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Recording 101 teaches us that the audio spectrum is 20-20,000 Hz and it is our job as recording engineers to manage those frequencies. For introductory level classes, that is a usable definition, but it often leads to misunderstandings. >Do we hear 20 Hz as much as 20,000 Hz? Do we hear those frequencies as well as 2,000 Hz? The answer to both is no. In fact, given contemporary technological limitations, it isn’t even possible to accomplish most of that.

For those of you who read Jay’s Primer on Audio Frequency Bands and made it all the way the bottom, you would have read some interesting things about broadcast standards and encoding algorithms.  Broadcast standards here in the US actually cut off frequencies above 15 kHz.  That is, radio and television broadcasts don’t even bother with the top 5000 Hz of the audible spectrum!  If there were such a thing as radio anymore, you’d know to laugh off any audio engineer who promises you “radio quality mixes.”  Also, cutoffs are employed in almost all digital encoding algorithms in order to prevent aliasing of upper frequencies.

On the other end of the spectrum, most playback systems are not designed to go below 30 Hz.  Currently, the lowest reproducible frequency by any JBL system is a live sound reinforcement loud speaker with woofer that goes down to 25 Hz.  They also have consumer and studio woofers with roughly the same specs.  You’ll notice that these are all woofer systems and not standard speakers for desktop and meter-bridge monitoring.  The standard studio monitors without a woofer falloff sharply at ~45 Hz.  With this in mind, you should know not to expect to hear anything below 40 Hz on a standard system without a woofer.  Furthermore, you should know that about 90% of your audience will not be able to physically reproduce anything below 50 Hz given the standard consumer set up.

This is not to downplay the psychological impact of low or high frequencies.  These play a very important role in psychoacoustics.  Low-lows, though inaudible, help us perceive lowness partially through feel rather than sound.  High-highs also help us perceive presence and therefore clarity by giving more emphasis to the minutiae of a sound that you’d only hear by being close to it in the real world.

Next week, I’ll clearly define the component regions of the audio spectrum and talk about the various ways to treat undesirable maladies afflicting them individually.

More from Phil’s Audible Spectrum series:

• The Audible Spectrum, Part 1: Introduction

• The Audible Spectrum, Part 3: Components
• The Audible Spectrum, Part 4: Sub-Bass Issues
• The Audible Spectrum, Part 5: Harmonics, Overtones, and Formants Primer

The blog has been on hiatus for the month but we will start writing again in the next 10 days or so. We will kick things back off with a bang, I promise.

(In the mean time, follow us on our shiny new Twitter page: @FixYourMix)

I have lived most of my life in the analog world.  That is a strange thing for someone under 30 to say, but it is the truth.  Sure I am fluent in Pro Tools and all things digital, but the very first studio I worked at was all analog.  I’ve worked at numerous all analog facilities and at my new studio of choice I am pioneering the slow but sure shift from analog to digital.

Keith and I have touched on a couple of the reasons why people prefer analog to digital, but one thing came to mind a while back when I was working on a piece of analog gear.  I was in the shop checking some components on a UREI LA2A when I realized that the box was still plugged in… 

I realized this then immediately felt a slight tinge or tightness in my arm.  It wasn’t necessarily painful—it was just strange.  I had never been so stupid as to electrocute myself while working on a piece of gear before (although other studio techs have comforted me by saying that you haven’t really worked on electronics until you’ve been electrocuted).  It only took one experience with this very undesirable sensation to realize that I would never be that careless again.

Of course my highly associative mind wouldn’t just let it go at that.  I started to think about BF Skinner and his experiments electrocuting rats for negative reinforcement.  Then I thought, well what other lessons have I learned through such experiences?  Perhaps the most pervasive lesson we all learn, rightly or wrongly, contributes to our perception that analog sounds better than tape.

Now before I go any further on this notion I have to give some background on Operant Conditioning.  Skinner used this term to describe how the consequences of an action affect our willingness to perform that action in the future. 

Imagine a rat in a cage with a metal floor that shocks the rat as soon as it touches the floor.  Inside the cage there is a button that, when depressed, stops the electrical shock.  You can imagine the rat scurrying around in the cage, frantically trying to avoid being shocked.  When the rat chances upon the magic button that stops this torturous experience, the rat then associates the button with the cessation of the electrocution pains.  With repetition, the rat learns very quickly that as soon as he’s put in the cage, he should run immediately to the button in order to stop the electrocutions.

