UNIT 3 STUDIO ACOUSTICS AND

RECORDING TECHNIOUES

1 Structure
3.0 Introduction .. -
3.1 Objectives
3.2 Studio Acoustics
3.2.1 Nature of Sound
3.2.2 Hearing Characteristics
3.2.3 Acoustic Quality of Enclosed Spaces
3.3 Recording Techniques
3.3.1 Sound Pick-Up
3.3.2 Signal Processing
3.3.3 Recording
3.3.4 Level Control
3.4 Recording Medium
3.5 LetUsSumUp
3.6 Check Your Progress: Possible Answers

3.0 INTRODUCTION
In the previous two units we have discussed the different components of the sound broadcasting
chain in general and microphones, loudspeakers and sound mixers in particular. You are already
conversant with various facilities and technical features of sound mixers and microphones. As
has been mentioned earlier, quality recording demands a good knowledge of studio acoustics
and optimum use of these elements.

This unit has been designed to give you an Insight into various concepts relating to studio
acoustics from the view point of a Programme Producer, and to provide an overview of sound
recurding techniques in general including practical tips for making quality sound recordings.

3.1 OBJECTIVES
After studying this unit, you should be able to:
explain the fundamentals of sound signal and hearing characteristics of the human ear;
identify the different acoustical effects related to enclosed spaces and make use of this
information while recording;
select the suitable microphone for different applications and use it appropriately for
efficient sound pick-up; and
describe the important characteristics of different recording media from a user's point of
view and be able to carry out signal processing effectively.

3.2 STUDIO ACOUSTICS

Acoustics refers to the science of sound. Environment plays a very important role in creating
sound recordings. In enclosed spaces like studios, auditoria etc., the quality of sound produced
depends on the design of studio and the materials used for sound proofing. Good acoustics
conditions for studios mean the following requirements:
the studio should have suitable reverberation and sound quality in accordance with studio
use. Requirements of reverberation are different for different types of audio programmes;
displeasing sounds should not be heard;
there should be no intrusive noise from the surrounding areas; and
sound from the studio should not leak to surrounding areas.

The above requirements are generally taken care of at the time of acoustic design of the
studio. However, as a Producer, you should understand the basics of sound and acoustics for
producing quality recordings.
 Studio Acoustics And
3.2.1 Nature of Sound Recording Techniques

Sound is a pressure wave which is generated naturally by the movement of surface, such as
strings or skins (like in string and percussion instruments), variation of air flow in a tube past
an obstruction (as in wood wind, brass or vocals) etc. Apart from these natural sources of sound,
there is ever growing family of synthesizers or electronic keyboards which generate sound
electrically and have no audible existence until the loud-speaker converts them into sound.

The sound waves propagate in air as very small pressure variations of the static atmospheric
pressure. A sound wave comprises of succession of compression (pressure higher than
atmospheric) and rarefactions (pressure less than atmospheric). One set of compression and
rarefaction comprise one cycle of frequency. There is no actual movement of air particles
from sound source to the listener. Sound propagates only by transfer of pressure variations.
MOVING
PISTON ALTERNATINW
I ,7-C O M P R E S S I O N S
/\ A N D RAREFACTIONS

PRESSURE
ALONE
'-
-
-ATMOSPHERIC
PRESSURE
WAVE

Propagation of plain sound waves

Sound wave propagation works exactly like propagation of ripples in a water pond when
disturbed by throwing a stone. The speed of sound wave propagation is about 332 rnlsec.
Have you ever wondered why thunder is heard later than the lightening flash, though both
originate at the same time? The reason is that light travels with a speed of 300,000 Krnl sec.
Thus lightening is seen immediately even if the clouds are a few kilometers away but sound
takes a few seconds to travel the same distance.

Frequency range of audible sound varies from 20 cycles Isec or Hertz (written shodly
as Hz ) to 20,000 Hz. Most musical sounds have their origin in simple harmonic motion
having a single frequency with multiple overtones. The particular overtones and their
relative intensities are the governing factors of the quality of sound. Faithful reproduction
of overtones brings naturalness to the reproduced sound.

