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Wednesday, May 09, 2012

In The Studio: Top Ways To Help Musicians Hear Themselves

The key is working together creatively

 
I’m often heard saying that the recording engineer’s job is to create an environment conducive to musical creativity and then capture that creativity.

Headphones are usually the only way that a musician will be able to hear themselves and (more importantly) how what they are playing works with the rest of the band.

Every musician will (eventually) ask to hear themselves much louder than everyone else. This makes sense as it will allow them to play the nuances of their instrument.

However, if they are only listening to themselves (or the click track) and not the everything else then what they play may be wonderful by itself but terrible within the mix. They may even compromise the art of their own playing as a result of a poor headphone mix.

For Example:
•—Guitar players who hear themselves too loudly will not “bear down” with the pick as much as they may need to.
•
—Piano players who hear themselves too quietly may not play with the full dynamic range of the piano if they cannot hear themselves play softly.
•
—And finally, any musician that cannot hear the full rhythm will cause a combined pushing and pulling of all the instruments, and no one will be together or “in the pocket,” even if they are overdubbing alone.

Remember, you must make the musicians feel like they are playing together in a room without headphones (in fact I prefer to record bands that way). They have to be able to hear and feel each other clearly.

Sometimes you may have the luxury of multiple headphone feeds, which will allow you to tailor different mixes for the players that require them. Even given the advanced personal mixer technology available today, always be wary of letting musicians mix their own headphones completely by themselves, as they will tend to want to hear only themselves.

A Few Pointers:

1. No matter how loud the drums may be in the room, everyone needs some kick, snare, hat and other drum microphones. The timing and feel of the drum mics will sound different from the drum sound in the room.

2. Panning can be your friend. Sometimes moving some instruments just slightly off center will make it easier to the players to hear themselves without increasing volume or resorting to making the moniutor mix a solo mix for certain individuals.

3. You can always change the sound musicians hear in their headphones without compromising the sounds you record.

Once, I was recording a large horn section that was used to a compressed edgy sound. I wanted to go for something full, so I recorded them using a combination of ribbon and condenser mics going flat from Neve mic pre’s straight into the tape machine.

The section was not happy and complained that the sound was not what they were used to. I did not want to lose the fullness the mics were giving me, so I EQ’ed and compressed the monitor channels coming off the tape machine. Suddenly they were all happy and played well.

When I mixed, I was able to use all of the sounds with absolutels no EQ or compression (until those effects were called for) and was very pleased with the results. If I had changed the sound I was capturing to match what the musicians were used to hearing in their headphones, the final sound of the section would have suffered.

4. Make sure the musicians hear enough of the band and even the beat that they can perform to the song rather than just lay down their parts. Musicians will (and should) be concerned with their performances, but do not let them lose sight of the fact that they are playing within a song along with other musicians.

If they do not hear the others they will not be able to interact with them, even if it is only on a subconscious level.

5. Some drummers will ask for loud click tracks in their headphones. If you have only one headphone feed and the drummer needs to share the cue with other performers, it may be tricky for you to keep everyone happy. You may need to ride the click.

And, speaking of riding the click….

6. Be prepared to ride the click track down in softer sections of a song, especially at the end.

There is nothing worse than trying to mix the very end of a song and having to fade out too quickly to keep the click from the drummer’s headphones from being heard.

Bruce A. Miller is a recording engineer who operates an independent studio and the BAM Audio School website.

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Tuesday, May 08, 2012

Composer-Arranger-Orchestrator Joe Trapanese Scores With ADAM Audio

In just a few years, Joe Trapanese has earned a reputation as a successful composer, arranger, orchestrator and music producer for film, television, multimedia, live theater and concerts.

“Theater work informs my music for film, TV and multimedia, much as the different work I do as a composer, arranger, orchestrator and producer influences each of those roles,” Trapanese states. “The work intermingles and makes everything better. A lot of what I do is problem solving and dealing with clients, and the more experience you have with that, the better off you are.

“My goal is to do what I do in the best possible way,” he continues, “no matter what role I play in the process. It’s just exciting to be in a room with great artists and be called on to do what I do best, which is to blur the lines between categories.”

Collaborating with other artists is a central element that comes up again and again: “I love collaborations because it’s the art of being part of something bigger than yourself. Working with directors, writers and producers elevates my work. I’m like an actor contributing another layer, so cooperation is essential. Musically I have been very fortunate to work with great artists like M83, Mike Shinoda, and Daft Punk.”

Before he sits down with his instruments, Trapanese usually has a general idea of what he’s looking for. “I start with piano and then surround myself with old analog synthesizers and all sorts of modern orchestral sounds and sound libraries that speak back to me creatively,” he explains. “I might have one idea but a certain sound leads me in a different direction and I’ll follow that. In that sense, it’s a very interactive process.”

Besides a brace of synthesizers, computers and software, his studio in Hollywood is equipped with Adam Audio monitors. When recording, he composes with Logic then uses Pro Tools like a tape machine and records it to audio so he can deliver it in the format all of the film and post houses are using. Sometimes he uses an engineer and records with a Euphonics console.

“I’m very dependent on my ADAM monitors because the way they sound is how I will hear things, the most important part of the process. I spent a lot of time listening to all kinds of monitors and eventually found that ADAM helps me deliver all of my ideas in a way that wasn’t possible with other speakers.

“The body of the sound is much better and I can more definition in the lower midrange- with film music, that’s where the body of your music is because that’s what will carry in theaters. It’s below the vocal range where you thrill people. If you have that area of the frequency range under control and you can hear it and work with it effectively, you’re mixes will translate better to the theater. The power and clarity of the ADAM A7Xs is breathtaking.”

Currently, Trapanese is scoring “Tron: Uprising,” an animated series for TV premiering June 7th on Disney XP.  Earlier this year, he contributed arrangements and orchestrations for French musician Anthony Gonzalez and his group M83’s double album “Hurry Up, We’re Dreaming.”

Adam Audio

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In The Studio: Three EQ “Fake-Outs”

Manipulating tracks with EQ for both good and evil purposes

 
We tend to think of EQ as “makeup” for our tracks. We use it to make things purdy.

But EQ can be a pretty handy tool for faking out your listener. And sometimes those fake-outs can be kinda cool.

Here are a few:

1. Fake Depth

Sometimes tracks are recorded so cleanly that they sound too “up front” no matter how much you turn them down.

It’d be nice if you could just make ‘em go sit in the corner.

Well, with EQ you can. Rolling off some high end can make them sound more distant.

It’s like walking to your car when leave a concert early. The farther away you get from the venue, the less highs you hear.

Up next…

2. Fake Tape Saturation

I’ve never owned a tape saturation plugin. (Go ahead, make fun of me.)

