Tuesday, September 15, 2015
Audio Basics: Not As Simple As They Look? Identifying & Solving Microphone Problems
You never know everything and there’s something new to be learned every day
Besides sound quality, there really isn’t much to think about when it comes to microphones, right?
Well, guess again!
Like all elements of a sound system, microphones present their own unique set of special problems.
Fortunately, a lot of these problems are relatively simple to solve. It’s just a matter of identification and appropriate action.
For example, most mic handles include a set-screw near the connector, with many models using this screw to ground the mic handle.
If the handle seems to be picking up hum when touched, check that the set-screw is fully secured down (turn clockwise until tight).
Inside the XLR connector on a mic cable is a ground lug, offering option of tying it to pin 1 or leaving it floating. If this ground lug is connected to pin 1, the connector shell is grounded.
Then, if the shell touches a grounded metal surface, a ground loop can occur, causing hum. So, a better approach is to float the shell.
Lighting cables and AC power cables radiate strong hum fields, which mic cables can pick up.
Keep mic cables well separated from lighting and power cables.
Keep a set of “tweakers” handy to tighten down that mic screw.
If the cables must cross, do so at right angles to reduce the coupling between them.
In addition, vertically separate the cables.
If your situation produces severe hum pickup when using dynamic mic models, try ones that include humbucking coils. (The Shure Beta 58 is probably the most popular example in this regard.)
Also, twisted-pair mic cable can reduce pickup of magnetically induced hum.
The more shield coverage, the less pickup of electrostatically induced hum.
Braided shield generally offers the best coverage; double-spiral wrapped is next best, and spiral-wrapped is worst.
Use twisted-pair cable to reduce pickup of magnetically induced hum.
And, routinely check mic cables to make sure the shielding is connected at both ends. For outdoor work, tape over cracks between connectors to keep out dust and rain.
Shocking But True
At times, electric-guitar players can receive an electric shock when they simultaneously touch their guitar and a microphone.
This is caused when the guitar amp is plugged into an electrical outlet on stage, and the mixing console (to which the mic is grounded) is plugged into a separate outlet across the room.
These two power points may be at widely different ground voltages, so a current can flow between the grounded mic housing and the grounded guitar strings.
This occurrence is especially dangerous when the guitar amp and the console are on different phases of the AC mains.
It helps to power all instrument amps and audio gear from the same AC distribution outlets. That is, run a heavy extension cord from a stage outlet back to the mixing console (or vice versa).
Plug all the power-cord ground pins into grounded outlets. That way, you prevent shocks and hum at the same time.
Also, put a foam windscreen on each vocal mic to insulate the guitarist from shocks. As a bonus, a foam windscreen suppresses breath pops better than a metal grille screen.
There are numerous clip-on options to get rid of mic stands on stage.
If you’re picking up the electric guitar direct, use a transformer-isolated direct box and set the ground-lift switch to the position with the least hum.
Using a neon tester or voltmeter, measure the voltage between the electric-guitar strings and the metal grille of the microphones. If there is a voltage, flip the polarity switch on the amp or reverse its AC plug in the outlet.
Fun With Clip-Ons
Nearly all mic companies offer miniature condenser models. These tiny units can offer the sound quality of larger studio mics, but in a compact package.
If they’re clipped to musical instruments, stage clutter is reduced by eliminating boom stands. Plus, the performer can move more freely around the stage.
Because a miniature clip-on mic is very close to its instrument, it picks up a high sound level. So you can often use an omni mic without feedback.
Omni mics generally have a wider, smoother response than their unidirectional counterparts, and pick up less mechanical vibration.
Another handy clip-on application: drums.
Try mini mics on a drum set as described earlier. Tape an omni mic near the bottom edge of the sound hole, and roll off some bass for a natural tone quality.
Tape one to a flute between the lip plate and finger holes, about 2 inches from the lip plate and 2 inches above the flute. It sounds much more natural than a pickup.
For a grand piano, tape two mini mics to the underside of the raised lid, over the bass and treble strings. If necessary, close the lid for more isolation.
You can also reduce clutter when using regular-sized mics by mounting them in holders that clip to drum rims and mic stands.
Squash The Squeal
Stage monitor speakers are the main cause of feedback, so it’s not the first time you’ve heard this: keep monitor levels down.
Loud monitors leak into vocal mics, creating feedback in addition to coloring sound. Musicians always want the monitors louder, so start with them as quiet as possible, and then when you’re forced to increase levels, they probably won’t be too loud.
One-third-octave (or even narrower) graphic equalizers can also be deployed to fight feedback. Connect the equalizer between the console’s monitor output and the monitor power-amplifier input.
With the equalizer controls centered, set up a normal monitor mix. Now slowly turn up the mixer’s master monitor volume control to bring up the volume in the monitor speakers. The system will start to feed back at a certain frequency.
Try to find this frequency on the equalizer by pushing down each control knob/fader in turn. The control that stops the feedback is the correct one.
Lower this frequency only down to the point where the feedback stops. Then turn up the monitor volume until the system feeds back again (usually at a different frequency). Lower the control for that frequency until feedback stops. (Rinse, repeat!)
Do this procedure several times, turning up the overall volume as feedback is suppressed, so that three to five frequency ranges are cut.
The monitor speakers should now be able to be played louder without feedback than before the equalization process.
There are “feedback fighters” that perform this function automatically. In each device, a microprocessor quickly senses feedback and determines its frequency, then assigns a narrow notch filter to that frequency and eliminates the feedback. Typically, several filters are assigned.
Finally, many sound people have come to love in-ear monitoring systems because they don’t leak into stage mics. And, the resulting house sound can be louder and more natural.
One last tip: At a Lenny Kravitz concert, the piano player (Ken Crouch) was playing an upright piano. The piano sound was excellent, and I complimented the sound engineer, Tom Edmonds, on his skill.
He later told me his secret: the pianist was really playing a Korg M-1 synthesizer mounted inside the upright piano, which was a prop!
The moral of the story is that you just never know everything, and there’s something new to be learned every day.
AES and SynAudCon member Bruce Bartlett is a recording engineer, audio journalist, and microphone engineer. His latest books are “Practical Recording Techniques 5th Ed.” and “Recording Music On Location.”
Thursday, September 10, 2015
Songwriter/Producer Monty Powell Chooses dbx 676 Tube Preamp Channel Strip
Highly awarded songwriter selects vacuum tube-based microphone preamplifier for new studio and educational projects.
Songwriter/producer and music industry veteran Monty Powell has written hit songs for Keith Urban, Lady Antebellum, Rascal Flatts and Tim McGraw and won numerous accolades including a CMA Triple Play Award for having written three #1 songs in one calendar year and SESAC’s Songwriter Of The Year in 2009.
Having recently completed construction of a new recording facility, Powell added the new Harman dbx 676 Tube Mic Preamp Channel Strip to his recording chain.
“Inspired moments in songwriting can be fleeting”, explains Powell.
“You need to be able to capture inspiration in an instant and dialing in a great vocal sound or tracking an instrument through the 676 is effortless. I can route any source to the 676 and comp together a song with confidence. Also, the 676’s compressor is really transparent. Controlling dynamics is a snap with its classic OverEasy setting.”
Powell’s songwriting talents are rivaled only by his generosity and willingness to share his craft, which has sprouted his latest endeavor, Musician Labs - a live, interactive, online instructional series and mentoring program for aspiring songwriters.
“Musician Labs is dedicated to teaching the principles necessary for sustaining a successful career in the music business,” said Powel. “I’m blessed to be able to share my knowledge as a songwriter/producer at a time when technology has removed the conventional barriers of communicating with other artists. I believe achieving success in the music industry is a science and cofounding Musician Labs has given me the perfect vehicle to teach my proven techniques and formula.”
Part of the conversation amongst songwriters inevitably lands on choices of gear.
“dbx compressors are a fixture in many of the Nashville studios where I first cut my teeth in recording music. The dbx 676 lives up to their legacy. It delivers a warm, natural, sound quality often missing from many digital interfaces at a price that isn’t out of reach for home studio enthusiasts. Not everyone can afford the luxury of tracking through a vintage tube mic. The dbx 676 can elevate the performance of any entry level mic. If you aren’t second guessing your signal path, your creativity can flow,” added Powell.
“Pairing the dbx 676 with an artist of Monty’s songwriting and producing abilities is a thrill. You can’t turn on a country radio station without hearing a song he’s either written or influenced and we can’t wait to hear how he uses the 676 on his future projects. In addition to being a force of nature in the songwriting world, Monty has a great ear. We’re honored to have the dbx 676 Tube Mic Preamp be a part of his recording and creative process,” stated Jason Kunz, market manager, portable PA, recording and broadcast, Harman Signal Processing.