Sound generally propagates in all directions. The intensity of sound reduces in proportion to
square of the distance from the source. When the sound waves strike boundaries of the studio,
a part of its energy is absorbed, a part is transmitted through the boundary and the remaining
part is reflected back. The quantum of reflection depends upon nature of the surface. Reflections
IS are strong from hard surfaces like stonebrick wall, glass surface, doors, tiled floorslwall etc.
whereas they are weak from sqft surfaces like curtains, carpeed floor, soft furniture, fibrous
material faced by perforated hard surfaces etc. The reflected sound may undergo multiple
reflections before becoming inaudible. Control of these reflections to a large extent decides
1 the acoustic quality of the studio.

.-
DIRECT~EARLY~REFLECTION~
MULTIPLE REFLECTIONS
--- ----
Process of reflections from different surface
-.
Recording You have studied in the previous unit that the intensity of sound is measured in terms of
sound pressure variations and is expressed in decibels ( a s ) . Typical sound pressure levels of
different sound sources encountered in day-to-day life are:

Forest 20 dl3
Quiet living room 40 dB
Business office 65 dB
Street traffic 80 dB
Pop Music 100 dl3
Jet take-off (100 m distance) 125 dl3

3.2.2 Hearing Characteristics

'
The human ear is a remarkable sensor. It can perceive sound levels right from rustling of i
I
leaves (level of about 30 dB)to the roar of an aircraft engine (level of about 120 dB). Thus
the human ear has a very high dynamic range i.e. level difference between the loudest and
quietest sounds. Such high dynamic range cannot be recorded on any recordmg medium.
You have to reduce the dynamic range by boosting the level of low level passages and
reducing the level of high level passages.

The ear has a non-linear response to both level and frequency. The ear is most sensitive
at frequencies around 2 KHz.At low sound levels, the ear is much less sensitive to low
frequency sounds than the mid frequency sounds. However at high sound levels, the
sensitivity is more or less the same as at low and mid-frequencies. Loudness pattern of
ear is given in equal loudness contours in figure below:

Equal ToiZdness contours of human ears

(Phon is a unit of Equal loudness contours of human ears- reference 1 Khz.)

intelligibility of sound reduces significantlyin the presence of noise. We have all experienced
the phenoaenon that we have to raise our voices to make ourselves heard on a crowded
railway platform. A difference of at least 20 dB between desirable sounds and noise is
I
essential for intelligibility. This is a very important criteria that decides the dynamic range
of recording medium.

The ear also has some very special characteristics. If a sound is received within about
30 milli-seconds of the direct sound, it is perceived as part of the original sound and adds
to its loudness. However if delay is 50 m seconds or more, it is perceived as a separate sound.
This effect is known as "Hass Effect". During acoustic design, care is taken to avoid
reflections with delay &weeding50 m seconds in the recording and listening area.
 The two ears cleverly decide the direction of sound. We would not have been able to perceive Studio Acoustics And
Recording Techniques
the direction of sound if we did not have two ears. Direction is decided by combination of
intensity and time difference of sound signal arriving at the two ears. The hearing mechanism
is so intelligent that it can precisely decide the direction of sound even when there are strong
reflections from the surroundings. The ears latch on to the first arriving sounds. These aspects
of hearing help in deciding stereo sound pick-up methods.

3.2.3 Acoustic Quality of Enclosed Spaces

We have mentioned earlier that studio or enclosed spaces like auditoria must have suitable
reverberation time depending on the use to which the studio is put to. Let us now discuss the
meaning of reverberation time and the various parameters connected with this.

Reverberation Time
As already mentioned, a sound signal urrlergoes multiple reflections from walls, floor and
It ceiling of any enclosed space b ~ f ~becoming
re inaudible. At every reflection, the level of
reflected sound is less than the incident sound. Initially, the reflections are spaced in time but
after the onset of multiple reflections, the reflections are very closely spaced as shown in
1 figure below.

BANG
Process of reverberation
.
Thus the sound once created does not die immediately. The time taken for a sound signal to
reduce in level by 60 dB is defined as Reverberation Time (RT). RT is the most widely
used acoustic quality parameter of an enclosure.
1
RT of studios varies fiom 0.3 seconds to more than 1 second depending upon use and volume.
Desirable RT is low for spoken-word studios and listening environment. The recommended
value is about 0.35 seconds At higher RT values, clarity of speech suffers. However, for
music stud~os,higher RT values are recommended. Desirable RT is related to volume of the
studio. The h~gherthe volume, the higher is the recommended RT. You can get a broad idea
about the RT of a studio by clapping once and listening to the response. Persistence of clap
can be heard in hlgh RT studios.

