We present four broad theoretical categories, and several practical studio demonstrations:

A) Filters & Equalizers

B) Interfaces for Filters and Equalizers

C) Circuits for using Processors

D) Introduction to Digital Filters using Waveguides

Q) Review Quiz

E) Practical Studio Demo's and personal studio experiments


A. Filters and Equalizers. Filters and Equalizers process sound in the frequency domain, and as a result they are used to modify the spectrum of the sound, that is, its frequency content, and hence its timbre. If you are unfamiliar with the concepts and representations of spectrum, it would be good to review the second Vibration module.

The main difference between filters and equalizers are that filters only attenuate (i.e. reduce) certain frequencies in the spectrum, whereas equalizers can either boost or attenuate the strength of particular frequency bands of the spectrum. A hybrid form of these models is called a shelf filter, which is somewhat of a misnomer, as it can boost or attenuate all of the high frequencies (a high shelf) or the low frequencies (a low shelf) above or below a certain frequency, respectively.

Another class of equalizer – and clearly the most powerful – is called a parametric equalizer, the term parametric referring to the fact that all parameters of an equalizer are controllable simultaneously. In the analog studio, these were quite complex units, whereas today you are likely to begin with a parametric plug-in which has all of the functions described here available together, so it is best if you understand all of them.

The simplest filters are the high-pass and low-pass filters which can only attenuate frequencies below or above what is called the cut-off frequency. Here we already have a potential point of confusion, so memorize this formula:
a high-pass filter passes the highs and attenuates the lows

a low-pass filter passes the lows and attenuates the highs

Where the confusion arises is that when you want to get rid of some low frequencies, as shown below, you need a high pass filter; keep in mind that the term “pass” refers to not affecting that range and passing those frequencies through unchanged.

At the left we have the high-pass and low-pass filters. They have two variables, the cut-off frequency which is where the signal is attenuated by 3 dB (that is, where the attenuation is regarded as significant), and the roll-off which is the slope of the filter’s response beyond the cut-off. In other words, the term “cut” is not an accurate description of the filter’s action as it implies removing something (true), but in a clean and precise manner (not possible). All filters, whether analog of digital, cannot eliminate, for instance, all frequencies below exactly 100 Hz, which would require a rectangular response.

Instead, all frequencies below the cut-off of a high-pass filter are gradually attenuated according to the slope of the roll-off. Because it can be thought of as a slope, the units are decibels per octave, in other words, it specifies how much attenuation there is with each octave. It might help to think of the slope of a highway, the grade, expressed as a percentage. With a roll-off of 12 dB/oct and a cutoff of 100 Hz (which is attenuated 3 dB by definition), the attenuation at 50 Hz (an octave lower) would be 3 + 12 = 15 dB.

The important distinction is that the larger the roll-off value, the more precisely it distinguishes between the desired frequencies that remain (i.e. are passed) and those which are attenuated. Today, digital filters typically have slopes of 16, 24 or more (e.g. 48 dB/octave) which is more than enough to isolate a frequency band cleanly. The limiting factor to having an extremely steep slope is that it adds phase distortion to the sound, hence the impossibility of having a rectangular cut.

In analog filters, the roll-off is fixed by the circuitry and cannot be changed. In digital filters, the roll-off is calculated as variables in an equation. Therefore, you can only select a desired roll-off and switch to it, as opposed to the cut-off frequency which can be swept up or down continuously at will, an interactive ability that is aurally very effective for hearing the changes in the spectrum.

Also, this type of sweep often produces an aurally interesting way of introducing a sound into a mix, starting with a high cutoff in a high-pass filter and lowering it gradually, or vice versa with a low-pass filter, an alternative to the conventional fade-in. A good digital filter will not have clicks when the cut-off is swept, so be careful with any that produce this kind of artifact.

The bandpass filter, shown above, is the combination (quite literally) of these two filters, the high-pass and low-pass. It is controlled by two cut-offs, low and high, with the distance between them referred to as the passband. Because there are two variables in a bandpass filter (plus the roll-off), a digital application will have to decide if there are two separate controls, and if so, which ones.

One choice that works well is centre frequency and bandwidth (to borrow terms from the Equalizer). If the control surface is a two-dimensional window with an X-Y axis, then this double choice could work well for a mouse moving around the space (for instance, as found in the GRM Tools approach, as shown below).

As briefly mentioned above, the shelf filter is a hybrid between a low-pass (or high-pass) filter and an equalizer. The difference is the lack of a continuous roll-off. All frequencies below the cut-off or turnover frequency in a low shelf filter are either boosted or attenuated (that is, + or - gain in decibels). Once the gain is decided, that is the gain for all frequencies below what is called the stop frequency or shelving frequency.

Something similar happens with a high shelf, except that the gain is for frequencies above the cut-off or turnover frequency. In practice, the shelf filter seems cruder than the bandpass, particularly when it is attenuating. In terms of low frequencies, the high-pass filter will progressively eliminate them, whereas a low shelf filter will merely lower them in intensity. Presumably the difference is whether removing or simply lowering those frequencies is the desired goal. Use of the shelf filter to boost all highs or lows should be done carefully, if at all.

