Tuesday, August 14, 2012

The Science Of Sound Recording: Digital Recording

The essentials of digital technology as it relates to recording

Here we present a portion of a chapter in the book “The Science Of Sound Recording” by Jay Kadis, published by Focal Press.


Whatever system is used to acquire digital data, we are faced with the same dilemma we encounter in the analog recorder when it comes to storing the information.

Although computer memories may store the data temporarily, most computers use dynamic random access memory (DRAM) chips that lose the data when the power is removed.

For long-term storage, magnetic media are used for digital recording as well as for analog.

Because digital data require only two states and not the complete linearity demanded by analog recordings, there is a difference in the way the process is applied.

Digital magnetic recorders use saturation recording, leaving all magnetic domains polarized in one direction or the other with no intermediate levels required, so the bias current used in analog recording is unnecessary in digital recorders. The density is quite high in digital recorders, which introduces some problems not encountered in analog recorders.

The number of bits per unit area of medium is limited in the longitudinal recording method used for analog record, although it is sufficient for the demands of analog recording.

Digital magnetic media may benefit from closer packing of domains, which is achieved by using perpendicular recording, in which the domain magnetic fields are magnetized perpendicular to the medium surface instead of along the surface as in analog recorders (Figure 1).

Digital data stored magnetically requires only two discernable states for binary information. This requirement is easily achieved by magnetizing domains fully in one or the other polarity.

Figure 1: Comparison of longitudinal recording used in analog magnetic recording and perpendicular recording used in many disc drives. (click to enlarge)

Though this approach avoids the nonlinear region of the M-H curve, it introduces another problem: the interference between closely occurring bits. If the write head and the medium are not capable of altering the magnetic polarities as rapidly as the bits are changing, the magnetization from the previous bit will affect the next bit, which causes the data to be altered because the overlap makes discriminating between ones and zeros unclear. This intersymbol interference limits the data density that can be stored.

We have several options for storing digital data, including dedicated devices using tape or discs as media and general-purpose personal computers with added interfaces to acquire and convert analog audio. The high data density required for storing digital data made early digital recorders quite complicated, requiring rotating magnetic head recorders designed for video recording or using high tape speeds with stationary heads requiring more than one data track for each audio channel to provide enough bandwidth.

The personal computer has largely replaced the mechanically complicated digital recorders as the preferred storage device for digital audio recordings. The low price and wide availability of large, fast disc drives has spurred a move to the computer as the digital audio recorder of choice, especially as the computer can take on the functions of editing, mixing, processing, and storing the entire project in a single device.

A recent development is the flash RAM chip, popular in USB memory sticks for example. This magnetic nonvolatile RAM – though slower to read and write than a hard disk – is becoming popular for non-time-critical recording such as backup of sound files and stereo sound file distribution.

The evolution of digital recorders has been rapid. Rotary-head modular digital multitrack recorders and stereo DAT recorders enjoyed only a few years of widespread use before the move to the general-purpose computer as the preferred platform for digital recording. These machines temporarily bridged the gap between high-cost stationary head professional digital recorders like the Sony DASH and Mitsubishi Pro-Digi systems and analog multitrack.

The Alesis ADAT and TASCAM DTRS machines used videotape, which was cheap and readily available, to provide inexpensive access to digital recording for a wide range of users. These machines, though initially inexpensive, suffered from their complexity when head wear and transport malfunctions required difficult repair and diagnosis procedures.

Yamaha produced the DMR/DRU series of recorders, which used stationary heads and proprietary tape cassettes, to deliver 20-bit 8-channel digital recording in the early 1990s, but they were expensive relative to the ADAT and DTRS machines and never caught on.

None of these tape-based systems survived the move to computer-based systems, and all have been phased out or will soon be retired. Although tape provides the advantage of removable media, the large size of hard drives and the availability of plug-and-play computer interfaces for storage media has diminished the attractiveness of tape-based digital recorders.

The ability to use inexpensive, mass-produced personal computers for digital audio recording and mixing has greatly expanded the accessibility of these tools.

The addition of a FireWire, USB, or Thunderbolt audio interface and some software is all that is required to create a digital studio entirely within the computer.

This change has had a dramatic effect on the recording studio and the music business in general.

Essentially, the entire recording studio can now be contained in a single piece of equipment, with the ability to recall the entire project and studio configuration in a few seconds.

The advantages of digital audio are hard to ignore, even for those dedicated to the analog studio paradigm.

Use of personal computers for audio recording has introduced a new set of difficulties.

Each operating system and hardware platform requires different software, and there are differences in the bus structures and interface ports available that complicate the choice of peripheral audio interfaces (Figure 2).

Figure 2: Computer interface speeds. (click to enlarge)

Input/output buses include FireWire and USB high-speed serial interfaces, both of which are possible choices for connecting multichannel A/D and D/A modules to the computer to provide audio access. The Thunderbolt interface protocol promises even faster device interconnection.

The software for recording interacts with the operating system to access these audio inputs and may do so with differing speed capabilities on different computers.With the main choices for personal computer operating systems – Macintosh OS X,Windows, and Linux – several types of interface are supported, but different recording programs are required and the performance of the audio interfaces may differ due to differences in the hardware and device drivers employed in the particular computer used.

Recording engineers must now have some knowledge of the internal workings of their computer. If something goes wrong with the recording system, it becomes necessary to isolate the problem by troubleshooting a complicated series of interactions between software, computer, and peripheral hardware that may not be well documented. Each manufacturer provides information about their part of the system, but no one company is responsible for the entire system, leaving the user to deal with the problem.

An issue we encounter with digital audio that is not found in analog systems is related to the time it takes to execute instructions. Even in complicated analog systems, computations occur in real time or instantaneously to human observers. Digital processes take varying amounts of time to complete, making parallel processes no longer synchronous.

When monitoring inputs through computer-based audio systems (software monitoring), there is a time lag between the sound input and the sound played back by the program. The delay is a function of the sample rate and buffer sizes chosen.

An alternative to this software monitoring is to monitor the analog inputs through a mixer at the input rather than through the software. Many systems include automatic delay compensation to resynchronize internal processes, but this function does not eliminate the delay we encounter from the A/D and D/A conversion processes.

When digital audio devices are connected, their clocking must be identical to maintain synchrony. Thus digital audio introduces a need for clock distribution that is not found in analog systems.

One of the advantages of software is the ability to refine and upgrade its performance over time. This ability can also be a drawback if the compatibility issues we encounter with continual updates continue to render code obsolete at a rapid rate. Not only the inherent performance of the software itself must be considered but also the interaction of the applications with the operating system used by the computer.

