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2. radio

How to Install a
Broadcast
Studio
A short training manual
for low-power FM stations
Written / Compiled by
David A. Casement
Broadcast Technician
Galcom International
September 2009
Version 1..3
Table of Contents
Legal/Disclaimer
3
INTRODUCTION
4
SECTION 1: POWER SOURCE
A. Introduction
B. Testing the Outlet
C. Plugs and Connectors
D. Surge Protection
E. Lightning Protection
5
5
5
6
7
8
SECTION 2: ASSESSING THE NEED
A. Channels Needed
B. Machine Inputs
C. Budget Issues
a) Selection of Equipment
b) Installation of Equipment
10
10
10
10
12
20
SECTION 3: WIRING THE MIXER
A. Cables and Connectors
B. Balanced or UnbIalanced
C. Inputs
D. Outputs
E. Testing the Signal through the System
F. Metering
22
22
24
25
26
27
28
SECTION 4: STUDIO OPERATION
30
SECTION 5: MAINTENANCE
A. Tape Recorder
B. Reel to Reel Tape Recorder
C. Compact Disc Player
34
34
38
38
SECTIONS 6: STUDIO ACOUSTICS
39
Written by Ethan Winer, Used by permission
CONCLUSION
53
BIBLIOGRAPHY
54
2
Notice
This manual is intended as a short “How To” guide to installing a low
power Fm studio and associate equipment. For those who have
limited knowledge and experience in doing so read the manual
carefully before attempting to installing any studio equipment or put a
radio station on the air. Incorrect procedures can cause great damage
which is very expensive to repair. Correct installation can provide
years of reliable service.
Legal
All copied material in this manual are used by permission. Copy write
material is the property of its original owner. Galcom International,
Africa by Radio, nor David Casement make any claim against such
material. This manual is provided for educational purposes by Galcom
International and Africa by Radio both of which are not-for-profit
entities. Any remuneration received will only be for the cost of media
used in distribution. All Logos and trade marks are property of their
respective companies. All trade marks and logos must remain in tact
when printing, copying, or forwarding this document.
Disclaimer
Safety must always come first when working with electricity and radio
equipment! Every effort has been made to provide accurate
information at the time of writing. As this manual is used outside of our
control and is provided for educational purpose, Galcom International
nor Africa by Radio take any responsibility for damage to equipment or
personal injury which may result from the use of this manual.
3
INTRODUCTION
In a radio station there is a long chain of equipment which the audio signals must pass through in order to be prepared for
transmission to the listener. This paper will focus on installing and wiring an audio mixer for broadcasting and maintenance on the
major pieces of studio equipment. The principles and information given should also be a good reference for installing most pieces of
studio equipment which are not covered in this manual.
TALLY
MIC 2
4
Main
MIC 1
COMPUTER
Cue
CASS
CD
Control room
CD
The diagram above shows the typical connection arrangement in an FM station. Most professional equipment will use XLR and 1/4 inch jacks
for the main outputs as well.
Section 1: Power Source
A. Introduction
One of the main things to establish before powering up a studio is that the electrical power source is at
the proper voltage and is stable. Improper voltage, surges (spikes) and low voltage (brown outs) can
cause serious damage to broadcast equipment. Also, improper wiring of the outlet can cause damage
to equipment. Before plugging in any electronic equipment, use a reliable voltmeter to check that the
voltage at the electrical outlet is correct. Also verify that the outlet is properly grounded. This is
procedure must be done with care and is described in section B.
Be sure that the voltmeter being used is in good condition and operating properly. Check the test
leads and probes for breaks or bare wires. Make any repairs or replacements needed before using the
meter.
B. Testing the Outlet
Each grounded or earthed electrical outlet has three terminals. Live, Neutral and Earth or Ground.
To verify that the outlet is properly grounded, place one lead in the live terminal and one in the neutral
and read the voltage. Now place the meter leads between the live terminal and the earth terminal.
This reading should be the same as the one taken from live to neutral. It is acceptable if the above
reading is within ten percent of its rating. If the outlet is 220 volts then the reading should be between
200 and 240 volts and should not be any lower or higher. This should not fluctuate very much while
taking the reading but be stable. If the reading from live to earth is zero or very low (only a few volts)
then the outlet is not grounded and should not be used until the ground terminal has been solidly
connected to earth with at least a number 14 or 2mm diameter earth wire.