The reverberation time is an overall acoustic quality parameter. AlIenclosures with the same
RT do not sound the same. There are many smaller parameters which are alsd quite important
for imparting character to an enclosure. Understanding of these parameters will help you in
deciding appropriate microphone placement. These are briefly described below:

Early Reflections
Early reflections play a very important role in the subjective quality of a studio. It is known
that early decay time is much more closely related to the reverberation perceived by the ear as
compared to the overall reverberation time. This aspect is kept in view while designing more
sophisticated studios.
Recording Echos
Echo is a phenomenon where the reflected sound is heard as a repetition of the direct sound
sometime after the direct sound has ceased. Usually it occurs when the reflected sound arrives
after a delay of more than 50 m. seconds and magnitude of reflected sound is high. As the
time difference of 50 m. seconds corresponds to a distance of about 17 meters, it implies that
an echo can occur if the distance of the reflecting surface is more than 8.5 meters. Thus echos
can occur in large studios but the smaller studios are free from this defect.

Echo phenomenon has an undesirable effect on sounds in the room just like it has on speech
intelligibility too. Care is taken at the time of acoustic design to avoid echos in the usable
area.

Sound Focusing and Dead Points'

Often in large rooms with barrel or dome shaped ceiling such as in a museum, church etc.,
although we can understand what people are saying to us in one particular place, we cannot
completely comprehend when we move slightly away from that place. This is due to the fact
that when the sounds hits the concave reflecting surface, the reflected sounds converge near
the focus of the concave surface. This is called the 'sound focus'. In an area away from the
sound focus, sounds become extremely small. These areas are the so called 'dead point'.
Recording at such locations is best avoided. However, if it becomes essential, you should
choose the recording location suitably.

Check Your Progress: 1

Note: 1) Use the space below for your answers.

2) Compare your answers with those given at the end of this unit.

1) Is there any sound propagation in a vacuum?

...........................................................................................................

............................................................................................................
2) Why do we hear echos in a hilly area?
........................................ ...............................................................
'.mp.

3) What should be the programme &toring level if the listener has to listen the programme
in an average living room?
......,......................................................................................................
. ~ ,.

............................................................................................................
4) What factors decide Reverberation Time of a studio?
............................................................................................................

5) Why do we require lower RT for speech than for music?

 Studio Acoustics And
3.3 RECORDING TECHNIQUES Recording Techniques

To a lay person, sound recording is a simple process of using a recorder to record music or
spoken-word. You might have recorded some conversation, music etc. at home for fun, using
a tape recorder. However, for a professional, making a good recording involves choice and
use of studio, microphones and processing the sound before recording it. Sound recording is
a technique involving sound pick-up, signal processing, recording including control of levels.

3.3.1 Sound Pick-Up

Sound pick-up is the most important factor in achieving quality recording because any
deterioration in quality at this point cannot be rectified at later stages. Hence, great care needs
to be taken in assessing the quality of environment, choice of microphones and placement/use
of microphones.

Choice of Microphone
We have already discussed the important characteristics of different type of microphones
in the previous unit. While dynamic microphones are less expensive and do not need any
powering sources, the condenser microphones have the advantages of better sensitivity,
extremely flat frequency response over entire audio band, a more stable polar pattern with
frequency variation and possibility of remote control of polar pattern. The choice of either
type depends upon the requirements of the programme under production.

A point of caution is in order as far as the frequency characteristic of the microphone is

concerned. A wide and flat frequency response is an important reqllirement in a microphone,
but care must be taken to see that bass response is not too far beyond that required for a
particular sound source. Many condenser microphones have a response down to almost zero '

frequency. Sometimes this can be very troublesome because a very slight air movement can
cause undesired rumble. A microphone with roll-off at about 40Hz should suffice for most
requirements. An extended high frequency response is far more important. The response can
also be equalised in the mixer channel.

The polar pattern indicates the response of a microphone to sounds coming from different
directions. Directional microphones have maximum output for sounds coming from the front
of the microphones with varying level for sounds coming from the sides (90 degree) or back
(180 degree). Direction of minimum output also varies for different types of polar patterns.
Due to reduced pick up of surrounding noises by directional microphones, such a microphone
can be placed at greater distance from the sound source.

