Phil O'Keefe

Inexpensive twin ribbon mikes / Blumlein stereo kit $399.00 http://www.cascademicrophones.com By Phil O'Keefe Ribbon microphones have seen a revival of interest over the past ten years or so, and their basic sonic characteristics make them a very good transducer choice for capturing many sound sources; especially when recording to digital, where their warmth, natural sound quality and complete lack of high frequency harshness tends to complement and offset some of digital's edgier characteristics. Cascade has been getting a lot of attention for their affordably priced ribbon microphones, and they were kind enough to loan me a stereo pair of Fathead II's for review. The Fathead II is a "short" ribbon design, using a 1 3/4" x 3/16" x 2 micron ribbon mounted symmetrically within the magnets inside the large 3" diameter head, which means the front and back side response and "sound" of the microphone is pretty consistent. As with most ribbon microphones, the Fathead II has a true bi-directional or "figure 8" polar pattern, and picks up equally well from the rear as it does from the front. The good news is that the side rejection is nearly perfect -- by carefully placing the microphones so that the front and rear are aimed to pick up what you want them to hear, and the sides are aimed at whatever you don't want them to pick up, it is possible to do things such as record a solo singer / acoustic guitarist and still maintain excellent isolation of each sound source on their individual tracks. Additionally, they're perfect for Blumlein stereo pair use, and the included stereo bar makes it easy to mount them in the proper orientation. OPTIONS, ACCESSORIES AND BUILD QUALITY When I opened the shipping box, I was looking at a nice aluminum "camera" style case. Inside I found the two Fathead II ribbon microphones, two sturdy shockmounts, a pair of chamois cleaning cloths and the Blumlein stereo bar all neatly packed into cutouts in the interior foam. (Figure 1) While individual Fathead II's are available for $219, getting the pair for $399 is actually a bit less expensive, at least on a cost-per-unit basis. You don't get the oiled wood box that apparently comes with the single mic package, but you still get the other goodies, which makes these fairly well accessorized. Figure 1: Cascade's stereo kit comes in a nice aluminum case. The Fathead II is offered with three different internal transformers. The price is $399 "direct" from Cascade for the stereo set with the stock transformers. For either the Lundahl or Cinemag transformer equipped versions, the price is noticeably higher, but still quite reasonable at $699.00 direct for the stereo package. In other words, you get two high-quality ribbon mikes with world-class iron for about the same price (or less) as you'd normally expect to pay for a single USA or European built ribbon mic. While definitely more expensive, the frequency response plots for these microphones shows extended reach in both low and high frequency ranges compared to the stock transformers. However, I specifically requested the standard transformer equipped models for this review. I wanted to hear with the basic model sounded like; with the assumption that things would only be better with the upgraded transformers. These are really nice looking microphones. I was sent the version with the brown body with gold grilles, which I think looks really classy, but if you want something with a touch less flash, the microphones can be ordered with black bodies and polished nickel grilles. Just to make sure everyone's preferences are covered, they are also available with black bodies and gold grilles (Figure 2). They're larger than I thought they were going to be too; they have sufficient size and weight to them that you're given the impression of a solid, professional tool, not a cheap toy. The overall build quality looks good, with nothing appearing to be out of alignment in terms of the body, grille and case. The ribbon appears to be properly tensioned, and while no claims were made of these being a "matched pair", they have sequential serial numbers and sounded virtually identical to each other. Figure 2: Three Cascade Fathead II ribbon microphones, showing the available color options. While they are made in China, Cascade Microphones inspects the build and audio quality of every mic they ship out before it leaves their facility in Washington state. This extra level of quality control is appreciated, and means you're less likely to get a "bad" mic; which is not unheard of with some Chinese built mikes. However, the overall build and sound quality of the two microphones I was sent was quite good, especially in light of their low price tags. SOUNDING THEM OUT Is it worth the extra money for the nicer iron? That will largely hinge on whether or not the sources you're recording will benefit from the enhanced frequency response. For instruments like electric guitar amplifiers, which tend to have a rather narrow frequency response that typically sits in the 100Hz - 6kHz range, the stock Fathead II should be fine. I was surprised by how close it got to the sound of my much used and loved Beyer M160 ribbon mic on guitar cabinets. It also does exceptionally well in both mono and stereo as the rotor mike(s) for a rotating speaker cabinet (Figure 3) but again, there's not much going on in terms of frequency response from most rotary speaker systems above 7 or 8 kHz, and the stock Fathead II remains fairly flat up to that point. Figure 3: The Fathead II works very well on rotating speakers such as this Leslie 142. Acoustic guitar could go either way. The detail is there, and the note attacks are quite precise and clear, but if you're looking for top end sparkle, you may want to opt for one of the transformer upgrades. While the Fathead II "takes EQ" very well, and a slight boost with a high frequency shelving EQ generally brightened things right up, too much added EQ will tend to bring out any noise issues in your mic preamp and recording signal path in general. I liked the sound I got using the Fathead II on my Taylor 510, which can sound a little thin sometimes, and the stock transformer equipped model would also be a good choice for recording a vintage small bodied "delta blues" style acoustic guitar. It's also great for taming the sometimes abrasive top end of things like shakers and tambourine. In general, it handles percussion very well, and the snappy transient response really helps with the articulation. However, for drum overheads, I felt the stock model was a little too lacking in high frequency response. If you plan on using your Fathead II pair for drum overheads on a regular basis, you may want to consider a transformer upgrade. The highs start to fall off above around 8kHz. They're still there, but they are down several dB in the highest audible octave, which translates to a somewhat subdued sounding top. The Fathead II probably isn't going to be the first thing most engineers reach for when tracking vocals -- especially for most modern music, where a fairly bright vocal timbre seems to be in vogue. However, if you're looking for a vintage flavor - and I'm talking about some old vintage; think 1930s through the early 1950s - or if you have a particularly nasal or thin sounding singer, then you may very well like the sound of the Fathead II for vocals. As with all ribbon microphones, just make sure to use a good pop filter in front of the microphone to protect the delicate ribbon from wind blast damage. And when I say "damage", I mean "destruction." Contrary to a popular myth, it's normally not high sound pressure levels ("loud volume") that damages a ribbon, it's wind. I recommend keeping the Fatheads in their case when not in use, but if you're going to leave them up on a stand, at least cover them with the plastic bags that they came packed in to help protect them. (Edit: Cascade indicates that the Fathead II Stereo kit normally comes bundled with a drawstring cover bag for each microphone. These were apparently inadvertently omitted from the review unit packaging, and are designed for keeping the mikes covered when not in actual use.) DO YOU HAVE ENOUGH JUICE? Speaking of mic preamps, as with most ribbon mikes, the Fathead II does best when coupled with a quiet mic preamp that has lots of gain on tap. The amount of gain you'll need will depend on the nature of and volume level of the sound source, and how close to it you position your microphones to it, but in general, having 60dB of gain on tap -- or more -- is a good idea. Another alternative is to use something like a Cloud Microphones Cloudlifter CL-1 or CL-2 to add up to 25dB of gain to whatever preamp you're currently using. These are small single (CL-1) and dual channel phantom powered gain boxes that are similar in appearance to the typical direct box. Just put one in inline between the microphone and and preamp and engage the preamp or board's phantom power. The Cloudlifter uses phantom power to provide power for its internal gain stages, and blocks it from reaching the ribbon microphone. For home recordists who have limited mic collections, these are an excellent choice. Ribbon mikes definitely have a different sound than a moving coil dynamic or condenser mic, and their natural figure-8 polar pattern is perfect for use with stereo microphone techniques such as Mid-Side and Blumlein stereo pairs. The included stereo bar (Figure 4) makes it very easy to position the pair of Fathead II's so that they are properly angled and positioned as a crossed Blumlein stereo pair. It's also beefy enough that you could use it to drive nails (not recommended, but it would work); this is one heavy-duty stand, and it keeps the fairly heavy microphone pair positioned firmly and securely. Figure 4: The included stereo bar makes positioning the microphones for Blumlein stereo very easy. I was very impressed with the sound and build of the Cascade Fathead II microphones. Look, I'm not going to say that I preferred these over vintage RCA and Beyer ribbons; the frequency response isn't as good and they're not quite as detailed, but they absolutely surprised me with just how good they do sound, and at a price that is one third (or less) of what most of those other ribbon mikes cost. Even in a well-equipped studio with a nice mic locker, I could see the base model Cascade Fathead II's getting a lot of use. And for home recordists who have limited funds, small mic collections and limited experience with ribbon mikes, these are a no-brainer. If you've been recording your guitar amp with just a SM-57 or similar dynamic mic, adding a ribbon mic to your setup will give you a new range of sounds that can be a real eye opener. Are you worried about picking up too much "room" from the equally sensitive rear side of the mic? Just use some baffling behind it to reduce the amount of room sound it picks up. (Figure 5) Figure 5: A baffle behind a ribbon mic can be a useful tool for reducing the amount of room reflections that are picked up. Ribbon microphones are known for their "natural" sound quality, and the Fathead II is no exception in that regard. There is a nice "open" quality to the sound, and the attack transients are extremely detailed and crisp, with none of the lag and smear you typically get with most dynamic mikes. As a true symmetrical bi-directional microphone, the Fathead II does have significant proximity effect, and when you move to within a few inches of the sound source, there will be increased bass. This can be used to good effect to fatten up thin sound sources, and for times when you don't want it, all you have to do is move the mic back a bit or use a high pass filter at the board or in your DAW software. All in all, the entire package is a winner, and I have no problem with giving it an enthusiastic thumbs up. It would not only make a good choice for your first ribbon microphone purchase, but would be a welcome addition to any mic collection. Specifications: • Type: Ribbon (velocity) Microphone • Ribbon type & dimensions: 99\\% Pure aluminum, 2.5 micron, 1 3/4" (L) X 3/16" (W) • Polar pattern: Figure 8 symmetrical design • Sensitivity: -56 db +/- 2 dB (0 dB=1V/Pa) • Frequency response: 30 - 18,000 Hz (+/- 3dB) • Output Impedance: <=200 Ohms • Recommended load impedance: >1000 Ohms • Max. SPL (1\\% THD @1000 Hz): 165 dB • Connector: 3-Pin male XLR • Size: Diameter: 3" (grille), 1 3/4" (body); overall length 7 3/4" 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 and Guitar Player magazines.

Mic placement and technique can significantly alter the sound of these classic rotary speaker systems By Phil O'Keefe The Leslie speaker (Leslie is a registered trademark of Hammond Suzuki USA, Inc.) was invented by Don Leslie, and it made its commercial debut way back in 1941. Consisting of a cabinet containing a power amplifier, a pair of speakers and a system of motors and pulleys to "spin" acoustic baffles that redirect the sound from the two speaker elements, it adds considerable complexity, motion and interest to whatever sound is routed through the system. While the Leslie speaker is probably most closely associated with Hammond organs, it can also be used with other sound sources, such as guitars and even vocals or drums and other pre-recorded tracks via re-amping. Capturing the sound of this electro-mechanical beast requires a little knowledge about how it works, so let's dig right in… WHAT'S INSIDE THE BOX A typical Leslie cabinet, such as the model 142 shown in Figure 1, is divided into three sections. The center of the cabinet houses the crossover and two speakers - a downward firing woofer and an upward firing high frequency compression driver or "horn". The crossover splits the audio into two frequency bands, and the sound below 800 Hz is routed to the woofer, and the sound above 800 Hz is sent to a compression driver. There is a rotating horn in the upper part of the cabinet, and the high frequencies from the compression driver get sent out through it. Interestingly, while there appears to be two opposing "horns", only one of them is open to the compression driver; the second one is closed off, and serves merely as a counterweight so the horn will remain balanced as it spins around. The bottom section of the cabinet contains a large rotating drum with a curved "ramp" or scoop; this redirects the sound of the downward firing low frequency woofer and "throws" it out of the lower front, sides and rear of the cabinet. This lowest section of the Leslie cabinet also houses the 40W tube power amplifier. While some later models may have a solid state amp, or rely on an external amplifier, the classic models, when viewed from the front (as in Figure 1) have a tube amp in the lower right hand corner. The location of this amp is important to know about, since it can be a source of noise. There is also switching with most Leslie speakers; this allows the musician to go from a slow rotary speed (sometimes called "chorale") to a fast ("tremolo ") speed. The solenoid switches that control the motor speeds are located over near the power amplifier, and will produce a loud "click" or "clunk" sound whenever the speeds are changed; for this reason, I recommend avoiding that "side" of the cabinet when placing your microphones; especially the low frequency / drum mic. Figure 1: A Leslie Model 142, as seen from the front. The sound changes in very interesting and fun ways when the Leslie's speed is switched. This is in part due to the different mass of the rotary horn and drum elements; when the speed is switched, the larger and heavier drum takes significantly more time to speed up or slow down than the horn does. The interaction of the sound from the two speakers -- the phase shifts and the sound reflections off the cabinetry, Doppler shift pitch changes and amplitude (volume / tremolo) fluctuations make for a very complex and interesting sound, especially when motor speeds are changed and as the drum and horn transition to the new setting. The effect on the upper horn is more FM, or frequency modulation in nature. As the horn spins towards the listener's position, the pitch of the sound rises slightly, and as it spins away from the listener, the pitch drops slightly. The sound of the Doppler shift changes, depending on the horn's speed. The horn rotates at about 50 RPM in Chorale mode, and about 400 RPM on the Tremolo setting. The drum has slightly slower speeds. It tends to rotate at about 40 RPM in Chorale, and 340 RPM in Tremolo mode. The drum can take five or six seconds to transition speeds, and the overall effect of the sound of the spin of the drum is more AM, or amplitude modulation in nature. Due to the lower frequency range, the effect is perceived as more or a up and down volume fluctuation as the drum turns towards and then away from the listener. Of course, there are some elements of AM and FM with both drivers, and the amount of this that you capture on your recording can be affected by how closely you place the microphone(s) to the cabinet, as well as the balance of horn to drum that you use in your mix. HOW MANY MICS, AND WHERE TO PUT THEM Ask any ten engineers where to place a mic, and you're bound to get ten different answers. As with most sound sources, where you place the microphones makes a big difference in the way the final recording will sound, but there are some unique characteristics to a Leslie speaker that you need to be aware of. First of all, since a Leslie cabinet has two speakers, and they're each reproducing different parts of the frequency spectrum, ideally you're really going to need at least two microphones in order to do the job properly. What if you absolutely have no choice but to do it with a single microphone? Then I would mic from the side or rear, with the mic about six feet back from the cabinet and three feet above the floor, and aimed