Equalizers (used to EQ a sound) come in many variations, the main one being how many bands are available, the more the better, in general. It is useful to think of an equalizer as a set of filters, where each band has a fixed bandwidth, usually defined in octaves and fractions thereof. However, unlike the filters we've considered, gain can be applied to boost or attenuate each band.

A third-octave bandwidth, meaning 3 separate bands per octave (with a total of between 24 and 30 bands to control) is a standard. When we study the ear’s resolving power for frequencies in a spectrum, called the critical bandwidth (that will be deal with in the second Vibration module), we will find that it is a little less than a quarter of an octave, so the 1/3 octave equalizer comes close to controlling exactly the range of frequencies that we can hear separately in a spectrum.

The diagram below, if taken literally, would not be a very good equalizer as it only has 7 bands to cover a 9-octave range of frequencies, even though they are distributed on a logarithmic frequency scale. So, each band covers over an octave, which might make it easy to use in a car audio system, but it is ill-suited for audio design work. The saving grace of the diagram is that it is easier to see what is going on than, say, with a 24-band equalizer.

The controls on an equalizer, for each band, are the choice of centre frequency and the gain, plus or minus, which is continuously variable up to or down to a maximum, here shown as +/- 12, but more typically +/- 15 or 20. In general it is the “curved” shape of this set of gains that is most effective, rather than the maximum gain. In fact, so much gain can be cumulatively applied with an equalizer that the sound will distort and/or be unpleasant to our ears, particularly if the boost is in the 1-4 kHz range where the ear is most sensitive.

Multi-band equalizer (a) and its frequency response pattern (b)

Parametric Equalizer. A parametric equalizer makes all of its variables controllable, namely:
  • Centre Frequency (CF) in Hz or kHz
  • Gain in + or - dB
  • Bandwidth as the ratio Q where Q = Centre Frequency / Bandwidth
If the last controllable parameter (Q) were actually bandwidth, it would be difficult to use because of the logarithmic nature of frequency. For instance, from 100 to 200 Hz is an octave, and the bandwidth is 100 Hz; the octave from 1 kHz to 2 kHz represents a bandwidth of 1000 Hz. So, if we kept the bandwidth constant at 100 Hz, and swept the centre frequency from 100 Hz to 1 kHz, we’d go from a very large bandwidth to a very narrow one perceptually, with resulting inconsistency in how the result would sound. Admittedly we could keep it constant as a ratio with an interval of, say, 1/3 octave, but that isn’t very easy to specify in general.

Therefore, by creating the unitless ratio of Q, being the ratio between the centre frequency and the bandwidth, we keep the actual bandwidth comparable at all centre frequencies. The usual range of Q is from 1 to 10, or higher in digital versions, which can also be thought of as a range of bandwidths from being equal to the centre frequency to being 1/10 of it for Q = 10.

Narrow bandwidths, with a Q above 5 or 6, may be narrow enough that, when applied to broadband sounds, a spectral pitch will emerge, somewhat similar to a vocal formant which is a narrow resonance region that helps to identify a vowel. The diagram below shows the range of Q from low (i.e. broad bandwidth) to high (i.e. very narrow bandwidth) at different gain levels for clarity.

In general, the Q factor should be judged carefully by ear to being just enough to give the sound more focus and presence, but not so much as to be annoyingly intrusive (since the auditory system is very focused on picking up such resonance regions). This type of boost in the 2-3 kHz region will give speech added presence and clarity, as demonstrated later.

Parametric equalizer frequency response for various values of Q

A useful subset of the parametric equalizer is the notch filter which provides a very narrow attenuation of a specific frequency. The most common use is for eliminating a 60 Hz hum (or 50 Hz in Europe). Similarly, a peak filter offers a single band similar to the parametric model.


B. Interface representations of filters and equalizers.

Low-pass, high-pass and bandpass filters. These include a range of graphic controls with virtual knobs and sliders, and a visual frequency response diagram – which is useful, but don't take the shapes too literally. Some allow the processing shapes to be stored for later use, and most have some kind of bypass function to allow the effect to be turned off and on, which is very useful for comparisons to the original, or in the case of multiple functions being used at the same time, to check the effect of each one separately and as a set. Although this is a limited set that compares 3 and 4 plug-ins, you should be able to find similar features in other ones that you have available.

Many plug-ins offer several filter/EQ functions that are combined in one interface, but despite it being in software, the companies don't bother to change the parameter names on the graphics, so you really need to know your parameters to use them efficiently.

This simple 4-band plug-in has the typical high-pass and low-pass curves as selected on bands 1 & 4. It allows you to sweep the white ball for each cut-off with a fixed roll-off.
The middle knob shows the cut-off frequency and allows control via the knob as a slider.
If the result is weak, the vertical slider at right allows for gain control. Bypass switches at the bottom.

These are the standard ProTools single-band filters.
Note the icons for all the modes at the lower left.
On the right what is actually the roll-off is incorrectly called Q and offers a choice from 6-24 dB/octave or more.
The cut-off frequency can be swept with the knob as a slider or by dragging the white dot.

(click to enlarge)

This is the standard parametric processor in Audition which includes all of the standard processes, including high-pass (HP) and low-pass (LP) at the far left and right respectively.