The selection of polar pattern is influenced by environment. For example, an ornni-directional

microphone could be useful for interviewing a group of people in an open space in the absence
of much enviroilmental noise but a cardiod microphone would be needed if there is a noise
source at 180 degrees to the desired sound source. Similarly, gun microphones are useful in
picking up remote sounds like effects in sports events1 questions in a press conference. You can
get the best from any given set of circumstances by exploiting these polar patterns. The more
familiar you are with different responses, the better your chances of making good recording.

Phasing of microphones should also be kept in mind. It should be remembered that the rear
lobe in case of bi-directional and hyper-cardioid microphones is in phase opposition to the
front lobe.

Dynamic Range of Recording

The range of levels of sound that can be recorded is limited. At the upper end it is limited by
the capability of the recording medium to faithfully reproduce hlgh levels and at the lower
end by noise. Noise assumes greater significance for low sound levels and for non-continuous
sounds like spoken-word items. Difference in level between the loudest and the softest
passage of sound is termed as Dynamic Range of the sound signal. Noise contribution to
a sound recording falls into two categories - Acoustic and Electrical.

Acoustic noise mainly results from atmospheric noises like traffic, air movement near
microphones, noise transmission from ventilation equipment through ducts and heavy
machinery (like power generation equipment) etc. In professional sound studios, atmospheric
and machinery noises are generally well taken care of. Maincontribution to overall noise
comes from ventilation and AIC system. Acoustio noise in well constructed studios commonly
lies between 25-30 dB.
41
Recording Electrical noise is inherent in any electronic equipment. Microphone channels of the sound
mixer also contribute noise to the sound signal.

The acoustic noise remains the dominant source of noise. It is needless to add that, for
programmes with higher peak sound levels like music, a higher dynamic range is possible
from the studio.

Placement of Microphone
This subject has wide scope. You will learn the best ways of using the microphone with
experience. A few useful tips are offered.

i. Whenever two or more microphones are used close to each other, it should be
ensured that their outputs are in phase. The following simple test may be adopted:

Place two microphones together. Ask someone to speak continuously into the
microphones. Fade in both the microphones one by one and monitor quality of
output. Now fade them in together. If the microphones are in phase, an increase iq
level should result. However, if they are out one of phase, there would be loss of
output level and deterioration in quality (cancellation of bass tones). The result can
be confirmed by fading out one of the microphones when quality and level should
restore. The phasing can be corrected by phase reversal switch generally available
in mixer channel or by reversing connections of one of the microphones.

11. Microphone should not be placed very close to reflecting surfaces such as, bare
walls, hard table top etc. because a hard reflecting surface gives rise to strong
reflections which disturb the acoustic quality near the microphone. In situations
where such microphone placement becomes unavoidable like recording a VIP in
his office, a simple solution of spreading a soft cloth like a shawl or cardigan under
the microphone helps in eliminating the problem to a large extent.
... '
111. Directional microphones should not be placed too close to a sound source to
avoid boosting of low frequencies due to a phenomenon called 'Proximity effect'.
A working distance of 30-45 crns is considered safe in this respect. Some of the
microphones (like two-way Cardioid AKG type D-222) are however, free from this
effect.

iv. For spoken-word recordings, the microphone should not be placed directly in line
with the mouth. It will result in 'P' blasting, a term commonly used to describe the
resulting noise when words containing letter 'P' are spoken. A position slightly to
the side of the mouth should eliminate this problem.

v. The talker should not hold the script between hisiher face and the microphone to
avoid shadowing effect. Where necessary, the script may be held in the dead zone
of the microphone, i.e., the zone where it cannot pick up sounds to avoid paper noise.

vi. The working distance of a microphone is also guided by the ambience. Close
microphone placement reduces the effect of reverberation and vice-versa. In a
lively environment, another way of reducing the effect of reverberation is to place
the microphone with its dead axis facing the live area.

vii. While recording various instruments, the direction of its maximum output and
wind blasts, if any, should be ascertained for choosing the right location of the
microphone. For a few instruments, the following guidelines should be helpful:

In stringed instruments like sitar, violins etc., on-axis of microphone is

kept normal to the front face of the instrument.
For instruments with large output, like tabla, h m s and other bass instruments,
the microphone should be placed facing the diaphragm
and well away from them.
For wood-wind instruments like flute, the microphone should be placed such
that the instrument does not face the wind blast directly.

viii. If a source of sound is placed close to dead axis of a microphone, it s o i d s as if it

is placed at a considerable distance from it. This effect could be us'ed to advantage ,
in drama productions.
 In outdoor performances, the stage floors are subject to intense low frequency Studio Acoustics And
ix. Recording Techniques
vibrations. In such situations, whenever microphones are mounted on floor stands,
some sort of mechanical resilience should be placed under the base of the stand to
reduce bumps and rumble. Slung microphones should be preferred to provide
complete isolation from floor vibrations. However support of the sling should be
examined carefully for any vibrations.