The dbx 676 is a vacuum tube-based microphone preamplifier that offers a host of flexible sound-tailoring options to deliver extraordinary audio quality in recording and live sound applications.
The dbx 676 employs a high-gain, Class A tube preamp section based around a 12AU7 vacuum tube that can be adjusted to be clean and pure-sounding or dirty and full of harmonic character. The 676 incorporates the Compressor/Limiter design from the dbx 162SL and a 3-band parametric EQ, enabling exacting control of dynamics and tonal balance.
The dbx 676 Tube Mic Pre Channel Strip has a U.S. street price of $999.95.
Posted by House Editor on 09/10 at 06:36 AM
Drawmer Now Shipping 1978 Stereo Tone Shaping FET Compressor
The 1978 gives flexibility to recording and mix engineers who wish to bless their tracks with an analog soul.
TransAudio Group is poised to begin shipping Drawmer’s new 1978 Stereo Tone Shaping FET Compressor.
The 1978 gives flexibility to recording and mix engineers who wish to bless their tracks with an analog soul.
Four character settings that interact with the compressor settings (threshold, ratio, attack, release) can be engaged alone or in combinations to produce a number of possible compression flavors, and an independent saturation knob dials in variable amounts of harmonic distortion.
Side-chain filtering options, which can operate on an external signal or on the input signal itself and which allow variable amounts of low- or high- end filtering, multiply the options even further.
Metering, wet/dry control, and output gain round out the Drawmer 1978’s feature set.
“With the 1978, Drawmer has given engineers a tool that can impart a huge range of genuine analog signatures on their tracks,” said Brad Lunde, founder and president of TransAudio Group.
“Of course, Drawmer is well known for designing musical compressors, and the 1978 excels in that regard. But the addition of the character settings and the saturation knob put the 1978 in a class of its own. With the affordable, single-rack space Drawmer 1978 Stereo Tone Shaping FET Compressor, engineers get the versatility and sonic possibilities that otherwise would require multiple racks of high-end, analog gear.”
Wednesday, September 09, 2015
Switchcraft Announces New White XLR Connectors
The AAA series is fast and easy to install, with no screws and only 2 pieces to assemble.
Switchcraft has expanded its “AAA” series XLR and Tini-QG Mini-XLR connectors, adding white as a color option in addition to the existing black and nickel.
The AAA series represents nearly 70 years of experience making XLR connectors, and with no screws and only 2 pieces to assemble it is fast and easy to install.
Already a favorite of the lighting industry, the white option opens new possibilities. “Many architectural lighting installations utilize white hardware,” said Switchcraft’s OEM marketing manager Dan Sima.
“Our customers asked for a connector to match and this product delivers that.”
Switchcraft has also created a white version of their popular Tini-QG line of Mini-XLR connectors.
Switchcraft developed the original Mini XLR connector more than 30 years ago and it has been used in applications ranging from microphones to medical devices.
With options ranging from 3 thru 8 contacts, this product is expected to be utilized in audio, instrumentation, and many other applications. “Every day, professionals depend on Switchcraft for their most critical connections,” said pro audio and broadcast marketing manager Wendy Charak.
“We’re constantly listening to their feedback and making sure we have the broadest line of products in the world to meet their needs.”
White AAA Series XLRs and Tini-QG Mini XLRs are in stock at many leading distributors.
Posted by House Editor on 09/09 at 02:37 PM
Tuesday, September 08, 2015
Vocal Mixing Basics
In 1878, a room full of people watched Thomas Edison’s new phonograph spin and heard a voice read “Mary Had A Little Lamb.”
Despite the excitement of hearing the first audio recording, I’ll bet someone thought, “That sounds like crap.” Having heard the recording, I agree.
Mixing the spoken word is a task in itself, but to mix singers and blend them with a band is an even more daunting task. Singers produce a range of sounds, good and bad, and no two voices are alike. This means each vocal must be uniquely mixed. What works for one person’s vocal isn’t right for another.
The good news is that I’ve identified seven areas of vocal mixing to focus on that take a lot of the hassle out of the process.
Roll It Off
There’s no reason for low-end frequencies to be in a vocal channel. Unless it’s an acapella group, musical instruments such as the drum kit, bass, and to a lesser extent electric guitar should be the only things that populate the sub-200 Hz frequencies.
A vocal microphone can pick up these sounds, either directly or through stage monitors, as well as any extraneous low end from the singer. Remove these by using a high-pass filter.
The filter can be fixed-point, such as rolling off frequencies below a set point, usually in the 80 to 120 Hz range, or it can be a variable filter. My personal preference is to roll off at as high a point as possible. For example, I regularly work with a singer that needs the filter set at 180 Hz. My process is to roll it higher and higher until hearing a negative impact on the voice, and then pull it back a few hertz.
Male vocals can have excessive low end, so console functionality permitting, also take a 3 to 6 dB cut in the 250 to 350 Hz range. This eliminates the muddiness in most male vocals.
There’s no such thing as perfect singing voice. Even the best singers have slight imperfections in the sounds they produce. (Just don’t tell them I said that.) These imperfections are usually in the 2.5 to 4 kHz range.
Find the sweet spot to remove the harshest frequencies. With an analog console, use it’s sweeping-mid or a graphical EQ frequency selector. Start at the 4 kHz point and apply a 6 dB cut. Then slowly sweep that frequency down until the vocals clear up. Next, decrease or increase the cut as required.
Analog consoles have a fixed bandwidth and therefore the cut will affect frequencies centered on the primary selected frequency, though in lesser amounts, like an upside-down mountain. However, this bandwidth (Q) can be altered on digital consoles. The tighter the bandwidth for cutting the better, because harsh frequencies are best removed with surgical precision – though without the worry of a malpractice lawsuit.
Turn On The Lights
Add brightness to the vocal with boosts to select high-end frequencies. The boost creates a bright and sometimes airy sound. The amount to add depends on the style of music, the song arrangement, the vocal, and what sounds good in the room.
Apply a gentle boost of 3 to 4 dB above the 6 kHz point. Sweep this point up until it produces the desired results. This is easy with consoles that have more than one sweeping-mid. In the case of consoles without, use the peaking high-end EQ control to increase that boost for all the high-end frequencies.
Make It Smooth
Despite the previous steps, a vocal mix can still be wanting. The bad stuff’s gone, and it’s got some sparkle, but it’s not quite there. Enter Mr. Smooth.
There’s a danger zone in the mid-range. One wrong move and the vocals can sound flat and dull or harsh and annoying. Welcome to the 1 to 2 kHz range.
Sweep a tight cut in this range This can be more of a problem area than the 2.5 to 4 kHz range, so when limited to the number of frequency manipulations, opt for which has the greatest impact.
Bring Out The Bass
Some lower-mids might be needed to add substance to the voice. Boost in the 200 to 600 Hz range. As noted earlier, vocal characteristics vary widely, so while some singers might have plenty of energy in this range, others might be in desperate need of it. Don’t make them sound like someone they’re not; rather, the goal is to make them sound like a better version of themselves.
Earlier, I mentioned cutting in the 300 Hz range for male vocals. But doesn’t this contrast with the aforementioned tip on boosting? Yes. No. Maybe.
Mixing is a process of additive and subtractive measures. The difficulty is in deciding what to do first. I’ve found the most success in removing as much of the bad as possible, and then listening to what remains and boosting where appropriate.
A vocal that’s devoid of much in the 300 Hz range is a vocal that’s not going to have the natural muddiness and therefore might be a prime candidate for such a boost. This doesn’t mean muddiness is added. It just depends on the specific voice characteristics as well as the style of music.
Time to work on the other channels. Much of the natural voice is in the mid-range frequencies, and so are the fundamental frequencies of most other instruments. Part of mixing a good vocal is making room for it in the mix. The vocal needs to own the primary area where it shines through. This doesn’t come by boosting only the good – it also comes by carving out space from other channels.
Look to vocal and instrument channels that clash with the vocal. Determine which “owns” that primary frequency area, and then adjust the others by applying a slight cut in that area. I’ve gone back and found I had two channels where both had the same frequency boost applied – of course they clashed.
The One Question
Audio production is part science and part art where too often the scientific mind is allowed to dominate. This happens a lot with EQ work. During any of the above processes, you might ask the question, “Does this sound good?” The question (and it’s answers) come from trial and error. Boost here, evaluate. Boost there, evaluate.
There’s another way. During the vocal mixing process, imagine how the vocalist should sound. Ask these questions: What frequency areas dominate? What areas are minimal? How does it fit into the overall mix?
Then go to those key mix areas, such as using the high-pass filter or adding brightness, and apply those measures so they meet the sound in your head. A great vocal mix can be imagined and then worked towards. It’s much harder and less likely to be obtained through trial and error.