A choice of cut-offs is offered, plus gain control at the left. Because this is a digital filter, the roll-off can be chosen from 6 dB/oct up to 48 dB/oct. However, note that all frequency changes in real time, such as the cut-offs, will result in clicks as they are moved.

This is the standard bandpass model in GRM Tools, which comes in both a mono and stereo version. Once you know how it works, the stereo version is excellent for treating left and right channels independently which will likely provide an interesting spatial spread to the timbre.
The x-axis is centre frequency and the y-axis is bandwidth, with numerical values at the top, so both low-pass and high-pass functions are neatly combined in a single gesture within the mouse paradigm. The actual values in Hz for the two cut-offs are indicated in small windows to the right of the individual manual controls for them. The other functions are standard for all GRM functions and can be consulted in the documentation.

High and low shelf filters. These are the filters used to boost or attenuate all low or high frequencies past a certain point.

The top diagram shows the low and high shelf filters (left and right) attenuated, and the bottom diagram shows them boosted (probably inadvisable), both in bands 1 and 4. The top two knobs control gain and cut-off frequency respectively. Note the shelving pattern beyond those points and compare it to the high-pass and low-pass filters above.

The top diagram shows the standard ProTools single-band filter in low shelf mode with attenuation, and the bottom diagram shows the high shelf with positive gain.

Note the stirrup-shaped icons for these. The top slider knob controls the steepness of the curve (called the Q); the cut-off frequency and gain are controlled with the other two slider knobs.

(click to enlarge)

This diagram shows Audition’s standard parametric processor in low and high shelf mode together (marked L and H, not to be confused with HP and LP which are nearby).
The lows are being attenuated and highs boosted. This example uses the “gentle slope” switch (above the L and H).

(click to enlarge)

This diagram shows Audition’s standard parametric processor in low and high shelf mode together (marked L and H, not to be confused with HP and LP which are nearby).
The lows are being attenuated and highs boosted. This example uses the “steep slope” switch (above the L and H).

Peak and notch filters. These are subsets of the general parametric model, offering one or two controllable bands.

The top diagram shows two bands (left and right) with a boost and attenuation respectively, with a low Q, i.e. broad bandwidth. The bottom diagram shows them with a high Q, i.e. narrow bandwidth, corresponding to what otherwise might be called peak and notch modes. All three knobs are in use to control (top to bottom) gain, centre frequency, and Q.

The top diagram shows the standard ProTools 1-band filter in peak mode with a boost (that could be changed to attenuation with negative gain) and choice of Q, in this case 1 which is quite broad.
The bottom diagram shows the specialized narrow-band notch filter where the gain is maximized negatively, the Q value determines the bandwidth, and the FREQ slider knob controls the centre frequency. Note the different icons for each of these.

(click to enlarge)

This diagram shows Audition’s standard parametric processor used with two bands simultaneously (marked 2 and 4). The lows are being boosted in the first band and the highs attenuated in the second, but with a low Q of 2, i.e. a broad bandwidth. Note that each band can be turned off and on.

(click to enlarge)

This diagram shows Audition’s standard parametric processor used with two bands simultaneously (marked 2 and 4). The lows are being boosted in the first band and the highs attenuated in the second, but with a high Q of 10 (the max), i.e. a narrow bandwidth. Note that each band can be turned off and on.

Full Parametric Equalizers.  The Audition examples shown above were all taken from its full parametric plug-in, so any of the various processes could be added together. With ProTools, the 7-band parametric could look intimidating, as shown next, despite being colour-coded, so again, it's best to learn each format thoroughly before using it.

Third-octave Equalizers. Unlike the interfaces for analog filters and parametric equalizers, the interface paradigm for the third-octave equalizer has not changed significantly in its migration into the digital domain. Each band of filters is represented by a vertical line where the vertical dimension represents gain above or below a zero point (i.e. where no gain is applied). The bands themselves are identified by the centre frequency based on the international standard of 1 kHz and its octaves and sub-octaves. For a third-octave equalizer, there will be two additional bands between each of those octaves.

Traditional analog 27-band graphic equalizer

Within a “mouse paradigm” where only one value can be adjusted at a time, a serious problem arises for this kind of processor – how to control its many variables efficiently. With the analog version, some skill in using both sets of fingers to control multiple bands simultaneously provided a certain kind of performability for dynamic changes. However, the implied norm for its use was for a fixed setting: once the controls were put in place, they stayed there. The same applies to most plug-ins, but steps can be taken for dynamic changes as discussed below.

We have already noted a significant extension to the “mouse paradigm” with GRM Tools’ Bandpass model where two variables in the x-y plane can be controlled by a the mouse on a small screen. However, unless you have an extremely steady hand to move a mouse smoothly, dynamic changes can sound jerky. GRM Tools solved this problem with the ability to interpolate between presets with a ramp whose speed can be specified in a small box below the presets (and stored with the preset). For filters and equalizers, among others, these smooth ramps (which can be continuously changed over time with multiple presets) offer an improvement over the analog version in terms of performability.