straight at the cabinet. Aim it or lower it further towards the drum if you want more low frequencies from the drum, and raise it or aim it more towards the horn if you want more top and brightness. You'll find that things get a lot easier when you add a second microphone. This allows you to use one to capture the horn, and the second mic to catch the drum. (Figure 2), resulting in better flexibility in terms of capturing the two. Record each mic to their own track and you will be able to adjust the balance in the mix. Figure 2: Using two microphones generally gives better results and provides more flexibility when balancing the sound at mixdown. Here a Cascade Fathead II (top) is aimed towards the horn, and an E/V RE320 (bottom) is aimed at the spinning drum. The horn at the top of the cabinet can be single miked, or for an even more dramatic effect, it can be miked in stereo. Because the high frequency range of the Leslie's rotating horn doesn't extend much past 7 kHz, there's no need to use mikes with extended high frequency response, and small diaphragm dynamic microphones or ribbon mikes are commonly used instead of condenser mikes. A few good choices include the Shure SM57, Audix i5, AKG D1000E and the Cascade Fathead II (pictured). Miking from the rear of the cabinet can result in a slightly more open and less diffused sound than miking from the front or sides and with the sound passing "though" the cabinet's louvers, but it's also possible to pick up more pulley and mechanical noises this way. Ideally, you'll need to experiment to see what sounds best to you. Sometimes you can get good results by putting the two horn mikes on each of the two opposing sides of the cabinet. Placing one mic at each of the rear corners and angling them in slightly towards the horn is another good approach. Another commonly used technique is to place one at the rear and one on one side (Figure 3). Figure 3: A pair of Cascade Fathead II's placed fairly close-in; one at the rear and one on the side, capturing the Leslie horn in stereo. Note the round black metal pop filter placed in front of the rear mic (left) to help protect it from wind gusts. The lower rotating drum is normally only captured with a single microphone. I prefer a large diaphragm dynamic for this. Some good candidates would be the Sennheiser MD421, Electro-Voice RE20 and RE320 (pictured in Figure 4), and the Audio Technica ATM250. Even when using fairly distant placement for the high frequency horn, some engineers will keep the low frequency drum mic placed in fairly close; if you try this, you may want to use a high pass filter set somewhere around 800 Hz for the horn mikes to reduce any low frequency bleed from the drum that they picked up due to their more distant placement. Figure 4: An Electro-Voice RE320 is aimed at the low frequency drum in this shot, which shows the mike arrangement used in the "close miked" audio clips. Note the use of Auralex Aural Xpanders behind the ribbon mikes to help reduce the amount of "room" picked up by the rear of these bi-directional mikes. THINGS TO BE AWARE OF Be aware of the fact that the power amplifier on many Leslie speakers is located in the lower left side of the enclosure (when viewed from the back side), and it's notorious for making noise. Not only electronic interference and hum, but also mechanical noises, such as "clunks and clicks" when the speed switch is triggered by the player. By positioning the microphone on the opposite side of the drum, you can minimize the amount of noise from the amplifier and switching that the microphone captures. Another concern with the drum is the large amount of wind noise it can generate. To help reduce the sound of wind "blowing into the microphone", I always use a foam windscreen over the drum mic. If it's really bad, you can also try inserting a vocal "pop filter" between the mic and the speaker to further reduce wind gusts that blow directly into the mic, but usually placement alone will make a big difference in the amount of wind noise you get. Speaking of wind, the one concern with ribbon mikes is that areas close to the cabinet openings and vents are prone to wind gusts from the spinning horn and drum, and a wind blast can destroy a fragile ribbon element. If you plan on using ribbon mikes, make sure you place them back a bit so they're not getting blasted with air, or use a suitable pop filter to protect your mic from wind damage. While minimizing the mechanical pops, clicks, hum and wind noise is generally something I try to do, I've given up on trying to eliminate it completely. First of all, it's pretty much impossible to do, and secondly, as long as it's minimized, it generally isn't going to be too obtrusive -- in fact, I'm convinced that the absence of some of that noise is what some listeners subconsciously listen for to determine whether they're hearing a "real" Leslie or a good simulation. Have a listen to the attached clips - I specifically waited until chords were ringing out and sustaining and fading away before switching the Leslie speeds; this was done to let you hear how much "click" we get with the microphones positioned as indicated in the photos by doing the switching in a quieter, more "exposed" part. I also left in some small sections where I'm not playing so you can listen to the wind and mechanical noise. Moving the microphones further away from the cabinet (Figure 5) generally gives a smoother sound with more "room" ambience and less dramatic volume fluctuations in the recorded sound, while close miking accentuates the tremolo effect; especially at high speed. While the microphones have all been moved back about 3 feet, even more distant placements can often work too. Some engineers like to keep the drum mike in close and just move the horn mikes further back; if you do so, don't forget to nudge the tracks back into phase and time alignment if needed. Figure 5: The mikes have been moved back about three feet in these pictures and for the "distant miked" clips. Don't be afraid to experiment with even more distant placement of the horn mikes. AUDIO FILES The recordings are of a simple classic Leslie guitar riff. It is played twice at slow rotor speed, then as the last note rings out, the Leslie speed is switched to fast (listen for the clicks), and once the Leslie has ramped up to full speed, the riff is played twice more, then as the last note rings out, the speed switch is again triggered and the Leslie speed ramps down. The signal path was kept straightforward - a Fender Duo Sonic II into a Speakeasy preamp and straight into a Leslie 142, with two Cascade Fathead II ribbon mikes for the horn, and an Electro-Voice RE320 on the drum. API 312 mic preamps were used on all three mikes, with no additional processing or filtering of any kind added… but feel free to experiment with high and low pass filters. The attached audio flies are available in two forms -- multitrack and stereo. If you just want to listen to the close and distant mic recordings as pictured in Figure 4 (close) and Figure 5 (distant), the stereo files should be all you need. If you'd like to dig a little deeper, download the multitrack versions. I've kept the clips very short so the downloads should not take too long. Once you've unzipped the multitrack files, drag or import all three files to "0" on your DAW's timeline, with each file on its own separate track. Then experiment. Try soloing just the low frequency drum track and listen to it. Notice the difference in ramp up and down speeds compared to when you solo one of the horn tracks. Also note the sound of the different frequency ranges that each speaker / driver covers. There's plenty to experiment with in terms of the high frequency horn tracks too. Start by muting off one of the horn tracks and listen to just a single horn mic paired up with the drum mike. Then try unmuting both horn tracks and panning them hard left and right, or as wide or as narrow as you prefer, with the drum track panned to the center. Notice how "bandpassed" the horn mike(s) sound when soloed - there's nothing much there in the ultra high frequencies (above 6 kHz), and nothing below 800 Hz. Of course, you can also create a second trio of tracks and drag the second (distant) mic placement version of the tracks into the same DAW "song" or "session"; this will allow you to quickly compare the sound of the close and slightly more distant mic placements -- collectively (use a mix group for each set of three tracks to solo or mute them collectively) and by soloing individual mikes (just disable your groups).

Multiple ways of creating more "space" at the start of a session to allow adding new material to the beginning of a per-existing recording By Phil O'Keefe Here's a question that comes up often enough on the forums and in email inquiries that I think it deserves a quick how-to article. Suppose you've created a session in Pro Tools, and started recording right at "0" (or 1| 1| 000) on the timeline, but later you decide the song would sound great with a new introduction added to it. But since you started at 0 on the timeline, there isn't any physical "space" to insert the new intro into. How do you "add space" to the beginning of the session? OPTIONS EVERYWHERE There's frequently more than one way to accomplish a task with a DAW program, and this one is no exception. If you're in a hurry, you can go to the Setup menu, select Session, and change the session's start time in the Session Start box. But if you're relatively new to Pro Tools, an alternative approach may be a better idea, because it's going to give you an opportunity to become familiar with some tools you'll be using on a regular basis. This article will walk you through those steps and introduce you to a few of those tools and time-saving keyboard shortcuts. OPEN YOUR SONG IN PRO TOOLS, THEN... Pull up the Edit screen. That's the one with the waveform displays. If you're on the Mix (mixer) window, you can toggle over to the Edit window by pressing Control / +. In other words, press and hold the Control key down on your keyboard, then while holding that, press the "+ ("=") key, located right below your F (function) keys. For you Mac users, the keyboard shortcut is Command / + (Command key, plus the + key). Each time you do, the display will toggle between the two main screens in Pro Tools. It will NOT work if you try to use the "+" key on your calculator keypad on the right side of your keyboard. Once you have the Edit screen displayed, you need to get your Groups displayed. They may be visible already, but just in case they aren't, have a look at the screenshot in Figure 1. The oval in the lower left hand corner is where you need to click to display (or hide) the Track list and the Groups. Try clicking on that spot a few times and you'll see what I mean. Figure 1: Click where indicated by the red arrow to reveal the Track and Groups Lists. (Click on images to enlarge) Once you have clicked there, you should be able to see the Track list as well as the Groups. I've used an arrow to show the location of the Groups. "All" is a default Group category, and all you need to do to select it is to click on it. Once you do, it will be highlighted, any edits or changes will affect ALL tracks, not just individual ones. Figure 2: Selecting the "All" group will enable you to edit all tracks simultaneously. A quick keyboard shortcut for enabling or disabling the Groups (so that you can edit a single track within the group without affecting others, or so you can go back to editing all items in the group together) is Control / Shift / G. Hold down the Control key, and the Shift key and the G key all together. Mac users, use Command / Shift / G. If you add a new key to the combination one at a time - hold the control key down, add shift and hold both, then add the G key) it works fine, or you can depress all three simultaneously - just as long as all three get held down together for a moment, it will work. Each time you repeat that keyboard shortcut, it will toggle between groups active, and groups disabled (greyed out). To create a new group, use Control / G. If you're a Mac user, hitting Command / G will do the same thing for you. Groups are a fantastic way of editing multiple tracks, such as a multitrack drum kit recording or multiple tracks of background vocals simultaneously. Get into the habit of using them, and you'll save a ton of time when working in Pro Tools. The next thing we're going to want to do is put Pro Tools into Grid mode so we can move the tracks accurately. To put Pro Tools into Grid mode (where all movement of tracks and events is done based on the Edit Grid), the easiest way, and the one I use regularly, is a keyboard shortcut; Just press the F4 key. Want to go back to "Slip" mode in a hurry? Just hit the F2 key. Alternatively, you can use your mouse and click on the Grid box -- which I have indicated with the red arrow on the LEFT side in the following image (Figure 3): Figure 3: The arrow on the left indicates the program is in Grid Mode, while the one on the right shows the location of the Grid Value ("grid size") box. See the red arrow on the right side of the image in Figure 3? That's the box for selecting the Grid size. In the picture, it is set for a quarter note. By clicking anywhere to the right of the word "Grid" within that indicated box, you will see a drop-down menu that will allow you to change the subdivisions of the Grid. Unfortunately, my screen image capture software doesn't want to catch the drop-down menu, but give it a try and you'll see what I mean. ACTUALLY MOVING THE TRACKS Now that you have the program showing the Edit window, and have put it into Grid mode and selected "All" in Groups, we're almost ready to move the tracks over and create some "room" at the beginning for our added intro. All we need to do is make sure that our Nudge amount is set to a one-bar increment. Changing this setting is very similar to changing your Grid size, and it is located immediately below the Grid size box -- I've indicated it with a red arrow in the following image (Figure 4): Figure 4: The Nudge value box (indicated with the red arrow) is set for a one-bar nudge. The maximum Nudge amount is one bar, which is what I've selected. Again, double-check to insure that you have "All" selected in Groups (lower left hand corner), and that you have Grid Mode selected (F4 / upper left hand corner of the screen). Now use your mouse and click directly on the waveform display - anywhere to the RIGHT of the longest track. While holding the left mouse button down, drag the mouse to the left, over to "0" on the timeline. If done correctly, you should see that you've highlighted the entire range of all of the tracks. Now we're ready to move them. There are two ways you can do this. You can simply hit the "+" key on your keyboard's numeric keypad (do NOT try to use the "+" key below the F / Function keys on your keyboard -- it won't work). Since we have changed the Nudge amount to one-bar, every time you press on that "+" key, the tracks will be nudged or moved one bar later in time, and because we've selected "All" in Groups, all of the tracks are nudged or moved simultaneously. Want to move them back the other direction? Just use the "-" key on your numeric keypad. Here's a screenshot of the same set of tracks after I've nudged them 12 bars later in time (Figure 5). The red ovals indicate the new start time of the nudged tracks (they now start at bar 12) and the Nudge amount (1 bar). The red arrow shows the direction they were nudged, and the 12 bars of new "space" at the beginning of the session that you can use to record your new intro. If you need more than 12 bars, you can obviously add more. Figure 5: The same set of tracks after being nudged 12 bars later in time in order to create space for additional intro material. Remember when I said there was a couple of ways you could move the tracks over? Instead of using Nudge, you can just put the session into Grid mode, select "All" in Groups, and then select all the files by dragging over them from right to left as we did with the previous example. Then press F8 to select the "Grabber" tool (the one shaped like a hand), and then click on the waveform display and merely drag everything over. When you do, they'll "snap" to the grid, at whatever resolution you have the Grid set for. To get back to your normal "I Beam" selection tool, just press F7. FUNCTION KEYS Oh, one more thing: Those Function keys on your keyboard are a real time-saver. Instead of using the mouse to switch between different editing tools, you can use the Function keys. F5 = Zoomer ("magnifying glass") F6 = Trimmer F7 = Selector ("I Beam") F8 = Grabber (or "hand") F9 = Scrubber F10 = Pencil tool Also, if you click on any of these more than once, you will cycle through the various alternative states / modes of operation for that particular tool. And by clicking on any two of the following - F6, F7 or F8 - for example, depress F7 and F8 simultaneously, you select the "magic" tool. It gives you the functions of all three of those tools (Trimmer / Selector / Grabber), depending on what part of the region (top half, bottom half, end of a region, or region crossfade points) you have the cursor sitting over. That's my favorite tool mode in Pro Tools - I can do a ton without having to change anything... but even if I do need to change tools, since I have all those Function keys memorized, it's automatic for my left hand to just reach up and hit whatever mode I need in order to do something different, such as switch to the pencil tool to fix a click or glitch in a wave file. And that's the key to speed with any program -- getting to the point where you hit the keyboard shortcuts without having to think about them. It takes time, but hopefully this article will show you the benefits and get you started.

