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How to get a great tonal balance in your mix

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How to get a great tonal balance in your mix

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One of the most challenging aspects of music production is mixing a track to sound well balanced across the frequency spectrum.

Having your high-frequencies too loud in the mix will make your track sound harsh. Too much energy in the mid and low-frequencies will make your track sound muddy. Having too little energy in these frequency ranges is equally as problematic.

This article will give you the information you need to get your mixes sounding as well balanced as your favourite mixes.

The Challenge & The Solution

Nailing the tonal balance in a mix is very tricky when you don't have acoustic treatment and monitors with a great low-end response. This is why so many home-studio enthusiasts struggle.

To solve this problem and help more people get their music sounding as good as their favourite tracks, we created a plugin called REFERENCE.

REFERENCE shows you visually what you would be able to hear if you were in a world-class studio. This isn't a simple 'slowed down' frequency analyzer like many other plugins out there that claim to do the same thing as REFERENCE. It's a complex algorithm created to specifically identify how the human ear perceives certain frequencies relative to the balance of the whole mix. With this information, we can make informed decisions regarding the tonal balance of a mix.

REFERENCE has a unique (and incredibly effective) way of showing you the perceived level in various frequency bands. The white level-line will drift into the lower half if those frequencies have less perceived volume than in your selected reference track. It will drift into the upper half if those frequencies have more perceived volume than in your selected reference track.

Fire up REFERENCE ( click here to download the free trial ) as the final insert on your master channel (unless you use any speaker/headphone correction software in which case REFERENCE should go before that) and drop in your reference tracks.

Loop the chorus of each of your reference tracks, as well as the chorus of your own production from within your DAW. Hit the 'level-match all tracks' button (top right corner of the waveform in REFERENCE) to make all the tracks play back at the same perceived volume, you're now set up and ready to nail your low-end.

In the example above, the 'low-end' white level-line in REFERENCE has drifted into the upper half. This tells us we should reduce the low-frequencies for a tonal balance that better matches our reference track.

Take your favourite EQ and use a low shelf to match the tonal balance to your reference track.

Balancing the mid-frequencies

The mid-frequencies range from around 250Hz to 3kHz. There are various sonic characteristics in this range; mud, warmth, thickness & harmonics to name a few.

Cheatsheet credit: Never Get Stuck Again - A Complete Guide To Turning Your Musical Idea Into A Finished Track.

If REFERENCE is encouraging you to boost frequencies in the mid-range, try to go for a broader 'Q' (the 'Q' is the bandwidth of the EQ). This is a more musical approach as a thin Q boost could give you a nasty 'boxy' resonance that would make the audio sound nasally.

If REFERENCE is encouraging you to cut frequencies in the mid-range, try to go for a thinner 'Q'. This surgical approach can help you cut the frequencies you don't want in your mix without dramatically altering the sound.

Balancing the high-frequencies

The high-end has the broadest frequency range of about 3kHz to 20kHz. I like to approach the high end slightly differently when I'm fine tuning the tonal balance.

Boosting the high-frequencies

Using an upward expander to boost high-frequencies is a musical approach that can increase the punch whilst minimising harshness.

Upwards expansion increases the volume of signals over the threshold, giving your audio more dynamic range in a transparent way.

In this example, I'll be using our plugin to dial in the expansion. The Expand module in ANIMATE can bring out the glistening top end of any channel that needs a high-frequency boost.

1. Set the filter to only react to the highest frequencies of the sound. You'll see the input audio glowing behind the filter, you can use this as a guide to select the relevant frequencies.

2. Set the ratio to around 1:1.5.

3. Change the threshold so the audio you want to effect surpasses the threshold.

4. Now increase the amount until the audio feels like the top end is glistening.

5. This effect will increase the perceived volume of the channel, so you'll want to reduce the output level. The output slider has an arrow to show you the perceived loudness of the audio before you added the effect. Click on the level match pointer and toggle the bypass button to make sure your mix decisions are improving your mix, not just making it louder.

To make a mix shine, you just need a few elements to occupy the highest frequencies. Too many channels with strong high frequencies will sound harsh and will give you and your audience ear fatigue.

Controlling the high-frequencies

Simply cutting the high frequencies will affect the tonal balance of your whole mix. That might be necessary, but more often than not you only want to reduce the louder high-frequencies. For this task, a dynamic EQ is perfect. You can set the band of the high frequencies you want to reduce, then set the threshold so the reduction only happens when those high-frequencies are too loud in the mix.

This is great as your cut doesn't make the quieter parts of your track sound less bright. You can use REFERENCE again to help you set this EQ to the ideal setting.

Getting the perfect tonal balance is not easy and takes a lot of practice. With these approaches and using , you'll speed up your progress and get closer than ever to the sound of your favourite tracks.

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How to get a great tonal balance in your mix

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One of the most challenging aspects of music production is mixing a track to sound well balanced across the frequency spectrum. Having your high-frequencies too loud in the mix will make your track sound harsh. Too much energy in the mid and low-frequencies will make your track sound muddy.

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© 2016 Steve Feinstein, InMusic Inc.

The Akai Professional RPM800. The Alesis Elevate 6. The Alto Professional TS215. The Denon DJ HP800. The Denon Professional DN-300C. The ION Plunge. The Marantz Professional MPM-2000. The M-Audio AV32.1. The Numark Red Wave.

All of our hardware brands have multiple units that specify their frequency response. The concepts of frequency response, frequency range, audible frequencies, how frequency relates to musical notes, etc. are absolutely central-critical-to understanding how audio equipment makes, reproduces or records sound and how that sound is related to the live sound produced by singers and instrumentalists.

The goal of this paper isn't to turn anyone into an engineer. Rather, the goal of this paper is to give people a working knowledge of the basic concepts of sound, how it's measured and the common terms and phrases we use to describe it.

That's very good info to have, because it gives you a solid overall view of how and why all our stuff fits together.

