Or: Everything You Ever Wanted to Know About Decibels but Were Afraid to Ask
At first glance, the decibel might seem like an oddly fundamental topic for a series called “Beyond The Basics.” But in practice, the average audio engineer’s understanding of decibels and how they work is often shaky at best.
Granted, you don’t need to fully understand dB to make great-sounding records. But it can never hurt to have a deeper understanding than you already do. Sometimes, going over the underlying theory causes new ideas click, or simmer and synthesize with old ones in the backs of our minds.
Making the Enormous Graspable
Plenty of you may easily remember that dB stands for “deci-bel”, and that this is one-tenth of one “bel” – a fairly obscure unit measurement named after Alexander Graham Bell, a man who is equally famous for inventing the first practical telephone, and for his premature male-pattern chin-baldness.
In essence, the decibel offers a manageable logarithmic scale for measuring changes in level or loudness.
For those who don’t remember too much precalculus, a logarithm essentially allows us to compress large numbers, shrinking the scale of digits down so we can talk about the relationships between them in a more intuitive and graspable way.
dB as a Measurement of Power in Watts
When it comes to loudness, the difference in power between the very quietest sound we can hear and the very loudest that we can stand before feeling pain is a ratio of about one-trillion-to-one.
That’s a 1 followed by 12 zeroes.
Things would get a little awkward and illegible if the numbers written next to our faders went from zero to a trillion, not to mention the cost in ink. So in the 20th century, we settled on dB as the standard measure of volume relationships.
When it comes to acoustic or electrical power (which is measured in watts) dB is easy enough to understand: Every increase of 10 dB corresponds to an increase of 10 times the power, which we tend to hear as a doubling of loudness.
Using this scale, if we were to take the lowest level we can hear and call it “0 dB”, that would make a sound one trillion times more powerful (a level known as our “threshold of pain”) register as “120 dB”.
In other words, if we were to take the quietest sound our ear could hear and label that as 0 dB, then a sound of 10 dB would be 10 times more powerful or twice as loud. 20 dB would be 100 times more powerful or 4 times as loud, 30 dB would be 1,000 times more powerful or 8 times as loud, all the way on up to 120 dB, which would have 1,000,000,000,000 times the power of our 0 dB signal
Adding 120 dB = adding 12 zeroes to the power = 12 increases of 10 dB = 12 doublings in perceived volume.
(At this point, it’s a good idea to hammer home the idea that doubling of power is not the same thing as a doubling in volume. A doubling of power would make for an increase of just 3 dB, which we experience as being just a little bit louder.)
This scale is especially handy for dealing with things like amplifiers, which measure their power in watts.
As you can infer from these numbers, a guitar amp that puts out 100 watts is not 90 times louder than a guitar amp that puts out 10 watts. All things being equal, you might expect it to put out just 10 times the power, and just twice the perceived loudness.
The truth is, it’s sometimes easier and more effective to get more clean level out of an amplifier by switching to a more efficient speaker system than by just upping the brute force of wattage.
dB is also a Measurement of Pressure or Voltage
If the lesson could end here, dB might not seem that complicated at all, and we could all go home for juice and cookies:
2 times the power in watts = +3 dB = just-a-little-bit-louder.
10 times the power in watts = +10 dB = twice as loud.
100 times the power in watts = +20 dB = four times as loud.
Ok, got it!
Unfortunately, things aren’t quite that simple. This scale changes somewhat when we look at voltage or pressure instead of power.
And in the studio world, we’re actually more likely to use this slightly different, slightly more complex voltage/pressure dB scale.
When we look at the voltage in an analog circuit or the levels in a DAW, things go a little something like this:
2 times the voltage = +6dB
10 times the voltage = +20 dB
100 times the voltage = +40 dB
So, when we’re looking at voltage:… if 0 dB were 1 volt, then 2 volts would be +6dB, 10 volts would be +20 dB, 100 volts would be +40 dB and 1,000 volts would be +60dB.
So, why the two slightly different scales?
Well, Power Law says that Power = Voltage * Current.
And Ohm’s Law (Current = Voltage / Resistance) says that if we increase the voltage of a signal by a factor of 10, we’ll also increase the current by a factor of 10 as well.
This means that an increase in power will basically be the square of an increase in voltage.
This means that increasing the voltage by a factor of 10 should lead to an increase in power by a factor of 100.
Or to put it another way:
10 times the voltage = +20 dB = 100 times the power
Maybe read that stuff above one more time. It’ll sink in, I promise.
The math above is pretty neat and tidy, and in the real world things are a bit less perfect. But still, this is the basic fundamental view of how these scales interact. A comparison between these two scales can be found above and to the right.
How Loud of a Difference Can You Hear?
Testing suggests that on average, people experience a tenfold increase in power (+10dB) as a doubling of volume, but keep in mind that this is a subjective measure.
I’ve also heard of tests that suggest people sometimes hear a doubling of voltage (+6dB) as a doubling of volume, and others that say it’s +12dB that sounds “twice as loud.”
One fact that’s worth acknowledging is that we are not equally sensitive to changes in volume throughout the range of our hearing. It’s said that on average, people can hear changes of about 1dB, up or down, but that depends a bit on what frequency is being tested, and how loud the sound is.
Our ears and brains are more sensitive in the upper mid-range and less sensitive in the bass and highs – although that does even out a bit as we bring sounds up in level overall.
Under the right conditions, even untrained listeners have been shown to hear changes as small as one-half-of-one dB. Trained listeners have even been known to hear as little as a 0.25 dB change, depending on frequency and overall level.
This is a remarkable sensitivity, for sure. But before we slap our own backs for our tremendous ability to hear, let’s take into account a few of our limitations.
Your Hearing is Not Flat
Using electronic devices, we can measure changes in pressure far smaller than those we can hear. And unlike a microphone, which may respond fairly evenly to level changes at different frequencies, our ears are biased toward the midrange – especially when we listen at low levels.
For a low 100 Hz soundwave to sound as loud as a 0dB 1kHz wave, we might need to put about 40 dB of extra energy into it.
But if we were to try to get that same 100 Hz soundwave to seem as loud as a 1kHz tone at 100dB, we might only need to put a handful of extra dB behind the tone.