The operating system (OS), the code that determines the operation of the CPU and peripherals, is developed either by the company that makes the computer or by an outside company that provides the OS software. The recording application software may be written by programmers without advanced knowledge of the new developments in the OS. Keeping the OS and application software synchronized can therefore become a major issue.

A full-time studio technical staff often provided such maintenance in the past, but the personal computer–based studio is frequently the responsibility of a much smaller staff or simply the engineer alone. The engineer must therefore become a knowledgeable computer technician in order to keep computer-based recording systems working smoothly.

“The Science Of Sound Recording” by Jay Kadis, published by Focal Press (ISBN: 9780240821542), is available here. Purchase the book from and use discount code FOCAL30 at check out to get 30 percent off and free shipping in the U.S.

Posted by Keith Clark on 08/14 at 09:38 AM
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Monday, August 13, 2012

RE/P Files: Studio Design And Construction

From the archives of the late, great Recording Engineer/Producer (RE/P) magazine comes a wealth of knowledge on live-end/dead-end acoustics which first appeared in the June 1982 issue.

“The most important piece of equipment in a recording studio is the control room,” says Phil Greene, chief engineer and part-owner of Normandy Sound, located in Warren, Rhode Island.

It’s that kind of thinking that led Normandy, one of the first 24-track studios in the region, to become the first facility in the six states to feature a certified Live-End/Dead-End control room.

Since the new room opened last October, business has been good, but that’s not necessarily due to the new control room.

One of the more important clients has been Billy Cobham, who came to this mill town from his home in Switzerland, on a blind recommendation from his bassist Tim Landers, to record two albums that were quickly picked up by Elektra/Musician.

R-e/p spoke with Normandy’s chief engineer Phil Greene on two occasions: the first was in January, while tracks for the Cobham session were being laid; and the second was in March, right after Cobham’s tour with Bobby and the Midnites, during the grueling 12-day mixing session.

Greene wastes no time explaining what the LEDE concept means to him. “It’s as close as you can get in reality to an anechoic chamber at the front of the room,” he says.

“Obviously you have the window and the console, but there is an essentially uncolored signal path between the speakers and the ears, with no phase or frequency-response abnormalities — you hear the speakers, not the walls.

“Of course, if the whole room were an anechoic chamber, you’d go crazy, so the rear wall is diffuse reflective, more or less centered towards the mixing position.”

“Not only does it recreate some ambience, so that you don’t put huge amounts of reverb on the tape, it also improves the sense of where sounds are coming from in the stereo field.”

Arguments can be (and are) made that such a “clinical” environment bears no relation to the outside world, and that this concept is just another room that a producer has to get used to before his or her product will translate well to the street.

But Greene still feels it is useful. “Of course it’s unrealistic; every listening environment is. But what this concept does is eliminate one generation of listening error, which is the almost random effect that a room usually has.’‘

The Conversion Process
Normandy’s old room was well respected for its accuracy, and it wasn’t an easy decision to rip it out and start over again.

“I wasn’t unhappy with the old room, to be totally honest,” Greene admits.

“It was ‘splashy,’ so I tended to mix things dry. Sterling Sound, who does most of our disk mastering, commented at one point that our product was a little dry. The room also had a hyper-preciseness that was unnatural.”

“It would tend to dry up the bottom, so that those tracks were always too separate — it was hard to get them to blend. We thought we should be able to do a better job monitoring.”

“If you’re secure in knowing that what you’re hearing is what’s really there, then you can make a good record, no matter what your equipment is like.”

“We’ve stressed that attitude ever since we first opened as an 8-track. We were comfortable with the old room, so going with LEDE wasn’t a necessity for us, but it was a good choice.”

“When I was looking around for a design, everybody sent me to Don Davis. I had agreed with the theory for years — it’s pretty hard to shoot holes in — but it seemed that it might be so far ahead of the real world that it might turn out to be too exotic.”

“We talked to the folks who designed the [New York] Record Plant, which is a great room, but they seemed to play a lot of it by ear, and I wanted a design I knew would work right the first time. It was a little discouraging when I went to a few rooms in the area that were uncertified attempts at LEDE, and they were total failures.

Studio diagram. Click to enlarge.

“What finally clinched it for me, however, was an interim step we took when we removed the ceiling from the old room, and just left the insulation and the cloth covering hanging there — sort of a Dead Top. I loved the mixes I found myself doing, and I realized this was the way to go.”

Considering the amount of work that had to go into the new control room, it was done very fast — downtime was about a month. Dan Zellman of Howard Schwartz Studios, New York, an engineer certified in LEDE design and measurement, came up to the studio and did the blueprints in a few days.

Then, the existing room was torn down to the outside walls, an asymmetrical outer shell was put in, and the symmetrical room was built inside of that. The carpentry was handled by two local builders, Alan Souza and Gary Fenster.

“Alan’s an artistic type,” offers Greene. “Most carpenters in a studio situation get blown out by the number of intersecting odd angles they have to deal with.”

“This guy loved what he was doing, especially when things weren’t rectangular. The blueprints were very precise, so there wasn’t much margin for error.”

“We started out with soft fiberglass on the front walls, but it was a little too anechoic for our taste— it soaked up too much sound, and the whole room was a few dB short on level. We replaced it with harder stuff: a thin version of the material they use to insulate boilers.”

The frame itself was heavily overbuilt, with double studding and two layers of sheet rock. “It’s got to be solid,” Greene says. “If the walls move, it defeats the whole purpose.”

Meanwhile, Bob Windsor, one of Normandy’s engineers, was working on the new wiring harness (along with the control room, Normandy was putting in a new MCI JH-600 console and JH-24 multitrack).

It took about 36 hours to install completely the harness and recording equipment after the room was finished. The final step was certification.

The Acid Test
“I’m not sure certification is absolutely necessary,” says Greene, “but if you’re going to go to all that trouble to build it, you might as well go all the way.”

Designer Dan Zellman spent a good day checking out the room, and gave it excellent marks. “He checks the sound coming from the speakers, the acoustical coupling with the room, the anechoic ‘hole’ following the initial blast, then the first reflection and the diffusion.”

“Obviously you can’t get rid of all of the uncontrolled sound, but the concept is valid as long as it’s within certain specs, which are pretty stringent,” says Greene.

In spite of the good evaluation, Greene noticed there were problems with the finished room. “It was a little short on the low-end. The kick drum wasn’t reaching out well.”

“On a hunch, I removed the drivers from the UREI 813A’s we had put in, and replaced them with our old Altec 604-8G’s. Even though they were out of time-sync, it cleared up the problem immediately.

