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Tuesday, May 15, 2012

PWS Walks The Red Carpet AT Billboard Latin Music Awards

Professional Wireless Systems (PWS) has been providing wireless solutions and gear onsite at the Billboard Latin Music Awards for the past 13 years.

This year the PWS team, including Brooks Schroeder and James Stoffo, created a custom wireless communications package for the show, with the company’s frequency coordination team working backstage to ensure all wireless operations went off without a hitch.

“This year’s package included all of the wireless microphones, in-ear monitors and IFBs for the technical production and musical elements, as well as the wireless communication for the production crew for both the main show and pre-awards red carpet show,” says Brooks Schroeder, project manager, PWS. 

Since 1999, The Billboard Latin Music Awards ceremony has been broadcast on the Telemundo television network, where it has become the network’s highest-rated music special.

For the main show, PWS used the new Shure Axient wireless microphone system along with the Shure PSM1000 personal monitor system.

“The Axient Wireless microphone system provides us with an innovative and fail-safe system,” adds Schroeder. “Features like the advanced planning, setup and control capabilities of the Axient make it an extremely reliable product, especially in the very populated Miami television environment.”

The PSM 1000 brings personal monitoring to its most advanced level yet. The PSM1000’s diversity bodypack receiver is ideally suited for large awards shows and other special events with a high noise floor from LED walls and the other wireless systems.

According to Schroeder, “We chose to use the PSM 1000s because it outperforms the other equipment on the market and comes loaded with operational features that no other equipment has. This is critical for live broadcast because you want to hedge your bets as much as you can by using the most reliable equipment available. This equipment allowed us to focus on all of the other things we needed to do.”

Additionally, for the pre-show red carpet activities, PWS’ John Garrido utilized Shure UHF-R wireless microphones and BTR 800 intercoms. In addition to the equipment provided, frequency coordination was another large part of the service PWS performed for the show. During the festivities, PWS’ team of experts was busy monitoring the RF spectrum to ensure there were no issues with the large amount of media covering the event. 

“Trying to manage the local and international press that are in attendance and make sure they do not interrupt the frequencies used for the shows is a big challenge,” concludes Schroeder. “Our goal is to have a perfect, interference-free show every time and we do whatever it takes to accomplish that.

“By using customized gear and bringing in our filtered products, RF monitors and antennas, we are able to pull off these large shows. This year’s Billboard Latin Music Awards was another successful event for us.”

Professional Wireless Systems

Clear-Com Introduces New CC-300 And CC-400 Headsets

Clear-Com introduces the latest additions to its line of professional headsets, the CC-300 and CC-400.

The CC-300 and CC-400 offer extra comfort, better performance and more flexibility for users who employ intercoms for long hours. They are compatible with Clear-Com’s wide range of wired and wireless intercoms.

Wearing headsets day in and day out while communicating with other team members over the intercom can quickly exhaust a user. The new Clear-Com CC-300 single-ear headset and CC-400 double-ear headset were designed to reduce fatigue and accommodate individual preferences by offering a clear and comfortable audio experience.

The CC-300 and CC-400 headsets’ microphone booms can be rotated 300 degrees, allowing the microphone to be worn on the right or left side of the head. Users can also make the headsets larger or smaller by manipulating the slide adjustments on either side of the headband. In addition, the enclosed headphones have very soft padding and a slight rotation to provide a better fit.

The CC-300 and CC-400 headsets are equipped with hyper-cardioid dynamic microphones and high ambient-noise attenuation headphones that deliver balanced audio performance to the user. The acoustic isolation capability significantly reduces external background noise and with a clear audio profile, including an up to 20-kHz frequency response, the new headsets are made for professionals who demand high-quality sound.

The CC-300 and CC-400 headset microphones can be turned on and off by moving the microphone boom. Users need only to pull the boom gently downwards to turn on the microphone and push the boom gently upwards to turn it off, in effect giving them a quick and simple mute to their intercom system. 

Interchangeable cabling is another major benefit of the CC-300 and CC-400 headsets. Both headsets come with a standard four-pin female connector, but users can easily change the cable and connector at the base of the headset by using a Phillips screwdriver in order to accommodate different connector types. The same process can be used to repair and replace a damaged cable on the fly.

“Busy crew members need a headset that they can slip on and immediately feel the difference in fit and performance,” says Stephen Sandford, Product Manager, Clear-Com. “The Clear-Com CC-300 and CC-400 deliver by providing long-lasting comfort and high-quality audio, and can be used in conjunction with virtually all of our intercom solutions.”

Clear-Com

Full Compass Hosts “Music Biz” Workshop For Young Musicians

Full Compass Systems will host a day-long workshop for aspiring musicians on Saturday, June 23rd 2012. 

“Making It in the Music Biz” is a special gathering of music professionals who will share insights and advice on the practical and business aspects of performing music.  The event, based on the popular Launchpad music program, is free to all participants.

“Launchpad celebrates the reach of music education, which is exactly what this event is designed to do.  While it targets youth, musicians of all skill levels, ages and genres are invited to attend for free and learn tips and tricks from the music professionals,” said Tim Wurgler, WSMA program director. 

“Making It in the Music Biz” features notable Yamaha recording artists and other music business professionals, including Full Compass staff.

The program covers a variety of topics: guitar and drum clinics; how to avoid costly mistakes in the studio and tips for recording; ways to get your band noticed; stepping it up with social media and online resources; the process of song writing; making music that makes money; making the most of practicing, rehearsing and performing; and panel discussions on publishing, royalties, copyright, contracts, artist management and touring.

Susan Lipp, who owns Full Compass Systems with her husband Jonathan Lipp, stated, “Jonathan and I are long-time passionate supporters of school music programs. In fact, we have recently returned from our seventh trip to Washington, D.C. lobbying for desperately needed funding. Hosting a workshop for these amazingly talented young musicians is just another way we can offer our support.”

Sponsors include Yamaha, Charter Communications, Nicholas Family Foundation, the Les Paul Foundation, Tilt Media, Sherwood Press, WTDY AM, WJJO FM, HOT JAMZ, Maximum Ink and the Onion.

Full Compass Systems

Allen & Heath GS-R24 Installed At COPA In South Africa

Allen & Heath’s GS-R24 Firewire recording mixer has been installed in the main recording studio at the Campus of Performing arts (COPA) in Johannesburg, South Africa, as a teaching console and to manage music students’ projects.

“I needed a mixer to satisfy my students’ obsession for all things digital, and to teach them the value of what they call ‘old school’.” explains COPA’s head of production, Neill Pash. “Enter the GS-R24, new kid on the block and what I believe to be the future of modern recording consoles.”

He continues, “Everything the modern day producer and musician needs has been included in this little gem: musical fully parametric EQ, two great-sounding onboard valve pre-amps which also serve as instrument DI’s, with onboard MIDI control of your DAW including transport and plug-ins, automated flying faders (GSR24-M) and the flexibility to route your DAW effects or stereo outputs to the two stereo returns, or route your outboard effects, is just brilliant.”

The mixer has already been employed to manage recording projects and demos of over 400 students with music ranging from metal, rock & pop, hip hop to jazz. The students particularly enjoy the impressive DAW and MIDI control features for producing drum & bass and electronic genre, mixed in the box using the channel faders as controllers, back to the DAW, then back again through the onboard valve preamps.

“This console is flexible and intuitive to use, has everything you need for today’s recording productions, and most of all, and as you would expect from Allen & Heath, it just sounds great. The students are so impressed many have bought the GSR’s little sister, the ZED-R16 Firewire for their home studios. The GSR is truly an all-round workhorse for any production studio,” concludes Pash.

http://www.allen-heath.com” title=“Allen & Heath” target=“Blank”>Allen & Heath

Rational Acoustics Announces Upcoming Smaart Training Classes

Rational Acoustics has announced several new Smaart classes in its 2012 training schedule.

