Tanvi Dhingra Report

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(6 TH JUNE-1 ST JULY 2011) Submitted By: TANVI DHINGRA College: SHAHEED RAJGURU COLLEGE OF APPLIED SCIENCES FOR WOMEN 1 | Page BROADCASTING HOUSE, ALL INDIA RADIO, PRASAR BHARATI, NEW DELHI

Transcript of Tanvi Dhingra Report

Page 1: Tanvi Dhingra Report

(6TH JUNE-1ST JULY 2011)

Submitted By:

TANVI DHINGRA

College:

SHAHEED RAJGURU COLLEGE OF APPLIED SCIENCES FOR WOMEN

CERTIFICATE

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BROADCASTING HOUSE, ALL INDIA RADIO,

PRASAR BHARATI,NEW DELHI

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This is to certify that this Project Report is the Bonafide work of:

TANVI DHINGRA

From: SHAHEED RAJGURU COLLEGE OF APPLIED SCIENCES FOR WOMEN

who have successfully carried out their Summer Training of SIX WEEKS under my supervision and guidance and I found them sincere towards their industrial summer training.

(Trainer In-Charge) Mr. S. C. PACHAURI Assistant Station EngineerALL INDIA RADIO, PRASAR BHARATI

ACKNOWLEDGEMENT

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This report is outcome of SIX WEEKS .Practical training which I received at Prasar Bharati (ALL INDIA RADIO), New Delhi. It includes the Structure, Procedure, function & performance of the station. Through this project report I would like to thank numerous people whose consistent support and guidance has been the standing pillar in architecture of this project.

The SIX Week Training Curriculum contained the study of following substations:

1. CES/STL2. BH Recording, Editing, Dubbing & Website Management3. Master Switching Room (MSR)

To begin with, my sincere thanks to Mr. S. C. Pachauri (Astt. Station Engineer), who provide me with the opportunity to undergo training in their reputed organization.

I would also like to thank and all the staff members who guiding me through the path of practically experiencing the art of broadcasting in A.I.R.

Last but not the least, I would like to extend my heartiest gratitude to all Teaching Staff, who took our classes and clarified the basic concepts, practical implementation of the technologies and were patient enough to answer all our doubts and queries apart from emphasizing upon the necessity of doing things on our own.

CONTENTS

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1. OVERVIEW1.1 Introduction1.2 History of AIR

2. CAPTIVE EARTH STATION (CES)2.1 Architecture of a Satellite Communication System2.2 Radio Networking Terminal2.3 Up-Link/Down-Link Chain2.4 DSNG Vans

3. STUDIO TRANSMITTER LINK (STL)

4. BROADCAST HOUSE (BH) STUDIOS4.1 List of Equipment’s in studio4.2 Studio Acoustic4.3 Studio Chain4.4 Sound Mixing4.5 Recording Room4.6 Dubbing Room4.7 Console Tape Recorder (CTR)4.8 ALL INDIA RADIO Website

5. MASTER SWITCHING ROOM (MSR)5.1 Control Room5.2 Master Switching Facility5.3 Master Switcher and Matrix Management5.4 Software Tools used in MSR5.5 Outside Broadcasting (OB)

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OVERVIEW

1.1 INTRODUCTION

All India Radio (Abbreviated as AIR), officially known as Akashvani is the radio broadcaster of India and a division of Prasar Bharati (Broadcasting Corporation of India), an autonomous corporation of the Ministry of Information and Broadcasting, Government of India. Established in 1936, today, it is the sister service of Prasar Bharati’s Doordarshan, the national television broadcaster.

The word Akashvani was coined by Professor Dr. M.V. Gopalaswamy for his radio station in Mysore during 1936.

All India Radio is one of the largest radio networks in the world. The headquarters is at the Akashvani Bhavan, New Delhi. Akashvani Bhavan houses the drama section, the FM section and the National service. The Doordarshan Kendra (Delhi) is also located on the 6th floor of Akashvani Bhavan.

The National Channel of the All India Radio in radio’s 3 tier system (National, Regional and Local) was originally conceived as the first stage i.e. National system with a broadcast of an 18 hrs. per day. But for various reasons the channel was limited to nighttime service taking the National programs and covering 65% of area and 76% of population of the country. Presently it broadcasts programs of entertainment and music with hourly News Bulletins for the entire country. This is the only channel available after most of the stations closed down.

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1.2 HISTORY OF ALL INDIA RADIO

Sound broadcasting started in India in 1920s with the proliferation of radio clubs. The radio club of Mumbai broadcast the first radio program in India in June 1923. It was followed by the setting up of a Broadcasting Service that began broadcasting in India in July 1927 on an experimental basis at Mumbai and Kolkata simultaneously under an agreement between Government of India and private company called the Indian Broadcasting Company Ltd. The operations of All India Radio began formally in 1936, as a government organization, with clear objective to inform, educate and entertain the masses.

When India became independent, the AIR network had only six stations located at Delhi, Mumbai, Kolkata, Chennai, Lucknow and Triuchirapalli with a total complement of 18 transmitters- six on the medium wave and the remaining on short wave. Radio listening on medium wave was confined to urban limits of these cities. The coverage was 2.5% of the area and just 11% of the population.

AIR today has a network of 232 broadcasting centers with 149 frequency (MW), 54 high frequency (SW) and 171 FM transmitters. The coverage is 91.79% of the area, serving 99.14% of the people in the largest democracy of the world. AIR covers 24 languages and 146 dialects in home services. In External services, it covers 27 languages, 17 national and 10 foreign languages.

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CAPTIVE EARTH STATION(CES)

Satellite Communication is the outcome of the desire of man to achieve the concept of global village. Penetration of frequencies beyond 30 Mega Hertz through ionosphere force people to think that if an object (Reflector) could be placed in the space above ionosphere then it could be possible to use complete spectrum for communication purpose.

Advantages of satellite Communication

The following are the advantages of satellite communication

- This is only means which can provide multi access two way communication. Within the coverage area, it is possible to establish one way or two way communication between any two points.

- The cost of transmitting information through satellite is independent of distance involved.

- Satellite can be used for two way communication or broadcast purpose with the covered area.

- Satellites are capable of handling very high bandwidth. Normally any satellite can accommodate about 500 MHz in C Band. For example the bandwidth of INSAT-I is 480 MHz in C Band and 80 MHz in S Band. INSAT-II has a bandwidth of 720 MHz in C Band and 80 MHz in S Band.

- It is possible to provide large coverage using satellite. For example Geostationary satellite can cover about 42% of earth surface using global beam.

- Satellite can provide signal to terrestrial uncovered pockets like valleys and mountainous regions.

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- Satellites can provide uniform signals for urban areas or rural areas unlike terrestrial service which will lay more signal to urban areas (where the transmitters are located) as compared to rural areas.

- It is easy and quicker to establish new satellite link using SNG terminal or VSAT terminal from any point to any other point as compared to any other means.