I think this operant conditioning might be a consideration when we consider analog versus digital.  In the digital domain, when you overload your DAW, you immediately get a big red light that sits at the top of your meters; your audio distorts with god-awful distortion that only the blackest of black metal heads would appreciate.  The red light is a stamp of your failures as an engineer and the distortion assailing your ears is your punishment.

In the analog world, if you overload your tape machine you don’t get the same kind of immediate and undesirable response.  If you overload in the analog world, you get tape compression/tape saturation.  Often these effects go unnoticed to the casual and professional listener.  Sometimes these effects are desirable: Rupert Neve now makes a tape simulator with saturation emulation capabilities.

Your analog meters don’t hold your overs infinitely.  There is no stamp saying that you couldn’t manage your level to tape, staring you in the face after laying down a track.  In fact, many engineers calibrate their VU meters so that the optimal level hits somewhere after the red starts.

On my analog board, I know that generally the optimal level to tape is when the meters just kiss the red.  Neve Boards are famous for sounding good when slammed into the red while SSL boards generally want to operate well below.

The thing about negative conditioning and conditioning in general is that the consequence has to be immediate.  When you are tracking 24 tracks of instruments on a tape machine, you don’t actively monitor how many times or even if your individual tracks tap into the red.  You just listen.  Sure you get a level before the song starts, but that is just a guess before the real show.  You may find later on at mix time that, during a particularly loud section, your trumpet track got a bit lively and had a hint of saturation on it.  No biggie.  It sounded good in the mix, it’ll still sound good in the mix, so no worries.

With digital, you see it immediately.  While tracking you see it go into the red.  Or maybe you didn’t see it go into the red, but when you look at your meters they are definitely red.  Now you are sweating it:  when did it peak?  Why didn’t I notice it?  Does it sound ok?  I’ll have to remember to go back and check.  Maybe I can keep some of it…Then the loud section comes.  Shit, everything sounds all distorted.  Gotta stop and start over.

For negative conditioning to work, the consequences have to be immediate and undesirable.  You can’t expect the rat to learn that the button turns off electrocution if it doesn’t stop the electricity until hours after the button is depressed.  Similarly, we often don’t experience the negative consequences of tape saturation until long after the tracks are laid down or we never experience them at all.  It’d be like putting the rat in a cage that was mildly humid instead of electrocuting the bejeezus out of him.

With digital overages, our experience is immediate and visceral.  We see the red; we hear the distortion.  We know that we do not want that stuff so we avoid it.  And it only takes a few times of this occurring for us to say jeez it was a lot easier with tape.  Like the rat, we quickly learn that the way to alleviate our situation is to run straight to tape.

If we can slowly and consciously divorce ourselves from our tainted histories with digital and analog (in more capacities than this example), we can finally have a real discussion about which sounds better—analog or digital—or if there is any palpable difference at all.  I think that if we can overcome the stigmas of our experiences with digital, we’ll find that it really doesn’t matter in most applications.

Over the course of hundreds of interactions with clients through Fix Your Mix, both in a mixing and mastering capacity, I have noticed that there is a great disagreement out there on the practical frequencies in audio.  This is strange to me because we have such a vague lexicon for our enterprise (boomy, boxy, tinny, etc.) that you’d think we’d all latch on to terms with such defined parameters as Low, Low-Mid, High, et al.

But nevertheless, every couple months I get a client who says “I love the mix, but I’d really like to hear more bass, can you boost 10 Hz by like 5 dB?”  So for all of you loyal readers out there and as a reference for future clients, I have composed a series of articles describing the portions of the frequency spectrum.

Here is an excellent primer for discussing frequency ranges. Jay works in post-production (television, film, etc.), so his end goals are different from those of us in the music business. He also neglects to emphasize the importance of upper frequencies for imbuing a recording with presence, clarity, and professional quality.  But other than that it is an excellent breakdown of the frequency bands.  For this week though, we’ll be talking about the audible frequency spectrum at large.

The audible frequency range is generally accepted to run from 20 to 20,000 Hz.  Some people hear more, most people hear less.  However, it is important to understand that this broad frequency range is supposed to include the frequencies that the average person is physically able to hear.  For the purposes of experimentation, frequencies outside of the range can be heard, but they have to be amplified to such an extreme that they are not worth measuring.