May 21-22, 2012 in Kerkrade, Germany.  This 2-day Fundamentals & Applications class will be hosted by Rational distributor AudioTec at the AD Systems facilities. It will be taught in German by Michael Haeck. Interested attendees will need to contact AudioTec directly for pricing, registration and logistic information by e-mailing .(JavaScript must be enabled to view this email address).

May 22-24, 2012 in Paris, France.  This 2-day Fundamentals & Applications class will be hosted by Rational distributor Haliotis and will be taught in French. Instructor TBD. Interested attendees will need to contact Haliotis directly for pricing, registration and logistic information by e-mailing .(JavaScript must be enabled to view this email address).

May 29-31, 2012 in Toronto, Canada.  This class will be hosted by Rational distributor SF Marketing and will be taught by Arthur Skudra, following the standard 3-day training format with Days 1 and 2 covering Smaart Fundamentals and Applications and Day 3 as the optional Practicum.  Interested attendees will need to contact SF Marketing directly for pricing, registration and logistic information and can register online.

May 29-31, 2012 in Manchester, UK.  This class will be hosted by Rational distributor Wigwam Acoustics and will be taught by Jim Cousins following the standard 3-day training format with Days 1 and 2 covering Smaart Fundamentals and Applications and Day 3 as the optional Practicum.  Interested attendees will need to contact Wigwam directly for pricing, registration and logistic information by .(JavaScript must be enabled to view this email address).

Don’t forget these previously announced classes:
June 18-20, 2012 in Las Vegas, NV
June 27-29, 2012 in Quito, Ecuador
July 2-4, 2012 in Lima, Peru
September 18-20 in Irvine, CA

Further information and registration details on all classes listed above can be found on the Class Schedule page of the Rational Acoustics web site.

In addition, Rational Acoustics Rational Acoustics will be exhibiting at the upcoming InfoComm show in Las Vegas, Nevada from June 13-15, 2012.  We will be located in the Central Hall, booth #C11346.  The latest enhancements to Smaart v.7 and various other peripheral measurement products accessories like the Smaart I-O and the Noise Stick will be on hand.

Jamie Anderson will also be presenting two sessions for InfoComm University on Thursday, June 14th.  The first is Session #IUX07: “Smaart Application: Reading the Phase Trace” which will run from 8:00AM to 12:00PM.  The second is Session #IUX10:  “Smaart Application: Spatial Averaging, or Measuring Something that Doesn’t Exist” which runs from 12:30PM to 4:30PM. 

Rational Acoustics

Monday, May 14, 2012

Producer Jake Gosling Chooses Audient ASP008

Music producer, songwriter and remixer Jake Gosling, best known for his collaboration with artists such as Ed Sheeran and Wiley, has just bought himself an Audient ASP008 mic preamplifer, which he is using in his Surrey-based Sticky Studios.

Supplied by high end studio sales company Kazbar Systems, Gosling couldn’t be more satisfied.

“Kazbar Systems lent me the ASP008 as we needed extra mic pres for our DAW,” he explains. “I was instantly impressed with how dynamic and punchy the pres sounded and we’ve been equally impressed with how well the instrument inputs perform when used with our synths. I would highly recommend ASP008 to prospective musicians.”

Already a fan of vintage analogue, Gosling has built Sticky Studios around his old DDA console (another product developed by Audient designer, David Dearden) so it seems perfectly natural to add the Audient ASP008 mic pre to his kit-list.  With eight channels of premium mic preamplifier housed in a compact 1U of rack, ASP008 bypasses the mic pres and converters in the multi channel interface to bring warmth, clarity and punch to every recording.

Most recently Gosling has co-produced Paloma Faith’s album “Fall from Grace” alongside Nellee Hooper. He has also released a number of official remixes under the alias ‘Sketch Iz Dead’ for Lady Gaga, Timbaland, Justin Timberlake, Kerri Hilson, Wale and Far East Movement and has played a pivotal role in developing a number of new artists.

Ed Sheeran, ‘discovered’ by Gosling when Ed was just 15 is counted among these, and has truly proved himself when his album ‘+’ went triple platinum (over 900,000 sales) in January of this year.

Audient

Audix Mics Up Recycled Percussion – And a Chainsaw

Recycled Percussion holds the distinction of the highest non-vocal finalists ever on “America’s Got Talent”. They’re now playing for audiences at the famed Tropicana Las Vegas Theater, performing six interactive shows a week using nearly 60 Audix mics to capture their brand of mayhem.

From the moment guests enter the theater and are each handed a drum stick and something to bang on, to the curtain raising to reveal… well nothing, it’s obvious that this will not be a typical Vegas show.

What follows is musical, mechanical madness as the group descends playing four ‘drum kits’ while bolted to a vertical wall. Wouldn’t it be a waste not to tip ‘The Wall’ fully upside-down before finally settling on the stage? Recycled Percussion thinks so.

From there on, percussionists Justin Spencer & Ryan Vezina, DJ Todd Griffin and guitarist Matt Bowman proceed to use, misuse and abuse pretty much anything one can find at hardware stores, junk yards and landfills. Oh, and a kitchen sink…

“The Audix D6 and D4 crushed the sound of the other microphones we were using and trying,” says Spencer; group founder and sound designer. “We have over twenty plastic barrels that we play. We need five dollar 22-gallon buckets from WalMart to sound like $2,000 kick drums and Audix, with some plate reverb, is the choice for us.”

Spencer and Vezina have nearly matching collections of junk, referred to as ‘Blue Kit’ and ‘Red Kit’. Each one features three 22 gallon plastic tubs, snares and various ‘cymbals’. D6’s and D4’s handle the drums, with overhead i5’s to cover the rest of the pile.

Twenty i5’s are used for all the snares and cymbals (and oxygen tanks, hubcaps, fire extinguishers, toasters, car mufflers etc.)  OM7’s are used for vocals and body percussion. Eighteen Audix mics alone are used for ‘The Wall’ of four drum kits; D2’s, D4’s, D6’s and i5’s. Another six D6’s and four i5’s cover ‘The Van’; a vehicle that splits open to reveal more percussive pyrotechnics. Additional D4’s and D6’s cover moving drums as necessary. Sound chaotic? It is!

The Audix D6 was selected for its unique ability to combine full low frequency response with a crisp and accurate high end. “We don’t play our kick (plastic tub) drums with pedals,” explains Spencer. Everything is played with sticks and we sometimes treat the drums like hi-hats, so we need that full in-your-chest sound, but with a crisp ‘snap’ and no flabby over-hang. The D6 is singlehandedly the best for this.”

“Shifting from some of the other well known kick-drum microphones out there to the Audix D6 was just a drastic reduction of the E.Q. needed at the console,” adds Tropicana front of house mixer Jeffrey James (JJ). “The plastics tubs that the guys use have an inherent pitch and tone to them, depending on how they are (duct) taped. Other kick drum mics weren’t able to capture that tone. The Audix D6’s make them sound like drums.”

“The 22 gallon plastic tubs are mounted to upside down folding metal chairs,” continues JJ. “The tubs are pushed down over the chair legs and mic-mounts clamped to the horizontal chair supports place the D6’s and D4’s pretty much in the inside center of the buckets”.

JJ, who also mixes “Dancing with the Stars,” is kept pretty busy during a typical Recycled Percussion performance; mixing both front of house and monitors. “I sub-group ‘The Wall’, ‘The Van’ and Justin and Ryan’s kits,” he explains. “I have six monitor zones going up and down to keep on-stage volume under control. Again, the Audix mics keep me from having to pay constant attention to E.Q.”

Recycled Percussion is on stage at the Tropicana Wednesday through Monday every week, in addition to other shows and corporate events. Their new concert DVD is available at recycledpercussionband.com. New shows are in the works for this fall.

Oh, and the chainsaw? “You mic it from a respectful distance with an overhead i5,” says JJ.

