LAB ELEVEN

REVIEW

Understand how to create panning automation.

Understand how to use stereo files, as opposed to mono, and

understand why you might still want to use mono files.

Understand how to create a soundscape composition in

practical terms, particularly the importance of time,
relationships, and processing.

ADVANCED PROCESSING
There are a number of other processes that you might come across
in audio editors that have not been covered in detail yet.
Within ProTools itself, there are the set of Dynamics II
AudioSuite plug-ins. These include a compressor, a limiter, a gate,
and an expander. The concepts of these processes, which deal
exclusively with amplitude processing, were covered in Unit Five
(Signal Processing). They are rarely used in musique concrte or
soundscape composition; however, they are widely used in
commercial music. The practical applications of these processors
will be left to you to explore.
Often, signal processing within audio programs can be broken
down into five separate types: filter, modulation, delay/reverb,
stereo, and miscellaneous. You should, by now, be very familiar
with filter processes, delay processes, and some of the modulation
processes, such as phasing and flanging. Pitch/Time
transformations should also be familiar.
here.

Some of the more unusual processes will be described briefly

Distortion
Distortion, which is a general term for an incorrect representation
of a signal, has come to specifically mean clipping. Clipping occurs
when a signal is amplified beyond the point the system can handle.

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When a signal becomes larger through gain change and the highest
signal points cannot be accommodated, they are clipped off.

A sine wave (left); the same sine wave amplified to the point of
clipping (right).
Notice in the diagram above that the smooth curve of the sine
wave is now flattened and closer to a square wave. Sine waves, of
course, have no additional harmonics, while square waves have a
lot of harmonics; therefore, the signal to the right will have extra
harmonics contained in the spectrum.
In digital systems, clipping can occur when the sample levels
are raised beyond the current bit rate. In a sixteen-bit system, the
range of numbers used to represent the signal is from 32,765 to
32,767. If a normalized sine wave is amplified by fifty per cent, the
following will result:

Digital clipping: much more dramatic changes.

The difference between the first examplewhich is
comparable to analogue clipping that may occur with guitar
amplifiers and mixersand the second is apparent even in the
visual representation: the much more dramatic change from a
curve to flattened top. This instantaneous change (compare it to the
smoother change in the first example) is essentially a discontinuity.
While both examples will increase the number of harmonics and
thereby make the signal richer, digital clipping will create many
additional frequencies, including signals that are above the Nyquist
frequency, thus causing foldover and aliasing.
(Analogue distortion was, in fact, a desired effect in predigital
days, particularly in the commercial recording studio. Because of
the more limited frequency response of magnetic tape and its lower
signal-to-noise ratio, the extra harmonics of tape distortion made
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the sound much richer and punchier. Early reactions to digital

recording often elicited the response that it sounded cold; in fact, it
was simply the lack of additional frequencies and noise in the
signal caused by analogue distortion, both intentional and not.)
Analogue systems have a certain amount of leeway above
their maximum signal level, called headroom. Although unity gain
on such a system indicates the maximum level of signal that can be
handled without distortion, exceeding this level will begin to clip
the signal but in a much more forgiving fashion than in a digital
system. In fact, most tape recorders have a headroom of 6 to 10 dB.
Digital systems have no headroom; a signal that exceeds the
maximum resolution will simply be held at that level.
Therefore, simply increasing the amplification on a signal
within a computer will not produce Jimi Hendrix type results
because of the unwanted digital artifacts. Distortion algorithms
used in plug-ins (those with names such as Tube/Tape Saturation)
take into account the potential for foldover and actually increase
the sample rate within their processing.
Sonic Decimator
This is a simple digital process that has no analogue equivalent.
These types of plug-ins will change either the sampling rate or the
bit resolution in order to create aliasing and digital noise. The
concept may seem utterly confusing: Why would anyone want to
do this? However, it can achieve a kind of retro sound of early
eight-bit 24 kHz sample rate samplers and create a unique
transformation of material that can be suggestive in itself.
In using both distortion and downsampling, you must
remember that these effects can quickly become tiring. There exists
a certain subset of electronica in which such sounds are the norm
(glitch music); however, remember that not everyone is interested
in listening to limited-frequency bands and noise-based samples.
Formant Shifter
One problem with pitch transposition, with or without time
correction, is the resulting aural suggestion of a change in sound
source. For example, transposing a text higher will suggest the text
was spoken by a smaller head. Transpose it high enough, and the
head will shrink to the size of a chipmunks. This effect will occur
with transposition of any material; however, we are most sensitive
to vocal sounds.

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The reason is that in the case of speech, the headspecifically

the vocal cavityis the main resonating body for the soundwaves
produced by the vocal cords. The vocal cords produce a fixed
waveform of varying frequency; it is the throat, mouth, and tongue
that filter and emphasize certain frequency bands. Try singing a
constant pitch, and moving from the syllable ahh to ooo to
eee, and notice how the vocal cavity changes. Then try singing
the same ahhh, and change the pitch over an octave; notice that
the vocal cavity does not change in this case.
What is happening is that by shaping the vocal cavity in a
certain way, a number of fixed frequency bands will be
emphasized: these are called formants.