To the left is the Fletcher-Munson Equal Loudness Curve, established in 1937.  It is probably the most cited graph in psychoacoustics (although the Robinson-Dadson Equal Loudness Curve of 1956 has been shown to be more accurate, since Fletcher-Munson is the most widely used, the following commentary will focus on that).  This graph plots sound pressure level (SPL) in phons against frequency.  The lines indicate equal apparent loudness.  That is, if you were to follow each line, from 20 to 20k, you’d see the variation in amplitude necessary to make each frequency sound equal in loudness.  For example, on the top curve, take 1000 Hz sounding at 120 phons as the baseline.  In order to hear 20 Hz at the same apparent level, you’d have to amplify it to 130 phons.  The same goes for 20k.

Another interesting phenomenon about this curve is how exaggerated the differences become at lower amplitudes.  For instance, when you look at 1000 Hz at 20 phons (the third line from the bottom), you can see that it takes almost 80 phons to sound at the same apparent level.

Now bear in mind, this is not to say that you want to go and quadruple your bass content to get a booming mix.  On the contrary, this is to say that you really shouldn’t expect to hear anything beyond a certain points in the mix.  In almost all instances of music recording, there will be frequency content below easy audibility.  The point of mixing is not necessarily to make them audible.  Sometimes these frequencies are meant to be felt rather than heard.  Other times, these frequencies don’t really add much to the mix at all—eating up large portions of the usable power spectrum and overloading your mix with unnecessary content that either will hurt fidelity due to digital encoding or broadcast algorithms, or will be cast off anyway due to physical limitations of sound reproduction systems.

Here is a graph of all the frequency ranges for common instruments and their notes as shown on a piano.  What you’ll notice is that the range for a concert bass is from ~90 Hz to ~350 Hz.  The absolute lowest note on the piano is around ~28 Hz, and that is a note that you will likely never hit.  Practically all the action in musical instruments occurs between 60 and 5000 Hz.  Allowing for formants, harmonics, and other sonic phenomena outside of the fundamental frequency of the note, it is safe to say that practically all usable and desirable sounds fall within 20-20K and that range could even reasonably be made smaller.

In next week’s article I will examine these specific limitations and discuss why the low frequencies are the most problematic.

More from Phil’s Audible Spectrum series:

• The Audible Spectrum, Part 2: Capture and Reproduction Limitations

• The Audible Spectrum, Part 3: Components
• The Audible Spectrum, Part 4: Sub-Bass Issues
• The Audible Spectrum, Part 5: Harmonics, Overtones, and Formants Primer

In 1978 the Yamaha NS-10 first hit the home audio market. The speakers were originally designed for the consumer rather than the professional sphere. The only problem was that the speakers sounded terrible and no one wanted them for that purpose. They were often described as overly bright and harsh and the frequency response was abysmal in the low end (criticisms which are founded and still exist to this day). However, despite its audiophilic shortcomings, Fate found other uses for this Little-Speaker-That-Couldn’t.

As New Wave, punk, and other lo-fi genres began to take hold on the world, a DIY spirit took over and smaller, cheaper recording studios were created that catered to a clientele who didn’t necessarily place a premium on fidelity. Near-field monitoring became the fashionable choice for these studios because it minimized the effect of listening environment on the sound of a mix. This allowed bedrooms, basements, strip-malls and other ostensibly acoustically unsound venues to become mixing environments.

In these situations the NS-10s weaknesses became strengths. Their lack of low-end capability meant that room nodes (standing waves in a listening environment which cause certain frequencies to be accentuated because of the geometry of the room) weren’t much of an issue since these acoustic phenomena are largely confined to the lower frequencies. Furthermore, their use with cheaper, lower output amplifiers (as was common in these smaller studios) meant that the program output was lower. These volume levels are generally agreed to be the NS-10s’ most accurate operating range. And of course the price, as a previously undesirable commodity, was just right for small studios.

Over the course of the 1980s, the NS-10 became a mainstay of the recording studio and their ubiquity, coupled with the fact that their poor sonic characteristics generally do not incite the individual characteristics of a listening environment, meant that the NS-10 could become a fairly universal reference. By and large, NS-10s were thought to sound reasonably similar in every listening environment. Thus, most mixing decisions are themselves adequately portable.

However, the NS-10 is only as useful as you are familiar with its sonic characteristics. A +7 dB peak at around 1500 Hz contributes to the audibility of some mid-range sounds such as the human voice and acoustic guitar. Operating without this knowledge may result in a weak vocal or acoustic in the mix when you take your songs to other environs.