Short scale bass with good sound, great playability and good looks at an insanely low price $269.99 MSRP, $169.99 "street" http://www.squierguitars.com By Phil O'Keefe Fender, the inventor of the first commercially successful electric bass (and Squier's parent company), has been making short scale bass guitars since 1966, and Squier has released a few different models over the years as well. Short scale instruments are often considered "student models", although they have also been used to good effect by many professional musicians over the years. Currently there are four Squier "Jaguar Bass" models, but only one that is a short scale bass. This model is officially called the "Squier Vintage Modified Jaguar® Bass Special SS (Short Scale)" (Figure 1), which is quite a mouthful, so we'll just refer to it as a Jaguar for the sake of brevity. Long favored not only by students and younger players but also by adults with smaller hands, short scale bass models usually have a 30" scale as opposed to the standard 34" bass scale length; the shorter distances between frets means less finger stretching, and this makes short scale models much easier to play. Figure 1: The Squier Vintage Modified Jaguar Bass® Special SS (Short Scale) PICKUPS AND ELECTRONICS The Jaguar's passive electronics are fairly straightforward. A split coil Precision Bass pickup is located in the middle position, and a Jazz bass single coil pickup is located near the bridge. Each pickup is connected to its own 250K Alpha mini volume potentiometer, while a third 250K pot and a .050 microfarad capacitor function as a master tone control. (Figure 2) The control and pickup cavities are painted with conductive paint for shielding. Hum is not an issue with the P-Bass pickup, although the J-Bass pickup of the review unit does suffer from a bit of noise. Curiously, it is also noticeably weaker than the P-Bass pickup, measuring 4.65K compared to the P-Bass pickup's 6.54K. The lower output and close proximity to the bridge give the J-Bass pickup alone a bright and thin sonic character. Fortunately, the P-Bass pickup is a real star. It sounds throaty and full, with great midrange note definition and punch, and not a trace of mud. It also has much better note definition and resonance on the low E string--a weak area for many 30" scale basses--than any other short scale model I have tried. I can foresee even the folks who love to modify their instruments leaving this stock pickup alone--it sounds really good. The two pickups also sound good when running together, but the J-Bass unit just can't measure up to the sound of the P-Bass pickup when running solo, so if you rely on the J-Bass pickup as part of your sound, you will probably want to budget for a pickup upgrade. It doesn't sound bad, but it is pretty weak. Figure 2: The Jaguar Bass uses 250K Alpha pots for the two volume controls and one master tone control FIT AND FINISH The Jaguar Bass (Short Scale) comes in three color combinations: candy apple red, silver, or black. Regardless of the color of the body, all three have a black headstock, black pickguard and rosewood fingerboard. Cosmetically, the satin finish on the maple neck isn't too bleached looking, but if you prefer a highly yellowed vintage neck tint, you may find the shade a little light. The neck inlays appear to be white plastic dots inlayed cleanly into the rosewood fingerboard. There are also white plastic side inlays to further help you find your way around. Generally, this bass looks really nice; even classy. But as you start doing a detailed inspection, little things show up. There is a "scuffed" spot in the satin finish at the very end of the peghead on the review unit. A small unfilled gap in the "skunk stripe" on the back of the neck. A small burr in the plastic near one of the nut slots. Fortunately, that never affected tuning, and none of these very minor blemishes are particularly egregious on an instrument at this price point, but they are part of the reason Squier can sell them so affordably; they're not using their best craftspeople to build them, nor are they using the finest materials and spending hours and hours sweating over the tiniest details of each individual instrument. And to be fair, these are really minor blemishes that in no way affect the playability of the bass. PLAYTIME Straight out of the box, the action was well-set; low enough to be comfortable and fast, but with none of the buzzing or fretting out that results from it being set too low. The tuners and top loading bridge are solid and tuning stability is very good. The truss rod was perfectly set and the neck was straight and true. The medium jumbo frets are really well installed. They are nicely crowned and have no sharp edges.The intonation was reasonably close too. Close enough that most novice players would probably never notice it was a touch out on two strings. The four saddle bridge can be individually adjusted for height and intonation, and along with the placement of the truss rod adjustment at the headstock, this makes fine tuning the "setup" (the action and intonation) a breeze. A pair of Allen wrenches for truss rod and bridge saddle height adjustment are the only accessories that are included with the bass. New and novice players are not likely to know how to adjust the action, truss rod and intonation, and they'll have to figure that out on their own. Unfortunately, there is no manual or instruction sheet included to help walk them through it, or to advise them on the care and maintenance of their new instrument. The smooth, fast and comfortable neck is a joy to play. The neck's profile is somewhat reminiscent of a Jazz Bass or full sized Jaguar Bass in that it is very narrow at the nut (1.5" wide) and gradually gets wider as you move further up the neck towards the body of the bass--although it never gets quite as wide in the upper registers as a Jazz Bass, and because of the much shorter scale length, things are much closer together and easier to reach, even for those of us with smaller hands. The offset waist Jaguar body makes the bass very comfortable to play, whether you are seated or standing and using a strap. A forearm contour for the player's right arm, and a rear "tummy contour" further increase playing comfort. The balance of the bass is pretty good, which isn't always the case with short scale models. When hanging from a strap, it doesn't really want to "neck dive" so much as "neck level" - it seems to want to hang with the neck parallel to the floor. PUT A BOW ON IT It plays really nice, and sounds pretty darned good. And nit-picky little niggles aside, it's a good looking bass. I really like the matching headstock on the black review unit. About the only thing I think that would "add" to the look would be a tortoise shell pickguard, similar to what Squier uses on some of the other Jaguar Bass models. If you wanted one, you could always have one custom cut from one of the various aftermarket parts companies. Speaking of customizing, this bass would be a good platform for modifications. Different knobs, pickups, a custom pickguard, strap retainers / locks - the Vintage Modified series Squier models are affordable enough that you can buy in at a low price, and modify and upgrade the instrument to suit your tastes and preferences, and still do it all on a very modest budget. It would make a great gift for a older child or teenager who wanted to learn how to play. As a student instrument, it's less intimidating than a full scale bass, and affordable enough that mom and dad are not taking a huge financial risk if junior's musical interests wane. Equally important, it's still nice enough that junior is actually going to be inspired to play it and want to practice. It is also of sufficient quality that he won't be handicapped in his efforts by a poor quality instrument that is difficult to play or that he'll quickly outpace. But again, I see this bass having broader appeal than just to novice players and students, and while the materials and workmanship are obviously not as nice as on a higher-end model, at this price point--an almost unbelievably low $170 "street" price; roughly the price of a decent stompbox--it's a very nice sounding and playing bass that guitarists, home recordists and those who just enjoy the fun of playing a nice short scale bass are going to dig. Even though I already have two other basses, I am more than a little tempted to purchase the review unit. It's just a really fun instrument to play. Chances are if you get one, you won't want to let it go either. Bravo Squier, Bravo! Specifications: Country of Origin: Indonesia Scale length: 30" Neck: Maple, C-shape Fingerboard: Rosewood Frets: 20, Medium jumbo Body wood: Agathis Finish: High gloss polyurethane on body and headstock, satin polyurethane on neck. Neck width at nut: 1.5" Pickups: One Jazz Bass single coil (bridge) and one split single coil Precision Bass (middle) pickup. Controls: One volume control per pickup, master tone control. Available colors: Candy Apple Red, Silver or Black body; all feature a gloss black headstock and black pickguard. Strings: Nickel-plated steel roundwound; .045 / .065 / .085 / .105 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.

Guitar compressor / gate pedal with studio-grade control capabilities $219.95 "street" http://www.mieffects.com By Phil O'Keefe There has been two main "camps" in terms of guitar compressor pedals, with each tracing their roots back to either the orange squeezer type of circuit and sound, or the grey box Ross type of sound. Both units offered minimal controls - just a knob or two at most. While some modern compressor variations have elaborated on the one or two knob control layout, few have taken things to the degree that Australia's MI Effects have with their Compressor--one of the most fully featured compressor pedals on the market. Figure 1: The MI Effects Compressor QUICK TOUR Housed in a 4.6" X 3.6" X 2.2" die cast metal box, and featuring true bypass switching, the light lime green powder coating with burnt orange silkscreened lettering on the Compressor immediately caught my eye. Pictures do not do it justice. It's certainly "different" - whether it's bold and daring, or slightly queasy and unsettling is a matter of opinion, but on a pedal with this many controls, I at least appreciated its legibility; it's really easy to see what control you're adjusting. The knob spacing is also set up for ease of use, with plenty of room to get your fingers in between the controls. And that was the second thing I noticed (after the color) - there are a lot of knobs on this compressor pedal; six in total, plus one internal trim pot--which is an unusually high amount of controls for a pedal compressor. According to MI Effects, they started with a full wave, transistor rectified control signal, which is similar in principle to the old Ross, and then added studio type features that generally aren't found on Ross style compressors--or for that matter, on most compressor pedals of any type. These types of controls are far more commonly found on compressors that are designed for studio use, where flexibility and versatility are in high demand. Input Level control with clipping LED Interactive Attack and Release controls Onboard noise Gate with level control and internal release trimpot For power, the Compressor offers the choice of using the 2.1mm ("barrel style") negative center pin 9V DC jack or 9V battery power. When using an adapter, you can hit it with up to 25V DC without harming the pedal. I didn't have a 25V DC adapter handy, but I did have a go at using an 18V DC power supply, and as I expected, the headroom and "openness" of the pedal improved, and the output increased noticably. While the large Volume and Sustain knobs feel beefy and firm, there's a bit of wobble to the four smaller knobs. Their location is about as far away from the Compressor's main bypass footswitch as you could possibly get, but they're still somewhat exposed, so if your clumsy friend with the big boots and "bad footswitch aim" wants to borrow your Compressor, you might want to say no; lest he return it with a snapped knob or two. The good news is that MI Effects has opted to use sealed potentiometers, so the controls should stay clean and crackle-free for a long time. The parts and construction appear to be of very high quality, in keeping with MI Effects reputation for first-rate pedal builds; high tolerance (better than 2.5\%) polypropylene and polyester capacitors, metal film resistors and beefy PCB construction are all in evidence, and it doesn't look like any corners were cut to try to save a few bucks. OPERATION The MI Effects Compressor should be "dialed in" for each guitar you use with it, which means an extra knob to adjust if you switch between guitars on stage a lot. The idea is to set the sensitivity of the pedal to best match the output level of the pickups. The process is similar to "gain staging" studio processors. You adjust the input Level control until the red Overload LED only flashes on the loudest peaks. The LED actually is calibrated conservatively, and lights up well before the onset of audible distortion, so occasional flashes of red as you play is acceptable. With some single coil equipped guitars, I was unable to get the LED to flash at all, even with the control set all the way up, but when I switched to a guitar with much hotter output humbuckers, I had to turn it down almost half way for best results, so there should be plenty of range available, no matter what kind of pickups you use. Once you have the input dialed in, you can set the output level with the large Volume knob. Setting it for unity gain--so it's equal in volume with the bypassed signal--is the usual approach, but if you like to use your compressor as a solo boost, there's enough level on hand for a pretty sizable volume increase. ATTACK, RELEASE AND SUSTAIN Studio engineers and recording enthusiasts will be familiar with attack and release controls. Attack allows you to adjust how much of the note attack--the very beginning transient of the sound--will pass through unaffected before the compressor circuit starts compressing the signal. As you turn the Compressor's attack knob further clockwise, the compressor allows less and less of the note attack to pass through un-modified; the compression "kicks in" faster, and tightens up the note attacks more and more. MI Effects recommends starting with the fastest setting and adjusting it to taste from there, and for general use, the fastest setting provided smooth and even results. However, if you want a little more "cluck to your pluck", you can roll off the Attack a bit and get some great Country approved "chicken picking" tones. The Release knob controls how fast the compressor "lets go" of the signal. The higher it is set, the longer the compression is maintained, and the slower the compressor's release. A faster release time is generally better for fast playing, and a slower release can help with long, held-out chords, although in real-world use, I was able to leave the Release knob near the middle of its range and still get good general purpose results. By adjusting the attack and release, a wide variety of different compression effects can be achieved - everything from slower attack, fast release poppy chicken pickin' to fast attack and slower release settings for maximum sustain on long notes and chords. The Attack and Release controls tend to interact with the Sustain control in a fairly interactive way, so you will have to play with them a bit to get a feel for how they affect each other. Speaking of the Sustain control, I found it helpful to think of it as kind of a "compression amount" control. It seems to be somewhat similar in function to a combined ratio and "threshold" knob; the amount of compression increases and the onset of compression happens sooner with higher settings on the Sustain knob. MI Effects recommends using only as much Sustain as you need, and no more. For general level control and dynamics taming, you only need to turn it up about 1/3rd of the way. At around noon on the Sustain knob, Nashville comes a calling. Crank it up, and it sustains for a long time; perfect for flowing, sustain heavy leads and big power ballad power chords. However, as you increase the amount of Sustain, the tone will sound more and more "processed." Rather than viewing this as a negative, I felt it was a great point in the MI Effects Compressor's favor. If you want subtle and transparent compression, this pedal will definitely give it to you, but it will also give you mangled and squashed too, which is exactly as it should be with a compressor that is designed for maximum versatility. The MI Effects Compressor doesn't force its sound on you--it allows you to find your own sound. GATE Compressors, by their very nature, tame your dynamics; lowering the level of loud peaks while bringing up the relative level of softer signals--including noise. The more compression (by way of the Sustain knob) you add, and the more make-up gain (Volume) you use, the worse the noise level will get. This will be further amplified and accentuated if you place an overdrive, distortion or other high-gain pedal after the compressor. A compressor in front of a Dirt pedal can be loads of fun, but it will amplify any hum or noise in your signal. This is just a byproduct of compression in general, and not a design fault with the MI Effects Compressor. In order to address this issue, MI Effects has included a noise gate. This gate automatically mutes your signal once it falls below a certain minimum volume threshold, which is set by adjusting the Gate knob. I found the gate "chattered" (rapidly opened and closed) a bit as it reached the threshold and closed--especially with high amounts of Sustain, but a little adjustment of the internal gate release time trimpot helped to smooth it out. (Figure 2) It would have been nice to have this seventh control mounted externally, but to be fair, it really is generally a "set it and forget it" type of control, and most people won't need to be making constant adjustments. Figure 2: An internal trimpot (red arrow) adjusts the Gate release time As with all gates, the trick with this one is to set the threshold level low enough to allow all of your notes--even the softly played ones--to pass through the gate, but just high enough so that low-level noise signals won't trigger and open it. I must admit to not being a big fan of gating in general, but that comes from my background as a recording engineer. I prefer to capture everything, and then get rid of what I don't need later, rather than rely on a device to make those decisions for me--but in a live situation, a gate can be a real lifesaver, and when properly set, the onboard gate does a good job of keeping background hiss silenced when you're not playing. Having the Gate included is a nice bonus, but you can always turn it off if you don't need it by turning the Gate knob fully clockwise; even MI Effects recommends using the gate judiciously, and only when needed, in the pedal's manual. Speaking of the manual, it is a single sheet of roughly standard sized paper, printed front and back, and it does a very good job of explaining the pedal and its features and functions. EXQUISITE SQUEEZING As you may have guessed by now, I'm not particularly crazy about the color of the pedal. That might be due to it clashing with my Sea Foam Green guitar (Figure 3), but who cares? It's not what it looks like that matters--it's all about the sound. Figure 3: Am I really going to complain about the color of the pedal clashing with my guitar? When there's not much else to complain about, this is what you get This is a compressor connoisseurs' pedal. If you demand "turn-it-up and go" simplicity, then the MI Effects Compressor is not for you. But if you need a wide variety of different compression effects, or you want to dial in your own individual "sound", and you know how to use (or are willing to learn) the extra controls, then you're going to love this compressor. The degree of tonal flexibility far exceeds that of most ordinary compressor and sustain pedals. While it will do more subtle and restrained compression sounds too, if you're a fan "squishy" tones like I am, you'll have lots of fun. Either way, this is a compressor pedal that lets you decide how you want it to sound. About the only thing it's missing is a "blend" knob (so you can do "parallel compression"), and maybe an alternative color scheme, but you'd still be hard-pressed to find a more versatile or better sounding compressor in pedal format. It may be a bit ugly, but it sounds fantastic! Specifications: Dimensions: 3.6"W x 4.6"D x 2.2"H Weight: 1 lb. Power requirements: 9V battery or 9-25V DC adapter with 2.1mm barrel style, center negative plug. Current Draw: 5mA @ 9V, 10mA @ 18V Bypass: True bypass with heavy duty 3PDT switch and clear lens amber LED for on / off status indication Warranty: Five year limited warranty

