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Archive for the ‘Producer Speak’ Category

What The Hell Is 0 dB? (A Quick Word About Decibels)

Posted by Keith Freund On January - 19 - 20124 COMMENTS

(Updated 2/4/12) Despite a formal education and real world training at some serious recording studios, I’ve always been amazed at how far along I had to come as an engineer just to learn some very basic information and points of confusion (a big reason for starting this blog by the way). When you do something for the better part of your life, you sometimes forget how much you really know. That’s part of why the best audio engineers are not always the best teachers.

When I started out, the decibel confused me.

I had read that a jet engine was 140 dB, a library was 30 dB, a rock concert was 115 dB, etc. Why then, I wondered, do digital and analog meters go from negative infinity (silence) to 0 dB (absolute loudest)? And why is it that track faders can go up to +6 or +12 dB?

I’m not going to get into all the gruesome details here, I’m just going to set forth one basic, but extremely important concept: there is more than one kind of decibel.

Decibels in the digital realm are called dBFS or “Decibels Full Scale.” Decibels in the real world (the kind used to measure a jet engine) are called dB SPL or “Sound Pressure Level.”

Now let’s apply this knowledge to a WAV file you might receive back from a mastering engineer. In the end, how loudly people hear your master is going to be determined by the playback system (iPod, car stereo, studio monitors, etc.) and whoever controls the volume knob.* This means that in the real world, even a recording that peaks at 0 dBFS and is -5 dBFS on average (very loud) can be as quiet as a library or as loud as a jet engine upon playback.

So the dBFS of your master really only indicates how loud your song will be relative to other songs when played back under identical circumstances. So if someone plays a Notorious BIG track that’s -8 dBFS on average, and your track is -6 dBFS, you’ll be louder than Biggie every time (unless some asshole decides to turn it down when your song comes on).

dB Adjustments Within Pro Tools and on Actual Recording Consoles

It’s also important to note that when you set your snare drum’s volume fader to +6 dB in Pro Tools or on a console, you’re not making the snare 6 dB SPL, your snare has not become one twenty-third of a jet engine or one fifth of a library. You’re also not making it 6 dBFS (which doesn’t exist) either. You’re just adding 6 dBFS to whatever the sound file’s initial volume was. So if you record that snare into Pro Tools hitting -5 dB on the meters, and set the fader to +4 dB, your snare is now -1 dBFS. And like I said before, the loudest anything can ever be in the digital realm is 0 dBFS. This also called “digital zero.” (Going over digital zero is what causes clipping.)

There are other types of decibels, but these two are the most important to grasp for now.

Any questions?

*Further still, final perceived loudness must also take into account the distance between the listener and the speakers, as well as dB SPL.

UPDATE: The graphic and article have been updated with a more accurate explanation of limiting.

Last night, I was giving one of our mastering clients some mixing advice regarding vocal compression. I thought I’d post the email here for those of you who might still be struggling with the concept (I know I did for years) or to refer your friends to. Also note that this article is about sound compression or even more specifically “dynamic range compression” and should not be confused with data compression (mp3, zip, rar, etc.) This isn’t a comprehensive explanation of compression, you can find that elsewhere. This is just the bare bones essentials.

I used drums instead of vocals in the diagram below because I felt that would be the easiest to understand visually. Click for full size:

What’s the point of compression anyway?

Back in the old days, vocal compression was used for one purpose:

Compression makes the volume of a vocal more consistent overall.
In fact it was originally called “Automatic Level Control.” So if you’re singing or rapping some words louder than others, compression makes for a less drastic volume difference between the loud and quiet parts.

So above all else, vocal compression makes the lyrics easier to understand and helps to keep the words from being drowned out by the music. Obviously you could just turn up the vocal track to solve this problem, but then you’d have a new issue: when the singer hits a loud note it will now be way too loud compared to everything else, startling the listener and causing them to turn it off because they don’t like being startled. Or more likely they just turn it down. And you want people to blast your music right?

How compression works.

It’s pretty simple: Every time the compressor hears a sound that goes over a certain volume level (that’s your “threshold”), it turns down the volume for that moment.