The takeaway from this is that if you monitor too loud, your mixes can end up sounding a bit weak in the low bass and high, shimmery treble when you play it back at a more normal volume.
dB: A Rolling Log
It’s also important to remember that in and of itself, dB has no value of its own. It’s merely a handy way to measure relationships between two levels. For dB to mean anything at all, we need to assign some concrete value to “0 dB.”
When we deal with SPL, the custom is to make 0dB the quietest sound the average human can hear – an exceedingly infinitesimal pressure of about 20 micropascal.
In digital circuits, we label the loudest sound possible before clipping as 0dBfs (for 0dB “full-scale”) and mark everything else as a negative ratio of that maximum level.
In the analog domain, the custom is to label the ideal operating level as 0dB. Depending on how the meters and circuit are designed this could be a level of 1 volt (dBV) or .775 volts (dBu) or even the power-based standard of 1 milliwatt (dBm).
There has to be some correlation between digital and analog, and often enough, 0dB on an analog VU meter will show up as somewhere between -22dBfs and -16dBfs on a DAW meter depending on calibration, with -18dBfs = 0dB on an analog VU becoming increasingly popular.
There’s another practical takeaway here as well: For the best possible gain staging, it’s often a good idea to record at a level that only reaches only about 2/3 to 3/4 the way up on your digital meters. It’s in this general area that your analog front end is often operating at its best.
Of course, if you like a little more grit, you can always push the input stage of your analog gear harder. But to keep levels well balanced between tracks, and to keep from overloading your master bus or plugins and outboard effects, a little bit of conservative meter-watching at the A/D stage can be just the thing. Especially in 24 bit systems, there’s no reason not to allow yourself a little headroom. Your gear was designed for it.
As always, rules are made to be broken — It just helps when you know what they are first. The rules we’ll always have the greatest difficulty overcoming of course, are the laws of physics. And that is the domain of the dB.
As we come up on the 2012 AES Convention - October 27-29 at San Francisco’s Moscone Center – we’d like to revisit what some of last year’s AES attendees told us was their favorite panel discussion. We happened to produce it, and now have it to share as video.
“AES Platinum Engineers: The Studio As An Instrument” brought an incredible panel of studio gurus into a discussion about creative recording techniques and fascinating collaborations. Those panelists included some of our favorite producers and engineers: Dave Fridmann (The Flaming Lips, MGMT), Peter Katis (The National, Jonsi), Damien Taylor (Bjork, The Prodigy) and Chris Shaw (Weezer, Bob Dylan). SonicScoop contributor, Justin Colletti, led the discussion.
The full version of the panel has been edited down slightly to just over 90 minutes in order to streamline the flow of the conversation without losing any of the insights or personality our panelists brought to bear. Check it out below – watch, listen, learn.
Last month, we brought you a roundup of some of the best extreme and unusual compressor plugins on the market. In that feature, we looked at the kinds of dramatic sound-shaping dynamic tools that are worth owning, but that you might not use on every mix.
Today, we present the inverse: A few of our favorite everyday workhorse compressors that are available for your DAW. Rather than sum up the biggest names in the field, we decided to take a closer look at a handful of the best cross-platform offerings from smaller brands. It begins with a two-man shop.
After a nearly year-long wait, Massey has finally released the much-anticipated update to their CT4 compressor. The original version inspired a kind of fevered loyalty that’s uncommon for compressor plugins that are designed to sound as transparent and unobtrusive as the CT4.
Part of that can probably be attributed to this small company’s reputation for affordable pricing and a customer-centered approach. But a large portion of the CT4′s success lies in the fact that users found it very hard to make this plugin sound bad.
It may not always add much mojo to the source, but as far as effective, pleasant-sounding dynamic control goes, the CT4 offered a great alternative to the clean compressor plugins designed by some of the longtime leaders like Sonnox and Waves. And, unlike the Oxford and Renaissance Compressors made by those companies, the CT4 offered a simple, intuitive analog-style GUI.
Now with the new CT5, Steven Massey has tweaked the plugin and added new controls, taking his compressor from useful to indispensable in the process. Although I enjoyed the original CT4, I’ll be the first to admit that it had its limiting factors. Thankfully, the most substantial of those have been largely corrected with latest update.
The CT5 adds a “medium” attack setting that allows this plugin to complement a far wider variety of sources. With the CT4, there were often times when the attack envelope I desired was just out of reach. Also new to the plugin is a limiter function, and a blend knob – a feature which has become nearly ubiquitous on new software compressor designs.
The official word from Massey is that the underlying algorithm remains largely unchanged. However, I believe I do hear a slight change in the sound. Although the design is still certainly a transparent one, with CT5, I hear the subtle – and admirable – tone of the plugin as soon as it’s instantiated on the track, before I even move a single knob. The VU meter seems to reflect this as well. Take the plugin out of bypass, and even with the controls set to minimum, the GR needle dances ever-so-gently to the beat.
Softube Tube Tech CL1B and Summit TLA 100A
Softube’s emulation of the Tube Tech CL1B is an everyday workhorse with heaps of character. Like all of Softube’s designs, even the GUI is rich, luscious and fun to look at. But far more than looks, it’s the sound of this virtual unit that’s unmistakably analog.
Even the gain control on the Softube CL1B sounds pretty good. A simple boost here seems to add tone as much as it adds volume. And ultimately, this plugin’s real strength lies in its ability to impart subtle body and gloss when compressing by just a few dB at a time.
When the CL1B digs in deeper than that, the sound of the plugin can become a bit grainy and saturated – but in a way that sounds authentic to me rather than inadvertent. It has a sound – much like the hardware it’s modeled after – that’s a useful and colorful alternative to vintage-minded workhorses like the LA-2A and its many copies.
But the CL1B isn not Softube’s only worthwhile venture into opto-style compressor emulations. Their rendition of the Summit TLA-100A fills in wherever the CL1B misses.
While it may be significantly less colorful than the Tube Tech CL1B, Softube’s Summit TLA-100A can deliver extreme levels of compression while staying largely transparent – just like the original hardware. It’s a natural everyday compressor for vocals, bass and acoustic guitars in particular, and can deliver heaps of dynamic control without significantly reshaping the tone of the source.
