The Behringer Ultramatch Pro SRC2496 is a Sample Rate & A/D/A Converter that allows you to transfer digital audio between devices with different sample rates, formats, or interfaces, and remove dropouts or jitter from digital media. The various digital outputs can be used simultaneously, allowing the unit to also serve as a digital signal splitter/patchbay. The SRC2496 also provides up to 24-bit/96 kHz A/D/A conversion, making it what could be the most affordable stereo A/D/A converter on the market.

Analog Connections

The SRC2496 boasts an impressive collection of inputs and outputs. First, there’s a pair each of balanced XLR input and output jacks. These connections are used when utilizing the SRC2496 as an Analog-to-Digital (A/D) and/or Digital-to-Analong (D/A) converter. A pair of balanced 1/4″ TRS plugs would have also been welcomed here, but for the price, I won’t complain. However, if you plan to use it with unbalanced gear of any kind, you’ll need to buy or solder your own cable with XLR Pin 3 shorted to Pin 1, as noted in the user manual.

The analog inputs allow you to connect up a piece of gear with analog stereo outputs, two separate single-channel mic preamps, or perhaps a dual-channel mic preamp (which is how I’m using it). The analog outputs can be connected to any other analog gear, but the most likely candidate would be to a mixing console, monitor controller, or directly to a pair of active studio monitor speakers. To round out the analog connections, there’s even a headphone output on the front panel, which is useful for monitoring the unit’s output over headphones if active monitors aren’t available.

Digital Connections

The digital inputs and outputs include a choice of 3 connectors: XLR, RCA, and Toslink. It’s important to note that these connectors are for DIGITAL signals, even though they use some of the same connectors (XLR & RCA) as common analog gear. It’s also important to use cables that are specifically designed for DIGITAL audio, rather than trying to use a regular old XLR microphone cable or analog RCA audio cables. Each of the digital inputs and outputs will send or receive either a professional-format AES/EBU signal, or a consumer-format S/PDIF signal. This makes the unit extremely flexible in terms of matching up disparate pieces of digital audio gear. Synchronization with other digital devices can be achieved in one of three ways: by utilizing the unit’s internal clock (making it the “master”), clocking to an external source via the digital input, or by clocking to the dedicated Word Clock input (provided via a standard BNC connector).

It’s worth noting that when using the unit as a D/A, it can only be synced to the digital input signal (“Dig In”). The BNC Word Clock input can only be used when using the unit as an A/D converter. In fact, when you choose to sync to the Word Clock input (you choose “External” for the Clock source), the digital outputs of the unit are muted. This caused considerable confusion for me when I first hooked up the unit, as I was trying to use it as an A/D/A, and clock it to the Word Clock input. After discovering this little nugget of information in the manual, I set the clock source to “Dig In”, and all was good.

A/D Conversion

I bought the SRC2496 to use primarily as a 2-channel A/D converter that would allow me to utilize the S/PDIF input and output of my recording interface (an RME Multiface), thereby freeing up a pair of my both my analog input and output channels. To set things up, I connected the analog outputs of a dual-channel mic preamp (an ART MPA Gold) to the XLR analog inputs of the SRC2496, then connected the S/PDIF output of the SRC2496 to the S/PDIF input of the interface. I also connected the S/PDIF output of the interface back to the S/PDIF input on the SRC2496 in order to also use it as a D/A converter, and to clock the SRC2496 to the RME Multiface, which is set as the Master.

I’ve been very pleased with the results so far. I’ve used it for recording vocals, acoustic guitar, bass, and clean electric guitar, and the unit adds no audible noise or distortion to the input signal, even when adding a considerable amount of gain for weaker signals. The SRC2496 provides clean, quiet A/D conversion, just as you would hope for when recording any such sources.

D/A Conversion

The SRC2496 has also gained considerable popularity as a D/A converter, especially among the consumer audiophile crowd. Many have reported that its performance compares favorably with D/A converters costing many times its price, making it a real bargain as a bang-for-the-buck D/A converter for a home theater or stereo system. I also tested it as a D/A converter in the control room of my studio, setting the stereo output of my DAW to the S/PDIF output of the Multiface, which in turn sends the digital output to the SRC2496. The analog outputs of the SRC2496 were then connected to my monitor controller, and then to my studio monitors.

Again, the unit provides clean conversion, adding no audible noise or distortion. I did a few listening tests, comparing it with an analog output pair on the Multiface with the SRC2496, and could tell no real difference, save the fact that the SRC2496 seemed to send a slightly hotter ouput signal, requiring me to turn the volume down slightly compared to the Multiface’s analog outputs.