Makeup gain.

After compressing a vocal track, it will be quieter than what you started with. To make up for this lost volume, most compressors have an output volume knob or “makeup gain” knob. (For our purposes, gain and volume are the same thing.)

So although technically compression turns down the loud parts with the quiet parts unaffected, if you raise your makeup gain enough you’ll effectively be turning UP the QUIET parts instead.

Threshold.

As I said before, your threshold is setting the cutoff point. Any word that’s quieter than the threshold will remain untouched. Everything above it will be turned down somewhat. For rap, you probably want to set the threshold so you’re compressing everything but the absolute quietest words. Until you get to the point where you can actually hear compression working (this can take years), you’ll want to use your gain reduction meter for this purpose.

Gain Reduction.

How you set your threshold will be relative to how loudly or softly your vocal was recorded. That’s a big part of why presets are only meant to be used as a starting point. The truly important value to watch is gain reduction, because a gain reduction meter tells you how much your vocal is actually being turned down at any given time.

If your compressor doesn’t have a meter, it’s hard to say where you should set your threshold. As I said before, compression is very hard to hear starting out. And your listeners will only hear it subconsciously. Or they might notice when it’s not there because it sounds like a demo and it’s not as smooth/consistent as a professional mix. But they won’t know why it sounds that way.

Ratio.

After using compression, the loudest notes will still be louder than the threshold, but not as much over it as before (see the diagram above). So post-compression, the difference in loudness between the loud and quiet words will be smaller. That difference is called “dynamic range”  and the ratio determines how drastic that difference is. High ratios (8:1 for example) make for a smaller range. Small ratios (2:1 or 3:1) will allow a bigger range.

Compression Today and the Effects of Attack & Release.

Today, compression is still used to make lyrics more understandable, but back in the day it pretty much only had one knob. More compression, or less. Today you’ve got more controls, so you can have different flavors of compression using attack and release. At the risk of oversimplifying, here’s the effect attack and release settings have on vocals:

Shorter attack and release times (smaller numbers) will make vocals sound more “energetic,” louder, and will also bring out the breaths between words.

Longer attack and release times (higher numbers) will make the sound “punchier” which obviously isn’t as important for vocals as it is for other instruments like kick and snare.

For vocals, generally you’re going to want the attack to be quicker than the release. It’s not worth getting into why for now. I will occasionally set the release time to its absolute shortest value if I’m having trouble getting a vocal to cut through or going for an extreme effect.

What a Compressor’s Attack & Release Knobs Technically Do (Versus Synthesizers and Drum Machines).

If you’re used to working with synthesizers and drum machines, you may be confused about attack and release settings for compressors. Unlike on a synth, the attack and release settings on a compressor do not tell you the attack and release times of the instrument the compressor is affect. You’re setting the attack and release of the compression itself.

Compression doesn’t turn on and off instantly. So attack is the length of time it takes your compressor to start compressing at full force. (How long it takes to go from zero to sixty, so to speak.) Release is the amount of time it takes for your compressor to stop compressing.

So what does all this mean in layman’s terms? Let’s say you’re compressing a snare drum and you’ve set a relatively slow attack time–80 miliseconds. And let’s also say you’ve applied this compression permanently to your sound wave and you’re looking at the waveform before and after. Visually, your snare drum probably already had a big volume spike at the beginning which trailed off, but now it will cut off even more abruptly since it was grabbed by the compressor after 80 miliseconds and taken way down in volume. You may hear this resulting sound as a punchier snare drum.

Now let’s say you set the attack to an EXTREMELY slow speed, like 1 second (1000 ms). You’re not going to be compressing the big pop of the snare drum much at all now because that pop lasts less than a second. We’re talking fractions of a second here. So by the time the compression kicks in you’ve already missed that spike.

If you want to control the attack and release of a recorded sound directly (like you would with a synth or drum sample) what you want is an “envelope follower” or “transient modulator,” such as Waves Trans-X, Envelope (built into Logic), or Oxford Transient Modulator. The most popular hardware version of this is called the SPL Transient Designer–for some engineers it was the secret to the 90s drum sound.