Third-octave equalizer in Audition (click to enlarge)

The GRM Graphic Equalizer follows the standard paradigm with a few differences. The EQ curve can be “drawn” by running the mouse over the surface – but not too quickly – to trace a general curve. Then, each band can be modified by dragging the gain of an individual band up or down (with the corresponding centre frequency and gain indicated at the top). Any configuration can be stored as a preset. However, note that there is no zero line, and in fact the gain could be lowered to zero, unlike an analog equalizer.

Rather unusually for GRM, the default group of presets are not particularly useful, some being quite flat. An unusual configuration for any equalizer is to have alternate bands up and down, or every third one up – the octave configuration. Given the 1/3 oct bandwidth, a spectral pitch will be heard even with noisy sounds, since any of these configurations resembles resonances in an air space. Musically, a third of an octave is a major third, and every second band is an minor sixth apart, so some kinds of tonal sounding “chords” can be created.

Tip: The GRM Equalizer default presets have some of these alternating band combinations, but since the lower gains go to zero, they do not cross-fade very smoothly, as the loudness drops in between presets. Raising the low gains to a higher value will correct this issue.


C. Circuits for using processors. In the analog studio where everything had to be patched (i.e. connected) together, creating a circuit or signal path was absolutely fundamental as well as very flexible and open-ended. Today, most signal paths are hidden or assumed, and at least one, the parallel circuit is much more difficult to create and use. Before we embark below on practical studio experiments, it would be useful to know about some of these types of circuits, where your challenge will be how to create them with your own equipment and software.

1. A direct recording refers to whatever route a signal takes from the microphone to a sound file, where "mixer" might refer to a simple level control. If you are importing a soundfile, then this step is not necessary.

2. A single transformation, or the "insert" version 2a, puts some kind of processor in between the source and its subsequent saved output. In the analog studio there was a subtle distinction about whether the transformation happened before it arrived at the mixer or after. In digital processing, the standard plug-in as illustrated above, is the equivalent to this form of single transformation, and is usually provided on every track.

3. Likewise, if the software allows multiple plug-ins, the assumption is that they are in a series configuration, that is, the output of the first goes into the second, and so on. When using such multiples, it is good to keep in mind that each process must be compatible with the previous one. If the first is a filter that removes low frequencies, for instance, they can’t be part of the processing in the second process.

4. The parallel circuit, in the analog tradition, relied on being able to split the signal into exact copies, that is, multiple versions, with no loss of strength. This was accomplished with a special patch bay wiring called a bridge or “multi” with several connections joined together electrically to avoid signal loss. With one input, all the other connections could be outputs, and the multi could be patched to another for even more outputs. These were then routed to independent processors and back into the mixer to be combined. The beauty and simplicity of this setup was that all signals were heard and processed in real time, and could be mixed in stereo formats at will.

Today, auxiliary circuits can be used on a DAW or mixer for a real-time version of the parallel circuit, but they are more complicated to set up for beginners. Both a real-time and non-realtime version will be described in the studio demo’s below as a kind of submix. All auxiliary circuit, whether analog or digital, allow the signal to be sent to the processing unit either independently of the playback level of the original (called pre, meaning before the playback level), or at a level that depends on the playback level (called post, meaning after the playback level and therefore dependent on it).

5. A feedback circuit is one where the signal is fed back and mixed with the original, but only where there’s enough of a delay involved that it doesn’t immediately go into distortion, as indicated in the second example above where the recording machine playback or a digital delay is used. It can also be called recirculation because a loop has been set up. The other requirement is a sensitive control over the feedback levels, which can increase the signal exponentially; that is, small changes become large ones. Most of the examples of this approach will be described in the modules on Time Delays. However, feedback with filters at modest levels can make them sound more like resonators.


D. Introduction to Digital Filters using Waveguides. The topic of digital filters from an engineering perspective is very complex, involving differential equations in their design, and not all of them will be related to sound design. However, it is still useful to consider some simple examples of what are called 1st and 2nd order filters that can be modelled using a short delay line also known as a waveguide. Basically, the waveguide is a memory array of n samples that are continuously stored and replaced once full. Therefore the contents of the waveguide reflect the most recent values of the waveform.

With these filter algorithms, only the previous sample, referred to as x(n-1), or the second previous sample, x(n-2), are used, so the delay line is very short, and in fact can be implemented with one or two variables that temporarily store these previous values.

The samples themselves are usually multiplied by a gain coefficient, often labelled g, and are combined with the direct signal which goes from the input, x(n), to the output y(n) in these diagrams. The 1st order examples are called that because they use only the previous sample, whereas the 2nd order use the 2nd previous sample in the calculation.

Introducing the concepts of delay lines and waveguides here will prepare the way for their use in longer lines for our presentation of phasing (with short delays) and echo and reverb models (with long delays) in later modules.

These diagrams show four basic filters in terms of their frequency response, equations and delay line circuits, from top to bottom. Here you will see that the two 1st order models describe the familiar low-pass and high-pass filters. The left-hand graphic circuit model shows that the input signal at left goes into a delay line D, gets scaled by .5 and is combined (the + sign) with the direct signal which is also multiplied by .5 to form the output signal at the right.

This is a formal way of saying we are averaging the current sample with the previous sample, since averaging involves adding two numbers together and dividing by two. However, in the frequency domain, we have learned that frequency is the rate of change of phase, with higher frequencies showing a more rapid change of phase. When we average out those rapid peaks, we are essentially filtering them out, hence the low-pass effect.