2.1 Architecture of a Satellite Communication System

Figure 1 shows the various components of a Satellite Communication System. Basically it comprises two elements :

a. Ground Segmentb. Space Segment

The Space SegmentThe space segment contains the Satellite and all terrestrial facilities for the control and monitoring of the Satellite. This includes the tracking, telemetry and command stations (TT&C) together with the Satellite control centre where all the operations associated with station-keeping and checking the vital functions of the satellite are performed. In our case it is Master Control Facility (MCF) at Hassan.

The radio waves transmitted by the earth stations are received by the satellite ; this is called the up link. The satellite in turn transmits to the receiving earth stations ; this is the down link. The quality of a radio link is specified by its carrier-to-noise ratio. The important factor is the quality of the total link, from station to station, and this is determined by the quality of the up link and that of the down link. The quality of the total link determines the quality of the signals delivered to the end user in accordance with the type of modulation and coding used.

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Fig. Architecture of a Satellite communication system

The Ground Segment

The ground segment consists of all the earth stations ; these are most often connected to the end-user’s equipment by a terrestrial network or, in the case of small stations (Very Small Aperture Terminal, VSAT), directly connected to the end-user’s equipment. Stations are distinguished by their size which varies according to the volume of traffic to be carried on the space link and the type of traffic (telephone, television or data). The largest are equipped with antenna of 30 m diameter (Standard A of the INTELSAT network). The smallest have 0.6 m antenna (direct television receiving stations). Fixed, transportable and mobile stations can also be distinguished. Some stations are both transmitters and receivers. Others are only receivers ; this is the case, for example with receiving stations for a satellite broadcast system or a distribution system for television or data signals.

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Space Geometry

Types of Orbits:

The orbit is the trajectory followed by the satellite in equilibrium between two opposing forces. These are the force of attraction, due to the earth’s gravitation, directed towards the centre of the earth and the centrifugal force associated with the curvature of the satellite’s trajectory. The trajectory is within a plane and shaped as an ellipse with a maximum extension at the apogee and a minimum at the perigee. The satellite moves more slowly in its trajectory as the distance from the earth increases.

Factors deciding the selection of Orbit: The choice of orbit depends on the nature of the mission, the acceptable interference and the performance of the launchers :

The extent and latitude of the area to be covered. The elevation angle of earth stations. Transmission duration and delay.

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Interference

2.2 RADIO NETWORKING TERMINAL

Introduction

The various All India Radio stations spread throughout the nation are required to relay certain programmes which are originating from Delhi. Similarly there are certain programmes which are originating from capital stations are relayed by the other stations in that region. In order to link Delhi and capital stations with other AIR stations, RN through INSAT is not only cost effective but also provide the good technical quality as compared to DOT lines and SW linkage.

The Radio Networking terminal located at AIR stations receive S-Band or C Band transmissions. The programmes thus received after processing are fed to the transmitter for broadcast purposes. Thus RNT acts as the ground terminal for satellite signal reception. The block diagram of S-band RN terminal.

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Fig. Block Diagram of S-Band RN Receive Terminal (RNT)

The RNT system consists of the following units :

i) Outdoor Unit 12 ft. Chicken mesh parabolic antenna and feed unit. Low noise Amplifier (LNA) Front End Converter (FEC).

ii) Indoor Unit Passive Translator Unit (FTP). Active Translator Unit (FTA). Synthesized Translator. Audio Demodulator Power Supply.

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Antenna

Antenna is usually a metallic device (as a rod or a wire) used for radiating or receiving electromagnetic waves. The radio frequency power developed at the final stage of a transmitter is delivered through cables/feeders, without themselves consuming any power to the transmitting antenna. This travels in the free space in the form of radio waves (electromagnetic waves). The receiving antenna picks up the radio waves and delivers useful signal at the input of a receiver for reception of signals. The transmitting and receiving antennae are reciprocal in the sense, any characteristics of the antenna in general applies equally to both.

2.3 Up-Link/Down-Link Chain

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2.4 DSNG VANS

Satellite News Gathering is the use of Mobile Communication Equipment for the purpose of World Wide News Casting through Geostationary Satellites.

The earliest SNG equipment used Analog Modulation, similar to conventional Television and Radio.

During 1990s, Digital Modulation supplanted Analog Modulation giving rise to the newer technology of Digital Satellite News Gathering (DSNG).

Two Types of DSNG’s are generally used:1. Fly Away DSNG2. Drive Away DSNG

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STUDIO TRANSMITTER LINK

(STL)

The high quality sound programs from AIR studio centers are normally transported to the AIR transmitting centers with the help of Department of Tele-communications land lines. These gave way to VHF-FM transmit/receive systems in some of the AIR centers. Now, AIR has introduced the new generation microwave studio-transmitter link (STL) for better reliability and quality.

AIR is having three types of STL called STL-01, STL-02 and STL-05. The numbers 01, 02 and 05 describe the number of base band (50 Hz - 15 kHz) channels that could be transported. STL-01 is taken up for discussion, as it is existing at number of stations. The description of the STL-01 system, the variations of STL-02 mainly in the base band, interface units and the measurements are also described in this article.

GENERAL DESCRIPTION

The Studio Transmitter Link (STL) system consists of a transmitting system (STL-TX) housed in the studio premises and a receiving system (STL-RX) housed in the AIR transmitting centre. A low loss cable connects the STL TX/RX to the two-metre dia microwave dish antenna usually mounted on a 50 m self-supporting tower at either end. In addition, a VHF service channel in duplex mode is provided at both the ends for voice communication between the AIR studio and transmitter ends through a multi-element Yagi antenna mounted on the top of the tower. The need for the

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service channel arises from the fact that there is no RF monitoring facility of the transmitted sound program at STL-TX.

The STL system is meant to operate unattended round the clock. The STL TX/RX is powered by an external power supply unit kept adjacent to the STL rack with floating batteries. This unit takes 230 V ; 50 Hz AC and supplier + 24 DC to STL Tx/Rx. The service channel is energized by another external power supply unit placed over that of STL TX/RX.

I. STUDIO TRANSMITTER LINK - TRANSMITTER (STL-TX)

The STL transmitter essentially consists of eight sections.

i) A single audio input transformer (LT 11) which splits the audio input into two equal audio outputs for (1+1) system.

ii) The base band unit (1+1) consisting of a music amplifier AT 01, and a base band interface unit GT 01 which is a 15 kHz low pass filter.

iii) The radio frequency unit (1+1) which generates the carrier, FM modulates and generates microwave (RF) power. (L1 TR SR04 A/B).

iv) An antenna change over unit which selects one of the (1+1) RF outputs for feeding to the antenna. (L1RF-Tx).

v) A low loss cable connected to a microwave dish antenna at suitable height above the ground.

vi) Two identical (1+1) dc-dc power supplies. (DC-11).vii) Two identical (1+1) monitoring (CM-01) and Alarm inter-face units. (AI 01).viii) One each of logic (LO 11) and parameter control card (PC 02) which selects one

of the RF outputs to be connected to the transmitting antenna

II STUDIO TRANSMITTER LINK – RECEIVER (STL-RX)

The STL receiver essentially consists of the eight sections similar to STL transmitter.

i) 2 m dia microwave dish antenna mounted on a tower of suitable height and a low loss cable connects the received RF power into the receiving system.

ii) Antenna filter and RF hybrid divider unit.iii) The radio frequency receiver unit (1+1) which recovers the base band signal

from the modulated RF carrier.