“I was actually pretty upset. I had ordered 813’s, which use essentially the same drivers as our old Altecs, but by the time we took delivery, UREI had stopped shipping them and sent us 813A’s instead. The difference is that the new speakers use ceramic magnets.”

“They’re always talking about how great they are, but the real reason is that AlNiCo, which they used to use, got too expensive.”

“But the ceramic magnets have poor low-end response, and they sound harsh and strident. The speakers measure out the same, but they sound totally different.”

“I had to take the horns off the Altecs and put the little blue UREI horns on them, and I had to buy different crossovers, because the ceramic magnets are shorter, and therefore use a shorter delay. Now they sound great. They’re about 2 dB less efficient, but I can deal with that.*”

As one might expect, Greene is very happy with his new room, and knows why. “It doesn’t wear me out nearly as much — I can work for a long time now,” he offers.

“Since the speakers and the room are all phase coherent, I don’t have to listen to phase distortion, which is very fatiguing. I’m also working 7 or 8 dB softer. I can hear things more clearly at lower levels, which also helps to make it sound better on the street.

“I realized that the other room tended to ‘smear’ the image, which had to do with reflections off the ceiling. It was tough to hear small panpot adjustments.”

“Also, now that I have the new speakers, the room sounds pretty much the same over a wide range of seating positions. Of course, you’re limited by the on-axis response of the tweeters, but I think the room even compensates for that a little.”

“I always tend to listen on headphones before 1 let anything out, and LEDE and cans aren’t really too far away from each other. You can’t really get a good idea of bass on phones, however, and there’s no ambience. So this is the best of headphone-type listening, yet without the drawbacks.”

The new room has changed some of Greene’s work habits. “Things that sound really bad will drive you out of your mind, he concedes. “For example, piano miking that used to sound fine now sounds as if the piano is inside out. You become hyper-aware of phase anomalies.”

“I find that I’m using a lot more coincident-mike placements, and paying a lot more attention to phase coherency. I’m also taking more care with mid-range EC). I can hear the subtleties better; of course, that has a lot to do with the speakers as well as the room. I’m mixing wetter, and I’m using the wall monitors a little more than I used to.”

“The room makes very little difference when you’re near-field monitoring — the primary reflection there is still off the console itself — but I never liked small console speakers anyway. They’re only really effective in the mid- range, and I have three different sets that sound completely different.”

The new room has also necessitated changes in monitor amplification. Because of the reduced efficiency, the old Spectro Acoustics 125 watt per channel power amps were replaced with a Mcintosh Model 2500. An intermediate setup used UREI power amps, but Greene found them short on headroom, and the damping factor to be too high for his speakers.

“The Altec-type woofers are made to move around, and the UREI held the cones too tightly,” he says. “The Mac is transformer-coupled, and although it has a high damping factor for that type of amp, it’s a lot lower than a direct- coupled amp. It’s much nicer to listen to.”

Reaction from Clients
Of course, the most important thing in any studio improvement is how it affects business, which is largely determined by artists’ and producers’ reaction to the change.

“Billy Cobham’s not the kind of guy who gets impressed with engineering concepts like LEDE, but if he takes the tape home and likes the way it sounds, that’s cool,” says Greene.

“If he doesn’t like it, it’s not cool. All the rest of the hype doesn’t matter to him. But most of our clients are long-term, and they’re saying that our mixes are much better. The new clients all like it too, even though they don’t have our old room to compare it to. But they do compare it, very favorably, to other rooms they’ve been in.

“The LEDE concept is so new that even a lot of people in the engineering end of the business aren’t that aware of it, so you can’t expect the creative people to concern themselves with it for some time yet. I think though, that artists and producers will care more in the future, and it will become a major issue.”

And at the bottom line, business for Normandy is up. It will be a while before the magic of a Billy Cobham record or two will draw clients to the studio, and a lot of the increased business was booked before the control-room conversion. But, as Greene says, “they won’t hurt.”

Greene figures that things will get even better, and puts it in this perspective: “With the current economic climate, people want state-of-the-art equipment with good personnel and service that will cost them in the $100 to $125 an hour range, which is where we are.

Fancy rooms will only be for established superstars with huge recording budgets. Otherwise, the record companies don’t want to hear about paying $165 to $200 an hour.

“Our equipment is not vast and awe- inspiring, but it all works, and you can make records in the place. I think that is what’s going to make us successful.”

*Garry Margolis, Sales Director of UREI, comments as follows:

“We noted Mr. Green’s comments on old versus new 813’s with interest. There are a number of objective and subjective differences between the old and new coaxial drivers. The older AlNiCo magnet was subject to partial demagnetization when hit by very heavy transients reproduced by a large power amplifier.”

“This demagnetization lowers the mid-range response of the driver, and, therefore, apparently increases the bass response. The newer ceramic magnet will not be demagnetized in heavy use, and will retain its sound character.”

“The new drivers have a crisper, tighter low-end, which may seem light to someone accustomed to a demagnetized AlNiCo driver. The mid- range response of the new driver has been considerably smoothed, and dis-persion broadened, when compared to the original system.”

“The new horn design uses slots to minimize the shad-owing of mid-band response from the cone, and utilizes a new diffraction buffer and padding in the horn to reduce reflections, improve dispersion, impedance matching, and smooth the out-of- band response.”

“We are soliciting user opinions regarding further improvements to our monitors, and we appreciate Mr. Green’s comments.”

Downloadable Media
Original Article (pdf)

Editor’s Note: This is a series of articles from Recording Engineer/Producer (RE/P) magazine, which began publishing in 1970 under the direction of Publisher/Editor Martin Gallay. After a great run, RE/P ceased publishing in the early 1990s, yet its content is still much revered in the professional audio community. RE/P also published the first issues of Live Sound International magazine as a quarterly supplement, beginning in the late 1980s, and LSI has grown to a monthly publication that continues to thrive to this day.

Our sincere thanks to Mark Gander of JBL Professional for his considerable support on this archive project.

Please send all questions and comments to ProSoundWeb Editor .(JavaScript must be enabled to view this email address).

Posted by admin on 08/13 at 04:35 PM
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Rome Selected As Host City For 134th AES Convention

May 4-7, 2013 At The Roma Eventi – Fontana di Trevi Conference Centre

The Audio Engineering Society has announced that the 134th AES Convention will be held in Rome, May 4-7, 2013.

Set in the heart of the “Eternal City” at the new Roma Eventi – Fontana di Trevi Conference Centre, the convention will be chaired by Umberto Zanghieri, vice president of the AES Southern Europe Region.

