2017. 06. 30.

Equalization for Rookies Soundbytes Techniques

Equalization for Rookies (special guest Christian Siedschlag,

DDMF)
soundbytesmag.net /equalizationforrookies/
2013. 11. 09.

Christian Siedschlag, the man behind the DDMF, knows a thing or two about
equalization since he created a respected EQ plug-in. He shares some of his expertise
with us here.

Nov. 2013

DDMF it is a name that suddenly popped in a field crowded with a big names, representing an independent one
man company that become a subject of a many rumors on various music related forums, such as Did you hear
about that cheap equalizer that sounds better than most of the expensive ones? At the time, I was involved in an
extensive search for my main equalizer.I tried most of the expensive ones and they sounded good. But some of
them didnt had enough bands, other offered editing only through the knobs, some of them even dont sound as they
should. Some sounded good but were CPU intensive. I didnt expect too much from this newcomer. What a mistake.

To make long story short, Christian Siedschlag, the man behind the DDMF, created an equalizer that became my
main secret weapon, one that Ive used for many years on every song on literally every track. His IIEQ Pro can
sound analog or digital, it offers a wide range of filters, and whats more important, it sounds right. It is light on CPU,
easy to operate and no matter what you use as a source, it always sounds natural, no matter if you cut or boost the
material.

We all use equalizers, but what do we know about them? Lets draw back the curtain to discover some basic facts
about this tool, and even more, to unveil the secret how those tools are made in the first place. So, lets hear the
truth from the big master Christian from DDMF.
More about DDMF on http://www.ddmf.eu/

Equalization for Rookies

Remember that old saying Talking about music is like dancing about
architecture? Well, to some extent that statement is probably also true
when talking about the art of mixing music: it remains an abstract
matter when you dont have actual sound examples at hand, which is
necessarily the case in an article like this. Plus, many of the worlds
greatest mixing engineers feel more than they know what to do in
which situation. Fortunately (for our purpose), Im definitely not one of
the worlds greatest mixing engineers but just a developer of audio
software. By nature, Im forced to translate my intuitive understanding
of things into working software code, which is why theres good reason
to hope that I might be able to explain a thing or two about digital
signal processing, and about equalization specifically, which is the
subject of the present article. The target audience, as stated by the
title of the article, is rookies so for the more experienced readers a lot
of things will probably sound familiar.

What is Equalization?
Lets start with the definition: what is equalization? These days we fortunately have Wikipedia, so we can readily
answer Equalization (British: equalisation) is the process of adjusting the balance between frequency components
within an electronic signal. This definition implies that any electronic signal has frequency components, and indeed
2017. 06. 30. Equalization for Rookies Soundbytes Techniques

it is this mathematical fact that lays the foundation for all techniques of equalization. Sound, if viewed from an
engineering perspective, is nothing but a sum of individual frequencies (sine waves) played together. The relative
strength and timing of these frequencies with respect to each other uniquely describe a certain sound (or a whole
piece of music) and is called the spectrum. Equalizers are devices (real or virtual) that allow the mixing engineer to
lower or raise parts of the spectrum without affecting other parts of the spectrum.

Fig 1: ACDCs Highway to Hell, spectrum at about 2:18

In Fig. 1 we see an example of a spectrum, frozen in time. The horizontal axis is the frequency axis, showing at
what frequency a contribution to the overall sound is oscillating, while the vertical axis measures the strength
(loudness) of each contribution. The frequencies are measured in Hertz (which are just oscillations per second) and
the loudness in decibels (dB). The huge peak at about 80 Hz is the bass drum. We see a lot going on in the range
between 200 and 5000 Hz, which is where vocals and all the other instruments are fighting to be heard, and then a
slow drop off towards higher frequencies. The sharp drop off at about 16 kHz is a result of mp3 compression, by the
way. The flat white line is the equalization curve, as the screenshot is taken from one of DDMFs EQs, without any
equalization applied.

The Task of EQing

Youve probably already come across the following situation a few times: you have a track with a number of
instruments and maybe some vocals which all sound great when played individually, but after mixing them together,
it just sounds muddy and crowded. That great guitar line suddenly is barely noticeable, and when you raise the
volume on the guitar track, the vocals start to disappear. This is where you need to reach for your EQ. EQing is all
about carving space in your mix so that the individual tracks dont get in each others way.
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Fig. 2: Typical frequencies of selected instruments, and attributes connected with frequency ranges (from
http://www.independentrecording.net/irn/resources/freqchart/main_display.htm)

In the beginning (and not only in the beginning) it is very helpful to have a chart that approximately shows where in
frequency space instruments typically have the strongest components. There are a lot of these charts available on
the internet, and one of them is shown in Fig. 2. These types of charts can help you when deciding where you need
to apply equalization. Also shown are typical attributes that are often associated with certain frequency ranges, e.g.
warmth at about 150-220 Hz. This means that when a track is lacking warmth that frequency area might be a
good starting point for a little peak filtering. Which brings us to the next subtopic, namely