It is also very difficult to judge a mix’s low-lows on NS-10s. The speaker simply was not designed to reproduce those frequencies. If you aren’t aware of this, then you may find yourself pumping in a ton of low-end just so that the sub frequencies are audible, but if you took it to the club, you’d probably blow out the speakers with all that 808!

It is now agreed in most professional circles that NS-10s are an excellent reference at low volume levels and for gross judgments that do not invoke the sub-frequencies. Armed with this knowledge, you’ll have a better understanding of how to use this omnipresent piece of gear and knowing how to properly use a tool is the most important part of the audio world.

Many recording enthusiasts like to fixate on “things” that separate them from the big boys. Some readers of Gearslutz.com would love to believe that the reason they aren’t Bob Katz is because they don’t have some $20k piece of gear. It isn’t because they lack experience or talent, it’s because of this one thing and if only they had it, they’d be certified hit-makers.

The fact is that once a certain base level has been achieved, the differences between engineers are almost entirely experiential and philosophical. If you took any unanimously great engineer and put him in your home studio, chances are he’d still make a better mix than you would if you had access to his professional facility. In the audio world, the clothes do not make the man.

So what is necessary for good mixing? In my experience, great mixers need only three things: an excellent listening environment, the ability to make targeted changes, and the experience to know which changes are necessary.

An excellent listening environment does not mean that you need to spend millions of dollars to get your room analyzed and treated. All you need is to really understand how your room sounds and how that translates to the outside world. Investing top dollar in studio acoustics is meant to accomplish one goal—to remove the elements that color a sound and make a room a unique listening environment. Acousticians strive for “portability” which is the ability to take something out of one space and have it sound the same in another. That way, the hundreds of clients who come into big-time studios can all trust that the decisions they make will translate when they leave the building.

For home recordists, it is not necessary for you to make your listening environment universal. Instead, you really just need to make sure that you understand your listening environment. If you understand that your room has a 120 Hz standing wave throughout it, then you know to compensate for that in order to get a good mix. This may mean that your mixes sound awful in your listening environment, but you know they will sound good everywhere else.

Yamaha NS-10s which became the studio standard back in the 70s. Even today, they are a mark of a serious recording studio. The reason these speakers are so prevalent is not because they “sound good” or because they are veristic in any environment. Rather, the NS-10 is a standard because practically anybody who is worth their salt knows what they sound like. The speakers themselves sound like absolute crap to put it plainly. But engineers, consciously or unconsciously, know how to make changes on these speakers so that the mixes sound good everywhere because they know how these speakers sound.

(Aside: it is important to note that despite their ubiquity, NS-10s are limited in their usability because they lack the frequency range to accurately depict low-end. Also, if you attach a sub to NS-10s, prepare to be laughed at.)

The ability to make targeted changes is the simplest criteria to fulfill. Digital Audio Workstations are so advanced that nowadays you can make practically any change that you would need. The real problem is knowing what changes are necessary and executing them properly.

That is the biggest issue that home recordists and nascent engineers face. The kind of calculation that is necessary for a mix is not intuitive. You can solo your individual tracks and have a gorgeous piano, a breath-taking drum set, and a rocking bass, but when you put them together suddenly everything gets muddled. Mixing is not an exercise in arithmetic—it takes the experience to know when less is more. Sometimes making one thing sound terrible by itself will ultimately make everything sound great when it all comes together.

So the next time you are ogling that Fairchild on eBay, remember that it’s not what you have, it’s how you use it.

There are some instances when a limited amount of knowledge can do a great deal of harm. For instance, you might know that a bit of sun is good for you. If you are not fully versed in the effects of sun exposure to the skin, you might be wondering what those strange, asymmetrical spots are that keep popping up all over your body. Get those checked out; seriously I worry about you sometimes…

Other times, a basic understanding of something might be helpful the most of the time. Take Euclidean geometry for example. If you aren’t an astrophysicist or a nuclear scientist, pretty much everything you need to know falls into Euclidean space.

But there are also times when the common sense understanding of something gets you by enough so that you don’t realize all the other times that it is absolutely wrong and leads you astray. This is the case with our friend the decibel.

I was working on a record a while back with producer/engineer extraordinaire Paul Kolderie (Radiohead, Pixies, Mighty Mighty Bosstones) and he mentioned something in passing that really caught my attention. I can’t really recall what the situation was, but we were setting up a session and he said to me “I can’t stand it when people ask me to change something by half a dB. A dB is the lowest possible change you can perceive, so saying half a dB is meaningless.”