Frequency Response is Important

There's just no getting around this. If you're going to understand anything about audio-how speakers and headphones work, the effect of recording studio acoustics, how position your mics, the concept of bass management/low-pass filters, anything about audio-you have to understand frequency response. This is the cornerstone of everything in audio. It all starts and stops with frequency response. Amplifiers, headphones, speakers/monitors, mics, cartridges on turntables, CD and DVD players, they all have to have as close to perfect frequency response as possible, or else they just won't sound right. Nothing else matters if the frequency response is no good. Can't make it any clearer than that.

Let's start of with the basics: Sound waves in the air that we perceive as music or speech or noise are measured in cycles per second, usually abbreviated as Hertz or Hz (in honor of the late 19th-century German physicist Heinrich Hertz, who did important pioneering work in the field of electromagnetic wave theory).

Humans can hear from approximately 20Hz in the bass to about 20,000Hz (20kHz) in the treble. (Well, women and young children can hear that high; a middle-aged man is lucky to make it much past 13-14kHz.
Your grandfather needs his hearing aid because his ears are toast by 3kHz. That's why he's always saying, "Huh?") Your pooch gets to 40k, easily, which is why 'dog whistles' are inaudible to humans. Bats fly around at night with unerring precision by using their super hearing as organic 'sonar' (they judge their location/position by analyzing reflected sound waves around them). Bats can hear as high as 80kHz!

The relationship between audible frequencies and musical octaves is 2:1 (or 1:2). So if humans can hear from "20 to 20" as the saying goes, that's a span of 10 audible octaves: 20-40Hz, 40-80, 80-160, 160-320, 320-625 (yeah, yeah, we fudged that one a little, to make the numbers that follow look nice), 625-1250, 1250-2500, 2500-5000, 5000-10k, 10k-20k. For musical reference, Middle "C" is around 262Hz. An octave higher is 524Hz; an octave lower is 131Hz.

Why Frequency Response is important:

An audio device with good frequency response is able to play all the low, middle, and high tones correctly-and in the proper proportion to each other-and that's what tells our ears whether or not this is a high-fidelity unit with rich, vibrant sound.

To help you understand frequency response, remember this: the loudness of sound is expressed in a unit of measure called the decibel, or dB (the "B" is capitalized in recognition of Alexander Graham Bell and his work in acoustics. No kidding.). Then we say how by many decibels (dB) the equipment varies from perfect, over a specified range of frequencies. 1dB is generally thought to be the smallest change in sound pressure level (SPL) that is consistently recognizable by someone. 3dB is a definite, but still relatively subtle, change in loudness. The dB scale is logarithmic (not linear), so 10dB louder is "twice as loud."

So for example, a speaker or headphone may be said to have a frequency response of 40Hz-20kHz
(that's the range), ± 3dB (that's the variation). If a manufacturer specifies a frequency response without a ± xdB variation, it should set off a "Danger, Will Robinson!" alarm. No specified dB variation = no value whatsoever. Stating "20-20kHz" by itself means absolutely nothing, and it undermines the believability and credibility of anything else the manufacturer says.

Frequency Response: 20-20kHz ±3dB = Good. Nicely done. Meaningful information.

Frequency Response: 20-20kHz = bogus, filler, something to hide, trying to pull a fast one.

Now it's time for another hard part:

Frequency responses are almost always shown as a graph. This graph is known as the "Frequency Response Graph."(Clever, no?) You have to know how to read a graph. No excuses. If you paid attention in Mr. Kelleher's 6th grade arithmetic class, great. If not, you'll be sorry now.

Look at Figure 1. This chart tells you what frequencies roughly correspond to bass, midrange, and treble sounds.

Now look at Figure 2. The black line is a speaker or headphone with excellent frequency response. The frequency response curve (so-called because a speaker's or headphone's frequency response will curve, or roll off, in the low bass and high treble) is pretty flat ("flat" is good, because it means the device is accurate), with no serious peaks, dips or other up-and-down variations. For speakers, headphones and microphones, ± 2 or 3dB is considered very good. Amplifiers, CD/DVD players and other 'strictly electronic' devices should be within ± 0.5 or 1dB at the most.

By contrast, the red line in Figure 2 shows a speaker's or headphone's frequency response with a big 7dB peak (so-called because the graph looks like a mountain's peak) in the upper midrange around 6 kHz, which will make it sound harsh and irritating.

Figure 3 shows what response curves look like that correspond to various commonly-used subjective descriptions.

For those of you who already know all this stuff, what this paper does is give you an easy way to present it and teach it to others who don't know it.

For those of you who don't really know this yet, if you learn this basic info on frequency response, the relationship between frequency and musical notes and octaves, how people hear, and how these various audio terms look as frequency response graphs, you'll be well on your way to a meaningful understanding of our audio equipment.

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Understanding Frequency Response - Why it Matters

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© 2016 Steve Feinstein, InMusic Inc. The Akai Professional RPM800. The Alesis Elevate 6. The Alto Professional TS215. The Denon DJ HP800. The Denon Professional DN-300C. The ION Plunge. The Marantz Professional MPM-2000. The M-Audio AV32.1. The Numark Red Wave. All of our hardware brands have multiple units that specify their frequency response.

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These terms sound more like medical tools/procedures used in gastrointestinal bypass surgeries!

Despite some of their more gruesome sounding names, these are all terms related to, and describing dynamics - and dynamics control in audio. Let's define a few items of interest.

Core Definitions in Dynamics

"Dynamics" in its simplest and most functional definition means this:

or:

A simpler definition is this:

Now obviously there is much more to it than that last sentence, but essentially what you are talking about is Volume, or actually... the changes in volume of a signal!

Now, one more definition we need before we move on:

Misc. note: There is A LOT more to dynamic range, but that is way beyond our scope for now. As a matter of fact, a full discussion of dynamic range is more suited for recording/mixing, and mastering discussions. If this subsequent topic is of interest to you, please let us know in the comments!

OK, I lied. One more definition. Sorry!
THIS is the important one... Remember it!
Dynamic Control

Now that have we our core definitions, all of this begs the question: How does all of this fit together?

Or, the more relevant question: Exactly how do we manipulate, regulate and control dynamics along with dynamic range?

Eventually, we will ask: Why would we want to?

First things first:

We control and manipulate dynamics with compressors.

