The Different EQ bands and What they mean (part 2)

Dyn3 7-band EQ (Avid Pro Tools free plugin)

If you look toward the bottom of the EQ pictured above, you will notice 5 different bands: 1. LF, low frequency, red; 2. LMF, low-mid frequency, orange; 3. MF, mid frequency, yellow; 4. HMF, high-mid frequency, green; 5. HF, high frequency, blue.

In today’s blog I will talk about these five bands. I want to start with band 1 and 5. These are typically used and referred to as “shelves.” Band 1, low frequencies, is the low shelf, and band 5, high frequencies, is the high shelf.

But these two bands each have two different settings. The small left icon, next to the LF and HF, is called a bell-type EQ. It kind of looks like -o-. This will either boost or cut a section of frequencies set by you with the frequency knob. The ‘Q’ knob will determine how wide or narrow the bell curve will be. A low Q setting will give you a wide band of frequencies, and a high Q will render a narrow band of frequencies. A good rule of thumb is wide when boosting and narrow when cutting.

The typical use for this is to, say, boost the lower frequencies to bring out a kick drum or synth bass. On the high end, with the HF knob, we can boost upper ‘air’ frequencies to make guitars or vocals stand out or sound brighter. Of course, we can also cut in these frequency ranges as well.

The other icon setting is called a ‘shelf.’ This is the more common use for these two bands. Typically we use a boost here (low or high). When boosted, it looks just like a “shelf.” If on the low shelf, we set the frequency knob to 125 Hz, then everything from 125 on down (to 20 Hz) is boosted the same amount. On the high shelf, we might add a shelf for vocals starting at 6 kHz. In this case everything from 6 k up will have a boost. Of course, we can also cut using a shelf, but this happens less often then a boost.

The Q factor is a bit more complicated and will have to be reserved for another post.

Bands 2, 3 and 4 allow for bell curve settings only. These are the same as the bell curves on bands 1 and 5. These are used for low-mid, mid, and high-mid frequencies. There are only three knobs: Frequency, Gain and Q. Frequency, of course, sets the frequency that you want to work with. Gain is volume (loudness) and can be plus (positive) or minus (negative). We might say boost 2 kHz 2 dB (2 dB) which is a positive gain. Or cut 1200 Hz 3 dB (-3 dB) which would be a negative gain.

As stated above, Q determines the amount of frequencies being altered by the EQ.

As always, I hope this helps!

And, HEY! Make it a Great day!

Tim

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The Different EQ bands and What they mean

Throughout my blog series on EQs I am going to refer to the free EQ plugin that comes with Pro Tools, the Digirack EQ III 7-band. First, let’s talk about the input/output LED meters and gain controls (top left of the plugin). This simply shows the input and output signal level running through the EQ. Always check to make sure there is no clipping going on. If on the input or output side the signal is clipping, hitting red, simply turn the respective gain knob down until there is no longer any clipping happening. It is normal to adjust these gain knobs. With the input gain knob is a symbol, Ø. This is the polarity switch “button.” This will invert the phase of the incoming signal. If you don’t know what this is, I will cover it in a later post. It’s a little more advanced, but easy to understand and know when to use. For now, it won’t concern us.

Just beneath the input/output section are two filters. There is a high pass filter and low pass filter (HPF, LPF).

High Pass Filter

Low Pass Filter

There is also a notch filter. It looks like a line with a ‘V’ in the middle of it. (I couldn’t find a pic of one.)

-∨- (notch filter)

The high pass filter alllows high frequencies to pass through the EQ, while cutting low frequencies, not allowing them to go through the EQ. Conversely, the low pass filter allows low frequencies to pass through and cuts high frequencies. The notch filter takes a small section of audio and makes a deep cut (-12 dB or more). It takes a “notch” out of a small section of audio frequencies. The frequency can be set by the user. One use for the notch filter is for plosives on a vocal track. When a vocalist pops, say a ‘p,’ set the notch filter at 100 Hz. It should diminish it greatly or make it go away completely. You may have to sweep the EQ up or down a little to take care of it.

The two filters each have an “IN” button to engage them. They will light up blue when engaged. The frequency, of course, can be set to whatever you want.

