Module 08 - Synthesis and Sampling PDF

Audio Masterclass Music Production and Sound Engineering Course
Module 08: Synthesis and Sampling
Module 08
Synthesis and Sampling

In this module you will learn the foundation knowledge of the processes of synthesis and sampling
in music technology. Although it is possible to operate and play synthesizers and samplers without
knowing much about how they work, it requires a full understanding of the principles by which they
create and manipulate sound to achieve their full potential.
Learning outcomes
To be able program and operate a subtractive synthesizer, either analog or digitally modeled.
To be able to program and operate a sampler. To create original samples and allocate them to
musical notes. To manipulate sound through sampling.
Assessment
Formative assessment is achieved through the short-answer check questions at the end of this
module.
Page 1
Module Contents
Synthesis 3
Brief note on additive synthesis 3
Subtractive synthesis 3
Waves 5
Filter 6
Voltage control 7
Voltage-controlled amplifier and envelope generator 9
Other features 12
Final point on subtractive synthesis 14
Sampling 15
The process of sampling 15
Sampler functions 22
Sampling rate 22
Sampling duration 22
Stereo sampling 23
Trim and extract 23
Normalization 23
Looping 23
Reverse 24
Time stretch 24
Making programs 25
Layering samples 25
Velocity zones 25
Triggering modes 26
Synthesizer-like functions 26
Mute groups 27
Having more than one program sound at the same time 27
Sampler specifications 28
Looping single note samples 29
Looping stereo samples 30
Appendix 1: MIDI note numbers 32
Appendix 2: Triggering from MIDI 33
Appendix 3: Transposition 34
Check questions 35
Page 2
Synthesis
The object of synthesis is to produce musical and
other sounds by purely electronic or digital means.
Brief note on additive synthesis

It is possible to analyze acoustic sounds mathematically.
In theory, any sound can be constructed by combining
sine waves of the correct frequency, amplitude and
phase relationships.
A sine wave is the simplest sound possible, consisting

of only one frequency component. It can be produced
by an electronic sine wave oscillator. The sound of a
tuning fork is the closest sound in real life to a pure
sine wave.
Phase refers to the time relationship between two

sine waves; where the peaks and troughs line up.
In theory therefore, it would be possible to imitate any

acoustic sound by combining the sounds of several
sine wave oscillators in the right way. This is called
additive synthesis, but historically it has been very
difficult to achieve in practice. Also, other methods
of synthesis have proved to be better for producing
sounds that are usable in a musical context.
Subtractive synthesis Classic Minimoog analog

subtractive synthesizer
Although it might seem sensible to build up complex
sounds from their basic components, in practice it is
simpler to start off with a complex sound, then take
away the components you dont need.
Musical sounds consist of harmonics. The pitch that

we perceive a musical note to be depends on its
fundamental, or fundamental frequency, which is also
considered to be the first harmonic. The note that an
orchestra tunes to, for example, has a fundamental
frequency of 440 Hz (440 Hertz - 440 complete cycles
of vibration per second). On top of this, when the
note is played by any acoustic instrument, there are
harmonics that give the note much of its character,
allowing us to differentiate between one instrument
and another. In musical sounds, the harmonics are
Page 3
generally whole-number multiples of the fundamental

frequency. For example, the harmonics of the note
with fundamental 440 Hz are these:
440 Hz
880 Hz (2x)
1320 Hz (3x)
1760 Hz (4x)
2200 Hz (5x)
2640 Hz (6x)
etc.
All string and wind instruments have this pattern of

harmonics. What makes them sound different is that
the relative strengths of the harmonics are different. In
the clarinet for example, the odd-numbered harmonics
are strong, while the even-numbered harmonics are
weak.
Therefore, if it is possible to start off with a sound

that is strong in all the harmonics, and control the
relative strengths of those harmonics in some way,
then in theory it should be possible to imitate acoustic
sounds.
In practice - theres always that difference between

theory and practice - the technology of the 1960s,
when subtractive synthesis was first developed, wasnt
able to shape the harmonics sufficiently accurately to
imitate acoustic instruments well. However, it turned
out that the sound that subtractive synthesizers of
the day could produce were very attractive to creative
musicians.
Page 4
Waves
As a starting point, we need a selection of harmonically
rich waveforms. The waveforms commonly used in
subtractive synthesis are these:
Sawtooth
Square
Pulse
Triangle
These waveforms are all harmonically rich, and they

are easy to generate with simple electronic circuits
called oscillators. The sine wave is not used because
it has no harmonics.
It is worth noting that there are two easy ways of

producing a sawtooth wave. For the technically
minded, one involves charging a capacitor from a
constant voltage source. This produces a sawtooth
with curved-edged teeth. If the capacitor is charged
from a constant current source, then the teeth have
straight edges. The two waves sound only slightly
different.
The square wave can be considered a special case

of the pulse wave. The pulse wave can have narrow
pulses or wide pulses. If the high-voltage segment of
the pulse is equal in width to the low-voltage segment
then we have a square wave, and we would say that
the pulse width is 50%. It also turns out that if the
pulse width is 50%, then the wave will only have
odd-numbered harmonics, similar to the sound of the
clarinet.
The triangle wave is not as harmonically rich as the

other waveforms, but it has a characteristic sound
that is useful in synthesis. Sine wave
Sawtooth wave
At this point, it is important to realize that the early
Square wave
subtractive synthesizers could only play one note at
Pulse wave
a time. Everything that follows from here will assume
Triangle wave
this.
Filter
To remove the harmonics from the wave, we need a
Page 5
filter. A filter is a fairly simple electronic circuit that

can remove high frequencies or low frequencies, both
high and low frequencies, or a band of frequencies in
the mid range. We would describe them as follows:
Low-pass filter: removes high frequencies