PSP oldTimer and BussPressor
While the Softube plugins that made this list model specific hardware units, PSP were content to imagine their own when they set out to create the affordable and popular oldTimer plugin.
What PSP lacks in GUI design here, they make up for in sound. The oldTimer has a bit of a Fairchild character – but not quite; a bit of an LA2A tone – but not exactly. At high gain-reduction levels it has just a touch of analog graininess, but stays smoother and more even, perhaps, than the CL1B.
This is not a glossy-sounding plugin by any means, but it rarely offends unless you intend it to. The oldTimer manages to reach a level of balance between grit and unobtrusiveness that’s difficult to achieve. It has personality, but rarely overwhelms. It sounds good on all manner of acoustic instruments, voice and clean electric guitars. And as a tool, it’s about as useful as a Phillips head screwdriver.
PSP’s BussPressor, which we reviewed in July, is a snappy, sinewy and flexible compressor inspired by the sound of SSL consoles. I’ve had great luck with it on drums, mix buss and some acoustic instruments. Once again, it’s a plugin that seems to find it’s way onto nearly every ITB mix.
Although I prefer the targeted approach of these two units, PSP also offers a plugin called the MixPressor, that’s designed for flexibility and seems to have a loyal following of its own.
McDSP Compressor Bank, Channel G and 6030 Ultimate Compressor
When McDSP emerged in 1998, the field of plugin developers was much smaller than it is now.
Waves and Sony were around, making a handful of forward-looking designs, and Dave Amel’s Bomb Factory had just gotten its start modeling vintage processors for an increasingly digital world.
Colin McDowell, meanwhile, decided that McDSP would do both of those things.
His now-classic Compressor Bank offered a multi-functional compressor similar to those being sold by Waves and Sonnox early on. The CB1, CB2 and especially, the CB3, were the kinds of tools that would only be truly feasible in the software world.
The CB4, however, was an earnest effort to model the specific vintage squeeze-boxes that users lusted after, limitations and all. Those models sound great even now.
Eventually, McDSP took the research that went into these algorithms and put them together with their flagship Filter Bank EQ and the flexible console and tape machine emulations from their Analog Channel, in order to create Channel G – a one stop shop for organic-sounding signal processing.
Unlike the other entries on this list, what makes McDSP fit here is not that they offer a single sound that works in so many places. Instead, each of their dynamics plugins is really a whole suite of tools – an entire palette of colors ready to be explored. While this is a key selling point for Channel G, Analog Channel and Compressor Bank, there is no place where this approach is more clear – or more simply executed – than in the 6030 Ultimate Compressor.
The 6030 is like a box full of crayons. It offers 10 different colors, each with a distinct personality, and encourages visceral, gutsy decision making. The layout is like a virtual 500-series lunchbox stuffed to the brim with boutique compressors that would easily run several hundred dollars a piece if sold as discrete hardware. But instead, this single plugin with its minimal, intuitive interface and cornucopia of distinctive models costs just $250 in its native version.
The models inside include compressors inspired by a Fairchild, LA2A, 1176, Vari-Mu and dbx, plus a few wholly original designs. Like classic hardware, each module has a minimalist layout. Some have just one or two knobs, and the busiest of the bunch have four.
As for the sound, each one is distinctive, although it’s apparent that they are “inspired by” more than they are faithful re-creations of classics. By and large, the individual compressors in the 6030 sound just a touch bigger, more modern, and more “hi-fi” than both the CB4 models, and the original hardware on which they’re based. Depending on your tastes (and your mix) that could be desirable.
Like any tool, this one isn’t perfect, or else, why would anyone own more than more? Most notably, the output section of the 6030 could use a little refinement to make level-matching a little quicker for AB listening. But plop this plugin on an insert, and it would be difficult to imagine an engineer who would be unable to find a sound in the right ballpark for the track.
Although almost any one of the modules in the Ultimate Compressor would be worth considering for this list on its own, when you put them all together, the 6030 transcends “workhorse.” It is a true swiss-army knife.
There’s no substitute for mixing with your ears. Too much visual stimulation can be a hindrance where audio craftsmanship is concerned, and because of this, one of my favorite plugins to date is a promotional freebie from Massey, called “Listen.”
When engaged, the Listen plugin will hide your entire workspace, making your screen glow a gentle solid blue, with one directive written out for you to consider: “Listen.” It’s great advice, and in general, it leads to better mixes.
Compared to other most mammals, our hearing is weak and even at our best, the most ‘golden-eared’ listeners have minds that are far more powerful than their ears. This is why we can turn an EQ knob and think we hear it working, before we realize that it’s not engaged, or that we’ve grabbed the wrong channel. And this is also why the power of the myths and stories that surround an artist often grab new listeners long before their music does.
The Value of Measurement
No good sound mix will ever be completed by relying on ones’ eyes alone. But because of the relative weakness of our ears when compared to the power of our minds, audio measurement tools will always be useful at times.
At its best, good measurement can help us come to a greater understanding of reality. It can keep us from engaging senseless, ego-based screeds over matters of taste and assumption. It allows us to set meaningless arguments of minutia to rest, and instead focus on the things that really matter. Needless to say, those are rarely found within any audio circuit.
I had a friendly debate with a colleague just the other day about whether EQs tend to have a more dramatic effect when they’re placed before or after a compressor. Although we disagreed, frequency analyzers make resolving questions like these easy, and they help keep debates civil and grounded in truth over rhetoric. We can quickly try things for ourselves, make repeatable measurements and share our results.
There are practical applications within mixes as well. I rarely pull out a frequency analyzer on more than one out of every 10 or 15 songs, but when I do, they can be enlightening.
A good analyzer is a little bit like a calculator. In the interest of exercise, I’d rather not rely on one too much, but they do help make awkward problems easy, and in extreme cases, make the impossible graspable.
If a mix is sounding a little boxy in the midrange and I’m having trouble deciding exactly where, an analyzer can point me in the right direction. If I’m having trouble getting great results out of a deesser on a particularly tricky vocal passage, I can figure out what exact areas are causing me grief. If I’m trying to match a punched-in tone and have only managed to get 95% of the way there with my naked ears, a frequency analyzer can help save time. And perhaps where they can be of the most help is in sorting out the low-end.