I’m not against ‘em, and I’ll probably own one eventually. But for now I’ll fake it.

Now, tape saturation adds extra harmonic content to the signal, which oftentimes softens the sound, removing a little of the harshness from the highs.

I was mastering an EP last week, and the highs were just a little harsh.

Since I didn’t have a tape saturation plugin, I reached my trusty friend, Mr. EQ. I just used an ever-so-gentle filter to roll off just a teensy bit of high frequencies.

And? It softened up the sound and worked nicely.

Now of course the EQ didn’t add any cool harmonics like tape saturation, but it still allowed me to “soften” the sound.

And finally…

3. Fake Reality

This one’s a little odd.

On my last album, there was one piano ballad. The piano was a fake one, a virtual instrument in Pro Tools.

It sounded good, but a little too good.

So I used EQ to make it sound worse, thereby making it sound more real.

Instead of a pristine, crisp, bright piano sound, it sounded a bit muffled and more like an imperfect recording of an actual piano.

And you know how I like imperfection.

So there are three ways to use EQ to fake-out your listeners.

To learn more about how to manipulate your tracks with EQ for both good and evil purposes (muahahaha), head over here: www.understandingEQ.com

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.

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Point Source Audio Conceives Miniature Lavalier Mic From Earworn Series

Point Source Audio has launched the new CO-7L Miniature Lavalier in response to customer demand for a lavalier version of its flagship CO-7 earworn microphone. 

The new CO-7L Miniature Lavalier fills a need for speakers in broadcast or theatre production who generally prefer or need an “invisible” option.  The CO-7L element is extremely small so it can be cleverly hidden in the hairline, behind a button, or even attached to eyeglasses.  It is also water-resistant to be more resilient against humidity and moisture stemming from breath or perspiration.

The CO-7L is a miniature omni-directional condenser lavalier microphone that handles up to 140dB SPL, so it should be the choice for actors, broadcasters, presenters, or pastors looking for maximum intelligibility and clean, accurate reproduction. Additionally, the CO-7L is offered in multiple colors to allow matching or blending to hair, skin or clothing.

“Customers that love our CO-7 Earworn Mic have been asking us for a sister lavalier for quite some time,” said Yvonne Ho, Vice President of Marketing for Point Source Audio. “There are definitely advantages and applications for both an earworn and lavalier; earworn mics are extremely easy to fit, while miniature lavs can be completely camouflaged.” 

The CO-7L Miniature Lavalier is shipping now and retails for $359 MSRP. The Miniature Lavalier is offered in a variety of terminations designed to work with all the most common wireless systems. Standard offerings are immediately available and special orders for custom wiring are welcomed as well.

Popular Standard Offerings:
CO-7L-AK - wired for AKG, available in Beige, Tan, Black
CO-7L-AT - wired for Audio-Technica, available in Beige, Tan, Black
CO-7L-SE - wired for Sennheiser Evolution Series, available in Beige, Tan, Black
CO-7L-SK - wired for Sennheiser SK Series, available in Beige, Tan, Black
CO-7L-SH - wired for Shure, available in Beige, Tan, Black

The CO-7L Miniature Lavalier is available at local pro-sound resellers and system integration contractors.

Point Source Audio

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Monday, May 07, 2012

Patrick Prothe Joins Biamp Systems As Marketing Communications Director

Biamp Systemshas announced that Patrick Prothe has been named the company’s new Marketing Communications Director.

In this role, Prothe will oversee Biamp’s marketing efforts worldwide with a focus on enhancing the company’s online presence, digital marketing assets and creative resources. Prothe will be based in Beaverton, Oregon and will report to Graeme Harrison, Executive Vice President of Marketing, Biamp Systems.

As Marketing Communications Director, Prothe will be responsible for Biamp’s strategic marketing direction, including developing and driving the global marketing plan, overseeing corporate communications, and directing the activities of the company’s external marketing agencies.

“With the continued growth in many of our global markets and the recent introduction of Tesira, it’s vital that our marketing efforts support the needs of our regions,” said Graeme Harrison, Executive Vice President of Marketing, Biamp Systems.  “Patrick is an ideal addition to our team because he understands the importance of closely aligned marketing and sales efforts. Patrick’s extensive experience and knack for creative vision will help us broaden awareness of Biamp and continue to strengthen our leadership position worldwide.”

Prothe comes to Biamp from Viewpoint Construction Software where he was Marketing Communications Manager, responsible for overseeing all marketing and client communications programs including social strategy, content development, public relations, media relations, and trade shows. Prior to this, Prothe served as Manager, Creative Services, at Knowledge Learning Corporation, where he managed corporate branding and creative direction. Prothe has also held various positions at Synesis Design, Xerox and WARN INDUSTRIES.

“Biamp has a solid industry reputation for providing the best support and excellent service to customers,” said Patrick Prothe, Marketing Communications Director, Biamp Systems. “Being able to join this team of smart, sharp-minded individuals is a true privilege. I look forward to working with Graeme and the rest of the Marketing team as Biamp continues to grow, and ensuring that our customers around the world are fully supported by our marketing efforts.”

Biamp Systems

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Audio-Technica Debuts New ATH-ANC9 QuietPoint Active Noise-Cancelling Headphones

Audio-Technica today announced the introduction of its ATH-ANC9 QuietPoint active noise-cancelling over-ear headphones, the company’s new top-of-the-line active noise-cancelling (ANC) model.

The ATH-ANC9 offers new features including exclusive Tri-Level Cancellation selectable noise-cancellation settings, an inline microphone and controller for answering calls and controlling music, and additional enhancements.

The ATH-ANC9 blocks up to 95% of outside noise—the highest ANC performance ever achieved by Audio-Technica QuietPoint headphones, while delivering superlative sound quality.

Audio-Technica’s new Tri-Level Cancellation provides three preset filters for noise reduction of up to 30 dB over a wide range of environmental noise conditions that are experienced in everyday life.
Mode 1 is ideal for use on airplanes, trains and buses and applies maximum noise-cancellation to low frequencies. Mode 2 is designed especially for use in noisy offices and crowded places, and targets midrange frequencies. Mode 3 is best for already-quiet locations like libraries and creates a pristine, peaceful environment ideal for study.

The ATH-ANC9 is the first over-ear QuietPoint model to feature a cable with an inline microphone and controller for answering calls and controlling music. The mic and controller support select products including the iPhone(TM), iPad(R)and many iPod(R)models. The microphone has an omnidirectional pickup pattern (it picks up sound from all directions) and is designed for high-quality, intelligible response, enabling the wearer’s voice to be clearly transmitted without having to speak directly into the mic. The controller enables the user to play or pause music, answer and end calls, and go to the next or previous track.