High-pass filter: removes low frequencies
Bandpass filter: removes high and low
frequencies
Band-stop filter: removes a band of frequencies
between the high and low frequencies
Bandpass is commonly written as one word, the

others sometimes are. There is another filter not
commonly used in synthesis called the notch filter,
which removes a very narrow band of frequencies.
In practice, the low-pass filter is by far the most

useful in terms of producing musical sounds. The raw
waveforms are very harsh, and need to be softened
considerably before they are usable. Many synthesizers
do only have low-pass filters.
The parameters of the filter are these:

Minimoog filter section
Cut-off frequency: the frequency above which
(for a low-pass filter) harmonics are removed.
Slope: the rate at which harmonics are
attenuated above the cut-off frequency. Slope
is measured in decibels/octave. A filter with a
slope of 24 dB/octave is most commonly used
in synthesis. A more gradual slope of 12 or 18
dB/octave is not sufficiently effective.
Resonance: the degree of boost applied at the
cut-off frequency
Page 6
Voltage control
In theory, we could take an oscillator, put the output
through a filter, then we would have a nice, musical
sound. We could control the oscillator from a keyboard
to produce different notes. The problem would be
however that the easy way of controlling the frequency
of the oscillator is by using the keyboard as an array
of switches. Each switch introduces a different value
component into the oscillator circuit, thus varying
the frequency. But this does nothing for the cut-off
frequency of the filter, which will remain the same. So
if you adjusted the filter to give the sound you wanted
for one particular note, a note of higher frequency
would be filtered more than you wanted, lower notes
would be filtered less, in proportion to the frequency
of the fundamental.
The key to musically successful subtractive synthesis

therefore is voltage control. Voltage control, invented
by Robert Moog, means that every circuit of the
synthesizer is controlled by a voltage input, rather than
component values. We have therefore the following
circuits within the synthesizer:
Voltage-controlled oscillator (VCO)

Voltage-controlled filter (VCF)
Voltage-controlled amplifier (VCA)
And also...
Envelope generator (EG)

Low frequency oscillator (LFO)
Lets take the example of the VCF. The VCF has a

number of inputs by which it can receive voltage
control. One would be via a knob on the front panel,
another is the keyboard, each key of which produces a
different voltage. To set the timbre you want from the
VCF therefore, you would play one note repeatedly
while turning the knob until you found the sound you
wanted. And then, when you play any other note, the
keyboard will send a voltage signal to the VCF so that
the cut-off frequency always stays in proportion to
the fundamental frequency of the note. This point is
vitally important to the understanding of subtractive
Page 7
synthesis.
Page 8
Voltage-controlled amplifier
and envelope generator
The VCA controls the level of the signal. In a simple
subtractive synthesizer, the VCA is not controlled
by the keyboard in any way, but principally via the
Envelope Generator (EG).
If you think of acoustic musical sounds, it is apparent

that some have a sharp attack and some start
gradually. The envelope generator is used to simulate
this. The parameters of the envelope generator are
these:
Attack time: the time taken for the sound to

reach its full level
Decay time: the time taken for the sound to
Minimoog envelope generator,
fall back to the sustain level
here called contour
Sustain level: the level at which the sound will
keep playing until the key is released
Release time: the time taken for the sound to
die away after the key is released
Note the initial letters: ADSR. The envelope generator

is sometimes called the ADSR.
Notice which of these are defined in terms of time

(attack, decay and release), and which is defined in
terms of level (sustain).
The envelope generator is used to control the level

of the sound from key-press to key-release, and is
retriggered for each note played. The signal that the
keyboard sends to the EG to inform it when a key is
pressed and released is called the trigger or gate.
The secondary function of the envelope generator is

to control the cut-off frequency of the VCF throughout
the duration of the note, and it is good for a subtractive
synthesizer to have two EGs, one for level and one for
the filter. Using the second EG it is therefore possible,
for example, to start off mellow, become brighter, fall
back to an intermediate mellow/bright balance, then
dull down again once the key is released.
At this point we are ready for a full block diagram of a

subtractive synthesizer...
Page 9
Page 10
Modulation
Modulation is the term we use for changing the
characteristics of a sound over time. There are three
principle types of modulation used in subtractive
synthesis:
Tremolo: a periodic variation in the amplitude