Lastly, there is a setting that is used for the HPF/LPFs that tells the filter how steep of a cutoff you want. If we are allowing high frequencies to pass through and cut out low frequencies, how strong do we want to cut off those low frequencies? The slope is set per octave. The setting choices are 6 dB per octave, 12 dB/oct, 18 dB/oct, and 24 dB/oct. As an example, let’s say I set a high pass filter at 200 Hz, with a 12 dB/octave slope. What this means is that only frequencies above 200 Hz will pass through the EQ (and anything further in the signal chain), and frequencies one octave down (100 Hz, remember from my previous post?), will be 12 dB quieter. Another octave down, 50 Hz it will be another 12 dB quieter. There are times we want a steep cutoff, like 24 dB/octave and other times when we might want 6 dB/octave.

Look on the left side of the GUI window, which shows the graphic interface. There are small numbers. On the center line is 0. This is where all EQ bands start. In out example, since we’re cutting at 200 Hz, at 100 Hz the downward slope will be at -12 dB. At 50 Hz it will be -24 dB.

I hope this hasn’t been too confusing. Try experimenting with these filters on a mix you’re working on. Keep your ears open when doing this. You can even experiment on a piano or acoustic guitar track. Set the HPF up higher, like 400, 500 Hz. Change the different octave settings. You should hear what’s happening.

As always,

Make it a GREAT day!

Tim

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.

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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

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

Compressors 103 – Going deeper

Today I’m going to go a little deeper into how compressors work. But first, I want you to do an experiment. This experiment will allow you to SEE what a compressor does. When I did this, things started to become a lot clearer for me. We have to listen differently when it comes to compressors. We are not listening for frequency, we are listening to dynamic changes (amplitude, loudness).

For this experiment it would be better to use a bass track. The reason for this is one, we ALWAYS use a compressor on a bass track and two, the reasons to use a compressor on a bass track is because we want to control the initial attack of a note and lengthen the sustain of the notes, as they fade out quickly. Try to use a track that has a regular-type bass line, not one that is super busy.

1. Find a bass track (or create one using a VI).  If a virtual instrument is used, use one that emulates a real bass guitar, not something like a synth bass. What is needed is a sound whose initial attack is strong (loud) and whose note decays after the initial onset of the note, like a real bass. Also, the track will need to be printed. For this test to work we need an audio clip, not a MIDI clip.
2. Put a compressor on the bass audio track.
3. Set the ratio to something higher, i.e. 6:1. Set a fast attack, i.e. 3 ms; fast release, i.e. 18 ms.
4. Lower the threshold until the meter shows roughly 8 dB of gain reduction. Then add some makeup gain. Set this to the amount being reduced. If the gain reduction meter shows 8 dB of reduction, set makeup gain to 8 dB. This way the volume remains the same.

What you should start to notice is that the initial attack of the notes (when the player first strikes the notes) no longer punches. Now the dynamics are a little flatter, smoothed out. The second thing you should notice is that the notes are more sustained. You will no longer hear the decay, but a nice solid note that holds out for it’s full duration of note value (i.e. quarter note, half note, etc.)

NOW, print the track again with the compressor engaged. What you should have are two printed bass tracks. One without compression and one with compression. Look at the differences between the two. The first track has a pronounced attack with high amplitude and fast decay. The second, the initial punch is now all or mostly reduced and the sustain of the note stays strong longer. Below is a picture of what this should look like. 

I have done exactly that here. I used an Instrument track with Trillian Bass module – played a bass line – printed it – ran compressor with 6:1, 3 ms attack, 18 ms release, 8 dB gain reduction, and 8 dB make up gain. I think this helps to drive home what a compressors job is. In this example my goal with the compressor was to lessen the attack and give it more sustain. If you want to see what the compressor does even more obvious than this, use the fastest attack possible, with 8:1 ratio, with a lower threshold for more gain reduction (10 dB).

I hope this helps – it did for me! Next time I will start to go into specific parts of a compressor, i.e. threshold, knee, attack, release, etc.

As always – HEY! Make it a great day!
Tim

Organize, Organize, Organize! (Sessions, Folders, Files)

Yes, more organization! 🙂

I think I have always been somewhat organized. But I have learned to be very organized since starting my studio. Things can get very unorganized, confusing and messy very quickly.

If you want to save yourself some headaches down the road, grab 20-30 mins to take some time when you’re not very busy and think about how you want to organize folders, files, sessions, and clients. 

Being able to find a client’s session quickly becomes key. For instance, I have one client who came in a few years ago with just one project. As the months and years have gone by she has done a dozen or so different types of projects. I made the mistake of thinking what she was bringing me was a one-time project, so I just threw it in with another of her sessions. I have now done that so many times that when she calls and asks if I have so and so I don’t know the answer. She is the exception for me. But her different projects are a MESS!!