SansAmp Character Series Bass overdrive pedal, preamp and amp sim/DI recording tool $225 MSRP, $169 "street" www.tech21nyc.com By Phil O'Keefe What have we here? The SansAmp VT Bass is part of Tech 21's amp emulating "Character Series" of pedals, and looks like your typical stompbox, but as I found out during the course of the auditioning and review process, it's actually much more - not only can it serve as a bass overdrive pedal with a wide range of classic and modern amp-like sounds, but also as a bass preamp, and as a direct recording tool and speaker emulator. Let's dive right in and have a closer look. OPEN THE BOX The VT Bass is packed in a cool "biscuit tin" metal box that includes a Tech 21 sticker, a pair of VT "sample settings" cards, and a well-written, thorough and easy-to-understand manual. The manual includes not only a full description of what each control knob does, but also some suggested applications and tips to get you started. The general visual vibe of the VT Bass (Figure 1) is very suggestive of a classic American built bass amplifier line, and gives you a good indication of the basic "character" that Tech 21 is shooting for. Housed in an all-metal 4 5/8" X 3 5/8" X 2" stomp box, the build quality of the USA made VT Bass pedal is solid and road-worthy. The controls have zero wobble and feel reassuringly solid, and are sensibly laid out--once you understand the Character knob and what it does, and the way it interacts with the other controls, it's a very intuitive pedal to dial in. Figure 1: The Tech 21 VT Bass pedal's graphics offer a strong clue as to the type of amplifier sounds it seeks to emulate I/O is on 1/4" unbalanced jacks. According to Tech 21, the input is designed to have "the same sensitivity and loading characteristics as a tube amp," and I noticed no issues with pickup "loading" when using either my Rogue VB-100's passive pickups, or the active EMG's in my Ibanez SR1100. The 1/4" out is a unbalanced 1kOhm low impedance "universal output" that can be plugged into a guitar or bass amp, other effects pedals, or even a low impedance mixing console or sound card input. The VT Bass doesn't have an XLR output, which may be an issue with some setups, but adapters and 1/4" to XLR cables are readily available to allow for direct connectivity with pro level recording systems and consoles, and the vast majority of users will have audio interfaces with line level 1/4" inputs that they can use instead. Like all Tech 21 pedals, the VT Bass features buffered bypass switching, with unity gain when bypassed. Discussions on the relative merits of true bypass and buffered output types aside, the output of the VT Bass was clean, and there was no "tone suck", even when the pedal was inline and bypassed. I think that on a pedal of this type, using a buffered output does make sense since it allows Tech 21 to tailor the output for low impedance and compatibility with a wide variety of devices - amps, mixing consoles, PA systems and so forth. Power consumption is a very low 5mA, which the manual suggests will translate to about 100 hours of battery life from a fresh alkaline 9V battery. I haven't managed to kill a battery yet, so I can attest that they last a long time. The LED indicator will visibly dim once battery voltage drops to about 6V. Of course a center negative 2.1mm ("Boss style") 9V DC adapter can be used instead if you're going to put it on your pedalboard and use it constantly, but if you'll be using it for recording the occasional bass part, it's nice to know you can just grab it, plug it in and get going quickly without the hassle of having to hunt around for a power supply - as long as you change the battery every few months or so, and remember to disconnect the input jack (to disconnect and conserve the internal battery) whenever you're not using the pedal. Battery access is tool-free, and through a heavy-duty plastic cover on the bottom of the pedal, so unless you have the pedal bolted down to your pedalboard, putting in a new battery is literally a snap. There is no battery or adapter included with the pedal, so make sure you take that into account when purchasing. BASIC CHARACTER The controls in general are very responsive and cover a wide range, so a little tweak of a knob can go a long way. The Character knob in particular makes big changes to the responsiveness and attitude of the sound. You can think of it as an "amp type" selector knob; with SVT tones in the lower third of its range, B-15 "flip top" sounds near noon, and with things getting progressively heavier and more raw as you wind through the last third of the knob's range. In most cases, I found it was best to dial up the Character knob first, then adjust the drive, level and tone knobs as needed. The Character control and how it is set "defines" the sound of the entire pedal. It not only modifies the frequency response, but it also affects the drive, distortion and attack characteristics of the emulated amp; as you turn it up, it becomes more modern and aggressive sounding, and it distorts more readily and at lower settings on the Drive knob, so you really have to work these two controls back and forth; adjusting the drive and EQ to taste after setting the Character knob for best results. DRIVE THE LEVEL The Level and Drive controls adjust the volume (overall output level) and amount of grit, grind and overdrive; again, both have a wide range and are very interactive with the Character knob - especially the Drive control. There is a large amount of dirt available from the VT Bass pedal; more than I personally will probably ever need, but if your tastes lean towards the exceptionally heavy and distorted, you should certainly audition the VT Bass for yourself--it can definitely rock. With the Character knob set fairly low, I was able to push the Drive control a bit harder without the sound becoming excessively dirty for the material we were working on. The best "clean, but with character" tones are with low to mid Character settings and relatively low Drive settings; it's easy to get good clean tones and very slight break-up sounds too, but this pedal wants to rock, and does so with gusto, which is what makes it so effective as a bass overdrive pedal in front of an amp. Just crank up the Drive knob and adjust the output Level to taste. EQ FOR THREE The three knob EQ on the VT Bass is very effective. It uses active "boost/cut" controls," as opposed to the passive ("cut only") EQ controls found on some pedals. The Low is centered at 125Hz, the Mid at 500Hz and the highs at 3.2kHz. I would have liked to have seen the midrange set just a touch higher - somewhere in the 700 to 800Hz range, where the note definition of the electric bass is typically found - but that would be untrue to the spirt of the amps it seeks to nail. In actual use, I was able to get just about any basic timbre out of the pedal that I wanted with very little effort. I could see some players wanting this pedal primarily because of its excellent EQ-shaping abilities. Between the three EQ controls, a wide range of bass tones are available; from Motown to metal, mellow-jazz fingerstyle to bright and aggressive rock plectrum tones, bright slap or deep funk; it's all in there. DIRECT IN I should point out that the VT Bass is not a digital pedal; it doesn't use "digital modeling" technology to emulate amps and speakers. The circuitry is all-analog, and frankly, I am amazed at how similar some of the sounds are to the classic amps that Tech 21 has set out to emulate. As part of the review, I put the VT Bass to use while recording so I could check out the sound of it going direct. I'm producing the new EP for Inland Empire local legends The Sedans, and both bassist Robert Gonzalez and I were impressed and pleased with the range of sounds that were available by running the VT Bass straight to Pro Tools; enough so that we wound up using it for all of the bass sounds on the project. Since the music of The Sedans covers a range of styles from Americana to Reggae, a variety of tones were called for, and due to its versatility, the VT Bass was certainly able to deliver. Since the VT Bass has a low impedance 1/4" output, I could have patched it directly into my Pro Tools HD system, but I wanted to add a bit of compression to the chain, so I ran it into a Groove Tubes Brick DI box and a FMR Audio RNLA limiting amp; everything else was handled by, and a wide range of tones achieved just by varying the controls on the VT Bass. SANSAMP - THE BUTTON Other than the bypass footswitch, the VT Bass only has one single push switch on it. It's located next to the EQ, and midway between a graphic representation of a speaker and the word "SansAmp™." This switch is a new addition with version 2 of this pedal, and when depressed, it engages the SansAmp speaker emulation circuit. On previous versions, the speaker simulation could not be disengaged. The smooth and very believable sounding speaker emulation is, according to Tech 21, designed to emulate the sound of a multi-miked 8X10" speaker cabinet. You can experiment with using or bypassing the speaker emulation, with the best setting depending on how you're using the pedal. I'd recommend starting with the emulation engaged when using the pedal for direct recording or when running it directly into a PA system, and bypassing it when using the pedal as a overdrive in front of an amp. I tried this out with my SWR LA12 (Figure 2), and the VT Bass definitely changed the sound of my little recording combo amp and gave it more of a "flip top" sonic vibe. If you really want to increase the effect, you can bypass your amp's onboard preamp by plugging into the power amp input / effects loop return jack and rely on the VT Bass for all tonal and volume controls while using only the power amp section of your amplifier. You can take that concept a step further and connect the output of the VT Bass directly to the input of a rack mount power amp - there is more than enough output level available to drive one directly - and use that amp to drive a full-range speaker system. Figure 2: An Ibanez SR 1100 with active pickups and roundwound strings, and a Rogue VB-100 with passive pickups and flatwounds were used along with a SWR LA12 combo amp for part of the review testing FOR WHO? MAYBE YOU! So, who is it for? Different types of musicians will appreciate what the VT Bass has to offer. Bass players who have a modern, clean, high-powered amp rig but who would like to add a little "vintage character" to their sound at the click of a footswitch. Recording musicians who want the sound of a bass amp on their tracks and need an affordable, versatile, easy to use package that can in many cases serve double-duty as a direct box. And finally, for those who need a ultra-small and portable bass preamp; as a backup to a larger bass amp, for running direct into the PA, or to drive a power amp and speaker. The VT Bass can do a great job in each of these applications. Do I have any gripes? Well, in some cases, the direct box functions would be easier if the pedal also included an XLR output, but this would have added to the cost of the pedal, and probably increased the size of it too. There is a Deluxe version of the VT Bass ($350 retail, $269 "street") that offers that feature as well as several more (memory locations, two inputs, effects loop) if that's important to you. Most home recordists will probably have 1/4" line inputs on their audio interfaces that will work just fine with the low impedance output of the VT Bass. The only other thing I could possibly complain about is the lack of panel labeling for the SansAmp speaker emulation switch, but you'll remember soon enough that when the switch is depressed, the speaker emulation is active, and when it's released, it is turned off. But that's so minor as to be quibbling, and outside of that, I've got nothing. It's a great pedal that has a ton of versatility, and I love versatile pedals that can cover a lot of sonic turf and serve multiple purposes. A rare few can actually do multiple tasks extremely well and sound great doing them. The VT Bass belongs on that short list. It's an outstanding tool for the modern bassist. Specifications:Input: 1/4" unbalanced instrument level (-10dBm / 250mV nominal), 1MOhm impedance Output: 1/4" unbalanced, 1kOhm impedance Bypass: Buffered bypass Switching: Silent switching via custom actuator Power: 9V battery (~100 hour battery life) or optional 2.1mm center negative, regulated or unregulated 9V 100mA minimum DC adapter (Tech 21 DC2 or similar) Power consumption: 5mA Dimensions: 4 5/8" X 3 5/8" X 2" Warranty: One year limited warranty

Identify and remove obstacles that get in the way of great recordings By Phil O'Keefe Why is it so hard to get tracks that kill? Mixes that scream with emotional impact--music that holds up to the work of the masters of our craft? Experienced pro or newbie neophyte, we all share a desire to improve the sound, relevance, and "vibe" of our recordings. But sometimes the way to do this isn't just by doing the right thing, but avoiding doing the wrong thing--and that in turn will indeed make things easier. BAD GEAR Everyone's favorite whipping boy, bad gear is often the first place many of us look to and point the finger at when something about our recordings doesn't knock us out. And let's face it: First-class gear sounds great, and that can't help but make things sound better--but only if you know what you're doing with it. I've been amazed by the quality of some recordings I've heard that were done on primitive or inexpensive gear, however, that says more about the engineer than the gear. Still, it's important to scrutinize your system from time to time and probe for weak links. Did you upgrade your mixer, but not your monitor speakers? Do you have a great microphone, but are using it with an old, noisy mic preamp? Nothing works in isolation, so consider where the best improvements can be made to enhance your system's sound quality as a whole, and don't obsess on any single area (like having the best mic cabinet in the world if you don't have preamps that are equal to the task). THE CURSE OF THE ADAPTIVE EAR Even in a well-designed control room with great monitors, our ears adapt to EQ changes very quickly--that's how you can enjoy hearing your favorite song on a cheap TV speaker or a high quality system. Our ears perceive the extremes of the audible frequency spectrum differently at different playback levels, with the flattest response being at about 85dB SPL. Our ears also tire after long hours, especially at unsafe monitoring levels. That EQ tweak that sounded great last night after 10 hours of playback at 105dB might not sound so hot the next morning. Having high-quality reference material that you can A/B with your mix can help you get back to reality when EQ changes start to throw off your perspective over time, and so can watching your levels and knowing when to quit when your ears have had enough for the day. NASTY CONTROL ROOM ACOUSTICS Thankfully, more people are starting to understand the importance of room acoustics (both for mixing and tracking), so we won't belabor the point. But some people just throw up their hands and say "My room was never designed to do acoustics, so it's hopeless." No it isn't! Just a few strategically-placed bass traps or diffusors can make a huge difference, as can monitoring at lower levels if you use near-field monitors; that way, the room acoustics will have less effect on what you hear, and the accuracy of your monitoring will improve. While you can learn to adapt for the quirks in your monitoring environment to some degree, having as honest of a system as you can makes sense, so improving your room acoustics is definitely something to consider if you have not already done so. POOR TECHNIQUE There's a certain charm in just randomly placing mikes and turning knobs until you find something that pleases you, and you may be lucky and come up with a masterpiece that way, but your odds are better if you have a basic understanding of how your gear was designed and how it works, and learn how to use it to best effect - including making it do things it wasn't originally designed to do. That means spending time learning the technical aspects and common techniques of modern recording and mixing, reading gear and software manuals, and also studying the work of the masters and of your contemporaries, then experimenting and listening on your own. Like a musician's influences, the idea isn't to be a "me too" of your heroes, but to learn what you can from them and then combine those various influences and apply them in your own way to your own musical voice, with your own distinctive style. SELF-DOUBT Arrogance isn't the answer, and a little self-examination and assessment of your weaknesses is definitely a good thing now and then, but a little confidence is also important. Second-guessing everything you do and doubting your own hard-earned experience and skills can kill you; if you think every mix you touch is going to suck, they will. Start trusting your gut. If something is sounding great to you, don’t worry so much about whatever the current cool trends are, go with it. Once you can make things sound good to you, and they make everyone else in the room nod, sing along, play air guitar, tap their feet or all of the above, you might be on to something. Those are the moments we live for, and they can be more addictive than any drug. You're not going to have many of them if you constantly doubt yourself. The best engineers and producers that I know are not boastful egotists, but instead carry themselves with an air of quiet self-confidence, and most of them are happy to pass their knowledge and experience on to anyone who shares a sincere interest - which are qualities I find more than worthy of emulation. SOUND SOURCES MATTER Crummy kit + shot heads + cracked cymbals = drum tracks that suck. Figuring out the inverse equation shouldn't be too difficult. It's a lot easier to get a beautiful sounding acoustic guitar recording if you start with a amazing sounding instrument. Unless you're tracking David Lindley (who can make the cheapest guitars sound great), you'll get better results from better instruments. If the artists you work with regularly do not bring them in, then consider renting or purchasing some high-quality instruments for your studio. Your recordings will improve dramatically when you are recording great instruments instead of bargain basement models. NO SUBSTITUTE FOR PERFORMANCES I consider myself to be a pretty good editor with tape or DAW; I've been doing it for over three decades, and I've gotten a fair amount of kudos from clients over the years. But I still need "something to work with", and the best edit is a performance that doesn't need one. If you have to edit, it's a lot easier to do if you have tracks with generally solid performances with few errors and great feel. Piecing something together from sub-standard performances is not my idea of a good time, and the musicality of your work is going to be much better if the musicality of the people you’re working with is already happening. When I work with brilliant musicians, my work sounds better - and so will yours. If things are not quite “there” with the artists you are working with, take some time to do some pre-production rehearsals before you get into the studio so that you can help get things as tight as possible before you start waxing tracks. Rehearse more, edit less. NUMBER ONE, WITH A BULLET Probably the number one issue is material. A so-so recording of a great song still leaves you with a great song. A great recording of a so-so song leaves you with a so-so song. Of course, we're not in the business of making so-so recordings, and everything matters, so take a moment to evaluate the weaker areas of your whole rig - and that includes your personal skills and musicianship - and plan out a strategy for improving each of them. Your recordings and productions will only get better as a result. Do you have some other areas that you struggle with, or ideas for helping others with some of these common problem issues? Then please feel free to stop by The Studio Trenches Forum here at HC and tell us about it! 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.

