Wednesday, 30 April 2014

Hi - Fi and Stereo

High fidelity is the attempt to reproduce by electronic and electromechanical devices (and without distortion) the full range of audible sound from 20 Hz to 20 kHz. A reproducing system is called as “Hi-Fi”, if it fulfills the above conditions and comes as close as possible to sounding like original. In other words, the reproduced sound should be exact replica of the recorded sound.
“Fidelity” means “Faithfulness”. An ideal hi-fi system should

1.Be complete free from noise. The S/N ratio should be infinite for complete AF band. In other words, there should be no frequency distortion.
2.There should also be no amplitude or phase distortions.
3.It should be able to detect the direction for which a particular sound is coming.
4.It should create stereophonic (3 dimentional effects).

The quality of reproduction system will depend upon many factors. The acoustics of the room and deficient (poor quality) electromechanical components (microphones, speakers) will also effect output of the system. However a Hi-Fi system at the minimum should contain an input source such as record player, an audio amplifier a loud speaker more elaborate system may and contain more than one input source, separate pre-amplifier and a multiple speaker system at the output.

The reason, why people prefer higher cost of separate components instead of a radiogram in a single cabinet is the better quality of each individual component especially the amplifiers and speakers and also because of the absolute need for the speakers to be housed separately for high quality (Hi-Fi) sound reproduction.
Stereophony:

The stereophony is made of two terms stereo + phone. Stereo means solid (3 dimensional) and phone means sound, i.e., stereophony means “3 dimensional sound”. Stereophony gives us original sound effects and we can feel its depth as well as direction. The human system of hearing is also stereophonic.

Stereophonic Reproduction:

The conventional method of mono-phonic (mono means single) reproduction over a single channel cannot do faithful reproduction, as it cannot preserve the minor details of the original sound.

Stereo-phonic (Stereo means two) reproduction is based on the fact that we have two ears. If we close one ear for some tome, we can realize the importance of two ears. One ear cannot preserve the depth of sound. The stereo requires broadcasting, recording and reproduction over two separate channels to satisfy our hearing by two ears. Stereo also satisfies our “two eared three dimensional” hearing. The monophonic reproduction cannot differentiate between left and right, up and down or front and back sounds. A stereo may or may not be a hi-fi.


Actually the stereo system involves separation of low and high frequencies. Human voice has low frequency, while the musical instruments have high frequency. The system records, amplifies and reproduces them on separate channels. Each channel may have one or more speakers.

Recording of CD (Compact Disc)

CD Recording: 

For recording, first a master disc is prepared. The laser beam acts as a stylus, which is modulated by digital audio signal. For this the audio signal to be recorded is obtained from the microphone, amplified and sampled @ 44.1 KHz with digital quantization of 16 bits.
Thus 7 mega bits (44.1 x 1000 x 16) per channel per second are used to modulate the laser. This modulated laser is made incident on the CD. When laser beam is ON, the output is 1 and when the beam is OFF, the output is 0. It forms pits on the photo resist material of the CD.

CD Playback: 

For replay a disc, a focused laser beam of suitable wave length, is made incident on the CD tracks through a mirror, A half polished mirror is used, as it allows the laser to pass through it, but does not allow it to return. The reflection of laser beam from pits represents 0 and from space represents 1. This reflected beam is again passed through a focusing system and then through a photo diode is fed to a digital to analog converter. The analog signal so obtained is fed to a loudspeaker for reproduction.


Tuesday, 29 April 2014

Need of DC and AC Biasing

Need of DC and AC Biasing: We have studied Hysteresis curve for magnetic materials which shows that the curve (characteristic) is non linear. While recording and reproduction, the magnetic tape passes through the hysteresis curve. The magnetizing force (H) is determined by the number of turns on the head winding, current through it, produced magnetism (B) and the induced magnetization in the tape.

When a particle on the magnetic tape comes across the head gap, it is magnetized upto E (Figure). When the particle leaves the head, though there is no magnetizing force (H) on it, even it has some magnetism called residual magnetism (OF in the figure).

This residual magnetism leads to distortions. Therefore, a technique is used during recording, reproduction and erasing, which makes the above operations only in the linear region of the hysterists curve, which makes the operations free of distortions and noise. This explains the need of biasing, for which Biasing oscillators are used.

Biasing is of two types: DC Biasing and AC Biasing.

