Chapter 1.

Terms
Table/Loop An audio clip repeated over and over again. In serum, one table consists of 256 sub-
tables.
Frequency Number of loops occurring per sec. High frequency=high pitch and low freq=low pitch.
Measured in Hertz.
Cent Measurement of musical intervals. 100 cent= 1 semitone
Fundamental The tuned frequency+(almost always) the lowest frequency that is present in the
sounding of a pitch
Harmonics/tone/timbre Sound DNA. All sounds has harmonics(cept sinewave) pitched overtones
that float above the fundamental frequency (the pitch of the note). With sounds that are termed
musical (guitar, piano or violin etc) these harmonics will be mathematically related to the fundamental
frequency. Each numbered harmonic, therefore, is a direct multiple of the fundamental frequency.
Overtones The harmonics of an instruments sound minus the fundamental frequency.
Oscillator A device generating a tone in a synthesizer
LFO Low-Frequency Oscillator-an oscillator puts out a low inaudible signal (1 Hz~10Hz) used for a
control signal.
Distortion Usually undesirable result of overloading sound equipment. Reducing the levels can
remedy the situation.
Harmonic distortion is the introduction of extra harmonics that are musically related to those
already present, resulting in a change in timbre.There is conflict on even and odd harmonics. Trust your
ear on this one instead of following rigid rules. There is no rules only tools. Usually it is said that Even-
order harmonic distortion tends to sound musically sympathetic, smooth, and bright in a constructive
way. Odd-order harmonic distortion tends to sound rough or harsh, gritty or edgy, and is often
associated with added 'richness' and 'depth'."
Beat The action of two audio signals mixing together causing regular rises &.falls in volume.
Dry+wet Dry simply means without any effects. Wet means with effects. Dry is the input signal, Wet
is the output (effected) signal, a Wet/Dry knob controls the mix of the two.

Chapter 2. Table Oscillator

1.Table Oscillator Anatomy
A table is a short audio clip played repeatedly in loop.
Rate at which loop is played back creates frequency (pitch) of
note. Fast playback rate make more high pitch/frequency
sounds. Imagine a spring and squeezing it to make the
wave(spring) more hi freq.
Tone (harmonics/ timbre)
of sound based on content in
waveform.
One table consist of 256 sub-
tables.
Activated sub-t is lit in
yellow.
Deactivated sub-t is lit in
green.
Inbetween sub-t is lit in gray. Like
motion tween- smooth inbetweens from
one table to another. Auto-generated not
actually designed.
Only 1/256 sub-t can be activated,
therefore heard at a time. Use WT Pos
knob to animate(automate) thru sub-
tables and choose which sub-table to
activate.
2. Oscillators
Sine wave Consists of fundamental frequency alone, no harmonics. Not suitable for sole
use in a subtractive sense, because if the fundamental is removed no sound is produced
(and there are no harmonics upon which the modifiers could act). Used independently for
sub-basses+whistling timbre.Mix with other waveforms to add extra body or bottom end to
a sound.

Square wave Produce only odd harmonics resulting in mellow, hollow sound. Good for
emulating wind instruments,adding width to strings and pads, or for creation of deep, wide
bass sounds.

Pulse wave Not square waves cuz width of the high and low states is adjusted, varying
harmonic content of the sound. Reductions in the width produce thin reed-like timbres
compared to wide, hollow sounds of square wave.
Sawtooth wave produces even and odd harmonics in series and therefore produces a
bright sound thats good for brassy, raspy sounds, gritty, bright sounds of leads and raspy
basses. Because of its harmonic richness, it is often employed in sounds that will be filter
swept.

Triangle wave The triangle wave shape is not as harmonically rich as a sawtooth wave
since it only contains odd harmonics. Ideally, this type of waveform is mixed with a sine,
square or pulse wave to add a sparkling or bright effect to a sound and is often employed on
pads to give them a glittery feel.

Noise wave Create random mixture of all frequencies rather than actual tones. White
Noise is like radio static. Pink noise produce heavier, deeper hiss. Noise good for making
percussive
sounds, snares and handclaps. Can also be used for simulating wind or sea effects, for
producing breath effects in wind instrument timbres or for producing the typical trance leads.

Creating complex waves Individual oscillators sound bad, to create interesting sounds, a number of
oscillators should be mixed together and used with modulation. First mix different oscillator waveforms together,
then detuning them(by odd rather than even numbers because detuning by an even number introduces further
harmonic content that may mirror the harmonics already provided by the oscillators, causing the already present
harmonics to be summed together.)
There is a limit to the level that oscillators can be detuned from one another. Oscillators should be detuned so that
they beat (The action of two audio signals mixing together causing regular rises &.falls in volume) ,but if the
freq/pitch of these beats is increased by any more than 20 hz the oscillators separate, resulting in two noticeably
different sounds. This can sometimes be used to good effect if the two oscillators are to be mixed with a timbre
from another synthesizer because the additional timbre can help to fuse the two separate oscillators. As a general
rule of thumb, its unusual to detune an oscillator by more than an octave.
Additional frequencies can also be added into a signal using ring modulation and sync controls. Oscillator sync,
usually found within the oscillator section of a synthesizer, allows a number of oscillators cycles to be synced to
one another. Usually all oscillators are synced to the first oscillators cycle; hence, no matter where in the cycle any
other oscillator is, when the fi rst starts its cycle again the others are forced to begin again too. For example, if two
oscillators are used, with both set to a sawtooth wave and detuned by 5 cents (one-hundredth of a tone), every
time the fi rst oscillator restarts its cycle so too will the second, regardless of the position in its own cycle. This
tends to produce a timbre with no harmonics and can be ideal for creating big, bold leads. Furthermore, if the fi rst
oscillator is unchanged and pitch bend is applied to the second to speed up or slow its cycle, screaming lead
sounds typical of the Chemical Brothers are created as a consequence of the second oscillator fighting against the
syncing with the first.

3.Table Osc Module

Power button: Top-left power button mute 1 oscillator (turn gray) free CPU.
Display: Click screen toggle between 2d+3d view mode. 2d show lit subtable wave. 3d show all
subtables.
Display Osc Pen: On top-right of display is pen icon. Click opens table edit.
Table Picker Click table name(---above)to show pop-up list of table presets. Click < > to move.
Save Floppy-disk icon to the left of table name. Please save own tables to the User folder, dont
overwrite the factory tables (or presets may sound different). When in doubt, always pick a new
name. By the way, Serum will always save user-made or user-edited wavetable data inside a
preset (your song) unless the wavetable is Factory.
Pitch Controls (0 | 0 | 0 | ----) Left>right: Octave, Semitones, and Fine (cents), and Coarse.
Unison Num box determines how many Unison voices are used. This lets you play multiple
notes(CPU consume)
of the same pitch, but slightly detuned. Classic magic number for unison is 7. Too much makes
cloudy sound.
Unison Detune usewhen unison set above 1. Detunes unison voices.
Unison Bland Use when unison set above 2. Level(volume) offset of the unison voices vs central
unison voice or voices.
WT Pos Choose 1/256 subtable to activate. Yellow activated, green and gray non activated. Gray
in-between frames.
Warp Menu Pick a warp to modify table. Knob will set the depth. Put display in 2d view ee how
warp mode affects table.
Phase Knob Determines where oscillator will begin playing back when note is triggered. May not
notice a difference changing this - especially if random knob is up high (because Random will be
altering the phase on each new note.
Random Knob Change the Phase Knob's value by a random amount for each new voice. Reasons
for using this:
1)to provide a different start or click/thump to each note. 2)To reduce or remove the Laser Zap effect when unison (slightly
detuned) notes are triggered.
3)To provide a random tone to each note (when layering multiple oscillators), as the phase cancellation between the oscillators
will vary with each new note.

