introduction to multimedia systems part 2, Study notes of IP Multimedia Subsystems

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 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 B. Okuku multimedia.

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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 device.B. Okuku

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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 signals, B. Okuku

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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 for short.

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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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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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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 produce a set of fixed pitches.B. Okuku

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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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• 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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