Spectrum of the vowel ahh showing three formant regions.

The vertical lines represent harmonics produced by vibration
of the vocal cords.
Because the formants are fixed within an individual, the
resonant frequencies will remain the same, regardless of the
fundamental frequency of the pitch. This is what allows us to
understand singing at different pitch levels. However, not only do
formants vary between individuals, they also vary considerably
between men, women, and children.
Transposition, whether analogue or digital, will also transpose
the formant regions, which explains why we perceive a
transposition of half an octave higher to sound more childlike. With
digital algorithms, however, it is possible to extract the format
regions from a recording, separate them from the underlying
frequencies, and then resynthesize the sound with higher (or lower)
frequencies while retaining the formant regions.

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In using formant shifting, as with any process, subtlety is most

often the key. Attempting to transpose a sound two octaves using a
formant shift will simply not work as intended, although it might
create an interesting effect.
Vocoder
Vocoding is the process of using one signal to create a set of filters
that are applied to a second signal. Analogue vocoders had (and
still have, since analogue vocoders are currently a popular sound in
contemporary commercial music) a series of bandpass filters that
analyze an incoming signal, called the modulator. For example, one
bandpass filter might be set to 1800 to 2000 Hertz. A signal that
passes through this narrow frequency band has its amplitude
continuously measured. This value controls the same frequency
band in the second set of bandpass filters, thereby matching the
amplitude of the incoming signal at that frequency band. Another
signal, called the carrier, is then sent through the second set of
filters, having its spectrum dynamically altered to (loosely) match
that of the first signal. Because the output of the vocoder is
triggered only by the carrier signal, there is no output unless both
carrier and modulator are active.
The more frequency bands there are in the vocoder, the tighter
the correlation between the two signals. However, in digital
systems, the trade-off is a higher computational load; real-time
vocoding is a computationally expensive process.
Certain signals work well with a vocoder. For example, one
common technique is to use a continuous synthesizer chord as the
modulator and a drum as the carrier. This effectively creates a
rhythmic chord triggered by the drum. Another current technique
is to use a synthesizer melody as the carrier doubling a modulating
vocal melody. The complex vocal characteristics modify the much
simpler synthesizer line, giving the effect of a talking synth. This is
the sound used by Chers producers in the song Believe, and it
has been used frequently since then.
Ring Modulation
Modulation is the technique of adding frequencies to a signal; ring
modulation gets its name from a very specific type of analogue
processor (a balanced modulator) that was built in a ring.
Ring modulation involves a carrier input signal and a second
modulating signal. In almost all cases, the modulator is a sine
wave. When the two signals are both active, the result is the sum

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and difference frequencies, called sidebands, of the carrier and

modulator.
For example, using an input sine wave of 400 Hz, and a
modulating sine wave of 25 Hz, the result would be a signal
combining the sidebands 375 Hz and 425 Hz. Note that the original
carrier frequency of 400 Hz is not present in the final signal. Using
a complex carrier such as the voice in ring modulation will produce
sidebands around every harmonic.
Although harmonic textures are possible using ring
modulation, most often the result is an inharmonic, bell-like or
metallic quality. The reason is that the sidebands are no longer in an
integer relationship to one another. In the first example, the 400 Hz
input became a combination of 375 and 425 Hz: a non-harmonic
relationship. Imagine the complexity of sidebands when a complex
sound has its harmonics replaced by inharmonic partials.
Most ring modulator plug-ins will have a balance control
between the source and the output of the process. Full output may
produce a harsh, unrecognizable sound, whereas slight (ten to
twenty per cent) output will give the sound a slight metallic sheen.
Lastly, choosing the modulating frequency intelligently will
greatly enhance the result. A modulating frequency in the normal
audio range (405,000 Hz) will produce a metallic sound
reminiscent of the Cylons in Battlestar Gallactica (see http://www.
battlestargalactica.com/). However, very low frequencies (110 Hz)
will produce interesting vibrato effects (since beating will result in
all the sidebands being only a few Hertz apart).
Another interesting technique is to use an extremely high
modulating frequency (14 kHz and higher). This method will create
sidebands in the extreme frequency range with a transposition and
inversion of the original spectrum.

The source is a harmonic sound with a 100 Hz fundamental.

Using a modulator of 15,000 Hz, sidebands are spaced at
equal distances on either side of the modulating frequency.
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ADVANCED CONCEPTS
As noted several times, multitrack programs like ProTools and
Audition are modeled after the traditional analogue studio
paradigm of a multitrack tape recorder, mixer, and external effects
processors.
Although they offer a high level of control over processes that
were formerly physical (such as moving a slider, which is now
replicated by the creation of an automation envelope), many
concepts in this virtual studio require an understanding of (if not
experience with) the traditional studio.

