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Okuku 1
Multi- means many; much; multiple
Medium means:
•An intervening substance through which
something is transmitted or carried on
•A means of mass communication such as
newspaper, magazine, or television

Multimedia is woven combinations of text,

graphic art, sound, animation, video and other
kinds of elements.
When a viewer of a multimedia presentation is
allowed to control what elements are delivered
and when, it is interactive multimedia.
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Multimedia is an inter-disciplinary subject
because it involves a variety of different theories
and skills:
•these include computer technology, hardware
and software;
•arts and design, literature, presentation skills;
•application domain knowledge

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Application domain — provides
functions to the user to develop and
present multimedia projects. This
includes Software tools, and
multimedia projects development
methodology.

System domain — including all

supports for using the functions of
the device domain, e.g., operating
systems, communication systems
(networking) and database systems.

Device domain — basic concepts

and skill for processing various
multimedia elements and for handling
physical
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 Everyday, we encounter many values
that change continuously, for example,
the voltage of the electricity that lights
up our room varies continuously over
time. These are also known as
analogue signals.
However, modern computers are built
to deal with entities in completely
different way. These are known as
digital computers because they work
with digits.
Because of this, when using a
computer to process continuous
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signals,
we first need to find a way to represent them so that
the computer is able to handle them. Usually, this is
a digital representation, i.e., we use a series of
numbers to denote the continuous signals.
Then, we have to convert the continuous signal
into the digital representation. This process is
known as digitisation.
The first step in the digitisation process is
sampling which takes samples of the continuous
signal. The number of samples taken during a time
period is known as sampling rate.
The second step is known as quantisation where
we restrict the value of the samples to a fixed set of
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 The unit of sampling rate is Hertz (Hz),
i.e., 1Hz means taking one sample per
second. For many signals, this is far too
slow, therefore, we often use kHz, i.e.,
kiloHertz.
Because modern digital computers often
store data in bytes, each sample is usually
stored using either 8 bits (1 byte) or 16 bits
(2 bytes). This corresponds to either 256 or
65536 levels for a digitised sample.
The device that we use to convert
analogue signal to digital signal is known
as an analogue-to-digital converter, ADC
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for short.
After being processed by the computer, the signal
will be played back , i.e., we need to reconstruct the
signal from the digital respresentation. One
commonly used technique is known as sample and
hold.
Clearly, if we want to reconstruct a signal that is
as closed to the original signal as possible, we
need to take sufficiently many samples, and we
need to have as many levels to record the sample
values in as possible.

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Nyquist sampling theorem tells us that, in order to
reconstruct the signal, the sampling rate must not be less
than twice the maximum frequency of the original signal.
For example, if the maximum frequency is 3000Hz, the
sampling rate must not be less than 6000Hz.
If we undersample, i.e., taking less samples than as
required by Nyquist sampling theorem, some of the
frequency components will be mistakenly converted into
other frequencies. This is known asaliasing.

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On the other hand, if we use too few levels to represent each
sample value, there will be large amount of error for each
sample.
This is known as quantisaton error. These errors can be
thought of as noise on the signal.
We measure the quality of a sample by its signal-to-noise
ratio (SNR). The higher the resolution, the smaller the noise,
and the better the quality. The unit of SNR is dB (deci Bel).
This is defined by

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 Sound is a physical phenomenon produced by the
vibration of matter and transmitted as waves.
However, the perception of sound by human
beings is a very complex process. It involves three
systems:
•the source which emits sound;
•the medium through which the sound
propagates;
•the detector which receives and interprets the
sound.