Compressors come in a variety of flavors (as do all effects): from the budget-minded stomp box (pedal), to studio spec. rack mount units, all the way to the multi-thousand dollar high-end units used by mastering engineers.

ALL of these compressors do this:
They raise the softest part of a signal, and lower the loudest part of that signal.
The result is a compromise that lets both softest and loudest parts be clearly audible. As a tangential effect, reducing the dynamic range to a narrower and overall consistent field allows the whole signal to be raised in overall volume. This is especially beneficial to mastering engineers, as this is one of the ways that they make mixes sound louder.

There, in the above paragraph, also lies the "why". With dynamics processing we can regulate individual or many instruments, and "herd" them into a specified range for the sole purpose of being able to hear the softest and loudest parts equally, as well as have all the sounds blend well together. In simple terms, we want our collection of sounds or instruments to play nice together!

That's it in a nutshell. That is the basic function of a compressor reduced to its lowest common denominator. It is simply a tool to manipulate dynamics and dynamic range.

Compression ear training

Here's a little taste - listen to the original funk riff, and then the compressed version:

Can you hear how the dynamic range has been reduced? The quiet parts are now louder, and the louder parts quieter, giving a more consistent sound throughout.

Misc. note: Some compressors impart a tonal color of their own, while other compressors are totally transparent... as in, all you hear is the compression with no added tonal altering. Pedal compressors used by guitar players usually color the original tone, and that can be a desirable trait in the overall tone creation. However, with studio compressors or especially mastering studio compressors, tonal alteration is viewed as a negative side effect to be avoided at all costs. And I do mean that fully literal as well, since transparent compressors that only alter dynamics are usually above the $1000 range! Of course, which compressor is right for you is entirely subjective, and dependent largely on your tonal needs for the job at hand.

Uses of Compressors

Here are a few basic uses of compressors:

Drums: Adding punch, or creating entirely new sounds.

Guitar: Adding sustain. Balancing out the strings so they are all equally audible.

Bass guitar: Same as guitar, plus enabling that slap/funk sound.

Vocals: Gently bring up the softest whispers and tame the peaks.

Full mixes: Get all the instruments to blend well.

Mastering: This is a whole separate topic on its own!

We'll see examples of some of these in the compression ear training examples which follow.

Compressor Controls

Moving on, of course there are different parameters that allow control over certain aspects of your compressed output. Here are the most common ones. Actually, let me restate that... the most important ones!

Here are the basics of each:

Threshold

The Threshold is what controls the actual point when the compressor clamps down on your signal and works its magic. The lower the threshold, the more signal gets compressed. The higher the threshold, the less signal gets compressed (since the high setting allows only the top part of a signal to get compressed).

Different threshold points are required for different instruments. For example, slap bass requires heavy compression, so naturally you want the entire signal processed, and hence a low threshold; whereas something like vocals that require very gentle manipulation would utilize a high threshold to maintain a natural, unprocessed sound.

A clean guitar would be a candidate for various thresholds. High threshold for just a bit of compression, and low for full processing and that "everything sounds equally loud" funk tone! By the way, because of the squeezing of soft and high sounds that a compressor does by default, instrument sustain is increased. That is why guitar and bass players love them!

There are many variants in between, but I think you get the idea.

Time to hear it in action! Please realize that the settings used to "demo" the various functions are set to VERY drastic, and to be quite honest, "obnoxious" levels... BUT this is done strictly to make the differences blatantly audible. In real life things would be set much more subtle; with attention to musicality and finesse!

With that said... Enjoy!

Threshold Examples

Light clean picking, same lick x3. Threshold set to 0, -25dB, and then -45dB.

Note how the sound gets "splattier" as the threshold setting is reduced, allowing for more compression.

Light clean picking with chord, same lick x3. Threshold set to 0, -25dB, and then -45dB. In addition to the "splatty" single notes, see how the chord at the end seems "suppressed" with reduced threshold settings, as if it is being choked.

Note: When the threshold is reduced, more compression takes place and therefore the overall volume level gets reduced. It is common practice to use a parameter called "make-up gain" to replace the volume that has been lost. In clips 1 and 2 this was used.

Here in clip 3 you will hear the EXACT same threshold settings WITHOUT make up gain.

Light clean chord, same lick x3. Threshold set to 0, -25dB, -45dB respectively. Make-up gain OFF.

Yeah, I know... Compression without make up gain is pretty useless!

Ratio

The Ratio parameter works in concert with threshold, and controls the proportion of the output that gets compressed compared to the input signal. It is the input/output ratio used when signals exceed the threshold.

Common ratio levels are 1:1, 2:1, 4:1, 6:1, etc. Here is how the math works: If you set a ratio of 2:1, that means that if the input signal exceeds the threshold by 2 decibels, then after compression, the output will only be 1 decibel above that same threshold. 4:1, (a guitar favorite) means that when the input signal exceeds the threshold point by 4 decibels, then the output is still only 1 decibel above that same threshold. The difference between the input and output is called gain reduction. In the 4:1 example the gain reduction is a total of 3 decibels. On a ratio of 6:1, it would be a gain reduction of 5 decibels.

Did you notice that a 1:1 setting will have no effect? 1dB change on input goes to 1dB change on output! And no gain reduction.

There is much more to say, but for now this will give you the idea that the threshold and ratio are the controls that govern just how much processing you want.

Ratio Examples

Now, let's have a little fun pushing the limits of good taste! Naturally it will be easier to hear the effect with the higher ratios. Once you've tuned into these, it should be easier to hear the more subtle lower ratios.

Two chords x5. Ratio settings: 1:1 (i.e. no effect), 3:1, 4:1, 6:1, 10:1.

Note how the higher the ratio, the more the sustain at the end of the last chord swells and "howls" more. Also note how the clip starts to "waver" and distort!

Chord comp x5. Ratio settings: 1:1 (i.e. no effect), 3:1, 4:1, 6:1, 10:1.

Note how the effects of clip 4 are even more prominent with this clip which contains more sonic information. Also notice how the chord comp becomes more fatiguing and noisier with each ratio increase. If there is a lesson here, it is this: Compressors are high powered sonic weapons that need to be handled with care! Truly, a little goes a long way!