Also note that the delay for the low-pass filter, D, is 1 sample (i.e. first order), and for the band-reject filter D = 2, the second previous sample, but otherwise the circuit is the same. Although simple, it turns out that these filters are not very useful because the slope of their roll-off is very gentle. Note also that the “zero” of the filter moves from the half sampling rate (Fs) for the first-order filter, to half that with the second-order filter (hence the term band reject).

For the two filters at the right, a similar process occurs, but instead of averaging the adjacent samples, we subtract them from each other. Since low frequencies move slowly in terms of phase, adjacent samples will show little difference in value, and therefore when samples are subtracted, a small value will result. As noted, the 1st order version is a high-pass filter, and the second order is a very simple bandpass filter. Note the location of the zeroes in these filters, as well as their “poles” where the output amplitude is highest. Higher orders in filter design using more scaled previous samples would be needed to improve the roll-off.

Finally, we should note that these filters are called FIR filters, Finite Impulse Response, because they settle to zero quickly, lacking any feedback in the circuit that would prolong them.

However, when we model the resonances on a string (as in the first Vibration module), there is feedback because of the waves are reflected at both ends. Therefore, in the simple Karplus-Strong model of the string, as shown below, there is an averaging function shown as a delay of one sample (engineers refer to the length of the delay line with a z with an exponent of -p where p is the length of the waveguide) in order to average the value. This acts as the simple low-pass filter shown above. Because of the recirculation of the values in the waveguide, even the simplest low-pass averaging filter will be effective because the filtering process happens again and again.

The K-S model feeds its values back into the delay line (whose length corresponds to that of the string), consistent with the standing wave phenomenon that is created in a real string. However, what we are “processing” with a string is the initial energy applied to it, namely the pluck, which is modelled by filling up the waveguide with random numbers, but once filled, adding no more, similar to a single pluck. Note the aural realism of the result including the decay of the sound lengthening with the length of the string/waveguide.

Karplus-Strong delay line resonator

Four plucks of the string model, with the waveguide length doubled each time and the resulting spectrogram

To complete this brief survey of digital filters, we have two more diagrams, the one at left showing an Infinite Impulse Response filter (IIR) which incorporates both the feed forward function that we saw above, that is, adding two previous input samples, x(n-1) and x(n-2), to the direct signal, as well as a feedback function that recirculates two of the previous output samples, y(n-1) and y(n-2). Note the gain values called coefficients are the various a, b values. This feedback function creates the “ringing” behaviour of theoretically infinite repetitions, similar to the various forms of audio feedback we will encounter later. Careful control over the feedback level will affect how long it lasts.

The right hand diagram shows an all-pass filter which means that all frequencies are passed equally, that is, with the same gain, but there are predictable phase shifts in certain frequencies.

Keep in mind that the two most salient points in this topic are that:
  • manipulation of the time domain in terms of delay samples affects the frequency domain (a characteristic of microsound)

  • filters can exhibit resonating behaviour when feedback is involved

Q. Try this review quiz to test your comprehension of the above material (not including Digital Filters), and perhaps to clarify some distinctions you may have missed. Note that a few of the questions may have "best" and "second best" answers to explain these distinctions.


E. Studio Demo's and Personal Projects. With some exceptions, we will not be referencing specific equipment in these demo’s, but for you to replicate them (definitely a good idea), you will need to find equivalent solutions with whatever software you have available. However, we are recommending that you use both a waveform editor with whatever plug-ins it is equipped with for processing, and a DAW (digital audio workstation) for assembling and mixing your files. Some waveform editors, such as Audition, include a small mixing module where multiple tracks can be combined. This can be useful for test mixes and submixes, i.e. where you combine multiple versions of your sounds into a mix that can be bounced into a cumulative file.

a) Demo's using filters and equalizers.

b) Personal studio experiment No. 1.

c) Parallel circuit models demo's.

d) Personal studio experiment No. 2.

a. Using filters and equalizers. In starting a project or experiment, the so-called raw recording you intend to use will likely need to be edited and cleaned up. Some users will prefer to keep the original recording intact, for future reference, whereas others might already delete any extraneous sounds and adjust levels (particularly when the record level is low) with the editor.

After you’ve made those choices, the next stage of clean-up might be to use a filter to get rid of unwanted low frequencies, for instance. Even if you end up using only a subset of the entire source soundfile (see the personal experiments below), it is a good idea to have all of it cleaned up in this manner first, so you won’t have to do it again if you go back to the original (and forget how you cleaned it up).

1. Using a high-pass filter.

Original scything recording with wind noise
Source: WSP Canada 32 take 10

Waveform prior to filtering

Recording processed with a high-pass filter

Waveform after filtering

Click on the link to see the exact high-pass filter settings in Audition. Note that the high-pass filter (HP) has a cut-off frequency of 609 Hz, and that the roll-off is set to the maximum value, 48 dB/octave which cleanly reduces the low frequency portion of the spectrum. Since low frequencies tend to have high amplitude waveforms, the difference in the before and after waveforms are telling. It’s actually unusual to be able to see such processing differences at the waveform level (we will rely on the spectrogram after this), but low-frequency noise is always quite visible. Given that so much energy was removed from the signal, note that a Master Gain of 8 dB was added to the filtering. This could have be done later, but was easy to predict during the processing setup.