Types of EQ Curves
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There are equalizers out there which, using a technique called FFT, allow you to change the spectrum of a track in a
free-form way. While this approach clearly offers the greatest amount of freedom and flexibility, you can also quickly
experience a phenomenon called choice paralysis, where you have so many options and variables that in the end
you cant really decide in which direction you want to move. Especially in the beginning, it is much better to stick to
tried-and-true EQ curves which over the years have been implemented time and again, partially because they
were relatively easy to achieve with hardware, but also because they produce predictable and pleasant sonic
results. These types of curves can be controlled with a very limited number of parameters, depending on the type of
EQ you have. An equalizer typically offers a number of bands which are individual filters through which the signal
passes in series. The simplest form of an equalizer, a graphical equalizer, only allows the change of one parameter
per band, namely, the gain of the band (in dB). In the age of software EQs, however, this is an unnecessary
restriction: here well be dealing with the more general parametric EQs in which each band can be controlled by
setting gain, frequency and something called Q or width, which determines the width of the range in the spectrum
the band is operating on.

1.) High-pass/Low-pass Filters

These filters gradually block all frequencies below (high-pass) or above (low-pass) the band frequency. Theres no
gain control, and the width can be used to change the filter response around the band frequency from very smooth
to (in the extreme case) resonance-like. These filters are very useful to clean up the upper and lower end of the
frequency spectrum. It is often a good idea, for example, to high-pass all but the kick and the bass at around 80-120
Hz. The lowest string of a guitar, for instance, has a frequency of 82 Hz, so everything below that can be safely
discarded. Also, many engineers apply a low-cut filter to the whole bus signal at around 5 Hz, as this area cant be
heard by humans anyway and is only eating up unnecessary energy.

Fig. 3: High-pass (left side, 100 Hz) and low-pass (right, 5 kHz) filters with a Q of 0.5 (upper row) and 1.0 (lower
row)

Shelving filters boost or cut the frequencies above/below the threshold frequency by a fixed amount of dB. When
cutting, they are a little less drastic than low- or high-cuts in the sense that they do not progressively lower all
frequencies above or below the threshold frequency. Again, Q is used to shape the response around the threshold
region.
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Fig 4: High-shelves with different Q values

3.) Peaking Filters

Peaking filters can be used to treat an isolated range of the frequency spectrum. They have a center frequency
around which the response is symmetrical. Theres a boost or cut by the specified amount of dB at the center
frequency, with a smooth fall-off around the center frequency. The width of the active frequency window is set by
the Q value. Peaking filters can either be used to enhance specific areas of a tracks spectrum in order to make it
heard more clearly (something which is easily overdone, though, so be careful!) or to reduce annoying areas. For
instance, in order to decrease the muddiness of a mix, it is often a good advice to apply a broad cut by 2 or 3 dB at
around 300-500 Hz to the master (sum) signal.

Fig. 5: Peak filters at 1000 Hz with a gain of 8 dB and different Q values

4.) Bandpass Filters

Bandpass filters leave the center frequency untouched but cut out an increasing amount of dB with increasing
distance from the center frequency. This type of filter can also be used to effectively decrease the necessary
bandwidth to transmit a signal. A famous effect is the telephone voice which can be generated by a single band
pass filter set to about 1000 Hz, with a bandwidth of about one octave. While often sounding thin when applied to a
single instrument, it can help to make the instrument sit better in a mix and create space for the other competing
tracks.
2017. 06. 30. Equalization for Rookies Soundbytes Techniques

Fig. 6: Bandpass (left side) and notch filter (right side)

5.) Notch Filters

A notch filter is the exact opposite of a band-pass filter: it completely cuts the spectrum at the center frequency, and
gradually less so around the center frequency, until it reaches 0 dB gain outside a window determined by Q. A notch
filter is very useful for removing annoying or problem frequencies (for instance a 50 Hz humming from the power
supply). A nice technique when you have the feeling that a track has some annoying component but you cant
exactly figure out where it sits is to apply a large-gain, narrow peaking filter and slowly sweep it across the
frequency spectrum. When you have localized the problematic area, apply a notch filter there, with a width thats
large enough to be effective but small enough to avoid any unwanted side effects.

These are the basic filter types that are available in almost all of todays (software) EQs. The implementation details
may vary, especially when you start comparing Q values almost every developer uses his own definition. Another
point thats influenced by the implementation is the CPU consumption. When the number of tracks to be equalized in
your projects is huge, this is something that will definitely become important at some point.

Design of (Software) EQs

Although you can happily use your EQs without knowing too much whats going on under the hood (pretty much like
you can drive a car without having to be a mechanic), it can be useful or at least interesting to know at least a bit
about how these filters are actually made. A thorough exploration requires quite a lot of background knowledge in
math and would be beyond the scope of this article, but Ill try to explain the process briefly, and in laymen terms as
much as possible.