Many nights I woke abruptly from sleep in a cold sweat tormented by what he had said. Something sounded so right and yet so wrong about that. I mean, if I told you to change something by half a dB twice—both equally insignificant changes by his definition—I would get a change of full dB, and therefore a significant change. Using some simple extrapolation, you can’t keep considering fractional changes in decibels as insignificant, because surely enough they add up.

So what exactly is a dB and what change in dBs is significant to our ear and in our mix? Well, without getting overly scientific about it and also restricting the question to audio applications (sorry electrical engineers), a decibel is a convenient unit of measure that expresses very large changes in magnitude against a reference level in a concise manner. Concision was important back in the days of hand calculation.

When they were busy wiring up the world for telephone usage, Bell Laboratories thought it’d be really swell if they could measure the amount of degradation in audio level over a mile of telephone cable. They did the calculations but soon found that expressing the quantities in conventional terms meant using insanely large and unwieldy numbers. So they decided to use a logarithmic function to bring the numbers to more manageable figures for simple calculation. Logarithms of numbers are useful because they have some of the same arithmetic applications as regular integers (for example, you can add two logarithms with the same base just like adding to regular numbers). The unit they came up with became known as a Bell in honor of the company and Mr. Alexander Graham Bell. So a decibel is actually 1/10 of a Bell.

So why do we talk about tenths of something? After all we don’t regularly deal in decimeters or decigrams. Well in the mid 1800s, some very clever psychophysicists began studying something called Just Noticeable Differences (JND) in sensation. A JND is the smallest incremental change in a sensation that is perceptible to the average person. This could be the JND in touch as measured in PSI or the JND in sight as measured in lumens. Someone discovered that a tenth of a Bell roughly correlated to the smallest detectable change in a sound to the human ear. As such, the decibel became a very important measurement in audio because it was simple to express changes that actually meant something with regard to common perception.

It is important to note that JNDs relate to the AVERAGE person. As such, musicians and audio professionals are often able to detect much more minute changes in audio level.

When studying JNDs, another useful but perhaps counterintuitive aspect of the decibel arose—a doubling of volume roughly correlated in a change of +/- 10 dB. This is useful but strange in that the arithmetic is skewed—you ’d expect a doubling in the perceived volume of something that sounds at +2 dB to be +4 dB. But then again, what is a doubling of something that measures 0 dB? This exposes some of the fundamental limitations in the simple definition of the decibel—human perception complicates the simple calculations.

Such problems spurred further investigation into situational applications of JNDs and Signal Detection Theory was born. In basic terms, the object of Signal Detection Theory is to figure out what extra factors go in to our perception of a sound and how it compares against “noise” or unrelated signals. For instance, does a +1 dB change to a signal still sound like an increase of 1 JND if the sound is played over white noise? What about if the original signal is 100 Hz sine wave? What about 30 KHz?  What if the original signal is a voice played over a country band?  Or a metal band?

It was discovered that the JND of a signal changes based on frequency range and initial level. A JND is around 1 dB for soft sounds at frequencies in the low and mid range—the frequencies we perceive most readily. Really loud sounds can have a JND of 1/3 to 1/2 dB. Really soft sounds on the edge of audibility might have JNDs of a couple dB.

Furthermore, other things can color sounds in such a way that you can take the same sound, add something to it and suddenly the JND might be more or less than a dB. Perceptual Encoding Theorists look for factors outside the Critical Band of Frequency for a sound (the frequency or frequencies that define a sound) that would alter our perception of it. For instance, adding a slight reverb in some cases might cause the JND to rise (meaning you need to turn the signal up more to get a perceivable change) or adding a harmonic exciter in most cases would cause the JND to lower (meaning you wouldn’t need to turn the signal up as much to get a perceivable change). This is because new nerve endings are being excited and these cause our minds to perceive the sound in a different way than we had previously.

As you can see, the decibel is not quite as simple as its common sense understanding in the audio world. So when you need to make something appear twice as loud, you know what to do. When somebody tells you to make their vocals 20 dB louder, you know that that is laughably extreme (for the most part) and you should adjust your corrections appropriately. When someone asks you to turn something down by 1/3 of a dB, you know that it is really only going to be detectable if that sound is already pretty loud.

Thanks to all of you over at Reddit for voting up this article. If you are a self-taught musician, you may find it helpful to check out my Solfege To Intervals Translation Chart to follow the melodic analysis.