The reason this example works so well is that, in terms of the spectrum, the “desired” component of the sound, the scything sound, occupies the region above 500 Hz, and the wind occupies everything below that. Therefore, they can be easily separated. However, if those components overlapped significantly, a compromise would have to be made as to how much to attenuate the desired sound in order to minimize the unwanted component.

Rolling off the low frequencies is also sometimes desirable in the spoken voice. Roughly speaking, the male voice will likely go down to around 100 Hz, and the female voice to 200 Hz. However, room resonances (called eigentones) will boost the low frequencies in a voice, as does a typical amplification of the voice in that space which for the moment we are limiting to medium-sized spaces which are not large enough (or empty enough) to produce significant reverberation.

British Columbia elder Herb George speaking place names in his native language, first filtered below 400 Hz, then opened up fully

Rolling off the lows with a voice recorded indoors will have the effect of making it sound farther away and possibly outdoors, particularly if ambience is added. It is surprising how seldom this simple processing is done in media contexts, particularly radio dramas. The scene is clearly set outdoors, but the voices are clearly studio recorded with noticeable low frequency resonances. Why do listeners (not to mention the producers) not find this illogical?

2) Using an equalizer and bandpass filter.

Original voice recording
(Liora Salter)

Waveform prior to equalizing

Recording processed with a high-pass filter and equalizer

Sound after EQ and editing out the "popping p"

The recording is of a female professor giving a public introduction, and is of medium quality (we’ve all heard much worse). A parametric equalizer in Audition was used to give the voice more presence, first by rolling off the low frequencies with the high-pass filter, with a cut-off of 176 Hz (there is little energy in the female voice below 200 Hz, but the room itself was somewhat boomy, so this was needed). A roll-off of 18 dB/oct was chosen to not attenuate the lows too much (which would have made the voice sound more distant and less resonant).

Then in EQ band 4 there’s a classic EQ boost of 8.6 dB around 2.5 kHz (where significant formant information is located in the vowels, hence the boosted clarity and presence). Note that the Q value is 5 to give it some bandwidth but not too much. Finally a low-pass filter (LP) rolls off the top octave above 10 kHz quite steeply. A modest Master Gain of +8 dB is added. Although the result does not repair all of the faults of the original, it certainly makes it clearer and easier to listen to. However, this EQ pattern is so commonly used, it would make sense to save it as a preset for inevitable future use.

Note that the plosive consonant “p” in the word “department” was very prominent in the original near the start, which is typical of the standard positioning of the mike in front of the speaker’s face (instead of slightly to the side). The air expelled by the plosive consonants, e.g. p, b, t, goes straight into the mic and is heard as a broadband noise, mainly low frequency, but if strong enough, some mid-range energy as well. It is clearly seen in the waveform near the start. As a micro-editing exercise, the so-called “popping p” was also excised from the EQ’d version with the editor, ensuring that no phase discontinuity was added. The result at the bottom right is somewhat more acceptable.

3. Using a strongly limited bandpass filter.

Original voice recording with multiphonics (Yves Candau)

Recording processed with a tight bandpass filter

The recording is of a male singer with a remarkable ability to sing what are called multiphonics. These are harmonics created by adjusting resonances within the vocal tract as controlled by the tongue. The pattern that was used in Audition strongly attenuated the frequencies below 770 Hz and above 4 kHz, leaving the harmonics present that were more than two octaves above the fundamental.

4. Using strong bandpass filter and parametric equalization.

Salvator Mundi bell, Salzburg
Source: WSP Eur 28 take 2

Click to enlarge

The recording is of the very large solo bell known as “Salvator Mundi” from the Salzburg cathedral, weighing 14,000 kg, and rung only on feast days. The bandpass that was used in Audition strongly attenuated the frequencies below 411 Hz and above 2 kHz, leaving the partials present that were more than two octaves above the fundamental. Then – strictly by ear, but also using the spectrogram – all 5 parametric bands were tuned to specific frequencies in the spectrum, giving them boosts from 13.6 dB to 17.7 dB. Note that the Q values are around 12-14, meaning quite narrow bandwidths. Given this amount of boosting, the Master Gain was dropped by 10 dB to prevent overload.

Although this kind of result is aurally interesting by itself, it may also be used in a mix with the original so that a smooth transition between the original and one enhanced in these partials is possible. Also, this harmonic cluster could be treated as secondary source material from which to derive octave transpositions, granular time stretching and convolution based transformations, among others.

b) Personal Studio Experiment No. 1

Based on the examples and information covered so far, you should be able to embark on some interesting experiments to work with actual sounds. The approach we’re taking here is the classic sound object exercise that has proved valuable over the decades as a technique for honing your listening skills, as well as advancing your technical and creative abilities.

It is based on using just a few sounds and maximizing their transformations, and as such it’s a good way to try out a lot of sound processing techniques. But what you’ll probably find out is that the more you manipulate a sound, the more you learn about it, and hopefully at some point along the way, the sound itself will lead you in some creative directions you’ve never imagined.