Limiting.

Limiting is the final effect in every mastering engineer’s chain but is also commonly used on vocals (most limiters are better suited to certain applications over others).

A limiter is a type of compressor which has an “infinite ratio”* which means that everything that goes above the threshold gets set to exactly the threshold’s level.

The controls on a limiter are also slightly differently from a compressor. Instead of using your threshold to push down the peaks and then compensating afterwards with makeup gain, for limiters you set the input gain, which pushes up the volume of everything at once, while the threshold is fixed at 0. (Go here to get unconfused about decibels.)

The end result is that although everything has been boosted by the same amount in theory, the volume of the quiet parts is effectively increased much more than for the loud parts, because the loud parts were closer to the threshold to begin with, and as I said before, a limiter doesn’t allow anything to get louder than the threshold.

If you’re working with vocals, you’ll probably want to use the limiter’s output gain control (if it has one) to turn the final result down.

A NOTE ABOUT LIMITING PLUGINS: Very confusingly, some of the most popular limiter plugins (including the Waves L1, L2, & L3) call their input gain control a “threshold” and use negative instead of positive values. In fact, the true threshold cannot be controlled at all–it’s fixed. Your L1 threshold is actually an input gain. The reason they’ve set it up this way is because it’s easier to understand visually. Unlike with most analog limiters, many plugin limiters have a vertical volume meter and use sliders instead of knobs. So the slider is positioned beside the volume meter, and you can drag the slider down as you watch the meter bounce in order to visually choose which peaks are limited and which are not. This allows you to use your eyes instead of your ears to decide how aggressively you want to limit something–do you want to flatten out the volume completely or only the loudest peaks? Or somewhere in between?

Feel free to leave questions in the comments.

*Some engineers consider anything with a ratio of over 10:1 to be limiting.


I got an email from my father today directing me to an internet forum debate regarding how over-compression in mastering is ruining music today and I responded with what you’re about to read–the simplest, most easy-to-read breakdown on this topic I could come up with–an explanation that anyone will be able to understand. No flowery or abstract adjectives, just the meat and potatoes. If something still isn’t clear, leave a comment. I’m mostly posting this article to save myself some time (to direct our clients to) but this post may be helpful for other mastering engineers who’d like to do the same.

Note: this post refers to sound compression, not MP3 or zip compression. Learn more about the differences between sound compression and data compression here.

What Is Mastering Compression?

squashed waveform 2

Mastering compression* (“limiting”) reduces the dynamic range of a recording. Dynamic range is the range between the loudest and quietest point in a recording or section of a recording. Dynamics and loudness are inversely proportional: the less dynamic a recording is, the higher the average loudness of that recording can be, and vice versa. Today, there is said to be a ‘Loudness War’ between labels trying to release the loudest records possible.


The Pros and Cons of Loudness


When we’re talking about the consequences of loudness, we’re really talking about two different things:

  • From a macro perspective: A louder/less dynamic recording means all of the sections of the song will be about the same volume. For ‘Verse Chorus Verse’ style songs, the benefit is that the song comes in loud right off the bat and stays loud from section to section. The downside is it means the chorus doesn’t “hit you” or sound as big as it otherwise might because there is little or no change from the verse to the chorus. In fact, if there’s more stuff going on during the chorus, individual elements may actually get smaller. For example, heavily compressed rock mixes tend to have a bigger snare sound during the verses than the choruses.
  • From a micro perspective: Compression from one beat to another is hard for the untrained ear to hear, unless it’s very extreme. And even then it is hard to explain what it is you’re hearing–you just know it sounds bad. At this scale, compression makes the mix sound more “exciting” right away, but if overdone can be fatiguing on the ear to listen to. Perhaps more importantly, the drums will often be less punchy if a mix is more heavily compressed.** A former coworker and assistant to one of my all-time favorite mixers once said to me, “mastering ruins everything.”


Is Louder Better?