The ATH-ANC9 has replaceable memory foam earpads for unmatched comfort, and is designed for the exceptional audio quality that Audio-Technica has offered for 50 years. Its precision 40 mm drivers and newly developed electronics provide clear, natural full range sound with authoritative bass, a detailed midrange, smooth, extended treble and precise imaging. The headphones offer an input sensitivity of 100 dB that will provide an ample listening level from portable music sources.

The ATH-ANC9 also works when the noise-cancelling function is turned off, and operates in passive mode without batteries.

The headphones fold flat for storage and come with two detachable cables (with and without inline controller), a 1/4-inch adapter, an airline adapter, a hard carrying case and an AAA battery.

The Audio-Technica ATH-ANC9 QuietPoint active noise-cancelling headphones are available now at a suggested retail price of US$349.95 at http://www.shopaudiotechnica.com, Best Buy Magnolia Design Centers, Airport Wireless stores and other select authorized retailers.

Audio-Technica

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In The Studio: Six Tips For Balancing The Bass And Drum Mix

Most bass drums and bass guitars have plenty of low end and don't need much more

 
Perhaps the most difficult task of a mixing engineer is balancing the bass and drums (especially the bass and kick).

Nothing can make or break a mix faster than the way these instruments work together.

It’s not uncommon for a mixer to spend hours on this balance (both level and frequency) because if the relationship isn’t correct, then the song will just never sound big and punchy.

So how do you get this mysterious balance?

In order to have the impact and punch that most modern mixes exhibit, you have to make a space in your mix for both of these instruments so they won’t fight each other and turn into a muddy mess.

While simply EQing your bass high and your kick low (or the other way around), might work at it’s simplest, it’s best to have a more in-depth strategy, so consider the following:

1) EQ the kick drum between 60 to 120 Hz as this will allow it to be heard on smaller loudspeakers. For more attack and beater click add between 1 kHz to 4kHz. You may also want to dip some of the boxiness between 200 to 500 Hz.

EQing in the 30 to 60 Hz range will produce a kick that you can feel, but it may also sound thin on smaller loudspeakers and probably won’t translate well to a variety of loudspeaker systems. Most 22-inch kick drums are centered somewhere around 80Hz anyway.

2) Bring up the bass with the kick. The kick and bass should occupy slightly different frequency spaces. The kick will usually be in the 60 to 80 Hz range whereas the bass will emphasize higher frequencies anywhere from 80 to 250 Hz (although sometimes the two are reversed depending upon the song).

Shelve out any unnecessary bass frequencies (below 30 Hz on kick and below 50 Hz on the bass, although the frequency for both may be as high as 60 Hz according to style of the song and your taste) so they’re not boomy or muddy. There should be a driving, foundational quality to the combination of these two together. 

A common mistake is to emphasize the kick with either too much level or EQ, while not featuring enough of the bass guitar (see the graphic on the left for a good visual of what it sounds like). This gives you the illusion that your mix is bottom light, because what you’re doing is shortening the duration of the low frequency envelope in your mix.

Since the kick tends to be more transient than the bass guitar, this gives you the idea that the low frequency content of your mix is inconsistent. For pop music, it is best to have the kick provide the percussive nature of the bottom while the bass fills out the sustain and musical parts.

3) Make sure that the snare is strong, otherwise the song will lose its drive when the other instruments are added in.

This usually calls for at least some compression, especially if the snare hits are inconsistent throughout the song.

You may need a small EQ boost at 1 kHz for attack, 120 to 240 Hz for fullness, and 10 kHz for snap. As you bring in the other drums and cymbals, you might want to dip a little of 1 kHz on these to make room for the snare.

Also make sure that the toms aren’t too boomy (if so, shelve out the frequencies below 60 Hz).

4) If you’re having trouble with the mix because it’s sounding cloudy and muddy on the bottom end, mute both the kick drum and bass to determine what else might be in the way in the low end. You might not realize that there are some frequencies in the mix that aren’t really musically necessary.

With piano or guitar, you’re mainly looking for the mids and top end to cut through, while the low-end is just getting in the way, so it’s best to clear some of that out with a hi-pass filter. When soloed, the instrument might sound too thin, but with the rest of the mix the low-end will now sound so much better and you won’t be missing that low end from the other instruments.

Now the mix sounds louder, clearer, and fuller. Be careful not to cut too much from the other instruments, as you might loose the warmth of the mix.

5) For dance music, be aware of kick drum to bass melody dissonance. The bass line over the huge sound systems in today’s clubs is very important and needs to work very well with the kick drum. But if your kick is centered around an A note and the bass line is tuned to A#, it’s going to clash. Tune your kick samples to the bass lines (or vice versa) where needed.

6) If you feel that you don’t have enough bass or kick, boost the level, not the EQ. This is a mistake that everyone makes when their first getting their mixing chops together.

Most bass drums and bass guitars have plenty of low end and don’t need much more, so be sure that their level together and with the rest of the mix is correct before you go adding EQ. Even then, a little goes a long way.

While these aren’t the only mix tips that can help with the bass and drum relationship during your mix (you can check out either The Audio Mixing Bootcamp or The Mixing Engineer’s Handbook for more), they’re a great place to start.

Remember, go easy on the EQ, as a little goes a long way.

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.

 

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Renkus-Heinz Expands Sales Management

Renkus Heinz has announced two major changes to their sales management force.

Industry veteran Phil Van Peborgh has joined the company as Eastern Regional Sales Manager, while Technical Sales Manager Ladd Temple has been promoted to Western Regional Sales Manager.

Van Peborgh comes to Renkus-Heinz after several years as a manufacturers’ representative with Highway Marketing and TechRep. He was also Director of Marketing for social media portal Knetwit.com, and General Manager of video games magazine Polygon.

Prior to moving to the US in 2000, Van Peborgh spent more than a decade working in live sound, touring the US and Europe extensively as a FOH, Monitor and Production Engineer. His resume also includes several years as a systems integrator.

“We’re very excited to welcome Phil to Renkus-Heinz,” said Rik Kirby, VP of Sales and Marketing. “His extensive career experience across a wide range of professional audio positions gives him a unique first-hand perspective into the many different markets in which our products are found, and his proven track record in sales makes him an ideal fit for our continued growth.”

“Renkus-Heinz is not just one of the original loudspeaker manufacturers in our industry, they’re also one of the most innovative,” added Van Peborgh. “Few companies can boast a track record of so many industry firsts, across so many different pro audio sectors. I’m proud to be joining such a forward-thinking team, and am very much looking forward to getting out and meeting our representatives and our end users.” 

Ladd Temple, who joined the company last year as Technical Sales Manager, has been promoted to the position of Western Regional Sales Manager. Temple, who came to Renkus-Heinz after several years with Peavey Electronics, has already been providing technical and sales support to Renkus-Heinz’s extensive representative and dealer network throughout the United States.