Attack

The Attack is simply what controls the speed in which the compressor starts to operate. Ranges are usually from a few milliseconds to a few hundred milliseconds - or even more. Here are the relevant facts:

Faster attack means almost instant compression. This means you do not get to hear the initial natural beginning (or attack) of the input signal.

Slower attack times simply means that you get to hear that natural beginning of your input signal. Great for drums, since the attack is the main component of the drum sound. This results in a very, very punchy sound with the control specified by the compressor. Classic studio trick used for loud punchy drum tracks.

Attack Examples

Now be warned: these examples will sound OBNOXIOUS!

Chick'n pickin' lick on bridge pickup x5. Attack settings: Original, .01ms, 2ms, 500ms, 1 second.

This is a peculiar control. Note how on the second time through (the .01msec setting) the signal distorted violently. Very fast attacks always do that. Remember that sonic phenomenon.

Past that, slower attack settings get more and more percussive on the transients of the phrase. How much you want is, well, up to you. But again, tread carefully!

Chick'n pickin' lick bridge and middle pickup x5. Attack settings: Original, .01ms, 2ms, 500ms, 1 second.

Just a different tone for you to hear. Also, I lowered the threshold to get more compression. Here, slower attack times work better since more signal is being squashed. The slower attack times seem to fill out the sound more, and are a little more musical. Subtle difference but it's there!

Also the real fast attack (.01ms) distortion phenomenon is still present here as well regardless of threshold settings!

Before I forget, in this clip, the 2ms setting almost distorted, producing a "wavering" sound. Hopefully you are starting to see how all these controls are very dependent on each other!

Release

The Release parameter works in conjunction with the attack control to specify how soon the compressed signal returns to an uncompressed state. Times vary from a few milliseconds all the way to a few seconds. Simply put, faster release times get the signal back to "natural home base" almost instantly, while slow release times let the signal stay compressed longer, and return to its natural uncompressed state in a slow deliberate way.

This where artifacts called "pumping", or "breathing" can come from. Too fast a release and the change between compressed and uncompressed becomes WAY too obvious. Too slow of a release, and now you're dragging the compressed signal out too long, which results in a "squashed" sound. Some compressors offer automatic attack and release to take out the guess work, but nothing beats the manual settings for specific surgical control!

Release Examples

Let's listen to the control over how quickly things get back to an uncompressed state!

Clean sustaining chord x5. Release settings: Original, 50ms, 500ms, 1 second, 5 seconds.

This particular example is subtle, but if you pay attention you can hear the decay of the chord wanting to die out but not being able to. The 50ms setting is the most musical. Even though the other settings aren't completely out of control, there are still subtle tell-tale signs of too long a release: very slight wavering, distorting, up and down pumping, etc. They are all very slight, but listen closely and you'll pick them out!

Homework

Listen to the tracks above for Threshold, Ratio, Attack and Release. Listen again! Try to tune your ear into how changing each parameter affects the sound. Read the explanations, and try to hear what's going on in the signal. The examples are going to be tougher next time, so make sure you get the basics of compression ear training down now!

There are many, many more aspects to discuss, but remember, we are laying groundwork here!

Next time we'll build on this with a look at limiters, expanders, gates, and single/multi-band compressors. These are all direct relatives or variations to compressors with similar control layouts, but their own uses and specialities. We'll also look in depth at real-world examples of compression and see how to choose parameters to get that particular sound you're going for.

Any questions? Don't be shy!

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Hearing Effects: Dynamics, Part 1 - Compressors | Musical U

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These terms sound more like medical tools/procedures used in gastrointestinal bypass surgeries! Despite some of their more gruesome sounding names, these are all terms related to, and describing dynamics - and dynamics control in audio. Let's define a few items of interest.

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Soundonsound magazine

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Welcome to Sound On Sound's indispensible, regularly updated, explanations of technical terms from the fields of Recording, Audio Production, Music Technology, MIDI, Music Software, Audio Plug-ins, Mac and PC Computing, Live Sound, Acoustics, Electronics and more...

If we do not explain a particular term below, please email glossary@soundonsound.com and we will add it to our next update.

B-Type Plug - A professional form of jack plug derived from the telecommunications industry and also known as the PO316. Widely used for balanced mic and line-level connections on professional patch bays. (cf. A-Type Plug)

Backup - A safety copy of software or other digital data. A popular saying is that unless data exists in three physically separate locations at the same time, it hasn't been backed up properly!

Back Electret - A form of electrostatic or capacitor microphone. Instead of creating an electrostatic charge within the capacitor capsule with an external DC voltage, an electret microphone employs a special dielectric material which permanently stores a static-electric charge. A PTFE film is normally used, and where this is attached to the back plate of the capsule the device is called a 'back electret'. Some very early electret microphones used the dielectric film as the diaphragm but these sounded very poor, which is why later and better designs which used the back electret configuration were specifically denoted as such. Designs which attach the PTFE film to the diaphragm are known as Front Electrets. Modern electret capsules compare directly in quality with traditional DC-biased capacitor capsules, and are available in the same range of configurations - large, medium and small diaphragm sizes, single and dual membrane, fixed or multi-pattern, and so on.

Balance - This word has several meanings in recording. It may refer to the relative levels of the left and right channels of a stereo recording (eg. Balance Control), or it may be used to describe the relative levels of the various instruments and voices within a mix (ie. Mix balance).

Balanced Wiring - Where protection from electromagnetic interference and freedom from earth references are required, a balanced interface is used. The term 'balanced' refers to identical (balanced) impedances to ground from each of two signal carrying conductors which are enclosed, again, within an all-embracing overall screen. This screen is grounded (to catch and remove unwanted RFI), but plays no part in passing the audio signal or providing its voltage reference. Instead, the two signal wires provide the reference voltage for each other - the signal is conveyed 'differentially' and the receiver detects the voltage difference between the two signal wires. Any interference instils the same voltage on each wire (common mode) because the impedance to ground is identical for each, and as there is therefore no voltage difference between the signal wires, the interference is ignored completely by the receiver.
Signals conveyed over the balanced interface may appear as equal half-level voltages with opposite polarities on each signal wire - the most commonly described technique. However, modern systems are increasingly using a single-sided approach where one wire carries the entire signal voltage and the other a ground reference for it. Some advantages of this technique include less complicated balanced driver stages, and connection to an unbalanced destination still provides the correct signal level, yet the interference rejection properties are unaffected. Effective interference rejection requires both the sending and receiving devices to have balanced output and input stages respectively.

