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CE Unit 2

The document provides an overview of audio systems, including their components, working principles, and applications. It covers digital consoles, FM tuners, and public address systems, detailing their functionalities, specifications, and the technology involved. Key concepts include the conversion between analog and digital signals, sound card features, and the importance of sound quality measures such as fidelity and intelligibility.

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0% found this document useful (0 votes)
10 views14 pages

CE Unit 2

The document provides an overview of audio systems, including their components, working principles, and applications. It covers digital consoles, FM tuners, and public address systems, detailing their functionalities, specifications, and the technology involved. Key concepts include the conversion between analog and digital signals, sound card features, and the importance of sound quality measures such as fidelity and intelligibility.

Uploaded by

oviya vetri
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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UNIT-2

Audio systems:Introduction to audio system, Digital Console – Block diagram, working


principle, applications, FM tuner – concepts of digital tuning, ICs used in FM tuner TD7021T,
PA address system, planning, speaker impedance matching, characteristics, power amplifier
specification.

2.1 INTRODUCTION TO AUDIO SYSTEM:


✔ Audio system refers to devices that reproduce, record, or process sound.
✔ Microphones, radio receivers,
✔ AV receivers, CD players,
✔ Tape recorders, amplifiers, loudspeakers.
✔ Audio system refers to devices that reproduce, record, or process sound.
✔ Microphones, radio receivers,
✔ AV receivers, CD players,
✔ Tape recorders, amplifiers, loudspeakers.
✔ Basic Sound System
✔ A basic sound system consists of an input device (microphone), a control device (mixer),
an amplification device (power amplifier), and an output device (loudspeaker).
✔ The primary goal of the sound system in sound applications is to deliver clear, intelligible
speech, and, usually, high quality musical sound.
✔ The basic principle of all electronic audio systems is to take sound waves convert them into
an electric current or voltage and manipulate them as desired, then convert back into sound
waves.
✔ A microphone is an example of a transducer, a device that changes information from one
form to another. Microphone changes information into patterns of electric current or
voltage.
✔ Computers are digital systems so they can only produce & manipulate audio in digital
format.
✔ The problem however is that in the real world audio is an analog entity.
✔ Speakers are expecting an analog signal so they can reproduce sounds, we can’t feed
speakers with a digital signal so D to A converter is used.
✔ Today all motherboards have an embedded sound card like on board audio.
✔ Some motherboards don’t come with the audio sections on the motherboard, but add on
cards that comes with the product being installed on a PCI slots.
✔ There is a chip called codec (coder/ decoder) on the motherboard, which is in charge of
converting digital audio signals into analog and vice versa.
✔ Sound Card
✔ A sound card (also known as an audio card) is an internal computer expansion card that
facilitates input and output of audio signals to and from a computer under control of
computer programs.
✔ Typical uses of sound cards include providing the audio component for multimedia
application such as music composition, editing video or audio, presentation, education and
entertainment (games).
✔ Digital Out: Used with surround sound or loud speakers.
✔ Line in: Connection for external audio sources, e.g. tape recorder, record player, or CD
player.
✔ Microphone or Mic: The connection for a microphone or headphones.
✔ Sound out or line: The primary sound connection for your speakers or headphones.
✔ MIDI or Joystick: (15 pin yellow connector) –used with earlier sound cards to connect
MIDI keyboard
✔ A typical sound card has:
✔ Digital Signal Processor
✔ Digital to Analog Converter
✔ Analog-to-Digital converter
✔ Read-only Memory (ROM) or Flash memory for storing data
✔ Musical Instrument Digital Interface (MIDI) for connecting to external music equipment
✔ Jacks for connecting speakers and microphones, as well as line in and line out.
✔ Sound Card Upgrades
✔ Sound-card upgrades are an option if the motherboard does not have a sound chipset built
in or if the user wants higher performance.
✔ A common upgrade path is to move from an ISA sound card to a PCI sound card.
✔ For some audio applications, such as telephony or certain games, full-duplex sound is a
must.
✔ Full-duplex sound has the ability to accept a sound input while simultaneously providing
sound output.
✔ USB Sound Cards
✔ USB sound "cards", sometimes called "audio interfaces", are usually external boxes that
plug into the computer via USB.
✔ The USB specification defines a standard interface, the USB audio device class, allowing
a single driver to work with the various USB sound devices and interfaces on the market.
✔ Drivers
✔ A driver provides a software interface to hardware devices, enabling operating systems and