Sample Rate Converter

Since I’m not using the SRC2496 as a Sample Rate Converter, I can’t really speak to its performance in this specific area. However, I can say with confidence that if I ever need a Sample Rate Converter, I’m certain it will be more than up to the task! The range of controls and connections found here leave little doubt that it’s a very capable unit in this regard, as well.

Conclusion

Behringer has built its business on providing highly useful pro audio gear for the studio and stage that boasts a boatload of features and a price/performance ratio that leave their nearest competitor in the dust. The Ultramatch Pro SRC2496 is no exception, providing an incredible package of features that would be welcomed in any studio or home theater system that’s mixing and matching digital gear with different sample rates or connections, or needing another two channels of quality A/D/A conversion.

Shop for the SRC2496 at Zzounds.com:

Ultramatch Pro SRC2496

For several years, I’ve had some Sennheiser e604′s that I’ve always used for recording the toms on drum kits. Lately though, I had been particularly unimpressed with how my tom tracks have sounded, so I decided to try out a pair of the much-loved-for-toms, CAD M179′s. I only bought two M179′s and set one mic between my two high toms (10″ & 12″) and the other between my two floor toms (14″ & 16″). I engaged the -20 db pad on the M179′s and set the pattern to hypercardioid to try and minimize bleed.

On the other hand, I had an 4 of the e604′s–one mounted on each tom using an old “Mic Eeze” drum rim clamp, since the stock clamps that came with the e604′s don’t play nicely with my RIMS suspension mounts. I had each of the e604′s positioned in what I would consider a fairly typical position for a tom or snare mic–a couple of inches above the head, and pointing down at an angle between the center and outside edge of the drum.

I pressed record, played for a while, then went to the control room to have a listen. I was literally blown away at how much better the M179′s sounded. The e604 tracks sounded terribly lifeless, dull, and honky. You would have guessed that the heads were totally worn out (especially on the 3 larger toms), which isn’t the case–the heads are relatively new.

Here are the tracks from the first pass:

e604 10″ Tom:
e604 12″ Tom:
e604 14″ Tom:
e604 16″ Tom:
M179 Hi Toms:
M179 Low Toms:

It was also strange that the toms sounded very inconsistent on the e604 tracks. The 10″ sounded ok, the 12″ not as good, the 14″–absolute crap, and the 16″ also quite terrible. On the M179 tracks, all 4 of the toms sounded fairly consistent, and generally quite good! I’ll also mention that I was soloing each of the e604 tracks in my DAW as I listened, so phase cancellation between tracks wasn’t the problem here.

After this little experiment, I was left scratching my head. The e604′s seem to be fairly well-respected mics, so I felt like I had to be doing something wrong. So later, I spent some time playing around with their positioning. I kept the mics on the Mic Eeze clamps, but tried pointing them down at varying angles, either more toward the center of the drum, or more toward the edge, where at the extreme, the mic was pointing pretty much straight down, with the face of the mic parallel to the head. Regardless of the angle, they still sounded like junk compared to the M179′s.

Finally, I removed the e604′s and put them on boom stands. Since the M179 is a side-address large diaphragm condenser, I tried positioning the e604′s pretty much the same way–perpendicular to the drum, parallel to the head, about an inch or so above the rim and pointing straight across the head. After thinking about it, I realized this is pretty much the same mic positioning that I use all of the time now when miking a snare drum. After positioning the e604′s in this way, I got a perfectly usable signal for all 4 toms that was much closer to the sound of the M179′s! The high toms sounded better than the floor toms, but they were all usable tracks. I still thought the M179′s sounded better, but the difference was much more subtle this time.

Here are the tracks after changing the position of the e604 mics (the M179′s were left untouched):

e604 10″ Tom:
e604 12″ Tom:
e604 14″ Tom:
e604 16″ Tom:
M179 Hi Toms:
M179 Low Toms:

I guess this is just another example of how critical mic positioning can be, and that some mics are more forgiving than others. So if you have a set of e604′s that you aren’t digging, try this position out, and see if it doesn’t work much better for you. If you don’t have any decent tom mics, definitely try the CAD M179′s. These are great mics, extremely versatile on a number of different sources (not just toms), and come with some nice accessories (huge case, nice shock mount, and nice clip mount for tight spaces). When positioned properly, both mics can give you outstanding results on the toms.

Special thanks to Jon Osborne of Imprint Studios who assisted me with this little experiment.