Start with the basics and grow your drum miking rig over time By Phil O'Keefe Miking up a drum kit can be one of the most challenging recording tasks; in part due to the fact that the drum kit is a complete "instrument" that is actually comprised of several individual elements. Each has its own sound, and due to their close proximity, each will tend to "bleed" into the microphones used to capture other parts of the kit. Additionally, there is a dizzying array of different drum miking techniques and approaches, from the simple to the highly complex. We're going to walk through a few of them; starting with one or two microphones and going up to ten or more. We'll be working with the basic four mic setup of kick, snare and two "overheads" and adding or subtracting from that for various configurations. SOURCES MATTER While drum selection, maintenance and tuning are beyond the scope of this article, I'd be remiss if I didn't mention their importance. If you want great sounding drum recordings, it really does help if you start with a great sounding drum kit and a talented drummer to play it. It doesn't have to be the most expensive kit, but it should be in good repair and have good heads on it, and those heads should be properly tuned. If you don't know how to do this yourself, hire a knowledgable drummer or drum tech to teach you how to set up a kit, or to do it for you prior to your recording session. This will make a huge difference in the quality of your drum recordings, because what you record---the sound sources--matters a great deal. KIT SIZE Drum kit size and the setup configuration are big considerations when miking. The larger the kit, the more complex the mic setup that will be needed to be in order to "cover" it properly. While your drummer may need to have a huge rig with lots of extras for live performances, less is often more in the studio. When recording, it might be better to run a more stripped down 4 or 5 piece kit (Figure 1) for the main drum parts, and overdub the wind chimes, tambourine, wood block, roto-toms and other "kit add-on" parts on separate tracks later. This will give you more mixing options, and ease the microphone and input channel requirements while tracking. Figure 1: A basic five piece drum kit in the studio; ready to be miked up (click on images to enlarge). INPUT CHANNELS AND NUMBER OF MICROPHONES The number of input channels on their audio interfaces will be the big limitation for many people. Most inexpensive computer audio interfaces only offer two mic inputs, and for tracking drums, it's always nice to have four or more. If you will be recording the kit by itself, an eight channel interface greatly expands your miking options; allowing you to track stereo overheads, snare, kick, and three tom mikes; or two toms along with either a room, ride or hi hat mic. For tracking a band or rhythm section all at once, having at least a sixteen channel interface is really the way to go--but you can record "just drums" with much less, and expand your recording rig and capabilities as your needs and budget allow. Two channel interface / two microphones: Your options are very limited here. While it is possible to capture a great drum sound with only one or two microphones, it is very difficult and requires extra care in their placement; not to mention a drummer who knows how to "balance" the sound of their kit as they play, and an exceptional sounding room to record in. Some microphone configuration options include: One mic. Wander high and low around the room until you find the spot where it sounds the best. Put the microphone there. Try about 6' in front of the kit, at about the same height as, and aimed at the drummer's chest. An overhead placement 3-4' above the snare drum level, and aimed straight down at the kit somewhere between the snare and rack toms is another alternative. (Figure 2) Cardioid condenser microphones such as the Audio-Technica AT4041 shown in the photo are the traditional choice for overhead microphones, although ribbon mikes such as the Beyerdynamic M160 can also work well. Figure 2: Basic overhead mic placement can be as simple as one mic centered over the kit and aimed down at the snare or kick pedal. Add in a second mic (right side) above the floor tom and aimed "across" the kit at the snare and hi hat for the "Glyn Johns" stereo overhead setup. One overhead mic, and one on the kick drum is a definite option with a two channel interface; hey, it worked on the early Beatles records, right? Put the overhead mic anywhere from 3 to 4 feet or so above, and aimed down at the snare; or aim it at a point somewhere between the snare and rack toms. Put the other mic on the kick. (Figure 3) This is generally my preferred approach for a two drum mic setup, but it is really a mono configuration; you won't have stereo tom fills unless you overdub those parts later. If your kick has no "hole" in the front head, you can mic it from a few inches in front of the head for a fuller sound, or around the other side, near the drummer's foot and the kick beater for a bit more "attack". If the drum head has a hole, I like to insert the mic about three quarters of the way into it, and aim the mic roughly midway between where the beater strikes and the side of the kick drum shell. (Figure 4) Figure 3: Large diaphragm dynamic microphones are the usual choice for kick drums. Figure 4: Aiming the kick mic midway between where the beater hits (white circle) and the side of the shell gives a good balance of shell resonance and beater attack. Rotate it towards the beater for more attack, and towards the shell for more resonance. One mic on kick and one on snare is an option if your music relies primarily on those two elements of the kit, but you will usually have to sacrifice toms and cymbals; again, this could be an option if you're willing to overdub those parts later. Various stereo miking techniques are also possible with a two channel interface. Blumlein, ORTF, XY or AB / spaced stereo mic pairs can all work well to capture the sound of the kit, but each relies heavily on the quality of the kit and the room acoustics, and none of them will give you a "big" sounding kick drum without additional help. Four channel interface / four microphones: Your options increase quite a bit with a four channel interface. The most obvious configuration is a stereo pair of overhead microphones, plus a kick mic and a snare mic. With careful placement, you should be able to capture a good balance of the toms and cymbals with the overheads. Move the overhead mikes out towards the cymbals or further in towards the toms until you achieve the balance you like. The well-known British recording engineer Glyn Johns pioneered what is probably the most often discussed four-mic technique in history. Put one mic overhead, about 3-4 feet higher than the snare and aim it down between the rack toms and point it towards the kick pedal. Put the second "overhead" mic much lower - only about 6" above the top of the floor tom, and aimed across the top of it and towards the snare and hi hats. An alternative approach is to raise this "side" mic a bit higher and back behind the kit a bit more, but regardless of the exact height and location you decide on, here's the important part: Make sure that both "overhead" microphones are exactly equidistant from the center of the snare drum's top head. A measuring tape or a few feet of string or mic cable can be used to check. This puts the snare in the "center" of the stereo image, and avoids phase issues. Pan the center overhead to about 3 o'clock and the tom side to about 9 o'clock and adjust to taste from there. Figure 5: The traditional approach to mic position on snare drum is from the top, just past the rim and angled downward. The snare drum and kick are also miked. We've already discussed where to place the kick mic. For the snare, you have a couple of options. Over the top edge of the snare drum and angled down (Figure 5) is the traditional approach, and this will give you solid attack, although it may be a little light on the rattle of the actual snares. I actually prefer to mic the side of the snare shell (Figures 6 and 7); this tends to give a good balance of note attack, shell resonance and "body" as well as the rattle of the snares, and it does it all without resorting to tricks such as separate top and bottom snare mikes. Avoid aiming the snare mic at a vent hole on the drum, or you'll get a big blast of air hitting the mic whenever the drummer hits the snare. Remember to record each of the mikes on separate tracks in your DAW software. This will allow you to adjust their relative levels when you mix, as well as to EQ and compress them differently. Figures 6 and 7: Miking the side of the snare shell can offer a good balanced snare sound. The curved metal piece behind the snare mic is a SE Mini Reflection Filter, which is being used to help reduce the amount of hi hat bleed that reaches the back side of the snare mic. You have a couple of options with a four input interface. A Blumlein stereo pair instead of the traditional overheads can give you great stereo imaging, and sound wonderful when placed just behind and above the drummer's head; aimed forward and pointing slightly down towards the center of the kit, they give a great "as the drummer hears it" sound to the recording. Adding a kick and snare mic to this setup makes for a nice optional four mic configuration when you're working in a good sounding drum room. If you have a small submixer, you can pre-mix two or three tom microphones and a pair of stereo overhead mikes and send the pre-mixed blend of toms and overheads to two line input channels of your audio interface and record the premixed toms and cymbals to a stereo pair of tracks, while recording the kick and snare to their own individual tracks. Six channel interface / six microphones: Stereo pair plus kick and snare, add two microphones as needed from there. If you need more toms on your recording, you can use those two extra microphones as spot mikes on a four piece drum kit's two toms. When recording a larger kit, you can aim one mic between "pairs" of toms; allowing you to cover up to four toms with the two additional mikes. Again, if you have access to a small mixer such as those made by Mackie or Alesis, you can use it to submix multiple tom mikes to two channels. If your song puts a heavy emphasis on the hi hat, you may decide to use an extra mic there, and then put the sixth channel to use for a figure-8 or omnidirectional "room" microphone. Eight channel interface / eight microphones: An eight channel interface offers you enough inputs to fully cover a five piece drum kit, with kick, snare and stereo overheads, plus three tom mikes and your choice of a hi hat or room mic. If you're recording jazz, you may instead want to use that last mic as a spot mic for the ride cymbal. Figure 8 shows the kit with tom mikes added, and the "side" overhead pulled back and up a bit, but keeping it the same distance from the center of the snare as the other overhead mic. Figure 8: Adding in three tom mikes and moving the "floor tom side" overhead up and back a bit. Figure 9: Close up view of the rack tom mikes. Figure 10: Close up view of the floor tom mic. Ten (or more) channel interface and ten (or more) microphones: The options are wide open. In addition to the basic five piece configuration outlined for an eight channel interface, you can have both the hi hat mic and a stereo pair of room mikes, or a mono room mic and a spot mic on the ride cymbal, or a second kick or snare mic. In the next photo (Figure 11), I have a modified Glyn Johns overhead mic setup, two mikes on the kick, one on the side of the snare shell, one on each of the toms, and a figure-8 microphone behind the drummer's head, with the two lobes of the figure-8 mic pointed to the "sides" of the room in order to pick up room reflections instead of the direct sound from the kit. Figure 12 shows a close up view of the second kick drum microphone - a Yamaha Subkick Figure 11: A full ten mic setup including two overhead microphones, two mikes on kick, one on each tom and the snare, plus the hi hats and a "room" mic. Figure 12: A second kick mic such as this Yamaha Subkick "speaker microphone" can help capture extra low frequency wallop. Always check it to make sure it's in phase with your main kick mic. THE EXPANDING DRUM MIC COLLECTION If your budget is limited and you can't afford to run out and purchase a new multi-input audio interface and a ton of drum mikes, don't despair. There is no reason why you can't start with what you have available, and augment it later. It's probably best to invest in a good multi-channel interface first, and then borrow or buy mikes to use with it as you can afford to. Ideally, you'll want to have a variety of microphone types for use on drum kits: Kick: Large diaphragm dynamic mikes such as the AKG D112, Audio-Technica ATM250, Shure Beta 52 and Electro-Voice RE20 are the preferred choices here. A large diaphragm condenser or Yamaha Subkick "speaker microphone" as an optional second microphone is a welcome addition, and is helpful for capturing extra bottom end beef. Record each kick mic to a separate track and blend them to taste later. Check phase! Snare: The usual choice is a small diaphragm dynamic mic such as the Shure SM57 or Audix i5. Condenser mikes with the pad switch engaged can also work well. Rack toms: Large or small diaphragm dynamic mikes such as the Sennheiser MD421 and Audix D2. As with snare, condensers are another option here. Floor tom: Large diaphragm dynamic mikes such as the Audix D4, Sennheiser MD421, and Audio-Technica ATM250, or a large diaphragm condenser. Overheads: Large or small diaphragm condenser microphones, or ribbon microphones will all work well. Audio-Technica AT4041, AKG C-414, MXL 603, Beyer M160, etc. Hi hats: Small diaphragm condensers like the SE Electronics SE2-A and AKG C-451E are traditional, although a ribbon mic can help "mellow" the sound if it is overly bright and harsh. Room microphones: Your large diaphragm vocal condenser mic can often serve double-duty in this application. Ribbon mikes also work great as room mikes. Don't forget to factor in the cost of mic stands and cables. Good stands make it much easier to position your mikes as you're setting up, and make them much less likely to slip out of position as you're recording. Also, make sure to take good care of your cables. If you lay them out neatly around the kit, they're less likely to be stepped on or tripped over. (Figure 13) Figure 13: Carefully and neatly routing your cables makes them less likely to be stepped on or tripped over. GENERAL TIPS AND GUIDELINES A quick word of advice about levels: Don't record too "hot" - leave yourself some "headroom" and avoid clipping your preamps or recordings. I like to record with my levels averaging about 15dB to 18dB below "0" on my DAW's meters. The occasional "peak" or hard hit can go higher, but the red clipping indicator should never light up on your recording tracks. Work around the drummer as much as you can. They'll be more comfortable and probably play better if you don't mess around with their setup too much. Sometimes that means taking a slightly different approach from conventional and traditional drum mic placements in order to accommodate unorthodox drum setups. Take it on the road. If your room isn't up to par, take the recording to a different "drum room" such as a local hall, church or warehouse. Laptop computers and mobile interfaces make this relatively easy to do, and if the room sounds great, it will make a significant difference in the sound of the final recording. Try setting the kit up in various different spots within the room. Some locations can yield better results than others. If you find something you like, make sure you take lots of pictures and measurements, or otherwise note exactly where you had things set up so you can duplicate the setup again at a later date. Walk around the room and listen while the drummer plays the kit. You should listen for any obvious problems that need attention (squeaky pedals, rattling hardware, out of tune drums), as well as to how well the drummer "balances" the various elements of the kit while playing. That can tell you a lot. If they hit the ride cymbal really hard, you might capture enough of it in the overheads so that you can forgo using a spot mic on the ride. The same is often true with hi hats (Figure 14); a spot mic aimed at the hi hat is always nice to have, but it is not uncommon to have "enough" hi hats from the room and overhead microphones, and from the hi hat "spill" into the snare and tom mikes. Listen for the impact and relative balance of all the different components of the kit as you move around the room. Pay attention to the ratio of direct sound verses room reflections and ambiance. Wherever you hear the best balance of all these sounds, that's where you should put your room mikes. Figure 14: A separate hi hat mic isn't always needed, but if you have enough channels available, it may come in handy. Sometimes cymbals can be a real problem in recording situations. Some drummers clobber the brass while hitting the skins relatively softly. I suspect this is due to the fact that cymbals are often not well miked in small clubs, so the drummers get used to hitting them hard so they'll project. In the studio, the opposite is often needed--less force on the brass and more solid and consistent hits on the drum heads. You can ask for the drummer to try to hit the drums harder, but they'll often revert back to their old ways. Swapping out "quieter," faster decaying cymbals can really help if you have access to them. Another option that can often help is to raise the height of the cymbals and hi hats a few inches above where the drummer normally places them. This will do two things: It moves the cymbals further away from any tom and snare microphones; thus reducing the amount they will be picked up by those mikes, and it changes the angles that the drummer hits them at. This tends to make them hit them less forcefully. Use your monitor controller or DAW's "mono" button regularly while positioning your microphones to make sure the mikes are in phase. Phase cancellation sounds "hollow" and lacking in bass when summed to mono. Inverting the phase on one mic preamp channel may help. If the drummer sets the cymbals really low and just isn't comfortable moving them higher, you might not be able to fit your tom mikes "under" them. In that case, you may have to resort to miking the tom shells or the bottom heads of the toms instead. There are no hard and fast rules for how to "mic up" anything; only what sounds good to you with that particular kit and drummer in that particular room, when it is being used on that particular track or song. Never take anything as gospel when it comes to setups-- they're all guidelines, and you should always experiment to see what works best in your particular situation. Avoid the direct center of the room if you can; acoustic phase cancellations are more likely to occur there than at any other spot within the room. Also be careful of corners; bass frequencies tend to build up in the corners of rooms, and can give your tracks an undefined and muddy character. Talk to the drummer. Chances are that they have some prior experience with recording and may be able to offer you insights about what has--and hasn't worked with their setup in the past. Talk to the drummer #2: If you value your microphones and don't want them getting smacked by a stick, don't forget to check with the drummer to make sure that where you have them placed will be "out of the way" as they play. Talk to the drummer #3: Ask them what elements of the kit they are going to play, and if there is anything they won't be using. No need to worry about how the kit mounted cowbell is going to sound on your microphones if they don't plan on playing it. Little movements can make a big difference, so before you try a completely new mic placement configuration, try making some adjustments to what you already have. For example, rotating the kick microphone an inch or two more towards where the beater strikes, or away from it and towards the drum shell can make a big difference in the amount of "attack" on each kick hit; moving the mic a few inches further into the drum or further back from the beater can also change the sound. Distance equals depth. The closer you are to the drum, the more direct sound you will get. Close mikes tend to capture a very up close and "in your face" type of sound, while microphones that are placed further back will capture more of the "room", and sound "further away" from the listener. "Bleed" (the sound from other drums besides the the individual one you're trying to capture) will also be greater the further back you pull the microphones, and "bass boost" due to proximity effect will be lessened. What the back of the mic is pointed at can be just as important as what the front of it is aimed at. Obviously you want to point the mic towards the "sweet spot" of whatever it is you're trying to record, but if you can simultaneously aim the "null point" of the microphone towards whatever you're trying to reject, that's even better. For example, if you're picking up too much of the hi hats, try to get the dead spot of the snare mic aimed towards the hats. Doing so will help reduce the amount of hi hat that your snare mic will pick up. Whenever possible, avoid having a nearby mic on the same plane as a cymbal, or so that the mic is aimed at the thin edge of it. When the cymbal is struck, you will get a very un-natural phasey sound as the cymbal pivots in and out of position relative to the mic.