There was a psychological study which showed that people consistently prefer recordings that are louder, even by an increment as small as 1 dB, even when they’re not told what the change is. (Anyone want to find the link for me?) If you want an extreme example of over-compression, just listen to the radio. They use a more complex system of compression to get recordings even louder than CDs. And yet for every person who puts up a fuss in the blogosphere/messageboardiverse about mastering ruining music today, I have a memory of someone in my childhood telling me how they like the sound of radio. It just has that magic je ne sais quoi. There have been studies which indicate that loudness has a strong effect on which radio station a person will stop on when channel surfing.


While I consider myself to be more of a ‘new school’ engineer rather than one pining for the days of yore, I sometimes wish more records today had bigger dynamic changes. My favorite example is “Quiet” by the Smashing Pumpkins, which came out in 1993 before the Loudness Wars really began. It will probably be hard to tell on YouTube, but when the guitar solo comes in on the record, the song just gets so much louder. If you already have the song cranked, the solo will hurt your ears a little bit. Now that, ladies and gentlemen, is rock & roll.


For a case study in consumer backlash against loudness, check out my post about the Metallica album Death Magnetic.


*The type of compression I’m talking about here is “limiting”, a specific type of compression that comes at the end of every mastering engineer’s signal chain. I’m using the term compression throughout the post because it will be more familiar to musicians and readers.


**If I know that a project I’m mixing will be mastered by someone else, I usually try to make my drums a little punchier than I want them in order to compensate for the effect that mastering will have, unless I know the mastering engineer tends to go easy on the compression. This is also why it is usually best to select a mastering engineer that your mix engineer has worked with in the past, so that the mix engineer can anticipate what will happen to the mix in the mastering stage and mix accordingly.

What is a Red Book CD? (Producer Speak)

Posted by FixYourMix On September - 17 - 20094 COMMENTS

468082-FBDespite the democratization of music production technology over the past ten years, most of the mastering process is still a black art to most people.  You can buy a cheap microphone for your computer at Wal-Mart to record your bedroom demos, but for some reason you still can’t get your music as loud and clean as Kanye.  The brave can attempt a lot of it at home with varying degrees of success, but an esoteric lexicon still exists that causes many to second guess their ability to deliver a professional product.  “Red Book” falls into this category.

 

If you just google “Red Book CD” or “Red Book Master” you’ll wind up with a bunch of questionable mastering sites offering a lot of warnings but not a lot of information.  They say “Oh, you can make a CD at home, but is it a Red Book??!?!”  Then they proceed with obfuscating generalities that encourage you to accept the fact that you are out of your depth and that you should use their service, in part, to obtain such a mystical artifact.

 

Well allow me to demystify the term and you’ll see that getting a Red Book master is not reason enough to choose a mastering house.

 

“Red Book” refers to a document created in 1980 by Sony and Philips.  A team of about 8 researchers was tasked with creating the specifications for standardizing compact discs (CDs if you’ve been living under a rock…on Mars…with your fingers in your ears and singing “la la laaaaa” to yourself at the top of your lungs for the past 30 years).   Researchers in the companies had a propensity to house their reports on the various forms of CDs in color-coded folders or books and refer to them as such.

 

folders03colored

These eventually became known as the “Rainbow Book Series”.  Lesser-known standards also included “Yellow Book” for CD-ROMs, “White Book” for video discs, “Beige Book” for photo discs, and “Scarlett Book” for super-audio discs.  The specifications found within these books dealt explicitly with standardizing production for the companies.  With Red Book, the virtual monopoly in CD technology at the time by Sony and Philips contributed to most of the other manufacturers adopting the same standards for compatibility across the board.

 

The Red Book stipulates that a standard CD should be 120 mm in diameter, 1.2 mm thick, and composed of specific materials (polycarbonate plastic substrate sandwiching some form of thin metal and coated in lacquer if you want to be explicit).  Pretty much every commercially available CD conforms to these physical standards.

 

BeethovenInteresting aside:  the companies originally wanted 60 minute of play time with 100 mm to 115 mm discs.  The ultimate choice of 74 minutes came from the suggestion by Herbert von Karajan, conductor of the Berlin Philharmonic, that the CDs should accommodate Beethoven’s 9th Symphony, which was recorded in 1951 at the Bayreuth Festival in Germany.   This increased time necessitated the increase to 120 mm diameter discs.  The first test disc ever made was pressed in Hanover in 1981 and featured Von Karajan conducting the same group, so his opinion must have been important to the researchers.