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iv) The base band unit (1+1) consisting of base band interface unit (GR 01) and the music amplifier (AR 01) and the muting switch (AS 11) which disconnects audio under squeltch operation.

v) A single audio line transformer which provides the audio output.vi) Two identical (1+1) DC-DC power supply units.vii) Two identical (1+1) monitoring (CM 01) and alarm interface units (AI 01).viii) One each of Logic (LO 11) and parameter control card (PC-02) which selects

one of the audio outputs.

III VARIATIONS IN STL 02 SYSTEM

In STL 02, the music amplifier AT 01 of the transmitter and AR 01 of the receiver are not present. Instead, a unity gain card EK 11 is provided to match coder/decoder to the base band interface unit GT01/GR01. Base band equipment is a stereo coder/stereo decoder made by Rhode and Schwarz for the transmitter/receiver respectively. These have no redundancy. The cut off frequency of the low pass filter in the base band interface unit of the transmitter (GT 01)/receiver (GR 01) is 250 kHz instead of 15 kHz in STL 01.

The noise unit monitor filter (BPF) – NF 42 in the carrier monitor unit CM 01 Rx used for evaluating S/N ratio of the received signal has a bandwidth of 100 kHz in STL 01 and STL 02.

IV. SERVICE CHANNEL (RT 33)

The service channel is mounted at the top of the transmitter and receiver racks. It is a VHF (68-88 MHz) trans-receiver. The transmitter output power is 15 watts. Yagi antenna mounted at the top of the towers on either end is used for the service channel. This antenna may be used both in horizontal and vertical polarizations. Normally vertical polarization is used. The hand set with a press to talk (PTT) switch is employed at either end for voice communication. These units can be removed from the racks and kept at any other convenient location at either end. M/s Meltron has developed an interface unit with which telephone facilities can be extended to the transmitter site with this service channel without the use of land lines.

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V. EXTERNAL D.C. POWER SUPPLY FOR STL TX/RX.

The external d.c. power supply unit from IMEC, Ahmedabad operates from the main supply to provide regulated 24V d.c. output at 5 to 7 A. This can change a set of floating batteries besides supplying the STL Tx/Rx for which suitable terminals are provided at the rear of the unit. The performance can be monitored. The current drawn can be boosted between 5 and 7 A with the help of a potentiometer etc. fuse provided can take care of the battery. Buzzer alarm sounds if battery draws more current.

VI. MEASUREMENTS ON THE LINK

The basic minimum measurements that are required to be made on STL links are :

- Frequency response- Harmonic distortion- Signal to noise ratio- Linearity check (level response)

These are not new to the AIR installations. But in the STL links, none of the above can be carried out at the transmitter in isolation. However, it is possible at the receiver end. In such a case for STL-01 one would require equipment like audio oscillator, DNL meter etc. which are normally available in AIR stations besides a microwave signal generator. To carry out such measurements on STL 02/STL 05, one would require microwave signal generator with stereo/PCM modulation capabilities. Therefore, the simplest way in the practical station situation is to take overall measurements from the studio centre to the transmitting centre. This will require co-ordination with the studio end.

Normally, when the system is handed over after installation, manufacturers have already made certain adjustments in various units. One is therefore not required to make any specific adjustments during the measurements. However it is worth while to mention them here.

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- Input at the LT 11 audio input transformer 0 dBm across 600 ohms at 10 kHz- Output of GT 01 (Tx.) Front panel attenuator at 0 dB –14 dBm across 75 ohms.- RF signal level at receiver input shall be better than –60 dBm.- Input to GR 01 – (Rx) –20 dBm across 75 ohms.- Output of audio output transformer 0 dBm across 600 ohms Front panel

attenuator at ‘0’ dBm position.

FREQUENCY RESPONSE

Feed 0 dBm across 600 ohms to the input of the transmitter and measure the output at the receiving end at various frequencies like 50 Hz, 125 Hz, 820 Hz, 1 kHz, 6.3 kHz, 14 kHz and 15 kHz. The frequency response shall be within + 1 dB.

SIGNAL TO NOISE RATIO

Feed 820 Hz at 0 dBm across 600 ohms at the transmitter input. Read the output of the receiver with a selective level meter. Switch off the signal and measure the noise. The signal to noise ratio should be better than 60 dB.

DISTORTIONS

Feed 820 kHz at 0 dBm across 600 ohms at the input of the transmitter. Measure the distortion at the output of the receiver, it should be less than 1%.

CROSS TALK

Feed normal level at studios on 1kHz and note the output level of any wanted channel. Remove the tone from this channel and feed it in any other channel. Measure the noise level in the wanted channel. Determine the cross talk level in the wanted channel below its normal level. This should be better than S/N ratio. Repeat the same with tone in all the other channels. Four such measurements indicate the cross talk of

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other channels on any one wanted channel. Repeat the same at any low and high frequencies also.

BROADCAST HOUSE (BH) STUDIOS

4.1 LIST OF EQUIPMENTS IN STUDIO

EQUIPMENT MAKE

Console Studer 1000Monitor 7500 CompaqKeyboard CompaqMouse CompaqCPU CompaqCD player TascamR-DAT TascamCTR MeltronMonitoring Amp. ComconA-mike stands audio TechnicaUPS (1KV) APCIntercom Usha DX 2000Speaker JBL 4200 seriesAmplispeaker Electro dyne YamahaWall Clock Leitch

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Sync DA SonyfexDigital DA Studer

4.2 STUDIO ACOUSTIC

Introduction

A broadcasting studio is a room in studio complex which has been specially designed and constructed to serve the purpose of originating broadcasting programs. Whenever any musician sings and we sit in front of a performing musician to listen to him, we enjoy the program by virtue of the superb qualities of our sensory organs namely ears. However, when we listen to the same program over the broadcast chain at our home though domestic receivers, the conditions are entirely different. We as

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broadcasters are continuously engaged in the task of ensuring the maximum pleasure for the listener at home when the artists are performing inside the studios.

In order to achieve our goal we must thoroughly understand the characteristic of the different components involved in the broadcast chain, and in this process we must preserve the original quality of sound produced by the artists inside the studio. The science of sound is often called ‘Acoustics’. It would be thus prudent to understand the field of acoustics as applied to broadcasting.