AUXILIARY SENDS AND DYNAMIC PROCESSING

So far, our use of processing has been static, in the sense that we
complete the process (for example, removing partials via filtering),
then place the processed file into a track. This is limiting in two
ways: first, the parameter settings remain the same for the entire
process (for example, the cut-off frequency); second, we cannot
create any relationship between the process and any other sounds
in other tracks.
Ideally, what we need is dynamic processing: processing in
which settings can change during the process. This is possible
through the use of inserts and auxiliary sends.

INSERTS
The simplest way to create real-time dynamic processing is through
inserts. These are simply a way of inserting a process (or
processor) in a track. Taken from the traditional mixer paradigm,
processes such as equalization could be inserted in a particular
track to alter only that one track. An example might be to add
equalization to a bass drum, or compression to a vocal. Because
each insert uses one process, you had to decide carefully when to
use inserts, since there were a limited number of processors
physically available (in a twenty-four-track mixer, you wouldnt be
able to have twenty-four inserts, since you probably didnt have
twenty-four different processors!). Similarly, in multitrack audio
programs, an insert takes up one plug-in; while you can use another
version of that plug-in elsewhere (since you are not limited to a
single usage of a plug-in), each version of a plug-in does take up
computer processing power.

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One additional characteristic of inserts is that they function in

real timeyou do not have to process the result to a new file; it is
calculated while playing. The benefit of this is that it allows us to
alter the parameter settings of the process in relation to other
tracks. For example, a drum track on its own may use only a little
reverb, but once other tracks are added, the minimal reverb is lost,
and more might be required. If you added the reverb in as a
process, and saved the new file (through destructive editing), then
you would have to return to the original and reprocess the file.
Using inserts allows you to change one parameter!

AUXILIARY SENDS
Certain processes such as equalization work very well as inserts.
However, other processes are best applied globally to several, if not
all, tracks. One such process is reverberation. While it is
conceivable that a separate reverb setting plug-in could be inserted
into every single track, it would be difficult to work with (since you
would have to adjust each reverb setting separately), and it would
also be extremely microprocessor intensive (since reverbs are
computationally expensive).
Instead, we want to use a single reverb process and send all
tracks to that process. Because only one process/plug-in is being
used, it is easy to control and not computationally expensive.
This working method is currently desirable and, like many
methods in the virtual mixer, has its model within the traditional
analogue studio. Auxiliary sends are one of the most complicated
aspects of a traditional mixer, and in order to understand a virtual
version, one needs to understand its genesis in the traditional
studio.
First, you need to understand how a mixer works. Below is the
ProTools Mixer window, with four tracks and a master fader (the
concept is the same in most audio programs). This models a fourtrack mixer with four inputs (from a multitrack tape recorder, for
example).
The volume sliders control how much signal is sent from each
track to the main output, and the master fader controls how much
of the combined signals get sent out of the mixer. All tracks send
their signals to the same outputthis signal path is called the main
signal bus. A bus is simply a signal path in the mixer.

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A four-track mixer: two mono& one stereo input, stereo output.

Most mixers have additional buses that can route signals to
destinations other than the main output. These are called the Auxiliary Buses; there are usually two to six auxiliary buses on a mixer.
They work exactly the same way as the main signal bus: there
is a potentiometer that controls how much signal is sent from each
track to that particular auxiliary bus and an auxiliary output that
controls how much of the combined signal on that bus gets sent out
of the mixer.
(Note that the ProTools Mix Window does not display the
individual auxiliary send pots to show how much signal is being
sent from each track to the bus, nor does it show the cumulative
output level of the auxiliary bus.)

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A four-track mixer with an auxiliary bus.

The output from the auxiliary bus is then sent to a processor,
such as a reverb unit. The output from the reverb is then returned
to the mixer in either the auxiliary returns (which get sent directly
to the master out) or additional input channels.
This configuration allows you to control the level of the
original signals (via the volume sliders); the amount that each
track/channel sends to the processor (via the auxiliary send for that
track); the amount of total signal that is sent to the processor (via
the auxiliary output); and the amount of the processed signal that is
sent to the main output (via either the auxiliary returns or the
additional input tracks/channels).
Creating and configuring Auxiliary Buses is discussed in the
individual software appendices.

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TO DO THIS WEEK
Start experimenting with the advanced features of your software.
These features will tend to make it easier to do things, or, to explore
more sophisticated processes and ideas.
Investigate other signal processing techniques. Perhaps look
for different plug-ins on the Internet and see if there are any demo
versions that you can try. Or, try using an audio editor and
exploring either the advanced built-in processing or the advanced
plug-ins, such as VST. The latter plug-in type has many free and
demo versions available online.
Try out the concept of inserts and auxiliary sends. While they
are complicated, both are extremely powerful, especially when
combined with the real-time automation.

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