4.Making Wavetables Balance between consistency *and* contrast. Too much

consistency/repetition is boring/constant. Too much contrast and it is difficult to draw connection between random
events .
The two Categories of wavetable types A) many tables, but correlated Typically a sound which has a
lot of frames to it, but they all fluidly work together- think a pluck string which decays across many cycles. There
might be a lot of individual cycles, but they all go together well / relate to each other a lot. Another example would
be a synthesizer sample of oscillator sync. You would want many cycles so it doesn't feel too 'steppy', however all
of the tables feel like part of a collection (a similarity / sweep across them).
B) few tables. Most of the factory tables in popular wavetable synths are only made up of 4-5 cycles. This allows for
some variety within the table, but not too many waveforms that it feels disconnected/crazy.
Table ordering It typically makes sense to have the tables progress from dull to bright, or vice-versa. You may
have a situation where you want 'peak' spectrum in the middle somewhere, but probably not. You can drag the
thumbnails at the bottom of the waveform editor to re-arrange them. This way when you move the WT Pos, it feels
like you are traveling in a straight line, instead of some zig-zag fashion, spectrally speaking.
Interpolation If we have 4 tables and automate (e.g. move knob) on WT Pos, we'll hear 4 discrete tones and it
will immediately jump from one to the next. This is typically undesirable. In the Editor you can select Morph->Table
Morph (Crossfade). This is what many wavetable synths do automatically, in every preset wavetable. However
there are times when you may prefer the discrete (non-interpolated) waveforms.
Making tables from 'scratch' There are several different ways to 'make' source audio for the WaveTable:
draw. Grid size in lower-right determines snapping. Try different sizes such as 6 or 12 to bring out +7 and +19
harmonics. draw in FFT bins at top. ctrl-click in bins for a popup menu with more options (random etc). move
drawing up to FFT bins (Curvy arrow on top left) and then add a new table ( > button in lower left). Then make
some adjustments and repeat. Again, about 4 tables tends to be a popular number of frames in many other
software synthesizers.

Chapter 3. Noise
Oscillator
Noise Osc Controls
Noise Menu Click name(AC hum 1 left)to show pop-up list of table presets. Click < > to
move.
OneShot/Looping Arrow icon. Deactivated=loop plays forever, looping start to end. When
Enabled, the sample stops when playback reaches the end of the sound. Looping is the
default and typical-use for noise sounds, but one-shot is useful for attack sounds, for instance
to add a percussive punch/attack transient to a sound.
Keytrack Switch A mini piano icon below one shot arrow icon. When enabled, puts the
Pitch knob into a different state (snaps to semitones). Usually not needed since pitch of noise
doesnt usually follow music notes.
Phase Knob Same concept as the Phase Knob for the oscillator. Determines where oscillator
will begin playing back when note is triggered. Knob can be automated/adjusted which results
in a sort of lo-fi scratching effect.
Random Knob Same concept as the Random Knob on the oscillators. Raising the Random
will prevent the noise from being identical each time you press a note (holding a chord is a
good example of where you would likely want to use Random, otherwise the exact noise
sound will exist for all 3 notes).
Pitch Knob Determines the base pitch/frequency for the noise oscillator. The default (50%)
is nominal pitch, that is the original pitch of the sound file (assuming key track switch is off).
Pan Knob the placement in the stereo(left and right channels) field for the noise oscillator
(Left-to-right).
Level Knob The output volume of the Noise Oscillator.

Chapter 4. Filter Module

Filter Type Click name or < > to change filter type. ([***] = for best result for Flange filters,
set mix knob at 50%)
Cutoff sets the primary cutoff frequency for the filter, with just a couple exceptions (such as
vowel for the formant filters).
Key-track switch Piano icon allows the cutoff to be offset from MIDI notes
(in most filter types, one octave of MIDI is precisely one octave of filter frequency control)
Res Change height(gain) of narrow band of frequencies, near the cutoff level, where the
sound is amplified.
Stereo At 50% (default) knob has no effect. When turned to left, Left channel cutoff will
increase, and the Right channel cutoff will decrease. When turned to right past 50%, Left
cutoff will decrease and the Right channel cutoff will increase
Drive The Drive knob increases the height(gain) of filter and can impart some mild distortion
to the sound.
Var The Var knob has a different function depending on the filter type chosen.
Mix The Mix knob allows for a wet/dry amount for the filter. The default(100%) means 100%
wet.
Level Hidden under the Mix knob . Basically adjusts volume of filter. Click on the Mix name
to change Mix into level.
[***] = for best result for Flanges menu items, set mix knob at 50%

Chapter 5. Filter theory(DMM)

Filters sculpt(remove parts) of oscillator signal to create desired sound.

1.Low-pass filter Can remove all other freqs except

fundamental. (Fundamental:The tuned frequency and the lowest
frequency that is present in the sounding of a pitch by an
instrument.) Why it might be useful to do this:
1. Using a variable filter on a bright sound allows you to determine
the colour of the sound.
2. In a struck piano string, Initial bright sound get duller as it dies
away. This effect can be made by opening the filter as the note
starts and then gradually sweeping the cut-off frequency down to
create the effect of the note dying away.

When cut-off point chosen, small

amt of harmonics that lie above pt
arent removed completely and are
instead reduced by a degree. The
degree of reduction depends on
transition band( the steepness of
curve/slope from which the hill drops
down to bottom of hill, if filter is
visualized as a hill) of the filter used.
The transition band matters since it
defines the sound of a filter. Steep
slope=sharp filter gradual=Soft filter.

2.Hi
-
pass filter
Opposite to lo-pass, first removing the low freqs from the
sound and gradually moving towards the highest.
This is less useful than the low-pass filter because it
effectively removes the fundamental frequency of the
sound, leaving only the fizzy harmonic overtones.
Because of this, high-pass filters are rarely used in the creation of instruments and are
predominantly used to create effervescent sound effects or bright timbres that can be laid over
the top of another low-pass sound to increase the harmonic content.
The typical euphoric trance leads are a good example of this, as they are often created
from a tone with the fundamental overlaid with numerous other tones that have been
created using a high-pass filter. This prevents the timbre from becoming too muddy as
a consequence of stacking together fundamental frequencies. In both remixing and dance
music, its commonplace to run a high-pass filter over an entire mix to eliminate the lower
frequencies, creating an effect similar to a transistor radio or a telephone.
Reduce cut-off control, gradually or immediately to morph track from a thin sound to a fatter
one, which can produce a dramatic effect in the right context.

3.Band-pass filter
 Band-pass filters, like high-pass filters, are often
used to create timbres consisting of fizzy harmonics.
They can also be used to determine the frequency
content of a waveform, as by sweeping through the
frequencies each individual harmonic can be heard.
Because this type of filter frequently removes the
fundamental, it is often used as the basis of sound effects
or lo-fi and trip-hop timbres or to create very thin
sounds that will form the basis of sound effects. Can
be used to thin a sound. Similar to notch filters which can
be used for a similar purpose.
4.Band-reject(notch) filter
Attenuate a selected range of freqs creating a notch in
the sound and usually leave the fundamental unaffected.
This type of filter is handy for scooping out frequencies,
thinning out a sound while leaving the fundamental intact,
making them useful for creating timbres that contain a
discernible pitch but do not have a high level of harmonic
content.
5.Comb filter
Some of the samples entering the filter are delayed in
time and the output is then fed back into the filter to be
reprocessed to produce the results, effectively creating a
comb appearance, hence the name.
Using this method, sounds can be tuned to amplify or
reduce specific harmonics based on the length of the delay and
the sample rate, making it useful for creating complex
sounding timbres that can-not be accomplished any other way.
As an example, if a 1kHz signal is put through the filter with a 1ms delay, the signal
will result in phase because 1ms is coincident with the inputted signal,equalling one.
However, if a 500Hz signal with a 1ms delay were used instead, it would be half of
the period length and so would be shifted out of phase by 180, resulting in a
zero. Its this constructive and deconstructive period that creates the continual
bump then dip in harmonics, resulting in a comb-like appearance when
represented graphically, as in Figure 1.17. This method applies to all frequencies,
with integer multiples of 1kHz producing ones and odd multiples of 500Hz (1.5, 2.5,
3.5kHz etc.) producing zeros.
The effect of using this filter can at best be described as highly resonant, and forms
the basis of flanger effects; therefore, its use is
commonly limited to sound design rather than
the more basic sound sculpting.