other computer programs to access hardware functions without needing to know precise
details of the hardware being used.
✔ A driver acts like a translator between the device and programs that use the device.
✔ Each device has its own set of specialized commands that only its driver knows.
✔ Sound Quality
✔ The three primary measures of sound quality are fidelity, intelligibility, and loudness.
✔ The fidelity of sound is primarily determined by the overall frequency response of the
sound arriving at the listener’s ear.
✔ The intelligibility of sound is determined by the overall signal-to-noise ratio and the direct-
to reverberant sound ratio at the listener’s ear.
✔ The loudness of sound is determined, in turn, by the intensity of the sound waves.
✔ Wireless Speaker
✔ Wireless speakers also work on the same principle as the wired speakers do. These speakers
also need to translate an electric signal to an acoustic one, but the only difference is that
they transmit these signals to the satellite speakers.
✔ Technologies used in Wireless Speakers:
✔ FM (Frequency Modulation) transmission in some speakers. This involves Radio
transmission of signals by the Transmitting system to all the receiving satellite
✔ Infrared Transmission is also used in some of the Home Theatre systems. Mostly used for
the speakers that create surround sound effect.
✔ Bluetooth Transmission is most common and advanced nowadays. It works on latest
Bluetooth technology.
✔ Most of the Wireless Speakers work at 900 MHz frequency range.
✔ Range of Wireless speakers is 150 to 300 feet from the Base Transmitter.
✔ All the Satellite speakers are generally Battery operated. They need to be charged
separately.
✔ Wireless speakers need extra care for their proper functioning. Generally manufacturers
claim that their speakers are weatherproof but extra care is needed for using in bad weather
conditions.
✔ Wireless Microphone
✔ Wireless microphones require a wireless transmitter, and a wireless receiver.
✔ The wireless transmitter is either built into the microphone itself (as in handheld wireless
microphones), or is connected by a short cable to a body pack transmitter (as in hands free
wireless microphones).
✔ All wireless transmitters require a battery
✔ The main advantage of wireless microphones as compared to wired microphones is
freedom of movement.
✔ With wireless microphones a vocalist or presenter has much greater flexibility to move
around a stage or among an audience.
✔ The biggest disadvantage is price, as wireless microphones are, in general, more expensive.
2.2 DIGITAL CONSOLE:
✔ A basic audio analogue mixer takes in multiple input streams and merges them into one.
✔ Simply connecting all the input signals in parallel and feeding them into a single input
won't work because they may interfere with one another.
✔ Separation of the signals is essential, allowing for independent regulation of each signal's
volume.
✔ In addition to basic mixing, a mixing console's many features include phantom power for
condenser/capacitor microphones, pan-control (which allows each signal to be placed in
any position in the stereo field), filtering, equalisation, routing, and monitoring facilities,
which allows any number of sound sources to be routed to the desired loudspeaker for
listening without affecting the mixer's main output.
Working Principle
✓ Professional audio engineers use a digital console to modify the characteristics of incoming
audio signals.
✓ Digital signal processing allows for the flexible manipulation of acoustic properties such
as mix, route, signal dynamics, and equalisation to meet a wide range of applications.
Multiple audio channels, each with its own digital processing, are combined into a single
"master" channel. Powerful signals in terms of control, processing power, and the desired
audio quality can be generated with the help of a professional digital mixing console.
✓ The audio mixing system's block diagram and power amplifier are depicted in Fig . The
✓ FM console employs a TDA 7021T integrated circuit.
✓ It is a low-power FM radio receiver IC that operates on just 3 volts and needs almost no
additional components.
✓ However, its low sound output necessitates an external amplifier which is IC LM386.
✓ The IC is protected against short circuits and operates over broad common-mode and
differential voltage ranges.
Applications of Digital Consoles
Live Sound
1. Concerts and Festivals: Digital consoles are used to mix and control audio for live
concerts and festivals, providing high-quality sound and flexibility.
2. Theater and Musical Productions: Digital consoles are used to control lighting and
sound systems, creating an immersive experience for audiences.
Broadcasting
1. Radio and Television: Digital consoles are used in broadcasting to control audio levels,
mix audio sources, and ensure high-quality sound for live broadcasts.
2. Podcasting: Digital consoles can be used to record and mix podcasts, providing high-
quality audio and flexibility.
Post-Production
1. Film and Television: Digital consoles are used in post-production to mix and edit audio
for film and television productions, ensuring high-quality sound and precise control.
2. Music Production: Digital consoles can be used to record, mix, and master music
productions, providing high-quality sound and flexibility.
ICs used in FM tuner TD7021