(level) of the sound
Vibrato: a periodic variation in the frequency of
the sound
Wah-wah: a periodic variation in the harmonic
content of the sound
In all cases, the source of the modulating voltage will

be the low-frequency oscillator (LFO). The LFO is an
additional oscillator that covers the range of around 1
Hz to 20 Hz. The most useful waveform for the LFO
is the sine wave, but triangle is also useful. A square
wave can be used for, for example, imitating the two-
tone siren of a police car.
To produce tremolo, the LFO is connected to

the VCA
To produce vibrato, the LFO is connected to the
VCO
To produce wah-wah (it sounds like a guitarists
wah-wah pedal), the LFO is connected to the
VCF
It is interesting to notice that modulation is only Modulation control in the Moog Voyager
musically useful over a narrow range of frequencies.
In fact if the LFO is set to produce vibrato, and the
frequency is increased to 20 Hz or more, the result is
not a particularly musical sound at all. In fact, this is a
crude form of Frequency Modulation (FM) synthesis.
Page 11
Other features
Although the above text describes the common
features of subtractive synthesis, real-life synthesizers
often have additional features.
Oscillator synchronization
Noise generator
Pulse-width modulation
Oscillating filter
Sample-and-hold
Ring modulator
Portamento
External input
Oscillator synchronization (sync) is where there are

two VCOs; VCO 2 is triggered to restart its waveform
each time VCO 1 starts a new cycle. If the two VCOs
are set to different frequencies, this can produce some
interesting effects.
The noise generator produces either white noise or

pink noise. White noise is a completely random signal
that contains every frequency in the audio band at
equal levels. Pink noise is produced by filtering white
noise such that each octave band contains an equal
amount of energy. In practice, most people only
remember that pink noise is not quite as bright as
white noise.
Although the noise signal isnt a great deal of use in

itself, once put through the VCF and controlled by the
EG, it becomes a valuable component of the sound
of subtractive synthesis. It can also be applied to
produce the sounds of explosions and wind.
Pulse-width modulation works only when the VCO

is set to square or pulse wave. The LFO is used as
the control voltage to vary the width of the pulse
waveform. This can produce a very rich-sounding
signal from a single oscillator.
If a synthesizer features an oscillating filter, it means

that the resonance parameter can be increased so
much that the filter itself starts to oscillate, with a
rather rough-sounding sine wave. This is often useful
Page 12
as an extra oscillator, but it also has a sound that

cannot be achieved in any other way.
Sample-and-hold is another method of producing a

control voltage. The noise signal is taken as a starting
point, and the value of the voltage is taken every so
often, say five times a second. This produces a voltage
that varies randomly several times a second that can
then be used to control the frequency of the VCO or
cut-off frequency of the VCF.
A ring modulator is a circuit that takes two signals

and multiplies them together. This produces
additional frequencies that are the sum of the two
input frequencies and the difference between the
input frequencies. These modulation products are
not musically related to the input frequencies, so the
ring modulator can be a source of very harsh sounds,
sometimes used as a component in imitating bells.
Portamento is a feature where the output voltage from

the keyboard changes only slowly as different keys
are pressed, rather than instantly as is normally the
case. This produces a sound that sweeps in frequency
between successive notes.
The external input is a means of accepting another

signal into the synthesizer, rather than using the
VCO, which can then be processed by all the other
components of the synthesizer. This is a remarkably
effective means of sound manipulation that is often
overlooked.
Page 13
Final point on subtractive

synthesis
Because subtractive synthesis was originally achieved
by analog circuits rather than digital technology,
subtractive synthesizers are commonly called analog
synthesizers. Modern subtractive synthesizers
however do use digital technology to perform the
same functions as the VCO, VCF, VCA, EG, LFO etc.
Digital subtractive synthesizers have the advantage
of polyphony - they can play many notes at the
same time. Although there were polyphonic analog
synthesizers in the past, they could only play a few
notes simultaneously, and they never really worked
too well.
Page 14
Sampling
Sampling is the process of digitally recording a
sound source and playing that recording from a MIDI
keyboard at different pitches according to which key
is pressed.
Currently, samplers have three distinct uses:
To sample a loop or breakbeat from a record

or CD to form the basis of the backing track
of a new song or rap. This is sometimes called
phrase sampling or loop sampling
To sample individual notes of a single
instrument so that the sound of that
instrument can be recreated from the keyboard
To sample sound effects so that they may be The classic Akai S1000 hardware sampler
played from the keyboard in a theatrical, live
broadcast or film/TV post-production context.
Of these the first looping function is currently very

popular and the other functions are almost ignored
by many. However, we must understand all of these
functions thoroughly.
The process of sampling

The process of sampling consists of a number of
steps:
Find the sound source to be sampled - for

example, find a suitable breakbeat on a record
Decide which MIDI note will play the sample at
its original pitch (usually this defaults to C3 -
middle C)
Set the recording level on the sampler
Cue the source so that it is ready to play
Put the sampler into armed mode so that it
is ready to record (there are a variety of ways
of initiating recording, which will be explained
later
Start the source sound and record it into the
sampler
If all is well, trim away audio that is not needed
Page 15
at the start and end of the sample - i.e. top

and tail the sample
Assign the sample to certain MIDI note
numbers so that it may be played from the
keyboard
Take more samples and assign them to
different MIDI note numbers
Page 16
Step 1. Finding the sound source