1.DC Biasing: This is done by adding a constant direct current (DC) to the audio signal to be processed and is operated in the linear region of hysteresis curve. This gives a distortion free output.

Advantages:

1.It improves the quality of the reproduced signal.
2.Output is free of distortions.

Disadvantages:

1.Enough dc bias is required for getting desired results.
2.The amplitude of the signal that can be processed is small.
3. The tape remains magnetized due to dc bias, even there is no audio signal. This produces a continuous hissing sound during replay.

2.AC Biasing: In this, high frequency alternating currents (AC) of the order of 100 KHz is added to the audio signal. The output is obtained on both sides of the magnetizing curve.
It is also be noted that the recording signal and the high AC bias are simply added without any modulation and the frequency of the pure sinusoidal ac bias voltage should be about four times the highest frequenct to be recorded.
Advantages:

1.Higher audio signals can be recorded.
2.Noise is reduced.

Disadvantages:

If AC bias applied is not proper:
1.Distortion  is increased, if the bias is low.
2.If the bias is higher, high frequencies are erased from the tape.

Comparison of AC and DC Biasing:

DC Biasing
AC Biasing
1. Here a constant DC is added to the audio signal to be processed.
1. Here high frequency AC of the order of 100 KHz.
2. It is operated in the linear region of hysteresis curve.
2. It is operated on both sides of the magnetizing curve.
3. The amplitude of the signal to ber processed is small.
3. Higher audio signals can be recorded.
4. The output is free of distortion.
4. Distortion increases if the bias is low.
5. Signal to noise ratio is small (25 db).
5. Signal to noise ratio is large (60 dB).


Magnetic Recording and Reproduction of Sound

Magnetic Recording and Reproduction of Sound: This is widely used system for recording audio signals. The basic principle involved in magnetic recording is the phenomenon of electro-magnetic induction. The audio and video tapes are coated with a magnetic material. The signal (audio or video) currents induce magnetism in the tape. The track is in the form of magnetic path on the tape.
For audio recording, the audio signal is given to the microphone which converts it into electrical pulses. This electrical signal is amplified and the amplified current is sent through the audio head coil. If magnetises the head. The magnetic strength of the audio head goes on changing in accordance with intensity of the audio signal being produced at the microphone.

The magnetized audio head is moved on the audio tape. The audio tape is covered with a layer of ferromagnetic material like ferric oxide. As a result tape is also magnetized; the strength goes on varying according to the signal. By magnetic induction, “magnetic track” of sound is formed on the tape which contains audio signal.

The voltage obtained at the output of amplifier should not have any distortion otherwise during replay; the same will come into play.

The core of the head is made of soft alloy, hence it does not have any residual magnetism, therefore it will not give any distortion. However, “demagnetisers” may be used to demagnetize the head immediately after it has done its job.


Magnetic Recording Methods:

For magnetic recording, generally the following methods are used.
1. Longitudinal Magnetic Recording
2. Transversal Magnetic Recording
3. Perpendicular Magnetic Recording

1. Longitudinal Magnetic Recording: In this method, the tape is moved longitudinally to the recording head. The North and South, both poles are kept on same side of the tape.
The method is simple and gives satisfactory results. This is commercial and a very popular method of recording.

2. Transversal Magnetic Recording: In this recording, the poles of the record head are kept transversal to the tape. In this case the record head needs special construction and also it is a complicated method. Results are also not so satisfactory as in the first case, hence it is not used at commercial level.

3.Perpendicular Magnetic Recording: In this method, the tape moves normal (at perpendicular) to the poles of the record head – one pole on each side.


Reproduction or Replaying: For reproduction of the audio signal “play head” is moved on the tape. This activates the play head. The recorded magnetic path (sound track) induces currents in the play head coil (by electromagnetic induction).These induced currents are amplified and given to speaker which by transducer action converts the currents into the original sound. 

Loud Speakers Types

Loud Speakers : A loud Speaker is a device which is actuated by electrical energy and radiates acoustic energy. The selection and installation of a speaker as well as its design should be guided by the problem of coupling an electrical signal source as efficiently as possible to an acoustic load. This involves the determination of acoustic load or radiation impedance and selection of a diaphragm, and means of coupling the loaded loud speaker to an electrical signal source. The performance of the speaker is intimately dependent upon the nature of its acoustic load and should not be considered a part of it. The nature of the radiating system and therefore the acoustic load impedances it sees, is primarily determined by space, acoustical environment and cost factors.