Shop for these mics at ZZounds:

Sennheiser e604

CAD M179

A few weeks ago, I was watching the movie Kung Fu Panda with my son. In the film, Po (the Kung Fu Panda), finally learns the secret of both Kung Fu, and of his adoptive father’s noodle soup: there is no secret ingredient. If you’ve seen the movie, you know what I’m talking about. A few days later, I started considering how this same revelation applies to the world of recording.

Just like Kung Fu and noodle soup, there is no secret ingredient to recording and mixing. At least there’s not ONE secret ingredient. Instead, there are hundreds, if not thousands of ingredients that can make or break a recording. But it seems that every recordist (myself included) goes through a phase of looking for that one secret ingredient, that one magic bullet, that will take your mixes from amateur to professional overnight. Many are willing to spend a fortune (and some do) on new equipment, only to be sorely disappointed when that new piece of gear turns out not to be the secret ingredient they were hoping it would be.

The reason that there is no one secret ingredient is that there are simply far too many variables involved in this very complex process. Let’s just try to name a few:

• The quality of the source being recorded–the actual sound being produced by the guitar, amp, drum, piano, horn, voice, etc.
• The acoustics of the room where the instrument is being recorded. Even when close-miking, a certain amount of “room sound” will be picked up by the mics, which affects the overall sound of the recording.
• The “vibe”–that intangible quality of a musician’s performance that’s almost indescribable, but you know it when you hear it (or when you don’t)
• The quality of the analog-to-digital converters in your recording interface
• The quality of your mics
• The quality of your mic cables
• The quality of your mic preamps
• Your mic selection, or which mic you choose for each source. For a variety of reasons, some mics are more flattering on certain sources than others.
• Your mic placement–where and how you position the mic relative to the source
• The room acoustics of your Mixing environment
• Your monitoring chain (digital-to-analog converters, Monitor Mixer/Volume Controller/etc.)
• The quality of your monitor speakers
• How you EQ your tracks (while tracking, mixing, & mastering)
• Your use of effects (compression, reverb, delay, etc.)
• Etc.

Don’t get me wrong–it’s debatable how much of an impact each of these factors contributes to the overall quality of a recording. Some would argue that some of these have little to no impact at all, and that’s okay. We can have that debate. What’s not debatable is that even if you cross a few of the more questionable items off the list, you’re still left with a significant number of variables, all of which contribute to the overall quality of the recording.

So what conclusion should we draw from all of this? I think the answer is clear: stop trying to find that (one) secret ingredient–there is no secret ingredient. Instead, try to make the most of each of these smaller, seemingly-insignificant ingredients to make the most of your mix. Strive to do the best you can to maximize each of these, to the extent that it’s within your power–given the time, talent, and gear that you have to work with. If you’ll do this, you’ll be well on your way to better recordings.

If you’re confused by the multitude of common digital audio connections available today on many recording interfaces, such as ADAT, S/PDIF, AES/EBU, S/MUX, Word Clock, etc., you owe it to yourself to read this excellent article from Presonus:

Digital Audio Connections and Synchronization

The Behritone C50A

For quite some time now, I’ve known about the merits of checking your mixes in mono to test for phase problems, and having some “grot boxes” (cheap, lo-fi speakers) to see what your mix will sound like on a less-than-stellar playback system. The idea behind the mono grot box is this: who cares if your mix sounds great on your studio monitors if it sounds like garbage on most listeners’ playback systems?

I followed the logic, but was never really compelled to invest in a grot box or mono monitoring solution until reading the book “Mixing Secrets for the Small Studio” (highly recommended reading) by Mike Senior from Sound On Sound magazine. For the reasons stated above, one of the book’s earliest recommendations is to get your hands on an Auratone subtitute. The Auratone was a cheap little cube-shaped speaker with a 5″ driver that recording studios began using back in the 70′s as a reference standard for other cheap speakers–like those you might typically find in boom boxes, TV’s, and the dashboard of your Ford Pinto.

The Auratone’s design also had some other benefits in that it’s unported, which avoids any problems typically associated with resonant porting frequencies, and it’s a single full range speaker (instead of a woofer + tweeter combination), so there’s no crossover, which can create some negative sonic artifacts of its own. Auratones have been out of production for many years, but there are several other companies that make similar alternatives, including the Avantone Mixcube, the Triple P Pyramid, and the Fostex 6301. I looked into all three of these models, but all were more expensive than what I could justify at the time.