In our recent interview with producer Bob Power for the Input\Output Podcast, Power reiterated the importance of carving out different homes in the low end for competing bass instruments. A bass guitar might occupy the deep registers, while the bulk of a kick drum lives in the upper bass range – or vice-versa.
At Bob Power’s stage in the game, this may be done by ear more often than not. But for beginners, or those who struggle with this balance, analyzers can be a godsend. They can also be fantastic teaching tool – in any register.
Nothing is a substitute for hands on practice and deep critical listening. But sometimes, the right tools can help us speed up our understanding, crystallize new concepts, or overcome stumbling blocks and plateaus.
So without further ado, here are some tools worth exploring.
Blue Cat FreqAnalyst
It’s available for both Mac and PC in a complete range of formats, including AAX, RTAS, VST and AU. You can even customize the skins and the display options. And did we mention it’s free?
If you’re just getting started with analyzers, our list could pretty much end here. This plugin does most of what you’ll ever need a frequency analyzer to do.
But if you’re interested in something more, Blue Cat also offers a whole “Analysis Pack” that includes a “Pro” version of this tool along with a stereoscope, oscilloscope, and precision peak meter.
Most powerful of all, perhaps, is the included FreqAnalystMulti, which is a multi-track version of the FreqAnalystPro. This is a tool that will allow you to put one instance on a kick drum, one instance on a bass, and see where their frequency spectrums conflict in the same readout.
This kind of functionality is a real leap forward when it comes to practical in-the-mix analysis. Use it on three competing guitar tracks and see their relationships together all on the same screen. It doesn’t beat using your ears, but it can be a fun supplement, or a unique learning tool.
These advanced tools range from $50 to $100 and can be purchased together in a bundle for $189.
Vintage-Style LED Analyzers
The one issue that comes to mind with Blue Cat’s tools is that they can be too advanced. For those who prefer a more minimalist readout, there are other options.
ReFuse Software makes a 30-band Real Time Analyzer inspired by the early LED analyzers that divided the frequency spectrum into 1/3 octave bands. The layout is minimal, easy to read, and packs a wallop of 80s nostalgia. It’s also free for users of the VST Pluggo architecture.
There are a couple of similar options for those who don’t use Pluggo:
Vertex DSP makes a free VST and AU tool called the Multi-Inspector which has a 31-band LED-style readout and multi-track functionality.
Meanwhile, Seven Phases makes a VST-only Spectrum Analyzer that is a lot like reFuse’s, only far more customizable, and RML Labs makes an add on Frequency Analyzer for their proprietary SAW Studio program that divides the spectrum into 50 bands.
So far I have not seen a similar tool compatible with Pro Tools. If you come across one, please let us know.
Sophisticated VST and AU only Analyzers
Other developers offer plugins that are on-par with Blue Cat’s once you get outside of Pro Tools’ RTAS and AAX environments and into the wide open waters of VST and AU.
Voxengo’s SPAN is free, and easily one of the most popular, but it’s not the only game in town.
schwa schOPE, by Stillwell Audio, costs $25 and does a lot. According to their website, schwa schOPE is a “multitool, multiview, multichannel waveform, frequency, and stereo phase analyzer that lets you superimpose and view multiple audio signals from every dimension.”
They recommend that you “scope pre- and post-compression signals together to see if you’re flattening peaks. Scope kick drum and bass together for EQ scooping. Scope DI and amped guitar signals together to fix phase problems.”
Nugen Audio’s $89 Visualizer is extremely sophisticated and worth a look as well. It includes precision level meters, spectrum analyzer, a vectorscope, phase correlation meter, stereoscope and more.
Melda Productions’ VST-only Manalyzer offers pre-filtering modes and magnitude normalization for easier screen-reading, along with a useful “Deharmonization” function which “reduces harmonics leaving only fundamentals, which is great for identifying relevant frequencies.” It’s also one of nearly two dozen free VST plugins that the company offers.
Speaking of free, Signal Analyzer by Robin Schmidt is another small one, as is Freakoscope by Mdsp @ Smartelectronix. Both are VST-only, and although the latter offers a neat note-readout feature, and both Fourier and LED-style display options, it is sadly, Windows-only.
You can also examine frequency and a host of other parameters across several stereo audio tracks at once without ever even firing up a DAW. Sonic Visualiser is 100% free software available under the GNU General Public License.
For those learning about the science of sound, or for long-time engineers looking to increase the depth of their familiarity with the physics that underlies their craft, this, or any of the tools on this list, can be invaluable in moderation.
In this episode of Input\Output, Geoff and Eli interview legendary “Soul Music Producer” Bob Power. In the 80s, 90s, and 00s, Power worked with some of the most influential artists of the Hip Hop and R&B world, including A Tribe Called Quest, De La Soul, Erykah Badu, D’Angelo, India Arie and Meshell Ndegeocello.
The hosts ask Power about his journey from session player to producer, rummage through his brain for tips and techniques and grill him about some of his favorite gear. They also take a detour to discuss the business of music, and how ask Power how he’s continued to adapt to a changing landscape.
Listen to the episode below, or right click here to download.
Input\Output is produced by Justin Colletti for SonicScoop.com. This episode was recorded at Stratosphere Sound, in Chelsea.
Click the link or player below to hear the episode!
Input\Output: The Bob Power Interview
I had a disconcerting moment earlier this week. While out for a mid-afternoon stroll, I rounded a corner and suddenly felt an unusual pressure in my right ear. In just a few footsteps, my hearing grew dull and I experienced the strange and unwelcome sensation that someone or something had plugged my ear canal full of miniature marshmallows.
Suddenly going deaf in one of your ears is enough to send any musician or engineer into a brief existential crisis. You can console yourself for a moment, remembering that hey, this kind of thing didn’t stop Brian Wilson or Beethoven. That is, until the far more realistic thought sets in: You’re not Beethoven.
The Leading Cause of Temporary Hearing Loss
Like many musicians, engineers and other freelancers, I’m one of the 60 million Americans who lack access to affordable health insurance. This means that when I’m stricken with some kind of unknown ailment, I tend to run to Google instead of my doctor, since I do not have a doctor.
