a stereo mix .WAV or .AIF file. There are a couple of methods you can use to route the audio. One is to physically connect a pair of outputs on your interface to a pair of inputs, but this requires an audio interface with at least four outputs. If you have a 4x4 audio interface and want to try this, first, turn down your monitors, and then: Route all of your DAW software mixer channels (tracks, aux returns, virtual instrument outputs, etc. - everything you want "in the mix") to outputs 3 and 4 of your audio interface. Click on the first track name to highlight it, then click on the last one while holding down the shift key - this will select all tracks. Now while holding down Shift/Option simultaneously, assign one of the tracks to output 3/4 - doing so with the shift/option keys depressed will change the assignment for all selected tracks at once, saving you considerable time. Physically plug the hardware outputs (3 and 4) into hardware inputs 1 and 2 on your audio interface. Create a new stereo track in your DAW. Assign inputs 1 and 2 to this stereo audio track. This will be your mixdown track. On this track, make sure the output is assigned to outputs 1 and 2 (the ones you typically have your studio speakers connected to). Make sure you DON'T assign it to output 3 and 4, or you'll create a nasty feedback loop! This is why I suggest turning down the monitors, just to be safe. When you're ready to do your mixdown, record enable the new stereo track, and when you're ready, hit record. The mix will play back, and simultaneously record on to the new stereo track. Whenever you want to make a correction, there's no need to re-run the entire mix, although you can if you want. In fact, I like using new Playlists in Pro Tools for saving all the different versions of the mix I create as I go along, but if you prefer, you can "punch in" on the previously recorded stereo file - just start recording on the stereo mix track a bit before the spot where you made the mix changes, and let it run a bit past the end of the changes until you punch out. You can then add crossfades and consolidate the file so it's a single continuous region. Then you can export that as a stereo .WAV file. I'll go into the specifics of how to do that in Pro Tools in a moment. The downside to this method is that the audio is making an additional "pass" through your system's converters - both D to A on the output, then another A to D stage before it is sent to, and then recorded on the stereo track. However, if you don't have a separate mixdown deck such as an Alesis Masterlink, or analog tape deck, and you want to avoid the bounce to disk function, this can be a valid way to do it, and the degradation caused by the extra pass through the converters can be fairly slight, especially with modern equipment. Another option, and one that gets around the extra converter stages, is to use your system's digital S/PDIF input and output instead of analog I/O. The basic steps are the same as listed above, except instead of using your analog I/O, you connect your computer audio interface's S/PDIF digital output to the S/PDIF digital input, then select S/PDIF Out for all the DAW tracks used in the mix, and S/PDIF In for the stereo mixdown track's input source. IN THE BOX - NO HARDWARE PATCHING REQUIRED There is another way to set everything up that can be done entirely in software, without having to make any physical patch connections on your interface. I'll be using Pro Tools as an example to walk you through it, but the basic concepts are the same regardless of the DAW you're using, although you may need to check your manual for the specifics for some steps with other programs. If the session / song file doesn't already have one, create a master fader. Assign it to your normal stereo output (typically Analog Out 1/2). Assign all of your other mixer channel outputs to an unused bus pair. I like to use the highest bus pair available - in this case, I'm using bus 127/128, but any unused pair will work - just make sure you don't assign, or use this bus for anything else! Again, selecting all the tracks at once first, then holding down shift/option while assigning the output of one to the bus of your choice will assign all of them to that output bus and save you from having to repeat this step over and over for each track separately. Double check to make sure that all mixer channel outputs are assigned to this same bus output. Don't forget to include all effects / aux returns, virtual instrument tracks and audio tracks - if it's part of the mix that you want to hear in the final stereo mixdown .WAV file, you need to assign it to the bus. Create a new stereo audio track. To quickly create a new track in Pro Tools, press Command / Shift / N (Mac) or Control / Shift / N (PC). Assign the INPUT for this track to the same stereo bus you used in step 1. In the example here, we've used bus 127/128. Assign the output of the new stereo track you just created to your main stereo output - again, typically Analog Out 1/2, or whatever outputs are physically connected to your studio monitors. (Fig.1) Figure 1: Setting the mixer