The process is entirely open-ended, and you should be mainly guided by your ears, but also by relying on what you have learned. Here are some steps that might help you to get started.
1. Select a soundfile that you think would be interesting to work with, and make sure it is cleaned up, editing out extraneous material, correcting any low levels, and filtering out frequencies such as the lows that may not be desirable. Keep in mind that computer speakers, and even headphones, are not the best way to determine the presence of lows. Save the result as your reference soundfile (which we will refer to as the source).

2. If there are some specific parts of your source that are also interesting, highlight those in your editor and save the selection to a new file (or copy and paste). If it has an abrupt start or end, use the amplitude tools your software provides to fade-in the start, and fade-out the end, even if it’s a very short fade. You can also try looping this selection to hear whether it works well in repetition, and make any adjustments to improve the break between the end and start. You can make more such short excerpts, but you’ll probably not have time to explore them all.

3. Start transposing your source or the excerpts up and/or down, depending on which range is strongest in the source. Although you can use any size of pitch transposition, doing it by octaves has several advantages. First, you can quickly get to the “end” of what is going to be usable in either directions. Remember the old saying: you don’t know you’ve gone too far until you’ve gone too far. Secondly, octave combinations always work well aurally, even if there are not actual pitches involved.

Ideally you’ll make two versions of each transposition, one that maintains the same duration, and one that multiplies it by two for each transposition down an octave, or halves it for the upward transpositions, similar to playing a tape at half or double its normal speed. For the downward transpositions, you may need to boost the level each time as it will drop off, particularly towards the lower frequencies.

Very important
: when you “Save as” each version, make sure you give it a file name that reflects the process, e.g. “Bell-1oct” and “Bell-1octSt” where St might be your reminder that it was stretched in length (or compressed for the upper ones). In general, lower transpositions work best and having some very long stretches may also be useful.

A variation on this approach is where you bandpass filter a part of the spectrum of your source, particularly if it is broadband. Listen to the source for differences in texture and character in the upper, middle and lower parts of the spectrum, perhaps with the aid of a bandpass filter. It is often true that a “part” of the sound in the frequency domain may be more useful to you than the entire range. And when these bands of frequency are transposed, the result will also be cleaner and more useful for combining in a mix. At some point you might want to EQ the sound as well, similar to what we did with the bell above.

As you transpose and/or stretch the sound, notice how the character of it changes, or what aural image it creates. Some sounds, like water, tend to stay recognizable as water no matter what transposition you make, probably since there are such a wide variety of water sounds in the soundscape. Others will have a different affect or image in different ranges. It’s also true that when you allow the sound to be stretched, you will notice small sub-events that passed by too quickly before. These might branch off the tree structure you’re creating into a new secondary source for further exploration.

4. At this point you may well have at least a dozen or more files (all well labelled of course) and a much better idea as to whether this sound source still holds your interest. If not, you’re always free to start again with another source. At this point, you’re probably not ready to start a piece, but instead, it would be good to hear some of this material in combination.

Some editors like Audition have the option of creating a session that mixes a small number of files with some basic level controls (usually graphic), maybe a looping function (or simply copy and paste multiple times), and some spatial panning options for Left/Right placement. Think of this step being a “proof of concept” stage, particularly if you can assemble several files together quickly and still maintain control over all of the elements and their levels. Don’t try to go beyond a minute in length for what we’ll call a sub-mix. Use the solo and mute functions in the session to add/subtract tracks and assess their suitability.

It is very important to listen, not just to the individual sounds you’ve created along the way (and you’re not likely to use them all), but to how they work together. If you’re lucky and have good instincts, you may discover a local version of another old saying: the whole is greater than the sum of its parts. This may be because it starts resembling a soundscape in terms of a coherent and balanced mix, or the energy and resulting emotional impact starts to build. Or, you might just find it boring or incoherent. But don’t give up, try to analyze what works and what doesn’t and keep an open mind as to where this all might lead.

c) Using parallel circuits. The series circuit has already been implicitly shown because with the parametric equalizer examples above, you probably noted that the one we used (from Audition) included all of the filter examples (high-pass, low-pass and shelf), along with the peak/notch style of equalizer. This means that when we use filters and some equalizer bands, we are using them as if they were in series.

Of course, nothing prevents the user from using them in an incompatible way – boosting and attenuating the same bands – but this will be reflected in the graphic response diagram. It is also possible to boost the same frequencies by several processes working in the same range of frequency, and conversely to attenuate them more (although given the high roll-offs available, this is seldom necessary). The main point is that all series-style processes are additive – each builds on the other.

The classic analog parallel circuit sends the source to several independent processes and then mixes the results together again. You may like to read a detailed account of a compositional example that featured this kind of processing in a high-level analog studio in France in 1979, as linked here.

In the digital world, we can make a distinction between a synchronous version, where all processed versions of the source are synchronized, and an asynchronous version, where the various soundfiles are placed in an arbitrary temporal position over multiple tracks. In theory, there is a difference between whether the synchronous version is being performed in real time (or “live”) as it always was in the analog tradition, or in non-real-time (which means it is set up as a sub-mix, but in practice the results may be quite similar.