This process isn’t easy for those new to the EQ process and frequency band characteristics. But learning is just a matter of time and practice. The key is asking the one question that matters: “Does it sound like I want it to sound?”
Use the above mixing areas to improve vocal mixes. Once the vocal channel is sounding great, reach for the reverb. Or don’t. It depends on a few things, now doesn’t it?
Chris Huff is a long-time practitioner of church sound and writes at Behind The Mixer, covering topics ranging from audio fundamentals to dealing with musicians – and everything in between.
Friday, September 04, 2015
The Essentials Of Compression & Using Compressors
Some of the most common questions I receive about audio:
What is a compressor?
How do I use it?
What do the controls do?
Compression is like a finger on a volume control (fader). If a sound gets too loud, it’ll turn the volume down. If the sound is softer, it can bring up the volume.
You can also tell it to wait a few moments before it makes an adjustment, and you can tell it to wait for a little bit after the sound passes before the volume will go back where it was before. In a nutshell, this is what the controls on a compressor do.
Here’s a breakdown of each control on a compressor, and what it does:
Threshold: Tells the compressor at what audio (decibel) level to start compressing at. Meaning, if you tell it to start compressing at -6 dB, then when the volume of an instrument hits -6 dB, the signal will start to be compressed when it reaches that level.
Ratio: Works together with threshold. Ratio is the amount of audio signal allowed to go above the chosen threshold. For example, if your ratio is set to 2:1, flip 2:1 upside down and turn it into a fraction = 1/2. So if the threshold is set to -6 dB and the ratio is set to 2:1, when the audio signal gets up to -6 dB, the compressor will only allow 1/2 of the original signal above the threshold.
If the ratio is set to 4:1, it will only allow 1/4 of the signal, and if it’s set to 6:1, only 1/6 of the signal will be allowed over the threshold, and so on. If the compressor is set to infinity, then it has been turned into a limiter and won’t let any sound above the threshold. Check out this handy dandy Compression Ratio Chart for reference:
(click to enlarge)
Attack: Tells the compressor how fast (or slow) to compress the signal once it hits the threshold. This is useful if you want to maintain some of the “punch” and dynamics of a signal, but still want to control its volume.
Let’s say that you have a kick drum that you want to preserve its initial transient (the first part of the hit)—you can dial the attack to start a little bit after the start of the drum hit happens, and then compress from there. This will in turn let the initial hit of the drum pass through uncompressed, and then start compressing at the designated time you’ve set the attack to.
For example, if you set the attack to 20 ms (milliseconds), the compressor will wait 20 ms after the signal reaches the threshold, and then once that 20 ms has passed, it will enact the compression.
Release: Tells the compressor how fast (or slow) to end the compression after it no longer reaches the threshold. So if you set the release to 150 ms, it means that the compression will end or “release” 150 ms after the signal goes below the threshold.
Make-Up Gain/Volume/Output Volume: This carries various labels on different compressors, but they all mean the same thing. Let’s use the example of the kick drum again. Make-up gain will bring up lower sounds and make them more audible, while the threshold will keep the louder sounds compressed. This in turn levels out the signal and makes it more consistent and even.
It’s very common to turn up the make-up gain to the same amount of decibels that the compressor is compressing. So if the compressor’s gain reduction meter shows -3 dB, you can adjust the make-up gain to boost 3 dB on the output.
Knee: Only some compressors offer this function. The knee tells the compressor how soft or hard to compress the signal once it compresses. It’s called a “knee” because of the slope the signal will take towards compression looks kind of like the bend of a knee.
Just know that a hard knee is typically better for a bass instrument like a kick or bass guitar, whereas a softer knee would be great for vocals, strings, or cymbals. The lower the register an instrument produces in the frequency spectrum, the harder the knee can be set. (There are some cases where this is not necessarily true, such as a hard compressed “spitty” vocal or a trashy sounding drum overheads track.)
The softer the knee, the more gradual the instrument will compress when it reaches the threshold, whereas with a hard knee, the signal doesn’t gradually compress but rather goes straight into compression.
(click to enlarge)
Auto: Many (but not all) compressors offer this function. It adjusts the attack and release times according to the source material it “sees.” This is pretty handy if you’re a beginner and are unsure about attack and release times, but note that it can yield less than stellar results on some material. Some compressors have an auto button for the release time only. The dbx 160sl is a compressor with an Auto button:
Practical tips for beginners:
• The “magic” attack time is 12 ms. This works for a variety of instruments in the mid and treble range.
• When just starting out in learning how to use a compressor, adjust the threshold where you’re only compressing 3-4 dB on the gain reduction (GR) meter.
• For bass type instruments, like electric bass guitar and kick drum , use higher ratios, i.e., 6:1 or even 20:1/Infinity depending on the player and what you’re going for.
• Lower ratios for mid-range and treble instruments, i.e., 2:1 to 4:1
• Any time when compressing strings (like violin, viola, cello, etc), use lower ratios for a more “transparent” response.
• The faster the release time, the less audible compressor artifacts are—meaning the “squishy” sound of a compressor.
• Limiters are great for bass instruments and sub-groups.
• When turning a ratio to its maximum setting, i.e., 20:1, you essentially create a limiter.
• Turn up the make-up gain (output) the same amount of dB the compressor is compressing. For example, if you’re compressing to -4 dB, turn up the make-up gain the same amount.
• The slower the attack time, the more initial “hit” or “pluck” you will get from the instrument.
• For more punch on a kick drum, set the attack to around 15 ms to 20 ms. This will let the initial transient through.
Ratio times (very general guidelines):
1. Kick—6:1 for rock, 3:1 to 4:1 for jazz or country.
2. Snare—2.5:1 to 4:1 for rock, 2:1 to 3:1 for softer music.
3. Toms—3:1 to 4:1
4. Overheads—2:1 to 3:1 Typically it’s best to go light on compression of overheads.
5. Percussion—2.5:1 to 4:1
6. Bass Guitar—4:1 to 20:1 (Infinity) depending on what you’re going for and how consistent the bass player is playing.
7. Keys—2:1 to 4:1
8. Acoustic Piano—Same as keys. And be careful.
9. Rhodes—2.5:1 to 4:1
10. Organ—3:1 to 6:1, depending on the type of organ and how aggressively it’s being played.
11. Electric Guitar—2:1 to 4:1
12. Acoustic Guitar—2:1 to 4:1
13. Contemporary Vocals—2:1 to 3:1
14. Rock Vocals—2.5:1 to 6:1
15. Strings (Violin, viola, cello, etc)—2:1 to 4:1
16. Harmonica—2:1 to 4:1
Final notes: When first starting out, it’s better to gravitate toward lower ratio settings, only compressing 3 to 4 dB. If your compressor has an “Auto” button, there’s nothing wrong with using it until you understand how to set attack and release times. If you have a dbx compressor, feel free to use it’s “Overeasy” function for “softer” compression, unless it’s being used with a bass instrument of some sort.
Casey Campbell is an audio educator who heads up Sound Instruction (www.soundinstruction.net), a website dedicated to live sound training.
Thursday, September 03, 2015
Ultrasonic Hearing? Paying Less Attention To The Extremes
It’s one of those debates that seems to go on forever, and has been discussed heavily in professional audio and hi-fi circles during the “digital era” that began in the early 1980s with the advent of the compact disc: Can people hear anything above 20 kHz?
And if so, does it matter?
As it turns out, this has been an important subject in scientific and medical circles.
In a 2003 article by Martin L. Lenhardt, Au.D., Ph.D., entitled Ultrasonic Hearing in Humans: Applications for Tinnitus Treatment, the author states “Human ultrasonic hearing has been independently ‘discovered,’ documented, and abandoned more than a dozen times over the last half century. So outlandish is the concept that humans can have the hearing range of specialized mammals, such as bats and toothed whales, that ultrasonic hearing has generally been relegated to the realm of parlor tricks rather than being considered the subject of scientific inquiry.”
In this particular article, the concept of treatment for hearing loss via the use of bone-conducted ultrasound is explored and offered as a viable option. In his summary, the author states: “Humans can detect ultrasound up to at least 100 kHz, but perception generally requires direct contact of the source with the body.”
This is interesting, if perhaps not completely practical or even relevant for those of us in the professional audio field.
But it’s more relevant than you might think.
LAW OF DIMINISHING RETURNS
Of course we should all be concerned about the future of professional audio and the consumer formats used by our ultimate customers.
But at what point do these efforts start to create little or no benefit for the increased cost and complexity? (Don’t even start me on the format wars…)
And let’s get one thing straight: consumers don’t really care if the original sampling rate was 44.1 kHz or 192 kHz. Most people probably wouldn’t notice even if it was 22.05 kHz!