“Ladd’s work with our sales reps has not only afforded him an unparalleled technical knowledge of the entire Renkus-Heinz product line, but has fostered relationships that will be invaluable in his new position,” added Kirby.

Van Peborgh’s territory will include the United States east of the Mississippi. He will be working from his offices in Knoxville, TN, and can be reached via email at .(JavaScript must be enabled to view this email address). Temple will now handle all territories west of the Mississippi, and will continue to be based in Texas. He can be reached at .(JavaScript must be enabled to view this email address).

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Friday, May 04, 2012

SSL Appoints Mark Davidson As Global Systems & Solutions Business Development Manager

Solid State Logic has announced that Mark Davidson has joined the company in the newly created role of global systems and solutions business development manager.

Davidson a wealth of industry experience, making him a valuable addition to the SSL team. Based in Munich, Germany, he will be responsible for the SSL global strategic business development strategies in relation to the SSL Integration I/O and Workstation Partner Products (WPP) product portfolio, with a focus on, but not restricted to, the broadcast and system integration markets.

“We searched for a long time to find the right person to represent SSL in the global install and integration markets, so it is simply great to have Mark on the team,” says Jim Motley, head of business, WPP for SSL. “With Mark on board, we have someone with a proven track record of establishing new routes to market.

“This capability is a key element for us and we are confident that he will help develop this growing sector for SSL. I’m confident that Mark will be a useful contact for both our existing partners and new clients alike.”

“I am really excited to join a company with the history and the pedigree of SSL,” states Davidson. “SSL’s reputation for quality and innovation, as well as the respect it has obtained from its peers, made accepting the offer to become part of the global SSL team a very simple decision. The company has some very exciting new products in development and I am looking forward to being part of the continuing SSL success story into the future.”

Davidson joins SSL with extensive sales and marketing as well as strategic business development experience within the broadcast, live performance and fixed installation markets on a pan global scale.

Most recently, he worked for Optocore, based in Munich, Germany, where he was responsible for the global sales and marketing function, as well as managing the back office sales and marketing and technical support teams.

Prior to this, Davidson worked at Clear-Com as regional sales manager for the Eastern United States region from a New York office, where, within a two-year period, he successfully oversaw an increase in turnover by more than double.

Before New York, Davidson was international sales manager for Clear-Com located in Hamburg, Germany, looking after the Eastern Central and Western Europe regions. Previously he also worked for Riedel Communications on a number of very high-profile broadcast communication infrastructure-based projects.

Solid State Logic

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In The Studio: DIY Subkick Microphone

An older but effective trick for kick drums

 
This is an old but very effective trick for mic’ing kick drums.

Take a Yamaha NS10 speaker cone and use that to capture the extra low frequencies of the drum.

Without going into too much theory about this, a dynamic microphone and a speaker are essentially the same thing: they’re both transducers. They take acoustical energy and convert it into electrical energy or vice versa.

So what you do is take the speaker out of the box and solder a male XLR plug on a short cable to the speaker terminals. Pin 2 goes to (+) and Pin 1 goes to (-) pin 3 is not used.

The matter of mounting this speaker to a stand is a different matter, and the main reason to go buy the Yamaha Subkick (pictured below), because of it’s great, easy-to-use mounting system.

That, and it’s also more durable likely than the home version.

One way to do it is to take a standard mic clip apart and fitting the slotted part securely to the corner mounting holes of the speaker; that is, if the speaker you’re using has the 4 corners and not just holes drilled just around the cone [square not a circle]. Or you can attach it to a microphone boom or gooseneck permanently.

The output of the subkick is very hot, meaning you’re going to have to attenuate the signal for it to be of any use to you. An inline -20 dB pad, a pad at the mic pre, or one built into the mic will need to be used.

This guy used a 10k Ohm in series with pin 2 and a 1k Ohm resister across pins 1 and 2 to drop the output about 20 dB.

Mic placement: These work really well at the edge of the drum parallel to the skin. Try it under a floor tom too.

Why the NS10? Most time you see these in a studio it will be with an NS10 cone, but why? From what I’ve been told it is because there are usually extra NS10s lying around a studio, all studios had NS10s, you could predict how it would sound, and they have a frequency response that works well. Don’t know how much truth there is to that.

You can use any speaker you want; it will obviously make a difference in the sound.

Finally, here is a picture I took of one of the two DIY subkicks at Metalworks Studios. Note mounting, placement, and inline pad.

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

 

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Middle Tennessee State University Wins Eighth Annual Shure Scholastic Recording Competition

Shure has announced that a team from Middle Tennessee State University is this year’s Grand Prize Winner of the eighth annual “Fantastic Scholastic Recording Competition.”

The three-student team of Taylor Bray, Jeff Braun, and Grant Hartford—with faculty advisor associate professor Michael Fleming—won this year’s contest with an original composition by aspiring singer-songwriter Rebecca Roubion entitled “Falliday.”

“We congratulate the winning team from MTSU and thank all of the students who participated in this year’s contest,” says Dave Mendez, market development specialist at Shure, who coordinated the competition. “This year’s competition was extremely close, which is a credit to the high quality of all the submissions, and hard work of the students and faculty of these fine recording programs.”

The judges for the competition were Ken Caillat, Leslie Ann Jones, Dave O’Donnell, Keith Olsen, and John Paterno. They evaluated the recordings on their overall fidelity, clarity, and sonic balance as well as creativity in selection and placement of microphones.

“Congratulations to all the participants in this project,” says judge Leslie Ann Jones, director of music recording and scoring at Skywalker Sound. “It is wonderful of Shure to provide such a great opportunity to these teams of soon-to-be engineers and producers, and the results are quite impressive. I was very happy to be involved.”

Each of the 10 student teams worked on a recording project that consisted of tracking and mixing a performance, exclusively using a “microphone locker” provided by Shure for the competition.  Teams submitted an unmastered stereo mix for review by a panel of industry professionals who were invited by Shure to judge the competition.

”We were thrilled to participate in this year’s competition,” notes Fleming. “The student team opened the mic locker like it was a Christmas present, and they really rose to the challenge of using a collection of great microphones, musicians, and acoustic sources to create a unique recording. They learned a lot from the experience and had a great time doing it.” 

Having the microphone locker enabled the students to gain experience with some microphones that none of them had previously used, and to experiment with different mics on different instruments and a variety of microphone placements.

In addition to the winning team from MTSU, there were nine other competing teams from Clemson University, Delta State University, DePaul University, New England School of Communications, The NYU Clive Davis Institute of Recorded Music, Tribeca Flashpoint Media Arts Academy, University of Miami Frost School of Music, University of the Pacific, and William Paterson University.

The runner-up in this year’s competition was the team from The NYU Clive Davis Institute of Recorded Music. The students from Delta State University received an honorable mention.