This could be a matter of listening to a record for
a suitable breakbeat, or getting hold of a musician
who can play the instrument you want to sample, or
finding the sound effect that you want to use in your
theatrical or other multimedia context in a CD sound
effects library.
Breakbeat
Hip hop and rap producers commonly use their record

collection as a source for samples (which they might
call loops). It is not uncommon for a producer to
have a collection of up to ten thousand vinyl records or
more. One of the skills of the producer is in knowing his
or her way round the collection; where certain sounds The Emulator II classic sampling keyboard
might be found, and remembering suitable samples.
If you have ever worked with such a producer, you will
realize that they have an amazing mental catalog of
sounds on which they can draw.
One of the useful features that most samplers do

not have is the ability to continuously record a sound
source into memory, always retaining the most recent
10 seconds or so. If a sampler has this feature, then
you can play the record and, when you hear a break
you like, hit the capture button, or whatever it is
named. This then stores the most recent 10 seconds,
ready for you to top and tail. Otherwise you have
to backtrack and go through a separate process of
recording.
Instrument
If the sound source is to be a musician, then obviously

a good musician is to be preferred, together with a
good instrument. The recording should take place in
good acoustics. It is almost certainly going to be more
efficient to record onto a medium such as DAT and
then transfer the samples into the sampler at a later
time. If you try to sample the instrument directly, it is
likely that the musician will spend more time waiting
for you to operate the sampler than actually playing
notes. (An alternative would be to record the audio
using a computer hard-disk recording software, then
top-and-tail the samples before loading them into the
Page 17
sample replay software).
If it is desired to mimic the sound of, say, a clarinet

from the keyboard, then the musician will need to
play a series of notes covering the full range of the
instrument. It is usually enough to sample three
evenly-spaced notes from each octave. Two would not
quite be enough, four might be better. Any more than
four and the task becomes excessively protracted and
painstaking for little additional benefit. This process is
called multisampling.
(It is worth noting that on a keyboard such as the

classic Korg M1, where samples are used as the basis
of the sound, a sample refers to a single sample that
spans the full range of the keyboard. A multisample
is a set of samples that together span the keyboard,
however the single samples that comprise the
multisample cannot be used independently. With a
sampler, you can use samples independently or as
part of a multisample as you wish).
It is important that all notes are played well, cleanly

and with a consistent attack and tonal quality. If,
for example, one note is a little spitty at the start, Akai MPC 60 classic sampling drum machine
which a softly-played clarinet note can be, this will
be an irritation when the final sampler program is
assembled.
Also, it is useful to sample sets of quiet, medium and

loud volumes.
As the duration of the note is limited by the musicians

breath control, it is best to get him or her to play notes
of a duration that is as long as possible, but consistent
with achieving a good steady sound. If you ask the
player to play for ten seconds, they can probably do it
but control might suffer.
Many instruments can be played with or without vibrato.

Either can be useful in sampling, but notes played
without vibrato are the most versatile as vibrato can
be added in the sampler. This will not sound quite as
good as natural vibrato. The two drawbacks of natural
vibrato are a) that it cant be taken away, and b) that
adjacent notes played on the keyboard will play back
the same sample. When the sample has vibrato, not
Page 18
only will the pitch of the note be transposed, the rate

of the vibrato will change too. This is not entirely
pleasant, although it may go unnoticed depending on
the context.
Sound effects
Sound effects are most commonly sourced from a CD

sound effects library, although you could record your
own onto a portable recorder, from which to sample
later.
Step 2. Set the record level

This is the same as for any other recording medium.
A recording level must be set so that it is as high
as possible, without clipping. It is generally possible
however for a sample that was recorded at a lower
level to be normalized so that it does come up to peak
level, of course without clipping. This sacrifices some EXS24 sample replay software
of the signal-to-noise ratio, but it is often important
that a series of samples, of a musical instrument
perhaps, should be consistent in level.
Step 3. Cue the source so that it is

ready to play
If the source is a record, or CD, then cueing it up is
simple. If the source is a live musician, or other live
source, then it is useful to have a hand signal so that
the musician knows when to play.
Step 4. Put the sampler into armed

mode
There are generally a number of ways in which
sampling can be initiated:
Triggered by the incoming audio reaching a

certain level
Manually triggered
Triggered by a MIDI note-on
Of these, the first method is by far the most commonly

used. When the sampler is armed, it listens for a
certain level at the input. Until this level, below which
is probably just background noise, is reached, no
recording takes place. Once it is reached, i.e. the
Page 19
musician starts to play, then sampling is initiated.
Manual triggering is obvious.
Step 5. Start the sound source and

record it into the sampler
Obvious
Step 6. Top and tail the sample

With most sampling there will be excess audio material
before and after the part of the sample you actually
want. All samplers make it relatively easily to set the
actual start and end points you want, so only the audio
in between will be heard.
It is possible to go one stage further and delete the

unwanted material. This has the advantage that
memory is freed up into which additional samples can
be recorded. Once you have deleted the pre-start
and post-end material, you cant always get this
Kontakt 2 sample replay software
audio back.
With modern samplers, there is generally more than

enough memory and you dont need to worry about
these small fragments. However, it takes time to store
and reload sample data, and the time saved by not
storing audio you dont want could be significant.
Step 7. Assign the sample to MIDI note

numbers
At this stage of the sampling process it is usually
possible to play the sample from the keyboard
straight away. Whatever note has been assigned as
the original pitch will play the sample back as you
recorded it. Higher and lower notes will play back the
sample proportionately higher or lower. The note can
be played from any key on the keyboard.
Usually however, it is desired to play different samples