The vintage-styled Behritone C5A

Shortly thereafter, I saw that Behringer was releasing it’s own Auratone substitutes in the Behritone C5A and Behritone C50A. Behringer is known for delivering solid gear for a fraction of the price of its nearest competitor, and the Behritone looked to be no exception. I chose the C50A rather than the C5A, mostly because it has a wider frequency response of 90 Hz to 17 kHz vs. the C5A’s response of 150 Hz to 15 kHz. In this regard, the C50A is actually more of an Avantone Mixcube clone than an Auratone clone. So in a sense, the C50A is a “clone-of-a-clone”. The C5A, on the other hand, not only has more of a “vintage vibe” in its woodgrain appearance, but its frequency response more closely resembles that of the original Auratone.

The C50A is an active/powered speaker, with a 30-watt class D amp built into the cabinet. It’s quite heavy and solid for its small size. It has an attractive glossy piano-black finish and no controls on the front–only a power/peak indicator light that normally glows blue, and changes to a purple color when powering on, or if the input is clipped. The back panel has a power on/off switch, standard IEC connector for the included power cord, and three sets of inputs: balanced XLR, balanced 1/4″ TRS, and an unbalanced RCA. There’s also an input trim control with a range from -6 to +6 dB. The bottom of the cabinet has a neoprene rubber pad pre-installed to help de-couple the speaker if placing it on a desk or table. It also has a handy threaded insert in the bottom for mounting the speaker on top of a mic stand, similar to what you see with the compact “hot-spot” style stage monitors.

The rear panel of the C50A

Since the purpose of a grot box is not to sound great, this speaker does its job well! It sounds like a typical full-range speaker of its size that you would commonly find in a portable radio, television, dashboard, or computer speaker. Even though it doesn’t deliver a “hi-fi” sound, it has a certain vintage charm to it that’s reminiscent of those days of crusing in the car and listening to your favorite songs on the radio. So it’s in no way unpleasant to listen to.

In use, I’m finding that the C50A helps to uncover potential problems in my mixes that I might not normally detect over my main stereo monitors. Specifically, it’s helping me to make more informed decisions about EQ, helping me to watch out for problems with phase cancellation, and letting me know what my mix will sound like for a typical listener. It also provides a nice “reset” for tired ears if you’re in the midst of a long mixing session, and simply need a break from your larger monitors. All-in-all I think the Behritone is a small, but smart investment for a studio of any size that’s looking for another inexpensive way to improve your mixes.

Shop for the Behritone at Zzounds.com:

Behritone C50A

Behritone C5A

Most recordists will agree that acoustic drums are perhaps the most challenging instrument to record (and record well). One of the reasons (among several) is that a drum kit has so many moving parts that are prone to rattles, squeaks, buzzes, and other annoying sounds. It’s also quite a task to tune a drumkit, which always needs to be done prior to any recording session.

So here in this article, I’ve attempted to list out some things that I’ve routinely done in the past to prepare a drumkit for recording. If you own a set of high-end drums, you may find that your drums already include some of these improvements (such as nylon washers, greased tension rods, stuffed lugs, etc.). But if you own a cheaper to mid-grade kit, your drums will definitely benefit greatly by making some of these tweaks and upgrades.

Replace Washers and Lubricate Lugs

Remove each of your lug rods and swap out the metal washers for nylon washers. Nylon won’t rust, rattle, bind, or squeak, and also allows for smoother tuning. If the threaded ends of your lug rods are ungreased, dip the tip of each one in Vaseline before inserting it back through the hoop into the lug. This will lubricate the threaded insert of the lug and allow for smoothing tuning.

Pack the Lugs

If you’re playing an older set of drums, the lugs probably have a metal spring inside that holds the threaded lug insert in place. Those springs are notorious to rattle. To remedy this, remove one of the heads from the drum, remove each lug, and stuff a cotton ball or a piece of foam inside it so that it fills the void inside the lug and touches the spring. This will help to dampen any vibrations in the lug and prevent the spring from rattling. If your drums are newer and have plastic inserts instead of springs, packing the lugs generally shouldn’t be necessary.

Lubricate Pedals

Lubricate the moving parts on your bass drum and hi-hat pedals. Use some 3-in-1 Oil or lithium grease (or whatever the owner’s manual for your pedal recommends). This will make your pedals faster, smoother, and should help to eliminate any squeaks. Be sure to lubricate any bearings, U-joints, or other moving parts on the pedal/s.