In fact, before the advent of the CD, the reigning consumer format was the cassette. Remember those things? You probably still have a few dozen or even a few hundred of them floating around, gradually becoming demagnetized audio horrors 1-7/8 inches per second at a time.
The compact cassette was never meant to be used for full-band audio, but instead was intended for speech only. But that didn’t stop it from becoming the number 1 format in terms of units sold during the decade between 1978 and 1988. There were even a number of movements in the 1970s to introduce improved versions of the cassette, with greater tape widths and faster tape speeds, all of which failed.
Dolby and dbx offered improved signal-to-noise performance but unfortunately were not universally used or understood. Why was the cassette so popular? Simple: it was inexpensive, portable, robust and recordable.
So what if the LP offered far superior audio quality if used properly? That last part is the key. For the average user, the LP was a fragile, expensive and frankly, crappy-sounding format that got worse with every play.
How many typical consumers did you know (or did your parents know) that had a really decent LP playback system? Sure, the potential was there but only very seldom was it reached.
Ironically, only after the advent of the CD did the true potential of the LP emerge during its “Indian Summer” during the late 1980s and early 1990s. Meanwhile, the cassette lived on in HUGE numbers well into the 1990s.
It was finally replaced by the CD - due to the superior audio quality. Wait! No, it wasn’t. It was replaced by the MP3 for the very same reasons that the cassette was popular in the first place: portability, recordability, and low cost. Remember Napster? The original one, I mean?
And what brought about this radical change was the increase in speed and power of the personal computer, allowing for the heavy number crunching required to turn PCM flles into the much smaller, compressed versions to save flle space and make Internet transfer practical. That’s right—MP3s were around for several years before the iPod came along.
Admittedly, the iPod made MP3s portable and truly user-friendly in the same way that the Sony Walkman did for the cassette. But the point is that sound quality was NOT what drove these formats, it was convenience features.
WHAT ABOUT THE IN-BAND CONTENT?
And here is the real kicker for our part of the industry, meaning sound reinforcement. I’ve mentioned this before, but it bears repeating: the equipment is rarely the problem in most of our situations. Sure, there are times when we are limited to using sub-standard or inadequate gear.
But hopefully, we are all wise enough to avoid those situations for the majority of our work. And at the same time, we all hope that our junior and less-experienced brethren become increasingly savvier and demand proper equipment, proper set up time, and generally aim to produce better sound.
Here’s the bottom line: our problems aren’t generally the lack of super-wide-band system performance, but instead what we often don’t get right between 100 Hz and 10 kHz.That’s right, the 6.5-octave range in the middle of the approximately 10-octave range our hearing offers the1youth among us.
And if we want to come right down to it, the really super-critical part is between 200 Hz and 7 kHz. Looking at the Equal Loudness contours first detailed by Fletcher and Munson at Bell Labs, we know that by far, the human hearing system is most sensitive in this range, with a distinct peak centering on 3-4 kHz.
Of course we want the bass to shake the walls, and of course we want the vocals to sparkle and of course we want the drums to pound everyone’s chest at the appropriate moments and of course we want the cymbals to shimmer. But really, if we can’t get things right between 200 Hz and 7 kHz, we aren’t doing our jobs properly.
Why is this range so important, beyond the obvious fact that our hearing system is most sensitive in this range? Well, first, this is where more than 98 percent of the vocal energy lies… and the drums and the guitars and the strings and the horns.
But the vocals are truly the most important thing, whether we’re doing a rock tour, a corporate event, or sound at our church. The audience can detect anomalies in the vocal sound much more readily than with any other source simply for the fact that we all hear vocals in speech every single day unless we’re in solitary confinement.
And even then, we hear voices in our heads. How many crazy people claim to hear snare drums in their heads? How many claim to hear guitars in their heads? No, it’s voices! Like the one telling me right now that I’ve beat this dead horse to a pulp.
HOW IT ALL TIES IN
Which brings me to my last point: that hearing loss and the sound reinforcement industry are not unrelated. How many of us already show signs of some loss, or of tinnitus? I’ll bet more than a few, including myself, have this problem. And what about the people who attend your events?
Hopefully we, and our audiences, won’t be so bad off after a few years listening to our mixes that they need some kind of ultrasonic system installed in their heads in order to hear the first words of their grandchildren.
So instead of concentrating on “digital vs. analog” and whether or not our microphones reproduce signals up past 50 kHz, let’s worry about what really matters: good sound in the mid-band.
Karl Winkler is the vice president of sales/service at Lectrosonics and has worked in professional audio for more than 20 years.
Wednesday, September 02, 2015
Brad Englander Selects The API 1608 For Bassmental Studios
Studio owner matches 1608 analog console to existing inventory of 512c preamps for Virginia based studio.
When Brad Englander was deciding how to expand his gear list to include a console, he was certain about one thing: “all roads led back to API.”
At the time, he already had four 512cs in an API lunchbox, but wanted even more of the “same clean and punchy sound,” so he turned to the 1608, along with a 2500, for the solution.
Englander owns “Bassmental”, a studio he runs from his home in Great Falls, Virginia.
The studio works largely with local bands and solo artists, across genres like rock, Americana, blues and country.
Englander says he has also recorded children’s music, audition recordings for young violinists and cellists, and language training narration sessions. While learning the ropes of his 16 channel, automated 1608, he has been “re-mixing some previously recorded materials and recording narration projects.”
When API spoke with Englander, he was excited to be mixing outside the box.
“The ability to use analog EQs, compressors and effects processors, rather than relying exclusively on digital plugins, is a new direction for me.” Even while phasing in the new console, Englander was already intrigued with his API modules. “I love the sound of the 512c preamps,” he says, which was what inspired him to find out more about analog consoles. “I read a lot of reviews, got a lot of advice, and researched the features of a number of consoles. All roads led back to API and the 1608.”
Englander has been able to blend his old studio setup with the flexibility of the 1608.
“Previously, I used a variety of pre-amps for tracking; the 1608 gives me the ability to track and mix more channels simultaneously using analog EQ’s and compressors.” While he managed to work with his old setup, he says the 1608 makes things much more efficient, even as he is continuing to learn.
“All of my pre-amps and processors are easily patched in. The aux sends are great too. I’m still configuring and learning the automation system; I expect it to be a hugely helpful tool.”
Englander closed out, letting us know that even the artists “have commented that the cue monitoring and playback have improved since I installed the 1608.”
Tuesday, September 01, 2015
It is said that “a picture is worth a thousand words” and nowhere is this more applicable than when trying to teach complex concepts.
A graphical depiction can often convey an idea better, and quicker, than a whole bunch of words. This is because our brains are mainly image processors, not word processors; the part of our brain that processes words is actually very small in comparison to the part that processes visual information.
Therefore visual cues help us to better store and retrieve complex information.
Bearing this in mind I’ve been exploring various ways of representing key audio concepts and terminology visually.
This invariably involves a certain degree of simplification but I think the results are a useful weapon in the battle against incomprehension.
Let’s start by looking at a simple way to represent the frequency content of a single sound, such as a kick drum, shown here against a vertical axis denoting frequency (Figure 1).
On the left is a representation of a kick drum that has been miked in a standard way, with a single microphone poking through the hole in the front skin. Here we can see there is slightly more energy in the bottom and top of the sound (i.e., the thud and the click) than there is in the middle – quite common when close miking a kick drum.
The right side of the image represents the same signal after a little EQ has been applied, in this instance the bottom end has been enhanced while the lower and upper mid range frequencies have been reduced slightly to give that classic kick drum sound.
EQ isn’t the only way we affect the frequency content of sounds so let’s take a look at some other methods (Figure 2).
On the left is a representation of a snare drum that has been miked up in standard manner – a single mic above the top skin. In the middle is the same snare with a high-pass filter applied, as indicated by the fade (which denotes the gradual reduction in the lower frequency content). On the right is the same snare after compression has been applied. In this instance the compressor is limiting not just the dynamic range of the snare (which is difficult to depict in a static image) but also it’s frequency content, resulting in a tighter and punchier sound at the possible expense of some of the finer detail.
Now that we’ve established a simple way to visually represent the different sounds, and the ways in which we can affect them, let’s take a look at a full drum kit. The kit as a whole has the widest frequency range of just about any instrument (with the possible exception of the pipe organ), from the low thud of the kick drum to the fizzy sparkle of the cymbals.
It comprises multiple elements that all need to be miked up in a way that enables us to treat each individual sound in relative isolation such that when they are combined, they complement each and work together as a whole. If we take a standard four-piece drum kit, miked up in a standard way (i.e., a single mic on each drum with a pair of overheard mics), and just bring up all the faders, it might “look” something like Figure 3.