As the winning school, MTSU takes ownership of a selection of Shure KSM recording microphones, which consists of one KSM313 ribbon microphone, two KSM44A, one KSM42/SG, two KSM32/SL, one KSM141/SL stereo mic pair with A27M stereo mic stand adapter, two SM27-LC, two BETA 181/S, two RPM181/O, and six SRH840 professional monitoring headphones.  The entire prize package is valued at more than $11,000. In addition, each member of the winning team will receive a KSM42/SG, valued at $999 MSRP.

Go here for more information about the winners and to listen to the winning song.

Shure

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Thursday, May 03, 2012

NAMM Announces Activities Of Sixth Annual National Wanna Play Music Week May 7-13

Once again, the National Association of Music Merchants (NAMM) is calling on all aspiring musicians to pick up a musical instrument and play during its sixth annual National Wanna Play Music Week (May 7-13, 2012).

“People spend too much time regretting the fact that they don’t already play a musical instrument, when it’s as simple as picking up an instrument and playing for the pure enjoyment of it,” says Joe Lamond, president and CEO of NAMM. “National Wanna Play Music Week is about finding the inspiration to strum that old guitar or play a keyboard at your local music store. Now’s the time to experience the fun, life-changing benefits of playing a musical instrument.”

Starting May 7, people of all ages and backgrounds can join in celebrating the benefits of playing music through various activities, including:

Monday, May 7: Music Monday–Music Monday kicks off National Wanna Play Music Week with nearly 600,000 students across Canada and the United States joining together by singing and playing instruments in a collaborative show of support for school music education and the fun and benefits of making music. This year, the cities of San Diego and Carlsbad, Calif. will join NAMM staff and special guests at NAMM’s Carlsbad headquarters to proclaim May 7, 2012 as Music Monday and join in the fun of playing music with their community.

Tuesday, May 8: Tech Tuesday–NAMM salutes the leading digital apps, websites and tech products that inspire people to learn how to play a musical instrument. From interactive video games to your personal pocket music teacher, learning to play a musical instrument has never been easier or more fun. Visit http://www.wannaplaymusic.com on May 8 to see which apps, sites and products top the list!

Wednesday, May 9: America’s Favorite Unexpected Musician–It’s surprising how many actors, athletes and politicians also play a musical instrument for fun. Find out which of these “Unexpected Musicians” are America’s favorites on May 9 when NAMM announces the annual public poll results at http://www.wannaplaymusic.com.

Thursday, May 10: Pledge to Play–Found at http://www.facebook.com/wannaplaybynamm, NAMM’s “Pledge to Play” is dedicated to empowering those who have always wanted to learn how to play a musical instrument or sing to take the first step. Starting on May 10, each pledge taker will receive monthly inspirational videos with insightful tips for staying motivated, focused and on a clear path to learning how to play music.

Friday, May 11: National Music Store Weekend–This weekend is designated to start the musical journey with a visit to a local music store. NAMM thanks the thousands of local community music stores who serve as music making’s biggest champions. To find local music retailers, visit: http://www.wannaplaymusic.com/dlocator.

NAMM’s national Wanna Play? public awareness campaign is dedicated to increasing awareness of the proven benefits of playing musical instruments for people of all ages. Since the campaign’s launch in November 2006, the key messages about the fun and many proven benefits of playing music have reached millions of people through national public relations efforts. In addition, Wanna Play? has the support of more than 100 celebrities, including John Stamos, Jack Black, Jeff Daniels, Robert Downey Jr., Band from TV, Orianthi, and Gary Sinise.

NAMM

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Wednesday, May 02, 2012

In The Studio: Mic With Your Ears, Not Your Eyes

When placing microphones, make sure you're placing it where the instrument sounds the best and not where the microphone looks the best.

 
Often, a young engineer will start to position microphones based on what they see done by others or read in a magazine. 

Sometimes they experiment and move the mics to see if the sound improves, but usually once someone ends up with a mic setup they like they stop trying to improve it.

There are certain standard approaches that have been successful, but even these approaches should never be considered “etched in stone”. 

Always experiment, especially if it just means putting up a second mic to try a new position without moving the mic you are already happy with.

Once upon a time I had fallen into a typical routine of going with what I was told worked or what I watched the engineers I had assisted use. 

I was recording piano with a pair of matching mics in an XY pattern around the hammers. I knew of many approaches (another mic at the far end of the piano and then pan that mic over to the bass side of the stereo spread, pair of PZMs taped to the piano lid, throwing mics under, over, and in the holes, etc). 

Sometimes I would use a pair of mics just outside the lid but only when I could get away with more warmth and less percussive clarity.

One day I was working with the talented pianist Warren Wolfe. I was setting up my mics and he said, “You know, nobody ever wants to hear my advice to get the best piano sounds, they always just put mics in the same places.”

I stopped what I was doing, looked him right in the eye and said, “OK, tell me.”

He then said, “All you have to do is to put your head in the piano and listen. Where it sounds good is where you put the microphones.”

So, I moved the mic stands out of the way and listened while he played. Fortunately he played in a way that allowed me to hear how the different sounds from the piano at different ranges and volumes bounced around the piano box…the resonating chamber. 

I then put mics where my right and left ears where (very different from the tight XY I usually used) and played with the angles until I felt they were closer to my actual ear positions.  When I threw up the faders, I was blown away.

The sound was full, and had a more intimate sound than when I used outside mics (click here for an example).  Now I always move my head around inside the piano while the musician played not only wide range material but the actual parts and ranges they would be playing that day. 

Sometimes I went back to the XY over the hammers or pair just outside the box, but in general I always found places in the piano I liked.

I now find it especially helpful to listen to all instruments before placing the mic, often getting weird looks from the musician while I walked around them getting closer and farther and moving my head up and down searching for the sweet spots (you would be surprised there can be more than one, each slightly different). 

Even guitar amps deserve listening to as each speaker sounds slightly different.

Yes, you can accomplish the same thing by having someone moving mics around from an eye driven position while you listen in the control room until your mic hits the sweet spot, but doesn’t it make sense to go find the sweet spots first?

Sometimes you may need to find different spots that emphasize different parts of a sound, or even different parts of a sound that must be captured independently. 

A good example of this is how I record Sanshin, which sounds sort of like a fretless banjo made of snake instead of paper played with rhythmic syncopated notes rather than arpeggios. 

When I walked around and listened while the Rinken Band played, I noticed a spot where it sounded rich, and that within that spot I could easily hear both an attack and a throaty twang. 

To capture that I used a condenser for the highs and an old ribbon for the throaty twang, both in the sweet spot I prefered. 

Rinken told me nobody had every captured the real sound of the Sanshin before. Had I not listened first and in doing so learned what was important to capture I would have ended up with something typical (thin) rather than strong.