The announcement was made by AES executive director Bob Moses.

Recently opened in the heart of Rome, a stone’s throw from the Quirinale Palace and, its namesake the Trevi Fountain, the modern Roma Eventi – Fontana di Trevi Conference Centre is situated in a neoclassical palace built in 1930. 

Designated by Benedict XV as home to the Gregorian University, the Conference Centre covers almost 30,000 square feet, and consists of 15 meeting rooms able to accommodate over 1,000 participants.

The 134th AES Convention will bring together audio engineers from around the world to “Listen, Learn, Connect” and, share the latest knowledge in audio research, development and applications.

This marks the first time an AES Convention will be held in Rome.

The 133rd AES Convention will be held at the Moscone Center in San Francisco, Oct. 26 – 29, 2012.


Posted by Keith Clark on 08/13 at 03:14 PM
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In the Studio: Recording A Great Guitar Sound

It’s not always easy to capture the great sound that you hear in the room
This article is provided by Bobby Owsinski.


Every guitar player, engineer and producer wants to record the ultimate guitar sound, but it’s not always easy to capture the great sound that you hear in the room.

In this excerpt from The Ultimate Guitar Tone Handbook (written with my good friend, television composer and great guitarist Rich Tozzoli), we’ll look different electric guitar mic’ing techniques.

While many believe there’s only one accepted way to mic an amplifier, you’ll be surprised to learn that there are as many ways as there are guitar and amp sounds.

Let’s look at some.

Using A Single Mic

It’s amazing what you can do with a single mic if you experiment a bit. Here are a number of techniques that have been used on popular recordings since the 50s.

They all work, but remember that what works for one recording may not work for another.

That’s why it’s good to always have an alternative in your pocket when you need one.

click to enlarge

The Classic Setup
Place a Shure SM57 about one inch away from the best sounding speaker in the cabinet. Place the mic about three quarters of the way between the edge of the speaker and the voice coil (away from the voice coil).

If you need more high end, move the mic towards the voice coil (the center of the speaker). If the sound needs more body, move it towards the outside edge of the speaker. Make sure that the mic does not touch the speaker cone when the loudest passages are played.

click to enlarge

The Old School Setup

The way amplifier mic’ing was consistently done in the 60s and 70s was to place the mic from one to two feet away from the center of the speaker or speakers. This allows the sound from the speakers and the cabinet to develop, but also captures some of the room, which can be a nice bonus.

The ideal distance on a cabinet with two speakers is where the output of both speakers combine. Move the mic to the side to capture more of the sound of one of the speakers voice coils if more high end is required.

click to enlarge

Mic’ing A Marshall 4x12

If you’re using a Marshall 4x12 cabinet, position a single mic 12 to 24 inches from the cabinet, dead center to all 4 speakers aiming for the logo plate. You can use this for other closed back cabinets as well, except their logo’s might not be in the same position.

On the typical 4x12 speaker cabinet (like the standard Marshall 1960 model), the four speakers usually become additive at a distance of 15 to 24 inches from the cabinet center (depending upon the speakers).

Using Two Mics

As much of a variety as you can get with one mic, you’ll get a lot more with two.

Over the years, many engineers discovered that they could more closely capture the sound that they were hearing in the room by adding a second microphone.

Here are some examples.

click to enlarge

The Classic Two Mic Setup
Place the SM57 near or against the grill cloth as in the classic method #1 above. Now add a Sennheiser MD 421 at the same position to the right of the 57, but aimed at a 45 degree angle pointing towards the voice coil.

Many sounds can be achieved from this setup by summing the mics at different levels and by flipping the phase on one.

Of course, you can use any mics you choose, but the classic setup uses the 57 and 421.

click to enlarge

Two Mic Variation #1

With an open-back amplifier (like a typical Fender), place a mic about a foot away from the rear of the amp, off center from one of the speakers, while using any of the single mic setups for the front of the cabinet.

Usually you’ll have to flip the phase on the rear mic, but try both positions and use the one that has the most low-end.

click to enlarge

Two Mic Variation #2

While using the mic setting from the single mic Classic #1 with the mic close to the grill of the cabinet, add an additional mic at the spot where the sound of the speakers converge 18 to 24 inches away.

This distance might be increased to as much as six feet depending upon the size and sound of the room, which will then increase the captured ambience.

Other guitar amp mic’ing setups can be far more sophisticated using a lot more mics, but one of these methods can get you where you need to go most of the time.

Don’t forget that everything starts with the player, instrument and amplifier first. If the sound is crappy at the amp, there’s nothing you can do to make it sound better via the mic’ing.

Get a great sound first, then pick your choice of mic’ing.

Bobby Owsinski is an author, producer, music industry veteran and technical consultant who has written numerous books covering all aspects of audio recording. For more information be sure to check out his website and blog.

Posted by Keith Clark on 08/13 at 09:30 AM
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Michael Bierylo Named Chair Of Berklee Electronic Production & Design Department

Intends to embrace advances in video game design, software development, and all aspects of computer music and video performance

Berklee College Of Music has announced that Michael Bierylo has been named chair of the Electronic Production and Design Department (EPD).

Bierylo, an electronic musician, guitarist, composer, and sound designer, has been a Berklee faculty member since 1995 and member of the band Birdsongs of the Mesozoic since 1991.

In his new role at the college, he intends to embrace advances in video game design, software development, and all aspects of computer music and video performance.

“Music technology is a moving target, and while new trends tend to disregard what precede them, EPD looks to celebrate all avenues and vintages of electronic expression,” says Bierylo. “We look to both analog and digital systems, lo-fi and hi-fi. Our students design software for iPads and hack Speak and Spells. They create thumping dance tracks, interactive audio-visual installations, and inspired sonic landscapes for video games.”

Bierylo’s commercial work includes music and audio production for Hasbro Interactive, the Smithsonian, Nickelodeon, and the Oxygen Network, as well as music and sound design for the Incredible Hulk Roller Coaster at Universal’s Islands Of Adventure.

As a composer, Bierylo’s work has been featured on A&E’s Biography, the Learning Channel, and Martha Stewart Living. Recent projects include work on the films Granito, the Reckoning, and Traces of the Trade, all featured at the Sundance Film Festival.

Bierylo holds a B.M. from Berklee College of Music and has completed additional studies in jazz composition and audio engineering. A Berklee faculty member for over 17 years, he received the Music Technology Division Excellence in Teaching Award in 2003, and was the 2009 recipient of a Newbury Comics Faculty Fellowship that funded two trips to Berlin to study laptop performance, modular synthesizers, and new music software.