output assignments in Pro Tools. Also note the stereo mixdown track at the far right, and its I/O assignments (click on images to enlarge) Also on the stereo track you just selected, click on the Input Monitoring icon. (Fig.2) This allows you to hear the mix from the other channels as it passes into the mixdown track. Figure 2: Input monitoring allows you to hear the output of the DAW mixer, even when not actually recording to the mixdown track. Unselecting it allows you to listen back to the recorded mix itself after you've recorded it Select the entire session by using the I beam cursor to drag across the entire length of the session's waveforms and MIDI tracks in the Edit window. (Fig.3) To give myself a little working room later, I like to leave a few seconds of blank space at both the beginning and end of the session, This can be trimmed away in mastering later, and is better than accidentally cutting off something at the very beginning or end of the song. Figure 3: Select the entire length of the session by dragging across it from beginning to end with the I-beam cursor in the Edit window. When you're ready to record the mix, hit record / play (Mac: Command / Spacebar, PC: Control / Spacebar) and let it run. (Fig.4) Figure 4: When you're ready to record the mix, record enable the mix track, and hit record/play PUNCHING IN FOR CORRECTIONS, AND EXPORTING THE FINAL STEREO MIX FILE Once you've recorded the entire song's mixdown track, you can go back later and punch in on a section of the mix and correct only the small area that is problematic. Then crossfade, consolidate the waveform, then export the track. This is significantly faster than waiting for a bounce to disk. To consolidate a track that consists of several regions in Pro Tools, select all regions by double-clicking in the center of any region, then press Control / Shift / 3 (PC) or Option / Shift / 3 (Mac) on your computer keyboard. This will turn the multiple-regions into a single one, which can then be exported. When doing an internal bounce, the files you record will be at the same sample rate and bit resolution as your session, so if you're recording at 24 bit, 96kHz, you'll need to do a sample rate conversion on the resulting stereo file as part of the mastering process. Mastering engineers invariably prefer getting the highest resolution version of the mix that they possibly can, so this is ideal from their perspective. If you're doing the mastering yourself, then you will also benefit from working with high-resolution files and saving the conversion to 16 bit and the dithering until the very end of the mastering process. Mastering within the DAW itself is becoming more popular, and some DAW programs, such as Samplitude, include some pretty impressive mastering tools, and there are plenty of mastering related plugins you can run within your DAW. Since you already have the stereo file in the DAW itself, you can disable all the other tracks, and focus on the stereo track; applying any editing, mastering EQ and compression or other processing directly to it, or to a copy of it. Once you're finished, or if you want to continue the mastering process in a different program, you can export your mix. Here's the steps to do so in Pro Tools: In the Edit window, double click on the stereo mixdown file to select it. (Fig.5) If you've punched in on the track and there are multiple regions instead of just one continuous file, make sure you consolidate the regions first. Figure 5: After any punch-in corrections and consolidating, double click on the stereo mixdown file (outlined here in red) in the Edit window to select it. Make sure no other regions besides the stereo mixdown are selected If it's not already displayed, open the Clips list by clicking where indicated in Figure 6. Figure 6: Opening the Clips list can be accomplished by clicking where indicated by the red arrow Right click on the header where it says "Clips" (circled in red in Fig.6) to pull open the drop down menu. Select "Export Clips As Files." (Fig.7) Since the single stereo mixdown file is the only region / file selected, it will be the only file exported. Figure 7: Exporting the stereo mixdown clip / region as a stereo file In the Export Selected dialog box that opens up (Fig.8) you can set the various parameters for the export, including sample rate, bit depth, file type, and the destination directory (file folder) you want to save the exported mixdown file to. Figure 8: The Export Selected dialog box That's it! You'll find that exporting files in this way takes a tiny fraction of the time that doing a full bounce to disk takes. Phil O'Keefe is a multi-instrumentalist, recording engineer / producer and the Associate Editor of Harmony Central. He has engineered, produced and performed on countless recording sessions in a diverse range of styles, with artists such as Alien Ant Farm, Jules Day, Voodoo Glow Skulls, John McGill, Michael Knott and Alexa's Wish. He is a former featured monthly columnist for EQ magazine, and his articles and product reviews have also appeared in Keyboard, Electronic Musician and Guitar Player magazines.