Band-pass Filter (BPF) - A filter that removes or attenuates frequencies above and below the centre frequency at which it is set, and only passes a specific range of frequencies. Band-pass filters are often used in synthesizers as tone shaping elements.

Bandwidth - The range of frequencies passed by an electronic circuit such as an amplifier, mixer or filter. The frequency range is usually measured at the points where the level drops by 3dB relative to the maximum. (See also Q)

Bank - A specific configuration of sounds or other parameters stored in memory and accessed manually or via MIDI commands.

Bass Response - The frequency response of a loudspeaker system at the lower end of the spectrum. The physical size and design of a loudspeaker cabinet and the bass driver (woofer) determine the low frequency extension (the lowest frequency the speaker can reproduce at normal level) and the how quickly the signal level falls below that frequency.

Bass Tip-up - see Proximity Effect.

Bass Trap - A special type of acoustic absorber which is optimised to absorb low frequency sound waves.

Bantam Plug - Also known as TT or Tiny Telephone Plugs. A professional form of miniature jack plug derived from the telecommunications industry and widely used for balanced mic and line-level connections on professional patch bays. (cf. B-Type Plug)

Beta Version - Software which is not fully tested and may include bugs.

Bias - A high-frequency signal used in analogue recording to improve the accuracy of the recorded signal and to drive the erase head. Bias is generated by a bias oscillator.

Binary - A counting system based on only two states: 1s and 0s. It is ideal for electronic equipment where it can be represented as high and low voltages, light on/off, N-S or S-N magnetic domains, etc.

BIOS - Part of a computer operating system (basic input-output system) held on ROM rather than on disk. This handles basic routines such as accessing the disk drive.

Bit - A contraction of Binary digit, which may either be 1 or 0.

Bit Rate (see also Sample Rate) - The number of data bits replayed or transferred in a given period of time (normally one second). Normally expressed in terms of kb/s (kilo bits per second) or Mb/s (mega bits per second). For example, the bit rate of a standard CD is (2 channels x 16 bits per sample x 44.1 thousand samples per second) = 1411.2 kilobits/second. Popular MP3 file format bit rates range from 128kb/s to 320kb/s, while the Dolby Digital 5.1 surround soundtrack on a DVD-Video typically ranges between 384 and 448kb/s.

Bi-Timbral - A synthesizer than can generate two different sounds simultaneously (see multi-timbral).

Blumlein Array - A stereo coincident microphone technique devices by Alan Blumlein in the early 1930s, employing a pair of microphones with figure-eight polar patterns, mounted at 90 degrees to each other with the two diaphragms vertically aligned.

BNC - A type of bayonet-locking, two-terminal connector used for professional video and digital audio connections. (See AES3-id)

Boom - A mechanical means of supporting a microphone above a sound source. Many microphone stands are supplied with a 'boom arm' affixed to the top of the stand's main vertical mast. The term may also be applied to larger, remotely controlled microphone supports used in film and TV studios, or even to the handheld 'fishpoles' used by film and TV sound recordists.

Boost/Cut Control - A single gain control which allows the range of frequencies passing through a filter to be either amplified or attenuated. The centre position is usually the 'flat' or 'no effect' position.

Booth - See Isolation Room

Bouncing - The process of mixing two or more recorded tracks together and re-recording these onto another track.

Boundary - A physical obstruction to sound waves, such as a wall, or a large solid object. When sound waves reach a boundary they create a high pressure area at the surface.

Boundary Layer Microphone - A specialised microphone where the diaphragm is placed very close to a boundary (eg. wall, floor or ceiling). In this position the direct and reflected sound adds constructively, giving a 6dB increase in sensitivity. It also avoids the comb-filtering that can occur when a conventionally placed microphone captures the direct sound along with strong first reflections from nearby boundaries. Also known as PZM or Pressure Zone Microphone.

BPM - Beats Per Minute.

Breath Controller - A device that converts breath pressure into MIDI controller data.
Buffer - An electronic circuit designed to isolate the output of a source device from loading effects due to the input impedance of destination devices.

Buffer Memory - A buffer is essentially a short term data storage facility used to accommodate variable data read or write periods, temporarily storing data in sequence until it can be processed or transferred by or to some other part of the system.

Bug - Slang term for a software fault or equipment design problem.

Bus - (Also sometimes referred to as a buss) An electrical signal path along which multiple signals may travel. A typical audio mixer contains several (mix) busses which carry the stereo mix, subgroups, the PFL signal, the aux sends, and so on. Power supplies are also fed along busses.

Byte - A collection of digital data comprising eight bits.

IC - An abbreviation of Integrated Circuit, a collection of miniaturised transistors and other components on a single silicon wafer, designed to perform a specific function.

IEM - In-Ear Monitor. A wirelessly-connected foldback monitoring system, often used by musicians on stage with in-ear earpieces.

Impedance - The 'resistance' or opposition of a medium to a change of state, often encountered in the context of electrical connections (and the way signals of different frequencies are treated), or acoustic treatment (denoting the resistance it presents to air flow). Although measured in Ohms, the impedance of a 'reactive' device such as a loudspeaker drive unit will usually vary with signal frequency and will be higher than the resistance when measured with a static DC voltage. Signal sources have an output impedance and destinations have an input impedance. In analogue audio systems the usually arrangement is to source from a very low impedance and feed a destination of a much higher (typically 10 times) impedance. This is called a 'voltage matching' interface. In digital and video systems it is more normal to find 'matched impedance' interfacing where the source, destination and cable all have the same impedance (eg. 75 Ohms in the case of S/PDIF).
Microphones have a very low impedance (150 Ohms or so) while microphone preamps provide an input impedance of 1,500 Ohms or more. Line inputs typically have an impedance of 10,000 Ohms and DI boxes may provide an input impedance of as much as 1,000,000 Ohms to suit the relatively high output impedance of typical guitar pickups.