There are two basic approaches: the time-based approach and the frequency-based approach. What does this
mean? Well, digital audio is, as you probably know, represented by samples that are being delivered at a certain
sample rate, typically 44.1 KhZ. The time-based approach calculates the output of a filter at any time by simply
calculating a weighted sum of the current sample, a certain number of sample before the current sample and (in
general) a certain number of previous outputs of the filter. This is called time-based since it only looks at how the
samples are coming in one after each other, theres no direct attempt to measure the frequencies that are present in
the signal. Nevertheless, with the correct weighting of samples, its possible to enhance or decrease the contribution
of only a range of frequencies, just like in the filter examples shown in the previous section.

A very simple, intuitive example is a filter that calculates its output by simply summing the current and the previous
sample. You wouldnt expect that theres anything useful coming out of this operation, but actually this is the
simplest form of a low-pass filter! You can convince yourself of this fact when you consider a signal that only
consists of the highest possible frequency that is available for the sample rate at hand (the Nyquist frequency). This
signal consists of the sequence +1, -1, +1, -1, +1, -1 wildly oscillating, as you can see. Now whats the output of
our simple filter? Its always either (+1-1) or (-1+1) depending on the sample position; in any case, its identical to
zero. This means that the Nyquist frequency is completely blocked. On the other hand, if we only have a DC
component in our signal, the sequence of samples would look like this: 1 1 1 1 1 1 1 1 (or any other number
different from 0, depending on the strength of the signal). Clearly, the output of the filter would be 2 2 2 2 2 2 so
2017. 06. 30. Equalization for Rookies Soundbytes Techniques

the DC component is enhanced. All frequencies between are interpolated, which gives the frequency response of a
low pass filter.

The time-based approach is working well in most situations, and theres a whole theory behind it that is also
concerned with how to find the weighted coefficients for the summing of the samples to match a given electric circuit
as closely as possible (which opens the door for simulating classic or not-so-classic pieces of hardware in
software). One issue, however, is that close to the Nyquist frequency, the filter responses often become less ideal
since youre always translating a system with continuous time (your analog filter) to a system with discrete time
(your simulated filter). There are remedies and tricks to avoid this to some extent, but ultimately the cleanest option
is to use frequency-based filters (or FFT filters, named after the technique of Fast Fourier Transformation which is
usually used for this). Briefly, what is done is instead of summing and subtracting the sample values live as they
are floating in, one waits a little while until one has enough samples at hand to perform an analysis of the
frequencies that are contained in the sample set. The nice thing is that there is a mathematical operation (the
Fourier Transform) which, for a number of samples, calculates the contained frequencies (the spectrum of the
signal) but also, for a given spectrum, calculates the samples that produced the spectrum. Theres a one-to-one
correspondence. This means that one can shape and bend the spectrum in any way one wants, and then calculate
the samples that result. So if one wants more gain at around 100 Hz, no problem, just add a peak there, easy
enough. Within certain limits, any desirable spectrum can be generated, which is why FFT based filters are usually
whats being used in free-draw EQs. But also the area around the Nyquist frequency, which is critical with the time
based approach, poses no problem for FFT filters.

There are two draw-backs, however: the FFT operation is taking more CPU power than the time-based approach,
and, since you always need a certain number of samples to get some precision in your frequency analysis, a delay
(latency) is inevitable. This is why FFT filters are usually not considered tracking EQs (of which you place at least
one instance on any track in your project), but they rather belong on the master bus.

Summary
While theres obviously a lot more to say about equalizers and equalization, the material presented here should give
you a good starting point to begin with your own journey into this field. I recommend that you not go into gear-
hunting mode during the first few months, but rather find a cheap or even freeware parametric EQ and try to learn
the basic principles first. Youd be surprised what a good engineer is able to produce already with low budget
plugins. One thing to look for is the option for mid/side EQing, which is a simple yet effective method to treat your
center and side signal separately (especially useful for creating a solid, centered bass and a more airy stereo
field).

In the beginning though, the most important part is to learn how your specific mixing setup sounds (that includes the
speakers and the room). It is definitely advisable to always compare your own mixes to one or more reference
tracks which sound more or less the way you would like to sound, and to switch back and forth between the
reference tracks and your own material frequently. The ear quickly adapts to changes in the frequency spectrum,
and after a few minutes a 10 dB boost at 1 kHz will sound almost natural if you dont have any standard reference to
compare it to. Also, when it comes to frequency analyzers (like the one presented in Fig. 1), you shouldnt use it too
much initially, but rather train your ears first.

Thats about it! Hope you enjoyed the material, and happy mixing!

by Christian Siedschlag, DDMF

2017. 06. 30. Equalization for Rookies Soundbytes Techniques