While I’ve been able to heal myself of two slipped spinal discs and an abdominal hernia using corrective exercises and remedies found online, I still don’t recommend this method to anyone, including you. It’s likely that you’re out of your element when it comes to medicine, just like I am, and trust me, I know firsthand that self-diagnosis is no replacement for a physician’s care.
With that said, we can talk about something that’s arguably even more important than cure: prevention.
As I continued my walk, I tugged downwards on my earlobe and was thankful to find that my hearing returned for a moment. As I released my ear, the canal closed up again and the hearing in my right ear was again reduced to a dull, pillowy murmur.
Because I’m a giant idiot, I decided to try something that each one of my sources would eventually confirm as being a very dumb thing to do.
Like so many similarly soft-headed morons around the country, I assumed the obstruction was due “too much wax” and grabbed the first thing I could think of to help me remove it. Even the Q-Tip package could have told me what I was about to do was wrong: “Not for use inside the ears.”
This is because what I experienced, along with more than 2.1 million other Americans each year, was not a foreign obstruction or what’s commonly mistaken for an undue buildup of ear wax. Rather, what took out my right ear for a good 24 hours was a condition that’s among the leading causes for temporary hearing loss in the U.S., especially in the summer months: “Swimmer’s Ear.”
This is a condition that’s so common, but so surprising to many of us, that it accounts for up to 40% of all the emergency room visits in southern states, where it’s most prevalent.
Here in the Northeast, where the condition is less common, Swimmer’s Ear still accounts for nearly 20% of all of our total emergency room visits. This is a staggering figure that is estimated to cost our healthcare system nearly $500 million each year.
Swimmer’s Ear is caused when bacteria or fungi – who think that water is the bees knees, and that your ear is a neat place to live – find their way in there and start getting fruitful and multiplying.
The most common causes of Swimmer’s Ear are freshwater swimming, saltwater swimming, pool swimming and tap water from the bath, in that order. (I got mine while whitewater rafting in the Delaware River.)
What To Do About It
Like so many millions of Americans, when I felt as if something was blocking my ear canal, what I was actually experiencing was internal swelling in response to an infection.
The number one thing NOT to do if you ever feel like there’s an obstruction in your ear is to use any kind of object to clear out the pathway. Counter-intuitively, trying to clear the obstruction or remove wax from your ear with a Q-Tip only makes matters worse.
This strategy actually increases irritation and swelling and tends to lead people into freaking out and driving to the emergency room. (Thankfully, I didn’t get that far.) And, if you are successful in removing a heap of earwax from your ear (rather than just impacting it against your eardrum, which doctors say is far more likely): then congratulations. You’ve just effectively removed your ear’s built-in cleaning fluid, and your body’s one natural defense mechanism against further infection.
However, if you’re lucky enough to catch your case of Swimmer’s Ear while it’s just beginning you may be able to effectively disinfect the area before it’s too late.
Instead of poking around in your ear with a stick like Cleetus the Slack-Jawed Yokel, you can make yourself a solution of 50% rubbing alcohol and 50% white vinegar.
If you’ve never had ear surgery and you’re certain that you don’t have a ruptured ear drum, you should be able to safely drip this solution into your ear using an eye-dropper, an empty contact lens bottle, or a sterile syringe (minus the needle, of course.)
If you’re as lucky as I was, this solution will kill the bacteria and fungus before they get a foothold and allow you to avoid a full-blown case of Swimmer’s Ear.
If you’re unable to catch it in time, a doctor will need to prescribe you up to 2 weeks worth of prescription antibiotic drops to cure your ear. If left untreated, the infection and swelling can reach a point where you wind up with a ruptured eardrum – And hearing that will never be quite the same again.
The best way to beat Swimmer’s Ear is by not getting it in the first place.
If you’re prone to these kinds of infections or if you plan to work with sound immediately after swimming, waterproof earplugs or a swimming cap can stop water from getting into your ears.
Some of the same folk remedies that are used to good effect in the early stages of Swimmer’s Ear can also be effective for prevention if your ears become filled with suspect water.
Rubbing alcohol kills germs and helps with the evaporation of trapped water. Both apple cider vinegar and white vinegar have anti-fungal properties that make them popular preventative remedies. And don’t underestimate the power of a good hairdryer in helping to gently dry out your ears before an infection can take hold.
If you’re like me and the countless thousands of Americans who have created an impenetrable wall of impacted earwax by jabbing a Q-Tip in there, please quit while you’re ahead. If you need to get past this obstruction in order to disinfect your ear, you can soften the earwax with Carbamide Peroxide, which is sold over-the-counter as generic “earwax removal aid” or under brand names such as Debrox.
But please, don’t go crazy with it. Remember that wax was put in your ears by nature to clean and protect your hear-holes. If you ever suffer from decreased hearing sensitivity or stuffy ears, the underlying problem is likely to be swelling due to infection or irritation. If you do start to produce more earwax than usual, odds are that it’s there to help fix that problem, and trying too hard to remove it is likely to make matters worse.
The Leading Cause of Permanent Hearing Loss
Although an ear infection can lead to deafness if left untreated, it’s relatively rare for Swimmer’s Ear to cause more than a temporary hearing loss in industrialized nations.
In countries like ours, the number one cause of permanent hearing loss is prolonged exposure to loud sounds – and you might be surprised by just how low our thresholds are.
Both The House Research Institute and The Center for Hearing and Communication estimate that over 37 million Americans, or roughly 12% of the population, suffer from significant hearing loss. Look at adults over age 65 and that percentage skyrockets to near 33%.
Ironically and sadly, that figure is likely even higher among musicians, audio professionals and music fans – Precisely the people who value their hearing the most.
To help us retain our hearing Marilee Potthoff, Director of Outreach and Education at The House Research Institute, says that her organization recommends we significantly limit our exposure to sounds exceeding 85 dB in level.
At 88dB, guidelines from House Research and The Center For Disease Control recommend no more than 4 hours of exposure. At 91dB, they suggest a 2 hour maximum. At volumes approaching 100 dB, they recommend just 25 minutes per day to avoid long-term hearing loss, and at 105 dB, no more than a few minutes is advised. Most troubling of all, at 115 dB – the volume level of a typical rock concert – Potthoff says there is no length of exposure that can be deemed safe.