I’ve now added panning information to the horizontal axis to denote the positioning of the sounds within the stereo field (at the moment everything is panned centrally). The one thing that this depiction makes obvious is the clutter that occurs where the sounds overlap each other, particularly in the mid range where the kick, snare and toms all produce sound energy.
This is a common cause of “muddiness” in the drum mix – something which can quite easily be addressed with a little EQ and panning (Figure 4).
Here we can see that a bit of EQ has been used to bring the bottom end of the kick drum out, the kick has also been compressed and a low pass filter has been applied. The snare has also been compressed and a high pass filter has been used to tame the lower mid energy. The toms have been EQ’d and panned to create some space for the snare and mimic their physical placement in the kit. The cymbals (or overheads) have been high passed and panned wide to give the top of the kit a nice sense of width.
Overall you can quite clearly see that the processing, while quite subtle, has created space for each individual element of the kit so they can be clearly heard but also so they work together and complement each other. This avoids the muddiness that can so easily bog down the drum sound and gives the kit the clarity and definition that will help it to sound good, even in the busiest of mixes.
Speaking of which, let’s take a look at other common mix elements in the form of a simple three guitar set-up, i.e., bass, rhythm and lead guitars (Figure 5).
This depiction shows the potential for a messy sound in the lower mid range where all three instruments produce energy. We can also clearly see the masking that commonly occurs when the fundamental frequencies of the guitar overlap and obscure the harmonics of the bass. Guitars also tend to have a pronounced low end as a result of the use of directional microphones in close proximity to speaker cabinets which exacerbates the proximity effect.
Thankfully this can all easily be fixed with high-pass filters and a little panning (Figure 6).
First, the bass has been compressed, which helps tighten it up and enables it to rumble away at the bottom of the mix without jumping out or dipping down. The guitars have been high passed and panned, which not only creates more room for the bass but also helps them to come across much more clearly. This may result in the guitars sounding slightly thin when listened to in isolation, but when combined with the bass, both instruments will come across much better while complimenting each other nicely.
Now that we have the tools to depict key elements and processing lets take a look at the mix as a whole.
Any musical performance that features more than one melodic or rhythmic element, be it live or recorded, requires these elements to be mixed.
Traditionally music was performed live and the mix was achieved by intelligently positioning the individual mix elements and augmenting where necessary. (You want the violins to be louder? Get more violins!)
In the early days of recording, where the performance was captured completely live with one or two microphones, a mix could be achieved by moving the musicians relative to the microphone(s), often during the performance to create dynamic variation.
In live performance we’ve evolved methods whereby each key element is miked or taken direct individually, so that we can treat them individually before combining them together in the mix.
So what happens if we combine our drum kit with the guitars and throw a vocal on top? (Figure 7)
It’s starting to look a little messy now but a mix is a complex interaction of multiple elements, so that’s quite normal. The vocals have been compressed to narrow their dynamic range and help them sit on top of the mix, but there’s still a risk of them being swamped by the other instruments that have energy in the same frequency range (such as guitars). Being as the vast majority of music we mix at gigs is song based, ensuring the vocals can be heard is a key challenge.
So what can we do to make them stand out a bit more? (Figure 8)
The answer, of course, is reverb, depicted here using an outer glow. They key is to use a reverb that’s small enough to ensure the vocals don’t sound too distant while exploiting the spectral smearing and stereo widening that can help to make the vocal sound bigger. A plate with a reverb time between 1.0 and 1.6 seconds and a pre delay below 10 ms usually does the job. I sometimes find that rolling off the top end (or adjusting the high ratio) of the reverb helps to make it sound more subtle – less like a digital reverb and more like a natural acoustic space around the vocal.
But that’s not the only way to make a vocal stand out in a busy mix (Figure 9).
Another trick I like to use is to apply a slap-back delay to the vocal, here depicted by the black drop shadow. A slap-back delay can be anything between 50 and 300 ms, but I find a setting of between 100 and 180 ms works particularly well on vocals in mid-tempo songs.
The feedback gain should be set so there is only one audible repeat – typically achieved by setting it to 10 percent. This creates a single echo very close to the vocal which has the effect of doubling it and helping it stand out against the background noise. Again rolling off the top end of the effect can help make it less obvious and more subtle.
As mix engineers we’re always striving to build better mixes, so hopefully these pictures have been worth a few thousand words and have provided a unique insight into the process. We might think we mix with our ears, but our brains are doing all of the hard work, so anything that can help us visualize such abstract concepts will enable us to better understand the nature of the mix and produce consistently high-quality results.
Andy Coules (andycoules.co.uk) is a sound engineer and audio educator who has toured the world with a diverse array of acts in a wide range of genres.
Oxygen College Outfits New Studio With Audient ASP8024 Console
The Geelong West music facility will use the analog console predominantly by the Advanced Diploma of Sound Production students.
Increasing numbers of students at the audio engineering branch of Oxygen College have resulted in the commission of a brand new, larger studio where a 36 channel Audient ASP8024 mixing console with Dual Layer Control (DLC) and patchbay now takes center stage.
Newly opened in May, this latest addition to the Geelong West music facility is in the care of head of audio Tom Isaac, who explains, “It is used predominantly by our Advanced Diploma of Sound Production students.”
He continues, “The students and staff are very excited and cannot wait to get in there to record. We have other analog studios here at the college but none as large as this.”
Keen to develop the students’ understanding of signal flow, the fact that the studio is analog is definitely advantageous, reckons Isaac. “The Audient desk fits in with our needs for a training platform which teaches students ‘out of the box’ mixing as well as being a great sounding desk.”
“In this very digital age it’s important that students still understand analog workflows, routing, gain structure, TT patchbays etc. It’s amazing to see their reactions when we say: ‘OK, lets mix this, but you cannot use ANY plug-ins. Only the Audient and the outboard in the rack. It really gets them thinking about their mixing craft and how records have been mixed in the past.
We have some great outboard units from TubeTech, Universal Audio, Manley, API, Drawmer, Lexicon and Eventide to work with giving students all they need to pull together a great sounding mix.’
“Flexible routing is one of the key strengths, having so many inputs available on mix down is the main reason the ASP8024 got the job. Our students are very excited about this feature.” Keeping the students’ minds open to analog has certainly had an interesting effect, as he explains: “Whilst they appreciate our studios based on a control surface, ProTools and ‘in the box’ mixing, a lot of students are very passionate about classic analog outboard and an analogue summing environment. They may not be able to pinpoint specifically what exactly they like better about the sound of an analogue summed project versus an ‘in the box’ summed project, but as their critical listening skills develop, it’s been a great conversation starter.”
As the largest studio at Oxygen College with a large live tracking room, plus vocal and isolation booths, it helps to teach the finer points of running a commercial facility: they have simulated budgets, worked out a cost per hour of hire and have strict deadlines for recordings. They also work in tandem with Advanced Diploma of Music students, who release EPs as part of their course.
“The Sound Production Diploma students record and mix these with guidance from their trainers. Some go on to be released and enjoy radio play,” says Isaac.
The Audient console’s patchbay is getting a thorough workout as well. “Students are very much looking forward to combining their favourite plug-ins with some high-end outboard dynamics that they can insert via the ASP8024’s patch-bay.”
Oxygen College has been a creative arts training hub for five years now, providing state-of-the-art, industry-standard and ‘real-world’ education, for a range of arts including music, photography, painting and drawing. The college’s commitment to a low student-to-facility ratio, ensures students have excellent access to studios, equipment and training staff.
Posted by House Editor on 09/01 at 09:25 AM
Friday, August 28, 2015
Creative Console Strategies
The first “big” mixing consoles I owned were a 12-channel Kelsey and a 16-channel Yamaha PM1000. The Kelsey saw the most use because the PM1000 weighed in at 110 pounds, and that was without the wooden case I built for it.
With a limited number of channels, buses and features available, I learned to be quite frugal when deciding what to mike onstage. For larger shows, the Kelsey sometimes served as a sub mixer for the drums and bass, feeding the Yamaha.
One day a buddy asked me to mix on his rig at a large outdoor jazz festival. It sported a 32-channel PM1000, and I was in heaven for two reasons. First, he didn’t ask me to help lift or move it, and second, I didn’t have to pick and choose what to put in the PA. With 32 inputs I could mike up everything onstage and still have empty channels.
Lacking, however, were monitor buses, but it was a problem easily solved back then by routing inputs to one of the four mix matrix buses and using those to feed stage wedges. Not as ideal as having individual aux sends on every channel, but musicians were aware of technology limitations and were happy to get more than one mix in those days.
Fast forward to today. One of my small digital consoles offers 66 processing channels and up to 14 mono buses in a rack-mount form factor. With onboard GEQs, FX units, comps and gates, there’s no need to carry outboard gear, and it can be picked up effortlessly.