Check the Bearing Edges

The “bearing edges” are the angled top and bottom edges of a drum where the head actually contacts the shell. It’s important that the bearing edges are smooth and even so that the head can be properly seated against the shell. To inspect the bearing edges, remove the heads, and wipe them clean with a damp cloth. There should be no cracks, gaps, or dents in the edge, or you may have problems getting the drum to tune properly. If an edge is badly damaged, you might consider having it re-cut by a professional drum-builder. If the edge looks good, you can simply replace the head. If the edge is a little rough, or there are small dents or voids in it, you can rub a light coat of paraffin wax on it to fill the voids. The wax will give a more even surface for the head to contact, and will help the head slide smoothly over the edge as you tune the drum.

Replace Your Heads

Replace any old, worn-out heads with new ones. Heads have a limited useful life, and when you’re recording, you want your drums to sound their absolute best. Remember, this recording will be around to haunt you for years to come, so you don’t want to cringe every time you hear that song and notice how dead the toms sound. Also, don’t forget to replace your resonant (bottom) heads once in a while, too. I’ve been guilty in the past of thinking that because I don’t actually hit the resonant heads, they should last forever, but that’s just not the case. These heads will lose their liveliness, too, which will adversely affect the resonance of the drum.

Tune Your Heads

Even if your heads are relatively new and in good shape, tune them up before each session. You’d be surprised at how quickly a drum head can go out of tune with even very little playing time. For tips on tuning drums, read my article on Drum Tuning Tips. Also, I highly recommend owning a Drum Dial. The Drum Dial is an ingenious device that makes tuning the drums infinitely easier than it is without one.

Inspect Your Snare Wires

Turn your snare drum upside down and inspect the snare wires. Cut away any wires that have broken or are loose. Be careful not to leave any sharp ends of the wire in place that could cut you or poke a hole in the bottom head. If more than 2 or 3 of the snare wires have broken, simply replace them with new ones. Also inspect the plastic strips or string that attaches the snare wires to your strainer and butt-plate. Replace them if necessary, too.

Cymbal Stands

Tighten all wingnuts (with the exception of the wingnuts that hold the cymbals down) so that they don’t rattle. If there’s a metal wingnut on the top of the stand that holds the cymbal down, replace it with a plastic wingnut, if possible. The plastic won’t rattle, and won’t damage your cymbals like metal can.

Muffling the Snare and Toms

If your Snare and Toms seem a bit too ringy, you can try muffling them. There are many different methods that can be used for this, but I have two specific ones that I will mention. The first is using the plastic o-rings, such as the RemOs by Remo, or the Aquarian Studio Rings that simply lay on top of your batter heads. These work pretty well, but don’t give you much control over tweaking the sound. Some kits, such as the RemOs Snare Kit will include a few rings of varying widths, giving you more control over the sound of the drum.

The second method (and the one that I prefer) is using Moon Gels. Moon Gels are sticky little gel squares that you simply stick to the batter head of a drum that helps to deaden the drum and kill some of the overtones. You can achieve a wide variety of sounds using a Moon Gel by simply changing the placement of the gel–just move it closer to, or further away from the edge of the drum until you like how it sounds. You may also try cutting the gels into smaller pieces to achieve a less pronounced effect (I usually cut them in half), or even try using multiple gels on a single drum.

Muffling the Kick Drum

Many drummers will use a pillow or blanket to muffle their bass drum. In my experience, this often deadens the drum too much for my taste. So I recommend muffling your bass drum using a piece of foam, like maybe from a mattress pad that you would typically use on a bed (sometimes called “eggcrate foam”), or even a piece of foam that you can buy at a hobby store for re-upholstering a chair. Any low-density foam around 2″ to 3″ thick should work. Cut a piece of the foam to fit the depth of the drum so that it touches both the batter and front head. Cut the length so that when you put it in the bottom of the drum, it covers the bottom and curves up the sides to a little less than half the height of the drum.

Cut a Hole in the Front Bass Drum Head

If the front head of your bass drum doesn’t have a hole, you should cut one into it to allow a microphone to be placed inside of the drum. You can even buy adhesive-backed plastic rings that you stick to the head first, and then cut around the inside of the ring with a razor knife. This plastic ring will reinforce the head around the hole to keep the head from tearing.

Conclusion

I hope that some of these suggestions will be helpful to drummers who are gearing up for their next recording session. Most kits will benefit from some, if not all, of these tweaks or upgrades. If you’ll follow the advice above, your next session will sure to be a “smashing” success.