In other words, a loud speaker is an electro-acoustic transducer normally intended to radiate acoustic transducer normally intended to radiate acoustic power into the air so that it is effective at a distance. In order to accomplish this, the loud speaker must be designed in such a way that it will cause the varying electric currents to set in vibration a diaphragm. The vibration of the diaphragm in turn sets the surrounding air molecules into motion. The vibration of this comparatively large volume of air produces the sound, and received by the ear.

The efficiency of a loud speaker is defined as the ratio of the useful acoustic power radiated, to the electrical power supplied to the load and is very low even in the most carefully designed systems (maximum of about 30 percent).

Generally, Loudspeakers consist of two main parts:
1. The driving unit or motor which changes the varying currents of the audio frequency amplifier into mechanical vibrations.

2. The other part is that which acts in conjunction with the driving unit to produce the vibration of the air molecules, and consists of a surface of various geometrical designs such as conical or flat shaped horns.

The horn has been known and widely used for centuries for increasing the radiation from a second source. The most commonly used are conical and exponential types.
The conical horn may be defined as one in which the cross-sectional area of the horn varies in direct proportion to its length, whereas in exponential form the area of the horn varies as an exponent of its length.
Loud Speakers are of general classes or types depending upon the principle involved in operation of the driving circuit.Loud Speakers Types,

Moving Coil (Magnetic) Loud Speaker:

In this type, the moving iron driving type is employed. The principle of operation is based on the varying of the magnetic polarity of the armature. These variations are caused by the electrical impulses flowing through the coil winding which encircles the armature.


The movement to the armature is effected by the induced magnetism, causing it to oscillate between the two poles of the permanent magnet.

Dynamic Loud Speakers:


A speaker of this type consists, principally of a field coil, a voice coil and a cone. The field coil is connected to a dc source, effecting a strong magnetic field across an air gap in which the voice is inserted. The signal current from the output terminal of the receiver, flowing through the voice or moving coil placed around the middle pole of a three pole magnet, causes the voice coil to oscillate corresponding to the oscillations of the signal current. The diaphragm being mechanically connected with the voice coil oscillates in a similar manner.

Microphones Types

Microphones: A Microphone is a device that converts the sound energy into electrical energy. Conversion of sound energy into electrical energy is important, because in this form it can be amplified, transmitted over electrical transmission lines and even radiated through free space over long distances. Without this conversion, the modern electronic communication systems like telephone, radio and satellite communications etc., would not have become a reality. Thus, from this pointof view, a microphone may be considered as the back-bone of the modern communications. Various types of microphones are discussed below.Microphones Types,

1. Dynamic Microphone:

A Dynamic Microphone works on the principle that if a conductor cuts a magnetic field, emf is induced in it. Sound waves striking a diaphragm cause a relative movement between a magnetic field and a conductor, there by inducing a voltage in the conductor. The figure shows the simplified construction of a moving coil dynamic microphone.

An aluminium foil is used as a diaphragm, one side of which is exposed to sound waves. On the other side of this diaphragm is attached a light weight former with a small coil wound over it. The coil is situated in the magnetic field produced by a strong permanent magnet and is free to move in the magnetic field.

Sound waves striking the diaphragm cause it to vibrate, giving to and fro-motion to the coil across the magnetic lines of field and inducing alternating voltage in the coil. The AC signal produced has a frequency equal to the frequency of vibration of the diaphragm. Amplitude of this signal is proportional to the extent of the diaphragm vibrations and hence on the intensity of sound waves.

The dynamic microphone has a flat frequency response in the range of about 60 Hz to 10 KHz. It does not require any polarizing battery for its operation. It can be built to have a light weight, small size and rugged construction at a comparatively low cost. Depending upon the intensity of sound waves, the signal output of this microphone loes in the range of -30 to -80 dbm (0 dbm = 1 mW).

The moving coil microphone has very low impedance, that is, in the range of a few ohms only. A step-up transformer is, therefore, always incorporated in the microphone casing to increase the impedance level, thereby matching it to the input impedance of the circuit where it is used.

2. Ribbon Microphone:

Another important microphone that operates on the principle of magnetic induction is the velocity-ribbon microphone. Figure shows the constructional features of this microphone.