from different keys. In this case, the sample is assigned
to a consecutive group of keys, commonly known as
a keygroup. Keys outside this range will not play the
sample.
The whole sampling process is repeated until sufficient

samples have been assigned to keygroups and the
Page 20
whole program - as it is commonly known - spans

the keyboard and is ready to play.
Page 21
Sampler functions
Sampling rate
It is confusing that there are two meanings of the
word to sample, both of which apply to samplers.
One is in the sense of digital audio recording where
an analog signal is converted to digital by sampling its
level around 40,000 times a second, then allocating
each level a digital number. Sampling, as in sample
a breakbeat is obviously a much longer timescale
process.
In digital audio, the sampling rate has to be more

than twice the highest frequency you want to capture.
Hence to capture up to 20 kHz, the accepted limit of
human hearing, the sampling rate (sometimes called
sampling frequency) has to be greater than 40 kHz.
In practice, the two most common rates are 44.1 kHz
and 48 kHz. Typically a sampler will offer 44.1 kHz
sampling as standard, and 22.05 kHz as an alternative.
The lower rate will only capture frequencies up to
around 10 kHz.
The reason why you might sample at a lower sampling

rate is because it conserves memory. However, modern
samplers have plenty of memory so this reason no
longer really applies. If the sampler offers a choice of
sampling rates, particularly very low sampling rates,
you may choose these lower rates because you like
their grainy sound quality as an effect.
Usually it is possible to convert a sample from one

rate to another.
Sampling duration
It is common to be able to choose the duration over
which sampling will take place. For example, if you
know that the sound you are sampling lasts just
under a second, you can set a sample duration of
one second and all will be well. However, if you are
manually triggering the sample, it is better to set a
longer duration, say four seconds. The excess can
be topped and tailed later. Setting a duration that
is very much longer than the sound is pointless. If
a sampling duration of 30 seconds was set and the
Page 22
sound only lasted one second, you would have to wait

the remaining 29 seconds until sampling completed.
Either that or abort the sample completely.
Stereo sampling
Stereo sampling is where the left and right channels
of a stereo signal are sampled simultaneously, which
is obvious. Depending on the sampler, this might
produce a single stereo file, or two mono files. Where
two mono files are produced, it is usual that they are
named automatically to show that they are linked,
and which channel is which: NAME-L and NAME-R, for
example.
Trim and extract

Topping and tailing (trimming) the sample has been
described earlier. However in some samplers it is also
possible to extract a sample from a longer duration
of audio. If this procedure results in a single sample,
then it is functionally equivalent to trimming. However
it is often convenient to sample a long duration, say
thirty seconds, and then extract several samples from
that, each of which would have to be given a unique
name; the original sample remains in memory during
this process.
Normalization
As mentioned earlier, it is possible to normalize a
sample so that it peaks at the maximum level possible
without clipping. This is useful to ensure that a group
of similar samples are all around the same level. It
is worth noting that the ear judges level according
to the average level rather than the peak, so some
adjustment may still be necessary. This adjustment
may sometimes be made to the sample itself. At
other times it is done when the sample is placed into
a program.
Looping
This technique is used for extending the duration
of a sampled note from a musical instrument. This
works well for a sustained sound such as a clarinet. It
can work, with some difficulty, for a piano note. It is
Page 23
pointless for a one-off drum sound.
Looping works by taking a portion of the sample

somewhere between the beginning and end, and
continuously repeating it until the key is released. It
is often difficult to create a successful loop as there
may be a click or other discontinuity in the sound at
the splice point of the loop. This is explained in the
section on looping later.
Looping is not used for breakbeats. Although you might

sample a two-bar phrase of drums with the intention
of repeating it over and over again as the basis of a
rhythm track, the looping function is carried out in the
sequencer. How this is carried out is described in the
unit on sequencing.
Although the breakbeat could be looped in the sampler

quite easily, it would be impossible to sequence this
with other instruments as the timing of the loop can
not be synchronized with the tempo of the sequencer.
However, if you only want to use the looped breakbeat,
and there were no other sequenced instruments,
synchronization would not be necessary and it would
work fine.
Reverse
It is generally possible to reverse a sample so that it
plays backwards.
Time stretch
Time stretch is an expression that is used to cover
both time expansion and time compression - making
the duration of a sample longer or shorter without
changing the pitch. This technique is generally time
consuming and may result in poor audio quality.
However the results, particularly on rhythmic samples,
may be interesting and musically useful. The musical
styles of jungle and drum n bass rely heavily on time
compression of breakbeat samples.
Page 24
Making programs
So far we have looked at sampling and samples, and
have only touched on the way in which samples are
allocated to notes on the keyboard. The following text
is mainly derived from Akai samplers, but all samplers
have similar features.
As we saw earlier, if we want to assign more than

one sample to notes on the keyboard, we need to
construct a program. The program will offer the
following features:
Between one and as many keygroups as you

like. Each keygroup will usually have one
sample (or one stereo sample)
Each keygroup may be retuned, or transposed
to a different pitch
The level of each keygroup may be adjusted in
relation to the other keygroups
Each keygroup can be panned. (If the sample
is stereo, the left half will normally be panned
hard left, the right half hard right)
Normally all keygroups in a program will
respond to the same MIDI channel, but
individual keygroups can be set to different
channels if required.
Layering samples
While it is usual for each keygroup to have one sample,
there are reasons you might want to have two or more
samples in a keygroup.
One is to layer sounds. If you assign two samples to