Acoustic Treatment

Good acoustics is a pre-requisite of high quality broadcasting or recording. Acoustic treatment is provided in studios, control rooms, and other technical areas in order to achieve the acoustic conditions which have been found from experience to be suitable for the various types of programmes’. In this section problems and design aspects of internal acoustics of a broadcast studio are explained.

a) Propagation of Sound Waves

Sound waves emanating from a sound source are propagated in all directions. These sound waves are subject to reflection, absorption and refraction on encountering an obstacle. Extent to which each of these phenomenon takes place depends upon the structure and shape of the obstacle, and also on the frequency of sound waves. In close rooms, the sound would be reflected and re-reflected till the intensity weakens and it dies down.

b) Reverberation Time(R/T)

In any enclosed room when a sound is switched off, it takes a finite length of time to decay to inaudibility.

The ‘hanging-on’ of the sound in a room after the exciting signal has been removed, is called ‘reverberation’ and the time taken for the sound to decay to one millionth of its initial value, i.e. 60 dB, after the source has stopped, is termed ‘Reverberation Time’(R/T).

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c) Factor Covering Reverberation Time

R/T of a room depends upon shape and size of room and on the total absorption offered on boundary surfaces.

For a room of given volume and surface area, the R/T can be derived by Eyring’s formula

Where R/T = Reverberation time in seconds

V = Volume in cubic ft.

S = Total surface area of room in Sq.ft.

= Average absorption coefficient

Average absorption coefficient ( ) is given by

Where S1, S2…….Sn are the areas (in sq. ft.) of different materials provided, and 1 , 2

……n are the absorption coefficients of these materials. of acoustic material is defined as the ratio of absorbed sound to the total incident energy of sound. An open window absorbs/allows to pass all of the sound energy striking it and reflects none. Thus it has of unity.

d) Effects of Reverberation on Programme

Reverberation is the most important single parameter of a room. It influences the audio programs in following ways:-Volume of program increases due to reverberation of sound. This is a desirable feature, however, too much of reverberation may impair the quality of program and, therefore, should be controlled.

Reverberation results in prolongation of sound inside the room. This leads to ‘blending of one sound with the next and produces a very pleasant continuity in

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the flow of music. Too much of prolongation, however, may create loss in intelligibility of program due to decrease in clarity.

Reverberation time of a room is dependent on frequency. Therefore, it modifies the frequency characteristics of the total sound field inside the room. High R/T at mid and high frequencies leads to increased ‘liveness’ and that at low frequencies increases ‘warmth’. This effect can be used judiciously for desirable qualities.

e) Optimum Reverberation Time

R/T value at each frequency of sound is fixed for most desirable results for different type of programmes.

Fig. 1 Reverberation Time vs. Volume

Optimum R/T values at other audio frequencies are dependent mainly on the type of programme for which the studio will be used. These values have been decided after detailed study and subjective listening tests. Optimum R/T for talk studio is generally flat, whereas for music, studio, Morris & Nixon’s recommendations (Fig. 2) are followed in AIR. For drama programmes, the optimum R/T is taken as an average of talks and music values at each frequency.

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Fig. 2 Recommendation – MORRIS & NIXON

f) Acoustic Absorbers

Acoustic absorbers are provided on the inner surfaces of the room to achieve optimum R/T characteristics. Different absorbers have different absorption characteristics. No single absorber generally provides uniform absorption over the complete frequency spectrum.

g) Design of Room Acoustic

Design for correct reverberation time consists of estimating the total absorption which must be present in the studio. This is calculated by Eyring’s Formula, some of the absorption is offered by windows, doors, flooring and artists inside the studio. For the balance requirement sound absorbing materials are provided on walls and ceiling surfaces. Calculations are generally made at six spot frequencies of 125, 250, 500, 1000, 2000 and 4000 Hz. Quantities of materials of known absorption coefficients are selected by trial and error method so that R/T requirements are met within +5% of the optimum R/T at all these frequencies. Computer aided design for the same has also

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been evolved. Thereafter these acoustic materials are distributed on various surfaces for proper diffusion of sound in the studio.

4.3 STUDIO CHAIN

Introduction

The broadcast of a programme from source to listener involves use of studios, microphones, announcer console, switching console, telephone lines / STL and Transmitter. Normally the programmes originate from a studio centre located inside the city/town for the convenience of artists. The programme could be either “live” or recorded”. In some cases, the programme can be from OB spot, such as commentary of cricket match etc. Programmes that are to be relayed from other Radio Stations are received in a receiving centre and then sent to the studio centre or directly received at the studio centre through RN terminal/telephone line. All these programmes are then selected and routed from studio to transmitting centre through broadcast quality telephone lines or studio transmitter microwave/VHF links. A simplified block schematic showing the different stages is given in Fig.

Fig. Simplified block schematic of broadcasting chain

Studio Centre

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The Studio Centre comprises of one or more studios, recording and dubbing room, a control room and other ancillary rooms like battery room, a.c. rooms, switch gear room, DG room, R/C room, service room, waiting room, tape library, etc. The size of such a centre and the number of studios provided depend on the programme activities of the station. The studio centers in AIR are categorized as Type I, II, III and IV. The number of studios and facilities provided in each type are different. For example a type I studio has a transmission studio, music studio with announcer booth, a talk’s studio with announcer booth, one recording/dubbing room and a Read Over Room. Type II has one additional drama studio. The other types have more studios progressively.

Broadcast Studio

A broadcast studio is an acoustically treated room. It is necessary that the place where a programme for broadcast purposes is being produced should be free of extraneous noise. This is possible only if the area of room is insulated from outside sound. Further, the microphone which is the first equipment that picks up the sound is not able to distinguish between wanted and unwanted signals and will pick up the sound not only from the artists and the instruments but also reflections from the walls marring the quality and clarity of the programme. So the studios are to be specially treated to give an optimum reverberation time and minimum noise level. The entry to the studios is generally through sound isolating lobby called sound lock. Outside of every studio entrance, there is a warning lamp, which glows ‘Red’ when the studio is ‘ON-AIR’. The studios have separate announcers booths attached to them where first level fading, mixing and cueing facilities are provided.

Studio Operational Requirements

Many technical requirements of studios like minimum noise level, optimum reverberation time etc. are normally met at the time of installation of studio. However for operational purposes, certain basic minimum technical facilities are required for smooth transmission of programmes and for proper control. These are as follows:

Programme in a studio may originate from a microphone or a tape deck, or a turntable or a compact disc or a R-DAT. So a facility for selection of output of any of these equipments at any moment is necessary. Announcer console does this function.

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Facility to fade in/fade out the programme smoothly and control the programme level within prescribed limits.

Facility for aural monitoring to check the quality of sound production and sound meters to indicate the intensity (VU meters).

For routing of programmes from various studios/OB spots to a central control room, we require a facility to further mix/select the programmes. The Control Console in the control room performs this function. It is also called switching console.