So I had to develop a “system” of sorts to stay organized. Here’s what I do:

I put the clients’ last name first, then first name. So Tom Smith becomes Smith Tom. Bands are simply listed by the name of the band. I always capitalize the main client folder and the main folder for a song (i.e. Smith, Tom).  I usually put a sub-folder inside the client folder with the song title – and I do one folder per song. I do not put multiple songs in the same folder! If they have three songs, they get three folders. The folder for the song I put in caps, like “The Setting Sun”. Then the session in that folder becomes settingsun. I always do ‘Save As’ with the sessions as I get something done, incorporating a number scheme. So the session I put settingsun_01 xxx. With the xxx being what was done during that session, i.e. vocal tracking, eq, rough mix, etc. Then settingsun_02,etc.
Although I don’t always do it this way, it’s good to know that ‘year/month/day’ formatted dates sort alphanumerically; ‘day/month/year’ (UK) and ‘month/day/year’ (USA) standards do not. Think about it: under the UK system, the 1st of December sorts in front of the 2nd of January because 1 comes before 2. And in America Jan the 1st 2008 comes before the December the 1st 2007 because 01 comes before 12.

This works for session files too. Versions of the same day get suffixed a, b, c etc. Dates are more logical than descriptives like… ‘final mix’, ‘final final mix’, ‘final final mix THIS ONE’, new final mix, ‘new new final mix unmastered reverb +EQ’ etc., etc. I don’t use descriptives this way.

If you use the YYYYMMDD format, they’ll sort alphanumerically on a computer and you’ll be organized!

I hope this helps!
Peace- And remember – make it a great day!

Tim

Compressors 102 (More of the Basics)

After learning the basics about compressors (see Compressors 101 earlier blog entry), then you can use this general guide of the type of overall effect you are going for.

If you want a Natural sound (the compressor is not noticeable):

Use a slower attack (longer than 75 ms) and gentle ratios (less than 2:1). Always allow the compressor to “relax” back to zero several times a measure.
For a Punchy Response:

For a harder, punchier sound, use higher ratios and thresholds, but keep an ear out for any distortion.

If you want a Thick and Dense sound:

For a thicker, denser sound use faster attacks, medium ratios, and lower thresholds. There will be much more gain reduction though.

If you want a Pumping Effect (for EDM, for instance):

For an overstated pumping effect use fast attacks, high ratios, and a longer release time.

DO – Avoid using extreme settings to begin with. This is especially true if you are just trying to control the dynamics.

DON’T – Add compressors to every channel just because you think you’re supposed to! Start with minimal compression and carefully choose where, when and why to add a compressor.

DO – Experiment with different kinds of compressors. There can be some big differences!

DON’T – Don’t forget to bypass the compressor occasionally to check that you’re getting good results.

DO – Remember to balance the output gain so the level doesn’t change when you engage the bypass. In other words the before and after volume level should be the same. We hardly ever use compression without changing the output makeup gain. If you add 3 dB of gain reduction (GR), then you should be able to add 3 dB or so of make up gain for the output.

DON’T – Don’t be afraid to experiment. Some of the greatest sounds in the history of recorded music came from misused and abused compressors!

The next blog about compressors I will talk about the “Knee” of a compressor! I really do hope this helps. It helped me!

As always – Make it a GREAT day!

Tim

Creating a Click Track (in Pro Tools)

Using a click track during recording is, of course, imperative. We can’t do our work if we can’t play to the beat! Luckily for us Pro Tools makes it easy to set up a click track! Just simply go to Track on the menu bar, scroll all the way to the bottom and choose Create Click Track. Pro Tools will create an auxiliary track and automatically put a metronome plugin on the track. The metronome will automatically set to the tempo (bpm) of the song.

The metronome is customizable. You can change the sounds used for beat 1 and all other beats (2, 3, 4, etc.). The volume for beat 1 can be set and the volume for all other beats can be set to something different. I usually have mine set so that beat 1 is louder than other beats and is a different sound. That makes it easy to find the downbeat while tracking.


When I set up my templates, I already have the click track set up and ready to go! Easy! And since I organize and colorize my tracks, for me, the click track is all the way to the left in the Mix window and a bright lime green. I always know where it is in the session, no matter working in the mix or edit window.

Lastly, you can save a preset of the type of click you like. On the click plug-in, select the drop-down arrow next to Preset, select Save As, and name it! That’s it!

Peace! And HEY! Make it a great day!