 

Additionally, Audio CDs must have three areas: the lead-in, the program, and the lead-out.  Every disc-burning program that writes audio CDs uses these areas.  The lead-in contains the table of contents and directs CD players to the track markers and song titles and so forth.  The program area is where the actual audio is housed (in Audio Engineering, program is just a term used to describe any kind of sound content).  The lead-out contains no data and indicates to players that the CD has ended.

 

Lastly, the Red Book stipulates what kind of files can be used and how they should be encoded and organized on the CD.  Discs can contain up to 74 minutes of audio, up to 99 tracks with a minimum of 4 seconds per track, with the possibility of 99 separate sub-divisions within each track.

 

The program content must be the standard 44.1 kHz sample rate, 16-bit depth and be two channel stereo.  Data is stored in frames of 1/75 second length and data is written in sector sizes of 2,352 bytes per frame.  Frames are encoded in such a way to minimize the effect of damage to a CD and house error correction and display information.

 

That last bit is a little abstruse, but basically all of these are the standard for every disc-burning program out there.

 

So what’s the big deal with Red Book masters?  The short answer is that there isn’t one—at least not today.  Understand that the Red Book was written in 1980, before commercial audio CDs ever even hit the market.  These were the standards the companies decided on before ever releasing a CD and were to be used as the template for mass producing CDs for the future.  In a large part, the Red Book is merely a description of what an audio CD is, not some uber-special type of audio CD that you need to have special gear to create.

 

Once upon a time, the means of creating audio CDs existed only in big manufacturing plants and the common-folk had to patronize these establishments to get their discs reproduced.  Now, technology has come so far that virtually every disc making tool available to the consumer can and does follow these standards.

 

I suspect that the only reason this term even exists anymore is because of stand-alone hard disc recorder/burners like the Alesis Masterlink, which give the option to record different kinds of discs.  Since many recorders are capable of recording at much higher (and some at much lower) quality than standard audio discs, the Red Book option is selected so that whatever you’ve recorded comes out playable from the burner.  I speculate that the “Red Book” option on these recorders is meant to be a short-hand for burning a playable disc as opposed to a data or archival disc.  Also, since many of the professional hard-disc recorders were made in the 80s and 90s, companies were still tinkering with Super Audio CDs and other forms of discs that might have been included as options.

 

Think back 10 years ago when you were burning CDs, maybe one in every ten or twenty didn’t work or some would only play on a certain brand of CD player, or maybe it only played on your computer but not your car or in your car but not your CD player.  Across the board, these problems have been reduced with error correction and more intuitive interfaces, not to mention the fact that the average consumer now knows the difference between a data disc and an audio disc and can recite the sample rate and bit-depth for Audio CDs.

 

The Red Book standard is in many ways simple antiquated jargon for specifications that we can safely take for granted anyway.  Some might latch on to this term because it’s something they can use to sound more professional than you.  Of course, by and large the people employing the term don’t know any better, they just know that they can push the “Red Book” button and make something without knowing what it is. 

 

If you burn your disc using an audio-CD writing program, using a normal CD-R, and using a modern CD Burner, then you’ve got yourself a Red Book disc.  I would suggest burning at the slowest speed possible to minimize errors, but other than that you are golden!

Common Terms in the High Frequencies, part 14

Posted by FixYourMix On July - 30 - 20092 COMMENTS

Airy:  Spaciousness, often a pleasant mostly treble-based reverb sound.  Extended frequency response that runs up through the top end of the bandwidth.

 

Brittle:  Peaking in high-frequencies, weak fundamentals with slightly distorted or harsh highs.

 

Crisp:  Good high-frequency response with good transient quality

 

Crispy:  Constant, but perhaps random high frequency sounds not unlike frying foods.

 

Delicate: Extended high frequency range without being harsh.  Emphasis on high frequencies extending to 20 KHz, but without the peaking. 