Before feeding the programmes to the transmitter, the response of the programme should be made flat by compensating HF and LF losses using equalised line amplifiers.(This is applicable in case of telephone lines only)

Visual signalling facility between studio announcer booth and control room should also be provided.

If the programmes from various studios are to be fed to more than one transmitter, a master switching facility is also required.

4.4 SOUND MIXING

As already mentioned, various equipments are available in a studio to generate programme as given below:

Microphone, which normally provides a level of –70 dBm.

Turntable which provides an output of 0 dBm.

Tape decks which may provide a level of 0 dBm.

CD and R-DAT will also provide a level of 0 dBm.

The first and foremost requirement is that we should be able to select the output of any of these equipments at any moment and at the same time should be able to mix output of two or more equipments. However, as we see, the level from microphone is quite low and need to be amplified, so as to bring it to the levels of tape recorder/ tape decks.

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Audio mixing is done in following two ways:

i) Required equipments are selected and then outputs are mixed before feeding to an amplifier. This is called low level mixing (Fig.). This is not commonly used now days.

Fig. Low level mixing

ii) Low-level output of each equipment is pre-amplified and then mixed. This is called high level mixing. (Fig. 3).

Fig. 3 High Level Mixing

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Low level mixing system may look economical since it requires one single pre-amplifier for all low level inputs, but quality of sound suffers in this system as far as S/N ratio is concerned. Noise level at the input of best designed pre-amplifier is of the order of –120 dBm and the output levels from low level equipment –70 dBm. In low level mixing, there is signal loss of about 10 to 15 dB in mixing circuits. Therefore, the S/N ratio achieved in low level mixing is 35 to 40 dB only.

High level mixing system requires one pre-amplifier in each of the low level channels but ensures a S/N of better than 50 dB. All India Radio employs High level mixing.

4.5 RECORDING ROOM

A block schematic of a typical recording room is shown in figure . Two numbers of CTRs and two numbers of Push Button switches have been shown. Outputs from various studios and switching consoles have been given to multiple pads 1,2,3 and 4. Outputs from the multiple pads are wired to PB switches. Three numbers of receptacles for cassette outputs have been provided. Transformers T1 and T2 transform the output impedance of the cassette recorder to 600 ohm. The output of CTR # 1 is wired to PB switch # 2 through MP # 6. With this arrangement output of CTR # 1 can be recorded on CTR # 2. Please carefully note the impedances and levels at various points. Red and green lamps are provided on the control panel for indications from and to control room and studios.

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4.6 DUBBING ROOM

A block schematic of a typical dubbing room is shown in figure 11. The arrangement is similar to the recording room except that an additional tape deck and a mixer unit have been provided. This arrangement allows mixing of programmes.

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Fig. Block schematic of Recording / Dubbing Room

JACK FIELDS

All the inputs to control panel are routed through a jack field (fig.) which is a single or double row of jack strips mounted on insulated material. These jacks are wired in such a way that for normal set up no patching is needed. However, for by passing any equipment and for changing the equipment from normal setup patching with patch cord is necessary.

A patch cord consists of a hollow brass sleeve, hollow ring and tip. Tip is insulated from ring and sleeve and similarly, ring is also insulated from sleeve. A small thin rod connects the tip and is carred back through insulated bushing to the body of plug where a terminal screw is provided for connection.

Similarly there is a terminal screw provided for ring and sleeve connection. Tip and rings are connected to live balanced line while sleeve is connected to ground (see fig.)

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Normal springs are also called as inners and swingers as outers. When patch cord plug is inserted inside the jack, tip rests on one swinger and ring rests on other swinger there by breaking the contact with normal springs (fig.).

Normally jacks are used in pairs one is called ‘Break Jack’ and the other is called ‘Make Jack’. The inners of the two are permanently connected while outers are connected as shown in Figure .

Fig. Jack-Field Fig. Tip-Ring-Sleeve Patch Chord

Fig. Balanced Jack Fig. Break & Make Jacks

CONSOLE

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Most of the studios have an attached booth, which is called transmission booth or Announcer booth or play back studio. This is also acoustically treated and contains a mixing console called Announcer Console. The Announcer Console is used for mixing and controlling the programmes that are being produced in the studio using artist microphones, tape playback decks and turn tables/CD players. This is also used for transmission of programmes either live or recorded.

The technical facilities provided in a typical announcer booth, besides an Announcer Console are one or two microphones for making announcements, two turn tables for playing the gramophone records and two playback decks or tape recorders for recorded programmes on tapes. Recently CD and Rotary Head Digital Audio Tape Recorder (R-DAT) are also included in the Transmission Studio. Audio block schematic of transmission studio is shown in Fig.

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Fig. Block diagram of Announcer Console

COMPACT DISK PLAYER

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A block diagram giving essential components of a CD player is shown in Fig. Power loading is usually implemented on the players where the disc is placed in a drawer.

Fig. Block Diagram of CD Player Showing the Data Path (Board Arrow) and Control / Servo System

Then the drawer is pulled into the machine, the disc is lowered on to the drive spindle and clamped at the center and this process is known as “chucking”. In simpler top

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loading machines, the disc is placed on the spindle by hand, and clamp is attached to the lid so that it operates as the lid is closed. The lid or drawer mechanism has safety switch to prevent the lower operating when the machine is open. Actually, there is very little hazard in a CD pick up. This is because the beam is focused a few milli meters away from the objective lens and beyond the focal point the beam diverges and the intensity falls rapidly. It is almost impossible to position an eye at the focal point when the pickup is mounted in the player. The data path consists of the data separator, time base correction and de-interleaving and error-correction process followed by error-concealment mechanism. The data separator, which converts the read out waveform into data, is LSI chip developed by Sony and also by Phillips. The separated output consists of sub code bytes, audio samples, redundancy and a clock. The data stream and the clock will contain speed variations due to disc run out and chucking tolerances, and these have to be removed by a time base corrector.

The time base corrector is a memory addressed by counters which are arranged to overflow, giving the memory a ring structure as shown below in fig.

Writing into the memory is done using clocks from the data separator whose frequency rises or falls with run-outs, where as reading is done using a crystal controlled clock, which removes speed variations from the samples, and makes wow and flutter un-measurable. If the time base corrector functions properly, the long-term data rate from disc is equal to the crystal-clock rate. To ensure this, the disc speed is controlled. The disc speed can be controlled by two methods.

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Fig. 13 Block Diagram Compact Disc Player CD02A

MICROPHONE

Introduction

Microphone plays a very important role in the art of sound broadcasting. It is a device which converts acoustical energy into electrical energy. In the professional broadcasting field microphones have primarily to be capable of giving the highest fidelity of reproduction over audio bandwidth.

Microphone Classification

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Depending on the relationship between the output voltage from a microphone and the sound pressure on it, the microphones can be divided into two basic groups.