Tunecore is a service for musicians that allows you to sell your music on the iTunes store and other digital retailers with minimal hassle. I recently had a client who was trying to get his music uploaded to Tunecore, but wasn’t sure how to create the specific file type that is required by them. Tunecore requests 16-bit, 44.1 kHz Wav files, and here’s how to create them in iTunes:

1. Open iTunes
2. In the menu click Edit>Preferences
3. Click the Import Settings button
4. Import Using: “WAV Encoder”
5. Setting: choose “Custom”. This will open the “WAV Encoder” window.
6. Set the Sample Rate to “44.1 kHz”
7. Set the Sample Size to “16-bit”
8. Set the Channels to “Stereo”
9. Pop in the audio CD and import it as you normally would. iTunes should create WAV files instead of MP3′s.
10. When you’re finished, be sure to change your Import Settings back to MP3.

After creating these instructions, I found something similar on the Tunecore FAQ, but I wouldn’t suggest following them, or you’ll be converting from the inferior, lossy, data-compressed MP3 format, to the higher quality, uncompressed Wav format, so the quality of your Wav files would actually be inferior to the original Audio CD.

However, if you follow my instructions above, you’ll be converting from an audio CD (which is already 16-bit, 44.1 kHz) to 16/44.1 Wav files, so there should be no degradation of the quality.

When working in the studio, some producers/engineers prefer to record each instrument completely separate from each other. First, they’ll record the drums, then the bass guitar, then the rhythm guitar, then the keyboards, etc. It’s almost like building a brick wall, with each instrument representing a single layer of bricks. Each layer is stacked one on top of the other until the wall (song) is complete. This technique gives you maximum separation between tracks, but may take a long time for each player to complete their parts, which will ultimately affect the cost of the song if you’re paying the studio by the hour.

Others like to record the entire band at once–with all of the musicians playing in the room together at the same time. The separation between tracks may not be quite as good as with the first method, but it also has its merits. In general, it seems to create more of a “live feel”, and helps the recording to sound more like the band does when you actually hear them play live. Since you may also be able to keep the tracks from multiple instruments for a given take, working this way usually takes much less time to complete a song, and therefore it won’t cost as much in terms of studio time. In both scenarios, each instrument is still recorded to its own track/s so the songs can be properly mixed later.

I prefer to take a “hybrid” approach when it comes to recording bands. First, I’ll record the basic rhythm tracks together. Then, we’ll go back and fix mistakes, do any overdubs for layered parts, record any guitar solos, and then finally record the vocals. Many top producers, such as Kevin Shirley (Aerosmith, Rush, Kiss, Journey, Iron Maiden, Dream Theater, Black Crowes, Joe Bonamassa, Mr. Big, etc.) prefer to work this way, so this is a proven method.

More specifically, I like to set up all of the rhythm players (drums, rhythm guitar/s, bass, etc.) in the room together, and give each of them their own headphone mix, which will include a click track, some of their own instrument, and some of the other instruments. I might also set up a vocal mic in the control room for the singer, which the players will also hear in their headphones, just to help everyone stay together. I can create completely separate mixes in each set of headphones, so each player can decide exactly how much of each instrument and click track they want to hear. When recording, I’ll record all of the instruments, but the only performance that’s really critical is the drums. I’ve got some acoustic baffles that I’ll put in front of the guitar and bass amps, so the drum mics don’t pick up enough of them so that they’re really audible. When the drums are done, we can go back and fix mistakes and do any overdubs for the guitar and bass tracks, or completely redo them if necessary.

While cutting the basic rhythm tracks, I’ll ask the guitar players not to play any solos. They will usually just drop out and stop playing completely during the solos, or they’ll keep playing the rhythm part, which we may or may not ultimately use. The reason for this is that unlike the rhythm parts, the solos will often be audible enough in the other mics that they won’t be covered up when we do the overdubs. Also, most guitar players don’t have their solos set in stone anyway, so when we actually do the solos (during the overdub process), we’ll just keep recording multiple takes of the solo until they play it to their own satisfaction. Finally, when we’ve fixed any mistakes, recorded any layered parts, and recorded the guitar solos, then we’ll record the vocal tracks to finish off the song.

As I said, each method has its own set of pros and cons. Recording everything separately gives you maximum separation between tracks, but you sacrifice some of the “feel” or “vibe” of the song. It also usually takes much longer in terms of time. Recording everything together generally goes faster and gives you a better feel, but you may sacrifice some of the separation. In any case, hopefully this article has given you a better understanding of these two different approaches to recording bands, and you’ll be better equipped to decide which method you would prefer to use in the studio.