6.*Resonance/peak
One final element of sound manipulation in a synthesizers filter section is the
resonance control.
This refers to amount of output of filter that is fed back directly into the input,
emphasizing any frequencies that are situated around the cut-off frequency.
This has a similar effect to employing a band-pass filter at the cut-off point,
effectively creating a peak.
Although this also affects the filters transition period, it is more noticeable at the
actual cut-off frequency than anywhere else. Indeed, as you sweep through the cut-
off range the resonance follows the curve, continually peaking at the cut-off
point.
In terms of the final sound, increasing resonance makes the filter sound more
dramatic and is particularly effective when used in conjunction with low-pass filter
sweeps.
On many analogue and DSP-analogue-modelled synthesizers, if the resonance is
turned up high enough it will feed back on itself. As more and more of the signal is
fed back, the signal is exaggerated until the filter breaks into self-oscillation.
This produces a sine wave with a freq equal to that of the set cut-off point and is
often a purer sine wave than that produced by the oscs.
Because of this, self-oscillating filters are commonly used to create
deep,powerful sub-basses that are particularly suited to the drumn bass and
rap genres. Notably, some filters may also feature a saturation parameter which
essentially overdrives the filters.
If applied heavily, this(saturation) can be used to create distortion effects, but
more often its used to thicken out timbres and add even more harmonics and partials
to the signal to create rich sounding leads or basses. The keyboards pitch can also
be closely related to the action of the filters,using a method known as pitch
tracking, keyboard scaling or key follow . On many synthesizers the depth of
this parameter is adjustable, allowing you to determine how much or how little
the filter should follow the pitch.
When this parameter is set to its neutral state (neither negative nor positive), as a
note is played on the keyboard the cut-off frequency tracks the pitch and each note is
subjected to the same level of filtering. If this is used on a low-pass filter, for example,
the filter setting remains fixed, so as progressively higher notes are played fewer
and fewer harmonics will be present in the sound, making the timbre of the
higher notes mellower than that of the lower notes. If the key follow parameter is
set to positive, the higher notes will have a higher cut-off frequency and the
high notes will remain bright ( Figure 1.19). If, on the other hand, the key follow
parameter is set to negative, the higher notes will lower the cutoff frequency,
making the high notes even mellower than when key follow is set to its neutral state.
Key follow is useful for recreating real instruments such as brass, where the higher
notes are often mellower than the lower notes,and is also useful on complex bass lines
that jump over an octave, adding further variation to a rhythm.

Chapter 6. Effect theory(DMM)

 1.Compression
Basics Compression restrict the range of volume/dynamics
to set threshold, making the volume more even (Difference in volume
between the loudest+quietest parts of audio are reduced ), and overall
louder(We determine overall volume of music from the average)
Change the overall character of timbre, often resulting in warmer, smoother+rounder tone which is
good and typical in edm. Helps recordings of real sounds fit better with other instruments by restricting
dynamic range (A edm track that uses a real bass guitar will have a fairly wide dynamic range, even if it was compressed
during the recording stage. This will cause problems within a mix because if the volume is adjusted so that the loudest parts fit
well within the mix, the quieter parts may disappear behind other instrumentation.By using compression more heavily on this
sound during the mixing stage, the dynamic range can be restricted, allowing the sound to sit better overall within the final mix.).
No hard and fast rules for settings, just trust your ear and know what each knob can do so you can
achieve the sound you imagine in your head! (while there are numerous publications stating the typical compression
settings to use, the truth is that there are no generic settings for any particular genre of music and its use depends entirely on
the timbres that have been used. For instance, a kick drum sample from a record will require a different approach than a kick
drum constructed in a synthesizer, while if sampled from a CD, synthesizer or film different approaches are required again.)
Visualize compression this way: Imagine putting a vibrating object in a small box(compressor), so all the vibrations that
exceed the the threshold(height of box) are confined to the max dimension of box(threshold).By routing the source sound
through a compressor and then into the recorder, you can set a threshold on the compressor so that any sounds that exceed this
are automatically pulled down in volume.

Threshold Sets the level where compressor will begin squashing input signal. Measured in dB. In
usual recording situation threshold is set so that average signal level always lies just below threshold,
and if any exuberant parts exceed it, compressor with reduce gain.
Ratio Amount of gain reduction on a sound exceeding the threshold. Ratio of Signals exceeding
threshold :levels that come out.
If 4:1, every time signal exceeds threshold by 4dB, the signal will be squashed so only 1dB
increase at the output of the compressor. For 6:1, every time when threshold exceeded by
6dB, theres an 1dB increase at output.
When the compressor is lowering the signal to set ratio it is lowering the entire signal, so that the
compressor is still working even after the input signal has fallen below the threshold, for an
amount of time determined by the release.
Highest ratio of :1 is often known as 'limiting'. It is commonly achieved
using a ratio of 60:1, and effectively denotes that any signal above the
threshold is brought down to the threshold level (except briefly after a
sudden increase in input loudness, known as an attack).
Ifa low-frequency waveform, such as a sustained bass note is
squashed, the compressor may treat the positive and negative
states of the waveform as different signals and continually activate
and deactivate. The result of this is an unpleasant distortion of
waveform. To prevent this from occurring, compressors will feature
atk+ release parameters.
Make-Up Gain If youve set the threshold, ratio, attack and release correctly, the
compressor should compress effectively and reduce the dynamics in a sound but this
compression will also reduce the overall gain by the amount set by the ratio control.
Therefore, whenever compression takes place you can use the make-up gain to bring the
signal back up to its pre-compressed volume level.
Attack/Release
The 'attack phase' is period when the compressor is
decreasing gain to reach level determined by the
ratio.
The 'release phase' is the period when the
compressor
is increasing gain to the level determined by the ratio,
or, to zero dB, once the level has fallen below the
threshold.
Atk and release controls are labeled in milliseconds(ms). This is the amount of time it takes for the gain to
change a set amount of dB, often 10 dB. For example, if the compressor's time constants are referenced to 10 dB,
and the attack time is set to 1 ms, it takes 1 ms for the gain to decrease by 10 dB, and 2 ms to decrease by 20 dB.
The envelope created is like ADSR envelope only instead of controlling vol, its amount of decrease in gain.
Control how quickly the volume is pulled down and how long it takes to rise back to its nominal level after the
signal has fallen below the threshold.
In other words, the attack parameter defines how long the compressor takes to reach maximum gain reduction
while the release parameter determines how long the compressor will wait after the signal has dropped below the
threshold before processing stops.
This raises the obvious question that if atk is set so that it doesnt clamp down on the initial attack of the source
sound, it could introduce distortion/ clipping before the compressor activates. While this is true, in practice very
short, sharp signals do not always overload an analogue recorder since these usually have enough headroom to let
small transients through without introducing any unwanted artefacts. This isnt the case with digital recorders,
though, and any signals that are beyond the limit of digital can result in clipping(Clipping: Distortion of a signal by
its being chopped off. An overload problem caused by pushing an amplifier beyond its capabilities.)

2.Standard Compression
As a generalized starting point, its advisable
to set the ratio at 4:1 and lower the threshold so
that the gain reduction meter reads between 8
and 10dB on the loudest parts of the signal.
After this, the attack parameter should be set to
the fastest speed possible and the release set to
approximately 500ms. Using these as preliminary
settings, they can then adjusted further to suit
any particular sound.
Its advisable that compression is applied
sparingly during the recording stage because
once applied it cannot be removed. Any
exuberant parts of the performance should be
prevented from forcing the recorders meters into
the red while also ensuring that the compressor
is as transparent as possible. Solid-state
compressors are more transparent than their valve counterparts and so are better suited for this purpose.
As a general rule of thumb, the higher the dynamic range of the instrument being recorded the higher the ratio
and the lower the threshold settings need to be. These settings help to keep the varying dynamics under tighter
control.and prevent too much fluctuation throughout the performance. Additionally, if the choice between hard or
soft knee is available, the structure of the timbre should be taken into account. To retain a sharp, bright attack
stage, hard knee compression with an attack setting that allows the initial transient to sneak through unmolested
should be used, provided of course that the transient is unlikely to bypass the compression. In these instances, and
to capture a more natural sound, soft knee compression should be used.
Finally, the release period should be set as short as possible but not so short that the effect is noticeable when
the compressor stops processing. After setting the release at 500ms, the time should be continually reduced until
the processing is noticeable and then increased slowly until it isnt.
Some compressors feature an automode for the release that uses a fast release on transient hits and a slower
time for smaller peaks, making this task easier. The settings shown in Table 2.1are naturally only starting points
and too much compression should be avoided during the recording stage, something that can only be
accomplished by setting both the ratio and the threshold control carefully. This involves setting the compressor to
squash audio but ensuring that it stops processing and that the gain reduction meter drops to 0dB (i.e. no signal is
being compressed) during any silent passages.
As a more practical example, with a simple four to the floor kick running through the compressor and the ratio
and threshold controls set so that the gain reduction reads 8dB on each kick, its necessary to ensure that the
gain reduction meter returns to 0dB during any silent periods. If it doesnt, then the loop is being overcompressed.
If the gain reduction only drops to 2dB during the silence between kicks, then it makes sense that only 6dB of gain
reduction is actually being applied. This means that every time the compressor activates it has to jump from 0 to
8dB, when in reality it only needs to jump in by 6 dB. This additional 2dB of gain will distort the transient that
follows the silence, making it necessary for the gain reduction to be adjusted accordingly.