In general, you are best off moving your head around the area of an instrument (including above and below, close and far), then placing the microphone where your ear hears the best sound. 

Start with suggested positions, but put your head there and listen before you automatically put a mic there and assume it is the best starting placement.

The key to mic placement is understanding what you are tying to capture, choosing the right mic and finding the location and positioning to most strongly capture the sound source. 

You may have to make sacrifices for the performance (moving the acoustic guitar mic because the musician is wildly throwing his picking arm around) or sacrifices due to available microphones (etc) but you will always capture the music if you mic with your ears instead of your eyes.

Bruce A. Miller is a veteran recording engineer who operates an independent recording studio and the BAM Audio School website.

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Why Quiztones Belongs In Your Studio Or Classroom As A Training Tool

A way to train aural skills that provides immediate feedback

This is a review of Quiztones, frequency ear training apps for Mac & iOS from Audiofile Engineering.

As engineers, we all have particular strengths and weaknesses. Some are musically gifted and play multiple instruments, while others naturally take to composition.

However, what about the most basic of skills - our hearing? Unless you have absolute pitch or synesthesia, we’re all playing with the hand dealt to us a birth. The only thing we can do is hone our auditory perception.

That’s why so many forms of ear training for musicians and eventually engineers have evolved over the years. Because, according to Quesnel & Woszczyk, “there is substantial evidence…that auditory perceptual skills can be improved by controlled practice and training.”[1]

Auditory perception is one of the most basic skills required of audio engineers as we go about our daily tasks of balancing, treating, and mixing audio. Therefore, providing new ways for engineers (especially students) to develop auditory skills is critical. Thankfully, Audiofile Engineering has created a Mac and iOS based ear training program for audio engineers, Quiztones, which does a fantastic job at helping the listener develop more acute listening and frequency recognition skills.

Deep Background: Why A Change Is Necessary

First, to understand why any one solution is ideal, it’s helpful to understand just a tiny bit of history. As audio engineers, we’ve always had some genuinely useful auditory training resources available to us like Dave Moulton’s Golden Ears, F. Alton Everest’s Critical Listening Skills for Audio Professionals (Thomson Course Technology), and even Jason Corey’s Audio Production and Critical Listening (Focal Press). Each one of these is a valuable tool on their own, however they’re a very passive way of learning.

That is to say, learners would read the text and then take auditory quizzes, which then required manual grading. However, educators across all content areas today recognize the value of learning technology within the classroom, which includes a broad range of communication and related technologies used to support learning, teaching, and assessment.[2] So, why not bring this into audio education? A wonderful parallel example of this comes to us from music education in the use of MacGamut, which allows for mastery-based drill and practice in Aural Skills of Intervals, Scales, Chords, and much more.[3]

As is illustrated by the success of MacGamut in music curricula, auditory training too must consist of truly interactive learning technology for learners to benefit the most, and this is precisely what Quiztones has accomplished.

The Solution: Quiztones

Quiztones has overcome the shortcomings of previous auditory training resources because it is a truly interactive training resource, presenting learners with auditory examples, multiple answers, and real-time feedback in the form of weighted grading. Interestingly, the product was born out of an undergraduate music production program internship and is the product of someone who understands exactly the needs of undergraduate music students. Both the iOS and Mac OS X versions contain the following trainers, which are truly impressive:

EQ Quizzes:

—Easy Frequency Boost (+10 dB)
—Hard Frequency Boost (+5 dB)
—Hard Frequency Cut (-10 dB)
—Expert Frequency Boost (+5 dB) – 1/3 Octave*

Tone Quizzes:

—Easy
—Hard

Gain Quizzes:

—Easy*
—Hard*

* Included in Mac App/In-App Purchase in iOS App

Over the course of using Quiztones, I was positively blown away. Initially, I didn’t perform as well as expected on some quizzes, however with consistent practice I’m pleased to say I’m now performing at the level I’d anticipated. This only further supports the already sound evidence that consistent practice at auditory drills will yield a dramatic improvement. And really, isn’t that what we’re all after?

However, what’s most impressive is the road ahead for Quiztones. In speaking with the apps’ creator, Dan Comerchero, it’s clear he intends this app to benefit engineers of all ages; whether they’re a seasoned pro looking to brush up their skills or a student just beginning ear training. This is evidenced by the development road-map which includes the addition of a practice mode to both apps, as well as content additions like reverb, delay, and compression trainers which will truly make the tool “feature complete” even when compared to the current industry standard of ear training, Golden Ears.

The most interesting thing Dan revealed to me, however, is the current development of a product called “Quiztones Author”. This is a utility which will give educators the ability to customize quizzes based upon the needs of their students and curricula. Currently in beta, this will be a separate utility that educational institutions can purchase as an additional tool for their faculty.

The creation of this utility is significant because it will provide university faculty the ability to utilize Quiztones directly as a part of their curricula; allowing the design, distribution and retrieval of scores for quizzes instead of simply recommending the tool as a supplement to classroom instruction. If you are an educator interested in beta testing Quiztones Author, contact dan via e-mail .(JavaScript must be enabled to view this email address).

Final Thoughts

You might be asking yourself what can “Quiztones really offer me? It’s seems too educational” or even “Why do we need to improve the old systems that were working so well”. To be honest, those sentiments and many more are perfectly understandable. We’re a legacy industry that doesn’t often accept change easily.

However, the reason is quite simple; everyone benefits from having better training tools, and the fact that Quiztones is built upon solid educational theory is only one of a dozen reasons to adopt it within your training regimen. Every engineer knows that better frequency recognition helps him or her in the development and discussion of sonic ideas, so why not train and improve aural skills with a system that provides immediate feedback?

And, fundamentally, fast frequency recognition helps engineers decide how to react if, for example, they hear X problem in the Y frequency band. So, using a system that helps engineers improve their accuracy over time with varied scenarios in a controlled environment is a tremendous asset.

Can I say that Quiztones is the absolute perfect aural training solution for you? Perhaps not quite yet, as I’d love to see more options in the quiz answers, and I think a “Match the Sound” style trainer would be incredible. Audiofile Engineering tells me this is the direction Quiztones is headed: hearing a modified audio loop and letting users utilize on-screen controls to try and match the modified sound while receiving feedback on accuracy. However, I can say without reservation that Quiztones is by far the best aural training solution currently available, and I urge you to give it a try in your studio or classroom curricula. I’m certain you find the tool worthwhile.

Quiztones is developed by Audiofile Engineering. Quiztones for Mac is available at the Mac App Store or Audiofile Engineering Store. Quiztones for iOS is available at the iOS App Store.