 

Edgy:  High frequency emphasis from about 3.5 to 6 KHz with harmonic content overly strong in comparison to fundamentals.  Can deal with high frequency distortion and rasp.

 

Piercing:  Hard on the ears, screechy and sharp.  Narrow peaks occurring between 3 and 10 KHz.

 

Raspy:  Harsh sounds in the 6 KHz range that sounds like a scratchy voice.

 

Sibilant:  S and Sh sounds are overly emphasized. 

 

Sizzly:  Similar to crisp, like the sound of the decay on a riveted cymbal.

 

Steely or Shiny:  Lots of top end from 3-6 KHz with peaky highs (as opposed to flat but boosted top end)

 

Sweet:  Flat high-frequency response without distortion and extended the full bandwidth.  

The final section of the audible spectrum is the high frequency or treble portion. Humans theoretically are able to hear up to 20 KHz (that is, newborn baby girls can theoretically hear up to 20 KHz at normal listening levels; for the rest of us, considerably lower). So what could happen in the 16,500 Hz range if no new instruments can sound there? 

 

It contains almost nothing but upper harmonics of treble instruments and room tone.  This helps solo instruments and vocals sound present and full, but also adds brightness and clarity to a mix

 

Most telephones cut off around 3.5 KHz yet you can still tell whose voice it is on the phone.  This tells you that practically everything needed to understand and distinguish any audio content pretty much lives below this range.

 

Pretty much only dog whistles operate in this range, so there is absolutely no need to worry about any more fundamentals or really any lower order harmonics getting in the way of any treatment you decide to apply.

 

Boosting in this range again helps with upper harmonics and upper harmonics are important to our brains in calculating proximity.  The closer we are to something, the more detail we can hear in the sound.  Similarly, the quieter an environment is when a sound is made, the more apparent that sound seems to us.  The upper harmonics of a sound are generally very soft and are the first things to go when we are either far away from a sound source or it is sounding in a noisy environment.  As such, the more upper frequency detail we can hear, the closer our mind perceives the sound source to be.  Furthermore, we perceive upper harmonic detail as clarity and salience.

 

Many mastering engineers, as a final polish job, will use a very hi-fi shelving EQ and boost the frequencies from roughly 16 or 18 KHz up about 3 or 4 dB.  The difference can be quite astonishing.

 

This range also gives you airiness and some pleasant room sounds.  But boosting this range can also have negative effects like hissing, piercing, and sibilance.

 

Sibilance, which is an overemphasis on frequencies ranging roughly from 6-8 KHz, is by far the most apparent and troublesome.  The beast way to deal with this is with a de-esser rather than an EQ so as not to sacrifice the harmonic content that you like that isn’t abrasive.  A de-esser is a frequency-dependent compressor, it only compresses a narrow bandwidth, usually between 4 and 9 KHz to tame sibilance.  It can be adjusted to work on cymbals or even hiss if it has the proper variables.

 

 

Next week, I’ll examine some of the commonly used terms associated with high frequency content and that will wrap up my series on the Audible Spectrum.  I hope you’ve enjoyed it so far!

Common terms in the Upper-Mid range, part 12

Posted by FixYourMix On July - 16 - 2009COMMENT ON THIS POST

Bright:  Emphasis on high-frequencies, specifically upper-mids with emphasis on harmonics.

 

Crunchy:  Exists between 2K and 4K, typically distortion based and generally pleasant.  Can lend rhythmic distinction to distorted rhythm parts.

 

Detailed:  Minutiae of the music are easily audible.  Present sounding, intimate and close with lots of articulation and transient response throughout the upper mid range.

 

Forward:  Present, in your face.  Detail present in transients and upper harmonics which lends a feeling of proximity.

 

Glassy:  Brittle sounding, too much upper-mid content especially with regard to harmonics in relation to fundamentals. 

 

Grungy:  Lots of distortion with emphasis on odd harmonics.

 

Hard:  Excellent transient response combined with an overemphasis on upper-midrange frequencies.

 

Harsh:  Peaking in the 2-6 KHz range.

 

Metallic:  Emphasis on upper-mid range frequencies, specifically those that deal with odd order harmonics in this range.