These often overlooked pedals can solve problems and give added tonal versatility to your other pedals By Phil O'Keefe Recently on the Harmony Central Effects forum, someone asked why EQ pedals aren't more popular with guitarists. One person speculated that it is due to the presence of decent tone and EQ controls on many modern pedals; and while in some cases the quality of EQ controls on "dirt" pedals has increased in recent years, there are still many pedals that lack EQ entirely, or that could benefit from more EQ adjustment than the onboard tone controls provide. Modifying the sound of distortion pedals is just one area where a dedicated EQ pedal can be very useful. Let's have a look at some of the types of EQ that you may find in pedal form, and what sort of problems they can solve; as well as some creative purposes they can be used for. WHAT IS EQ? EQ, which is short for "equalization" was first used in the telecommunications industry to "equalize" the sound at all frequencies, and to compensate for high frequency signal loss over long telephone transmission lines. EQ uses filters and amplifiers that affect only portions of the audio frequency range, allowing the user to adjust the tonal balance of low, midrange and high frequencies in various ways. Eventually the audio and electric musical instrument industries adopted equalizers for similar corrective use, as well as for tone shaping purposes. With "creative" use, equalizers can take a sound and mangle the tone in all sorts of different ways, making it sound far different than it did to begin with. TYPES OF EQ PEDALS There are two basic types of EQ pedal that you're likely to encounter. The most common type is a "Graphic EQ" such as the Boss GE-7 and Danelectro Fish and Chips 7 Band EQ (Figure 1). Figure 1: The Danelectro Fish & Chips 7 band graphic EQ pedal (click on images to enlarge). A graphic EQ pedal usually has anywhere from six to ten EQ filters at different pre-set frequencies; each usually spaced one octave apart and controlled by a center-detented up / down slider. Pushing the slider up boosts or increases the level at that frequency, while moving it down "cuts" the sound at the indicated frequency. Each frequency band can be boosted or cut; typically by up to 15dB. The lowest frequency band is adjusted with the slider on the left, and highest on the right; the controls themselves visually form a "graphic" representation of the frequency range (from left to right, low to high) and show exactly what the EQ is doing to it. Typically a six band graphic EQ like the Maxon GE601 will have controls for the frequencies from 100Hz up to 3.2kHz, while a ten band unit like the MXR M-108 covers a wider range; from 31.25Hz up to 16kHz. Although less common than graphic equalizers, the second type of EQ pedal you may run across is the Parametric EQ. A true parametric EQ has a minimum of three controls: Sweepable Frequency Select Boost / Cut Q (or Bandwidth) The sweepable frequency control selects the center frequency of the EQ, while the boost / cut control adds or subtracts gain (volume level) at the selected frequency. The Q, or bandwidth control affects how much of the signal above or below that center frequency will also be affected by the boost or cut. This allows you to dial up anything from a narrow band "notch" filter (great for reducing 50/60Hz ground loop hum) all the way to a wide two octave bandwidth range, which would be better suited to overall tonal balancing, such as for cutting some bass overall, or adding in more sparkle and brightness in the treble region. KIND OF, SORT OF Despite how they're sometimes named, many "parametric" EQ pedals are more accurately called semi, quasi or pseudo parametric EQs; these pedals typically have the sweepable frequency select and cut / boost controls, but lack the flexibility that comes along with the bandwidth or "Q" control of a "true" parametric design. Most, such as the Carl Martin 3-Band Parametric Preamp and the Catalinbread VariOBoost (Figure 2), have the bandwidth pre-set at a one or two octave range, which makes them useful for general tone adjustments, but less effective as a "problem solver", or for narrow bandwidth, "resonant filter" type EQ sound effects purposes. Figure 2: The Catalinbread VariOboost is a 0-20dB boost pedal that also includes a two knob, one-band semi-parametric EQ. WHERE IN THE CHAIN First think about what you want to accomplish. Do you want to equalize the sound of the guitar that's feeding into the distortion pedal, or sculpt the sound that is coming out of it? EQ pedals can go practically anywhere in your pedal chain, but they will respond differently depending on where they are positioned relative to your other pedals. For example, if you place an EQ before a distortion pedal (Figure 3), you can control the frequencies that will be clipped. Anything you boost on the EQ pedal will receive more "grit" from the dirt pedal, and anything you cut will be less distorted. If you place an EQ after a distortion pedal (Figure 4), the EQ will act more as an overall tone control; affecting the balance of the sound overall, and having no real effect on the amount of clipping--only the relative frequency balance of the final sound. You can use this to your advantage. For example, the Electro-Harmonix Big Muff Pi is a well loved fuzz pedal with a huge low frequency response, but the midrange can be a little "soft"; an EQ pedal in front of it in the chain can be used to tame the bottom and bring the mids out a bit more so the guitar "cuts through" better onstage or in a mix. The Ibanez TS9 Tube Screamer oevrdrive has the opposite EQ sound - a boosted midrange "hump" and less in the low frequency range… try placing an EQ pedal after it in the chain and adding a bit of bass and pulling the mids back a touch to balance out the sound. Here are a few more "pre / post dirt" EQ guidelines: Too much bass "going in" can lead to a flabby, muddy sounding low end. An EQ in front of the distortion pedal with the lows pulled back a bit will help with definition and clarity. Boosting a frequency pre-distortion tends to accentuate that frequency and causes more distortion in that band. If your distortion sounds too shrill, an EQ after the dirt pedal can be used to roll off some of the high frequencies and smooth out the sound. Adding low frequency EQ after distortion is usually a better way to get "fullness" than adding it in front of a dirt box. Remember that these are just some suggested starting points. Ideally, you should experiment with placing your EQ pedal before and after any fuzz and overdrive pedals you use, and see which approach you prefer. Figure 3: Use EQ before an overdrive, distortion or fuzz ("dirt") pedal to accentuate the frequencies you want to distort, and de-emphasize the ones you don't. Figure 4: Use EQ after a "dirt" pedal to shape the overall sound. Adding bass after dirt usually works better than adding bass going into it. OTHER NIFTY EQ USES AND TRICKS Wah pedals use a relatively narrow band resonant EQ filter that is swept in frequency when you rock the pedal back and forth. You can simulate the resonant sound of a "parked" wah by boosting heavily in a narrow frequency range with a graphic or parametric EQ. Cutting some of the frequencies above and below the boosted frequency range can make the effect even more dramatic. Since a wah is essentially a foot controlled, sweepable EQ filter, don't forget to try it before and after your dirt pedals. Fuzz pedals in particular can react very differently when placed in front of, or after a wah pedal. Using a parametric EQ, dial up a narrow "notch filter" at 60Hz and cut heavily to reduce ground loop hum. Does your semi-hollowbody guitar feed back at high volumes? Use an EQ pedal to lower the level at the frequency that is causing the feedback. Are you looking for feedback in a certain frequency range or on a particular note? Skillful EQ boosting in the appropriate frequency range will often help bring out feedback. Try placing a graphic EQ in front of a phase shifter or flanger pedal. Boost or cut a band to accentuate or reduce the "swish" of the pedal in that frequency range. Some delay pedals have a built-in effect loop - by patching in an EQ pedal, you can make the delay repeats get progressively brighter or darker. Low frequencies require more power to amplify than high frequencies, so if you're trying to get the maximum "clean volume" out of your amp, try rolling off some of the low frequencies; in particular, the area below 100Hz usually adds little to the sound of the guitar, but it can be a drain on headroom. Do you ever switch back and forth between two different guitars? If one guitar has humbuckers, and the other uses single coils, it may be difficult to find an EQ setting on your amplifier that works well with both instruments. An EQ pedal can be used to compensate; just kick it on or off when you switch instruments and you won't need to be constantly adjusting your amp settings. When trying to make sonic "corrections" with a parametric EQ, boost heavily and then slowly "sweep" the frequency knob until the offensive sounding frequency really "jumps out" at you--then just cut it back until it sounds good. Many EQ pedals include an output level slider. Turn it up and use your EQ pedal as a "clean boost", or to kick your "on the edge" tube amp into full overdrive. Add in some mid and high frequency EQ boost, and your solo will cut through the densest mix. Running an EQ pedal in your amp's effects loop allows you to adjust the overall sound of the preamp right before it hits the output / power amp stage. Small all-tube practice amps such as the Fender Champion 600 or Vox AC4 sound great, but are usually very restricted in terms of their EQ sections (Figure 5); if they have any tone controls at all, it is usually limited to a single knob. Adding an EQ pedal right before the amp will allow you to adjust the tone to a far greater degree than you can with the amp's limited EQ. Figure 5: An EQ pedal can greatly increase the tone shaping capabilities of small tube practice amps with minimal controls, such as the Fender Champion 600 (top) and Vox AC4 (bottom).