3.DIGITAL DELAY
To the dance music producer, digital delay (often referred to as digital delay line DDL) is one of the most
important effects to own as if used creatively it can be one of the most versatile. The simplest units will allow you
to delay the incoming audio signal by a predetermined time which is commonly referred to in milliseconds or
sometimes in note values. The number of delays produced by the unit is often referred to as the feedback, so by
increasing the feedback setting you can produce more than one repeat from a single sound. This works by sending
some of the delayed output back into the effects input so that its delayed again, and again, and again and so
forth. Obviously this means that if the feedback is set to a very high value the level of the repeats end up
collecting together rather than gradually dying away until eventually youll end up with a horrible howling sound.
While all delay units will work on this basic premise, the more advanced units may permit you to delay the left
and right channels individually and pan them to the left and right of the stereo image. They may also allow you to
pitch shift the subsequent delays, employ fi lters to adjust the harmonic content of the delays, distort or add reverb
to the results and apply LFO modulation to the fi lter. Of all these additional controls (most of which should require
no explanation) the modulation is perhaps the most creative application to have on a delay unit as it allows you to
modulate the fi lters cut-off or pitch, or both, with an LFO. The number and type of waveforms on offer vary from
unit to unit but fundamentally most will feature at least a sine, square and triangle wave. Similar to a synthesizers
LFO parameters, they will feature rate and depth controls allowing you to adjust how fast and by how much it
should modulate the fi lter cut-off or pitch parameters.
One of the most common uses for a delay in dance music is not necessarily to add a series of delays to an audio
signal but to create an effect known as granular delay. As we touched upon in Chapter 3, we cannot perceive
individual sounds if they are less than 30ms apart,which is the principle behind granular synthesis. However, if a
sound is sent to a delay unit and the delay time is set to less than 30ms and combined with a low feedback setting,
the subsequent delays collect together in a short period of time which we cannot perceive as a delay. The resulting
effect is that the delayed timbre appears much bigger, wider and upfront. This technique is often employed on
leads or, in some cases, basses if the track is based around a powerful driving bass line.

4. EQ EQ is a frequency-specific volume control tone control that allows you to intensify/ attenuate specific frequencies.
3 controls needed:
A frequency control allowing you to home in on the frequency you want to adjust.
A Q control allowing you to determine how many frequencies either side of the centre frequency you want to
adjust.
A gain control to allow you to attenuate or intensify the selected frequencies. Notably not all EQ units will offer
this amount of control and some units will have a fixed frequency or a fixed Q, meaning that you can only adjust
the volume of the frequencies that are preset by the manufacturer. EQ plays a much larger role in mixing than it
does in sound design, so this has only been a quick introduction and well look much more deeply into its effects
when we cover mixing and mastering in later chapters.

5. DISTORTION Distortion is pretty much self-explanatory; it introduces an overdrive

effect to any sounds that are fed through it. However, while the basic premise is quite simple, it has
many more uses than to simply grunge up a clean audio signal. As touched upon in Chapter 3, a sine
wave does not contain any harmonics except for the fundamental frequency, and therefore applying
effects such as flangers, phasers or filters will have very little influence. However, if distortion were
applied to the sine wave it would introduce a series of harmonics into the signal giving the
aforementioned effects something more substantial to work with.

6. Reverb Reverb is used to describe the natural reflections weve come to expect from
listening to sounds in different environments. We already know that when something produces a
sound, the resulting changes in air pressure emanate out in all directions but only a proportion of
this reaches our ears directly. The rest rebounds off nearby objects and walls before reaching our
ears; thus, it makes common sense that these reflected waves would take longer to reach your
ears than the direct sound itself. This creates a series of discrete echoes that are all closely
compacted together and from this our brains can decipher a staggering amount of information
about the surroundings. There should be little need to describe all the differing effects of reverb
because youll have experienced them all yourself. If you were blindfolded, you would still be able
to determine what type of room youre in from the sonic reflections.
Ultimately, while compression is the most important processor, reverb is the most important effect
because samplers and synthesizers do not generate natural reverberations until the resulting
signals are exposed to air. So, in order to create some depth in a mix you often need to add it
artificially. For example, the kick may need to be at the front of a mix but any pads could sit in the
background. Simply reducing the volume of the pads may make them disappear into the mix, but
by applying a light smear of reverb you could fool the brain into believing that the sound is further
away from the drums because of the reverberation thats surrounding it.
Ratio (Sometimes Labelled as Mix) The ratio controls the ratio of direct sound to the amount of
reverberation applied. If you increase the ratio to near maximum, there will be more reverb than
direct sound, while if you decrease it significantly, there will be more direct sound than reverb.
Using this, you can make sounds appear further away or closer to you.
Pre-Delay Time After a sound occurs, the time separation between the direct sound and the first
reflection to reach your ears is referred to as the pre-delay. This parameter on a reverb unit allows
you to specify the amount of time between the start of the unaffected sound and the beginning of
the first sonic reflection. In a practical sense, by using a long pre-delay setting the attack of the
instrument can pull through before the subsequent reflections appear. This can be vital in
preventing the reflections from washing over the transient of instruments, forcing them towards
the back of a mix or muddying the sound.
HF and LF Damping The further reflections have to travel the less high frequency content they
will have since the surrounding air will absorb them. Additionally, soft furnishings will also absorb
higher frequencies, so by reducing the high frequency content (and reducing the decay time) you
can give the impression that the sound is in a small enclosed area or has soft furnishings.
Alternatively, by increasing the decay time and removing smaller amounts of the high frequency
content you can make the sound source appear further away. Further, by increasing the lower
frequency damping you can emulate a large open space. For instance, while singing in a large
cavernous area there will be a low end rumble with the reflections but not as much high-frequency
energy. Despite the amount of controls a reverb unit may offer, it is also important to note that
units from different manufacturers will sound very different to one another as each manufacturer
will use different algorithms to simulate the effect. Although it is quintessential to use a good
reverb unit (such as a Lexicon hardware unit or the TC Native Plug-ins included on the CD), its not
uncommon to use two or three different models of reverb in one mix.
7.CHORUS
Chorus emulate the sound of two or more of the same instruments playing the same parts
simultaneously. The result is a series of phase cancellations. This is analogous to two synthesizer
waveforms slightly detuned and playing simultaneously together; there will be a series of phase
cancellations as the two frequencies move in and out of phase with one another. A chorus unit
achieves this same effect by delaying the incoming audio signal slightly in time while also
dynamically changing the time delay and amplitude as the sound continues.
Phase cancellation occurs when two signals of the same frequency are out of phase with each
other resulting in a net reduction in the overall level of the combined signal. If two identical signals
are 100% or 180 degrees out of phase they will completely cancel one another if combined. When
similar complex signals (such as the left and right channel of a stereo music program) are
combined phase cancellation will cause some frequencies to be cut, while others may end up
being boosted.
To provide control over this modulation a typical chorus effect will offer three parameters all of
which can be directly related to the LFO parameters on a typical synthesizer. The first allows you
to select a modulation waveform that will be used to modulate the pitch of the delayed signal,
while the second and third parameters allow you to set the modulation rate (referred to as the
frequency) and the depth of the chorus effect (often referred to as delay). However, it should be
noted that because the modulation rate stays at a constant depth, rate and waveform it doesnt
produce the authentic results you would experience with real instrumentalists. Nevertheless, it
has become a useful effect in its own right and can often be employed to make oscillators and
timbre appear thicker, wider and much more substantial.

8. PHASERS AND FLANGERS

Phasers and flangers are very similar effects with subtle differences in how they are created, but
work on a principle comparable to the chorus effect. Originally phasing was produced by using two tape
machines that played slightly out of sync with one another. As you can probably imagine, this created
an irregularity between the two machines, which resulted in the phase relationship of the audio being
slightly different, in effect producing a hollow, phase-shifted sound.
This idea was developed further by applying pressure onto the flange (the metal circle that the tape
is wound upon) of the second tape machine. This effectively slowed down the speed of the second
machine and produced a more delayed swirling effect due not only to the phase differences but also to
the speed. With digital effect units, both these work by mixing the original incoming signal with a
delayed version but also by feeding some of the output back into the input. The only difference
between the two is that fl angers use a time delay circuit to produce the effect while a phaser uses a
phase shift circuit.
Nevertheless, both use an LFO to modulate either the phase shifting of the phaser or the time delay
of the fl anger. This creates a series of phase cancellations since the original and delayed signals are
out of phase with one another. The resulting effect is that phasers produce a series of notches in the
audio fi le that are harmonically related (since they are related to the phase of the original audio signal)
while fl angers have a constantly different frequency because they use a time delay circuit.
Consequently both fl angers and phasers share the same parameters. They both feature a rate
parameter to control the speed of the LFO effect along with a feedback control to set how deeply the
LFO affects the audio. Notably, some phasers will only use a sine wave as a modulation source but most
fl angers will allow you to not only change the shape but also control the number of delays used to
process the original signal. Today, both these effects have become a staple in the production of dance,
especially House, with the likes of Daft Punk using them on just about every record theyve ever
produced.