Tim

6 Recording Myths – Busted!

It is hard to learn how to record and mix music today. With so much information available on the web, sometimes it is hard to know if the information is true or not – whether it can be trusted or not. Here are six myths that are not true! Ask anyone who really knows his stuff and is experienced and successful.

Myth 1 – You can’t use ribbon mics on loud sources

This myth is a good one to start with because like the best myths, there’s just enough of a grain of truth to it to keep it going. It’s true that the actual ribbon element can be more fragile than the diaphragm of a moving coil or condenser microphone. It’s also true that in the early days of ribbon mics, those classic RCA mics from the 1940s would fail readily if you tried to use them on a screaming guitar amp or a kick drum. However, that hasn’t been true for decades. These days, arguably the most venerated guitar cabinet mic, the Royer R-121,  is a ribbon mic. Ribbon mics these days can easily withstand extremely high Sound Pressure Levels (SPL) and can be used on any source. Some ribbon mics such as the Shure KSM313/NE utilize a ribbon made of Roswellite, a substance created using carbon nanofilm technology that is virtually unbreakable and can endure levels up to 146dB SPL.

Myth 2 – Always record as hot as you can

This is another myth that has roots in the early days of recording to tape. Back when your recordings had to stay above the noise floor of the tape, tracking too quietly could render your recording noisy and unusable. Not only that, but recording engineers realized that for rock music, slamming your recording levels produced a very pleasing tape compression and “heat” that could make things sound great. With digital recording, however, both of these are no longer true. With 144dB of dynamic range (24-bit recording) you can even record at -40dB and have 100dB of dynamic range. Early analog-to-digital converters (from decades ago) did sound better when recording near the top of their range but that is no longer the case. In fact, with digital recording, overloading your recording levels is decidedly unpleasant, resulting in a digital distortion when clipping that is ugly and abrasive.

Myth 3 – External digital clocking improves the sound of your audio interface

If you’re interconnecting a lot of digital gear you may want to use a master digital clock. Get the best clock you can afford, and make sure everything is connected properly via Word Clock cables. In many cases, the master clock won’t have a drastic influence on the sound; the uniform clocking simply makes everything work together without digital pops and ticks. Just taking your audio interface and hooking it up to an external clock isn’t going to improve the sound quality of its digital-to-analog and analog-to-digital converters unless the clock in your interface is really poor. If you really want to improve your recorded sound, get the best mics, preamps, and audio interface you can. Only buy an external digital clock after you’ve made sure the rest of your audio chain is the best it can be.

Myth 4 – Egg cartons or mattress foam are good acoustic treatments

No, not even close! And despite what you may read on the internet, they don’t sound-proof anything. Materials such as drywall, insulation, and acoustic foam can be great acoustic treatment materials. With these materials and proper construction and application methods, you can effectively tackle the two general aspects of studio construction: isolation and acoustics. First, if you’re concerned with keeping sound from getting in or out of your recording space, you’ll need to tackle isolation. This is best done with some form of mass-air-mass construction. A wall with drywall and insulation, empty space, then another identical wall with drywall and insulation will provide a great start. For controlling the acoustics inside your space, you’ll need a combination of absorption and diffusion. There are myriad ways and a long list of proper materials to implement this — egg cartons and mattress foam are NOT on the list!

Myth 5 – External hardware always sounds better than digital plug-ins

In the early days of digital, this may have been true, but definitely not today. Sure, there are hardware compressors, equalizers, and effects processors with a certain mojo that sound amazing. But there are also digital software processors that sound incredible and offer a level of precision and recall that you’ll never get with external hardware. There’s a reason that nearly every pro studio has a ton of high-quality plug-ins even if they already have and use great outboard gear. You may like the sound of a piece of hardware, but you may like, or even prefer, the sound of a digital processor. The days of digital being second best are far behind us.

Myth 6 – There’s a “correct” way to record

It might seem counter-intuitive after all these “wrong” myths to proclaim that there’s no “right” way. But it’s true! One way of doing things may not get you the results you’re after, but then there are multiple ways that will. The name of the game is experimentation! Never stop experimenting and searching to find techniques that work for you, your music, your musicians, your studio. If you wonder if something will work, even if it seems patently false, give it a go! At worst you’ll need to redo it. At best you may add another unique tool to your toolbox. And that’s what recording is all about!

These are truths that all of us can learn from. I hope this helps musicians and engineers alike get better at their craft!

Peace – and HEY! make it a great day!

T