A three-head tape deck, or even a tape delay simulator pedal can be configured as a tape delay unit for use with your DAW, or used to give your digital tracks the sound of analog tape By Phil O'Keefe In a recent thread on the Harmony Central Effects forum, a member named "Loobs" was asking about the delay effects used on Jimi Hendrix recordings. While I do not know the specifics regarding the actual equipment they used for his classic albums, I mentioned that using studio tape decks (like the Otari MX5050 IIb pictured in Figure 1) for delays was a common practice during the 1950s and 1960s. This led to a request for details about how to set this up, which led me to write this article. Fig. 1: An Otari MX5050 IIb 1/4" three-head reel to reel tape deck. THE CONVERSATION Loobs: You hear a lot of echo on Hendrix's studio stuff. I know he didn't gig with any sort of echo box but what would have been used on his studio stuff? It always sounds awesome. Echoplex EP-1 in the studio? Were there any other outboard tape delays that could've been used? Phil? Phil: I don't know this for a fact (but the next time I see Eddie Kramer, I'll ask him if I remember to do so), but I would suspect they used one of the studio's reel to reel decks, configured as a tape delay. That was very common in studios in the 60s, and was being done even earlier than that by guys like Sam Phillips and Les Paul. But you're right - tape delay units such as the Watkins and Echoplex existed at that time, and given his interest in effects and new sounds, Jimi certainly would have likely known about them. Loobs: Thanks. So presumably with a reel-to-reel configured as an echo you'd only have one single repeat, given that they only have one play head? Phil: No... not at all. You can set up regeneration fairly easily... THE SETUP You need a three-head (erase / record / playback) tape deck that can monitor off of the playback head in near-real time while recording. Most 3 head decks can do this, and even some cassette decks have 3 heads and the ability to monitor off the playback head. It's essential that the tape deck be configured to monitor off of the playback head while recording, and not from the recording head like you normally would when doing an overdub on an analog multitrack deck. This way, instead of the recording and playback signals being synchronized, the recorded signal is delayed; the incoming sound is recorded by the record head, and then several milliseconds later as the tape winds through the machine, the recently recorded tape will pass over the playback head (Figure 2), where the delayed signal is played back and routed back out to the mixing console or computer audio interface / DAW. Fig. 2: A three head tape deck, showing the location of the three heads and the direction of the tape travel (click on images to enlarge). Getting things set up is fairly simple. Route the mixing console's aux send into the tape machine input. Route the tape deck's output back into a second channel (return channel) on the mixing board. Turn up that aux send knob on any channel you want to route to the delay. If you're doing this with a DAW, you can assign a software aux bus to a hardware output of your audio interface, and route that to the tape deck's input; then send the tape deck's output into an input on your audio interface (Figure 3). The software routing on the virtual mixer would be set up exactly the same as on a physical mixing console--use an aux send to route the signal out of the audio interface and into the physical tape deck. Plug the tape deck's output into an input on the audio interface, and assign the "echo return" from the audio interface's input to an unused DAW track (or an aux input mixer channel) to return the delayed signal back to the DAW. Fig. 3: Raise the aux send on the track you want to add delay to, and return and record the delayed signal to a new track. Since this interface only has two analog inputs and outputs, I'm patching interface output #1 to the delay unit, using a pre-fader aux send, and hard panning the return signal to the opposite side (output #2) and muting the source track to prevent unwanted feedback; if you have a multi-channel audio interface, you can assign the aux send and return to secondary outputs and inputs while leaving your main monitoring outputs unaffected. Put the tape deck into record and roll tape. Hit play (or play and record, if you're re-recording the delayed signal) on the DAW. This basic setup will give you a single delay. Raise the level of the aux send on the RETURN channel (Figure 4) for feedback / regeneration / more "repeats". Be careful though: if you raise it too much, you'll get runaway regeneration and uncontrolled feedback. Fig. 4: Raise the aux bus level on the return track for regeneration. Be careful--a little goes a long way! THE DETAILS The delay time will depend on two different factors: First is the speed the tape machine is running at. If you have the machine running at 15 IPS (inches per second), the delay time will be exactly half of what it will be if you instead run the tape deck at 7.5 IPS. Slower tape speeds = longer delay times. You can use a VSO (variable speed oscillator - the tape deck's "varispeed" knob) to fine tune the speed of the tape deck, and thus the delay times. Most varispeeds give you a +/- 6 to 10\% adjustment range. Second is the physical spacing between the record and play heads. If they are spaced further apart, the delay times will be longer for a given tape speed. In other words, if the heads are spaced 1.5" apart, the delay time will be twice as long than it would be if the heads are spaced .75" apart. Here are some examples: At 15 IPS, a 1.5" head spacing gives you a 100ms delay time, and .75" spacing gives you a 50ms delay time. Drop the tape speed to 7.5 IPS, and the lengths of the delay times double: with a 1.5" tape head spacing, you get a 200ms delay, and with a .75" spacing, you get a 100ms delay time. You can get different delay times by increasing or decreasing the tape speeds. Common tape deck speeds are 1 7/8 IPS (cassette and some reel to reels), 3.75 IPS (some cassette decks and reel to reels), 7.5 IPS (fastest speed on most "consumer" model reel to reels), 15 IPS and 30 IPS - both of which are "pro machine" speeds. When using a DAW's virtual mixer and audio interface, your delay times will be slightly different than if you're using an analog console for the sends / returns. This is due to the slight delays involved due to the A/D and D/A conversion stages, and will vary depending on the interface you're using and the buffer sizes you select on your computer. And of course, when re-recording the delayed signal back to the DAW, you can adjust the delay time after the fact by merely sliding or nudging the processed track to a new location relative to the original source track. ALTERNATIVES TO TAPE The simplest is to just use a plugin, but half the fun is in the experimentation, and in getting your own unique sounds. Here's an alternative approach: If you don't have a three head reel to reel deck, but want to experiment with "tape delays" with your DAW, you can utilize a tape delay simulator pedal, such as the Skreddy Echo (Figure 5) or Strymon El Capistan... even a "real" tape delay such as the Fulltone TTE or an old Echoplex or Roland Space Echo can be used instead of / in place of an analog three-head reel to reel. Just put it into the chain in the place of, and instead of the analog reel to reel; the setup is basically the same, but make sure you have the mix control on the delay unit set to 100\% "wet". Set the repeats to minimum if you want to simulate a tape deck, or adjust to taste for additional repeats and regeneration. Fig. 5: A Skreddy Echo pedal is designed to simulate a classic tape delay, and can be used in place of a reel to reel deck. I've dialed in a bit of "warble" to better simulate the "wow and flutter" pitch fluctuations of an old analog tape deck. If you route the delay return to a new track instead of an aux input (return) channel, you can record the results; this way, you can mix in the "delays" later by simply raising or lowering the level of the recorded track, without having to reconnect the aux send and return and the external delay device. TAPE DECK (OR TAPE ECHO SIMULATOR) AS A TRACK SIGNAL PROCESSOR One final nifty trick that I learned years ago from Harmony Central Editor-In-Chief Craig Anderton: This same configuration can be used to "analogize" your digital recordings. Set the external tape machine (or tape echo simulator pedal) up as outlined previously, and record the output of the tape deck to a new track in your DAW. You'll want to make sure you have it set up for only a single delay, with no repeats or regeneration set on the return channel's aux send, or on the pedal's controls. The track will be a reproduction of the original source track, but with the analog processing of the tape deck (or tape echo pedal) applied to it. The tape return channel will be delayed relative to the digital source track (Figure 6). Once you are satisfied with the sound, just nudge it over until they line back up, mute the digital track, and then use the "tape effected" sound instead (Figure 7). Unlike when recording straight to analog tape, you can experiment with how hard you "hit" the tape deck (how loud the recording level is set, and how noisy, clean or distorted the resulting sound will be) and adjust the quality of the "tape sound" after the fact--after you've captured the performance safely to digital, and without wearing the musicians out with countless repeat performances while you experiment with different analog recording levels. Fig. 6: The signal will be delayed relative to the source track. To find out how long the delay time is, drag your cursor from the beginning of the first note on the original track, to the same spot on the first note of the delayed track, and read the length of the delay time on your DAW software's display. Then you can nudge the track by the same amount to get them into exact alignment. Fig. 7: When used as an analog-sound effect processor for your digital tracks, drag or nudge the delayed sound into alignment with the original source track, then mute the original track and use the replacement "analogized" recording in its place. Using outboard equipment in this way isn't just a way to copy "old school" techniques--it's about getting distinctly unique sounds that help your music stand out from the crowd, so experiment and get creative with the sounds you use, and the ways you process them. Sometimes that means thinking outside of the box... SAMPLES OF A DIFFERENT KIND I've included a couple of audio examples so you can hear this technique in action. Rather than utilize my studio tape deck and multi-output Pro Tools HD setup, I used gear that you are more likely to have available--an Avid Mbox interface and a tape sim delay pedal (Figure 8). In the audio samples, the original mono drum machine loop is hard panned to the left, while the delayed signal that was processed by the Skreddy Echo pedal is hard panned to the right. This will allow you to use your DAW's balance controls, or better yet, to split the stereo tracks into two mono files in your DAW so you can listen to either signal independently, or together, in any ratio you desire. Turning the delay track up or down in volume relative to the source track is conceptually similar to raising or lowering the mix knob on a delay pedal--try it! Fig. 8: The setup used for the MP3 examples--an Avid Mbox, and a Skreddy Echo, patched together and ready to go; with the software aux send routed out of the interface and into the delay, and the delay's output routed back into the interface and to a new track in Pro Tools. In the first MP3, the signal is processed with no regeneration - the "repeats" knob on the delay unit is set to zero, and the aux send on the software mixer's return channel is also at zero (Figures 3 and 9). For the second MP3 example, I gradually raised and then lowered the aux send on the return channel while recording, which created more regeneration in the middle of the audio file (Figures 4 and 10). Fig. 9: The audio track waveform display of the first MP3 file; no regeneration, just straight delays. Fig. 10: Raising the return channel's aux send while recording results in regeneration, or "echoes of the echoes", which are clearly visible on the waveform display, and audible in the second MP3 file.

Advance preparation can help you avoid problems on recording sessions By Phil O'Keefe There are a seemingly endless amount of details you may have to deal with when tracking music; some of which can not be anticipated, and that you may have to address on the fly. However, many things can be taken care of in advance, and doing so can take a considerable amount of pressure off of you, and allow you more time to deal with anything unforseen that may arise on the day of the recording. It also allows you more time to serve the needs of the musicians - a crucial consideration for making them comfortable and relaxed. In general, musicians tend to be at their best when they are comfortable, relaxed and happy rather than when they are ticked off, frustrated and grumpy. COMMUNICATE WITH THE BAND Discuss the session with the band beforehand. What type of music and instruments will you be recording? Is it a five piece or a seven piece drum kit? What "vibe" or type of "sound" is everyone going for? If you get the chance to hear a live gig, band rehearsal or demo tape of the band in advance of the session, do it. It will give you an idea about the songs and arrangements and general vibe of the music. Whenever possible, the day of the session is not the time to have them spring that on you. Are they going to play with a click track? If so, you can save time if the tempos are calculated and the tempo maps are done prior to the session. The band should also have rehearsed with a click track or metronome, or at least be experienced with them - the day of the session is not the first time to try playing to a click. Make sure the band understands the importance of pre-production and rehearsals and is musically prepared, with no illusions about writing in the studio or fixing everything in the mix. Of course, in order for these conversations to have the most benefit, they should occur well in advance of the session, to allow for sufficient time for rehersal. Be ready to answer any questions from the band about what to expect, and tell them how they themselves can prepare for the session. Make sure they have their instruments checked out, set up and in good condition. Go over what gear they need to bring, and remind them to bring the all extra little things like spare strings, picks, drum sticks, batteries and other essentials they might need with them too. PREPARE FOR THE UNEXPECTED Just in case they forget, have an emergency kit onhand with a drum key, spare strings, picks, drum sticks, guitar stands and straps, snare drum head, speaker cable, duct tape, cymbal felts, hand tools, allen wrenches, IEC power cable, 9V batteries, tubes, etc. If there's a lot of extra people hanging out, they will eventually distract the band and / or get in your way. Advise the band to leave their friends at home. Mind and mood altering substances can also wreck a session - you're there to record music, not to party and get wasted. Make sure everyone understands this! Check out your studio the night before to make sure everything you plan on using is functioning correctly and set up for the next day's session. You'll have plenty to do the day of the tracking session, so don't put it all off until the musicians show up. Having spare patch cables handy can really save the day if a cable goes bad or you need to connect something you hadn't anticipated. Make sure you have all the standard cable types, and don't forget USB, firewire and MIDI cables. A variety of adapters and "problem solver boxes" such as DI, inline pads, impedance transformers, hum eliminators, dummy speaker load boxes, and speaker attenuators will further increase your connectivity options and might really save the day if you need to connect something unusual or overcome an unexpected connectivity challenge. PREP AND TEST YOUR EQUIPMENT Get mikes, stands and cables out the night before, and have them ready to go in the room, and connected to the preamps you want to use. Nothing kills an artist's vibe as quickly as a bad cue mix, so check the headphone system. Walk out to the tracking room and listen to all the headphones. Replace any blown or distorting cans with working ones. Check your microphones too while you are out there, and make sure they're all passing clean signals. Set up a way to monitor the headphone / cue system from the control room so you can reference what the musicians are hearing. This will help you to quickly dial in headphone mixes, and it lets youu hear what the artist is referring to when they complain of a problem or ask for changes to their cue mix. Many monitor controllers allow you to select between the cue feed and the stereo mix for just this sort of monitoring. Null the board and get your software ready. Templates are your friend, because they save you setup time, so make preset templates of your most common setups - audio routing, type, name and number of tracks, commonly used plugins such as a reverb aux send and return - all the basics you typically need to get you rolling. Have your song files, recording paths and backup drives prepared for the project in advance. Be prepared to take notes and properly document the session. Have plenty of notepads and pencils and pens onhand for your own notes and for the use of the musicians. Use your software's comments fields to note what signal path you used, who the performer is on each track, and - most importantly, be sure to name your tracks BEFORE you record them so your multitrack WAV files aren't all randomly named. Make sure any and all tube rack equipment and microphones you'll be using for the session are turned on for a bit before the session, and don't wait to power them up until right before you roll a take. You want to give things a chance to warm up and let the circuits - and sound - stabilize. PREP YOURSELF It may be hard to do, especially before a "big session" that you're excited or nervous about, but make sure you take some time for yourself. Get enough rest and make sure you're eating properly. Don't expect to party hard late into the night before a session and be at your best the next day. Show up on time, and with a positive attitude, and ask the band to do the same. Actually, the engineer should show up early. If being late is unavoidable, at least call the client to let them know about it, and ask the client to do the same if they're running late. Learn your gear before relying on it for a session. Installing a new piece of critical and complicated gear, such as a digital mixing console or new DAW program, the night before an important session is nearly guaranteed to result in lots of frustrating moments, especially if you spend absolutely no time experimenting with it on your own beforehand. Trying to learn complex features and functions the day of the session is never a good idea. THE CREATIVE VIBE Patching something in to experiment with creating different sounds can often be fun, but frantically running around at the last minute trying to get everything connected for an entire session while the musicians are all standing around waiting on you almost never is. You want to have the time available to do the former, which means you have to avoid the later. The way to do that is to be prepared. You can't predict or control everything, but if you prepare and check your rig in advance, know what you're going to be doing and have a plan of attack, you'll generally have a smoother session, with fewer disruptions, "equipment casualties", delays and distractions - all of which leads to a better overall vibe, and a more creative and productive session. Vibe is everything. The band is looking to you to be the expert. You shouldn't be a "know it all", but you should be confident of your equipment and skills, and unflappable in the face of adversity. If you freak out when some little thing goes wrong, not only will it blow the session's vibe, and make it harder for you to think your way through the problem clearly so you can find a solution, it will completely ruin the band's faith and confidence in you and stress them out. So remember - stay calm, prepare in advance and keep the session fun! 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.