Chapter 7. Effects
1.Basics
clickFX at the top of the window to access effects. The 10 fx modules on left can be re-
ordered by dragging.
Enable fx button by clicking power button on left. It will light up upon click and module will
appear on right.
All audio output from the oscillators+filters route to the FX module (and then to the master
volume, and output. The signal flow is top-to-bottom through the FX. )

2.Common controls to each FX module

MIX: Most effect has Mix knob on far right. Sets wet/dry balance of signal (knob at 0=100%
dry, knob at full= 100% wet). (Dry:without fx. Wet:with fx. Dry:input signal, Wet: output
(effected) signal, a Wet/Dry knob controls the mix of the two.)
Bypass: At right edge of FX module is a power button. Temporary mutes a given FX. To
permanently deactivate an effect it is recommended to use the very leftmost Enable fx
buttons, because these will also hide the module from view.
3.Modulating FX parameters You can also modulate most FX parameters.
It is worth noting however The effects are like plugin inserts after Serum. Therefore when playing
polyphonic synth parts (strummed chord, for example) you should keep in mind that automating FX
controls with per-voice mod sources (such as an envelope) will result in the effect parameter
modulation will be getting modulated/re-triggered.

4. Individual Effects modules explained

Reverb
Size: Sets room size (reverb time + dimension)
PreDelay: Amount of time in milliseconds before reverb occurs. Using predelay allows you to
give the impression that a sound is close to you but in a large room, separate your transient
from the reverb (A transient is a high amplitude, short-duration sound at the beginning of a
waveform that occurs in phenomena such as musical sounds, noises or speech. ) or create a
delay-like echo.
Low Cut: Suppress low frequencies from reverb. 0% = no effect on low freq, 100% = no lows
in reverb at all
Hi Cut: Suppress high frequencies from reverb. 0% = no effect on high freq, 100% = no high
frequencies in reverb at all
(Cut: 1.To reduce gain of a particular band of frequencies with an equalizer. 2.To not pass a
particular band of frequencies said of a filter)
Damp: An additional high-frequency cut for the reverb. The knob controls how fast this high-
frequency attenuation occurs. 0% = no damp, 100% = maximum damping. (Increasing the
damping produces a more "muted" effect. The reverberation does not build up as much, and
the high frequencies decay faster than the low frequencies. Simulates the absorption of high
frequencies in the reverberation.)
Width: Expand or collapse stereo(2 channel) width of reverb. (Width is another term for
Depth, the amt of change in the controlled signal by the control signal ) 100% =maximum
width. (Sets the apparent "width" of the Reverb effect for stereo tracks only. Increasing this
value applies more variation between left and right channels, creating a more "spacious"
effect. When set at zero, the effect is applied independently to left and right channels. )

EQ
Two 3-state switches: in middle of the EQ section allow you to set the type for each of the two
bands.
Left band: is intended for low-frequency (LF) adjustment and allows for Low Shelf, Peaking, or hi
pass filtering.
Right band: is intended for high-frequency (HF)adjustment and allows for High Shelf, Peaking, or
Lowpass filtering.
Freq (L):set the frequency in Hz for the low EQ band. Freq (R):set the frequency in Hz for
the high EQ band.
Q (L):set the Q (resonance) for the low EQ band. Q (R):set the Q (resonance) for the high
EQ band.
(Resonance: narrow band of frequencies, near the cutoff level, where the sound is amplified. )
Gain (L):set gain boost/cut in dB for the low EQ band. This knob has no effect if you selected Hipass
as your low EQ band type.
Gain (R):set gain boost/cut in dB for high EQ band. This knob has no effect if you chose Lowpass as
your high EQ band type.
(Gain: The amount of increase in audio signal strength, often expressed in dB. )

Distortion 13 different distortion types, including 2 dual-waveshaper modes which allow you to
create own custom distortion.
Mode: Click on distortion name and < > to navigate
Off/Pre/Post Switch: This switch enables the filtering, which can be set to either Pre-distortion or
Post-distortion.
When enabled you'll notice the filter display will turn orange showing filter is active.
F (frequency):Number-box under filter display sets cutoff frequency for filter.
Q (resonance):Sets resonance for filter. Allows for high values,for squelchy feedback. Typical use is a lower
setting.
(Resonance: narrow band of frequencies, near the cutoff level, where the sound is amplified. )
LP/BP/HP:Sets the filter type for the distortion filter. Allows for morph between lowpass-to-bandpass, or
bandpass-to-hipass.
Drive: Essentially the Amount control; typically the amount of gain boost for the distortion. The exceptions
are...
*on Downsample effect type, the Drive knob controls the amount of sample rate reduction.
*On the X-Shaper and X-Shaper (Asym) modes, the drive knob is a 'morph' between the the two waveshapes.

Flanger
BPM Sync: Syncs flangers sweep to host BPM. When on, the rate knob snaps to beat. When off, the
rate knob is in Hz.
Rate: Control rate of flangers sweep.When BPM on, the rate knob snaps to beat. When BPM off, the
rate knob is in Hz.
Depth:How much flanger changes input sound, how much flange fx is taking place.
Feed:Feedback amount of the flanger circuit, makes the effect more pronounced (ringing).
Phase:Stereo(having 2 channels) phase offset(The time interval by which one wave leads or lags another.)for
the LFO influence over the flanger (Basically, Left flange and right flange offset).
0% = both L and R have the same frequency.
50% = 180 degrees - L and R have opposite frequency,in other words the flanger sweep will be rising on the left while falling
on the right, or vice versa.

Phaser
BPM Sync: Syncs phasers sweep to host BPM. When on, the rate knob snaps to beat. When off, the
rate knob is in Hz.
Rate: Control rate of phasers sweep.When BPM on, the rate knob snaps to beat. When BPM off, the
rate knob is in Hz.
Depth:How much the Phaser's LFO changes input sound, in other words how much/deep the phase
effect is taking place.
Freq:Sets the base frequency for phaser effect.
Feed:The feedback amount of the phaser circuit, makes the effect more pronounced (ringing).
Phase:Stereo(having 2 channels)phase offset(The time interval by which one wave leads or lags another.)for
the LFO influence over the phaser (Basically, Left flange and Right flange offset).
0% = both L and R have the same frequency.
50% = 180 degrees - L and R have opposite frequency,in other words the phaser sweep will be rising on the left while falling
on the right, or vice versa.

Chorus The chorus module is a 4-voice chorus effect, with 2 L and 2 R chorus taps.
BPM Sync:Syncs chorus's modulation rate to host BPM. When on, the rate knob snaps to beat. When
off, the rate knob is in Hz.
Rate:Control rate of chorus's modulation rate.When BPM on, the rate knob snaps to beat. When BPM
off, the rate knob is in Hz.
Delay 1:Sets the amount of delay time in milliseconds between the dry signal and the first stereo
pair of chorus voices.
Delay 2:Sets the amount of delay time in milliseconds between the dry signal and the second pair of
chorus voices.
Depth:Sets how much the delay times above are modulated by Chorus's LFO. In other words, how
much pitch warble occurs.
Feed:Feedback amount of chorus voices, in other words how much of chorus voice's output appears
back at input of the chorus module. This creates a more pronounced ringing to the chorus.
LPF: Set the cutoff frequency the lowpass filter which is after the chorus wet effect.
Delay
Feed:Sets amount of delayed signal appearing back at input of delay,control how many repeats of the
delay are audible.
BPM Sync Switch:Put delay times in tempo-based units ( note, etc) or in milliseconds.
Link Switch:When enabled, the right-channel delay times will be linked to (kept the same as) the
left channel delay times.
Delay times:2 columns of number boxes, labelledLEFT and RIGHT. The upper box is the base
Delay time for the respective Left or Right audio channel. The lower box is an offset for the above
delay time, for instance a value of 1.1 means that the above delay time will now sound 110% of its
value. You'll also notice when dragging these lower boxes you'll see TRIP(triplet in music theory) and
DOT(dotted value in music theory)appear when you reach 133% (1.333) and 150% (1.5) respectively.
Freq (F):This number box sets the filter cutoff frequency for the delay's filter.
Q (Q):This sets the amount of bandwidth(The range of frequencies affected by an equalization setting.) for
the delay's filter (How much lowpass and hipass are applied. In this case, a maximum value means
minimum filtering / maximum bandwidth).
Mode Select (Normal / Ping-Pong / Tap->Delay):
Normal is a 27 standard stereo delay with independent L and R channels/times.
Ping-Pong, the output of the L delay and the output of the R delay feed into one another.
Tap->Delay, both delays are put in mono, in series: the L signal fires once with no feedback (tap)
followed by the R signal operating as a typical delay (feedback is applied here).