a keygroup, then a thicker sound will be produced. It
may be possible to detune the samples, one slightly
higher in pitch, one slightly lower. This will further
thicken the sound. A similar effect could be achieved
with two single-sample keygroups, both assigned to
the same group of keys.
Velocity zones
Another reason for having more than one sample in a
keygroup is to create velocity zones.
Page 25
Normally, samples will respond to all MIDI velocities

from 1 to 127, and will vary proportionately in loudness.
However it is possible to assign one sample to, say,
velocities 1 to 63, and another to velocities 64 to 127.
This will create a velocity switch. In the early years
of samplers, one common use was for bass guitar,
where lower velocities produced a normally-picked
note. Higher velocities produced the same pitch, but
as a slapped note.
Another application of velocity zones is to take up to

four samples of a note from, say, a piano at different
levels of loudness. The sound quality of any instrument
varies considerably with the level at which it is played,
so if this can be mimicked by placing soft, medium
soft, medium loud and loud samples in four velocity
zones so that they will be triggered by different MIDI
velocities, then the result can be very natural indeed.
It must be said however that all of this involves a
massive amount of work, so it is only appropriate for
sample libraries, not really for individual musicians.
It is possible to overlap zones to create a velocity

crossfade.
Triggering modes
There are two main triggering modes:
One-shot: the sample plays all the way to the

end, even if the key is released. This is useful
for drum hits, breakbeats and extended sounds
As played: the sample stops when the key
is released. This is useful for instrumental
sounds.
The terminology for these functions will vary according

to manufacturer.
Also, it is possible to set a constant pitch mode so

that the sample will always play back at the pitch at
which it was originally recorded, no matter what key
is pressed. This is useful for drum and percussion
sounds, otherwise they will be transposed according
to which key is played.
Synthesizer-like functions
Page 26
Once keygroups have been created, they can be

subjected to synthesizer-like processing. This will
include envelope shaping via an envelope generator
with attack, decay, sustain and release parameters.
There will also be a low-pass filter, which may or may
not feature resonance. The filter will probably have a
separate envelope generator.
Mute groups
A mute group is where the triggering of one keygroup
prevents another from sounding, or cuts it off if it is
already playing. An example of the use of this is the
hihat of a drum kit. The hihat has open and closed
sounds, which a real drummer commonly alternates.
Obviously, it is impossible for both sounds to occur at
the same time. If both the open and closed sounds
were sampled and assigned to the keyboard, then
it would indeed be possible for both to sound at the
same time. However, if they are both assigned to the
same mute group, only one is allowed to sound at a
time. This makes the result much more realistic.
Having more than one program sound

at the same time
There is another way of layering sounds besides
assigning more than one sample to a keygroup. That
is to have two or more programs sounding at the same
time. This is very easy to do - just assign them to the
same MIDI channel.
Page 27
Sampler specifications
We need to know what makes the difference between
one sampler and another, apart from the facilities
provided. What are the fundamental specifications?
Memory: Early samplers had as little as ten seconds

worth of memory. It requires 5 Megabytes of memory
to store 1 minutes worth of mono samples at CD
quality. To give a couple of useful comparisons - to
sample the entire lead vocal of a song would take
2 to 3 minutes of mono sampling time (this used
to be common before hard disk systems became so
prevalent). Another would be to sample every note
of a grand piano, for the full duration of their decay
at several velocities. This takes several hundred
Megabytes. Some samplers can store data on hard
disk to provide almost infinite storage.
Voices: Each sample that it is required to play

simultaneously requires one voice. If you wanted
to play a five-note chord, then five voices would be
required. If keygroups have been layered so that each
note plays two samples, then ten voices are needed.
Modern samplers can have 64 voices, but these can
easily be used up if a complex composition is played
from a MIDI sequencer.
Outputs: Samplers are inherently multitimbral,

meaning that they can play different sounds over
different MIDI channels. Ideally, a sampler should
have as many outputs as possible. Sixteen would be
a good number since there would be one per MIDI
channel. The outputs could be routed independently
to a mixing console. Even then, more would be
desirable. A drum/percussion program would benefit
from a separate output for each note!
Page 28
Looping single note samples

It is often necessary to loop a single note sample
so that it can be played for any duration. Take the
example of a clarinet note. It would be practical to
sample three or four seconds, and loop the sample so
it could be played for any length of time from a MIDI
keyboard.
Here is how the loop works:
The note-on message triggers the sample to

start playing from the beginning
The sample plays until the end point of the
loop
The sample then plays repeatedly from the
start point to the end point, for as long as the
key is held down
When the key is released, the sample stops
immediately. If a release time is set, the loop
will continue until the sound has faded away
In some samplers, it is also possible to set
a release loop. This means that the sample
plays and loops as before until the key is
released. When the key is released, the note
continues into the release loop phase and loops
until the sound has faded away
Note that the portion of the sample after the
end point of the loop (or after the end point of
the release loop) is never heard.
One of the problems with looping is that there can

easily be a click every time the loop cycles:
In the top example of the diagram the end point of the