Audix

In The Studio: The Latest On “Mastered For iTunes”

This article is provided by Bobby Owsinski.

 
On Friday I had the pleasure of sitting in on a meeting with three representatives from Apple at Oasis Mastering, as they briefed us on the latest info on the new “Mastered for iTunes” program.

While there’s already some info on the program currently available, it was good to finally get the most up-to-date details directly from the source. Here’s the gist of the program.

“Mastered for iTunes” at its most basic is iTunes finally opening up to hi-res masters. This means a number of things:

1) iTunes now prefers that you supply the master audio files at 96kHz/24 bit, but any sample rate that’s a 24 bit file will still be considered “Mastered for iTunes.” Music files that are supplied this way will have a “Mastered for iTunes” icon (like on the left) placed beside them to identify them as such.

The reason why they’re asking for 96/24 is so they can both start with the highest resolution source material for a better encode, but also for a bit of future proofing in the event that iTunes later converts to a better format or a higher encode resolution (it’s now 256kbs, but more on this in a second).

2) “Mastered for iTunes” doesn’t mean that the mastering facility does anything special to the master except to check what it will sound like before they (or the record label) submit it to iTunes, and then check it later once again. All encoding for iTunes is still done by Apple, not by the mastering houses, record labels, or artists.

The reason for this is to keep the encodes consistent and to prevent anyone from gaming the system by hacking the encoder, but also to avoid any potential legal problems that might occur when a mastering house sends the files directly to iTunes instead of the label without their permission, or uses different specs, etc.

3) As stated above, the mastering house doesn’t do any encoding directly, but Apple has provided a number of tools that they can use to hear what the final product will sound like when it’s encoded. That way they can make any adjustments to the master to ensure a good encode.

One unique aspect of “Mastered for iTunes” is something that’s not been publicized called a “test pressing.”

When Apple finally encodes the file, they’ll send a copy back to the label/engineer/artist to check. If they sign off on it, the song then goes on sale in the iTunes store.

Of the few mastering houses that are currently participating in the program (all of the major ones), it was surprising that most of the time a test pressing was rejected not because of the audio quality, but because it was the wrong master.

Yes, as record companies seem to do, someone would actually send the un-mastered file or a completely different song or version. Luckily, the problem is now able to be caught in the test pressing stage.

4) Speaking of the sound quality, iTunes is now using a completely new AAC encoder with a brand new algorithm and the sound quality it produces is stunning. It provides an excellent encode if you use a few common sense guidelines (more on this in a bit), and if you do, the result is almost impossible to hear (at least on the music we listened to).

I mean, there we were, mastering engineers Eddy Schreyer, Gene Grimaldi plus myself, listening in this fantastic listening environment, and we literally couldn’t tell between the source and the encode most of the time.

Now there were some where we could hear the difference too, but it wasn’t that big a difference and certainly didn’t sound anywhere near as bad as the typical MP3.

So what are the tricks to get the best sound quality from an iTunes encode?

It turns out that the considerations are about the same as with MP3 encoding:

  a) Turn it down a bit. A song that’s flat-lined at -.1 dBFS isn’t going to encode as well as something with some headroom. This is because the iTunes AAC encoder outputs a tad hotter than the source, there’s some intersample overs that happen at that level that aren’t detected on a typical peak meter, and all DACs respond differently. Something that won’t be an over on your DAC may be an over on another playback unit. If you back it down to -.5 or even -1 dB, the encode will sound a lot better and your listener probably won’t be able to tell much of a difference anyway.

  b) Don’t squash the master too hard. Masters with some dynamic range encode better. Masters that are squeezed to within an inch of their life don’t. Simple as that. Listeners like it better too.

  c) Although the new encoder has a fantastic frequency response, sometimes rolling off a little of the extreme top end (16k and above) can help the encode as well.

5) “Mastered for iTunes” is only an indication that a hi-res master was supplied; it’s not a separate product. There will always be only one version of the song on iTunes at the same price as before. “Mastered for iTunes” doesn’t mean you get to charge more, or that iTunes charges you more. Everything is like it was before, you just supply a hi-res master so it sounds better.

6) So how do you supply that hi-res master? This is where it gets a bit tricky. If you’re signed to a major label, they’ve been contacted my their Apple reps and everything is in place, so no problem there. If you’re with an indie label, insist that they contact their Apple rep for instructions.

If you use CD Baby or Tunecore, at the moment they’ll tell you they don’t take 24 bit or high sample rate masters. Insist that they contact their Apple rep and don’t take no for an answer (this is what the Apple iTunes guy told us).

Apple is greatly encouraging everyone to get with the program, so the more pressure you put on them, the quicker it will become a standard. Of course, if you can find out who your local Apple rep is (ask the local label), that could expedite things too.

The bottom line is that “Mastered for iTunes” is a great thing for digital music. As far as I can see, there’s no downside to it (except maybe for the initial hassle you may go through as an indie), and you’ll be giving your fans a much better sounding product as a result.

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.

Friday, May 11, 2012

In The Studio: Record Great-Sounding Drums Using Only Four Tracks

Here’s a simple, common-sense method to record a great-sounding drum kit on only four tracks. I’ve always been a follower of the less-is-more philosophy, and this kit technique goes all the way back to my analog 4-/8-/12-track days when track economy was a must.

There have, of course, been volumes dedicated to recording “trap kits”, from only two microphones to two mics on each drum! I think the concept of a drum kit as a group of separate instruments is off-base. A drum kit is an ensemble and has a “group sound”.

To me, using a mic on each drum is like multi-miking piano strings. And using so many mics can become a balance, EQ, and panning headache—as well as a phasing nightmare!

Now back to square one: What exactly are you looking for in a kit sound? Most pop, rock, R&B, jazz and country music requires a good separate kick and snare sound and maybe a separate hi-hat depending on the musical arrangement.

But how about all those tom-toms? They have essentially the same fundamental sound with different harmonics depending on shell size, diameter and tuning. If struck properly, they will have approximately the same volume level, and most pro drummers will do this subconsciously. So a simple mic technique can capture them well.

If you are looking for an earthy, realistic kit sound that can easily be manipulated, try this out. You’ll need one pair of stereo mics, one good snare mic, one good kick mic and last but not least, a good location for the drum set in the studio.

Setup
Location is very important and is an oft-neglected starting point. I prefer to set up the kit across a corner. A wall will do as well.

I use baffles from the floor up to about three feet, and approximately four feet wide behind the drummer in the corner. My baffles are low tech, made of 3-foot by 4-foot jalousie window frames with 6-inch wooden slats. There are four layers of wool moving blankets behind them. Thank you U-Haul! 

Other baffle materials that work well are cork faced bulletin boards with combos of styrosheet and blankets behind them. Portable office walls work well too.

The baffles reduce, but do not completely remove the resonances and reflections of the tom, snare and kick. We want some of these reflections for our kit sound.

Microphone Placement
Once the kit is set up with baffles in place, and tuned as you and the drummer want it, you can proceed with mic placement. It’s always a good idea to watch the drummer play for a while to observe where he places his hands and sticks while going around the kit. This will help you put the mics where he won’t hit them or have to move around them. We want him to be comfortable.

Walk up in front of the kit, put your head over the tom-toms, find a spot where the drums seem to focus, and listen for the toms and reflections off the corner. What you’re hearing is a larger percentage of top skins, some bottom skins and wall reflections.

I usually find this spot about two or three feet above the toms and two-thirds of the way over the toms. That’s fairly close but out of the drummer’s stick path. This will be the position for the stereo mics. I have been using a Crown SASS-P MKII stereo mic for this job for more than a decade. Any good pair of mics in a stereo configuration should work well.