This week, I’m going to break down the music theory behind one of the most unusual pop songs to come out in years: Beyoncé’s  “Single Ladies.”

Tempo: 87 BPM*
Key Signature(s): E major, E minor
Special Songwriting Devices Used: No back beat, Polytonality (technically polymodality**), Resolution using a Minor 6 chord, Starting a melody on sol

Several months ago, I was having a conversation with a friend of mine about whether or not this single would flop. Pop music has certainly gotten interesting over the past 5 years, but this song was, well, too interesting. To put it bluntly: “Single Ladies” is just downright bizarre. And yet as time went on, I began to see that it has what I call the Spice Girls Factor–designed to make groups of adolescent girls dance around in their bedrooms, sing into hairbrushes, and post videos of the whole ordeal on YouTube for their friends to watch.

To start, let’s take a look at the groove. In pop music, there is almost always some kind of clap, snap, or snare on beats 2 and 4, also known as a back beat (read my post on back beat options here). “Single Ladies” breaks the mold, especially for a pop song, with claps on every 8th note, which gives the song an uptempo-feel. In fact, to me these claps give the song more of a “1 feel” rather than strictly 4/4, which would mean every quarter note is an equally strong beat. Normally only beats 1 and, to a lesser extent, 3, are considered strong beats. Strong and weak beats become important when understanding how melodies and chord changes affect perceived key signature or tonality. This “1 feel” theory is reinforced by the dancing in the music video, in which the choreography consists largely of Beyoncé jolting around on every beat.

But it doesn’t stop there.

There is a snare drum in this song, and like virtually all hip-hop out right now, it’s not used as back beat. However, where normally hip-hop draws the line at syncopated southern-style fills or dotted 8th note patterns a la “A Milli“, there is a snare hit on the last 8th note of each measure (AKA the “and” of beat 4). This, combined with the 8th note claps, plays a big role in giving “Single Ladies” its memorable feel.

Now let’s move on to the harmony. During the song’s call-and-response section (”All the single ladies, all the single ladies”), she sings solfege syllable*** sol (as in do re mi fa sol) then riffs on mi, re, and do. Sol is a very common beginning note for a pop melody, adding strength (rather than color) to the harmony. Also note that she skips fa, which is common practice for melodies sung over a root chord because it forms a weak interval, a perfect fourth.

As I talked about in last week’s analysis, in traditional harmony and counterpoint, we only need a major or minor third interval to imply a chord. Beyoncé does exactly that during the verses: solidly establishing the key of E major by singing only an E and a G# with the occasional F#. The only “music” during the verses is a pitched noise, though the notes are indistinguishable, keeping in line with the current pop minimalism trend (see: 5 Pop Songs With No Music).

Pretty basic stuff so far. Now here’s where things get really interesting:

During the chorus, a bass synth comes in and goes from B to C, which is the bVI chord borrowed from the key of E minor. I will be talking a lot about borrowing chords from related keys and tonalities (aka modal interchange) in future Compositional Analysis posts, but what makes “Single Ladies” downright bizarre is that the melody doesn’t reflect this change in harmony at all, so what we’ve got is music in E minor and a melody in E major. This is called polytonality**, a technique normally reserved for highly esoteric jazz and classical music.

The result is a striking juxtaposition: a nursery rhyme-esque melody with a powerful, sinister bassline beaneath it, creating a bitter, almost shocking melancholy which underscores the “strong woman” image for which Beyoncé has become an archetype. The melody is distinctly feminine and “cute” while the bassline is aggressive and forceful (usually thought of as masculine traits). It is probably no coincidence that the bassline enters with the line, “if you like it then you shoulda put a ring on it.” Here, the woman asserts her control over a man.

All this being said, she could not have pulled this song off were it not for a sparse arrangement, an exceptionally catchy beat, and the clout of being a well-established, top female artist, not to mention a role model for a generation of young, ambitious women.

Some music scholars might take issue with my assessment, in fact some don’t believe in polytonality at all, saying our ear cannot perceive two tonalities at once. With an arrangement this sparse, though, their case holds little weight.