Here's a different kind of board for your studio By Phil O'Keefe While there’s something incredibly cool about the sound of a great guitar plugged straight into a great amp, you can get a lot of other useful tonal colors with some good effects pedals — and if you own a bunch of pedals, you need a nice pedal board. I got a lot of great tips and ideas from the folks on the Harmony Central Effects forum; I’d like to share a few of them with you, and walk you through how I built my pedal boards. GETTING YOUR EFFECTS IN ORDER Lay out your pedals of choice on the floor in a variety of arrangements until you find one that provides easy access to all your pedals so you can stomp on them as needed, without messing up your settings on the other pedals, yet doesn’t take up an excessive amount of space. As to the order of effects, different setups provide different types of sounds. For example, wah-before-fuzz sounds significantly different than wah-after-fuzz, so experiment with the pedal order to find what sounds best. If you’re really stumped, start with fuzzes and filter pedals first, followed by compression, overdrive and distortion pedals, modulation effects such as chorus and flangers, and delay and reverbs last. EQ can work well both before and after your dirt boxes, so try it in both spots. Once you have an effects layout and order, measure around it to get an idea of the needed board size. You can opt for one of several different commercial boards (e.g., Furman’s, which despite being a pedal board for stage use is very “studio-friendly”), or build one yourself. I built two separate boards — one for dirt and filters (Fig. 1), and a second one for modulation and delay effects; otherwise it would have taken a 6' wide board to hold everything the way I wanted. I used standard 1" x 3" lumber for the boards, and screw and glue construction. A few coats of flat black spray paint completed the build. I could have used a large sheet of plywood, but the “slat” style construction allowed spaces for routing my daisy chained power connections under the board, and also cut down on weight compared to using a solid piece of lumber. Fig. 1: A completed pedalboard for the studio. GETTING ATTACHED The next issue is how to mount the pedals to the board. There’s always Velcro, and while it offers quick and easy pedal repositioning, sticking it on the back of pedals leaves a mess, and I wanted my pedals a bit more firmly secured to the board. A great trick I learned on the forum was using bike chain links as mounting brackets (Fig. 2). These allow you to use the existing case screws on the pedals; a few wood screws (or nuts and bolts if you’re using a plexiglass or metal top board) then clamp the pedals firmly and directly onto the pedal board. Fig. 2: Attach bike chain links to your pedal, then attach the links to your board. Practically any bike shop or hardware store will have bike chains and chain repair tools (the pair cost me under $12). You simply use the tool to disassemble several links from the bike chain by setting the chain into the grooves on the tool; line up the pin on the chain link with the pin on the repair tool, and turn the handle on the tool to “punch out“ the chain pins (Fig. 3). For some pedals, like the Fulltone OCD, which have case screws on the lower sides instead of on the bottom, I used bent chain links to mount the pedal to the board. It involved a blowtorch and careful bending of the links using a vise and pliers, so if you try it at home and get hurt, you never heard of me . . . Fig. 3: Using the chain repair tool to separate links. Other pedals required other approaches. With my Line 6 MM4 and Danelectro Reel Echo, the bike chain links weren’t long enough to reach to the screws on the pedal case. For those, I used 1.5" galvanized metal pipe clamps. Again, these are available at hardware stores and come in a variety of sizes. I cut them in half with tin snips, and used two pairs of pliers to bend them either at right angles or flat as the situation called for, then drilled screw holes in the appropriate places. Don’t forget to round off any sharp edges with a grindstone or Dremel tool. If there are no screws on the pedal that you can use to mount the brackets, sometimes you can disassemble the pedal, drill holes into the bottom plate, and screw the bottom of the pedal to the board with sheet metal screws and then reattach the top. If there’s a set screw underneath, as on the Danelectro mini pedals such as the Fish & Chips EQ, you’ll need to drill an access hole through the deck of the board so you can adjust the assembly screw. While I have tried to take a “harm the pedals as little as possible” approach, sometimes you don’t have much choice . . . but don’t deface any expensive or vintage pedals with this technique. Fortunately for me, my über-expensive Lovetone pedals have screw holes in their base plates that made mounting them with sheet metal screws possible without modifications. Eventually I was able to screw down all of my pedals except for my tuner (I used Velcro for that one); I don’t step on it anyway, as I use an A/B switch to bypass the amp and route the signal to the tuner for “silent“ tuning. MORE POWER, CAPTAIN! Regarding power, batteries were out of the question — the arrangement I laid out puts the pedals too close together to unplug all of the input cables easily and bypass the batteries, and battery access isn’t from the top on most of these pedals, so AC power made sense. I used a Visual Sound 1 Spot power adapter (http://www.visualsound.net) on each board to power the majority of the pedals. If some of the pedals hadn’t had different power requirements, such as different voltages or polarities, I could have powered everything with just the 1 Spots and a few of their daisy chain power plugs. But as I had to have more than one wall wart on each of my boards, I needed a power strip into which I could plug everything. Most power strips don’t attach easily to a board, don’t have detachable AC cables, and take up a lot of room. I found a cool alternative at an Ace Hardware store: a Yellow Jacket five-outlet adapter (http://www.woods.com). It’s compact (Fig. 4), costs $9, you can plug a detachable extension cord right into it (it has a male 3-prong AC input and five AC outlets), and it includes “cable locks“ that came in handy for wrapping up the unused extra length of DC power cable. I put screws through the wide, flat plastic panels between the outlets on either side to mount it to the board, plugged in my two power adapters, and ran the daisy chain power cables to each of the pedals — and whenever possible, routed them underneath the board and then up through and between the slats to each pedal. This, along with some nylon cable clamps, allowed hiding a lot of the wiring underneath the board, thus keeping everything organized and tidy. And because the Yellow Jacket has extra, unused outputs, a short extension cable allows plugging in the power for the second board without having to run two extension cables to the wall. Fig. 4: The Yellow Jacket five-outlet adapter. GOT CABLE? For audio wiring, I used four of the Planet Waves pedal board kits. These come with 10' of cable and ten right-angle 1/4" plugs — enough to connect up to six pedals. Other plug types are also available, including straight 1/4", “thin line” 1/4", and even plugs with built in “circuit breaker” switches; these are very handy for quietly switching guitars without buzzes and pops. Assembly is easy, but when you insert the cables, make sure you really have the cables inserted all the way into the plug. Don’t just stop at the first sign of resistance — push hard. A slight quarter-turn “twist” on the cable while pushing inwards seems to help, as does maintaining pressure while setting the locking screw. Also, test each finished cable with a multimeter set for continuity checking. That way you know all is well before you hook up everything. I really like these cables; not only are they solderless and fast to make, but you can cut them to the exact lengths you want, and the connectors can be reused if you change your board layout and need a different cable length. And most importantly, they sound great. With a little thought, some inexpensive and readily available materials and a afternoon’s worth of work, I was able to build two boards. My clients appreciate having the extra toys to play with, and I appreciate the extra tonal options. So even if you’re a dedicated “mix in the box” fan, don’t overlook what having a different type of board can bring to your studio! And for some tips on what to do with your new pedalboard, be sure to check out Phil's Pedal Pointers.

Marshall Class 5 1X10 all-tube combo amp ($399 street) Classic Marshall sound from a small, all-tube 5W amp. http://www.marshallamps.com/product\\_range.asp?productRangeId=29 By Phil O'Keefe A lot of people have been asking for a small Marshall tube combo that offered the classic Marshall attitude in a smaller, lower wattage version; something that would be better suited for small gigs, recording use and home practice than their larger combos and stacks are. Marshall has delivered with the new Class 5. (Fig. 1) Let's take a closer look. Fig. 1 The Marshall Class 5 5W all-tube 1X10 combo amp. OVERVIEW Housed in a "Bluesbreaker" styled combo, the Class 5 sports a familiar vintage look. Controls are on the top of the amp, and consist of only the basics - volume, treble, middle and bass. There is no master volume control (Fig. 2), which some people may miss; despite its relatively small size (19.5"W x 16.33"H x 9"D) and 5W Class A power rating, this puppy can get loud enough when cranked to get apartment dwellers with cranky neighbors evicted. The speaker is a Celestion branded G10F-15 (Fig. 3) that was developed specifically for this amplifier, and is rated at 15W RMS with an efficiency rating of 96.5 dB @ 1W / 1m. The rear panel includes a headphone output with selector switch and a 16 Ohm external speaker output - using either the headphone output or external speaker jack mutes the onboard speaker. Fig. 2 The Class 5's control panel is simple yet effective, but lacks a master volume control. Fig. 3 The Celestion G10F-15 speaker. SOLID CONSTRUCTION Some owners complained about rattling issues with a few early examples of this amp. Rest assured - Marshall has addressed this, and current Class 5's are completely rattle free. I got a good look at the construction quality of the amp when I disassembled it (Fig. 4) to check out the tube access, which requires the removal of 11 bolts and screws and pulling the amp chassis from the enclosure; don't expect to do this in five minutes at a gig. On the positive side, the construction quality is solid - the steel chassis and vented back panel are heavy gauge, and rubber gaskets and stoppers / standoffs at critical locations assure rattle free operation. The Sovtek EL84 output tube is held in place in its ceramic tube socket with a simple but effective spring loaded wire retainer, and the tube sockets for the two (Marshall branded) ECC83 (12AX7) preamp tubes hold the tubes tight and secure. I have had no reliability or noise issues with the amp, and was generally pleased with the sound of the stock tubes. Fig. 4: Replacing the tubes requires the removal of 11 screws and bolts and partial disassembly of the amp. CLASSIC SOUND While the clean tones are fine for recording and practice purposes, you can't expect an amp in this wattage range to stay clean at loud enough levels to hang with a full band. If that's what you're after, you should look elsewhere. What this amp excels at is grit and dirt. The amp stays clean up to about 9 o'clock on the volume knob; hitting about 85-90dB SPL (measured at two meters) in the process. You'll start to notice it getting some grit in the tone by 10 o'clock. Turn the volume control up to about 1 o'clock, and it is nearly as loud as it will get - moving it beyond that point tends to add more distortion than volume. Cranked to the max, it will hit about 105-107dB SPL at two meters from the amp, which is loud enough to cause hearing damage if you play without hearing protection for extended periods. The Class 5 interacts very well with guitar control adjustments. I like to set the amp's volume knob to about the 1 o'clock position; this allowed me to use my guitar's volume knob to quickly adjust from clean to slightly overdriven to all out distorted tones. The basic sound of the Class 5 is somewhat reminiscent of a Bluesbreaker combo, with hints of JCM800 thrown into the mix. It's aggressive and meaty, thick and full sounding, with a touch of snarl and spit, depending on how you dial up the controls and whether you're using single coils or humbuckers. Unless you add a dirt pedal, the distortion sound is more classic rock than "metal", but since it takes pedals so well (the Catalinbread DLS is a standout with this amp), you can get it sounding considerably heavier in the dirt department when desired. Frankly, as long as you're in a situation where you don't have to worry about anyone complaining about the noise, it's a blast to just plug in, crank up and wail through this amp. And if you really want to move some air, plugging it into a 16 Ohm 4X12 cabinet will give you extra push and thump. The stock speaker benefits from a bit of a break-in period before it sounds its best, but don't underestimate it - it sounds bigger and fuller than you'd expect from a 10" speaker in a relatively small combo amp. CONCLUSION This is a great amp for recording. It has plenty of grit and punch, and a surprisingly full sound, despite its relatively diminutive size. It takes pedals extremely well, and if you're looking for classic all-tube Marshall crunch at moderate volume levels and are on a tight budget, this will definitely get you there. The lack of a master volume control is really my only complaint, and apartment dwellers may want to use a good overdrive or distortion pedal instead of cranking the volume, because it can definitely get loud. But that's OK for me - I have a well isolated studio, and this Class 5 is staying right where it is - I purchased the review unit. Class 5 Specs: Power Output: 5W RMS Class A Preamp Tubes: ECC83 (X2) Power Amp Tube: EL84 Internal Speaker: 1X10" Celestion G10F-15 15W, 16 Ohm Channels: One Controls: Volume, Treble, Middle Bass, Power switch, Power lamp Inputs: One, 1/4" Headphone Output with selector switch; bypasses internal speaker 16 Ohm Extension Speaker output (bypasses internal speaker) Dimensions: 19.5"W x 16.33" H x 9" D Weight: 26.5 lb.

For some applications, this venerable mic technique still can’t be beat By Phil O’Keefe While working as an electrical engineer for a Bell Labs subsidiary, and later for EMI, Alan Dower Blumlein (1903-1942) had a profound effect on the fields of telecommunications, TV, radar, and audio recording and reproduction. He developed the first “weighting networks” to compensate for the non-linearity of our ears, designed moving coil mics, and much more. But probably Blumlein’s biggest impact was that he essentially invented stereo recording and playback. Back in 1931, he also created the Blumlein Pair, one of the most useful stereo mic techniques ever (UK patent 394,325; “Improvements In and Relating to Sound-Transmission, Sound-Recording and Sound-Reproducing Systems”). THEORY The Blumlein Pair is a crossed coincident pair of figure-8 velocity (ribbon) mics, each placed 45° off-axis from the sound source, while 90° off-axis from each other — similar to setting up two cardioid mics in an XY stereo configuration. Placing two figure 8 (“bi-directional”) mics in this “X” position provides a very detailed stereo image, and the rear lobes of the two figure 8 mics pick up a significant amount of “room tone” or ambience and reflections. As with the Mid-Side technique (see "Mid-Side Recording"), you place the mic capsules as close together (“coincident”) as possible; but unlike M-S, no polarity inversion or decoder is required. A Blumlein Pair requires two closely matched figure-8 mics to insure balanced stereo recordings. While the patent calls for a pair of bi-directional ribbon mics like the two RCA 74b Junior Velocity mics shown in Fig. 1, two identical, multi-pattern condensers with each set for figure-8 polar patterns are often used instead. While preferred, you don’t have to have a factory matched mic pair; you can use any two mics of the same type and model, as long as each has (or can be set to) a bi-directional polar pattern. If you already own one multipattern condenser mic, or one bi-directional ribbon, consider purchasing a second mic of the same model so you can exploit the Blumlein Pair technique. Fig. 1: Two RCA 74b Junior Velocity mics configured as a Blumlein Pair. PRACTICE Set one bi-directional mic so it’s aimed 45º to one side of the center (“aim point”) of the sound source, and place the other mic directly above the first, aiming it 45º to the other side of the sound source. Make sure each mic’s front side points forward, toward the sound source, so that each is in phase. Referring again to Fig. 1, orient the mics so they’re “aimed” at the sound source; the camera that took the photo is positioned at the sound source location. Route each mic into a separate mic pre and record each to a separate track, or to a single stereo track in your DAW, and pan the two tracks hard left/right. As with M-S recordings, you’ll pick up a significant amount of room ambience, so adjust the mic placement for the desired ratio of direct sound to room ambience (closer to the sound source for more direct sound, further back to pick up more reflections and reverb). Despite its age, the Blumlein Pair technique still has significant advantages. Unlike spaced pair stereo recordings, phase issues are generally not a problem due to the coincident placement of the two mics; and as Blumlein Pairs rely on amplitude differences as opposed to differences in wave phase to generate stereo information, mono compatibility is quite good. Blumlein Pairs also provide “realistic”-sounding stereo techniques, where the stereo imaging is very similar to what you hear if you stand where the mics are placed. In fact, one of my favorite applications is to set the pair just behind and above a drummer’s head, pointing forward toward the kit. This captures an “as the drummer hears it” perspective of the entire drum kit, along with a healthy amount of room reflections. Blumlein Pairs are also suitable for live ensemble recordings, brass sections, and stereo recordings of individual instruments (e.g., acoustic guitars and classical piano) — basically any situation where more ambience is desired than what an XY stereo configuration or close miking provides. So the next time you’re looking for a sense of space and ambience in a recording, say hi to Mr. Blumlein. 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.