Remember, tom-tom and snare spill is actually an important part of the overall sound. If you listen to the drum solo tracks on the Beatles Anthology CDs, you’ll hear a great example of this: Ringo’s Ludwigs drone along just beautifully in “Strawberry Fields”  Another example is Levon Helm’s kit on all The Band’s classics and—oh yeah, Atlantic R&B.

Once you’ve positioned the stereo mics, the rest is straightforward except for the optional hi-hat. For snare and hi-hat use a mic that has plenty of proximity effect. Position the mic at the snare-drum edge between the drum and the high-hat. This should keep the mic out of the drummer’s sticking path.

A Shure SM57 will work OK. I prefer a condenser, a Neumann KM84 or AKG 451 type. The Asian clone mics are recommended here.

You want the mic nice and close to exploit the cardioid proximity effect to get the snare drum “bulge” sound. I prefer positioning at a slight angle. The mic will pick up the high-hat thanks to leakage into the side of the mic.

Finally, the kick mic is whatever you’re comfortable with. Your criteria should be good low frequency response, excellent transients and—very important—ability to handle high sound pressure levels at low frequencies.

Find a sweet spot where you hear a definite increase in volume and tone. I use a Sony ECM 322 (an ancient cardioid condenser) inside the drum under a layer of blanket, about 4 inches away, parallel to the drum head. This is for a one-head kick drum. For two heads, I use a Neumann U47 FET in front. Here a large-diaphragm, Asian mic clone will also work fine, but be aware of room noise.

If you really need overheads for the cymbals, add them. If arrangement calls for a hi-hat played open and closed, use an extra mic.

Finally, record as hot as you can without clipping. This gives you the dynamic range needed for a good drum kit sound. I don’t recommend compression while recording. 

Mixing
That’s it for setup—now on to mixing where we will tailor—not create—our kit sound. As always, if you got it right in the recording, the mixing will be easy and breezy and not a time consuming chore.

If at all possible don’t mix immediately after tracking because of listening fatigue. It’s much better to come in fresh with the concept of simply getting the right sound.

I personally hate to mix after tracking. While recording I have a pure “techno Nazi” head: ears tuned for all the bad stuff, rattles, hum, clipping, pitch, meter, mistakes, etc.

While mixing, first listen with no effects, EQ, reverb, etc. Start with only the overhead pair. You should have a nice overall kit sound that’s almost usable by itself. Listen to the sound and use EQ to trim out any unneeded resonances.

Be careful not to cut into the floor toms’ fundamentals. If your board has minimal EQ, beg, borrow or steal a pair of graphics or parametrics. A single sweepable mid EQ will make life difficult here. You’ll need to EQ mids at more than one frequency.

Now’s a good time to add some reverb. You will use the drums to “trigger” the reverb and they will complement each other.

Don’t smother the drums with too much low-end EQ on either the track or the reverb. Always remember less is more! 

Listen for significant tom fills and cymbal crashes. You should be able to tweak low mids and mids and separate upper mids/highs for the crash cymbals. Also cut out high-end in the reverb. We don’t want any reverb on the cymbals.

Next listen to the snare/hi-hat mic. Again, start flat. Trim the bottom for unwanted room rumblings. Work for a big snare sound, usually found in the low mids and even the upper bottom. Add the reverb and work the two.

Next, on to our optional hi-hat. Many engineer/producers make the mistake of going too high in frequency looking for a hi-hat sound. Cymbals have a broadband signal and reach well down into the mids.

Look for an effective stick and brass strike and then add highs to sweeten the sound. Not too much! Make them peek through and you’ll have the real deal.

Is the sound O.K. now? The kit should sound sort of like Levon Helm and The Band. But what if that’s not what you want? Do you need more balls, more commercial sound, more funk?

Let’s whip out the compressors. Some compression on the toms will even them out and make them cut through nicely. Again, not too much. Play with the ratio. It’s a good idea to let them build to a threshold for more dynamics. I prefer RMS compression for a more natural sound.

Snare is more critical with compression but more fun. If the drummer is consistent in volume, you can use a higher ratio, but watch the threshold. Let it limit only the top. Watch the lil’ red lights and make ‘em dance to the beat. This way you can control the ring tone of the drum and make it really funky.

This same snare technique will work fine on the kick drum. You can control the attack and tone. Watch the bottom end with the EQ. Don’t overdo it or it will get lost when heard with the bass player. Look for EQ frequencies that separate the kick from the bass.

I use little or no reverb on kick drums. You want to trigger the reverb with the compressed signal from the snare and toms to get a naturally reverberant sound. This is exactly what a properly tuned and played kit in a decent acoustic environment would sound like…….with a little help from our electro friends.

Now you should have all the control you’ll ever need. You can raise and lower kick and snare independently as needed in the mix.

You can also pan the tom-toms as you like. I usually use a medium pan on the overheads. This way when the drummer plays a fill across the kit, it will bloom across the sound field and then settle down the way a real kit would—unlike with the hard synthetic panning of individual toms that always stay separated from each other. This naturally occurring sound will also help the drummer, as his kit will sound the same in the cue ‘phones as it does live.

I usually leave the kick and snare near center, but not on top of each other. It’s best to slightly separate kick and bass. Bass that is too far to either side is bad news for the mastering engineer.

During tom fills, the effect of natural buildup is due to the toms’ resonance enhanced by the corner walls and your compressor.

Tweaks For Different Genres
Try this technique when you have time and you’re not under a deadline. It’s well worth the effort. If you can nail it, it will work with little variation on many types of music.

Some suggestions:
- R&B: toms medium spread, kick and snare tight-panned.
- Country: tight-panned snare and kick, medium tom spread (just like R&B).
- Jazz: close-mike the snare and kick, mike the toms not too close, and use very little “room program” reverb.
- Doo-wop: mike very close for mono sound, and use little or no reverb.
- Reggae: mike snare and kick very close, pan toms wide, use tight EQ and mucho reverb.

Neat huh?  Good luck!

Ward Lionel Kremer is a lifelong musician, producer, and recording engineer, who cut his first hit at age 17. In the 1960’s he recorded and performed in the New York pop/R&B music scene with The Four Seasons, The Chiffons, Joey Dee, The Temptations, and Ike & Tina Turner. In the ‘70s he worked in the Miami music scene with TK records, KC & The Sunshine Band, George McRae, and The Ritchie Family. Ward also recorded and produced soca, reggae, and jazz festivals in Italy, USA, and Mexico. He did live sound and recording for Randy Bernsen and Ken Basman. As Ward says, “There’s no music I can’t appreciate if it’s performed with soul, sincerity and love!”

The Audio Expert: Lies, Damn Lies, and Audio Gear Specs—Part 2

Jonathan: “You lied first.”
Jack: “No, you lied to me first.”
Jonathan: “Yes, I lied to you first, but you had no knowledge I was lying. So as far as
you knew, you lied to me first.” — Bounty hunter Jack Walsh (Robert De Niro) arguing with white-collar criminal Jonathan Mardukas (Charles Grodin) in the movie Midnight Run

When it comes to audio fidelity, the four standard parameter categories can assess any type of audio gear.

Although published product specs could tell us everything needed to evaluate a device’s transparency, many specs are incomplete, misleading, and sometimes even fraudulent.

This doesn’t mean specs cannot tell us everything needed to determine transparency—we just need all of the data.

However, getting complete specs from audio manufacturers is another matter. Often you’ll see the frequency response given but without a plus/minus dB range. Or a power amp spec will state harmonic distortion at 1 kHz, but not at higher or lower frequencies where the distortion might be much worse. Or an amplifier’s maximum output power is given, but its distortion was spec’d at a much lower level such as 1 watt.

Lately I’ve seen a dumbing down of published gear reviews, even by contributors in pro audio magazines, who, in my opinion, have a responsibility to their readers to aim higher than they often do. For example, it’s common for a review to mention a loudspeaker’s woofer size but not state its low-frequency response, which is, of course, what really matters.