✓ Here is a convenient quality sensitive FM recipient circuit or FM tuner circuit. The circuit
works around the TDA 7021T IC which is an FM radio receiver IC.
✓ The IC requires just barely any external parts and the necessary voltage is just 3 volts. The
sound yield is low so we have further amplified it with LM386 amplifier IC.
✓ You can likewise utilize other high-power sound speakers with it to expand the sound more.
A high-frequency tune antenna speaker circuit utilizes the antenna input segment of the
circuit for expanding affectability.
✓ You can change the recurrence with the assistance of a 25pF variable capacitor. The Coil
L1 is an air-cored coil equivalent to 5 turns of 1mm silver plated wire turned on a 4mm
composition.
✓ This FM radio circuit is entirely versatile and easily fits in a little box with batteries.
Applications and Uses
✓ FM receivers are generally used in fin FM radio telecom.
✓ It is likewise used in seismic prospecting, telemetry, radar, and monitoring newborns for
seizures through EEG, music synthesis, two-way radio systems, some video-transmission
frameworks, and magnetic tape recording systems.
2.3 FM TUNER:
✔ FM tuners operate in the very high-frequency regions and consequently have operating
requirements different from those of a.m. tuners.
✔ The r.f. circuits and components must be of optimum design to provide sufficient
sensitivity to pick-up f.m. stations under all conditions, and the audio aspects all circuitry
must be of sufficient quality of design and construction to receive and pass audio
frequencies from 15 to 20 kHz as may be broadcast by f.m. stations.
✔ High fidelity AM reception is usually limited by the presence of high-level man-made noise
and atomospheric disturbances which AM receivers cannot reject without loss of fidelity.
✔ Also, because of the propagation characteristics of frequencies used for standard a.m.
broadcasting, out-of-area broadcast station signals can interfere with local reception
(especially at night).
✔ At the frequencies used for FM reception, distant stations cannot interfere, and the inherent
noise-rejection characteristics of the FM receiver minimize the noise problem.
✔ Receivers for frequency modulation are of the superheterodyne type and are somewhat
similar to ordinary amplitude modulation superheterodynes. Block diagram of the most
widely used FM receiver is shown in Fig.
✔ The most important difference between AM and FM receivers is in the detector circuit. A
number of FM detectors have been developed. The ratio detector removes the audio signal
from the carrier and at the same time rejects amplitude impulses which may accompany it.
✔ The Foster-Seeley discriminator is sensitive to amplitude impulses and in order to eliminate
them before detection, limiter stages must be provided. The limiter removes all amplitude
fluctuations from the carrier before if is applied to the detector.
✔ Practically all current f.m. tuners are transistorised. Many FM tuners are combined with
AM (AM/FM) tuners. Superheterodyne configuration is universal; however, considerable
variation of detail is employed.
✔ For example, low priced FM tuners used a single transistor as a combined oscillator and
mixer (converter), whereas deluxe type tuners provide separate oscillator and mixer
transistors.
✔ Again, economy-type designs generally utilise ganged tuning capacitors, while the more
elaborate designs provide permeability-tuned arrangements. Solid-state designs are based
on the transistor-for-tube principle, except for systems employing IC packages.
✔ An f.m. broadcast signal is basically a high fidelity transmission, whether the mono or the
stereo mode is used. The high fidelity capability of an FM. system is due to its
comparatively wide channels.
✔ Each FM broadcast channel has a bandwidth of 150 kHz, or + 75 kHz. Although a high
fidelity FM signal can be transmitted in a narrower channel, wide channels provide a better
signal to noise ratio.