Pressure Operated Type

In such microphones only one side of the diaphragm is exposed to the sound wave. The output voltage is proportional to the sound pressure on the exposed face of the diaphragm with respect to the constant pressure on the other face. Moving coil, carbon, crystal and condenser microphones are mostly of this type. In their basic forms, the pressure operated microphones are Omni-directional.

Velocity or Pressure Gradient Type

In these microphones both sides of the diaphragm are exposed to the sound wave. Thus the output voltage is proportional to the instantaneous difference in pressure on the two sides of the diaphragm. Ribbon microphone belongs to this category and its polar diagram is figure of eight.

Types of Microphones

There are many types of microphones. But only the most common types used in broadcasting have been described here.

Dynamic or Moving Coil Microphone

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Fig. Dynamic Microphone (Moving Coil)

Ribbon/ Velocity Microphone

Fig. Ribbon Microphone

Electrostatic or Condenser Microphone

Fig. Condenser Microphone

Electret ‘Microphone’

It is a modified form of condenser microphone in which the polarizing voltage is avoided. In fact a plastic polymer containing metallic dust keeps the metal particles permanently charged within the plastic insulation and such a polymer within the diaphragm foil or fixed plate delivers the electrical signal on the principle of the condenser mike. The hissing noise gets avoided since there is no external polarizing

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resistor as a load. The microphone has high impedance and is generally having FET pre-amplifier. The microphone costs very little but develops excellent quality designs in many forms.

Lip Ribbon Microphone

It is also called noise-cancelled mike since the ribbon even if held close, does not pick up breathing noises due to a guard in between. The stainless steel mesh acts as a wind shield. The design and other features resemble the ribbon mike.

Lapel Microphone

Both carbon and ribbon types are available. The microphone is very small and light- weight and is suspended around the neck keeping the mike just below the chin. It is most suitable for running commentary or in a lecture.

Contact Microphone

It is generally a dynamic microphone of lower sensitivity. It is normally placed close to the source of sound, when it is not supposed to pick up other stray noises.

Gun Mike

It has two forms, (short gun and long gun). A dynamic mike placed at the end of a perforated tube extends its directivity in the front. The short gun about 18” long can pick up a talk from about 10 feet distance and a long gun with a tube of about 3 feet length can pick up sound from a distance of about 20 to 25 feet. The quality suffers but is intelligible. This microphone is useful when sound from a distant spot is to be picked up. An example is the picking of the sound of bat hitting a cricket ball.

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4.7 CONSOLE TAPE RECORDER

Magnetic tape recording system has got many special features, which makes it unique in Sound Broadcasting, Television and Computer field. These are instant and simultaneous replay during recording.

The recording medium i.e., the magnetic tape can be used again and again after erasing the previous recordings, which generally takes place along with the recording of the new programme. The editing is simple and accurate. This can also be done electronically, without physically cutting the tape.These facilities combined with excellent quality and reliability has made magnetic recording system very popular in the field of entertainment and all direct recordings are first done on magnetic tape.

The SystemThe magnetic tape recording system may be studied under three sub-systems :

The magnetic system comprising the magnetic tape plus record, replay and erase heads. Tape transport system comprising the two spooling motors, the capstan motor, the brake mechanism and the control cum interlock system. The electronic system comprising the amplifiers equalizers and power supplies etc.

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Recording Principles

The magnetic material used in recording is magnetic oxide of iron Fe2O3 and Fe2O4 or a suitable mixture of the two with small quantities of the oxides of Nickel and Cobalt. This is mixed with suitable adhesives, plasticizers, fillers etc. and applied in the form of an extremely smooth, even and thin coating (0.4 to 0.6 mils) on to a PVC backing (1.0 to 1.5 mils thick). This magnetic coated tape has a remanance of about 500 to 1000 gausses, coercively of about 300 to 500 oersteds. The permeability is rather low (5 to 10). This tape gets magnetized when it comes in contact with a recording head with audio frequency signal currents flowing through the head windings, and as it passes on forward, retains the magnetism induced, due to the magnetic properties of remanance and coercively. Thus if the tape is moved across the head at a constant speed of V cm/sec. and the signal current is of frequency "f" Hz, the signal current variations in time, will be recorded as magnetic intensity variations along the tape length. Thus a single cycle will be recorded on a tape length V/f cm. This is called recorded wavelength and will be given by

So in tape recording, we really record wavelengths and from a recorded wavelength any frequency signal can be obtained by running the tape at play back speeds different from the one at the time of recording. Normally, the record and replay speeds are exactly the same for a faithful reproduction of the recorded signal. For a fixed frequency audio signal pure tone, the magnetic conditions on the so magnetized tape can be approximately depicted in the form of recorded wave lengths condition of an array of half wave bar magnets placed end to end along with the tape length as shown in figure .

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Fig. set up of tape recorder

The Erase ProcessErasing the previously recorded signal is essential for using the tape repeatedly. A satisfactory method for this is to feed the erase head with a high amplitude signal of about 100 kHz and the tape passes over this erase head before it passes on to the record head (see fig. 6). In this arrangement every part of the tape passes the erase head gap (about 15 mil) and is subjected to about 200 cycles of alternating magnetic field, starting from low value at the start of the gap, increasing to saturation value in 44 | P a g e

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the middle of the gap and again steadily dropping to low value of the field, as the tape leaves the gap. These repeated magnetising cum de-magnetising cycles erase the signal completely and leave the tape in completely unmagnetised form similar to a virgin tape without a magnetic history.

The Recording ProcessMagnetic recording is made possible due to magnetism remaining behind after the magnetising force is removed.

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The curve showing the relation between remanant magnetism (Br) and magnetising field (H), indicates that the relation is non-linear in the beginning. Therefore a signal recorded as such will have a high component of harmonic distortion of the wave form as shown in the Fig.8 and is not useful. This can be partially improved by using D.C. bias to avoid the instep of the initial Magnetisation curve (I.M.C) and shifting the operating point to a more linear middle region as shown in the fig. This method of D.C. bias does give a fairly satisfactory quality of recording but the noise and distortion in the reproduced signal leaves much to be desired. Also it may be seen that only one half of the Br-H used suggesting scope of further improvement. This above method of D.C. bias recording is still used in cheaper version or cassette recorders for domestic use. A greater improvement in recording quality was achieved during forties when the high frequency bias, a tap off from the erase head was also applied to the record head along with the audio signal to be recorded. This H.F. bias current agitates the magnetic particles on the tape, sufficiently that they settle down to the flux value, corresponding to the superimposed signal current, overriding the in step non-linear portion of the I.M.C. This is diagrammatically depicted in Fig. Both halves of the Br-h curves are utilized, flux recorded is high, signal to noise ratio is good and distortion is low as a result of H.F. bias.

The Tape Transport System

During record or replay, the tape has to pass across the heads. Also there should be provision for spooling fast rewind. A standard tape transport format used in all professional machines is shown below.