If you’re one of the 50 million Americans who have experienced Tinnitus, or “ringing in the ears” after a loud concert, this means that you’ve suffered a loss in hearing that will never come back. Hearing loss from high volume music is cumulative, which means that repeat exposure only makes matters worse. But it also means that it’s never too late to stop more damage from occurring in the future.
What To Do About It
Unfortunately, once you’ve suffered long-term hearing damage from loud SPLs, there’s no way to get it back.
Unless you’re a fish, who are among the few creatures that can replenish their hearing capacity (we imagine they’re also immune to Swimmer’s Ear), your ears are not going to fix themselves.
We audio-loving air-breathers have to get smart now, or suffer the consequences long into our ever-expanding working years.
Nothing makes teenagers turn a deaf ear like the word “prevention.” No wonder that so many of them are also going deaf in a literal sense. Today, 15% of adolescents between the ages of 6-19 have measurable hearing loss in at least one ear.
Some of this can be blamed on earbuds that are regularly blasted at high volumes. But this isn’t just a matter of kids being irresponsible with their ears. If you ride the subway, chances are that you jack up your listening device loud enough to hear over the oppressive rattle of the train. The sound level of that train may approach 100dB, which is safe for only about 15 minutes a day – So if your iPod is louder than that, you do the math.
This means that one of the best ways to keep your headphones at a reasonable level without compromise is to switch from conventional headphones or earbuds to something that blocks out more exterior noise.
The key here is acoustic isolation, so active headphones that use fancy phase-based noise-canceling technologies are of no help unless they also block out significant amounts of exterior noise through good old-fashioned padding and mass.
The same principle applies for uber-loud full-band tracking sessions. If you have solid headphones that provide ample acoustic isolation, such as the Sennheiser HD-280 or the Direct Sound EX series, your players won’t have to crank them quite so much to hear themselves clearly.
Alternately, you could become an advocate of tracking at reasonable volumes, perhaps even without headphones like on so many of your favorite-sounding classic records – But that’s fodder for a whole other article.
It used to be that the only earplugs you could easily find were the big clumsy foam ones that block out tremendous amounts of sound with little elegance. Although earplugs like these can reduce by as much as 40dB near the center of our hearing range, they do so at the cost of nuance and musical sensitivity.
Thankfully, much better models are available today at a reasonable cost, and today there’s no excuse not to have a few pairs of high-quality reusable earplugs like High-Fidelity Hearos or the E-A-R UltraFits at hand.
For those who want to rock out in style, any good audiologist can fit you for custom-made, reusable, and practically invisible earplugs like those made by Sensaphonics or Etymotic.
Some of the most sensitive and transparent earplugs on the market reduce SPL by as little as 10 – 20 dB. While this is little enough to keep from degrading the sound of the music you hear and perform, it also goes a long way to bringing an ear-splitting 105 – 115 dB concert down to a manageable 85 – 95 dB. And at volumes like that, you can still feel that satisfying thump in your chest without going deaf for it.
Kevin Killen is a Grammy-winning engineer and producer best-known for his work on classic albums by U2, Peter Gabriel, Elvis Costello, Kate Bush, and Tori Amos – not to mention a long list of well-known contemporary singers, songwriters and bands.
On this episode of Input\Output, hosts Geoff Sanoff and Eli Janney sit down with Kevin to talk about the two albums he worked on that most influenced them as engineers: U2’s Unforgettable Fire and Peter Gabriel’s So.
The stories behind both of these records are fascinating. For Unforgettable Fire (1984), Killen worked in an 18th-century castle his native Ireland with producers Daniel Lanois and Brian Eno. On So (1985), Killen unexpectedly found himself in the middle of a ten-month “mixing” session in a converted farmhouse on the English countryside.
Since relocating to the United States in the late 80s, Killen has been living and, largely, working in NYC. A longtime proponent of in-the-box mixing, he’s been outspoken about his all-digital workflow – a topic which Geoff and Eli dig into here as well.
Listen to individual segments below, or (better yet) download this episode in its entirety to listen wherever you go!
When we last checked in with producer/engineer Ben Rice, he was working out of a “glorified home studio” called Newkirk Recording, where he tracked indie rock bands like The Mooney Suzuki, as well as an assortment of aspiring singer-songwriters and small jazz groups.
Since then, business has been brisk enough that Rice has decided to take the plunge and build out a full-fledged commercial recording studio. His new space, Degraw Sound, occupies a modest 1,100 square feet on the Gowanus/Park Slope border, but feels larger thanks to a crafty design dreamed up with help from David Ellis – the man who helped create The Lodge Mastering, Engine Room Audio, LoHo Studios and Saltlands.
In addition to a control room large enough to house Stratosphere Sound’s old Trident 24 mixing console, the new studio layout includes a spacious 450-square-foot live room, dedicated isolation booth, lounge, and even a small B-room that is operated by longtime friend and former client Gian Stone.
“My overhead has gone up maybe fifteen-fold over the old space, but I think it’s still competitive,” Rice says about his new studio’s rate. It falls firmly in the affordable “mid-priced” category.
Aside from the new console, compressors and Pro Tools HD rig, much of the outboard gear and microphone collection has been transplanted from Rice’s old room. A small wall of amplifiers and guitars, an old Wurlitzer and a vintage Ludwig drum kit round out the equipment list, and nothing feels lacking.
What’s most strikingly new is the sheer space as well as the upgrade in amenities and décor. Rice’s old control room at Newkirk would barely fit into Degraw Sound’s studio B, much less house a full 24 channel desk. The feel of the new studio is professional-but-casual, in much the same way that a classy Park Slope barroom is. Vintage-style Edison bulbs hang from the live room’s ceiling, glowing like so many vacuum tubes, and richly stained barn doors hide all the cases and cables from view.
“People have just been bugging out about the new space,’ Rice tells me. He says that old clients like indie rockers The Reckless Sons have already returned to work with him at the new studio during the two months before his grand opening party late last week.