 

Pinched:  Narrow-bandwidth, often relegated to the upper-mid range frequencies.  Try boosting lower frequencies to balance.

Consisting of frequencies from 1.2 to 3.5 KHz, we once again have a nice range of frequencies to play with.  The high-mids contain lots of harmonics, especially the lower to mid order harmonics for the mid-range instruments.  The range of  2300 Hz gives you plenty of room to work with in carving out specific places for various instruments to sit.  Only the highest instruments can really play in this range:  the piccolo can sound in this band and it also accounts for the top octave or so of the piano, which most people know not a whole lot happens there.  So there are no new instruments sounding fundamentals to get in the way and cover up the harmonic content that you are treating in this band.

 

Additionally, this range is very important because it contains much of the sudden transient content.  Attack transients, sibilance, consonants and more all live in this register, so it is very important for understandability as well as punchiness, presence, and dynamics. 

 

The human voice is the most dynamic instrument on the planet.  When I work on TV and movies, I’m always amazed at how suddenly the voice changes dynamics.  Looking at the waveforms, the T, C, D, B and other consonant sounds are so short and quiet while the vowel sounds are exponentially louder and longer.  This can be a problem when mixing music because you might miss out on an initial or ultimate consonant sound that totally changes the meaning of a song without that consonant.  I remember working on a Christian rock album and the line was “We know that we can’t live with out you.”  When the mixing was done, they loved the track, but the “t” in “can’t” had disappeared.  Of course in a religious context you don’t want to be saying that you know you can live without God, so we had to spend a little extra time making sure that came across without being overbearing.

 

Just a crazy little factoid, almost all consonant sounds sound the same no matter who says them.  The majority of the time, you can fly in a t from one person, paste it in, and no one would know the difference.  You really only know from the vowel sounds what somebody sounds like.  Consonants are just air pushing against your lips, teeth, tongue, and mouth and we are all roughly equal to each other in body composition—at least enough that it isn’t imminently audible in most instances.

 

This is also the range where attack sounds live:  picks strumming strings, sticks striking cymbals, this is the range where you can hear all that.  Giving a boost to those sounds in this range can lead to a more present sound.  After all, your mind thinks your closer to something the more detail you can hear of it.  So if you can hear a stick tapping a drumhead, by god you must be close to it.  We’ll talk more about psychoacoustics and proximity in the next article.

 

Also, many big time producers believe that in this band lies the frequency that makes digital sound abrasive and therefore worse than tape, which centers around 2 KHz.  While this may or may not be the case, it certainly cannot be argued that harshness, edginess, and abrasiveness live in this frequency band.  Raucous and in your face sounds like screeching guitars and sailing synths need this range so that they can cut and make your ear drums bleed.

 

Next week, I’ll look at some common terms for upper-mid range problems and some common solutions

Common Mid-Range Terms, part 10

Posted by FixYourMix On July - 2 - 2009COMMENT ON THIS POST

Honky:  When you cup your hands and sing into it, that is pretty much what honkiness is.  This is a frequency buildup around 500-700 Hz, so cut in that area or boost the lows.

 

Nasal:  Like when you pinch your nose and speak.  This is very similar to honky except that it is a bit higher around 800-1000 Hz.

 

Radio-Filter:  The most overused pop cliché out there.  I wish everyone would stop doing this, but to do it properly you should know:  Old radios had small speakers which meant poor bass response and sometimes weak highs as well.  They also had poor construction which means limited dynamic range.  So use high and low pass filters centered around 1 KHz.  Most of the effect will be accomplished by the high-pass filter.  The low pass filter can be adjusted to taste.  Compress heavily to limit dynamic range.

 

Tinny:  Sounds like it’s coming through a tin can.  To me this also indicates peaky mids which would be a significant bump at around 1 KHz.  Perhaps some high-end reverberations of a metallic variety.  Can be remedied by boosting lows.

Earlier I defined the mid frequencies as the ones between 600 and 1200 Hz.  These would contain higher portions of the harmonies, higher melodies, and a whole bunch of harmonics.