The rewind and forward motors are ordinary induction motors. Their torque depends on the voltage applied. The rewind and F/F motors freely rotate in opposite direction as indicated. In fast forward mode, the F/F mode is supplied full voltage and the R/W motor, a fraction of it, so the tape moves forward under a small reverse drag to keep the tape taut. In rewind mode, the R/W motor is given full voltage and the F/F motor a small voltage. The tape rewinds fast under a small reverse drag. In play back mode,

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the motors are given equal voltage and about half the full voltage. The tape as such remains taut and should not move in any direction, if freed from the capstan motor pinch roller. During record/replay mode the tape must move with constant speed, as otherwise the frequency reproduced will be different from the one recorded. Also the speed should remain constant and not vary around a mean average speed. This simply means that if the nominal tape speed is 7.5 inches/sec. The tape should not only move forward 7.5 inches/sec but also 7.5 mili inches every milisecond to say. This is achieved by the capstan motor and pinch roller combination. The capstan motor is heavy duty, under loaded synchronous motor. Its speed of rotation i.e. no. of revolutions per second, are synchronous to the power supply frequency. Voltage variations have no effect on sub multiple of it, depending upon the pair pole windings. The tape is pinched between the capstan motor shaft and pinch roller and can move forward only when the capstan shaft rotates at the precise rate of 'd' inches per rotation where 'd' is the capstan shaft circumference. Thus the tape speed is maintained constant as short duration frequency variations mains power supply frequency (frequency fluctuations) are rare.

Brake MechanismFor stopping the tape from any transport mode, the supply is switched off to the motors by pressing the stop button and simultaneously applying the brakes to kill the movement of the shutting motors due to inertia. It may be noted that the capstan motor is not switched off. The pinch roller engages with the capstan shaft in record/replay mode only and in shuttling mode, this remains disengaged and as such capstan motor though running continuously does not play any part in tape shuttling. The brake system used is invariably of differential braking type where the braking torque depends upon the direction of rotation of the shuttling motors.

Measurements and AdjustmentsIn the tape transport system following quantities are specified by the manufacturers. These should be checked at least once a week and adjustments carried out if the measured values are not within the tolerance limits specified. Worn out brake bands and pinch rollers need replacement as and when necessary. The measurements and adjustments are :

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- Rewind reel brake torque-clock and anti clockwise.- F/Forward reel brake torque-clock and anti clockwise.- Capstan Pinch roller pressure.- Rewind and F/Forward motor torques in all the three modes separately in

pulling and release form.

THE ELECTRONIC SYSTEMThe electronic system in tape recorders comprises of power supplies, recording amplifier chain including equalizing circuits, playback amplifier chain with play back equalizer, HF bias oscillator, metering and monitoring amplifier etc.

The individual circuit details vary from model to model and detailed study should be done from the particular equipment manual. However, the circuits are quite conventional.

The Recording ChainAs is evident from the above block diagram, the input may be from a microphone or from a high level controlled, amplified, correct amount of HF boost is provided to pre compensate for the HF record process losses as described earlier. Also, in addition to amplification and pre-equalisation, it converts constant voltage input into constant current output. This is required because the record head is a current operated device and the magnetic flux is proportional to the current flowing in the record head coil.

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The Playback ChainThe output of the PB head is rather low and rising with frequency. This means that the output at lower frequencies is rather very low as compared to mid range frequencies. Therefore, great care is taken in electrically and magnetically shielding the P.B. Head to avoid hum pick up. Also the head output cord is kept small and shielded and the first amplifier stage is placed close to the head to minimise hum pick up. Equalising is incorporated after one or two stages of pre-amplification.

Electronic System AlignmentIt is obviously necessary that for every machine the overall record replay chain response versus frequency should be flat. This can be achieved in various ways by complementary equalisation in the recording and replay chains. However, such machines will not reproduce satisfactorily the recording made on other machines i.e. the equipment will not be compatible for programme exchange. For this reason, a standard recorded flux characteristic is generally specified keeping in view the best signal to noise ratio and minimum distortion. The CCIR characteristics is followed by large number of countries all over the world including India and the recording

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machines are standardised using the standard tape and then the recording chain is standardised using the standard play back chain as reference.

The Standard Tape: The CCIR standard tape for 7.5 inches/sec. (10 cm/sec) speed has three sections as discussed below:

Level SectionThis section has 1 kHz signal recorded on it at a recorded flux level of 32 mili Maxwell/mm recorded track length.

This is the maximum permissible level recorded on present day commercially available tapes and such tapes should give a total harmonic distortion of 3% or less when reproduced from a standard playback chain. The average or normal recorded level is 10 dB below this maximum level. For chain alignment the playback output attenuator is set at -20 dB mark and play back amplifier gain control adjusted to give 0 dB reading on the VU meter. The attenuator is advanced by 10 dB i.e. to a reading of -10 dB and left undisturbed. Now when the recording input gain control is so adjusted that the play back chain output reads zero ensures that the recorded level on the tape is 10 dB below reference level or 32 m Maxwell.

Head Azimuth SectionThis section has 10 kHz signal recorded on the standard tape with a record head having true azimuth. The tape is played and the play head azimuth adjusted for maximum quite constant i.e. output variations not more than 0.5 dB. The PB head is left undisturbed. Now a 10 kHz signal is given to the record chain. The record head azimuth adjusted for maximum and stable output from the play back chain, which was earlier azimuth, standardized with standard tape. This completes azimuth alignment of the complete record replay chain.

Frequency Response SectionThis section of the standard tape has recordings on it, various frequencies from 60 Hz to 100 kHz corresponding to the specified recorded flux versus frequency characteristic, as per the CCIR specified impedance Vs. frequency characteristic of an RC parallel connected network of 70 micro second time constant. The reference flux at 1 kH being 10 dB below the maximum specified flux of 32 mili. Maxwell/mm. This 50 | P a g e

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standard tape section is played and play back equalization adjusted to obtain flat frequency response within permissible limits as given below in Fig.

Overall Chain ResponseBy the above procedure using standard tape, the play back chain is standardized for level, azimuth and equalization. Now for over all chain response, a set of specific frequencies from 50 Hz to 10 kHz are fed to the recording chain at constant level and record equalization adjusted till the output monitored at the play back chain output is flat within specified limits as above.

MAGNETIC TAPE

Tensile Strength The tape should stand a steady pull of 3.5 Kg. Wt. and impulse load test of 100 gm. falling from a height of 250 mm.

Elastic Elongation Residual elongation should not be more than 0.3% for a steady load of 1 Kg. Wt. for 24 hours.

Overall Thickness 0.050 mm + 0.005 mm.

Coating Thickness Max. 0.015 mm min. 0.01 mm.

Magnetic Anchorage The magnetic layer should show no evidence of anchorage failure, when an adhesive tape of 1" length is stuck and pulled away.