There have been new clients too, from across genres. In between tracking sessions with indie bands like The VeeVees and The Courtesy Tier, composer and historian Allen Lowe recently brought in an “epic cast of musicians” including pianist Matthew Shipp and free jazz saxophonist Maurice McIntyre.
“He actually heard about me from the article you wrote, I think. So yeah – SonicScoop works,” Rice laughs. “It definitely gets the word out.”
By now – if you’re a Pro Tools user – you’ve at least considered upgrading to Pro Tools 10 if you haven’t taken the plunge already. On this episode of Input/Output, our hosts Geoff Sanoff and Eli Janney offer a comprehensive review of this *latest release* in use, outlining the pros, the cons, and the most useful, most worth-the-investment new features and functionality.
Specifically, Geoff and Eli talk about the benefits of clip gain, real-time fades, longer delay compensation, 32-bit floating point processing, OMF/AAF/MXF file interchange, and disk cache – the full power of which is only accessible to Pro Tools HD10 and Complete Production Toolkit users (PT10 native users get 1GB of disk cache) – among other new features.
Then, tune in as Geoff and Eli interview Avid’s Tony Cariddi on some of Pro Tools 10’s more elusive features, advanced functionality, the new AAX plug-in format, and the future of the platform.
Should you upgrade to Pro Tools 10? Now? Later? Tune in and find out.
As far as polarizing figures go, Ethan Winer is an unlikely candidate. Now 63 years old, Winer is a former audio engineer and computer programmer who plays the cello, owns a successful acoustics business, and frequents online messageboards in his spare time.
In the most pervasive photo of him on the web, Winer is pictured wearing oversized spectacles and an oversized sweatshirt, as he holds his equally-oversized pet cat.
In person and online, he is generally pleasant and mild-mannered, with a ho-hum attitude and a genial, nerdy kind of charm. Based on the innumerable essays and videos he’s produced over the years, Winer doesn’t seem to take himself especially seriously, and tends to be exceptionally reasonable – almost to a fault.
But somehow, Ethan Winer did become a polarizing figure in certain pockets of the audio community – mostly for his unflagging persistence and steadfast devotion to no-nonsense empiricism. One niche audio forum has even barred him from posting, and keeps “stickies” (permanent posts glued to the top of every page) which aggressively mock him in the kind of way that would be easy grounds for a libel lawsuit (if Ethan Winer was the kind of guy who would think to sue someone for libel.)
To these hardcore “subjectivists,” Ethan Winer is like a pedantic Bond villain (which of course, would explain the cat.) But to mainstream audio scientists, Winer is just a smart, quirky man with a dry sense of humor, a tireless typing hand, and some very sensible ideas about sound.
This Spring, Focal Press published his all-new book The Audio Expert. In a sentence: It’s a painstakingly well-researched, 650-page reference guide that seeks to fill in the gaps of knowledge so prevalent in the audio community today.
Background and Credentials
Ethan Winer has racked up hundreds of thousands of posts on online audio forums. Usually, this is the kind of feat that suggests a lack of any real credentials and propensity to rarely do anything useful for society. But in many ways, Winer is an exception.
He’s written articles for Tape Op, Sound on Sound, EQ, Mix, Electronic Musician, Keyboard, Recording, and Trust Me, I’m a Scientist, among others. He’s been an audio engineer, a session musician, a computer programmer, a circuit designer, a studio owner and an acoustics consultant at RealTraps.
Perhaps most impressively, Winer sold his software business at the age of 43 to take up the cello. He went on to release a video called “Cello Rondo” that features an original piece comprised of no less than 37 cello parts, which he played entirely himself. In the video, Winer is dry, goofy and unassuming as ever. So far, it has received than 1 million hits on YouTube alone.
Winer’s written opus, The Audio Expert, was inspired by his memorable Audio Myths panel at the 2009 AES convention. The book is the work of a clear-thinking and disciplined empiricist, and in it, Winer aims to present facts and casts light on poorly understood concepts. The Audio Expert is not intended for beginners, and even its first chapter, “Audio Basics”, is anything but.
As he notes in the introduction, Winer assumes some level of audio knowledge as he begins, and readers without any college-level background or equivalent work experience could get lost quickly. As Winer begins, he assumes you have some basic familiarity with the harmonic series, decibels, impedance, routing and the Nyquist Theorem, and he assumes you’ve heard of summing and dither and jitter and THD. But he also assumes, rightly, that to most audio engineers (even working professionals) the details behind these concepts are often half-understood or misremembered.
One of the great strengths of Winer’s writing stems from the fact that he’s spent so much time reading and answering frequently-asked questions on the web. Because of this, he’s able to immediately zero in on some of the most commonly confused concepts in audio with laser-like precision.
Sure, you may already understand that phase-shift is a necessary aspect of all traditional equalizers, but do you understand how it functions in the circuit, and what the side-effects are, if any? Winer has found this to be largely misunderstood even among active engineers, and he’s not wrong. He also provides tests that reveal whether phase-linear EQs really perform better, and the results may be surprising to some.
Likewise, if you’re an average working engineer, chances are that you have some knowledge of how dB works, and you may even understand that it’s a logarithmic way to describe voltage in a circuit or sound pressure level in the air. But do you really understand what that means? Do you carry around in your mind the fact that if you lower a signal by 80 dB, then you haven’t decreased its voltage or SPL by 80 times, but rather by a factor of 10,000? And are you aware of whether or not noise at this level is audible to you?
Similarly, Ethan assumes you may understand the basics of THD and IMD and have some sense of how to read a spec sheet, but do you understand how these measurements are made, what they show, and what they can hide?
It’s advanced concepts that Winer illuminates simply and cleanly in the opening chapters of his book. Even if you’re only able to get through the first 100 pages of it, chances are you’ll come out knowing more about the most misunderstood subjects in audio than many of your peers do now.
But be warned: Although reading The Audio Expert is likely to be enlightening – even if you’re an audio expert yourself – there are times when it may make you squirm.
In the tightly-packed opening chapters of his book, Winer pursues myth-busting in earnest, and rest assured, even the smartest and most accomplished engineers among us sometimes get things wrong when it comes to the underlying science of audio. With detailed and largely airtight refutations, Winer soundly busts common myths such as these:
–“Analog has higher fidelity than digital.”