Fig 2.1 Block diagram of FM Superhetrodyne Receiver


✔ It follows that an f.m. receiver should have a bandwidth of 150 kHz. The f.m. broadcast
band extends from 88 to 108 kHz, and f.m. station occupies (including its guard band) a
range of 200 kHz.
✔ Thus, a maximum of 100 FM broadcast stations can be accomodated in a given service
area.
✔ Co-channel interference from distant areas is much less troublesome in the case of f.m.,
than for f.m. This comparative immunity of an FM receiver to co-channel interference is
due to he “capture effect”, which means that the stronger signal tends to suppress the
weaker signal.
✔ If one signal is twice as strong as the other signal, the suppression is practically complete.
Even if one signals is only 50 per cent stronger than the other, the suppression is very
effective, provided that the receiver is properly adjusted.
✔ All f.m. receivers use an intermediate frequency of 10.7 MHz. At the 50 per cent of
maximum response points (6 dB down) on the response curve, the bandwidth is typically
200 kHz.
✔ The comparatively high value of intermediate frequency reduce the circuit complexity that
is required. If a lower intermediate frequency, such as 455 kHz, were used, elaborate
circuitry could be required to obtain a flat-topped response over a bandwidth of 150 kHz.
✔ When the bandwidth of an amplifier is increased, its gain is decreased. Therefore, the i.f.
stage gain is considerably less in an f.m. receiver than in an a.m. receiver.
✔ Therefore, an f.m. receiver employs more i.f. stages. At least one more i.f. stage is provided
in most tuners.
✔ When two more f.m. i.f. stages are included, the last stage operates as a limiter; since the
limiter is essentially an overdriven amplifier, its gain is considerably less than that of the
other i.f. stages.

Fig 2.1 Block diagram of Typical FM Tuner


2.4 PUBLIC ADDRESS SYSTEM:
✓ Because the intensity of a sound diminishes with distance, amplification is required
whenever a large group is to be addressed; this allows those seated further away from the
stage to stillhear the speaker clearly and without discomfort.
✓ A public address system (PA system) serves this purpose. It is used to reach people in far-
flung places (like airports, hospitals, and factories) and at large gatherings (like town hall
meetings and concert halls).
Block diagram of a basic PA system
The microphone in a public address system converts acoustic signals into electrical ones. The
loudspeaker is another transducer that transforms electrical audio signals into sound waves by
amplifying and processing them. A typical PA system's block diagram is depicted in Fig.
Components of a PA system are as under:
1. Microphones: Sounds are picked up and transformed into audio signals, which are merely
variations in electrical current. Amplifiers typically feature multiple
microphone inputs and an auxiliary input for a record player or tape player.
2. Mixer: A microphone's output is routed to a mixer. When multiple channels are being fed into
a single power amplifier, the mixer stage is there to ensure that none of them interferes with one
another. It can be a standalone plug-in or an integral part of the structure.
3. Voltage amplifier: The mixer's output is further amplified by this device.
4. Processing circuits: Tone controls (bass/treble controls) and a master gain control are features
of these circuits.

5. Driver amplifier: Because of the voltage amplification it provides, the internal resistance of the
subsequent stage is lowered when the signal is fed into it. In this way, it triggers the power
amplifier to produce a greater output.
6. Power amplifier: It boosts the signal's strength. In order to eliminate even harmonics from the
output and prevent the transformer core from becoming saturated, a push- pull circuit topology is
used. A matching transformer connects the power amplifier's output to the loudspeaker so that the
low impedance of the loudspeaker doesn't limit the amount of power that can be transferred to it.
7. Loudspeaker: It takes in electrical audio signals and transforms them into pressure variations,
which we perceive as audible sounds.