But as full-featured as these smaller boards are, bigger is often still better because clients always seem to need another feed or send somewhere, and there’s almost always extra inputs that show up at the last minute.
An 8-channel version of the author’s 12-channel Kelsey console, circa the mid-1970s.
Regardless of the size of the console, sometimes we have to be a little “creative” to get the desired results. Here are some things I do.
When running both front of house and monitors from the same console, it means that the monitors either share the same channel EQ dialed in for the mains (post EQ sends) or they do not get any EQ at all (pre EQ sends). This might not cut it for a picky performer or an acoustic instrument.
What I do is use a simple splitter box to send the microphone or DI to two channels instead of one. The first channel is for the house mix and the second (usually adjacent to the first) can be “dialed in” with an acceptable EQ for the monitors.
On smaller acoustic shows, I might place every input into two channels, effectively providing separate house and monitor consoles. If there aren’t enough splitter boxes handy, I can use a channel’s direct output to feed the second channel. On a digital console that offers channel patching, simply patch an input to more than one channel in the menu.
I’ve also used a second channel for singers who want a significant amount of effects in their monitors but don’t want to hear the effects when they’re talking in between songs. Sure, I could mute the FX masters, but on most of my consoles they’re on a different layer by default. Using the second channel for effects to the monitors, I can simply press the convenient mute button to stop the effects as needed.
It’s also easier to dial in a good mix of “dry” verses “wet” vocals in the monitors because I can simply send dry effects to the monitors from one channel, and then wet it up as needed with the second channel.
Another use for second channels is making a killer board recording. Many of us make recordings of live shows, and there are a lot of ways to do it. “Down and dirty” board tapes can be had by taking a copy of the main L+R outputs and sending them to a stereo recording device.
Newer digital consoles may offer the option of recording a stereo feed to a USB drive, but the mix and some instruments may not sound “right” because they were equalized and balanced to be heard through the PA rather than a recording.
Multi-track recordings can be attained by sending the channel direct outputs to a recorder or splitting off the inputs with a splitter snake or grabbing the inputs off a digital network – but this involves using a stand-alone multi-track recorder and possibly a lot more extra gear.
Sometimes all that’s needed is a good stereo board tape. Sure, you can set up a mix using an aux send, but this raises the problem of sharing the EQ with the house PA. Using second channels on instruments or vocals that have been “overly adjusted” to sound good in the house can result in a better recording because you can have control over difficult stage sounds as well as EQ directly for the recording.
For years I carried around a line-level distribution amplifier in my rack because I was always running out of outputs on the corporate-type shows that make up the majority of my work. I might only need a few inputs but dozens of outputs for the main loudspeakers, delay and fill loudspeakers, as well as feeds to the venue for underbalcony fills, lobby systems, overflow rooms, onstage and backstage monitors, video and safety recordings, intercoms and dressing rooms, etc.
This is why I gravitate to consoles that have extensive matrix sections. In its most simple form, a matrix takes a selection of inputs (usually derived from the group and main output buses) and allows routing of those signals, complete with level control, to a series of outputs. Complex matrix systems offer the ability to choose from a variety of inputs, including specific channels or external sources, and may supply processing that includes EQ, compression, limiting and even signal delay.
Even when you don’t have this many channels and options, a matrix section can considerably expand flexibility.
A matrix adds a ton of flexibility to a console and gives the user a lot of easy solutions to routing problems, like adding a support act console.
While there are many ways to tie two or more consoles into a single PA, more than a few times I’ve simply patched the support act console into the external matrix input on the main console and fed the PA from both consoles through the matrix out.
Another good use for a matrix is creating mix-minus feeds. This refers to a program feed that has been remixed to exclude one or more input components. Sometimes the video people might want the program audio minus the playback audio they’re sending to front of house, or teleprompter operators want to hear the program feed but with less music. I can whip up a quick mix-minus by routing the various parts of the program through subgroups and into the matrix. Levels of each feed can then simply be adjusted as needed.
A trend I’ve been seeing of late is providing audio feeds for remote meetings. I need the audio from the remote site in the PA, but don’t need to send it back to them. so I’ll create a mix minus the remote audio by using the matrix, and then add processing like leveling and compression before sending to the remote site.
More than a few times I’ve been mixing monitors from a smaller front of house board and have run out of aux sends. Using a matrix, I’ve set up side fill mixes as well as individual performer mixes.
One great feature about larger consoles is, of course, more channels to use. I can employ back-up mics or run back-up lines without having to re-patch.
Ever wonder why there are two mics on the podium at high-level events? One is normally not on, used as a spare that’s already in place in case there’s a problem with the main podium mic. Simply un-muting the spare keeps the show rolling. It’s the same reason we place two lavalier mics on important presenters at corporate shows or on lead actors in theatrical performances.
Note: sometimes two different pattern podium mics are used, like a cardioid and supercardioid, with the mix engineer choosing between the two, depending on the person speaking.
Larger channel counts allow me to do some things normally not pursued when I’m running out of inputs. For example, instead of choosing between two overheads, or a single overhead and a ride cymbal mic, I’ll probably use two overheads and a ride cymbal mic on the kit. Same with snare drum. With enough channels, I often opt for a bottom snare mic to pick up the snap, capturing a better overall drum sound.
Even smaller-format consoles like this Yamaha QL1 afford capabilities not imagined a relatively short time ago.
Extra channels can also be turned into an ad hoc intercom system. I place a mic at front of house, plug it into a spare channel and send it to a powered loudspeaker placed backstage via an aux send. A mic placed backstage is routed to a second spare channel, and by pressing that channel’s PFL button, I can hear the person backstage on my headphones. Not perfect, but when the intercom power supply loses its magic smoke 15 minutes before a cue-heavy show, you do what you have to do.
One more use for extra channels is “phantom mixing.” Ever get to the point where you’re satisfied with the mix and then a person walks up to FOH and tells you that they can’t hear their girlfriend, boyfriend, wife, child, niece, etc.? A quick twist of an unused channel knob and a sincere “Is that better?” usually gets them out of your hair…
Senior contributing editor Craig Leerman is the owner of Tech Works, a production company based in Las Vegas.
Thursday, August 27, 2015
Preventing “Hollow” Sound Through Microphone Techniques
Suppose you’re reinforcing a singer/guitarist in mono, with one microphone on the singer and another mic on the acoustic guitar.
The vocal sounds funny - sort of hollow or filtered. What’s happening?
Both microphones are picking up the singer, with the mic for the guitar about one foot farther from the singer’s voice than the mic for the vocals (Figure 1, below).
So there are two vocal signals in the mix – one is direct and the other is delayed.
When a signal is combined with its delayed replica at equal levels, certain frequencies cancel out, depending on the length of the delay.
In the frequency response of the combined sounds, “notches” occur at frequencies where the sounds cancel out each other.
This is called a comb filter effect, because the frequency response looks like the teeth of an inverted comb.
In general, if two mics pick up the same sound source at different distances, and their signals are fed to the same channel, it might cause phase cancellations.
These are dips or notches in the frequency response caused by sound waves at certain frequencies combining out of phase.
The result is a colored, filtered tone quality.
3 To 1 Rule
To reduce phase cancellations between two mics, follow the 3 to 1 rule: The distance between mics should be at least three times the mic-to-source distance (Figure 2).
For example, if two mics are each 4 inches from their sound sources, the mics should be at least 12 inches apart to prevent phase cancellations.
How was the 3:1 rule determined? When you add a signal to its delayed replica at equal levels, you get severe comb filtering with deep notches.
But when you mix direct and delayed signals at different levels, you get less deep notches.
Specifically, if the delayed signal is 9 dB less than the direct signal, the comb-filter notches are only +/- 1 dB, so for all practical purposes, they are inaudible.
How do we make sure that the delayed signal, picked up by a distant mic, is at least 9 dB below the direct signal picked up by the closer mic?
Put the distant mic at least three times farther from the source than the close mic is. Due to the inverse square law, the level drops 9.54 dB when the distance to the source is increased three times.
So the 3:1 rule insures that the level at the distant mic will be down at least 9 dB, so the mixed signals will have comb filtering of +/- 1 dB or less.
A ratio of 4:1 or more is even better. The 3:1 ratio is the minimum to avoid audible comb-filter effects.
Suppose the close mic is picking up a loud voice, and the distant mic is picking up a quiet acoustic guitar.
You’ve placed the mics following the 3:1 rule, but you have to turn up the guitar-mic gain a lot because the guitar is so quiet.
If so, the 9 dB separation might be negated. That is, the vocal signal in the guitar mic might be less than 9 dB below the vocal signal in the vocal mic, because the guitar mic’s gain is so high.
So there’s more to it than just the 3:1 placement. The idea is to get at least 9 dB difference between mic levels for the same instrument.