Compressor
Thresh:Sets threshold in dB for the compression to start engaging. 0 = 0 dB = no compression
(unless overloaded input signal). 100% =-120 dB = almost always compressing. Typical use would likely
be somewhere in middle of range, eg. around -12 dB, but it is more dependent on your input signal
strength, and how much compression is desired.
Ratio:sets the strength of the gain reduction. Typical compression is between 2:1 and 4:1. If you set
the compression knob to maximum, you will see Limit appear. This is a completely different DSP
circuit at that point (true peak limiter, no longer a compressor) and therefore the other controls will
behave different when this Limiter is engaged.
Attack:the amount of time in milliseconds for the gain reduction to engage. A longer attack
(slow attack) is useful for letting some signal through before the gain reduction takes place,
resulting in a punch,snap or bite (or other words in your preferred language). A shorter
/fast attack will tame peaks more completely.
Release:The amount of time for the gain reduction to get removed.
Gain:The amount of makeup gain. This is a good way to boost quiet signals. Allows for ~36 dB
of boost so be careful, as that is a large amount of gain! A little can go a long way.
Multiband:Enabling this switch makes the compressor become a multiband up/downwards
compressor. This is an extreme setting but you may find a use for it. The individual bands are not
user-adjustable separately.

Filter
The effect filter operates identical to the per-voice synth filter found on the main OSC tab, except
running as a master effect. Please refer to the chapter titled Filter Module(chapter 6, the previous
chapter) for specifics about the controls.

Hyper The Hyper effect is a micro-delay chorus effect with a variable number of voices (1-7). In
addition the Hyper effect can be configured to re-trigger on every MIDI note, which adds to the
potential simulation of a unison.
Rate:The speed at which the various Hyper voices are oscillating sharp/flat in pitch.
Detune:Sets the amount / depth for the Hyper voice oscillations in pitch.
Retrig:When enabled, the Retrig switch will reset all of the Hyper voices to start over from a zeroed
pitch offset. This provides a laserlike zap effect on each note on. Typical use probably doesn't involve
enabling this switch, but you may decide otherwise, in particular on monophonic patches it may be of
use.
Unison:This number box sets the number of chorus voices make up the Hyper effect.
Mix:The Hyper effect has its own Mix control, independent to the Dimension controls listed below. If
you only wish to use the
Dimension effect and not the Hyper, it is recommended to set the Unison control to 0.

Dimension
The Dimension effect is a pseudo-stereo effect made out of 4 delay lines summed out-of-phase and
slowly amplitude modulated to provide a subtle amount of motion to the effect. This is useful for adding
a perceived of width to an otherwise mono signal.
Size:Sets the amount of delay time for the delays.
Mix:The Dimension has its own Mix control. When set to 0%, the Dimension effect is disabled.

Chapter 8. Envelopes
The Envelope module provides 3 modulation sources.
1st envelope, is special as it is dedicated for
Amp (output volume of each voice), however
you can also connect it to additional parameters
if desired. Clicking the names Env 1, Env2,
Env3 along the top of the envelope will choose
which envelope to show+edit.
Assigning an Envelope to a
parameter
In order for an Env (Env2 or Env3) to
activate, you need to first drag it's
modulation source tile to a destination
parameter. [note: Env1 is always active because it always controls the volume of the note]. To do
this, drag the Env 2 tile to a knob, such as Cutoff on the filter (be sure to enable the Filter
module at the top of Filter). Now lower the sustain knob below the envelope in lower-left and press
a note. You should hear the sound become more filtered as the note is held down.
Env View modes There are two different zoom display options for the envelope. A lock
button in the top-right corner of the Env determines which mode:
Locked: The Envelope always fills the display area. In other words zoom is set automatically so
the envelope will fit perfectly in the display. In other words adjusting envelope times will simply
change how much time is represented (you'll see the background time ruler change scaling).
Unlocked: dragging in the zoom area below the lock will determine the zoom level for the
envelope display.

Chapter 9. Modulation Routing

Serum offers 16 Modulation Matrix slots in a patch, which allows for up to 16 destination
parameters to be modified and/or scaled by up to 32 modulation sources.
Drag-and-drop Routing / Assignments
Serum contains a drag-and-drop method of routing for common connections.
For instance, to assign LFO1 to OSCA Pan, click+drag the Tile name LFO 1 to the
pan knob in the Osc-A panel located nearby (diagonally up to the left about an inch
from the LFO 1 tile.
As you are dragging, you should notice the LFO 1 tile is hovering with the mouse
cursor, and the mouse cursor will get a + sign as you hover over the pan knob.
This + is indicating that you are over a valid mod destination. When you let go of
the mouse, the connection will automatically be made, and the LFO1 is now affecting
OSC-A's stereo pan position.
You'll also notice a number 1 will appear on the right side of the LFO 1 tile. This
is indicating that the LFO 1 has one destination. Hover mouse to see where a mod
source is used. You can hover the mouse over the LFO 1 tile, and it will display the
assigned destination(s) (A Pan in this example).

Hover mouse to see where a mod source is used can You

hover the mouse over the LFO 1 tile, and it will display the assigned destination(s) (A
Pan in this example).

Modulator depth control

A blue halo will appear around the Pan knob. This is indicating the LFO1 Depth on
the pan knob (in other words how much influence the LFO has over the panning
position). An Up/Down arrow will also appear in blue to the left of the knob. You can
click+drag on this blue arrow to change the modulation depth amount (tip: you can
also alt click+drag on the knob to have the same function as clicking on this arrow,
the arrow simply saves you from having to touch the computer keyboard).
The miniature arc in the top-left of the knob allows you to adjust this depth amount
by clicking the mouse and dragging. You will notice a grey arc as well on controls
which have a modulator assigned, but not the currently selected mod source. To
adjust the modulator depth in that case, you must either A) first click the associated
mod source tile, or B) visit the Mod Matrix window and adjust the depth from there.

Negativity
Ifyou lower this modulation amount beyond zero, you will notice the halo around the
knob becomes a slightly different hue. This color is simply indicating that there is a
negative depth amount, in other words the value is inverted (as the LFO output goes
up, the influence to the pan knob goes down)

Context-Menu on controls
If you Right-click on a control such as a knob, a context-menu will appear. With the
following choices:
Bypass Modulator: Bypass/cut off/mute the connection between this tile and the modsource tile.
After making this menu selection, you will see the 'halo' around the knob turn grey, indicating that the
modulation connection is bypassed. You can un-bypass this connection by selecting the menu item
again (uncheck).
Remove Modulator: Similar to above, except it removes the connection between the modulation
source and this knob completely.
Remove All Modulators: Removes all connections to this knob from all Modulation Sources.
Reset Control: Resets this knob/control to its default value this is the same as CTRL-click
(windows).
Mod Source (submenu): Displays all of the possible modulation sources to control this parameter.
This allows another way
to make a connection to a control without having to drag from the ModSource tile or visiting the Matrix
window.
MIDI Learn: Selecting this menu item puts Serum in a MIDI Learn Mode, where it waits for an
incoming MIDI CC value. Once it receives a MIDI CC, the MIDI Learn is deactivated and the CC# is now
assigned to this control. This assignment is saved with the preset/patch. Revisiting this pop-up context
menu again will display the CC# and give you the ability to either re-learn it to a different CC#, or else
remove it.