Impulse Response - An impulse respsonse is the time-domain equivalent of the much more familiar frequency (and phase) responses in the frequency-domain. A very brief click (technically, a Dirac delta function) which theoretically contains all frequencies at equal amplitude, is passed through the device under test. The resulting output is the 'impulse response' of that device and uniquely describes its signal processing behaviour. Impulse responses are very convenient for digital signal processing applications as the source impulse is very similar to a single digital sample value.

Inductor - A reactive component that presents an increasing impedance with frequency. (Also see Capacitor.)

Initialise - Resetting a device to its 'start-up' state. Sometimes used to mean restoring a piece of equipment to its factory default settings.

Insert Points - The provision on a mixing console or 'channel strip' processor of a facility to break into the signal path through the unit to insert an external processor. Budget devices generally use a single connection (usually a TRS socket) with unbalanced send and return signals on separate contacts, requiring a splitter or Y-cable to provide separate send (input to the external device) and return (output from external device) connections . High end units tend to provide separate balanced send and return connections. (cf. Effects Loop)

Input Impedance - The input impedance of an electrical network is the 'load' into which a power source delivers energy. In modern audio systems the input impedance is normally about ten times higher than the source impedance - so a typical microphone preamp has an input impedance of between 1500 and 2500 Ohms, and a line input is usually between 10 and 50k Ohms.

Insulator - A material that does not conduct electricity. (Also see conductor)

Instrument Level - The nominal signal level generated by an electric instrument like a guitar, bass guitar or keyboard. Typically around -25dBu. Instrument signals must be amplified to raise them to line-level.

Interface - A device that acts as an intermediary to two or more other pieces of equipment. For example, a MIDI interface enables a computer to communicate with MIDI instruments and keyboards.

Intermittent - Something that happens occasionally and unpredictably, typically a fault condition.

Intermodulation Distortion - A form of non-linear distortion that introduces frequencies not present in and musically unrelated to the original signal. These are invariably based on the sum and difference products of the original frequencies.

I/O - The input/output connections of a system.

IPS - Inches Per Second. Used to describe tape speed. Also the Institute of Professional Sound (www.ips.org.uk)

IRQ - Interrupt Request. Part of the operating system of a computer that allows a connected device to request attention from the processor in order to transfer data to it or from it.

Isolation Room - A separate room or enclosure designed to provide acoustic isolation from external noise. Often used alongside a studio's main live room to record vocals or drums, for example, without spill from other instruments.

Isolator (also decoupler) - A device intended to prevent the transmission of physical vibrations over a specific frequency range, such as a rubber or foam block. The term can also be applied to audio isolation transformers, used to provide galvanic isolation between the source and destination, thus avoiding ground loops.

Isopropyl Alcohol - A type of alcohol commonly used for cleaning and de-greasing tape machine heads and guides.

Jackfield - A system of panel-mounted connectors used to bring inputs and outputs to a central point from where they can be routed using plug-in patch cords. Also called a patchbay.

Jack Plug - A commonly used audio connector, usually ¼ inch in diameter and with either two terminals (tip and sleeve known as TS) or three (tip, ring, sleeve called TRS). The TS version can only carry unbalanced mono signals, and is often used for electric instruments (guitars, keyboards, etc). The TRS version is used for unbalanced stereo signals (eg for headphones) or balanced mono signals.

Jargon - Specialised words associated with a specialist subject.

k - (lower-case k) The standard abbreviation for kilo, meaning a multiplier of 1000 (one thousand). Used as a prefix to other values to indicate magnitude, eg. 1kHz = 1000Hz, 1kOhm = 1000 Ohms.

K-Metering - An audio level metering format developed by mastering engineer Bob Katz which must be used with a monitoring system set up to a calibrated acoustic reference level. Three VU-like meter scales are provided, differing only in the displayed headroom margin. The K-20 scale is used for source recording and wide dynamic-range mixing/mastering, and affords a 20dB headroom margin. The K-14 scale allows 14dB of headroom and is intended for most pop music mixing/mastering, while the K-12 scale is intended for material with a more heavily restricted dynamic-range, such as for broadcasting. In all cases, the meter's zero mark is aligned with the acoustic reference level.

K-Weighting - A form of electrical filter which is designed to mimic the relative sensitivity of the human ear to different frequencies in terms of pereceived loudness. It is broadly similar to the A-Weighting curve, except that it adds a shelf boost above 2kHz. This filter is an integral element of the ITU-R BS.1770 loudness measurement protocol. (See also A-Weighting and C-Weighting)

Q - The 'quality-factor' of a filter which defines its bandwidth and indicates a filter's resonant properties. The higher the Q, the more resonant the filter and the narrower the range of frequencies that are allowed to pass.

Quantisation - Part of the process of digitising an analogue signal. Quantisation is the process of describing or measuring the amplitude of the analogue signal captured in each sample, and is defined by the wordlength used to describe the audio signal - eg. 16 bits.

Quantiser - A means of moving notes recorded in a MIDI sequencer so that they line up with user defined subdivisions of a musical bar, for example, 16s. The facility may be used to correct timing errors, but over-quantization can remove the human feel from a performance.

Unbalanced - A 2-wire electrical signal connection where the signal conductor is surrounded by a screen which provides a 0V reference and also guards against electrical interference.

Unison - To play the same melody using two or more different instruments or voices.

Unity Gain - A condition where the output signal is the same amplitude as the input signal; the overall system gain is then x1 or unity.