loop has a positive level, the start point of the loop is
negative (this is correct, they do come in this order).
The discontinuity in the waveform will create a click.
Contrast this with the bottom example. In this case

the level of the wave is the same at both ends of the
loop. So is the slope. Since there is no discontinuity,
there is no click.
So, two factors are necessary for a successful loop:
Page 29
Same level
Same slope
In the past, it has often been said that the loop must
be made through a zero crossing point. This is where
the level is zero. If the level is zero at the start and
end point of the loop, then the first criterion is fulfilled.
But if the slope is not the same, then there will be a
click.
Even if the criteria above are fulfilled, the loop may

still not be successful.
A good example is a piano note, which obviously decays

over time. In this case, the start and end point of the
loop will be at a different position during the course of
the decay, so there will be a sudden increase in level
each time the loop repeats. The best way to deal with
this is usually to make the loop very short, and as late
as possible in the sample so the discontinuity is not
very loud.
A further problem is if the quality of sound is different

at the start and end point of the loop. Often, this can
be an insurmountable problem.
The final solution to major looping problems is

crossfade looping, where a short crossfade should
cover any remaining discontinuity. However in
practice it is found that crossfade loops sometimes
work, sometimes they dont. Results can be difficult
to predict.
Ultimately, sometimes it is impossible to get a really

good loop and all you can do is either not loop at all,
or be content with the best you can do.
Looping stereo samples

It is often difficult to find a good loop in a mono sample.
Trying to find a loop that works for both channels of
a stereo sample can be almost impossible. Usually a
short crossfade will be necessary, even in a sample
with a very regular waveform. The difficulty of finding
acceptable loops in stereo samples has been known
to make people dedicate themselves totally to mono
sampling, with the exception of breakbeat loops,
Page 30
where the loop is created by the sequencer, or one-

shot sounds like drums that dont need to be looped.
Page 31
Appendix 1: MIDI note

numbers
MIDI note numbers on a typical five-octave 61-key
keyboard range from C1 to C6.
Notes in the octave above each C have the same

number.
Hence the note immediately below C3 is B2. The notes

immediately above C3 are D3, E3, F3 etc.
The note one semitone above C3 is C#3. It is never

D flat in MIDI.
Most equipment conforms to this standard. However

there is another standard where middle C, instead of
being called C3, is called C4. This is unfortunately a
little confusing.
Deciding which MIDI note will play the

sample at its original pitch
If you are sampling a breakbeat, and that will be the
only sample you use, then you can leave this at the
normal default setting of C3 - middle C. Other notes on
the keyboard will play the sample at proportionately
higher or lower pitches (and tempos).
However, if you are recording a note from a musical

instrument, say A3, then you would want this note to
play back at the correct pitch when you play A3 on the
keyboard. Usually, you can set this after sampling,
but it is probably more convenient to set it before
because unless you took written notes on what you
recorded, it might be difficult to tell what the original
note was.
Page 32
Appendix 2: Triggering from

MIDI
Imagine the case of a MIDI sequencer synchronized to
a multitrack tape recorder. The system incorporates a
sampler that can be triggered by the sequencer.
Suppose there is one note of a vocal track that falls in

pitch towards the end, and you desire to correct this.
Record a MIDI note-on into the sequencer just

before this vocal note - perhaps one second or
so.
Sample the vocal note using the MIDI note-on
on as a trigger to initiate sampling.
Correct the falling pitch in the sample by
recording a pitchbend message into the
sequencer.
Record the sample back to the vocal track
on the tape, using the multitrack recorders
punch-in function.
The vocal note sample will be triggered by the
sequencer.
Obviously there are some additional details of

operation, but the process is simpler than it appears,
once practised.
Page 33
Appendix 3: Transposition
The essence of the function of the sampler is that any
sample can be played at different pitches according to
which key is pressed on the MIDI keyboard.
To do this, the sample is simply speeded up or slowed

down. This changes the pitch, and also the duration
of the sample.
When samples are transposed downwards, they

have to be filtered so that the sampling frequency
(which is also transposed) is not audible. The cutoff
frequency of the filter is proportional to the degree of
transposition.
However, many early samplers did allow the sampling

frequency to become audible when samples were
transposed down. Also, you could hear the aliasing
products, which is where frequencies greater than
half the sampling rate were allowed to get through
into the recording, which according to digital theory
should not be allowed.
Aliasing causes additional frequencies to be generated,

which are not musically related to the frequency of
the sample being played. The result is a grainy sound
quality.
This might sound bad, but some old models were