3.3.2 Signal Processing

The microphone output is routed through the audio mixing console before recording. All
modern audio consoles have, in each input channel, standard facilities like sensitivity control,
phase reversal, phantom powering for condenser microphones, controllable equalisation,
channel assignment selection directly or through Panorama Potentiometer.(Pan-pot), reverb
send selection and one or more additional auxiliary channel output selections.

The output channels are equipped with a high level input for reverb return signal, auxiliary
output selection and a limiter/compressor for level control. All channels are also generally
equipped with a peak level indication. The mixers are provided with elaborate monitoring,
metering and talkback facilities. It is also possible to insert external devices like time delay,
noise reduction equipment, compressors, reverb units etc. in any of the input or output
channels. Because of such flexibility, you can mould the sound signal as you desire.

Ideally, the audio mixer is considered to be a creative tool. For efficient operation of the
mixer, you need musical insight, keen ears and an analytical mind. You must master every
hnction of the console; handling of controls must become instinctive and second nature as
you concentrate on sound quality. You can do this job best after acquiring good understanding
of what goes on behmd the buttons and knobs.

The signal processing job in the mixer is a complicated one and there can hardly be a set
procedure for achieving optimum results. It is experience that helps most.

Application of Equalisers
All recording audio mixers are equipped with at least three band equalisers - namely bass,

Band width restriction for reducing scratch noise of old recordings.

Shelving Low Frequencies and boosting High Frequencies to improve older recordings.
Removal of hum and other unwanted signals.

Some of the Creative applications are:

Fullness may be added by boosting Bass Equaliser in 100-300 Hz range.
A broad peak in 800 Hz to 2 KHz projects the sound.
Articulation transients are heightened by a broad peak in 1-4'KHz region.
Crispness in percussion instruments can be heightened by shelving boost above 102 KHz.

A few points of caution may, however, be noted for use of equalisers:

Peaking filters should be preferred over shelving filters wherever possible. This is
because the peaking filter boosts or cuts only the desirable frequencies while the shelving

Boosting should be used sparingly on metal against metal transients as encountered in

musical instruments like cymbals.
Equalisation should never be used as substitute for better microphone placement.

Artificial Reverberation
Artificial Reverberation is often used to compensate for acoustically 'dead' studios. Till the
development of digital techniques, only mechanical reverb devices like spring and plate
reverberators were popular. However, digital reverberation units are now quite common.

43
Recording The digital reverb devices can even be operated remotely whch is a big advantage from the
operational point of view. Some people, however, still prefer the plate type reverberator for
its distinct quality. In general, digital reverberation devices are being preferred to their
mechanical counter-parts.

Ratio of the un-reverberated and the reverberated signal can be controlled in the mixer.

Application of LimiterlCompressor
As we have already discussed, control of dynamic range of the incoming signal is essential
to contain the signal within available dynamic range of the recording system. This can be
achieved by pre-deciding the level of the lowest level signal of the composition and by
controlling high level signals with the help of compressor/lirniter usually available in the
master module of the mixer. The compressor/ limiter changes the input to output level
characteristics of the mixer.

Other Ancillary Equipment

Other signal processing equipment normally used in recording include noise reduction
equipment and time delay units.

Check Your Progress: 2

/
t
Note: - 1) Use the space below for your answers.
2 ) Compare your answers with those given at the end of thls unit.

1) Why is phasing of microphones important?

............................................................................................................
............................................................................................................
............................................................................................................
2) List the important factors for deciding working distance of the'mi~ro~hones
to balance
different sources for a recording.
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............................................................................................................
..............................................................................................................
3) Why are peaking filters preferable over shelving filters?
.............................................................................................................
............................................................................................................
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4) Why do we need a compressor/ limiter in the recording chain?
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3.3.3 Recording
Two basic techniques of recording have evolved over the years. These are: Direct Recording
and Multi-Track Recording.