References
1. René Quesnel and Wieslaw R. Woszczyk, ‘A Computer-Aided System for Timbral Ear Training’, Audio Engineering Society Convention 96, 1994. (link)

2. What is Learning Technology? The Association for Learning Technology. (link)

3. What is MacGAMUT? MacGAMUT - Music Software International. (link)

Kyle P. Snyder is an audio engineer with innumerable credits in the public and private sector, writing about audio engineering, recording technology, and audio education. Find out more about Kyle on his website or follow him on twitter.

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The Audio Expert: Audio Fidelity, Measurements, And Myths—Part 1

Only four parameters are needed to define everything that affects the fidelity of audio equipment

Here we present a portion of a chapter in the new book “The Audio Expert” by Ethan Winer, published by Focal Press.

———————————-

“Science is not a democracy that can be voted on with the popular opinion.” — Earl R. Geddes, audio researcher

In this chapter I explain how to assess the fidelity of audio devices and address what can and cannot be measured. Obviously, there’s no metric for personal preference, such as intentional coloration from equalization choices or the amount of artificial reverb added to recordings as an effect. Nor can we measure the quality of a musical composition or performance.

While it’s easy to tell—by ear or with a frequency meter—if a singer is out of tune, we can’t simply proclaim such a performance to be bad. Musicians sometimes slide into notes from a higher or lower pitch, and some musical styles intentionally take liberties with intonation for artistic effect.

So while you may not be able to “measure” Beethoven’s Symphony #5 to learn why many people enjoy hearing it performed, you can absolutely measure and assess the fidelity of audio equipment used to play a recording of that symphony. The science of audio and the art of music are not in opposition, nor are they mutually exclusive.

High Fidelity Defined

By definition, “high fidelity” means the faithfulness of a copy to its source. However, some types of audio degradation can sound pleasing—hence the popularity of analog tape recorders, gear containing tubes and transformers, and vinyl records. As with assessing the quality of music or a performance, a preference for intentional audio degradation cannot be quantified in absolute terms, so I won’t even try. All I can do is explain and demonstrate the coloration added by various types of audio gear and let you decide if you like the effect or not.

Indeed, the same coloration that’s pleasing to many people for one type of music may be deemed unacceptable for others. For example, the production goal for most classical (and jazz or big band) music is to capture and reproduce the original performance as cleanly and accurately as possible. But many types of rock and pop music benefit from intentional distortion ranging from subtle to extreme.

“The Allnic Audio’s bottom end was deep, but its definition and rhythmic snap were a bit looser than the others. However, the bass sustain, where the instrumental textures reside, was very, very good. The Parasound seemed to have a ‘crispy’ lift in the top octaves. The Ypsilon’s sound was even more transparent, silky, and airy, with a decay that seemed to intoxicatingly hang in the air before effervescing and fading out.” — Michael Fremer, comparing phonograph preamplifiers in the March 2011 issue of Stereophile magazine

Perusing the popular hi-fi press, you might conclude that the above review excerpt presents a reasonable way to assess and describe the quality of audio equipment. It is not. Such flowery prose might be fun to read, but it’s totally meaningless because none of those adjectives can be defined in a way that means the same thing to everyone. What is rhythmic snap? What is a “crispy” lift? And how does sound hang in the air and effervesce?

In truth, only four parameters are needed to define everything that affects the fidelity of audio equipment: noise, frequency response, distortion, and time-based errors. Note that these are really parameter categories that each contain several subsets. Let’s look at these categories in turn.

The Four Parameters

Noise is the background hiss you hear when you raise the volume on a hi-fi receiver or microphone preamp. You can usually hear it clearly during quiet passages when playing cassette tapes. A close relative is dynamic range, which defines the span in decibels (dB) between the residual background hiss and the loudest level available short of gross distortion.

CDs and DVDs have a very large dynamic range, so if you hear noise while playing a CD, it’s from the original master analog tape, it was added as a by-product during production, or it was present in the room and picked up by the microphones when the recording was made.

Subsets of noise are AC power-related hum and buzz, vinyl record clicks and pops, between-station radio noises, electronic crackling, tape modulation noise, left-right channel bleed-through (cross-talk), doors and windows that rattle and buzz when playing music loudly, and the triboelectric cable effect. Tape modulation noise is specific to analog tape recorders, so you’re unlikely to hear it outside of a recording studio.

Modulation noise comes and goes with the music, so it is usually drowned out by the music itself. You can often hear it on recordings that are not bright sounding, such as a bass solo, as each note is accompanied by a “pfft” sound that disappears between the notes. The triboelectric effect is sometimes called “handling noise” because it happens when handling poor-quality cables. The sound is similar to the rumble you get when handling a microphone. This defect is rare today, thanks to the higher-quality insulation materials used by wire manufacturers.

Frequency response describes how uniformly an audio device responds to various frequencies. Errors are heard as too much or too little bass, midrange, or treble. For most people, the audible range extends from about 20 Hz at the low end to slightly less than 20 KHz at the high end. Some youngsters can hear higher than 20 KHz, though many senior citizens cannot hear much past 12 KHz.

Some audiophiles believe it’s important for audio equipment to pass frequencies far beyond 20 KHz, but in truth there’s no need to reproduce ultrasonic content because nobody will hear it or be affected by it. Subsets of frequency response are physical microphonics (mechanical resonance), electronic ringing and oscillation, and acoustic resonance. Resonance and ringing will be covered in more detail later in this and other chapters.

Distortion is a layman’s word for the more technical term nonlinearity, and it adds new frequency components that were not present in the original source. In an audio device, non-linearity occurs when a circuit amplifies some voltages more or less than others, as shown in Figure 2.1. This nonlinearity can result in a flattening of waveform peaks, as at the left, or a level shift near the point where signal voltages pass from plus to minus through zero, as at the right. Wave peak compression occurs when electrical circuits and loudspeaker drivers are pushed to levels near their maximum limits.

Some circuits compress the tops and bottoms equally, which yields mainly odd-numbered harmonics—3rd, 5th, 7th, and so forth—while other circuit types flatten the top more than the bottom, or vice versa. Distortion that’s not symmetrical creates both odd and even harmonics—2nd, 3rd, 4th, 5th, 6th, and so on. Crossover distortion (shown in Figure 2.1) is also common, and it’s specific to certain power amplifier designs. Note that some people consider any change to an audio signal as a type of distortion, including frequency response errors and phase shift. My own preference is to reserve the term “distortion” only when nonlinearity creates new frequencies not present in the original.

When music passes through a device that adds distortion, new frequencies are created that may or may not be pleasing to hear. The design goal for most audio equipment is that all distortion be so low in level that it can’t be heard. However, some recording engineers and audiophiles like the sound of certain types of distortion, such as that added by vinyl records, transformers, or tube-based electronics, and there’s nothing wrong with that. My own preference is for gear to be audibly transparent, and I’ll explain my reasons shortly.