We will first look at a synchronous parallel digital circuit using multiple auxiliary lines, starting with a relatively simple version set up in an older version of ProTools.

(click to enlarge with the zoom tool)

The diagram shows a relatively simple parallel circuit in ProTools, consisting of a stereo original track (of waves on a shore), with four Auxiliary send buses selected (at left) in “pre” mode to make them independent of the original signal, with four Auxiliary returns (in the middle) to mix the results with the original. The four Auxiliary tracks have inserts that each go to the simple 1-band filter/equalizer plug-in. These have been chosen for high-pass, low-pass, a 1 kHz narrow peak, and a 4 kHz narrow peak, respectively. The Auxiliary returns are panned left and right in the mix. The Master Gain is kept at -3 dB to avoid saturation. It would be a good idea to save such a session as a template for future use.

The 6 tracks are grouped into 3 stereo pairs to bring in the original and the processed sounds together in stereo pairs with similar level control. To keep the example simple, all of the levels are shown in “latch” mode which means that all level changes will be recorded during the playback. The mouse control is therefore limited to one of the three stereo sets of tracks at a time, but a simple alternation between them can still be effective.

(Click to enlarge)

If you examine the above graphic of the level controls, you’ll see the classic cross-fade between each pair such that they overlap. Note that an aurally effective cross-fade goes in three stages: (1) bring in the first stereo track (2) bring in the second stereo track to the desired level while leaving the first track level intact (3) fade-out the first track level, leaving the second one intact. This pattern is repeated for all three combinations of auxiliary signals, finally returning to the original. All levels were controlled in a single playback.

This cross-fade pattern, which is a good aural exercise requiring careful manual control if you want to keep it smooth, is the opposite of what might be implied by the symbol for cross-fade, namely X. A literal X pattern, where the levels cross each other in the middle, will leave a gap in loudness level that is undesirable when a smooth result is wanted.

This example is quite subtle in terms of the changes (the transition at 0:20 to the split high and low filters, then at 0:40 to the narrow EQ pair, and then returning by the same route to the original). Something more dramatic could have been achieved by using a multi-band processor so that the processed sound was a combination of a narrow bandpass around the levels that were being boosted, and therefore the middle of the exercise would be just two narrow bands.

Waves processed with a parallel circuit (source: WSP Can 41 take 2)

Of course, individual tracks can have their levels individually controlled by soloing them (or not) and latching only one set of stereo levels at a time. However, that kind of track by track control lacks some of the interaction that is desirable in a parallel circuit. If you are fortunate enough to have a digital mixing interface for the DAW, then you can break free of the “tyranny of the mouse” which only allows you to control one thing at a time. The traditional mixer or its digital version capitalizes on manual dexterity to control many levels at the same time using your fingers, while guided by your ears.

Parallel mixdown of the wave mix

Here is a similar result achieved in a synchronous parallel session in Audition. Again, there are three stereo tracks, the original, the high-pass & low-pass split spectrum, and a more dramatic narrow band filtering with EQ boost, as mentioned above. Audition simplified the process by allowing the left and right channels to be modified separately (by muting them one at a time when applying the process to the unmuted channel) to allow the higher and lower parts of the spectrum to be distributed between the two channels for a broader stereo effect. The session is synchronous because each file starts at the same time.

The session, shown graphically above, had its levels determined by amplitude breakpoints shown as yellow lines, with points of transition determined by command clicks on the line, at which time you see the specific dB level being chosen. Since the waveform is silhouetted behind the yellow line, it was simple to co-ordinate the peak levels in each transition with a particular wave (note the transitions at 0:26, 0:40 and 0:52). Otherwise, classic cross-fades were followed.

d) Personal Studio Experiment No. 2

Based on the above examples, there are many experiments that you could try out, using one or more of the sounds you created in the first set of experiments. Here are some suggestions:
1) Try setting up a parallel session with a DAW that can be saved as a template for future contexts. Experiment with a single stereo sound source and get used to how the various channel groupings and processor settings are going to be controlled. Practice the classic cross-fade when you start latching (i.e. recording) level settings. If your DAW doesn’t allow this, then use breakpoint amplitudes for the same purpose. The main point of the exercise, besides practicing the setup and starting to feel comfortable with the Auxiliary circuit, is to be able to create smooth, dynamic mixes of variations of the same material.

2) If your editor incorporates multi-track sessions as shown for Audition, try to duplicate the DAW model in non-real-time. Otherwise, set up a DAW multi-track session where you import the already processed files. Notice how the process differs and how the way you think about it can evolve. Start with a synchronous version just to keep things simple.

3) Try extending these processes to an asynchronous session, using sounds that have been stretched during the pitch transposition process from Experiment 1. Start by lining up the files to get an additive effect, being careful to control the level of each so they are balanced but don’t oversaturate. Shorter files can be copied one or more times so they still sound while the longer ones are progressing.

If your sound has a sharp attack, you may need to shift the position of the lower pitched versions because of the stretching. Usually this can be done by zooming in on the files and lining them up visually with a vertical cursor line, then excising any extra silence. At the end of each experiment, bounce the result to an interleaved stereo file and check its levels, etc. in an editor.