Direct recording
In this system, all the sound sources are premixed into oneltwo channels for monoistereo
recording. Balancing of various sources has to be well rehearsed because balance once
recorded cannot be altered later. Direct recording is quite useful as long as the complement
of participants is small. However, when the number of microphones being used becomes
large (say 6 or more), direct recording becomes a difficult task. Because of lack of flexibility,
balancing of various channels becomes tedious and time consuming thus exerting a strain both *
on the recording team and performers. If too many retakes are necessary from the performers,
this may adversely affect their performance. For larger or complex recordings, the direct
recording system has given way to multi-track recording system.

The direct recording system is in use by many broadcasting and recording studios all over
the world.
44
IVLUI~I-~rack Recording
In this system, each microphone or a group of microphones may feed an indiviaual track on
the magnetic tape or other multi-track recording medium. Upto 16 tracks are quite common
in such recorders and machines with many more t~ackshave already become available.

The storage of original recording material on a multi-track recorder is an intermediate step.

These tracks are combined to obtain a mono or stereo master recording in the mix-down
process.

In the multi-track recording philosophy, the final recording may bear very little resemblance
to the original pre-recorded tracks. Sometimes the Recordist may exercise greater artistic
responsibility during mix-down than the Music Producer during the recordiilg. Involvement
of the Music Producer in the mix-down proczss tends to produce better results.

The multi-track recording gives following advantages:

a) Flexibility of building a recording piece by piece. The tracks can be recorded at different
timings or even at difft-rent locations depending upon availability of the artistes. Such
piece-meal ::cording sessions are called 'overdub sessions' where headphones carrying a
t e m p c a y n i x of what has been recorded earlier supplies the performers with necessary
timing cues.

b) Better control to create the desired balance in the end product.

c) More economical as it is possible to record a musician separately when his particular

track has not come off well without bothering to re-record the other tracks.

However, it has the following limitations:

a) There is h~ghernoise due to mixing of many narrow tracks.

b) It requires close coordination between the musicians, sound recordist and the programme
producer. To create good separation between instruments, the musicians are separated
from each other physically and acoustically. Extremely dead acoustics are employed and
the musiciatls have to play to a cue track over headphones. All these factors sometimes
are a hindrance to the creation of a good recording.

Despite the limitations, the multi-tracks recording is in vogue for popular music recordings.
However, of late there is a tendency to prefer direct recording with fewer microphones by
many people as it is supposed to give more spontaneous recordings.

Recording Process
The process of multi- track recording may be broadly divided into two activities: Separation
Recording, and Mix-down and Master Recording.

Separation Recording
This sphere of activity comprises microphone selection, mi&ophone placement and recording
of individual channels on a multi-track recorder. As long as the number of microphones does
not exceed the number of input channels of the recorder, each microphone can be allotted an
independent track. On many occasions, however, a large number of microphones may have
to be catered for. In such cases, a certain amount of mixing before separation recording is
unavoidable. A few microphones are judiciously mixed and assigned to different sub-groups.
Each sub-group output is routed to a specific track of the nlulti-Rack recorder.

The flexibility and success of multi-track recording stands or fails solely on the ability to
provide tracks separated enough from each other to allow reasonably independent handling
of each track in the subsequent mix-down. Just how much separation is required in multi-track
recording? Usually 15-20 dB separation is aimed at. Because there is compatible sound on the
adjacent tracks, the above separation allows sufficient freedom to establish relative
dominance between performers.
For simultaneous recording, the separation has to be created at the source ~tselr.uncc r l l ~
sound picked up by microphone suffers in separation, this can not be rectified. ~ l t h o u g h
a separation of 15-20 dB may appear to be quite an easy requirement and is in fact SO for
the recorder, it is not easily achievable at microphones points. Separation between the
sound picked up by various microphones will depend upon;
Closeness of each microphone to its source
Directivity of microphones
Distance between the sources
Relative output level of various sources

A good separation can, therefore, be achieved by using an adequate number of directional

microphones each placed close to its s o w e and well separated fiom its neighboring
microphones. Besides, location of various performers/groups have also to be chosen
optimally. Separation can be further improved by using baffles and barriers between the
sources, studios with comparatively dead acoustics, electronic gating devices with adjustable
threshold for rejecting low level signal and contact transducers. Finally, extreme forms of
separation are made possible by the over-dubbing technique.