In the ribbon microphone, the magnet is mounted below the pole pieces. The ribbon is made of very thin and pure aluminium foil, which is corrugated to give flexibility and prevent curling. This thin metallic foil ribbon is placed between two elongated pole-pieces. The ribbon acts as the diaphragm as well as the conductor. The ribbon has an extremely small resistance (usually less than 1 ohm).
The pole-pieces are shaped to provide a cavity on each side of the ribbon. These cavities are broadly resonant and ensure that the high frequency response is maintained.

The extremely low impedance ribbon necessities the use of a step-up transformer, similar to the dynamic microphone and a transformer is, therefore, invariably incorporated in the microphone housing.

3. Crystal Microphone:

If certain materials are mechanically deformed by bending or twisting, a difference of potential is produced between the faces of the material. This is termed as the Piezo-electric effect and is exhibited by crystals of quartz. Rochelle salt and few other materials. Rochelle salt (sodium potassium tartrate) is much more sensitive than quartz and is, therefore, commonly used in crystal microphones.

These plates of the material are cut from the crystal of Rochelle salt and are coated with a conducting material on their large faces. Two such plates are joined to form a pair with inner conducting surfaces being in contact and forming one of the output points, as shown in figure.


The two outer faces are joined together and form the second output terminal. Such an arrangement is termed as bimorph element and is found to be more sensitive than a single crystal.

Microphones employ bimorph crystal elements in different forms. In one form of crystal microphones, a pair of bimorph elements is fixed in a frame parallel to each other, with a small distance between them. The inner cavity is made air-tight. The arrangement is termed a sound cell. In another form of microphone, the sound pressure is made to act on a diaphragm, rather than the crystal elements and the diaphragm is made to actuate the crystal unit. This form of microphone gives larger output, because the diaphragm can be designed to provide matching between relatively low impedance of the air and the high impedance of the crystal.

Crystal Microphones exhibit high capacity impedance and can be directly connected at the input circuit of a JFET or tube amplifier.

4. Capacitor Microphone:

The Capacitor Microphone operates on the principle of a change in the capacitance. When subjected to incident sound waves. Figure gives the simplified representation of a condenser microphone. The capacitor microphone consists of a thin light metal alloy diaphragm and a heavy metallic back plate. These two act as the plates of a parallel plate capacitor. The stretched diaphragm is of circular shape and is clamped around its edge. The diaphragm is positioned in front of the back plate but is insulated from it. A DC polarizing potential is applied across the capacitor plates.


The alternating pressure of incident sound waves on the diaphragm causes it to vibrate. This gives rise to an alternating change in the capacitance of the microphone and causes an alternating voltage to be developed across the microphone, provided the load resistor R is large enough to prevent appreciable change in the charge on the microphone electrodes. The alternating voltage across the microphone due to changing sound pressure is equal to the voltage across the load resistor R.

The capacitor microphone has a very low output as compared with the carbon microphone and requires amplification. The output is relatively free from non-linear distortion and has a fairly uniform response. The microphone has high and capacitive impedance. A very commonly employed capacitor microphone is called the electret microphone and is employed in most of the commercially available magnetic tape-recorders, because of its extremely small size. A junction field effect transistor is always included in the electret microphone capsule to raise the signal level.


Frequency Response and Equalization

Frequency Response and Equalization: In audio systems i.e. AF amplifiers, microphones, loud speakers, various types of recording and reproduction of sound, the response is not uniform to various frequencies of Audio Signal. Due to this, frequency distortion results. The quality of the sound is not good. Fidility will be poor.

In AF voltage amplifiers, the voltage gain falls at low frequencies (BASS) due to reactive coupling elements. At high frequencies (Trebble), the gain falls due to inter-electrode capacitances and stray capacitances.

In Accoustic devices i.e., microphones, loud speakers, the frequency response is not uniform. When sound is recorded/ reproduced either by Disc/ Tape/ Film/ CD it is observed that frequency response is not uniform.

To cite an example, while recording/reproducing sound on magnetic tape, the dimension of head-gap has an effect on its performance.

Similarly while recording/replay of sound on Disc (Grammaphone), the frequency response is not uniform.

Equalizer: This is a network having opposite frequency response of the audio circuit/ device, which when included in the circuit gives an output of uniform frequency response. Thus, fidility of the audio system improves. For example, pre-emphasis and De-emphasis networks are equalizers.

Pre-emphasis and De-emphasis Curve:

The high frequencies have low energy content. So, Emphasize the signals above 1 KHz at 6db per octave. During playback, the higher frequencies are more attenuated. Curves are shown in figure.
In sterio-systems, para-meteric or graphic equalizers are adopted. In parameteric equalization, a boost up to ± 15 db is possible. If there is an unwanted peak in response at a particular frequency, it can be cut-down.