As a member of Birdsongs of the Mesozoic, Bierylo has performed throughout the U.S. at venues as diverse as the Knitting Factory, Honolulu Academy of Arts, Duke and Emory universities, and Dartmouth College.

Bierylo’s compositions are featured on the group’s albums Dancing On A’A, Petrophonics, the Iridium Controversy, and Extreme Spirituals, all on Cuneiform Records. As a solo artist, Bierylo has performed in the U.S. and Berlin, Germany, including a concert with Grammy-nominated electronic musician BT in 2012.

Electronic Production and Design (formerly Music Synthesis) teaches the musical and creative use of electronic production and sound design tools and technologies. Working in professional-level 5.1-equipped studios, classrooms, and labs, students learn electronic composition, synthesizer programming, interactive performance systems, digital signal processing, music with integrated visuals, alternate controllers, and more. The curriculum provides a solid foundation for continued learning and effective performance in a profession that is constantly changing and evolving.

Berklee College Of Music

Posted by Keith Clark on 08/13 at 09:16 AM
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Friday, August 10, 2012

Matt Ward Joins Manley Labs As Strategic Advisor

Veteran of Universal Audio, E-mu Systems, Otari and Studer Revox will develop and implement new strategies for growth

Manley Labs has announced that Matt Ward, former president of Universal Audio, has joined the company as a strategic advisor, where he will work directly with president EveAnna Manley on all aspects of Manley’s business to create strategic goals and a tactical plan to achieve them.

“We’re thrilled to have someone with Matt’s experience and expertise on board,” states EveAnna Manley. “He will help us devise a clear strategy to deal with the challenges Manley faces in the ever-changing audio marketplace.”

“EveAnna and I have been friends for years,” Ward adds, “and we’ve always conferred with each other about the challenges of growing a business so it was natural that when I became a consultant, my first phone call was to EveAnna. 

“Manley is a revered brand with solid products and a stellar reputation for customer service so I’m excited to be joining the team.”

Prior to his 10 years in executive management at UA, Ward worked in product management for other prominent companies in the professional audio and music industries including Studer Revox, Otari, and E-mu Systems.

Manley Labs

Posted by Keith Clark on 08/10 at 02:00 PM
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Five Creative Uses Of Loudspeakers That Can Enhance Recordings

Vibe up dull snare drums, energize a piano, add some "room" to sounds, enhance that kick and build a tunnel

1) Adding More Snares to Snare Drums
If you’re presented with an “inherited recording” to mix (one you didn’t engineer) with live drums where no bottom mic was used on the snare drum, or the track sheet says “snare” but all you’ve got to work with is a dull thump, try this: Route an aux send bus output from your mixing console to a small powered loudspeaker (or, if you have an extra power amp, a regular small passive loudspeaker) you’ve placed out in the studio room or vocal booth.

I’ve done this, putting my small, powered 5-inch Yamaha loudspeaker right on top of a decent sounding snare drum sitting on its stand.

Use a spacer so the loudspeaker itself does not dampen the snare drum head too much.

I used the plastic protective ring from a two-inch reel of tape for a spacer, strapping it and the loudspeaker down to the drum’s shell with gaffer’s tape.

Then I put my favorite bottom snare drum mic on the bottom, and brought it up in the mix on another mic input fader.

While sending on the aux send bus from the original snare track, slowly add in the bottom mic. I sometimes “hard gate” the aux send signal to the loudspeaker if leakage causes too much snare buzzing in between snare hits.

Also be sure to do equalization, and also try flipping phase—one way will sound better than another, and you shouldn’t need much of this to “vibe up” that dull snare.

2) Recording Sympathetic Vibrations
The strings inside a piano can be energized with a loudspeaker as well.

Use non-residue tape to clamp down the piano keys in the key of the song. I usually hold down all the octaves of the key of the song, but you can experiment with forming chords too.

You also need to put something heavy on the sustain pedal to keep the damper off the strings.

If you place the loudspeaker (or attached it) underneath the piano right up under the soundboard, you’ll hear it vibrating the harp and strings.

I usually send the bass, guitars and keyboard tracks, but try sending only vocals for a very interesting vocal effect.

Of course, you’ll want to place a couple of microphones over the harp on the other side of the soundboard and add them to the mix as a stereo pair.

3) Adding Unique Room Ambience
Another trick is to place the loudspeaker out in a room and pick up it’s sound with a mic—a basic echo chamber.

Some mixers routinely set up two loudspeakers with a stereo send and stereo mics just to add more “room” to sounds that are too dry or were recorded direct.

If it’s a good sounding room, this is a winner.

4) Kick Drum Mic
I’ve used old Auratone loudspeakers when recording kick drums - the small five-inch woofer gets a low mid-range sound quality when placed close to the kick drum.

Moving further away produces a more hollow sound good for special effects.

I’ve also used the woofer out of the ubiquitous Yamaha NS-10M since I usually find them around the studio in various stages of disrepair. The 8-inch Yamaha woofer gets a lower tone - almost a TR-808 (drum machine) sound with a lot of ‘hang time’ (that’s ghetto-speak for decay length).

After soldering a mic cable to it, I prop it up on tape boxes and place it about a foot in front at an angle since air blasts from the drum can cause trouble.

These “microphones” are not going to give you anything above about 2 kHz,  so you’ll have to mix in a real microphone for the rest of the drum’s sound.

5) Bass Drum Tunnels
A lot has been written about bass drum tunnels, and I’ve seen a few versions. The usual way is to build a tunnel using standard studio gobos or baffles, which I’ve seen as long as 15 feet.

A long tunnel presents the option of using more distant miking and still maintain isolation from the rest of the kit. Tunnels must have a roof made of more gobos and cartage blankets laid on top of all of it.

I’ve also read that a heavy cylindrical cardboard concrete casting form works as a prefab tunnel, and various diameters of these (up to about 24 inches) can be acquired at home improvement centers in just about any length you want.  They’re much quicker to set up than dragging out all those gobos.

The mic should be put in the normal place—maybe just inside the hole of the front head or right in front of the drum. Then put a distant mic down near the end of the tunnel.

Achieve a balance and sound with just two mics: the distant one down the tunnel, combined with the close mic.

Flip phase around and try different positions before reaching for EQ, and if you’re recording to DAW, you can shift (in time) the distant mic’s recorded waveform closer to the close mic’s signal waveform timing. This sometimes helps and sometimes hurts.

And, I’ve used a shotgun mic for the distant mic for a faraway sound yet sonorous presence.

Have fun!