But just for the hell of it, I’m going to do a standard harmonic analysis of this tune anyway, as if it were all in one key. Things often get vague when it comes to analyzing modern pop music because the harmonies are so fragmented. You rarely hear a full triad or seventh chord in rap and dance-oriented R&B these days (though I believe this trend is about to change) and “Single Ladies” is no exception. The result is often some funky looking chords with half the notes missing. Case in point:

Roman numeral analysis
1st Measure: V (no3), IV-/b3, III+ (no #5)/3, bVI (no3)
2nd Measure: V (no3), IV (no3), IVmaj7 (no3), IV-6 (no3)

Chord chart
1st Measure: B (no3), A-/C, G#+/3 (no+5), C (no3)
2nd Measure: B (no3), A (no3), Amaj7 (no3), A-6 (no3)

Chords in laymen’s terms

1st Measure: B with no third, A minor first inversion, G# augmented first inversion with no (augmented) 5th, C with no third.
2nd Measure: B with no third, A with no third, Amaj7 with no third, A minor 6 with no third.

Here, the two chords to watch are III+/3 and IV-6. The third chord in the progression does sound like a III augmented in that it is especially dissonant, but it’s also not functioning in a way that augmented triads are supposed to function (such as leading to the IV chord). And unlike major and minor triads, you are technically supposed to have the fifth when it comes to augmented or diminished chords. Augmented and diminished fifths cannot be implied. This again leads me back to polytonality because we only have two notes from the chord.****

The very last chord in the chorus sounds like it’s implying an A minor 6 chord (minor triad with a major sixth–A C E F#), though only the sixth is present. I say this is minor six rather than a II-/3 because I hear a strong pull back to the I, something a IV-6 has and a II- does not.

I hope you’ve enjoyed this analysis. I realize that this song is not for everyone, but it’s very important for songwriters to think about songs like Single Ladies, the songs that stretch the boundaries and yet are still wildly successful. It can speak volumes about how people connect with music, the future potential of music, and the realm of what is “commercially viable” (if you care about that sort of thing).

Buy Beyoncé’s “Single Ladies” on iTunes

*While I have the tempo listed at 87 BPM, you could certainly argue that “Single Ladies” is in the upper 160-200 BPM range, making the claps quarter notes and the kick drum pattern repeating every 2 bars. For the sake of discussion, though, I chose to analyze this song at a typical hip hop tempo. This makes the snare and kick drum patterns one instead of two bar phrases.

**As many readers have pointed out, it’s actually more accurate to call this polymodality because the ‘tonal center’ is still E even though the scale is different from E minor to E major.

***Maybe it’s because I’m a guitarist and singer, but I like to think of melodies in terms of solfege syllables because they are instantly transferrable from one key to the next.

****The third chord could also technically be a III/3 chord, but in every music course I’ve ever taken, teachers have advised against analyzing something as a major III chord, let alone an inversion of it. Such a chord does not exist in any mode, so it could not be borrowed. The only other real possibility would be if it had a flatted seventh, making it a V7/VI- chord, but there is no indication that this is the case nor is that possibility even within the scope of this post.

If you’ve ever had a 9-5 office job, then you are well aware of the break room refrigerator. It is a place of great temptation and enormous possibility. When you roll in to the office in the morning and unload your brownbag special onto the bottom shelf, you can’t help but see that beautiful Salisbury steak in the Tupperware container on the shelf above. My friends, there is nothing more tempting than someone else’s lunch.

Do you remember the lunch time trades in middle school: swapping Lays for Doritos, peanut butter and jelly for ham and swiss, Fruit Roll-ups for Fruit by the Foot?  It’s not even that you like Lays better than Doritos, but when you sit down at the lunch table or peer into the refrigerator, you know what you have. You’ve known since you packed it that morning. You grow to hate that tuna on white because it is a foregone conclusion. But somebody else’s lunch—that is new and exciting. It is surprising because it materializes out of the ether: without you spreading the mayo, without you packing it up, without it damning you from the passenger seat during morning traffic. It is a wholly unexpected option that presents itself without any input on your part whatsoever.

I think the same scenario applies with user interfaces: tape versus DAW, plug-in versus outboard.  They all can do the same sort of things, and you know that once you get to a certain point there really isn’t that much difference between them, but for some reason analog just plain sounds better. Part of it is that the inner workings of an analog piece are obscured by the interface.

Below are two common de-essers, one outboard analog the other a plug-in.  Which one seems more intuitive to you?

With analog outboard gear, the exact inner workings of this magic box are a mystery. Sure you know that you turn this knob and the compression ratio changes or the EQ bandwidth shrinks, but exactly how it does it is a mystery to most. The hieroglyphics on the faceplate often have no intrinsic merit, but rather are arbitrary increments that represent something else more accurately. (Think of that infamous amplifier in This is Spinal Tap. It goes to 11, but you could just make 10 louder and then there is no need to go to 11. 10 and 11 are just arbitrary units that co-relate to a specific amount of gain, but that amount is obscured by the interface.)