USB - Universal Serial Bus. A computer interface standard introduced in 1996 to replace the previous standard serial and parallel ports more commonly used. The USB1.1 interface operated at up to 12Mb/s, but this was superseded in 2000 by USB2.0 which operates at up to 480Mb/s. Most USB interfaces can also provide a 5V power supply to connected devices. USB3.0 was launched in 2008 and is claimed to operate at rates up to 5Gb/s, but it is only now (2011) starting to appear on hardware. USB conncetors can be indentified by having a blue insert in the Type-A sockets and the letters SS (SuperSpeed). The latest edition, USB3.1 (launched in 2013) offers transfer rates of 10Gb/s, approaching that of the first generation of Thunderbolt interfaces. Like earlier USB interfaces, the USB3 format provides a 5V power supply, but the current rating has been increased from USB2.0's 200mA (1.0 Watts) to 900mA (4.5 Watts), and sockets designated as charging sockets can manage 1500mA (7.5 Watts).

USB-C - is the latest (2014) incarnation of a Universal Serial Bus interface, carrying bidirectional data at speeds of 10Gb/s. USB-C is not compatible with previous generations (USB 1.1, 2.0, 3.0, 3.1) as it has a very different 24-pin connector which is symmetrical and employed at both ends of a free cable, and all devices have the same socket. Power is optionally available over USB-C conenctions at 5V with a current capacity of either 1.5A (7.5W) or 3A (15W)

Valve - Also known as a 'tube' in America. A thermionic device in which the current flowing between its anode and cathode terminals is controlled by the voltage applied to one or more control grid(s). Valves can be used as the active elements in amplifiers, and because the input impedance to the grid is extremely high they are ideal for use as an impedance converter in capacitor microphones. The modern solid-state equivalent is the Field Effect Transistor or FET.

Vari-Mu Compressor - An audio compressor that employs a valve (tube) as the variable audio attenuator. Mu is an engineering term for gain, so this is a variable-gain compressor. In essence, the side-chain signal continuously adjusts the bias o the valve to alter its gain appropriately. Vari-Mu compressors are fast and smooth, with low distortion.

VCA - Voltage Controlled Amplifier. An amplifier in which the gain (or attenuation) is controlled by an external DC voltage. VCA's are used in a wide range of audio and musical equipment, such as fader-automation systems in large format mixing consoles, audio compressors, and synthesizers.

VCA Compressor - See VCA. VCA compressors tend to be fast-acting (at least in comparison to opto-compressors), a wide dynamic range, and low distortion.

VCA Group - Found in large mixing consoles. The fader levels of a number of separate channels assigned to the VCA group can be adjusted together by the VCA Group fader but without mixing their signals together. Usually referred to as a DCA Group in a digital console.

VDU - Computer display screen (See also Monitor).

Velocity - The rate at which a key is depressed. This may be used to control loudness (to simulate the response of instruments such as pianos) or other parameters on later synthesizers.

Vocoder - A signal processor that imposes a changing spectral filter on a sound based on the frequency characteristics of a second sound. By taking the spectral content of a human voice and imposing it on a musical instrument, talking instrument effects can be created.

Vocal Booth - See Isolation Room

Voice - The capacity of a synthesizer to play a single musical note. An instrument capable of playing 16 simultaneous notes is said to be a 16-voice instrument.

Vibrato - Pitch modulation using an LFO to modulate a VCO. (cf. Tremolo)

VU Meter - An audio meter designed to interpret signal levels in roughly the same way as the human ear, which responds more closely to the average levels of sounds rather than to the peak levels. (cf. PPM)

XG - Yamaha's alternative to Roland's GS system for enhancing the General MIDI protocol so as to provide additional banks of patches and further editing facilities.

XLR - A very robust and latching connector commonly used to carry balanced audio signals such as the outputs from microphones or line level devices. An XLR is a type of connector developed by US manufacturer, Cannon, and used widely in professional audio systems. The company's original X-series connector was improved with the addition of a latch (Cannon XL) and a more flexible rubber compound surrounding the contacts to improve reliability (Cannon XLR). The connector format is now is available in numerous configurations, from many different manufacturers, and with several different pin configurations. Standard balanced audio interfaces - analogue and digital - use three-pin XLRs with the screen on pin 1, the 'hot' signal on pin 2 and the 'cold' signal on pin 3.

X-Y - A specific way of mounting two directional microphone capsules such that they both receive sound waves from any direction at exactly the same time. Information about the direction of a sound sources is captured in the form of level differences between the two capsule outputs. Commonly, the two microphones in an XY array are mounted with a mutual angle of 90 degrees, although other angles are sometimes used. The two capsules will have the same polar pattern, the choice of which determines the stereo recording angle (SRA). The XY configuration is entirely mono-compatible because there are no timing differences between the two channels.

Y-Lead - A form of adapter cable that passively splits the output of a source to feed two or more destinations. Y-leads may also be used in unbalanced console insert points in which case a stereo jack plug at one end of the lead is split into two monos at the other for separate send and return connections. A Y-lead must never be used to combine signals.

Zenith -The parameter of tape head alignment relating to vertical alignment head and whether it is leaning forward or back relative to the tape path. (cf. Azimuth and Wrap)

Zero Crossing Point - The point at which a signal waveform crosses from being positive to negative or vice versa.

Zipper Noise - Audible steps that can occur when a parameter is being varied in a digital audio processor.

Last updated 19 November 2019.

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Welcome to Sound On Sound's indispensible, regularly updated, explanations of technical terms from the fields of Recording, Audio Production, Music Technology, MIDI, Music Software, Audio Plug-ins, Mac and PC Computing, Live Sound, Acoustics, Electronics and more... If we do not explain a particular term below, please email glossary@soundonsound.com and we will add it to our next update.

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Two Ways To Listen To Music Productions

Critical listening, and analytical listening are two basic ways to listen to music productions when working on them professionally. These two methods of interpreting recordings are the foundation for all of your music production work.

The purpose of this audio program is to distinguish the two different methods of listening and how you use them. You need to listen to music as both an engineer and as a producer to achieve great results. Both methods play a huge role in how music is perceived and accepted by the consumer.

Critical Listening

Critical listening primarily comes from the engineer's point of view. From the engineer's perspective you would be listening to the physical details of music. Frequency response, dynamic range, tone, imaging, and how your instruments are blended together.

Having control of the technical aspect of sound is important because it allows you to work with the issues that are present in musical performances and the way they integrate together. Sometimes, specific frequency areas between performances overlap each other and prevent each performance from being heard clearly. Sometimes the sound of the particular instrument wasn't quite what it should've been and needs some form of processing to come across more realistically in the production.