Mod Matrix Window

Mod matrix is a list view, which allows you to select the routing and amounts for the
various modulation connections. Unique to Serum is the fact that the Mod Matrix and
drag-and-drop style of modulation routing both exist. Drag a mod source to a knob,
and this routing will appear in the matrix, or vice-versa. This gives you additional
flexibility in viewing/creating/altering mod assignments.
Mod Matrix: Sources
Some mod sources only exist in the Mod Matrix (no 'drag source' tile):
Aftertouch (aka Channel Pressure)
Chaos 1 and 2 (Chaos controls are on the Global tab).
 Note-On Random 1 and 2 (two separate random numbers generated on a note on, in case you
have a need for 2 different random values on each note on)
Noise Osc (the Noise Oscillator's output can be used as a modulation source, try pitching the
Noise Osc low for more of a chaos LFO type of effect)
Pitch Bend
Fixed (not really a mod source, but you can use this to allow a modulation assignment to get a
'fixed' value with slider depth control, should you want that for some reason). only exist in the Mod
Matrix Source menu (Src). Therefore you will use the MATRIX window if you wish to use these
mod sources. Furthermore the MATRIX window is used for assigning the secondary
source (Mod Src) and/or it's Mod type (Mod) and its amount (Mod Amt).
Mod Matrix: Amount Slider
The Amount slider is bi-directional. As mentioned in the above heading Negativity, the slider to
the left of middle will have a Negative amount value, in other words the mod source's output will
be inverted before heading to its destination.
As with other controls Serum, Command-Click (OSX) / CTRL-Click (Windows) will reset the value
to default (Zero in this case).
Mod Matrix: Destinations
A pop-up menu for selecting the mod destination (the parameter which you wish to have effected
by the mod source).
Mod Matrix: Type
This determines if the modulation is unidirectional (displayed as an arrow pointing to the right: -> )
or bi-directional ( displayed as arrows pointing both left and right: <->). You can achieve similar
sonic results with either, it depends on whether you prefer the destination control (think Knob
position) be at the beginning (Unidirectional) or the middle (bi-directional). When creating
modulations with the drag-and-drop method mentioned at the beginning of this chapter, the
default Type assigned depends on whether or not the control is centered. For instance, if you drag
to a pan knob which is centered, Serum assumes you wish your Mod Source to pan both left and
right, so bidirectional is chosen. If you drag to a control which is not centered, such as Filter
Resonance, Serum assumes you want the modulation to add value only, and Unidirectional is
used.
Tip: You can change the Type between Uni/Bi without visiting the Mod Matrix window, by shift-alt-
clicking on the knob with a visible (blue/yellow) modulation assignment.
Mod Matrix: Mod
The Mod drop-down gives you some bonus ways to manipulate modulation assignments.
* Default. The two modulation sources are
multiplied together, so that one is scaling
the other. For instance, if LFO1 is the Src,
and ModWheel is the Aux Src, then we will
not hear the LFO1 influence unless the
ModWheel is raised above zero (and the Aux
Amt slider is also above zero).
*(inv Same as the above, except the secondary
) source (Aux Src) is value-inverted. For
instance in the above example, there would
be no modulation if the ModWheel was at
maximum, and there would be full
modulation if the ModWheel is at minimum.
byp This row of the modulation matrix now has
ass no effect.

Chapter 10.
LFO
4 LFOs, each contain an entire set of
independent controls. You select which LFO
graph to view or edit by clicking the
Modulator Source tiles (LFO 1, LFO 2,
LFO 3, LFO 4) directly above LFO Graph.
LFO Graph Display
The graph area is where you draw the shape
for the LFO. Here are the mouse/hotkey
functions for graph editing:
double-click: Add/Remove points
shift-click: draw steps at the Grid Size
(step sequencer)
alt-click+drag Points: snap points to the Grid Size
click+drag on background: multi-select points
alt-click+drag any Curve Point: move all curve points at once.
ctrl-click+drag a point: multi-select points for a relative movement (rainbow color
show on points, dragging them in this state will make closer points move more and further points
will move less).
 right-click: bring up a pop-up menu for additional features, such as setting the segment
'shape' for shift-click, deleting all multi-selected points, or assigning the start or loopback points.
shift-control-click on a point when in Env mode : set this point as the
Loopback position (or the very last point if you wish no Loopback position). TSimply a shortcut to
avoid the menu. No need to remember this one since could use menu instead.
The LFO Controls The LFO module is in the bottom-center of Serum window. These are its
controls:
Grid Size Number box allows set grid size of LFO graph. See visual grid background of the LFO
graph change while adjusting this number. The Grid exists to be used in conjunction with the Alt-
click (snap points) or the Shift-clicking (draw step segments) on the LFO graph.
Load/Save menu This folder icon will display a pop-up menu when clicked. There are many
presets included which will overwrite the visible LFO graph. The bottom option of this menu is
Save LFO Shape.. which will allow you to save your own LFO Shape for future use.

LFO Mode (Trig / Env / Off) Allows you to select how the LFO behaves to new notes:
Trig: re-triggers LFO. LFO will start with new note. Useful for when you want the LFO to always
behave the same timing-wise with new notes.
Env: similar to Trig, except the LFO will stop once it reaches the right-edge of the graph (plays
once, like an envelope). It is possible to loop a segment of the LFO, for instance add a point to the
middle of the LFOGraph and shift-command-click on it (shift-ctrl-click on windows) and you'll see
an orange marker appear above the point. Now when you play a new note and hold it down,
assuming you have routed this LFO to a destination, you'll see+hear it looping this second half
of the LFO graph (from the marker to the right edge).
Off: Note on's are essentially ignored, the LFO will run mindlessly on its own like an analogue
LFO on a synthesizer would. This is useful for creating a more free-form modulation which ignores
new notes, and/or one which is always synced to your
song's time position.
BPM switch Enabling this will make the time
value snap to tempo-based units ( note, 1/8th note,
etc).
Dot/Trip switches(BPM switch must be
enabled)Allow for triplet and dotted times on the rate
control, respectively. These switches are useful for
avoiding triplet or dotted times when automating the
LFO Rate, that is, for avoiding doted and triplet times
when you know you want an evenly beat-divisible
time(s).
Anchor switch(BPM switch must be
enabled)This determines whether or not the LFO's
playback position will jump if you automate the time
control: if Anchor is enabled, then the time is
anchored to your song time, in other words changing
the tempo from note to 1 bar means the time
position may jump in order to have the playback properly affixed to the bar cycle. Conversely, if
the Anchor is disabled, the play position will not jump when you change the LFO Rate, but now
may feel rhythmically off-the-beat. This is how a traditional LFO behaves.
Rate the playback speed of the LFO. Said another way, this determines the amount of time
represented on the LFO graph area. The LFO Rate is in beat-synced units by default (BPM switch
on) but can also be made free to a Hz value (BPM switch off).
Rise this is the amount of time taken for the LFO graph shape to have influence over the LFO
output. In other words, the LFO will begin with a fixed output (imagine the LFOGraph is a flat
horizontal line, with whatever value the left-most point of the LFOTool graph contains) and slowly
(based on the Rise time) becoming the shape of the visible graph. This knob is useful for getting
the LFO to slowly have influence over your sound.
Delay Amount of time before rise begins. LFO will have a fixed output and after the delay time
period, the rise will begin. Smooth this smooths the LFO output. This is useful for avoiding
abrupt jumps in the LFO output, without having to draw ramps on every segment of the LFO
graph.
*Protip: lt-click+drag one of the LFO 1, LFO 2, etc tiles to another, to copy all LFO settings
from one LFO to another.

Chapter 11. Voicing &

Portamento
Voicing Settings
In the lower-right corner of Serum you'll find the Voicing section. These controls allow you to
change how Serum behaves to multiple notes playing at once.
Mono (Monophonic) Switch
When enabled, serum will only allow one active note at-a-time. If a new note is pressed while one
is playing, old note ends.
Legato Switch
Legato is only audible with Mono switch is enabled. When a mono voice is interrupted, the state of
the Legato switch determines whether or not the Envelopes/LFO's will re-trigger. When Legato is
enabled, the envelopes do not re-trigger, which allows for a smooth change to the new note which
is pressed. However sometimes you want the envelopes to re-trigger so each note has the same
definition, in these situations you'd want Legato to be set to off.
Poly (Polyphony) number box
Sets the number of simultaneous notes which can be played. Sometimes, often for CPU reasons,
you would want to place a limit on the simultaneous notes. For instance if you were sending a
flurry of notes (arpeggio) to a patch which had a long release (say 10 seconds) this could end up
being a whole lot of voices, were there no limit (and your CPU would probably overload). Typically
speaking 8 is enough and 16 is generally a bunch. This box is greyed out (as pictured) when the
Mono switch is enabled.
Poly (Polyphony) count display
Underneath this number-box is a numeric readout. In the picture above you'll notice that it says
0 / 1 in this display. This means 0 out of a total possible number of 1 voices is playing. If I held a
note it would display 1 / 1. If I enabled a 2nd oscillator, it would display 0/2 when no notes are
playing. In other words, Serum totals the number of voices you have active in a patch.
Portamento Settings
Below the voicing settings (in the very lower-right corner of Serum) are the Portamento controls.
Portamento allows for a slow glide/pitch bend from one note played to another. It is most
commonly used/most useful when Mono is enabled as well. This way if you hold one note and play
another, it will slowly change the pitch from the first note to the second.
Porta (Porta time) Knob The Porta time knob controls the rate of glide from one note to another.
Porta Curve Control The Porta curve, adjusts the contour of glide from one note to another. If set
convex (typical use),note pitch will depart the beginning pitch quickly and slow down as it nears
the note destination frequency. If set concave (dragged down below half),opposite is true: the
pitch will slowly depart the source pitch, and later rapidly arrive at the destination pitch.
'Always' switch When activated, the portamento will occur on a new note even if no note is
currently playing. When Off, a note must be held for portamento to take place on the (2nd) note.
'Scaled' switch This is potentially useful for melodic leads where you wish there to be a less
noticeable portamento on short intervals. When activated, the portamento rate is adjusted based
on the distance of transversal between the source and destination pitches. In other words, if the
portamento is a glide between two notes 1-octave apart, the rate knob's time value will be used. If
the portamento is a glide between notes less than one-octave apart, the time will be faster, and
vice-versa (notes larger than an octave will be progressively slower).

