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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

Wednesday, May 09, 2012

In The Studio: Top Ways To Help Musicians Hear Themselves

This article is provided by BAMaudioschool.com.

 
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.

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

In The Studio: Three EQ “Fake-Outs”

This article is provided by Home Studio Corner.

 
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.

Monday, May 07, 2012

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

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

This article is provided by Bobby Owsinski.

 
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.

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

In The Studio: DIY Subkick Microphone

This article is provided by Audio Geek Zine.

 
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.

(click to enlarge)

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.

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