Various microphone signais are given a minimum of processing as they are being recorded.
Only those things are done to the signal which improve the signal/noise ratio or which
compensate for deficiencies in the source, save of course, for a certain amount of pre-mixing,
if necessary. It is best to use absolute minimum of limiting and equalisation in the original
recording because it retains the greatest for mix-down manipulation to get the desired results.

Mix-down and Master Recording

The maximum signal processing occurs during mix-down. Flexibility of the mixing console
comes into play during this process. The Sound Recordist and the Music Producer, however,
must have the mix-down in mind from the beginning. The way the instruments are arranged,
the way the microphones are placed, all must be carried out according to a logical plan.
Although the mix-down offers flexibility of repeated 'cut-and-try; it is necessary to have
a well-set mental image of the finished product right fiom the beginning.

3.3.4 Level Control

There are three possible locations for level control in a normal sound mixer. These are:
the sensitivity control in the microphone channel$, channels fader and the master fader.

The sensitivity control is used to equalise sensitivity of different types of microphone used.
This is done by manipulating the sensitivity control keeping the master and channel fader
at 0 dB position to equalise the loudness level of different microphones.

The channel fader is used to adjust the level of the individual microphone and the master
fader for adjustment of the overall level of the recording. The multi-track mixers will have
additional level control points in the form of sub-group faders which are used for controlling
sub-group level. Most mixers are provided with peak level indicators in each channel. At
times there is overload before the channel fader which will not disappear even by recording
gain at the channel fader. In such situations, sensitivity control has to be reduced to ensure
that the signal does not get distorted before reaching the channel fader.

The magnetic tape has the tendency to saturate beyond a particular recorded level. This
results in onset of distortion. The level control, therefore, assumes great significance.
For sudden changes of programme level, it is not practical to adjust the level manually.
Compressors are, therefore, quite often introduced in the 'signal chain feeding channel of
the recorder.

There are basically two types of programme level meters in audio mixers. These are VU
meters or PPM meters. W ~ m e t e r o rVolume Unit Meter indicates broadly the loudness
level of the signal being recorded and does not respond adequately to fast programme peaks.
The PPM or Peak Programme Meter on the other hand responds virtually to the peaks of
the programme and thus does not give an indication of the average or loudness level.

. .
. - . -~- ~ .- - - .- ,
 mc normal level. In the PPM mixers one has to rely on the
level.
The digital recording media is prone to severe distortion if the recorded level exceeds a given
threshold. Hence detection of peaks becomes all the more important for such applications.
Many organisatioils are slowly switching over to PPM for level monitoring due to the above
reason.

- Iheck Your Progress: 3

Note: 1) Use the space below for your answers.
2) Compare your answers with those given at the end of this unit.

1) List three important advantages of multi-track recording.

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2) List the way of in~plcvingseparation between sources for multi-track recording.
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3) List the primary functions of the sensitivity control and channel fader.
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4) List the essential difference between W Meter and PPM.
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3.4 RECORDING MEDIUM

As a producer, it is not necessary for you to know the Recording Medium in great technical
detail, it is desirable to know its broad characteristics. The recording media currently popular
are analog tape, digital tape, digital opiical discs and the computer hard disc.

As already mentioned, the magnetic tape starts saturating beyond a certain recorded level.
This means that the recorded level is not able to increase in the same proportion as the
increase in incoming signal. This results in the onset of distortion. The distortion is gradual
in the beginning and is severe if the signal level increases further.

On the other hand, distortion of digital signals is independent of the recording medium.

I
The digital audio signal is recorded in the form Lr'binary numbers. There is a largest number
(usually 16) allocated in the system beyond which the audio signal cannot be increased.
This level is usually denoted as 0 dB. Increase in audio level beyond this level causes
severest form of distortion. Hence every care is taken to avoid signal level reaching the
above limit. This is usually done by employing compressors with'sharp gain reduction
beyond a level usually 6-8 dB below maximum level mentioned above.

Best recorded dynamic range can be achieved by optimum utilisation of available dynamic
range of the recording medium. Available margin above the normal level of recording
corresponding to 0 VU or 0 PPM level on the level meter and signal to noise of the recording
medium should be known to the Progranlme Producer for optimum control of operating level.
Knowledge of the medium, whether it is analog or digital, also helps in taking appropriate
level coiltrol decisions.