 

For most of music history, solo singers who could sing very high were coveted.  Coloratura sopranos and castrato singers were great assets because their voices could soar audibly above the rest of the orchestration.  Their vocals pierce because the sit above the normal range for the rest of the instruments.  This frequency bandwidth aligns pretty well with the upper reaches of the soprano voice and the high-flying notes of 80s lead guitar.

 

Now accompaniment instruments such as guitar and piano might also play in this register in band situations, however in this range the emphasis tends to be on notes other than the root or melody.  This allows the soloist or lead instrument to have the spotlight in this frequency band.

 

This is also the register where the frequency range starts to increase.  Previously, we were dealing with relatively small increments between notes and registers, but here we have a gamut of 600 Hz as opposed to the low-mids which were only 300 Hz.  This allows much more room to play with sonics using EQs, harmonic exciters, and other effects which is great because this section houses most of the lower order harmonics other than the fundamental.

 

As mentioned in the primer, harmonics help us distinguish one instrument from another.  Even harmonics give a warmer, organic, and natural sound while odd harmonics impart a more harsh and metallic sound.  Smooth guitars through tube amplifiers have rich even harmonics while harsh distorted heavy metal guitars have more odd harmonic content.  Brass instruments have more of an emphasis on odd harmonics while strings have more even harmonics.

 

So now going back to bass instruments like the kick and bass guitar, another good way to distinguish them from each other is by treating their harmonics in this range differently.  This range is better for this kind of treatment because it avoids putting the changes intended for emphasis in the frequency band with a lot of build-up like the low mids.  This range contains mostly harmonics and solo instruments, so there isn’t a lot to get in the way of hearing these subtle alterations and they are still low enough to be significant to the fundamental sound.

 

So if we have a bass guitar playing mostly root notes down in the key of A, we’d know that the bass is playing notes in the frequency range of 55-110 Hz.  This would mean second harmonics from 110 to 220 and third harmonics from 220 to 440.  These are great to try and treat, especially if you are dealing with sparse mixes, but they aren’t really helpful in densely orchestrated tunes because other instruments will be taking up those frequency bands.  The next harmonics would be from 440 to 880.  These harmonics are in this frequency range, so a nice wide EQ centered at 660 and subtly boosting around might give the bass the audibility you need and it would be nice and smooth since it emphasizes the even harmonic (4th).  You could also try and emphasize the next batch of harmonics which would be 880 to 1760.  This would put the center right at 1.2 KHz, right at the top of our range.  And this would impart a more harsh and aggressive tone.

 

You might de-emphasize those frequencies in the kick drum or even choose to emphasize frequencies that fall on the outskirts of bass guitar’s harmonic ranges.  If you find that the bass rarely sounds harmonics in the 900 range, it’d be a perfect place to emphasize the kick drum and maybe carve out the bass there.  All you need is a little spot in the mix for your ear to key on and you’ve got audibility.

 

Now the bass guitar I’ve used in my example is 1-2 octaves below the other non-bass instruments I’ve talked about in my hypothetical mix.  That means that treating these instruments in the mid-range will be emphasizing lower order harmonics and can really alter the instruments’ sounds.  But you do have a nice wide range to work with, so treating each instrument individually with a different portion of the bandwidth for emphasis can help benefit audibility.  Plus, this is where many of the sounds intrinsic to specific instruments exist, so emphasize the frequency band that makes a trumpet really sound like a trumpet can help to keep it audible but prevent it from overtaking the lead vocal.

 

One thing I want to emphasize here is that most instruments play a range of notes, not just one note like a kick drum.  In the bass guitar example, you saw how wide a range a bass guitar’s frequency content can have from just playing in one octave.  I didn’t give any specifics about the tune other than the key, we don’t know how often it plays what note or which note, we just know the key.  Many experts and magazines will like to give you helpful frequencies to try when mixing.  Bear in mind that these are only guidelines and could not possibly be a one stop fix for all mixing needs.  If somebody tells you to cut 450 in every instance to make a mix better, it would really be a shame for songs in the key of A whose mid-range instruments would be getting de-emphasized when they play the root note…

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