prized for this quality because it sounded, in a strange
way, more exciting.
Page 34
Check Questions
Subtractive synthesis
Briefly describe additive synthesis.
Why was subtractive synthesis developed before additive synthesis?
Describe the harmonic structure of a note from a string or wind instrument.
Can subtractive synthesis accurately imitate the sounds of acoustic instruments?
List the four waveforms commonly used in subtractive synthesis.
Why is the sine wave not used as a sound source in subtractive synthesis?
Describe the relationship between the square wave and pulse wave.
How many notes could early subtractive synthesizers play at the same time?
What is the function of a low-pass filter?
Which type of filter is most commonly found in a subtractive synthesizer?
Describe cut-off frequency.
Describe slope.
Describe resonance.
Why is voltage control necessary to design a workable analog subtractive synthesizer?
What is a VCO?
What is a VCF?
What is a VCA?
What is an LFO?
What is an envelope generator.
Explain ADSR.
In what order are the VCO, VCF and VCA normally connected?
Name and describe the sound produced by modulating the VCA with an LFO.
Name and describe the sound produced by modulating the VCO with an LFO.
Name and describe the sound produced by modulating the VCF with an LFO.
What is oscillator synchronization (sync)?
What is the noise generator used for?
What is pulse width modulation?
What is an oscillating filter?
What is sample and hold?
What is a ring modulator?
What is portamento?
What is an external input used for?
Page 35
Sampling
What is a breakbeat?
What is the usual reason for sampling notes from a musical instrument?
How are samplers used for sound effects?
Describe the process of finding and sampling a breakbeat from a vinyl record.
When sampling notes from an instrument, why is it necessary to sample several notes?
Why should notes from an instrument be consistent in tonal quality?
Describe the pros and cons of sampling an instrument with or without vibrato.
Describe the term original pitch.
Give the names of the two white notes adjacent to C3 on a MIDI keyboard.
What does it mean when the sampler is armed?
List the three ways by which sampling can be initiated.
What is topping and tailing (trimming)?
What is a keygroup?
Why is it not a good idea to set a sampling duration of thirty seconds if the sound you want to
sample only lasts one second?
How does extract differ from trim, as described in the text?
What is normalization?
Why might it be necessary to loop a sample?
Why are breakbeats not looped in the sampler?
What is the difference between timestretch, and the normal transposition that occurs when a
sample is played away from its original pitch?
What are velocity zones?
Describe the two triggering modes listed in the text.
Why might it be desirable for the sample to play at constant pitch?
Describe the function of mute groups.
Why might you assign more than one program to the same MIDI channel?
If a sampler has 128 Megabytes of memory, what is the total maximum duration of mono
samples that can be stored in the memory, at CD quality, approximately?
What does it mean to say that a sampler has 64 voices?
Why should a sampler have lots of outputs?
What criteria are necessary for the successful looping of a mono sample with a regular
waveform?
What is a crossfade loop?
Why is it difficult to loop a sample of a piano note?
Why is it more difficult to loop stereo samples than mono samples?
Page 36

Module 08 - Synthesis and Sampling PDF

Hochgeladen von

Dokumentinformationen

Originaltitel

Copyright

Verfügbare Formate

Dieses Dokument teilen

Dokument teilen oder einbetten

Freigabeoptionen

Stufen Sie dieses Dokument als nützlich ein?

Sind diese Inhalte unangemessen?

Copyright:

Verfügbare Formate

Module 08 - Synthesis and Sampling PDF

Hochgeladen von

Copyright:

Verfügbare Formate

Audio Masterclass Music Production and Sound Engineering Course

Module 08: Synthesis and Sampling

Synthesis and Sampling

Brief note on additive synthesis

A sine wave is the simplest sound possible, consisting

Phase refers to the time relationship between two

In theory therefore, it would be possible to imitate any

Subtractive synthesis Classic Minimoog analog

Musical sounds consist of harmonics. The pitch that

generally whole-number multiples of the fundamental

All string and wind instruments have this pattern of

Therefore, if it is possible to start off with a sound

In practice - theres always that difference between

These waveforms are all harmonically rich, and they

It is worth noting that there are two easy ways of

The square wave can be considered a special case

The triangle wave is not as harmonically rich as the

filter. A filter is a fairly simple electronic circuit that

Low-pass filter: removes high frequencies

Bandpass is commonly written as one word, the

In practice, the low-pass filter is by far the most

The parameters of the filter are these:

The key to musically successful subtractive synthesis

Voltage-controlled oscillator (VCO)

Envelope generator (EG)

Lets take the example of the VCF. The VCF has a

If you think of acoustic musical sounds, it is apparent

Attack time: the time taken for the sound to

Note the initial letters: ADSR. The envelope generator

Notice which of these are defined in terms of time

The envelope generator is used to control the level

The secondary function of the envelope generator is

At this point we are ready for a full block diagram of a

Tremolo: a periodic variation in the amplitude

In all cases, the source of the modulating voltage will

To produce tremolo, the LFO is connected to

Oscillator synchronization (sync) is where there are

The noise generator produces either white noise or

Although the noise signal isnt a great deal of use in

Pulse-width modulation works only when the VCO

If a synthesizer features an oscillating filter, it means

as an extra oscillator, but it also has a sound that

Sample-and-hold is another method of producing a

A ring modulator is a circuit that takes two signals

Portamento is a feature where the output voltage from

The external input is a means of accepting another

Final point on subtractive

Currently, samplers have three distinct uses:

To sample a loop or breakbeat from a record

Of these the first looping function is currently very

The process of sampling

Find the sound source to be sampled - for

at the start and end of the sample - i.e. top

Step 1. Finding the sound source

Hip hop and rap producers commonly use their record

One of the useful features that most samplers do

If the sound source is to be a musician, then obviously

sample replay software).

If it is desired to mimic the sound of, say, a clarinet

(It is worth noting that on a keyboard such as the

It is important that all notes are played well, cleanly

Also, it is useful to sample sets of quiet, medium and

As the duration of the note is limited by the musicians