Barry Rudolph is a veteran L.A.-based recording engineer as well as a noted writer on recording topics. Visit his website at



More Reviews & Articles By Barry Rudolph On PSW:
Does The WAVES Hybrid Line Of Plug-Ins Enhance The Creative Process
The Shure 55 Microphone Has Deep Roots, But How Does It Hold Up Today?
Thumbs Up Or Down For The Marshall MXL V89 Studio Condenser Microphone?
Inside The Peluso P12 Tube Condenser Microphone
Barry’s DAW Toolkit: Review Of The Novation Nocturn With Automap 3 Pro
Barry’s Recording Tips: Figure Of Eight Royer For Electric Guitars
Review Of The X-Tempo Pok DAW Wireless Footswitch Controller
Barry’s Toolkit Of Handy DAW Products
Recording Gear Hits At The 2009 Winter NAMM Show
Working At Recording Success: Taking Elemental Steps Can Make All The Difference
Recording Tip: Successfully Dealing With A Dead Room

Posted by Keith Clark on 08/10 at 01:57 PM
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Genelec Introduces SpeakerAngle App For iOS Devices

New app allows users to correctly configure and match the angling of stereo or surround-sound speaker systems

Genelec has introduced the SpeakerAngle app for iOS devices, which helps users correctly set and match the angling (“toe-in”) of both stereo and 7.1 surround sound loudspeakers.

SpeakerAngle was co-developed by Genelec and AudioApps (a new mobile apps company) and is compatible with iPhone 4 and later, iPad 2 and later and iPhone Touch 4th Generation and later.

In the app, dedicated onscreen loudspeaker icons move as the actual speaker is rotated, while number boxes below each speaker icon continuously display the angle of the speaker.

The number boxes also change color to let users know when their speaker is angled within industry recommendations, and when it is angled to the same degree as the other one in the pair (i.e. left and right in stereo systems; front left / front right, rear left / rear right and side left / side right in 7.1 surround systems).

Detailed information screens provide a tutorial on speaker angling, as well as step-by-step instructions for using SpeakerAngle.

“The new Genelec SpeakerAngle app is a convenient and intuitive way to quickly set and confirm the angle of your speakers in your listening environment,” stated Will Eggleston, Genelec USA marketing director. “This is a perfect tool for home theater owners, recording engineers, system installers and anyone else working to get optimum sound from any loudspeaker system.

“The possibilities are exciting, and we look forward to audio enthusiasts everywhere putting it to good use.”

To use SpeakerAngle, the user simply selects the desired mode of operation (Stereo or Surround), then places the iPhone, iPad or iPod Touch on top of the speaker to be angled.

Set the speaker so that it is facing straight towards the user, then touch the corresponding speaker icon so that it lights up.

Next, touch the icon’s number box to let SpeakerAngle know that the selected speaker is currently at “zero-axis.”

The speaker is then physically rotated inward (that is, towards the listening position). The selected speaker icon will move accordingly, and the number box below it will display the degree to which the speaker is angled.

When the speaker is angled within the industry recommendation of 20 degrees to 45 degrees, the number box changes color, from red to green.

Touch the speaker icon once again (or select any other speaker icon), and the number box changes color to orange, and “freezes” the currently displayed angle.

Next, touch the paired speaker icon (for example, the right speaker in a stereo system if you have just angled the left speaker).

Pick up the iPhone, iPad or iPod Touch and place it on top of that speaker, then touch the number box below the speaker icon. Physically rotate the speaker inwards until the number box changes color to yellow, indicating that this speaker is now angled to the same degree as the first one.

To continue experimenting with different speaker angles, start from scratch at any time by pressing SpeakerAngle’s RESET button.

SpeakerAngle is available now at the iTunes App Store (here), at a cost of 99 cents.


Posted by Keith Clark on 08/10 at 12:53 PM
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Sennheiser Provides Authentic Sound Via Wireless Headphone Technology At Newport Folk Fest

Attendees were able to experience performances from festival artists via Sennheiser RS 120 wireless headphones

Sennheiser, in conjunction with event partner Paste Magazine, recently hosted a live broadcast/recording studio session housed in an alcove of an 18th century fortress, located on the site of this year’s Newport Folk Festival.

By donning a pair of Sennheiser RS 120 wireless consumer headphones, music fans were able to experience performances from festival artists, including Of Monsters and Men, Tom Morello, Jonah Tolchin and 26 others.

Wherever possible, the entire audio chain consisted of Sennheiser related technology — including microphones from subsidiary Neumann and Sennheiser, pre-amplifiers from distributed brand TRUE Systems and wireless RF technology from Sennheiser.

The two-day recording session presented challenges, including a live sound stage located just 100 yards away and an extremely reflective — and somewhat leaky — recording environment.However, the wireless headphones were still able to deliver a clean and quiet performance, faithfully representing the artists’ sounds.

Microphones on stage and in the room included several Neumann TLM 49s and KM 184s, two U 87s and a pair of Sennheiser e 906s. The microphones were connected to a pair of TRUE Systems Precision 8 preamplifiers, which — through a special feature on the back panel of the unit — split the signal and subsequently routed it to both a multi-track recording rig as well as a live mixer.

Each of the Sennheiser wireless headphones received a live stereo mix of the multitrack recording sessions courtesy of Nashville-based engineer Steve Ledet. A Sennheiser A5000CP antenna was strategically placed in the rear of the grotto, providing a generous amount of RF coverage both inside and outside the grotto.

In addition to having many pairs of wireless headphones on hand, Sennheiser set up a special VIP seating area where listeners could audition an assortment of Sennheiser’s audiophile and professional headphones, including the HD 600, HD 650, HD 700, HD 800 and the new Amperior.

“Each of the performers we hosted at the Sennheiser Sound Lounge at the Paste Ruins takes an enormous amount of pride in the craftsmanship and honesty of their songs — this is of paramount importance to them,” says Tim Moore, artist relations manager, Sennheiser. “By selecting a complete signal path of Sennheiser family gear before and after the mixing console, we were able to ensure the integrity of the audio at almost every stage. As a result, the performers were able to establish a more direct and honest connection with their fans.”

Singer/songwriter/guitarist Jonah Tolchin, who performed on the second day of the festival, found the Sennheiser Sound Lounge at the Paste Ruins particularly inspiring. “This is just incredible,” he states. “In this environment, with all these great mics, you are so zoned in and focused with all your heart and soul.”

Clips of the performances will be made available for viewing via Sennheiser’s social media outlets over the next several weeks. 