And remember - you want at least 9 dB of separation, not exactly 9 dB of separation.
Tips & Techniques
Here are some ways to prevent phase cancellations between mics that are fed to the same channel:
Place mics close, then turn down the excess bass with EQ.
Spread instruments farther apart.
Use a pickup on the guitar instead of a mic.
Delay the vocal mic signal by about 1 msec (millisecond). Then it will align in time with the vocal signal picked up by the guitar mic.
Use directional mics, and angle the mics away from each other. For example, aim the vocal mic up and aim the guitar mic down. If the close and distant mics are two cardioids aiming opposite directions, the mics can be closer than 3:1 and still get enough separation.
Use coincident directional mics, aimed up and down, so that the vocal signal arrives at both mics at the same time.
Another tip to prevent phase cancellations: Don’t use two mics when one will do the job.
For example, use just one mic on a lectern. If you must use two mics mixed to the same channel, place them so their grilles touch, one above the other. That way, there is no delay between their signals, and thus no comb filtering.
What if two mics pick up the same instrument at different distances and they are NOT mixed to the same channel? The result is stereo images, rather than phase cancellations.
The location of the instrument’s image between the house loudspeakers depends on the delay between mics, the levels at those mics, and where they are panned.
Suppose one mic is panned hard left and the other is panned hard right. If the delay between mic signals is 0 msec, and the level is the same at both mics, the image will appear in the “center” between the loudspeakers.
If the delay is 0.5 msec, the image will be about halfway off-center. If the delay is 1.5 msec or more, the image will be at one loudspeaker.
AES and SynAudCon member Bruce Bartlett is a recording engineer, audio journalist, and microphone engineer. His latest books are “Practical Recording Techniques 5th Ed.” and “Recording Music On Location.”
Wednesday, August 26, 2015
Clarifiying Common Misconceptions About Sound
Over the years, I’ve heard people tell me a lot of things that they believe to be true, but aren’t.
They hear it from other musicians and pass it on and pretty soon, people start accepting it as absolute fact.
The actual truth gets buried in history and that’s the way legends are born. It’s charming, but inaccurate.
Let’s examine some of these beliefs.
Bass drum ports
There are a lot of drummers that cut a small hole (usually around 4 to 6 inches in diameter) in their front head to “port” their drum.
Somebody may have told them that this tunes the drum (like a bass reflex speaker) to improve the bottom end. Is it true? Yes and no.
Cutting a hole will provide a vent which can be tuned to resonate the air inside the drum, but that’s what the second head does anyway - it’s like a passive radiator, driven from the pedal head.
Putting a hole in the front head is kinda like putting a hole in your speaker to port your hi-fi system.
So, why do drummers put that hole in their drums? Primarily to use as an opening for a kick drum mic, without removing the head entirely.
A lot of drummers still print the band’s name on their drum head and that’s important to them. That hole is important when you walk into a studio to record. Live drummers saw their studio counterparts using the hole and thought it looked cool, and adopted it.
Where the hole is located is very important, but not for any reason you’d normally think about.
It should be above the center line of the drum, so that a short mic stand will work, and the mic stand boom arm angle will let the engineer position the mic to point directly at the spot where the beater hits the head.
The hole diameter should be around 6 inches to allow for various size microphones.
The center of the hole should be above the center line of the drum, so that the entire opening is in the upper half of the drum. Any 6 inches opening above the 9:00 to 3:00 line will work.
Tubes are better than transistors
When transistors first appeared, their distortion characteristics were very different than tubes.
Once you exceeded their output range, they simply gave up, all at once and distortion went straight up very quickly.
A transistor amp which hit 100 watts at .05 distortion might put out 110 watts, but at 35 percent distortion. A tube amp distorted slower and more gracefully, often generating 2nd and 4th harmonics - which made the sound even better.
The newer breeds of MOSFET transistors a able to mimic this kind of distortion, and the gap narrowed.
With the new breed of computer modeling amps, and some of the new DSP chips, the gap between tubes and transistors is getting even narrower.
Tubes are noisier than transistors
Nope, it depends on the circuit. You can build ultra-low noise tube circuits if you’re willing to take the time to do it right.
And let’s get rid of the tubes won’t reproduce high frequencies myth too.
For many years, tubes ruled the high frequency roost in the megaHertz range.
The main advantage to transistors over tubes is less heat, less susceptible to shock and vibration, and now, lower cost.
There’s no difference in cables and cords
Somewhat true for loudspeaker cables, once you get past the teeny size wires.
Not as true for guitar and audio cables. Bad shielding, high capacitance, and poor construction can seriously degrade your sound in any cable carrying low level signals.
There are now even some wire companies selling “directional cable,” which is pure bull. Basically, it’s all just hype.
Different batteries sound different
Hmmm. Some people swear they can hear a difference in batteries. I remain skeptical of their claim.
Some batteries do put out more current then other batteries and that might change the sound but I think different batteries of the same actual voltage and peak current output should sound the same. The jury is still unconvinced on this one.
A condenser microphone is the best kind of mic
Best for what? If that was really true, they would use nothing but condenser mics in major studios. They don’t.
Every studio has dynamic mics, like the Shure SM 57, the AKG D112, the Sennheiser 421, and usually several ribbon mics and a wide assortment of general purpose mics.
Why? Because there is no such thing as the one perfect mic for everything.
For big ballads, it’s hard to beat the sound of a great big diaphragm condenser mic like the old Neumann U47, which now sells for around $10,000 in primo shape.
But even that mic occasionally gets beat out by a Shure 58 or an old ribbon mic for some voices or some songs.
A good engineer doesn’t go by price - they will pick whatever works best for that particular sound.
Amplifier wattage should match loudspeaker wattage
Usually you can double the wattage of the power amplifier to prevent clipping.
But if you’re going to be playing loud, invest in a good stereo compressor to go across the output of the mixer to prevent huge spikes from blowing the loudspeakers.
Guitar amps and speakers: 4 ohms, 8 ohms, or 16 ohms; is one impedance better than the others?
Impedance is simply the working load the speakers put across the amplifier’s output terminals. Maximum safe power transfer occurs when the amplifier is correctly matched to the speaker load.
On stage, multiple loudspeakers can provide more output compared to a single speaker, and also provide increased power handling, but these factors aren’t all that important in the studio.
Many of the big groups use a little 15-watt amp with a single speaker to record.
Loudspeakers can be wired in many different combinations so that is why many amplifiers have impedance switches on the back of the head.
Which impedance is best? Any of them will work fine as long as the head impedance is set correctly - it depends on the loudspeaker configuration you’re using.
Electric bass - how low does it really go?
The main output of a bass E string is primarily around 84 Hz, not the 42 Hz most bass players imagine.
The reason is simple; the string length is too short to produce much fundamental. Yes, it produces some 42Hz, but most of the sound is an octave above that.
Which brings up the next question; how do you get more bass out of a system? It’s very simple - you need to move more air.
Low bass must move more air, so the answer is more power (to make the speakers you have move further), more speakers (so that each speaker doesn’t have to move as much), or a more efficient ported or horn-loaded cabinet (so that the port and/or horn adds more air motion).
Harvey Gerst is a long-time recording engineer and owner of Indian Trail Studios in Texas.
Thursday, August 20, 2015
Cenzo Townshend Chooses Audient For Decoy Studios
Former owner of Electric Landlady mobile recording, Cenzo’s new studio is packed with high-end, esoteric gear including an ASP8024 console.
Two-time MPG mixer of the year award winner and British analog gear aficionado, Cenzo Townshend has been wall-to-wall busy since the opening of the new Decoy Studios last summer.
Set in the picturesque countryside of Suffolk, Cenzo’s studio is packed to the gills with high-end, esoteric gear, alongside which is his trusted Audient ASP8024 mixing console.
“It’s my dependable rock. It’s been all over England with me back when I had the mobile studio, Electric Landlady” he says, adding, “If that console doesn’t work then I know there is something fundamentally wrong with the studio.”
Housed in one of the two large control rooms, he continues, “We use it for tracking and overdubbing, but also for mixing - it’s getting a lot of use every day.”
When asked why it was so useful in this capacity, he explains, “It’s a very stripped down console with an awful lot of features. It’s very dependable, robust and sounds great. We try a lot of different equipment in that room too, most recently we’ve been using the new iZ Radar so we’ve had a lot of people coming in to do listening tests. It always sounds great through the Audient.”
A broad range of artists come through the doors of this recording studio, from unsigned bands, rising star Rhodes [Ministry of Sound], to George Ezra and The Maccabees - even pop’s Robbie Williams was mixed here last month. Producer, Richard Flack did a 5.1 mix of the show using Audient’s ASP510 surround sound monitor controller. “It all happened in the Audient room,” adds Cenzo.