Audio magazine reviews often include impressive-looking graphs that imply science but are lacking when you know what the graphs actually mean. Much irrelevant data is presented, while important specs are omitted. For example, the phase response of a loudspeaker might be shown but not its distortion or off-axis frequency response, which are far more important.

I recall a hi-fi magazine review of a very expensive tube preamplifier so poorly designed that it verged on self-oscillation (a high-pitched squealing sound). The reviewer even acknowledged the defect, which was clearly visible in the accompanying frequency response graph.

Yet he summarized by saying, “Impressive, and very highly recommended.” The misguided loyalty of some audio magazines is a huge problem in my opinion.

Even when important data are included, they are sometimes graphed at low resolution to hide the true performance. For example, a common technique when displaying frequency response graphs is to apply smoothing, also called averaging. Smoothing reduces the frequency resolution of a graph, and it’s justified in some situations. But for loudspeakers you really do want to know the full extent of the peaks and nulls.

Another trick is to format a graph using large, vertical divisions. So a frequency response line may look reasonably straight, implying a uniform response, yet a closer examination shows that each vertical division represents a substantial dB deviation. The graphs in Figures 1—3 below were all derived from the same data but are presented with different display settings.

For this test I measured the response of a single loudspeaker in a fairly large room with a precision microphone about a foot away. Which version looks more like what loudspeaker makers publish?

Figure 1: Loudspeaker response as measured, with no smoothing.

Figure 2: The exact same data but with third-octave smoothing applied.

Figure 3: The same smoothed data as in Figure 2, but at 20 dB per vertical division instead of 5 dB, making the loudspeaker’s response appear even flatter.

Test Equipment

“Empirical evidence trumps theory every time.”

Noise measurements are fairly simple to perform using a sensitive voltmeter, though the voltmeter must have a flat frequency response over the entire audible range.

Many budget models are not accurate above 5 or 10 kHz.

To measure its inherent noise, an amplifier or other device is powered on but with no input signal present; then the residual voltage is measured at its output.

Usually a resistor or short circuit is connected to the device’s input to more closely resemble a typical audio source.

Otherwise, additional hiss or hum might get into the input and be amplified, unfairly biasing the result.

Most power amplifiers include a volume control, so you also need to know where that was set when the noise was measured. For example, if the volume control is typically halfway up when the amplifier is used but was turned way down during the noise test, that could make the amplifier seem quieter than it really is.

Although it’s simple to measure the amount of noise added by an audio device, what’s measured doesn’t necessarily correlate to its audibility. Our ears are less sensitive to very low and very high frequencies when compared to the midrange, and we’re especially sensitive to frequencies in the treble range around 2 to 3 kHz.

To compensate for this, many audio measurements employ a concept known as weighting. This intentionally reduces the contribution of frequencies where our ears are less sensitive. The most common curve is A-weighting, as shown in Figure 4.

Figure 4: A-weighting intentionally reduces the contribution of low and very high frequencies, so noise measurements will correspond more closely to their audibility.

In the old days before computers were common and affordable, harmonic distortion was measured with a dedicated analyzer. A distortion analyzer sends a high-quality sine wave, containing only the single desired frequency with minimal harmonics and noise, through the device being tested.

Then a notch filter is inserted between the device’s output and a voltmeter. Notch filters are designed to remove a very narrow band of frequencies, so what’s left are the distortion and noise generated by the device being tested. Figure 5 shows the basic method, and an old-school Hewlett-Packard distortion analyzer is shown in Figure 6.

Figure 5: To measure a device’s harmonic distortion, a pure sine wave is sent through the device at a typical volume level. Then a notch filter removes that frequency. Anything that remains are the distortion and noise of the device being tested.

Figure 6: The Hewlett-Packard Model 334A Distortion Analyzer. (Photo courtesy of Joe Bucher.)

Intermodulation distortion is measured using two test tones instead of only one, and there are two standard methods. One method sends 60 Hz and 7 kHz tones through the device being tested, with the 60 Hz sine wave being four times louder than the 7 kHz sine wave.

The analyzer then measures the level of the 7,060 Hz and 6,940 Hz sum and difference frequencies that were added by the device. Another method uses 19 kHz and 20 kHz at equal volume levels, measuring the amplitude of the 1 kHz difference tone that’s generated.

Modern audio analyzers like the Audio Precision APx525 shown in Figure 7 are very sophisticated and can measure more than just frequency response, noise, and distortion. They are also immune to human hearing foibles such as masking (1), and they can measure noise, distortion, and other artifacts reliably down to extremely low levels, far softer than anyone could possibly hear.

Figure 7: The Audio Precision Model APx525 Audio Analyzer. (Photo courtesy of Audio Precision)

Professional audio analyzers are very expensive, but it’s possible to do many useful tests using only a Windows or Mac computer with a decent-quality sound card and suitable software. I use the FFT feature in Sony’s Sound Forge audio editing program to analyze frequency response, noise, and distortion.

For example, when I wanted to measure the distortion of an inexpensive sound card, I created a pure 1 kHz sine wave test signal in Sound Forge. I sent the tone out of the computer through a high-quality sound card having known low distortion, then back into the budget sound card, which recorded the 1 kHz tone. The result is shown in Figure 8. (Other test methods you can do yourself with a computer and sound card are described in Chapter 22.)

Figure 8: This FFT screen shows the distortion and noise added by a consumer-grade sound card when recording a 1 kHz sine wave.

As you can see in Figure 8, a small amount of high-frequency distortion and noise above 2 kHz was added by the sound card’s input stage. But the added artifacts are all more than 100 dB softer than the sine wave and so are very unlikely to be audible.

Low distortion at 1 kHz is easy to achieve, but 30 Hz is a different story, especially with gear containing transformers. Harmonic distortion above 10 kHz matters less because the added harmonics are higher than the 20 kHz limit of most people’s hearing. However, if the distortion is high enough, audible IM difference frequencies below 20 kHz can result.

Sadly, many vendors publish only THD measured at 1 kHz, often at a level well below maximum output. This ignores that distortion in power amplifiers and gear containing transformers usually increases with rising output level and at lower frequencies.

The convention these days is to lump harmonic distortion, noise, and hum together into a single THD + Noise spec and express it as either a percentage or some number of dB below the device’s maximum output level.

For example, if an amplifier adds 1 percent distortion, that amount can be stated as 40 dB below the original signal. A-weighting is usually applied because it improves the measurement, and this is not unfair. There’s nothing wrong with combining noise and distortion into a single figure either when their sum is safely below the threshold of audibility.

But when distortion artifacts are loud enough to be audible, it can be useful to know their specific makeup. For example, artifacts at very low frequencies are less objectionable than those at higher frequencies, and harmonics added at frequencies around 2 to 3 kHz are especially noticeable compared to harmonics at other frequencies.

Again, this is why A-weighting is usually applied to noise and distortion measurements and why using weighting is not unreasonable.

———————————————————————————————————————

1) The masking effect refers to the ear’s inability to hear a soft sound in the presence of a louder sound. For example, you won’t hear your wristwatch ticking at a loud rock concert, even if you hold it right next to your ear. Masking is strongest when both the loud and soft sounds contain similar frequencies.

Audio Transparency

As we have seen, the main reason to measure audio gear is to learn if a device’s quality is high enough to sound transparent.

All transparent devices by definition sound the same because they don’t change the sound enough to be noticed even when listening carefully.

But devices that add an audible amount of distortion can sound different, even when the total measured amount is the same. A-weighting helps relate what’s measured to what we hear, but some types of distortion are inherently more objectionable (or pleasing) than others.

For example, harmonic distortion is “musical,” whereas IM distortion is not. But what if you prefer the sound of audio gear that is intentionally colored?