Posted by Keith Clark on 08/10 at 12:33 PM
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Thursday, August 09, 2012

Hosa Technology 2nd Generation Elite Series Mic Cables Now Shipping

Available with Neutrik XX-Series connectors plus a new nylon webbing over the cable’s PVC jacket, the Elite Series delivers performance attributes that are every bit on par—if not superior to—more costly boutique cables.

Hosa Technology is pleased to announce that a significant upgrade of the company’s popular Elite Series microphone cables is now shipping.

Available with Neutrik XX-Series connectors plus a new nylon webbing over the cable’s PVC jacket, the Elite Series delivers performance attributes that are every bit on par—if not superior to—more costly boutique cables.

Available in both Lo-Z (XLR3F to XLR3M) and Hi-Z (XLR3F to ¼ inch TS) configurations, the cable used in the Hosa Elite Series is a vitally important contributor to overall audio performance.

These cables use 20 AWG Oxygen-Free Copper (OFC) conductors that reduce resistance in order to facilitate maximum signal transfer.

Polyethylene dielectrics reduce capacitance for crystal-clear high-frequency transmission while conductive PVC reduces handling noise.

Further, a 95% OFC braided shield is employed for noise-free signal transmission. Take all this and complete it with nylon webbing over the cable’s PVC jacket, which is cut- and abrasion-resistant for a lifetime of trouble-free use, and the end result is a cable audio professionals can place their trust in.

A clean, reliable connection to one’s mic preamplifier or console input is of paramount concern for any audio engineer or recording enthusiast, and the Hosa Elite Series microphone cable cuts no corners in this regard.

The Neutrik XX-Series connectors employ gold-plated contacts for corrosion resistance and superior signal transfer and utilize a zinc die-cast housing for rock-solid reliability.

With a polyurethane gland to prevent cable kinking for longer cable life and chuck-type strain relief for maximum cable retention combined with a sleek, ergonomic design for easy handling, these connectors deliver the ideal blend of performance and long-term reliability.

“These second generation Elite Series cables offer exceptional audio performance and have been field tested for maximum reliability,” said Jonathan Pusey, Hosa Technology’s Vice President of Sales and Marketing. “With a rich feature set and a highly competitive price, we are extremely optimistic that this product line will find favor with a wide range of customers.

“The Elite Series has always been an exceptional product and now, with the addition of Neutrik connectors and abrasion-resistant nylon webbing for a lifetime of dynamic, noise-free sound quality, I’m confident these mic cables offer the best possible combination of performance and value.”

The Hosa Elite Series microphone cables are available in lengths from 3 to 100 feet and carry MSRP pricing that ranges from $42.75 – $203.55.

The Elite Series is in stock and available now.

Hosa Technology

Posted by Keith Clark on 08/09 at 09:24 AM
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In The Studio: Favorite Ways To Use Compression

Compression can be an insanely versatile tool
This article is provided by Home Studio Corner.


I love compression.

I think it’s is a fabulous tool.

Also, I hate over-compression. (Too much of a good thing and all.)

That said, compression is one of the most enjoyable tools I use during recording and mixing.

And here are some of my favorite uses:

#1 Bass

I haven’t found a bass track that doesn’t love to be squashed. Compression helps even out the performance and create a big, thumpin’ bass track.

#2 Vocals

It’s hard to hear the lead vocal over a nice full mix, no matter how much you adjust the volume. Compression allows you to bring out the consonants in a vocal, so you can actually understand what he/she is singing.

#3 Electric Guitars

I love using compression to add sustain to my big electric guitar tracks. Big guitars have a lot of sustain to begin with, but compression can help ‘em ring out even longer (sounding even more huge than before).

#4 Drums

The sound of kick and snare drum that you hear in modern recordings? It’s hard to get that sound without compression. You can literally change the tone of a snare drum dramatically by simply changing the compressor settings. It’s pretty wild.

I also love using compression on the drum bus. This usually allows me to use less compression on the individual tracks.

#5 Reverb

I’ve only started doing this recently, but in a big mix, sometimes it helps to compress the reverb track. It lets you hear the reverb without having to crank it up and wash out the entire mix.

So, those are five of my favorite uses for compression.

Of course, one of the reasons I really enjoy compression is because I understand how to use it.

And it really is a lot of fun. So much fun, in fact, that I want to add 11 more items to that list.

These are ALL things you can do with a SINGLE compressor.

You can:

—Add punch
—Take away punch
—Add sustain
—“Fix” inconsistent performances
—Bring something to the front of the mix
—Push something to the back of the mix
—“Dirty up” a clean recording
—Add character
—Add snap
—Take away snap
—“Glue” a mix together

As you can see, compression is an insanely versatile tool. And you can use the same, simple, stock compressor that came with your recording software to do all of the cool stuff listed above.

It’s all about how you tweak the settings.

To learn more, go here.

Joe Gilder is a Nashville-based engineer, musician, and producer who also provides training and advice at the Home Studio Corner.Note that Joe also offers highly effective training courses, including Understanding Compression and Understanding EQ.


Posted by Keith Clark on 08/09 at 08:36 AM
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Sennheiser & Full Compass Sponsoring Recording Clinic Led By Leslie Ann Jones Of Skywalker Sound

Grammy Award winning engineer to Illustrate vocal mic techniques and best practices

Sennheiser and Full Compass Systems are co-sponsoring an audio recording clinic on Tuesday, September 11 at the Full Compass facility in Madison, WI, featuring Grammy Award-winning sound engineer Leslie Ann Jones, who will demonstrate vocal recording techniques and cover best practices when recording live vocals in the studio.

Attendees will be provided with a pair of Sennheiser HD 449s, enabling them to monitor both recording and playback.

The event will feature door prizes including a K-array Piccolo audio system, a Neumann TLM 102 microphone and a TRUE Systems P-SOLO microphone preamplifier.

Leslie Ann Jones, who is director of music recording and scoring with Skywalker Sound, has been a recording and mixing engineer for over 30 years.

She began her career at ABC Recording Studios in Los Angeles in 1975 before moving to Northern California in 1978 to accept a staff position at the legendary Automatt Recording Studios. There she worked with such artists as Herbie Hancock, Bobby McFerrin, Holly Near, Angela Bofill, and Narada Michael Walden, and started her film score mixing career with “Apocalypse Now.”

From 1987 to 1997, Jones was a staff engineer at Capitol Studios located in the historic Capitol Records Tower in Hollywood, where she recorded projects with Rosemary Clooney, Michael Feinstein, Michelle Shocked, BeBe & CeCe Winans, and Marcus Miller, as well as the scores for several feature films and television shows.