2.4.1 Planning a public address system


1. Acoustic Feedback: In order to prevent an extremely loud howling noise caused by acoustic
feedback, the microphone must be kept well away from the loudspeaker.
2. Distribution of Sound Intensity: The high notes are what allow us to understand what are being
heard by us while the low ones provide the volume. Intelligibility decreases at greater distances
because high notes attenuate more quickly than low notes when propagated further. Because of
this, the sound pressure level needs to be split between multiple speakers so that it can reach even
the farthest corners of the room.
3. Reverberation: Because sound waves in a reverberating medium overlap with one another,
intelligibility is diminished. If the direct sound is getting lost in the echoes, the PA system should
boost the volume there. Multiple low-power loudspeakers placed strategically around the
auditorium are preferable to a single, large one because they reduce echo.
4. Orientation of Loudspeakers: Loudspeakers should be positioned so that the maximum output
of the PA system is directed toward the audience and not the walls. In addition, they should be
elevated one metre from the ground, with their axes roughly at the level of the seated audience
members' ears. Reflections from faraway buildings can also cause audible echoes that need to be
managed.
5. Ambient Noise: When there is a lot of background noise like in a busy market, the high
frequencies (treble boost) need to be amplified in order to restore intelligibility. The high levels of
amplification is required in a noisy environment which makes the sound seem artificial because
bass is emphasised over treble. Therefore, in a noisy environment, the PA system should tone
down the bass to preserve the quality of the sound. In addition, microphones with built-in noise
reduction are useful.
6. Dynamic Range: Good public address systems have a level limiter built into the amplifier,
which prevents the volume from rising above a certain threshold.
7. Microphone selection: The best microphones for a public address system are cardioid types,
which pick up neither reflected sound nor sound from the loudspeakers themselves.
8. The direction of arrival of sound: The loudspeakers need to be positioned so that the sound
appears to originate in that area. In general, the human ear can determine the source of a sound
within a few seconds of hearing it. Accordingly, small speakers can be placed in such a way as to
point listeners in the right direction, while larger speakers can be placed further apart to increase
volume. Alternately, in digital systems, the signal can be artificially delayed by 10–20 ms before
it reaches the loudspeakers, causing the loudspeakers to be heard after the sound directly from the
speaker. Whatever the placement of the speaker, the desired volume and directionality will be
achieved.
9. Phase delay: When multiple loudspeakers are playing at once, it is possible to hear both the
sound coming from the closest loudspeaker and the other loudspeakers. When there is a delay of
45 milliseconds or more, the intelligibility of the delayed sound is compromised. The lag time is
roughly 16 metres. Therefore, the maximum distance between speakers should not be more than
16 metres. When maintaining a safe distance, 10 metres is ideal.
10. Grounding: Earthing the chassis and shield of electronic devices and coaxial cables via
plumbing is essential.
11. Choice of Loudspeakers: The selected loudspeaker must be able to cope with the amplifier's
output power. Speaker columns with woofers, squawkers, and tweeters can be used to achieve high
fidelity. Moreover, the efficiency of a public address system is improved by using horn-type
speakers rather than cone-type speakers.
12. Placement of Microphones: Microphones should be positioned to pick up all of the program's
sound without picking up any background noise.
13. Speaker Impedance Matching: Maximum power transfer from the amplifier to the
oudspeaker is achieved by matching the total loudspeaker impedance to the amplifier's output
impedance. Thus, the loudspeaker series-parallel combination should be optimized for maximum
power transfer. For example, contemporary solid-state amplifiers, often known as "bridging"
devices, take a voltage signal from an audio source and produce an amplified version of that signal.
Output voltage and power are both dynamically regulated, and the output impedance is low.
14. Power Amplifier: The audience can hear us clearly from a distance thanks to the PA
system's amplification. The amplifier's output power can range from a few watts for use in a
classroom or at a small event to several hundred watts for use at a large public event or sporting
