EQ Overview and Introduction

In the next series of blog posts, I’m going to go through EQ, or equalization. I will talk about why we use it, and when and how to use it. I think EQ is easier to understand than compression (my last series of blog posts), but when I see EQs added by young producers and engineers, I realize they are just as lost using EQ as a compressor. Partly, this is because they don’t understand frequencies.

In this blog I am going to start with an overview. I think to understand EQ and use it properly, one must understand frequencies, our ear’s perception to frequencies, the frequency spectrum (or range), and frequency specifics of individual instruments.

To start with, the ear hears 20 Hz to 20,000 Hz (or 20 kHz). This is, of course, ideal, but starts to become less (mostly on the high end), soon after we’re born. If you’re serious about a career in music, it would serve you well to NOT listen to loud sources for very long. Personally, I wear ear protection when using my leaf blower and shop vac!

If you think about an acoustic piano, the lowest note is A0, 27.50 Hz, and the highest note is C8, 4186 Hz. I bring this up because I think it helps us equate pitch with frequency. Next chance you get, go play specific notes on a piano (acoustic or digital), and then consult a chart as to the frequency of that note. For instance “middle” C is 262 Hz. A440 (the A just above middle C) is 440 Hz.

Do you know what an octave is? An octave, at least on a piano, is from, say, middle C up or down to the next C. This happens to be 8 white keys, thus octave. Using octaves, the frequency either doubles (up an octave) or halves (down an octave). So middle C (C4), 262 Hz, up an octave goes to C5, 523 Hz. (Technically, C4 is 261.63 and C5 is 523.25 Hz.) The A above middle C, A4 is 440 Hz. Up an octave is 880, down an octave is 220. Down another octave is 110, then 55, then 27.50, the lowest note on the piano. All instruments, of course, can go up or down octaves at a time.

The lowest note on a guitar is E2, 82 Hz. Guess what a bass guitar’s lowest note is? One octave down, 41 Hz. This is important. For one reason, when EQing either of these instruments, I know there is no useable information below those frequencies, so I will use a high pass filter set just below those frequencies. This helps to clean up the sound of these instruments, make them less muddy.

The frequency spectrum (20 Hz – 20 kHz) is broken into ten octaves:

1. 20 – 40
2. 40 – 80
3. 80 – 160
4. 160 – 320
5. 320 – 640
6. 640 – 1280
7. 1280 – 2560
8. 2560 = 5120
9. 5120 – 10, 240
10. 10,240 – 20, 480

So, the lowest note on a bass guitar, 41 Hz is in octave 2; lowest guitar string is octave 3; middle C on piano is octave 4; A440 is in octave 5. Where do vocals sit? Fullness, for example, is 140 – 440, octave 3 to 5.

A different and more effective way to think about the frequency range is to break it up into five broader ranges:

1. 20 – 100         Bass (Sub Bass)
2. 100 – 500       Mid Bass (Upper Bass)
3. 500 – 2 kHz   Mid Range
4. 2 k – 8 kHz    Upper Mid Range
5. 8 k – 20 kHz  High (Treble)

Bass:                           Depth, Power, Thump

Upper Bass:              Warmth, Body, Fullness

Mid Range:               Bang, Nasality, Horn-like, Fullness of high notes

Upper Mid Range:  Presence, Edge, Punch, Brightness, Definition, Excitement

Treble:                        Brilliance, Sizzle, Treble, Crispness, Airiness, Breathiness

As example, electric guitar has too much “edge,” cut in the upper mid region. Vocal sounds a little nasal, cut in mid range area. The overall track needs more power and punch, boost bass region.

————————–

To go deeper, the human ear (and mind) hears and perceives sound differently at different frequencies and levels of loudness. Generally speaking, we are more sensitive to mid range and upper mid range frequencies. The ear is less sensitive to low frequencies at lower volumes, and slightly less sensitive to higher frequencies compared to mid range frequencies at the same volume. Being more sensitive means we hear it easier and more readily.

Another way to say this is at low listening volume, mid range frequencies sound more prominent, while the low and high frequency ranges seem to fall into the background. Conversely, at high listening volumes, the lows and highs sound more prominent, while the mid range seems comparatively softer. Confusing? Yes. But extremely important to understand.

To illustrate – Let’s say you’re working on the EQ of a mix, and as you listen back at low levels, you think the lows and highs could use a boost. So you boost them, and it sounds great. The next day you listen back at a high volume, and notice the lows and highs are too loud, so you cut them back down some. Sound familiar? This is the Equal Loudness Contour effect.