The two basic types of distortion are harmonic and intermodulation, and both are almost always present together.

Harmonic distortion adds new frequencies that are musically related to the source. Ignoring its own inherent overtones, if an electric bass plays an A note whose fundamental frequency is 110 Hz, harmonic distortion will add new frequencies at 220 Hz, 330 Hz, 440 Hz, and subsequent multiples of 110 Hz. Some audio devices add more even harmonics than odd, or vice versa, but the basic concept is the same.

In layman’s terms, harmonic distortion adds a thick or buzzy quality to music, depending on which specific frequencies are added. The notes created by most musical instruments include harmonics, so a device whose distortion adds more harmonics merely changes the instrument’s character by some amount.

Electric guitar players use harmonic distortion—often lots of it—to turn a guitar’s inherent plink-plink sound into a singing tone that has a lot of power and sustains.

Intermodulation distortion (IMD) requires two or more frequencies to be present, and it’s far more damaging audibly than harmonic distortion because it creates new sum and difference frequencies that aren’t always related musically to the original frequencies.

For example, if you play a two-note A major chord containing an A at 440 Hz and a C# at 277 Hz through a device that adds IM distortion, new frequencies are created at the sum and difference frequencies:

Sum: 440 Hz + 277 Hz = 717 Hz
Difference: 440 Hz + 277 Hz = 163 Hz

717 Hz is about halfway between an F and F# note, and 163 Hz is slightly below an E note. Neither of these are related musically to A or C#, nor are they even standard note pitches. Therefore, even in relatively small amounts, intermodulation distortion adds a dissonant quality that can be unpleasant to hear. Again, both harmonic and intermodulation distortion are caused by the same nonlinearity and thus are almost always present together. What’s more, when IM distortion is added to notes that already contain harmonics, which is typical for all musical instruments, sum and difference frequencies related to all of the harmonics are created, as well as for the fundamental frequencies.

Another type of distortion is called aliasing, and it’s unique to digital audio. Like IM distortion, aliasing creates new sum and difference frequencies not harmonically related to the original frequencies, so it can be unpleasant and irritating to hear if it’s loud enough. Fortunately, in all modern digital gear, aliasing is so low in level that it’s rarely if ever audible. Aliasing artifacts are sometimes called “birdies” because difference frequencies that fall in the 5
10 KHz range change pitch in step with the music, which sounds a little like birds chirping. An audio file letting you hear what aliasing sounds like is in Chapter 3.

Transient intermodulation distortion (TIM) is a specific type of distortion that appears only in the presence of transients—sounds that increase quickly in volume such as snare drums, wood blocks, claves, or other percussive instruments. This type of distortion may not show up in a standard distortion test using static sine waves, but it’s revealed easily on an oscilloscope connected to the device’s output when using an impulse-type test signal such as a pulse wave.

TIM will also show up as a residual in a null test when passing transient material. Negative feedback is applied in amplifiers to reduce distortion by sending a portion of the output back to the input with the polarity reversed. TIM occurs when stray circuit capacitance delays the feedback, preventing it from getting back to the input quickly enough to counter a very rapid change in input level. In that case the output can distort briefly. However, modern amplifier designs include a low-pass filter at the input to limit transients to the audible range, which effectively solves this problem.

Time-based errors are those that affect pitch and tempo. When playing an LP record whose hole is not perfectly centered, you’ll hear the pitch rise and fall with each revolution. This is called wow. The pitch instability of analog tape recorders is called flutter. Unlike the slow, once per revolution pitch change of wow, flutter is much faster and adds a warbling effect.

Digital recorders and sound cards have a type of timing error called jitter, but the pitch deviations are so rapid they instead manifest as added noise. With all modern digital audio gear, jitter is so soft compared to the music that it’s almost always inaudible.

The last type of time-based error is phase shift, but this too is inaudible, even in relatively large amounts, unless the amount of phase shift is different in the left and right channels. In that case the result can be an unnaturally wide sound whose location is difficult to identify.

Room acoustics could be considered an additional audio parameter, but it really isn’t. When strong enough, acoustic reflections from nearby boundaries create the comb filtered frequency response described in Chapter 1. This happens when reflected sound waves combine in the air with the original sound and with other reflections, enhancing some frequencies while canceling others.

Room reflections also create audible echoes, reverb, and resonance. In an acoustics context, resonance is often called modal ringing at bass frequencies, or flutter echo at midrange and treble frequencies. But all of these are time-based phenomena that occur outside the equipment, so they don’t warrant their own category.

Another aspect of equipment quality is channel imbalance, where the left and right channels are amplified by different amounts. I consider this to be a “manufacturing defect” caused by an internal trimmer resistor that’s set incorrectly, or one or more fixed resistors that are out of tolerance. But this isn’t really an audio parameter either, because the audio quality is not affected, only its volume level.

The preceding four parameter categories encompass everything that affects the fidelity of audio equipment. If a device’s noise and distortion are too soft to hear, with a response that’s sufficiently uniform over the full range of audible frequencies, and all time-based errors are too small to hear, then that device is considered audibly transparent to music and other sound passing through it. In this context, a device that is transparent means you will not hear a change in quality after audio has passed through it, even if small differences could be measured.

For this reason, when describing audible coloration, it makes sense to use only words that represent what is actually affected. It makes no sense to say a power amplifier possesses “a pleasant bloom” or has a “forward” sound when “2 dB boost at 5 KHz” is much more accurate and leaves no room for misinterpretation.

Chapter 1 explained the concept of resonance, which encompasses both frequency and time-based effects. Resonance is not so much a parameter as it is a property, but it’s worth repeating here. Resonance mostly affects mechanical transducers—loudspeakers and microphones—that, being mechanical devices, must physically vibrate. Resonance adds a boost at some frequency and also continues a sound’s duration over time after the source has stopped. Resonance in electrical circuits generally affects only one frequency, but resonances in rooms occur at multiple frequencies related to the spacing between opposing surfaces. These topics will be examined in more depth in the sections that cover transducers and room acoustics.

When assessing frequency response and distortion, the finest loudspeakers in the world are far worse than even budget electronic device. However, clarity and stereo imaging are greatly affected by room acoustics. Any room you put the speakers in will exaggerate their response errors further, and reflections that are not absorbed will reduce clarity. Without question, the room you listen in has much more effect on sound quality than any electronic device.

However, the main point is that measuring these four basic parameters is the correct way to assess the quality of amplifiers, preamps, sound cards, loudspeakers, microphones, and every other type of audio equipment. Of course, to make an informed decision, you need all of the relevant specs, which leads us to the following.

“The Audio Expert” by Ethan Winer, published by Focal Press (ISBN: 9780240821009), is available here.

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