competition.
2.4.2 Speaker Matching Systems
✔ Speaker matching systems permit higher operating efficiency in audio power systems. The
70.7 volt system is preferred for high-power PA networks.
✔ A matching transformer is installed to match a speaker or a group of speakers to the 70.7
volt line. This is called a constant-voltage system because the line voltage is comparatively
unaffected by switching speakers on or off.
✔ Network calculations are as follows :
✔ 1. Determine the power rating of each speaker.
✔ 2. Add the power rating to find the total power demand and use a 70.7 volt amplifier with
a rated power output at least equal to this demand.
✔ 3. Select a 70.7 volt matching transformer for each speaker, or for each group of speakers
✔ 4. Connect the primary terminals of each transformer across the 70.7 volt line from the
amplifier output. Note that a primary mismatch up to 25 per cent is tolerable.
✔ 5. Connect the secondary terminals of each transformer to its speaker, or group of speakers,
observing the matching ohms taps.
2.4.3 PA System Characteristics
✔ In speech reinforcement systems the listener hears sound from two sources. Part of the
sound comes from the person making the speech; yet another part of the sounds comes
from the PA system.
✔ The total sound that arrives is greater than that from either source alone, but the two
portions do not arrive simultaneously. From a practical viewpoint, the maximum tolerable
time delay between the two arriving wavefronts is approximately 25 milliseconds.
✔ A longer time delay is likely to result in a confusing separation of the two sound sources;
the listener’s attention jumps from one source to the other and becomes distracted.
✔ Since the human ear has a natural tolerance for echo effects, an echo tends to be disregarded
under ordinary conditions.
✔ However, if the echo arrives too long after the original wavefront, it is no longer
disregarded and the listener becomes uncomfortably aware of the split-sound sources.
✔ Therefore, it is good practice to plan a PA system so that no listener will be exposed to two
arriving wavefronts that have a time delay greater than approximately 25 milliseconds.
That is called the Haas effect.
✔ A cluster of speakers, or a speaker column, will provide considerable sound coverage.
However, in large buildings that have appreciable reverberation, a large speaker column
may need to be supplemented by smaller columns placed farther back from the sound
source.
✔ A supplementary speaker column, can be installed with a time-delay circuit so that the
sound from the first column arrives 5 millisecond before the sound from the second column
is radiated.
✔ The intensity of the delayed sound should not be more than 10 dB greater than that of the
first-arriving sound, otherwise separation is likely to become evident.
✔ Time delay units for PA systems generally employ a tape deck with spaced playback
recording and playback heads, or a magnetic disc.
2.4.4 Public Address Amplifiers
✔ A PA amplifier may be designed as a unit, or it may consist of separate amplifiers.
✔ When more than one audio input source is utilised (two-microphones, for example)
operating controls are provided to change from one source to the other, or to mix the
outputs of the sources, The mixer and fader section is designed to provide flexibility of
operation without sudden increases or decreases in volume.
✔ A fader serves to gradually reduce the volume of one source while gradually increasing the
volume of another source. A mixer may or may not provide fader action.
✔ A microphone amplifier provides preliminary step-up of the microphone output signal. The
power amplifier is driven by an intermediate amplifier which is basically similar to a
microphone amplifier except that it may employ high power transistors.
✔ If a power amplifier has sufficient gain it may be driven directly by the microphone
amplifier. The complementary symmetry public address audio amplifier. For stereo
operation, the amplifier circuit is doubled, using the single power supply for both channels.
✔ Since this is a basic power amplifier, a suitable preamplifier is required. The balanced
bridge power supply helps eliminate ripple, resulting in noise-free reproduction. A dual
voltage output is obtained across dropping resistor R15.
✔ This amplifier can be used in a number of applications, provided there is suitable drive
from a preamplifier. Typical applications include high fidelity.
✔ It can also be used in an electronic organ, as a playback amplifier for a tape deck, and the
25 watts of audio power is sufficient for use as a musical instrument amplifier.

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