In a basic sense, you turn this knob and things suddenly sound better without you necessarily knowing the specifics of what is being changed and by how much. It’s simpler, it’s more elegant, and the results are not visual.

With a plug-in, you often see the changes being made. In a digital EQ, you see a visual representation of the frequency range, the shape, and the amount of change. The visual feedback causes you to judge things differently than turning a knob divorced of stimulus response. (Have you ever heard somebody say that your plug-in settings don’t “look” right?  Isn’t this the audio business?  How does it sound?)  Users are more prone to making larger adjustments with visual feedback systems than they are when turning knobs—just look at any plug-in’s presets and you’ll see that they are usually wildly extreme.

This is not to say that the digital version of an 1176 is exactly equivalent to its analog counterpart. There are indeed harmonic differences, distortion characteristics, and sonic qualities that manifest when electronic components age, but these traits would cause an outboard 1176 to differ sonically from any other outboard 1176 as well. Indeed, given the kind of modeling analysis some digital emulators go through, the sonic characteristics of a digital 1176 are probably identical to at least one analog 1176 out there in the world.

So if the difference isn’t sonic, then the sonic difference likely stems from differences in how we tend to use analog and digital effects. I’ve heard numerous engineers say “Oh that plug-in sucks, I have to turn the compression up to 20:1 before it sounds like anything.” While this maybe true for cheap plug-ins that don’t go through rigorous modeling algorithms, I don’t think we can rule out the possibility that we simply believe a plug-in is different from it’s analog counterpart, so we perceive it differently, and therefore treat it differently.

Some readers out there may be familiar with legendary producer Brian Eno’s Oblique Strategies. These are ways of randomizing your approach to a given situation. Oftentimes the results are surprising and pleasant and the same principle may be at work when we deal with analog interfaces—the fact that we aren’t intimately aware of the mechanics of what is happening when we apply an effect causes us to enjoy the outcome a little more.

So if some vintage piece of analog gear is too obscure to locate or maybe just prohibitively expensive, try out the digital version of it. Approach it like the real deal and let me know if the result is more desirable.  I’d really like to know the results of approaching digital plug-ins with a sort of analog mindset. It just might be that my sandwich and your sandwich are interchangeable and we only perceive a difference because of how we regard them.

Artist: Noisettes
Album: What’s the Time Mr. Wolf?
Released: 2007
Sound: Indie Rock
For Fans Of: Yeah Yeah Yeahs, Janelle Monae
Recommended Tracks: “Scratch Your Name,” “Don’t Give Up”

To get a feel for Noisettes, you have to start by watching frontwoman Shingai Soniwa’s electrifying performance in their music video for “Scratch Your Name.” This is classic case of a lead vocalist taking an act from good to great.

I saw them play a small Brooklyn night club a few summers ago. The show unexpectedly got combined with Battles at the last minute. (This was right around the time Mirrored came out, one of my favorite albums in the last 5 years.) The energy was unreal throughout the night. Since then, they’ve have toured with Bloc Party, TV on the Radio, and Muse.

While their sound is not “revolutionary” per se, Noisettes doesn’t particularly sound like anyone else. It would be a disservice to compare them to the decidedly darker and grimier Yeah Yeah Yeahs (though Shingai does list Karen O as an influence).

When Janelle Monae came out last year, I immediately thought “hey, she’s ripping off Shingai’s look,” (although she’s equal part Andre 3000). Musically, though, Janelle is a solid Motown-era soul singer while Shingai is an epic, class-of-her-own, wouldn’t-want-to-challenge-her-to-a-thumb-wrestling-match rock vocalist. Her signature is when her voice squeaks in just the right place at exactly the right time. You have to hear it to understand.

“Don’t Upset The Rhythm,” the second single from their yet-to-be-released sophomore album, has reached #2 on the UK Singles chart after being featured in a Mazda commercial. This track, along with the other single released from their upcoming sophomore album, ironically seem to indicate a directional shift for Noisettes towards a more processed, dance-y, Janelle Monae-esque sound. Guess indie rock wasn’t paying the bills. I will reserve judgment, however, until the new album drops.

Wild Young Hearts comes out April 20th on Universal.

Noisettes on Myspace