Between November 2010 and March 2011, I had the opportunity to do something I had never done before–record the audio for a short film. Some friends of mine who have a video production company here in Northeast Arkansas (Anthem Pictures), were producing it, and recruited me to handle the audio.

I recorded everything in the field using their Zoom H4N Handy Recorder, along with an Audio Technica shotgun mic and boom pole–standard equipment for this sort of gig. It was often a challenge to get clean audio of the actor’s lines, while trying to minimize any background noise and keep the mic out of the camera frame at the same time. There were some takes where I wasn’t quite sure how well the audio would turn out, but with a little “audio voodoo”, the film’s editor was able to make it work.

“Rise Up” tells the story of Eli, a troubled youth who battles with mental health issues, but ultimately gets help through a mentoring program called “Action For Kids”. Eli later grows up to serve as a mentor himself, and helps out another troubled teen. The script is based on the real life experiences of several different families who have participated in the Action For Kids program. It contains plenty of drama, humor, and even some action sequences (try running down an alley at full speed carrying a boom pole and a mobile recorder around your neck sometime). As is often the case, the scenes were all filmed out of sequence, so it was interesting to see the story unfold as the scenes were played out.

The schedule was pretty grueling at times. The film came in at just over an hour, and the whole thing was shot over the course of eight days, which I’m told is quite a feat. Some of the cast and crew were driving into Jonesboro from Memphis and Nashville, so we had to make the most of every minute we had them, and worked a few really long days–I think one day we worked for 19 hours straight, taking breaks only for meals.

Back at my studio, I also got to record an original song (also entitled “Rise Up”) that was used in one of the scenes of the film. The song was an acoustic number, with only a few layered guitars and vocals. It was written by the Executive Producer of the film, and sung and performed by some friends of his. I finished mixing and mastering it, and sent it off to the editor only 5 days before the premiere, but he managed to work it in. I was worried that he wouldn’t have time, but I later learned that they finished the final cut the very day before the deadline!

The Jonesboro premiere was held on Tues, May 24th at the Malco Hollywood Cinema. There was a packed house, and everyone was excited to see a legitimate movie that had been filmed mostly here in our community. The film was shot using a Red One Digital camera, the same camera used on several major motion pictures, and it looked fantastic. The premiere was really quite an event, and a somewhat surreal experience to see and hear something that I had worked on come to life on the big screen.

In conclusion, I’m not sure if I would ever want to be a full-time audio engineer for films. I’m really quite happy recording bands in the comfort of my own studio. But this was certainly a cool experience, and a project that I’m glad I got to be part of. I got to meet and work with some really great, talented people, and the end result is something that we can all be proud of.

How many times has this happened to you? You’ve been slaving away for hours, even days, mixing your latest musical masterpiece in your home studio. Everything sounds absolutely perfect. Then, you burn it to a CD, jump in the car, and pop it in your CD player. You’re ready to hit the open road and jam out to the greatest tune you’ve ever recorded. Then, the CD starts playing. You’re underwhelmed. The CD sounds dull and lifeless. There’s not enough high end. There’s too much low end. And what happened to the snare drum? You can hardly hear it at times because it’s being buried in the mix by other instruments.

This is a familiar scenario, and it reveals one of the greatest challenges of recording and mixing audio: creating a mix that’s “portable”, or “translates” well on other playback systems. In a nutshell, this means that your mix sounds good on a variety of playback systems: your studio monitors, car stereo, home stereo, computer speakers, Ipod earbuds…well, you get the picture. One of the biggest hurdles to creating a portable mix is accurate monitoring in your control room or mixing space. Notice that I said “accurate monitoring”, and not “accurate monitors”. The reason I say that is because accurate monitors are only part of the equation when it comes to accurate monitoring.

That being said, accurate monitoring should start with accurate monitors. You want monitors that have a reasonably flat frequency response and are not “flattering” to the music. In other words, you want monitors that are honest, and don’t artificially hype certain elements of the mix to make it sound better than it really does. If your monitors hype certain elements, it makes it very difficult to make decisions while you mix.

For example, I once owned a set of monitors that were very bright. They over-emphasized the highs so that when my mixes were played back on other systems, they sounded dull. I owned another set of monitors that were very small. Because of their small size, their bass reproduction wasn’t that great. As a result, I would over-emphasize the bass in my mixes. When I played back my mixes on other systems, the bass was too much, because I was over-compensating during the mixing process.

Hence the need for accurate monitoring. If you can’t hear a true representation of how your mix sounds, you will under-emphasize the elements that are being emphasized by your monitoring. When your mixes are played back on other systems, those elements will be lacking. Conversely, if your monitoring system under-emphasizes certain elements, you will over-compensate, and those elements will be too prominent when played back on other systems.