Chapter 12.Global page

settings
All of the Global settings are saved in your Preset/Song/Patch, except the Preferences which are
truly Global.

Oscillator Settings In the top-left of the Global page is the

Oscillator Settings:
Pitch Tracking OSC A switch
This switch determines whether or not OSC A will follow the MIDI note to determine pitch. When
off, all incoming MIDI notes will trigger OSC A at the same (low) pitch. If you wish to make this a
different pitch, you can use the OCT | SEMI | FINE and/or COARSE controls for OSC A on the main
OSC tab.
Pitch Tracking OSC B switch
This switch determines whether or not OSC B will follow the MIDI note to determine pitch. When
off, all incoming MIDI notes will trigger OSC B at the same (low) pitch. If you wish to make this a
different pitch, you can use the OCT | SEMI | FINE and/or COARSE controls for OSC B on the main
OSC tab.
Noise Fine
This allows an adjustment in cents for the Noise oscillator. This is useful when Key-track is enabled for
the Noise oscillator (noise pitch knob snaps to semitones in this mode), and the Noise oscillator has a
discernible pitch which sounds out of tune.
Oversampling This sets the quality for Serum. This setting applies to situations using a
Warp mode (OSC Sync, FM, etc) for an oscillator. Otherwise the standard oscillator quality is
always maximum!
Chaos 1 | Chaos 2
Chaos 1 and 2 are different types of chaos LFO (low frequency oscillators). To hear one or both,
you must make the assignments in the Mod matrix (or by ctrl-clicking (OSX) or right-clicking
(Windows) on a desired destination control (e.g. filter cutoff) and selecting the desired Chaos
under the Mod Source submenu).
Chaos 1 has a more periodic (repeating) and more bipolar (two separate tendencies, up or down).
Chaos 2 has more of a chaotically 'restarting' sort of sound to it.
A good use for the Chaos modulators is subtlety, for instance very small fine-tune deviations to
simulate the drift of analog oscillator pitch. However you may of course find your own uses!
Chaos BPM sync
Although by nature chaos is not predictable, since both Chaos oscillators do have a sort of natural
frequency in the algorithms, the Chaos BPM sync will snap the LFO rates to musical divisions
based on your host BPM. However you should not expect rhythms out, but nonetheless can be
useful at the very least so you can get the sort of chaos rate you had in mind (if you think in
musical units!)
Chaos Mono switch
When disabled, every voice (if you're playing a chord) will have it's own chaos oscillators. However
sometimes you may wish all voices to share the same chaos motion (for instance if controlling
filter cutoff, you may want all notes to have this same chaos movement). Enabling the mono
switch will make all synth voices use a single Chaos oscillator for this purpose.

Unison Settings
On the global tab you can find the more advanced Unison settings for OSC A and OSC B, respectively:
(Unison) Range (maximum detune amount)
Sets the range in semitones that the Oscillators Unison Detune knob (on OSC panel) will cover between
0 and 48 semitones. The default value is 2 semitones.
(Unison) Width (stereo spread)
Sets the stereo panning for the individual unison voices. 0 = all voices panned center, 100 = voices
panned as much as 100% Left and Right.
(Unison) Warp
Sets the amount of warp offset for each of the individual unison voices. To hear this, you must have a
Warp mode is selected (Sync, FM, etc) and of course Unison raised above 1. An easy way to think about
this is imagine the Warp knob on the Osc page has a little lower or higher value for each unison voice.
The more this Warp offset is raised, the wider the spread for each unison-voice this +/- assignment,
based from the Warp knob's current value.
(Unison) WT Pos Similar concept to Warp (above) but instead the offset per-unison-voice is made
to the WT Pos. This way a single note with unison enabled can be playing multiple sub-tables in your
wavetable simultaneously. The greater this control's value, the more spread.
(Unison) Stack The stack control provides multiple options for having the unison voices playing
different pitches. The default is off where all voices play the incoming pitch (note you played). The
first option: 12 (1x) means that every 2nd unison voice will play up an octave (0, +12, 0, +12, 0 +12,
etc). The next option: 12 (2x) means that this transposition can be either 1 or 2 octaves (0, +12, 0
+24, 0 +12, etc). The choices with +7 means that a perfect fifth will also be included in these layerings.
The last choices: Center-12 and Center-24 means the center voices of the Unison stack will be lowered
in pitch 1 or 2 octaves, respectively.
(Unison) Tuning
Determines the mode of detune for the detune knob. These changes are also reflected visually when
adjusting the detune knob on the main oscillator panel, you'll notice the vertical stalks which draw on
the waveform area will have a different look, depending on this Tuning Setting.

Linear which means all voices will be detuned by equally spaced amounts.
Super is a little less even than Linear, the detune has a slight emphasis towards notes being the
correct (in-tune) pitch. This
simulates a little closer how an ensemble of instruments playing the same note might sound (a couple
people are out of tune, but the average is closer to in-tune).
Exp., short for Exponential, favors notes closer to being intune even more. This might be closer to
how a more talented ensemble trying to play the same note might sound ;)
Inv., short for Inverted, is a little abstract, note tuning is pushed more to the outside edges. This
gives the ear a bit more of a
sensation of 2 pitch centers (sharp and flat).
Random selects random detune amounts every time there is a new note-on. This might sound more
natural in some situations, as of course humans don't play out of tune the same amount every
note/time.
Chapter 13. Wavetable
editor
Click on subtable: View/edit particular subtable Click +drag subtable:
reorder subtable
Shift-click on subtable after
clicking on subtable: Select a range
of tables
Draw tools(most are self-
explanatory but the harder
ones are in pic on right side)

Chapter 14. Import

audio
To import a multi-cycle wav file, drag the audio file from your Finder/Explorer or Host Sequencer's
file browser to the Waveform display (green waveform) on Serum's main window. Note: Dragging
files directly from the host arrangement windows or region bins is not possible in most hosts, but
the host standard file-browser should work. However many hosts have a way to show the (parent)
sound file in the host file browser, and you should be able to drag from there. As you drag over the
waveform, you will see a variety of choices. Where you 'drop' the file (release the mouse)
determines the load method. Serum will then analyze the sound and create a new wavetable set in
RAM. The specifics of analysis depend on what method you chose for import (pictured above, also
found in the Import menu of the WT Editor). The load 'choices' (pictured above) are as follows:
Import: constant frame size (pitch avg) *If in doubt... try this one first! This is typically the
best choice when a sound has a fixed frequency, such as a one-shot from a synthesizer (in other
words you hear it as a perfect or near-perfect constant-pitch, with no pitch bend or vibrato to
speak of). In this mode, Serum will analyze the entire file for an average pitch, and then uses this
number of samples as the import length. Because some sounds contain half-cycles, silence,
multiple notes, among other artifacts, serum might not guess the desired pitch. Fortunately,
Serum will display the number of samples it is using perframe in the WT Editor Formula area (read
on...) and will switch over to this 'fixed' value found from analysis unless changed or cleared from
the formula field.
import: normal (dynamic pitch zero-snap) This scans the audio file and builds a pitch-map. It
then attempts to locate zero-crossings that fall near the pitch map. While this can work wonders
on simple sounds, complex sounds unfortunately don't adhere to having sensible zero crossings,
so you'll end up with some glitches at best. It is recommended to use this mode when you have a
sound with a non-fixed fundamental (pitch bend or vibrato) and the sound is pretty simple, for
example a sawtooth wave with little filter sweep/resonance.
import: normal (dynamic pitch follow) Similar to the above, this also will build a pitch map
and import a varying-sized segment of audio for each sub-table, based on the analyzed pitch.
Unlike the above, pitch-follow import does not attempt to locate zero crossings. This means it is
better suited for complex sounds, such as a source sample that might have a touch of
chorus/unison, resonance, or background noise/notes.