Plasticity The tape should fall against a sharp edge under its own weight of 5 cm length.

Erasability When a recording at maximum record level is passed through an erase field of 1000 oersteds, the residual signal level should be 60 dB below the recorded level.

Distortion At the maximum recorded level of 32 mili Maxwell/mm at 1 kHz, the total harmonic distortion should not be more than 3% and at 10 dB below the above, the total

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distortion should be less than 1%.

LOUD SPEAKERS

A loudspeaker performs an opposite function to a microphone, i.e. it converts electrical signal into sound wave.

Moving Coil or dynamic loudspeaker: It consists of a permanent magnet and a voice coil for carrying audio signals. Voice coil is having a few turns of wire, wound on paper, plastic or aluminium former. It is attached to a peper that radiates sound. The coil is suspended with the help of “spider”, made of flexible material. Spider permits forward backward motion but no lateral motion.

HEADPHONEHeadphones basically work on the same principles which are applicable to loudspeakers. However, with headphones the acoustical loading is achieved by intimacy of the ear units to the ears. Thus even very small units are capable of providing very good bass performance. Most headphones used for high quality applications are either moving coil or electrostatic. Headphone impedances range from 4 to 1000 ohms. Specifications of a stereo headphone type EM 6201 (Philips) are given below :

Frequency range 20 to 20 kHzMatching impedance 4 to 32 ohmsMaximum input 0.1 watt.

For checking levels on a studio chain headphones with higher impedance should be used. Headphones are classified into mono, stereo and four channel headphones according to the number of channels.

4.8 ALL INDIA RADIO Website

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MASTER SWITCHING ROOM

(MSR)

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For two or more studios set up, there would be a provision for further mixing which is provided by a control console manned by engineers. Such control console is known as switching console. Broad functions of switching console in control room are as follows

Switching of different sources for transmission like News, O.Bs. other satellite based relays, live broadcast from recording studio.

Level equalisation and level control.

Quality monitoring.

Signalling to the source location.

Communication link between control room and different studios.

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Fig. Block Schematic of Control Room

5.2 MASTER SWITCHING FACILITY

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If a single transmitter is to be supplied with a programme, facility for master switching is not required, however when many transmitters are simultaneously being supplied with different programmes or the programmes to be fed to a single transmitter is periodically changing, a switching facility becomes necessary and is provided by Master Switching Console. To understand the requirements of a Master Switching console, an example of feeding of two transmitters from any one of six sources.

5.3 MASTER SWITCHER AND MATRIX MANAGEMENT

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5.4 SOFTWARE

In the AIR we us basically some software for networking, editing, recording, dubbing, routing of program, scheduling for various works. These are:

1. DALETPLUS SOFTWARE:- main software2. NEXUS SOFTWARE:- for execution of switching3. ASPA SOFTWARE:- for scheduling the program4. ADOBE AUDITION AND SOUND FORGE:- for recording and dubbing

DALETPLUS OVERVIEW

DALETPLUS is a professional and industry-recognized line of software that offers a full suite of applications for multi-media editing and digital asset management for professional TV and Radio broadcasters. With DALETPLUS, you'll be able to search and access a wide range of media – newswires, audio, video, web l inks, texts, and contacts – from a single standard desktop PC connected to your company's network. Intuitive collaboration tools make it extremely easy to share your work and distribute it, without leaving your workstation. Integrated editing applications let you modify the content of audio, text, image, and video files that you want to prepare for broadcast or publication. In the role of a studio technician or presenter, you'll find intuitive playout applications. With its integrated set of full-featured modules, DALETPLUS lets you optimize the workflow and media processing in your organization – from media ingest, cataloguing, editing, broadcast scheduling to ONAIR applications, Web publishing and Archiving.

DALETPLUS MODULES

DALETPLUS integrates several software modules in a single collaborative work environment. Different users have access to different modules, based on access rights. Customizable workspaces allow users to organize their desktop so that only the relevant modules are visible at each stage of their work.

The DALETPLUS modules cover the following functionality:

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A) BrowserB) Asset ManagerC) Search D) Media Production ToolsE) Audio RecorderF) Trackfiler PlusG) NewswiresH) EditingI) AudiosurferJ) Activelog Transcription Player

5.5 OUTSIDE BROADCAST’s (OB)

Outside Broadcasts (abbreviated as OBs) form a substantial portion of programmes radiated from a Radio Station. Major events that occur at different parts of a country, such as sports events, important functions of political, cultural and national important and other such programmes that originate from outside the broadcast studio are covered as OBs.

Different Types of OBsOBs can be classified into two types :

i) Live BroadcastEvents of national importance such as Independence Day Celebrations, sports events etc. are generally radiated as Live programme.ii) Spot RecordingsMost of the OB programmes are recorded at the OB spot with the help of a portable, battery operated OB amplifier and or an Ultra Portable Tape Recorder (UPTR) or a cassette tape recorder. Some programmes, depending on their importance are recorded at the studio end. In this case, it is necessary to book telephone lines, from the OB spot to CR. Normally three such lines are booked. One for feeding the programme to CR, one for inter communication between the OB spot and CR using a magneto telephone, and one as a standby programme line.

EQUIPMENTS NORMALLY USED IN OB’s

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i) OB AmplifierAn OB amplifier is a portable mixing unit. Normally four low level microphone inputs and one high level input from a PTR or UPTR, can be mixed and controlled by this unit.

ii) MicrophonesThe choice of the correct type of microphone and its proper handling and placement is very important for the success of an OB. The microphones used in OBs must be robust, insensitive to wind noise and popping effects, and having a good front to back ratio to avoid feedback. Different Types of Microphones used are:Omni directional MicrophonesOmni-directional microphones are sensitive to sound from all directions equally.Short Gun MicrophonesIn OB situations such as cricket test match or athletics coverages, the sound is tobe picked up from a distance and hence we require a microphone with a narrowacceptance angle.Radio/Wireless MicrophonesIn sports coverages, there may be situations such as in a big stadium wheredifferent athletic events take place simultaneously where it is not possible to laycables. Radio microphones are best suited for these locations. Use of Wind shieldsWhen microphones are used out of doors, in windly conditions, wind shields areused.

iii) Tape RecordersSpot interview and glimpses of the various events taking place in a particular city, are covered by spot recordings done with Ultra Portable Tape Recorders (UPTRs) and cassette tape recorders.

OB Van

AIR has acquired a few OB Vans recently. The vans are of the size of a matador vehicle and incorporate equipment of latest technology. Each van has been provided with a 6 channel audio mixer 3 UPTRs and a Public Address Amplifier. Provision is available to meet most of the requirements of production, recording, editing and dubbing etc. The van can also meet the requirements of a live coverage.

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Provision will be kept for installing a VHF/FM transmitter and a video camera along with a monitor inside the van in case these are required for certain types of coverage.

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