He’s a fan of the smaller Audient products, too. “I’ve not had anything from Audient that isn’t a good box,” he says, citing the Centro and a more recent purchase of the iD22, although he does confess to not having used the audio interface as much as he could have. “We keep lending our iD22 to clients, actually. If they come in with their laptops and need to plug something in or work next door, it’s ideal. The Maccabees had it for about a year! (They have since given it back and bought their own.) I love it though, it’s so beautifully built - everybody that picks it up says the same,” he adds.
Of course he loves it, the mic preamps are the same as those on the ASP8024 console, and he describes them as, “...really good and solid. We’ve got Neve preamps as well, but the other day we recorded a band and we ended up just using all Audient for speed and reliability. We didn’t plug anything else in. It’s great to be able to do that with the Audient.” He likes how they sound too: “The current album we’re working on with Rhodes is a very pure sounding record, very organic - the Audient desk is really helping with that.”
When pushed, Cenzo admits he is still very proud of winning an MPG award for best mixer two years in a row; in particular the moment he walked up to collect it and Brian Eno shook his hand. “That will certainly stay with me,” he says wistfully.
Looking back over his award-studded mixing career, Audient asked what advice would he give his 20 year-old self. “Learn to listen. Listen to instruments, bands; go to classical orchestral concerts and really listen to what instruments sound like. Hear how musicians balance internally without somebody turning faders up and down,” he says.
“And learn to record - properly. A lot of people just want to mix, overlooking the skill and necessity of making a good recording, until they start mixing and realise that it’s badly recorded and impossible to mix well.
“Also, listen to records - old records (I mean vinyl of course) - it gives you a completely different perspective. It’s very grounding to listen to that and to hear what you’re actually up against.”
Wise words from the ever lovely Cenzo there, who can be found at Decoy Studios or his brilliantly quirky personal website.
Wednesday, August 19, 2015
Cut ’Em Off At The Pass: Effective Uses Of High-Pass Filtering
So there I was, system engineer at a county fair gig. The act of the day was a traveling ’60s reviews with three or four artists who were, shall we say, past their prime.
They weren’t carrying engineers, so we got the duty.
Soundcheck went fine. The artists cruised through their paces and the hired back-up band was surprisingly good. Nothing to do but hit catering and wait for the “white hair, blue hair and no hair” crowd to show up.
Show time. The band started the intro, everything was rocking in an old school sort of way and the emcee/star came out. He was much more animated than he had been at soundcheck – running around the stage, exhorting the crowd to put down their walkers and dance, generally getting them in the mood.
Suddenly I heard a phantom kick drum that was waaaay off the beat. I cued up my cans and began to solo channels.
The offending thump came and went, but I finally put my eyes and ears together and realized that the star, we’ll call him “Frankie” for the sake of this article, was running around clapping his hands while holding his SM58.
At first I tried riding the mute button on his microphone, but I was spending so much time on him I couldn’t mix the rest of the show.
So I reached for the variable high-pass filter knob and ran it up to 300 Hz. It thinned his voice out a bit but I doubt anyone noticed but me.
Combat The Unwanted
High-pass filters are probably one of the most under-utilized features on the console. The most common use has traditionally been to combat unwanted proximity effect, which is the tendency of directional mics to increase their output at low frequencies as the sound source gets closer to the mic.
Cardioid and hypercardioid mics get their directional characteristics from ports in the mic capsule that allow sound to impinge on the rear of the diaphragm as well as the front. The added length of the ports creates a difference in path length between sounds hitting the front of the diaphragm and the rear.
Pressure differences between the front and rear of the diaphragm are what make it move. These different path lengths cause a difference
in pressure because of two factors: phase and amplitude.
The phase component is dominant at higher frequencies. A 20 kHz wave is slightly more than a half-inch long. The path length difference from the front of the diaphragm to the rear is large as a percentage of the wavelength, so almost complete cancellation can occur.
This is one reason why microphone directivity breaks down as frequency decreases, and it is also why the diaphragms of cardioid mics are damped at about 6 dB per octave as the frequency rises. Remember: more pressure difference equals more diaphragm movement.
But the key to proximity effect is the amplitude disparity. The inverse square law tells us that every time we double the distance from the source to the diaphragm, we lose 6 dB. This is very powerful at short distances; for example, the difference between a singer being a quarter-inch from the mic and a half-inch from the mic is 6 dB.
It also means that the difference in path length from the front of the diaphragm to the rear becomes more and more significant as the source gets closer.
Since phase cancellations are a fixed percentage of amplitude at any given frequency the amplitude factor becomes much more dominant at close distances than the phase factor. The phase part of the equation has less and less effect at longer wavelengths while the amplitude part holds true at all frequencies.
Hence proximity effect.
Proximity effect can go as high in frequency as 500 Hz depending on the mic, although 200-300 Hz is more common. The amplitude gain can be as much as 16 dB! This is probably why high-pass filters were put on mics and into consoles in the first place.
But sweepable high-pass filters can also be used to help you clean up your overall mix.
One of the things we learn from audiology is that lower frequency sounds obscure higher frequency sounds, but not the other way around. This is one of the principles that makes sound masking work.
It’s useful in sound masking systems, but in a live performance situation, not so much.
Many live mixers react to this unconsciously when they reach for the house graphic and hack away at 125 Hz and 160 Hz. True, many rooms react poorly in that frequency range, but the room is only one part of the problem.
Let’s think about the physics of low frequency sound waves.
A 100 Hz wavelength is 11.3 feet long (at sea level, at 72 degrees Fahrenheit etc., etc.). This is typically above the crossover point for subwoofers, so it’s probably being reproduced by the main arrays.
In order to provide good directivity at any frequency, the array must be larger than the wavelength. If the array is not larger than the frequency of interest, the sound waves wrap around the array and it behaves as an omnidirectional source.
Even if the line array is fairly long, you only get the directivity benefits in the vertical axis. Chances are, the array is four feet wide (or less), which means that in the horizontal plane, pattern control starts to break down at around 250-300 Hz.
What is in close proximity to the array on the horizontal axis? The stage. And the mics on the stage.
Even if the subs are being run from an aux send on the console (which I highly recommend), there is still energy from the sources being routed to the subs that finds its way back into the stage mics.
Because the same laws of physics hold true for stage sources as for main arrays, the mics are picking up the desired musical content in these frequency ranges – plus the adjacent instruments and floor wedges, plus the room resonances, plus the wraparound from the main system in the longer wavelength frequencies below about 300 Hz.
This is happening even if we don’t consider the artist clapping with a mic in his hands or tapping his foot on the mic stand base. And to compound the problem, the cardioid pattern of the mics breaks down in the lower frequencies as well.
The inverse square law (minus channel compression) is your only friend at this point!
So, what’s a poor sound engineer to do? Directional cardioid subs and cardioid sub arrays can help enormously with the least directional part of the mains, which is often closest to the stage.
We’ve already made gains in cleaning up the stage sound (at least in some cases) with tools like in-ear monitoring and instrument amplifiers located off stage in isolation cabinets.
While these techniques are incrementally helpful, there’s another tool at our disposal: the console channel’s variable high-pass filter.
The earlier we can deal with these issues in the signal chain, the better, which is why high-pass filters are found on many outboard mic preamps as well. If your mics have a shelving filter, try that first. If it doesn’t degrade the instrument sound, leave it switched in.
Next, at soundcheck, start your equalization process for each mic by sweeping the high-pass filter up until you hear it affect the sound. Obviously, there are some inputs that might be left out of this process like the kick drum, bass guitar and a low piano mic. DIs and other direct feeds don’t count because they aren’t picking up ambient sound.
A sharper knee and a steeper slope will allow you to set the filter to a higher frequency without degrading the natural tone of the source up to a point. Too steep of a slope can cause a filter to “ring.” Filters have resonances too.
Then, during the show, solo each mic with headphones that provide good low frequency isolation and response (I like beyerdynamic DT770s), and you may find you can cheat your high-pass filters upin frequency a little higher.
Oh and by the way, have the monitor engineer try this too, only he/she can be quite a bit more aggressive with it. The performers on stage don’t have high-pass filters on their IEMs, and many ear molds don’t do a great job of isolating lower frequencies.
Using this approach should lead to cutting less in the 125-200 Hz range on the system EQ because you are solving the problem at the source.
You’ll also be surprised at the increased clarity in your overall mix. The system will have more headroom as well since the frequency ranges we’re dealing with are real energy hogs.
Remember, garbage in garbage out. Why deal with it in your mix when you can cut it off at the pass?
The high pass, that is.
Bruce Main has been a systems engineer and FOH mixer on and off for more than 30 years. He has also built, owned and operated recording studios and designed and installed sound systems.