In graphic equalization, slider-type Potentio meters are arranged on front panel of the equipment. The AF is divided into a number of bands (3 or 6 or 8 or 10 bands). Each band of frequencies is provided with a control. The user will be able to adjust the shape the frequency response curve as per his desire.

Equalization:

This is a process of compensating for the losses at both low and high frequencies. Electronic circuits can be used to over-come the frequency dependant losses while recording and replay. When it is done while recording, it is called ‘pre-equalization’ and if it is done in player it is called ‘post-equalization’.

A process called pre-emphasis is adopted to enhance low intensity sounds prior to recording. While replay, a process called De-emphasis is adopted to obtain original signal. Other wise fidility will be lost. Hence, combined process of pre and de-emphasis can be called equalization.

Equalization has to be standarised for simple playback. There are CCIR, IEC, DIN and NAB standards. The basic difference between the CCIR and NAB standard is the bass boost at low frequencies for NAB. Bass boost improves hum rejection. But CCIR standard prefers simple equalization. Currently Dolby’s method of equalization is popular.

In pre-emphasis circuit, the capacitor which is in series offers low reactance at higher frequency. Thus, attenuation for ‘Trebble’ is low.


In De-emphasis circuit, the capacitor is in paralleland output is developed across it. The capacitive reactance is low at trebble and output is less for high frequencies. 

Properties of Sound

Properties of Sound: Some of the important properties/characteristics of sound are as given below:
Intensity: While propagating from source, intensity of sound decreases as distance from source increases, because sound gets distributed in all direction as they travel. It is inversely proportional to the distance from the source.

It can be defined as the average rate of transmission of energy in a given direction measured over one square meter of area at right angles to the direction. It’s unit is watts per square meter (w/m2). Intensity of sound is usually represented in decibels (db) with reference to Threshold of Hearing.

Threshold of Hearing: The ear is very sensitive to sound intensity. It can sense very low sound intensity (10-3 w/m2). However, sensitivity of ear is not same for all frequencies.
The minimum sound intensity which an ear can hear is called Threshold of Hearing. It varies with frequency usually at 1000 Hz frequency is taken as standard (10-12 w/m2).

Threshold of Pain: The upper limit of hearing by human ear without causing pain is called Threshold of Pain. This also varies with frequency.
The power Ratio between Threshold of hearing and pain is very large i.e., 1012. Its high to low power ratio is very large, Decibel (db) scale is generally used. In this method, when Threshold of Pain is 120 db above.

Loudness: This is subjective quantity which can not be measured like Intensity. Intensities of tones at 1 KHz are taken as standard levels for comparison. It is observed that two tones of different frequencies with same intensity will not cause equal loudness. PHON is the unit of equal loudness. The Threshold of Hearing is taken as ZERO Phon.
Hence, Threshold of pain is equal to 120 phon. Unit for loudness is Sone.

Pitch – Frequency Relation: Pitch is decided by frequency of sound wave. When frequency is constant, intensity can change Pitch. The unit of Pitch is Mel which is defined as below.
1000Hz tone having a loudness of Sone produces a pitch of 1000 mels.
The relation between pitch and frequency is Non-linear. A frequency of 1 KHz causes a pitch of 1000 mels. But a frequency of 2 KHz causes a pitch of 1500 mels only. Similarly 400 Hz frequency causes 500 mels pitch.

Harmonics: Overtone and timber

Harmonic: An integer ratio between two frequencies is called Harmonic. 2 KHz is second harmonic of 1 KHz. Similarly 4 KHz is fourth harmonic of 1 KHz.
Harmonic should not be considered equal to octave. In Above example,
1 KHz and 2 KHz are of one Octave.
4 KHz and 1 KHz are of two Octave.

Harmonics are Integers multiples of fundamental frequency where as Octave can be in fraction.

Over Tone: All frequencies greater than fundamental including harmonic are termed as overtone. Music is Harmonic blending of fundamental frequency and Harmonics.


Timber: The quality of sound as heard by the ear is called Timber. Timber depends on the proportion in which Harmonics are present. We identify different tones produced by Persons or Instruments by Timber.

Monday, 28 April 2014

Nature of Sound

Nature of Sound: Sound is a longitudinal wave motion. It has compressions and rarefraction in medium of travel. Air acts as medium for sound propagation. Sound follow simple Harmonic motion.