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Sounds we heard everyday are very complex.
Every sound is comprised of waves of many
different frequencies and shapes. But the simplest
sound we can hear is a sine wave.
Sound waves can be characterised by the following
attributes:

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 Period is the interval at which a
periodic signal repeats regularly.
Pitch is a perception of sound by
human beings It measures how
‘high’ is the sound as it is
perceived by a listener.
Frequency measures a physical
property of a wave. It is the
reciprocal value of period

The unit is Herts (Hz) or kiloHertz

(kHz).
Musical instruments are tuned to
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produce a set of fixed pitches.
The other important perceptual quality is
loudness or volume.
Amplitude is the measure of sound levels. For a
digital sound, amplitude is the sample value.
The reason that sounds have different loudness is
that they carry different amount of power.
The unit of power is watt. The intensity of
sound is the amount of power transmitted
through an area of oriented perpendicular
to the propagation direction of the sound.
If the intensity of a sound is ,
we may start feel the sound. The ear may be
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This is known as the threshold of feeling. If the
intensity is we may just be
able to hear it. This is known as the threshold of
hearing.
The relative intensity of two different sounds is
measured using the unit Bel or more commonly
deciBel (dB). It is defined by

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•Dynamic range means the change in sound levels.
For example, a large orchestra can reach 130dB at
its climax and drop to as low as 30dB at its softest,
giving a range of 100dB.
•Bandwidth is the range of frequencies a device can
produce or a human can hear.

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•Sound waves are continuous while computers are
good at handling discrete numbers.
•In order to store a sound wave in a computer,
samples of the wave are taken.
•Each sample is represented by a number, the
‘code’.
•This process is known as digitisation.
•This method of digitising sound is know as pulse
code modulation (PCM).
Refer to Unit 1 for more information on digitisation.

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•According to Nyquist sampling theorem, in order to
capture all audible frequency components of a
sound, i.e., up to 20KHZ, we need to set the
sampling to at least twice of this. This is why one of
the most popular sampling rate for high quality
sound is 4410HZ.
•Another aspect we need to consider is the
resolution, i.e., the number of bits used to represent
a sample.
Often, 16 bits are used for each sample in high
quality sound. This gives the SNR of 96dB.

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The size of a digital recording depends
on the sampling rate, resolution and
number of channels.

Higher sampling rate, higher resolution

gives higher quality but bigger file size.

For example, if we record 10 seconds of stereo music at

44.1kHz, 16 bits, the size will be:

High quality sound files are very big, however, the file size can be reduced
by compression.
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•The most commonly used digital sound format in
Windows systems is
.wav files.
•Sound is stored in .wav as digital samples known as
Pulse Code Modulation(PCM).
•Each .wav file has a header containing information
of the file.
•type of format, e.g., PCM or other modulations
•size of the data
•number of channels
•samples per second
•bytes per sample
•There is usually no compression
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Other format may use different compression
technique to reduce file size.
• .vox use Adaptive Delta Pulse Code Modulation
(ADPCM).
• .mp3 MPEG-1 layer 3 audio.
• RealAudio file is a proprietary format.

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•Recording and Digitising sound:
•An analog-to-digital converter(ADC)
converts the analog sound signal into digital
samples.
•A digital signal processor(DSP) processes the
sample, e.g. filtering, modulation, compression,
and so on.
•Play back sound:
•A digital signal processor processes the sample,
e.g. decompression, demodulation, and so on
•An digital-to-analog converter(DAC) converts the
digital samples into sound signal
•All these hardware devices are integrated into a
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few chips on a sound card
•Different sound card have different
capability of processing digital sounds.
When buying a sound card, you should look at:
•maximum sampling rate
•stereo or mono
•duplex or simplex

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•Windows device driver — controls the
hardware device.
Many popular sound cards are Plus and Play.
Windows has drivers for them and can
recognise them automatically. For cards that
Windows does not have drivers, you need to get
the driver from the manufacturer and install it
with the card.
•If you do not hear sound, you should check the
settings, such as interrupt, DMA channels, and
so on.
•Device manager — the user interface to the
hardware for configuring the devices.
•You can choose which audio device you
want to use
•You can set the audio volume
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Mixer — its functions are:
•to combine sound from different sources
•to adjust the play back volume of sound sources
•to adjust the recording volume of sound sources

Recording — Windows has a simple Sound Recorder.

Editing — The Windows Sound Recorder has a limiting
editing function, such as changing volume and speed,
deleting part of the sound.

There are many freeware and shareware programs for

sound recording, editing and processing.

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