Understanding the critical listening aspects of sound is the foundation of engineering work but it is not everything. Technically perfect productions don't always feel great and can easily come across as cold or lifeless. It is equally important to understand how the technical aspects of sound affect the feeling of a performance.

Analytical listening

The other type of listening is analytical listening. Analytical listening is all about feeling and meaning. Sound gives us meaning and perspective in our lives. For example, someone talking to you can say hello. On the surface it seems friendly, but the way they say hello to you can convey a lot of different meanings. If they say it in a very short and abrupt way, it may be apparent that they are angry at you, not happy to see you.

These gestures come across in a more subtle way with music. It is important to understand that the emotional intention of a musical performance is reflected in the sound. Everything in a music production must reflect this intention in order for the feeling of the song to be properly conveyed to the listener. The following will go into more detail about the different aspects of Analytical listening.

Meaning is the essence of every music production. If the meaning of a song does not come across clearly in the production, you will confuse the listener. The meaning of a song is conveyed by adapting the characteristic feeling into the instrumentation. The setup of each instrument and way you choose to record them will greatly influence the way those instruments express themselves.

If the sounds don't weave through the vocal performance and support the meaning of the song then you will be missing something that is critically important. There is no such thing as a perfect sound for every style of music. There is no one drum sound that works for every type of recording and every style of music. As a producer or engineer you need to be able to define what type of feeling or "sound" it is that you're going for. These decisions go way beyond the critical listening characteristics of frequency.

Each Song Is Unique

Most of the sound and meaning of a recording goes into controlling the performances of the musicians and asking them to respond according to the needs of the song. However, if you use the same miking techniques, equalization, and other forms of processing that you would use for a heavy rock track and apply them to a ballad it's going to sound off. It will also impair the ability of the musician to interpret the meaning of the song.

Just because the musician changes what they are playing doesn't mean that the sound translates to another style. Even within the same style of music, a change in the tempo will greatly affect the sounds. If the tempo is slower the sounds can be more open and sustaining than it would for an up-tempo song. The selection of instrumentation, their sonic tuning and the performance is critically important.

With analytical listening, you're looking for the meaning and interpretation of what is being played. Your ability to see discrepancies in the approach, and adapt those performances accordingly, is all you really need to get great sounds.

Language is something that we've grown up with our whole lives for the purpose of communication. Because we have done it most of our lives, it is very easy for any person to interpret the meaning and emotions that come across from another person. The same process can be applied to music but the meaning and feeling maybe a little less obvious. The meaning must be conveyed on a deeper level that may not be so consciously apparent.

Educating Your Ears

Educating your ear to any music production style is like studying language. It's really important to listen to music you want to record. Whatever type of music you enjoy, you will need to listen to it with both an analytical and a critical ear. You need to study the best of the best. Study the people that really convey the feeling and emotion of a particular style of music well. When you study in this way, you learn the nuances and subtle things that are done in the music production that make it work. The selection of sounds, the performances, the balances of instruments all will allow you to understand the essence of the style.

Understanding and applying this concept is really a make or break thing for any song. Ultimately the critical aspects of the audio recording will have importance, but a poorly recorded track that is an incredible performance can still strike peoples feelings. Conversely, a well engineered recording that is a very poor performance will not reach anyone.

Balancing Analytical And Critical Listening

The secret to making great recordings is to plan the critical listening aspects of the recording around an environment that allows the artist to be the most expressive. This is a balance that works well for both the engineer and the artist. Remember that analytical listening is really about the interpretation of meaning and feeling. The critical listening aspect deals with the different types of sounds and the frequencies areas in them that are most important for capturing the intended feeling.

It may be warmth, brightness, or a shrill harshness that's conveys the emotion of a performance best in a particular type of music. If a song is about being agitated and angry, a shrill harshness may help to convey that feeling across to listener better. In a love song, that harshness and abrasiveness will not convey a feeling of love. You will likely be going towards warmer sounds with an open high frequency response. Analyzing different styles of music, in this way, is critical to learning how to convey the feelings and emotions associated with them.

Frequencies

Critical listening is much more about engineering skills and the interpretation of how different frequencies areas affect us emotionally. It is a huge part of educating your ears. Understanding what frequencies play the most significant role in different instruments is fundamental to engineering work. Understanding the harmonic content of any given instrument and being able to work with it is also a key component of making those instruments sound alive.

A deep understanding of frequency and harmonic areas will allow you manipulate and move the sounds around the speakers. Knowing what frequencies with different instruments interfere with each other when being blended together and which frequency areas add clarity when blended together gives you enormous control of your recording and mixing. There are many approaches and many ways that you can work with these concepts.

The ability to hear compression and know how to use compression is also a critical listening aspect of sound. There is a significant difference between the amount of a frequency are versus the density of a frequency area. Density adds power without interfering with other frequency areas. This is the art of compression. For example, if an engineer is working on a record and the producer asks for the drum sound to be bigger and more aggressive, the engineer might be more inclined to add compression to the drums rather than EQ. Compressing a particular frequency range may help to help bring out the ambient energy of the drums and thus make the space around the drums seem larger.

Interpretation

The ability of an engineer to analyze and interpret through the eyes of a producer or artist what is needed on the critical level is the definition of an engineer. This requires a lot of critical listening and analysis of great mixes. Once defined, those approaches recording and mixing must be practiced with a critical mindset until they become second nature. Experience is always the best teacher.

In the next audio, Educating Your Ears, a practical approach to this process is outlined. The sooner you begin to use these techniques and practices, the sooner you will be able to get great results with your music production and engineering work.
The Decibel Fletcher and Munson Selective Hearing Speed of Sound and Wavelength Acoustical Phase The Sound Envelope

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Critical Listening vs. Analytical Listening

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Critical listening, and analytical listening are two basic ways to listen to music productions when working on them professionally. These two methods of interpreting recordings are the foundation for all of your music production work. The purpose of this audio program is to distinguish the two different methods of listening and how you use them.

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