Now, before you even ask it, I’ll try and answer one of the most oft-asked questions on recording and pro audio forums: which monitors are the best, and which should I buy for my home studio? The answer is: it depends. It depends most notably on your budget and your needs. You can spend as little as $100 up to many thousands of dollars on studio monitors. My advice is to buy the best monitors that you can to meet your needs, using as much money as you can spare within reason.

When I say “to meet your needs”, what I mean is this: if all that you plan on doing is recording and mixing your own songs in your home studio for your own enjoyment, you can get by with spending a lot less than someone like me, who also pursues this studio thing as a business. In fact, if you’ve got a home studio for the primary purpose of writing and recording demos, I would actually recommend that you keep it simple–spend most of your time and energy on writing and playing instead of spending all of your time, energy, and most notably MONEY on less-than-stellar recording gear. Then, when you’re ready to record a serious CD for commercial release, come see me at my studio (for more on this, check out the article Home Studios are Killing Music, by Producer/Engineer Ronan Chris Murphy).

Ok, so how do you decide what monitors are the best for the money? My advice there is to research the forums, read buyer reviews on websites and online stores, and read reviews in professional pro audio publications and on their accompanying websites. If the general consensus seems to be that they’re good, they’re probably good. If the general consensus seems to be that they’re bad, they’re probably bad. However, you need to recognize that some folks are cheerleaders when it comes to certain brands and haters when it comes to others. That’s just a fact of life. I tend to dismiss any review that says something like this: “[FILL IN THE BRAND NAME HERE] sucks. I wouldn’t buy their products if they were the last ones on earth. One time I bought one of their speakers, and it got so hot it caught on fire and burnt my house down.”

Let’s be honest. Most brands make some products that are good, and others that are not so good. Everyone’s entitled to a failure every once in a while. Just keep in mind that if you’re buying a cheaper brand that’s not known for durability and reliability, you may want to check into the return policy for the store you’re buying from, and find out what kind of warranty repair or replacement service is offered by the manufacturer. Otherwise, if your research bears out that a certain product from a cheaper brand performs well and provides good value for the price, by all means, buy it and try it.

If you’re able, don’t just rely on online reviews. Get your butt out of the studio, drive to local music store that has several models set up, and audition each of them using some familar CD’s as reference material. Maybe even take some of your own mixes (which you should be REALLY familiar with) on CD to try them out, and see how the various models stack up. Ultimately, choose the monitor that you think is best that’s within your budget.

Now, to my earlier point of “accurate monitoring” vs. “accurate monitors”. My point here is that even the best, most accurate monitor will still sound bad if placed within in a bad acoustic space. For example, I’ve played my drums in certain venues where they sounded great, and others where they sounded terrible. Same drums, same heads, same tuning, same sticks, same drummer, etc. The only difference was the room in which they were played. The same holds true with speakers. If your room sounds bad, your speakers will most likely sound bad, or at least be inaccurate to the point that it’s detrimental to your mixes. So, not only must you spend some money on the speakers themselves, you also need to spend some money on acoustic treatments to improve the acoustics of your room. Acoustic treatments are an education in and of themselves, and are beyond the scope of this article, so let’s leave it at that for now.

Even after you’ve treated your room, there may still be some acoustic anomalies working against you. In this case, there are still some options available. Many active monitors on the market today include some kind of “room compensation” or eq settings. These can help you to tailor the speaker to your room by cutting or boosting certain frequencies based upon how well your room is treated and the position of the monitors in the room (if they’re near the walls, near the corners of the room, etc.). If you buy monitors that are equipped with such settings, consult the owner’s manual for guidance on setting them up properly.

Another option is to use some kind of “room correction” system in conjunction with your DAW. These generally consist of some kind of measurement mic and some software for performing a room analysis. Based on the room analysis, your system will be configured (EQ’d), usually via a plugin that you’ll place on the stereo master bus in your DAW. This plugin will attempt to correct the frequency response of your system so that it’s optimized at the listening position, making it easier to make accurate mix decisions. Then, when you mix down in your DAW, you’ll need to bypass or remove this room correction plugin so that the resulting mix is unaffected by it (this way, it only affects your monitoring).

By following the suggesions above, you should be well on your way to more accurate monitoring. When your monitoring system is accurate, you’re able to make more informed decisions about your mix, creating a mix that’s portable and translates well on a variety of different playback systems.