In the 1960s, when I became interested in recording, ads for most gear in audio magazines touted their flat response and low distortion. Back then, before the advent of multilayer printed circuit boards, high-performance op-amps, and other electronic components, quality equipment was mostly handmade and very expensive. In those days design engineers did their best to minimize the distortion from analog tape, vacuum tubes, and transformers.

Indeed, many recordings made in the 1960s and 1970s still sound excellent even by today’s standards. But most audio gear is now mass-produced in Asia using modern manufacturing methods, and very high quality is available at prices even nonprofessionals can easily afford.

Many aspiring recording engineers today appreciate some of the great recordings from the mid-twentieth century. But when they are unable to make their own amateur efforts sound as good, they wrongly assume they need the same gear that was used back then.

Of course, the real reason so many old recordings sound wonderful is because they were made by very good recording engineers in great (often very large) studios having excellent acoustics. That some of those old recordings still sound so clear today is in spite of the poorer-quality recording gear available back then, not because of it!

Somewhere along the way, production techniques for popular music began incorporating intentional distortion and often extreme EQ as creative tools. Whereas in the past, gear vendors bragged about the flat response and low distortion of their products, in later years we started to see ads for gear claiming to possess a unique character, or color.

Some audio hardware and software plug-ins claim to possess a color similar to specific models of vintage gear used on famous old recordings. Understand that “color” is simply a skewed frequency response and/or added distortion; these are easy to achieve with either software or hardware, and in my opinion need not demand a premium price.

For example, distortion similar to that of vacuum tubes can be created using a few resistors and a diode, or a simple software algorithm.

The key point is that adding color in the form of distortion and EQ is proper and valuable when recording and mixing. During the creative process, anything goes, and if it sounds good, then it is good. But in a playback system the goal must be for transparency—whether a recording studio’s monitors or a consumer playback system.

In a studio setting the recording and mixing engineers need accurate monitoring to know how the recording really sounds, including any coloration they added intentionally. With a consumer playback system you want to hear exactly what the producers and mix engineers heard; you’ll hear their artistic intent only if your own system adds no further coloration of its own.

“The Audio Expert” by Ethan Winer, published by Focal Press (ISBN: 9780240821009), is available here. To read part 1, Audio Fidelity, Measurements, And Myths, go here.

Metric Halo On The Road With Rising Hip-Hop Stars

The team of rapper Macklemore and producer Ryan Lewis is making some of the most forward-thinking hip-hop music on the planet

Lewis, who also handles many of the engineering duties for the duo, recently committed to purchasing a
Metric-Halo ULN-2 preamplifier/interface so that they could cut studio-quality tracks for their forthcoming album while on the road.

“I’m in the middle of producing a full-length LP with rapper Macklemore,” said Lewis. “I needed a small unit that could deliver audio quality worthy of our debut LP.

“The goal is for Macklemore to be able to record at home or on the road, tracking vocals wherever he wants or needs to.”

Lewis learned about Metric Halo through online research and discussions with audio engineers in their hometown of Seattle.

“One of the largest consumer products for introductory home recording is the two-channel USB or FireWire interface,” he continued. “Almost every company has their own version. Nevertheless, there honestly wasn’t a lot out there that could compete with the ULN-2.

“Finding high-quality AD/DA conversion is hard, and finding high-quality AD/DA conversion together with fantastic preamps is almost impossible. The Metric Halo ULN-2 was exactly what we needed. It alone could match the quality of vocals we’re tracking in the studio.”

As an active member of the Seattle music scene, Macklemore earned a hometown following early on. But it was with his 2004 release Welcome to Myspace – which was heavily hyped by the then-new social networking site – that Macklemore earned respect and an eager following at the national and international levels.

Taking the long view, Lewis cited Metric Halo’s dogged commitment to future-proofing its equipment via upgrades as another excellent reason to go with the ULN-2.

“Metric Halo is a company that is focused on the future of pro audio and that’s why I’m so excited to be working with their products,” he said.

Later in the year, Lewis and Macklemore plan to follow up the ULN-2 with an eight-channel ULN-8, Metric Halo’s flagship product, for use in their studio.

Metric-Halo Labs

Telefunken And Manley Labs Support WAM Recording Session With Kronos Quartet

Women’s Audio Mission, a non-profit dedicated to the advancement of women in music production and the recording arts, recently partnered with Manley Labs and Telefunken Elektroakustik, who provided microphones for a recording session with the GRAMMY Award Winning Kronos Quartet.

Telefunken generously loaned WAM a pair of ELA M 260s, AR-51s and CU-29 Copperhead microphones and Manley Labs loaned a Manley Gold Reference specifically for the session.

The Kronos Quartet recorded material for their upcoming performance of “Women’s Voices” on May 11th, a concert series featuring critically acclaimed female composers and performers from around the world, including Vietnamese multi-instrumentalist Van-Anh Vo. Vo, who joined the Kronos Quartet in-studio at WAM, has written and arranged five pieces for the quartet that she’ll also perform during “Women’s Voices.”

Women’s Audio Mission founder Terri Winston and staff engineer Laura Dean engineered the session, assisted by Jenny Thornburg. With help from Manley and Telefunken, the session was an incredible success and WAM’s engineers were able to optimally capture the Quartet’s signature sound.

“It was great to have Telefunken and Manley on board for this project,” says Winston.“It made all the difference in the world to have that confidence in our mic selection.  Kronos Quartet especially liked the Copperheads on violins and the ELA M 260 and Gold Reference on cello.”

Winston continues, “We couldn’t be happier to have collaborated with Telefunken, Manley and the Kronos Quartet on something that is also a huge part of our mission – to expand the voice of music and media and ensure that women’s ideas, interests and points of view are conveyed throughout our culture and society.”

SoundPure.com coordinated the microphone loans from Telefunken and Manley; the WAM team recorded to Pro Tools HD 2 with Lavry Blue converters, Avedis MA-5, Great River MP-2NV and Millennia HV-3R mic pres, and Earthworks QTC 30 microphones.

The material recorded at WAM will debut at the “Women’s Voices” performance at the Yerba Buena Center for the Arts in San Francisco on May 11th and 12th.

Telefunken

Thursday, May 10, 2012

In The Studio: What Engineers Should Know About Meters

This article is the second part in a series on decibels, excerpted from Bob Katz’s book Mastering Audio: The Art and The Science. The first part is available here, and be sure to check out all our videos featuring Bob Katz.

We Won’t Get Fooled Again. Recording engineers rely heavily on their favorite meter, and this is not intended to change people’s favorite.

But as practicing engineers, it is prudent to learn the defects and virtues of each meter we encounter.

The VU Meter. Relative newcomers to the industry may have never seen a VU meter, and some of them may be using the word “VU” incorrectly to describe peak-reading digital meters.

VU should only be applied to a true VU meter that meets a certain standard. The first thing we must learn is that the VU meter is a dreadful liar… It is an averaging meter, and so it cannot indicate true peaks, nor can it protect us from overload.

However the VU does do one thing better than a peak meter—it comes closer to our perception of loudness, but even so, it is a very inaccurate loudness meter because its frequency response gives low frequency information equal weight, and the ear responds less to low frequencies.

Another problem is that the VU meter’s scale is so non-linear that inexperienced operators think that the greater part of the musical action should live between -6 and +3 VU, but this is wrong.

A well-engineered music program has plenty of meaningful life down to about -20 VU, but since the needle hardly moves at that level, it scares the operator into thinking the level is too low.

Only highly-processed (dynamically compressed) music can swing in such a narrow range; in other words, the VU scale encourages over-compression.

Hence the VU meter should only be taken as a guide. A much better averaging meter would have a linear-decibel scale, where each decibel has equal division and weight down to -20 dB.

Digital Peak Meters
Digital Peak meters come in three varieties:
       1. Cheap and dirty
       2. Sample-accurate and sample-counting (but misleading)
       3. Reconstruction (oversampling)

Cheap and Dirty Peak Meters. Recorder manufacturers pack a lot in a little box, often compromising on meter design to cut production costs. A few machines even have meters which are driven from analog circuitry—a definite source of inaccuracy.