There are two different charts one could consult to dig deeper into this important, albeit nerdy and technical subject – Fletcher-Munson Curves and Equal Loudness Contour. There are many articles available online regarding these two subjects, so I will not get into them. BUT, it is extremely important to realize how important these affect your work as an audio professional!

I went a little more in depth than normal in this post, but I hope it helps you to understand what is involved when learning to become an successful producer or audio engineer.

Peace –

And, HEY! Make it a GREAT day!!

Tim

Advertisement

Compression Dos & Don’ts

To wrap things up regarding compressors, I will offer 3 Dos and Don’ts as my final word for now. These are things to always keep in mind when working with compressors. Some may have been previously stated in an earlier blog post.

DO          Avoid using extreme settings to begin with, if you are just trying to control the dynamics.

DON’T   Add compression to every channel by default. Start off with minimal compression, and carefully choose where to add compressors.

DO         Experiment with different types of compressors – hardware and software. There can be differences in how they sound. Compressors can and do sometimes sound different from one another.

DON’T  Forget to bypass the compressor occasionally while setting to check the results.

DO         Remember to balance the output gain so the level doesn’t change when engaged and bypassed. This way you can accurately compare before and after. Also, typically compression is added AFTER the mix has been balanced. So you don’t want to alter levels with either compression or EQ.

DON’T  Be afraid to experiment. Some of the greatest sounds in the history of recorded music came from misused and abused compressors.

Compressors 201 – Threshold

A compressor has a lot of knobs and settings. They can be confusing at first. In this blog I am going to talk about one of those knobs – Threshold.

A compressor is an automatic volume control. We try to make the signal somewhat the same, static. Without a compressor, we would have to do it manually, turning the signal down, then up, then down, etc. But with a compressor, it can do that job for us.

When a signal gets loud (and crosses the threshold), it turns it down. If there’s any makeup gain, it will turn the softer signals up making them louder (along with everything else, of course).

The threshold setting tells the compressor when to start working. Put a compressor on, say, a vocal track. Pay attention to the input signal on the compressor. Let’s say the input signal is -10 dB. Now set the threshold knob to -16 dB, and the Ratio 2:1. What we’re telling the compressor to do is this: Any signal that is stronger than -16 dB, I want it to compress (lower) the signal. When the signal crosses the threshold it will get cut (attenuated) in a 2 to 1 ratio. So in this scenario, the signal is coming in at -10, with the threshold set to -16 and a 2:1 ratio. This means 6 dB is going to get compressed in a 2:1 ratio. The signal will get cut down to 3 dB over. You can think 2 becomes 1, 4 becomes 2, 6 becomes 3, etc.

If the ratio was 10:1, the signal would get compressed more. The higher the number, the more the compression. If the signal crossed the threshold by 10 dB, then it would be reduced to 1 dB.

The threshold determines how much of the signal the compressor is going to affect. You can change where the threshold is set usually in two ways. Using the Dyn3 compressor/limiter, grab the orange arrow on the signal led and slide up or down. Or to the far right at the bottom, grab the threshold knob and set up or down.

TIP: If Pro Tools is your DAW, use the Dyn3 Compressor/Limiter (free). It helps to make understanding what and how a compressor works easier.

I hope this helps!

and HEY! Make it a great day!

Tim

Timebase in Pro Tools

In Pro Tools, material (audio or midi) on a track is associated with a type of Time Scale. All track types can be set to either sample-based (for the Sample Time Scale) or tick-based (for the Bars/Beats Time Scale). Different tracks can be set to different timebases as needed.

Audio tracks are sample-based by default. This means that audio clips have absolute locations on the timeline and are tied to specific sample locations. If you change the tempo or meter the audio will not move. This is helpful, for instance, if you import an audio clip and want to build other audio or midi tracks around it and end up changing tempos or meters a few times. You don’t want to affect the original clip. 

However, MIDI and instrument tracks are tick-based by default. This means that midi clips are fixed to bar and beat positions and move relative to the sample timeline when tempo and meter changes. So if you change the tempo, the midi will either speed up or slow down accordingly. 

A good tip to keep in mind is Elastic Audio-enabled tracks can be switched to tick-based in order to automatically follow tempo changes in your session and conform to the session’s tempo map. 

And lastly, you select whether a track is sample-based or tick-based when you create it, but you can change timebases later as needed. 

Hope this helps!
Peace – and HEY make it a great day!
Tim