It doesn’t, and if you enjoy analog recording (as I do) this is not the reason why.
–“Digital can’t sum properly.”
It can, and if you enjoy analog summing, chances are that the “summing” part of it is not the primary reason.
–“The ‘stacking’ effect can cause exaggerated buildups at particular frequencies.”
Winer demonstrates that if a preamp or microphone is perceived to sound okay on one signal, but not so great when used on several sounds in a mix, this is not because its effect on frequency and noise is additive. As a measurable fact, that would only be the case if the device were used in series – not in parallel, as is normally the case with preamps and microphones.
(Unfortunately, on this point, Winer over-corrects slightly and writes off the term “stacking” completely. Instead, he might have conceded that a piece of gear could be thought to “stack well” for other reasons – namely that its EQ curve might be flattering on a wide variety of sources, regardless of whether or not that curve is additive in normal use.)
By and large, Winer’s evidence is well-presented, well-supported and hard to argue against. This is what makes it so enraging to many hardcore subjectivists who maintain that “science” is unable to measure what the ear can hear.
While Winer can be flexible on isolated points, when it comes to arguments that question the basic validity of measurement and scientific testing, Winer would reply that, in reality we can measure more than the human ear can hear. The truth, he says, is that test equipment can pick up extremely low levels of noise, distortion and frequency incongruity that cannot be heard by the human ear – Not the other way around.
As the book progresses, some of the chapters deal with less advanced ideas from time to time, but I was still surprised to find how many fascinating and under-reported factoids show up even in seemingly elementary chapters, like the ones on audio connectors and musical instruments.
Towards the end of his book, there’s another welcome treat as Winer devotes a full 150 pages to the subjects of monitoring and acoustic treatment; both areas where his professional expertise is especially welcome. This is one section where Winer clearly transitions from skeptic to advocate, and it becomes clear that his career in acoustics has stemmed from his love of measurement and analysis, rather than the other way round.
Quibbles and Qualms
If I ever quibbled with Winer while reading his book, it tended to be over matters of taste, presentation and emphasis – almost never over accuracy.
For instance, Ethan Winer is a vocal proponent of recording “clean and flat” and deferring judgments about EQ, compression and effects until later. While this may be sensible advice for beginning recordists and for some styles of music, nearly all of the great creative engineers I know tend to get a sonic vision in mind and pursue it ruthlessly from square one.
Once you have the confidence and experience, our side would argue, why would you defer decisions, leave thousands of options open and risk “analysis paralysis”, when you can instead make the music sound the way it should right now and then again every step along the way? I’ve definitely drunk the Kool-Aid on this front and tend to roll my eyes like a teenager whenever Ethan advocates the opposite.
Although I respect Ethan’s efforts to restore sense and balance to an often insane and unbalanced field, there are a few isolated moments where Winer may make minor rhetorical over-corrections in his mission to dispel widely-held misconceptions.
For instance, when Ethan debunks misunderstandings about phase-shift in equalizers or the stacking effect of preamps, he under-emphasizes the fact that both of these can still be considered measurably audible phenomenon under certain circumstances Still, Ethan’s corrections on these subjects are valid, and reading his take will add depth to anyone’s understanding of these ideas.
There are a very few, minor internal inconsistencies in this 1st edition of the book, which I hope would be corrected in a future edit. At one point, Ethan states that “there’s no such thing as subharmonics”, but later tells a story of how instruments can create subharmonics through acoustic IM distortion.
Similarly, his wording is contradictory in a brief passage when he discusses whether distortion and noise are meaningful side-effects of recording at low levels in 16-bit digital devices.
In both cases, these come across as a fleeting semantic oversights rather than full-on errors, and over the course of 650 pages they were the closest things I could find to inaccurate statements.
These are minor quibbles, and thankfully, the author does us the service of making his biases plainly known so that readers can filter through his personal opinions if desired.
More often than not, his positions are balanced. Although Winer stresses the importance of accounting for the effects of expectation bias and placebo in our perception, he also clearly states that there are often times when it is possible to hear real differences between devices. He also allows that other engineers are not “wrong” for preferring gear with a “colored” sound, even though his own preferences skew far in the other direction. And finally, he admits to sometimes enjoy the sound of printing to tape or using digital sound-manglers to similar effect – even though his own ideal is to capture and playback audio with gear that is as transparent as possible.
The Final Analysis
It’s difficult to say exactly what kind of book The Audio Expert is. On one hand, it’s well-researched and thorough enough that it could be used to teach a college-level audio class. On the other hand, it’s too personal, too opinionated and has too much of a sense of humor to pass as a straight-ahead textbook. In the end, the effect lies somewhere between academic reference, manifesto and consumer guide.
Winer is guided by some basic principles in his writing – Chiefly, that all audio which reaches our circuits and our ears consists of measurable components; and that when it comes to purchasing audio devices, scientific testing and blind listening are preferable to magical thinking and blind superstition.
“The amount of public misinformation about audio is staggering,’ Winer writes in his introduction to The Audio Expert. “This book, therefore, includes healthy doses of skepticism and consumerism which to me, are intimately related…There’s a lot of magic and pseudo-science associated with audio products, and often price has surprisingly little to do with quality.”
This ingrained skepticism and bias toward consumer advocacy is a welcome perspective in a field that so often becomes obsessed with its tools and the hype behind them. Winer’s critics suggest that his is an attempt to rain on the pro audio’s parade and to make professional engineers look foolish.
But although he can seem prematurely dismissive from time to time (especially on the 71st page of a 91-page forum thread) Winer’s real mission is to shift the focus from the things that barely make a difference and toward the places where the benefits are measurably huge.
To him, this means focusing less on mic preamps and more on microphones, less on high-end clocks and more on high-performing speakers, less on D/A converters and more on room acoustics. In every case, he’s accurate to say that upgrading the former will usually make for a tiny improvement, compared with the benefits of the latter.
For all of the thought Ethan Winer gives to audio, it’s also clear that he understands first hand that even bigger gains come from a factor that lies outside the circuits: Time spent in study and practice of the music and the craft. If you like to spend some of that time reading, rest assured that there’s no way you could read through The Audio Expert and not come out smarter than you went in.