In 2003, Jones was nominated for a Grammy Award for Best Engineered Recording, Classical, and received a Grammy Award for The Kronos Quartet’s recording of Berg: Lyric Suite, which won Best Chamber Music Album. This year, she won a Grammy Award for Best Engineered Album, Classical for Quincy Porter: Complete Viola Works by Eliesha Nelson & John McLaughlin Williams.

Go here for more information on this event. Seating is limited.

Full Compass Systems

Posted by Keith Clark on 08/09 at 06:51 AM
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Audio-Technica U.S. Appoints Javier Tiburcio As Support Specialist For Mexico, Central America

Will be responsible for providing product training and sales support to A-T distributors and customers

Audio-Technica U.S. has appointed Javier Tiburcio as training & sales support specialist for the territories of Mexico and Central America.

He will be responsible for providing product training and sales support to A-T distributors and customers in Mexico, Central America and South America, as he works as part of the team to grow the brand and increase sales within these regions. The announcement was made by Philip Cajka, Audio-Technica U.S. president and CEO.

Tiburcio has broad experience in microphone sales and training. A native of Acapulco, he has lived and worked his entire career in Mexico, and will continue to be based there. 

His previous positions at Hermes Music and Grupo Imis have included sales and marketing responsibilities for the A-T brand in Mexico.

An experienced musician and audio engineer, Tiburcio studied music at the School of SUTUM, and digital audio and audio for broadcasting at CECAT, gaining expertise in microphone placement for live sound and recording.

He has visited music industry dealers throughout Mexico and conducted clinics and training seminars at a variety of music expos with musicians like Moderatto Drummer Elohim Corona.

“Javier is a tremendous asset for us to expand our sales and training strategies in Mexico, Central America and South America,” said Cajka. “He comes to Audio-Technica with a wealth of experience, and we look forward to all of the great things we can accomplish together.”


Posted by Keith Clark on 08/09 at 06:17 AM
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Furman Now Shipping Contractor Series CN-15MP MiniPort

Extends SmartSequencing technology to components outside the equipment rack

Furman is now shipping its Contractor Series CN-15MP (15A capacity) MiniPort.

Designed for components outside the equipment rack, it combines Furman’s SmartSequencing technology with power protection and optional compatibility with Panamax/Furman’s BlueBOLT hosted remote power and energy management platform.

The CN-15MP features one pair of AC outlets with configurable delay on/off options, while Extreme Voltage Shutdown circuitry protects connected equipment against under/overvoltage conditions.

When connected to a Furman SmartSequencer (CN-1800S or CN-2400S), the CN-15MP’s SmartSequencing technology allows bidirectional, safe sequenced power on/off of remotely located equipment with the simple press of a button or turn of a key.

Remote control/monitoring is available via Panamax/Furman’s BlueBOLT cloud-based platform or third-party control systems when utilizing a BB-RS232 adaptor with the SmartSequencer.

The CN-15MP can also be integrated with legacy (non-Contractor Series) and non-Furman power conditioners/sequencers via remote terminal blocks.

“Packed with features for the ultimate in flexibility, the CN-15MP provides a convenient way to extend the benefits of our SmartSequencing technology to remotely located equipment,” says Dave Keller, senior vice president of sales and marketing for Panamax/Furman. “When combined with our SmartSequencers, installers can also take advantage of our cloud-based BlueBOLT platform for remote outlet control, energy monitoring, email alerts during power events, and more.”

The CN-15MP is available now to Certified Furman Contractor Series resellers. More information is available at


Posted by Keith Clark on 08/09 at 06:09 AM
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Wednesday, August 08, 2012

In The Studio: Synthesizer Basics—Components, Controls, Wave Shapes & More (Includes Audio)

Experiment with combining the various wave shapes and get to know how each one sounds
This article is provided by Audio Geek Zine.


When you first look at a synthesizer, it can be a little intimidating with all the knobs and switches and buttons.

Believe it or not, most synthesizers have the same blocks of components, and once you figure out what they do, you can pretty much use any synthesizer.

The main components you’ll find in a synthesizer:

—Oscillators (Red)
—Mixer (Pink)
—An Envelope Generator (Orange)
—LFO (Yellow)
—Filter (Green)


Additionally you’ll usually find options for:

—Ring Modulation
—Noise Generator
—And More

I’m not going to attempt to cover all that at once, this lesson will cover just a few of these.

Oscillator Section

An oscillator is an electronic circuit that creates a repetitive signal. There are four basic wave shapes that an oscillator will produce.

—Sine Wave
—Triangle Wave
—Square Wave
—Sawtooth Wave


A sine wave is a single tone with no harmonics. The waveform has round peaks and troughs and smoothly changes from positive to negative polarity.

This is a boring sound on its own, but is essential for creating certain sounds.

Here is the sound of a sine wave from a synthesizer. (Listen)

A triangle wave adds odd harmonics to the sine wave. This makes the waveshape linearly alternate positive and negative.

Here is the sound of a triangle wave from a synthesizer. (Listen)

A square wave has only odd harmonics, like the triangle wave. It has instantaneous transitions between high and low levels.

Here is the sound of a square wave from a synthesizer. (Listen)

A Sawtooth wave is named for it’s resemblance to the teeth of a saw. There is a linear rise to the highest value then an instant drop to the lowest level.

A sawtooth wave contains both odd and even harmonics.

Here is the sound of a sawtooth wave from a synthesizer. (Listen)

Because the square wave and sawtooth wave shapes are rich in harmonics, they are a good starting point for creating sounds.

Any one of these wave shapes on their own is not enough to make really interesting sounds which is why you’ll usually find two or more oscillators in a synth.

Mixer Section

You control the blend of the multiple oscillators and other sound generators in the synth in the mixer section. This is most often just a simple knob for each oscillator, turn up the ones you want to hear.

More Oscillator Controls

In addition to the wave shape part of the oscillator section there are also some controls you should be aware of.

There is usually an octave selection for selecting which range the oscillator operates in. There are also pitch controls, usually a semitone adjustment and a fine tune or sometimes labeled detune control.

Here is what it sounds like when you take two identical oscillators set to sawtooth and slightly detune one. (Listen)

The oscillator section may have some other controls for envelopes and LFOs, but I’ll come back to those in another lesson.

I invite you to open up all of your virtual synths and locate the common components. Experiment with combining the various wave shapes and get to know how each one sounds.

More on synthesizers soon…

Jon Tidey is a Producer/Engineer who runs his own studio, EPIC Sounds, and enjoys writing about audio on his blog To comment or ask questions about this article go here.

Posted by Keith Clark on 08/08 at 09:24 AM
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