VU meter operators are often fooled into treating the top and bottom halves of the scale with equal weight, but the top half has only 6 dB of the total dynamic range.

Some manufacturers who drive their meters digitally (by the values of the sample numbers) cut costs by putting large gaps on the meter scale (avoiding expensive illuminated segments).

The result is that there may be a -3 and a 0 dB point, with a large unhelpful no man’s land in between. When recording with a meter that has a wide gap between -3 and 0, it is best practice to stay well below full scale.

Sample-Accurate and Sample-Counting Meters. Several manufacturers have produced sample-accurate meters with 1 dB (or smaller) steps, that convert the numeric value of the samples to a representation of the sample value, expressed in
dBFS.

The Paradox of the Digital OVER. When it comes to playback, a meter cannot tell the difference between a level of 0 dBFS (FS = Full Scale) and an OVER. That’s because once the digital signal has been recorded, the sample level cannot exceed full scale, as in this figure.

We need a means of knowing if the ADC is being overloaded during recording. So we can use an early-warning indicator—an analog level sensor prior to A/D conversion—which causes the OVER indicator to illuminate if the analog level is greater than the voltage equivalent to 0 dBFS.

If the analog record level is not reduced, then a maximum level of 0 dB will be recorded for the duration of the overload, producing a distorted square wave.

After the signal has been recorded, a standard sample-accurate meter cannot distinguish between full scale and any part of the signal that had gone over during recording, it shows the highest level as 0 dBFS. However, a sample-counting meter can analyze a recording to see if the ADC had been overdriven.

This meter counts contiguous samples and can actually distinguish between 0 dBFS and an OVER after the recording has been made! The sample-counting digital meter determines an OVER by counting the number of samples in a row at 0 dB.

If 3 contiguous samples equal 0 dBFS, the meter signals an OVER, because it’s fair to assume that the incoming analog audio level must have exceeded 0 dBFS somewhere between sample number one and three.

Three samples at 44.1 kHz is a very conservative standard; on that basis, the recorded distortion would last only 33 microseconds and would probably be inaudible.

While an original analog signal can exceed the amplitude of 0 dB, after conversion there will be no level above 0, yielding a distorted square wave. This diagram shows a positive-going signal, but the same is true on the negative-going end.

While this type of meter was sophisticated in its day, current thinking is that the sample-counting meter is only suitable for evaluating whether an ADC has overloaded.

Authorities now feel that meters which display the digital value of the samples and which count samples to determine an OVER are no longer sufficient for mastering purposes and should be used with caution during mixing. Their place is taken by…

The Reconstruction Meter: Even More Sophisticated As long as a signal remains in the digital domain, the sample level of the digital stream is sufficient to tell us if we have an OVER. However, signals which migrate between domains can exceed 0 dBFS and cause distortion.

This includes any signal that passes through a DAC, a sample rate converter, or is converted through a codec such as mp3 or AC3. During the conversion from PCM digital to analog or mp3, filtering within the converter yields occasional peaks between the samples that are higher than the digital stream’s measured level, which can be higher than full scale.

This next figure shows that contrary to what we might assume, filtering or dips in an equalizer which we’d imagine would produce a lower output can actually produce a higher output level than the source signal. B.J. Buchalter explains that:

“the third harmonic is out of phase with the fundamental at the peak values of the fundamental, so it serves to reduce the overall amplitude of the composite signal.”

“By introducing the filter, you have removed this canceling effect between the two harmonics, and as a result the signal amplitude increases. Another reason for the phenomenon is that all filters resonate, and generally speaking, the sharper the filter, the greater the resonance.”

Equipment designers have known for years that because of filtering, the analog output level of complex audio from a DAC can exceed the sinewave value of 0 dBFS but very few have taken this into account in the design.

TC Electronic has performed tests on typical consumer DACs, showing that many of them distort severely since their digital filters and analog output stages do not have the headroom to accommodate output levels which exceed 0 dBFS!

While typical 0 dBFS+ peaks do not exceed +0.3 dBFS, some very rare 0 dBFS+ peaks may exceed full scale by as much as 4 or 5 dB with certain types of signals— especially mastered material which has been highly processed, clipped (turned into a square wave on top and bottom), and/or brightly equalized.

By oversampling the signal, we can measure peaks that would occur after filtering. An oversampling meter (or reconstruction meter) calculates what these peaks would be, but these meters are still rare. Products from TC Electronic (System 6000) and Sony (Oxford) have an oversampling limiter and reconstruction peak meter. RME’s Digicheck software includes an oversampling meter.

Reconstruction meters tell us not only how our DAC will react, but what may happen to the signal after it is converted to mp3 or sent to broadcast, both of which employ many filters and post-processes. Many DSP-based consumer players cannot handle the high levels at all and exhibit severe distortion with 0 dBFS+ signals.

Armed with this knowledge, no mastering engineer should produce a master that may sound acceptable in the control room but which she knows will likely produce severe distortion when post-processed or auditioned in the real world.

If the reconstruction meter is not enough to convince the client, she should also demonstrate that this “loud” signal becomes distorted, ugly, and soft when it is converted to low bit rate mp3. All the harmonics which made the signal seem loud in the control room have been converted to additional distortion.

Practice Safe Levels
What this means is that if you are mixing with a standard digital meter, keep peaks below -3 dBFS, especially if you are using aggressive bus processing.

The more severely processed, equalized or compressed a master, the more problems it can cause when it leaves the mastering studio.

We didn’t start hearing about this problem, or at least the severity of it, before the loudness race and the invention of digital processing which could be egregiously abused. Maximizing engineers should try to use a reconstruction meter and/or an oversampled brickwall limiter. If these are not available, use a standard peak limiter whose ceiling is set to -0.3 dB (see Chapter 10) and exercise caution.

But even the oversampled brickwall limiter is not foolproof; I’ve discovered that such limiters do not protect from very severe processing and can still make a consumer DAC overload unpleasantly. The best solution is to be conservative on levels. Clipping of any type is to be avoided, as demonstrated in Appendix 1.

The Myth of the Magic Clip Removal
If the level is turned down by as little as 0.1 dB, then a recording which may be full of OVERs will no longer measure any overs.

But this does not get rid of the clipping or the distortion, it merely prevents it from triggering the meter.

Some mastering engineers deliberately clip the signal severely, and then drop the level slightly, so that the meters will not show any OVERs.

This practice, known as SHRED, produces very fatiguing (and potentially boringly similar) recordings.

Peak Level Practice for Good 24-bit Recording
Even though 24-bit recording is now the norm, some engineers retain the habit of trying to hit the top of the meters, which is totally unnecessary as illustrated at left.

Note that a 16-bit recording fits entirely in the bottom 91 dB of the 24-bit. You would have to lower the peak level of a 24-bit recording by 48 dB to yield an effective 16-bit recording! There is a lot of room at the bottom, so you won’t lose any dynamic range if you peak to -3 dBFS or even as low as -10 dBFS, and you’ll end up with a cleaner recording.

Since distortion accumulates, if a “hot” recording arrives for mastering, the mastering engineer doing analog processing may have to attenuate the level to prevent the processing DAC from overloading. A digital mix that peaks to -3 dBFS or lower makes it easier to equalize and otherwise process without needing an extra stage of attenuation in the mastering.

In black is a complex wave. When the high frequency information (light orange) is filtered out, the result is a signal (orange) that is higher in amplitude than the original!

A number of 24-bit ADCs are advertised as having additional headroom, achieved by employing a built-in compressor at the top of the scale, claiming that the compressor can also protect the ADC from accidental overloads. But this is specious advertising.