1 '
47
 ost widely used recording medium in India, a tew
precautions are listed below for care and use of magnetic tape:
ensure that the tape is properly threaded in the machine and the coated surface is facing
the recording head;
make sure that the heads of the machine are clean;
stack tape spools vertically;
do not place recorded tape near strong magnetic fields like Loudspeakers, Transformers,
CD Players etc;
safeguard against excessive variation of temperature and humidity;
discard faulty spool; and
if a tape has been stored for a long time, it is advisable to spool it once before use.
-. -

3.5 LET US SUM UP -

In this unit, you have learnt the basic concepts of sound, hearing characteristics of the human
ear and studio acoustics. The importance of reverberation time (RT) in an enclosed space, i.e.,
studio and the advantage of a higher RT for music recordings were explained while discussing
the recording techniques the need for assessment of the quality of environment, choice and
use of appropriate microphone and how signal processing equipment help the recordist
were analysed. This will help you in analysing the reasons for any particular sound effects
produced in the recording environment. It will facilitate taking appropriate action by way
of optimum microphone placement to avoid undesirable effects to the extent possible. The.
advantages of multi-track recording over single track or direct recording and the recording
process were also explained.

Producing quality recording is an art. You must master the controls of the mixing console,
intimately know the quality of your microphones and understand the characteristics and
limitations of the recording medium. Knowledge of different types of musical instruments
also helps in deciding optimum pick up from the musical instruments. Rehearsing before
a recording is a must for optimum dynamic range control and balancillg. You can get the
best out of given circumstances with some of the practical hints given in this unit. You can
improve your recording skill with further reading and experience.
----

3.6 CHECK YOUR PROGRESS: POSSIBLE ANSWERS

Check Your Progress: 1

1) There is no sound propagation in vacuum as it requires presence of a medium like air,

water, metal etc.

2) nearby hills reaches the ear after a gap of more than 50 m.

Sound after reflection f i : ~ m
seconds and any sound arriving with such delay is perceived by the ear as a separate
sound. That is why'we hear echo in hilly areas.

3) If one has to listen to a programme in an average living room, a monitoring level of about
70-80 dB sound level should suffice as the background noise in an average living room
will not exceed 50 dB.

4) The nature of walls, floor and ceiling are the factors which decide the reverberation time
of a studio. The harder the surfaces, the more is the RT.

5) We require lesser RT for speech than music because higher RT means that sounds takes
longer time to die. Thus previous words are still audible when new words are spoken.
This causes interference which affects intelligibility of speech. On the other hand,
musical sounds improve with higher persistence of sound. We are all familiar with
bathroom singing effect.

Check Your Progress: 2

1) Phasing of microphone is important because if microphone outputs are not in phase,

output of one microphone will cancel part of the signal picked up by the out-of-phase
microphone while the microphones are mixed in the audio mixer. The result will be a
very dull recording.
2) Working distance of different microphones should be decided keeping in view the Studio Acoustics And
Recording Techniques
following parameters:
a) Level of different sources feeding a single microphone;
b) Requirement of ambience pick-up;
c) Proximity effect of the microphone;
d) Artistic requirements like distant or near sounds etc.

3) Peaking filters are preferable because the peaking filter boosts or cuts only the desirable
frequencies while the shelving filter boosts or cut all frequencies in the bass or treble
range.

4) We need compressor/limiter in a recording chain because recording media have limited

dynamic range. Sudden programme peaks do not allow manual level coptrol for
controlling the dynamic range of the picked up signal. A Compressor/limiter does this
job effectively.

Check Your Progress: 3

1) Tlwee important advantages of multi-track recording are:

a) better flexibility;
b) better control;
c) more economical for large recordings.

2) The following are some ways of improving separation between sources for multi-track
recording:
a) by employing large number of directional microphones close to their sources;
b) by separating different sources from each other by use of sound isolating screens;
C) by use of contact microphones; and
d) by overdub sessions.

3) The sensitivity control is used to equalise loudness of different microphones used in

different channels while the channel fader is used mainly for balancing different sources.

4) The,VU Meter indicates average level of the programme. It does not respond adequately
to fast programme peaks. The PPM on the other hand responds virtually to the peaks of
the programme and thus does not give an indication of the average or loudness level.