When sound wave strikes the ear drum, they are changed into Audio signals. These signals are carried by nerves to the Brain, where they are sensed as sound. Sound has three dimensional motion in air.
When source of sound i.e., tuning fork is gently hit by wooden hammer, a pure tone is generated. This tone remains for some time and gradually decreases.

Due to vibrations of tuning fork, compressions and rarefractions are produced in air. When they reach the Human ear, they cause sensation of sound. Observe that the direction of propagation is longitudinal. There are compressions and shown with closed vertical lines and rare fractions have been shown with vertical lines which are apart. The changes in pressure produced by compressions are shown by curve of continuous line, whereas pressure changes produced by rarefraction are shown in figure.

A wave has Amplitude, frequency, wave length, phase and velocity. Let us know about them.
Amplitude: It is the intensity of compression and Rarefraction produced in the medium. For a pure tone the intensity can be represented by a sine wave.

A cycle has compression (+ve side) and rare fraction (-ve side).

Frequency: The number of repeated compressions and rare fractions in one second is termed as Frequency. Unit of frequency (f) is cycles/second or Hertz (Hz).

Time Period: The time taken to complete one cycle of sound wave.
Time Period, T = 1/f Hz Seconds.

Wave Length: Wave Length is the distance travelled by the sound wave in one cycle. It is denoted by λ.      λ = V/f
V = Velocity of sound in meters / second.
f = Frequency in Hertz
λ= Wave length in meters

Velocity: Distance travelled by the sound wave in one second. The velocity of sound in space is 344 m/s at 20 degree Celsius. At 0 degree Celsius it is 332 m/s. It is clear that sound travels slowly when compared with light.

The relation between Velocity and Temperature can be given by,
V2 = V1 √(T2/T1)
V1 = Velocity at T1 Degree Kelvin
V2 = Velocity at T2 Degree Kelvin

So, Velocity of sound is affected by Temperature, but not by pressure.
Also, velocity of sound depends on density of Atmosphere.
V = √(rp/d)
V = Velocity in m/s
p = Pressure of gas in dynes / cm2
d = Density in gm/cm3
r = A constant of medium of propagation for air. It is 1.41.

So, we can observe from the above, velocity of sound is inversely proportional to density of medium.
Velocity of Sound is slow in gases.
Velocity of Sound is faster in liquids.
Velocity of Sound is fastest in Solids.

Phase: It gives instantaneous motion relative to some reference. Phase can be expressed in angle or wave length or time period.The phase angle shown above cvan be mentioned as 90 degree or T/4 or λ/4.

Audio and Video Systems

Audio Systems: An audio system is a system which involves an audio signal. In other words, Audio Systems “process” sound. The examples of audio systems/equipment/components are: Radio,Tape Recorder/Player/Stereo, Microphone, Loud Speaker and Telephone.

Video Systems: A Video system is a system which involves a video signal. In other words, video system “process” light i.e., picture. The examples of video systems/equipment/components are : the picture tubes which are used in television, VCO, VCR, etc. Camera is also a video system.

Audio Video System: An Audio/ Video system is that which simultaneously process sound as well as picture signals. The examples of AV Systems are:

(a) Human AV systems: Human AV systems compresses of two ears and two eyes.
(b) Man Made AV systems: Television, VCP, VCR, Cinema Projector, Radar, Picture Telephone

Importance of AV systems: Everybody know the importance of Radio, Tape Recorder, Telephone, TV, VCR etc. These have been the necessity of every home now.

Now AV Systems have very much entered also in the field of education. Video cassettes have been prepared and these are displayed in “Video Schools”. Lectures of eminent professors/ educationists are prepared on all subjects and same can be viewed on TV at home. There is no need to join computer classes. All computer languages can be learnt just sitting in the bedroom and watching a TV.

Now it has become a fashion to prepare video films on all occasions. It may be a birthday party, or a funeral procession, we can keep our memories afresh everytime. In old age, we can see our childhood.
AV systems have entered also in the world of crime. Video cameras can be installed in any place or house. All activities of smuggling/ killings can be recorded on a video. The same can be presented in the courts as a witness. How a criminal will deny!

Now in medicinal sciences, say a patient is being operated upon in the operation theatre. A video cassette can be prepared of the same and patient can see at any time later on. Further by using close circuit TV. The relatives of the patient sitting outside can see what is happening inside the operation theatre.