Level accidents don’t occur in a mix studio; engineers have control over their levels and when tracking live musicians, it is better to turn off the ADC’s compressor, drop the level and leave plenty of headroom for peaks. The only possible use of this function of a compressor is if you like its sonic qualities and are trying to emulate the sound of tracking to analog tape.

But since tracking decisions are not reversible, I suggest postponing “analog simulation” to the mixing stage. It’s easier to add warmth later than try to take away some mushiness due to an overdriven compressor. As we have just seen, there is no audible improvement in SNR by maximizing a 24-bit recording and no SNR advantage to compressing levels with a good 24-bit ADC.

How Loud is It?
Contrary to popular belief, the levels on a digital peak meter have (almost) nothing to do with loudness.

Here is an illustration. Suppose you are doing a direct to two-track recording (some engineers do still work that way!) and you’ve found the perfect mix.

Leaving the faders alone, you let the musicians do a couple of great takes. During take one, the performance reached -4 dB on the meter; and in take two, it reached 0 dB for a brief moment during a snare drum hit.

Does that mean that take two is louder? No: because in general, the ear responds to average levels, not peak levels when judging loudness.

If you raise the master gain of take one by 4 dB so that it too reaches 0 dBFS peak, it will sound 4 dB louder than take two, even though they both now measure the same on the peak meter.

An analog tape and digital recording of the same source peaked to full scale sound very different in terms of loudness. If we make an analog tape recording and a digital recording of the same music, and then dub the analog recording to digital, peaking at the same peak level as the digital recording, the analog dub will have about 6 dB more intrinsic loudness than the all-digital recording.

Quite a difference! This is because the peak-to-average ratio of an analog recording can be as much as 12-14 dB, compared with as much as 20 dB for an uncompressed digital recording.

Analog tape’s built-in compressor is a means of getting recordings to sound louder (oops, did I just reveal a secret?). That’s why pop producers who record digitally may have to compress or limit to compete with the loudness of their analog counterparts.

The Myths of Normalization

The Esthetic Myth. Digital audio editing programs have a feature called Peak Normalization, a semi-automatic method of adjusting levels.

The engineer selects all the songs on the album, and the computer grinds away, searching for the highest peak level on the album and then automatically adjusts the level of all the material until the highest peak reaches 0 dBFS. If all the material is group-normalized at once, this is not a serious esthetic problem, as long as all the songs have been raised or lowered by the same amount.

But it is also possible to select each song and normalize it individually, but this is a big mistake; since the ear responds to average levels, and normalization measures peak levels, the result can totally distort musical values. A ballad with little crest factor will be disproportionately increased and so will end up louder than a rock piece with lots of percussion!

The Technical Myth. It’s also a myth that normalization improves the sound quality of a recording; it can only degrade it. Technically speaking, normalization adds one more degrading calculation and level of quantization distortion.

And since the material has already been mixed, it has already been quantized, which predetermines its signal-to-noise ratio—which cannot then be further improved by raising it.

Let me repeat: raising the level of the material will not alter its inherent signal-to-noise ratio but will add more quantization distortion. Of course material to be mastered does not need normalizing since the mastering engineer will be performing further processing anyway. Clients often ask: “do you normalize?” I reply that I never use the computer’s automatic method, but rather songs are leveled by ear.

A 24-bit recording would have to be lowered in level by 48 dB in order to reduce it to the SNR of 16-bit. The noise floors shown are with flat dither.

Average Normalization
This is another form of normalization, an attempt to create an intelligent loudness algorithm based on the average level of the music, as opposed to the peak.

But when making an album, neither peak nor average normalization nor any intelligent loudness algorithm can do the right job, because the computer does not know that the ballad is supposed to sound soft.

There’s no substitute for the human ear. However, average normalization or better, a true intelligent loudness algorithm can help in situations where every program needs the same loudness, even if that doesn’t sound natural, such as radio broadcast, ceiling loudspeakers in a store, a party or background listening.

Judging Loudness the Right Way
Since the ear is the only judge of loudness, is there any objective way to determine how loud your CD will sound? The first key is to use a single DAC to reproduce all your digital sources and maintain a fixed setting on your monitor gain.

That way you can compare your CD in the making against other CDs, in the digital domain. Judge DVDs, CDs, workstations, and digital processors through this single converter.

Eventide Blackhole Native Plug-In Now Shipping

Turning for the first time to its stompboxes for inspiration, Eventide today announced that its Blackhole Native plug-in for AAX, AU or VST is now available. 

The Blackhole Native plug-in is being offered free with the purchase of any Eventide Stompbox effective immediately through June 30, 2012. Today’s announcements follow the recent introduction of Eventide’s Omnipressor and 2016 Stereo Room Native plug-ins.

Blackhole has evolved over the years. In its earliest incarnation on the DSP4000 and H8000 studio processors, it was regarded by some as a secret weapon. With today’s announcement Space’s widely-acclaimed Blackhole is now available as a native plug-in for desktop and laptop recordists. 

“The response to Blackhole from our beta testers has been gratifying,” said Ray Maxwell, Vice President. “We are delighted to bring this other-worldly reverberator to a whole new group of users.”

Blackhole may not be your first choice to simulate an enclosure on the surface of the earth but, if you’re at all interested in ‘sculpting sound’, Blackhole just might pull you in. Its possibilities appear to be endless; can’t see beyond the event horizon. 

Blackhole is available now at your authorized Eventide plug-in dealer or at Eventide.com for a special limited time only introductory price of $99. 

Eventide

Drummer Rick Latham’s Groove Workshop Tours Europe With AKG Drumset Groove Pack

World-renown drummer and “groove” activist, Rick Latham recently entertained and educated aspiring and professional drummers in a workshop dedicated to the music business, highlighting the art of drumming. 

Armed with AKG Drumset Groove Pack Latham’s educational sessions touched on multiple facets of musical education, wowing students and fans during the four-week group workshop.

The workshops focused on mastering the drums; from essential playing tips and helping individual artists find their own sound, to miking, recording and mixing final products.  “That style,” or the LA/American sounding drum sound, has been Latham’s focus during his career and the workshops elaborate on the art of perfecting the sound for the enthusiastic students.

Latham began the 2012 Groove workshop with two clinics/master class performances in Northern Germany – in Lubeck at Musikhaus Andresen and Rendsburg at Musickmarkt – where the drummer demonstrated his finer points of groove and feel for percussion artists.  Playing funk and R&B, Latham stressed the importance of great live sound miking and proper tuning.

The next leg brought Latham to Mannheim, Germany, where his master class was held at Pop Akademie – one of the premier contemporary music schools in Europe.  There, he played with Frank Itt, the renowned German bassist who holds a teaching position at the school. Latham then stopped at Paderborn and Schlangen, Germany, where his audience was a high school music class curious about becoming professional musicians.  That night, Latham offered a clinic for a local community theatre and performed with enthusiastic local musicians. 

Latham rounded out his tour through Istanbul, Turkey, Milan, Italy and returned to Germany for the final stretch.  During each event, Latham utilized the Drumset Grove Pack to demonstrate great sound and the proper mechanics to mic a drum kit. 

“The AKG Drumset Groove Pack does just that, Grooves,” stated Latham.  “It sounds cliché, but the most important aspect of music is sound and the Groove Pack allows my signature sound to flow as it should be heard.  I truly believe in the functionality of the AKG mics, and their ease of use and quality allow me to provide a great example of professional grade music during my workshops.”

Latham’s audience was offered additional captivating information during sessions as the drummer went into the fine details of the business of music.  With a vast range of participants, ranging from young children, to teenagers and professional artists, Latham described his business experiences, which are essential in today’s competitive music space. 

“Today’s professional musician is more involved in all aspects of the business and being successful takes a lot more effort than just being able to play well,” stated Latham.  “With my workshop, the students, both young and old are exposed to the ins and outs of the business and drumming and it opens their eyes, inspiring them to be engaged in multiple levels of the industry.”

AKG