The next reading, with the ground/earth terminal grounded, should be from earth to neutral. The
reading should be zero or just a few millivolts. The earth and neutral should both be connected to
earth/ground but the neutral should also go back to the transformer which feeds power to the building.
The earth wire should travel the shortest path to ground and be connected to a ground/earth rod.
The connection to the transformer is done by the electricity supply company. If the outlet and electrical
system in the building are wired properly the reading between neutral and earth should read zero or in
the millivolt range. If the reading is high, such as several volts or higher, the outlet is not wired
correctly and the equipment connected to this outlet will not function properly and can even be
damaged. This condition can cause surge protectors to be ineffective or even be permanently
damaged! In extreme cases some protection devices can even EXPLODE!
Surge protection power bars have devices from live to earth and neutral to earth and depend on the
outlet being wired properly so the voltages on earth and neutral are in proper range. If either the earth
or neutral are “floating” too high then the limits of the protective devices can be greatly exceeded. If
any wiring beyond the outlet itself is required, call a qualified electrician to repair the wiring.
If the voltage in your location fluctuates or has frequent outages, a voltage stabilizer or an
Uninterruptible Power Supply (UPS) will be needed for a studio and/or low power transmitter such as
30 to 200 watts. This unit should have sufficient output capacity to power essential equipment for a
broadcast studio. Consult an electrician or UPS professional.
Make sure that the outlets used for your station equipment are wired properly and are supplying the
proper voltages BEFORE plugging in any equipment! Many times basic items such as lights and
portable fans will appear to operate on power sources which are less than ideal. This does NOT mean
that the power source is correct. The above test must be done!
5
Type G
Mainly Great Britain
Rated at 15 amps.
Type M
South Africa, Swaziland, Lesotho
Rated at 15 amps.
Neutral
Earth
Live
North American Type B
Rated at 15 amps, 125 volts
C. Plugs and connectors
Electrical equipment often comes from the factory with molded ends on the power cords which plug
into the electrical outlets. Some manufacturers provide two or three cords with different ends or plugs
for various types of outlets which are used in different parts of the world. If the power cord(s) which
are provided with your equipment are not the correct ones for the outlets in your country, the correct
one must be installed properly.
Modifying electrical plugs to fit into outlets or removing earth pins is an unsafe practice and must not
be done. The earth pin is there for safety reasons. Removing it can cause severe electrical shock to
personnel and can also cause problems with the functioning of the equipment. The practice of not
using a plug on the end of an electrical cord and just pushing the bare ends of the wires into an outlet
is also very unsafe. This practice can lead to electrical shock and the connection will be unreliable at
best. Wires by themselves in the outlets will not make proper contact with the inside of each terminal
which can lead to arcing in the outlet and burnt wires. It is imperative that the proper plug be used for
the outlet. If a different plug is needed on the power cord provided it is best to purchase a new power
cord at an electrical shop. The second option is to cut off the plug on the power cord and properly wire
on a new one. Before doing this make sure the colors for live, neutral and earths are known on both
the cord and the plug. (Normally in Africa on power cords, brown is live; blue is neutral and green/
yellow is earth.) If there is any uncertainty have an electrician wire the new plug on the cord.
6
D. Surge protection
This is a very important part of a radio station. Surge protectors are just what the name says; they
protect the equipment from voltage surges or “spikes”. It is a good practice to use multiple outlet
power bars which have surge protection built in for studio use. With adequate power ratings these can
be used for low power transmitter sites as well. The power handling rating must be well above the total
power consumption of the transmitter plus any associated studio equipment. That is, the total of all the
equipment plugged into the power bar.
Along with this kind of surge protection, it is also very important that there
be a higher capacity surge protector in the system. There are specific units
which are designed to be wired into the circuit breaker panel. Other units
are plugged into an outlet and are in series with the equipment they protect
but are larger physically and will handle greater power surges than what is
built into power bars. Power bars tend to respond to surges faster than the
larger units. It is best to have both kinds of surge protection in the system.
All of these systems rely on the electrical wiring to be correct and a good
connection to earth! Without a proper connection to earth they will not
provide adequate protection. Again, make sure the wire or cable used to
connect to earth is a larger diameter than the other wires in the system.
Superior Electric makes the PT series of surge suppressors which wire
parallel to the load. It can be wired in the breaker panel or even on a plug
and inserted into an outlet.
Tripp Lite is on manufacturer that makes different
models of Power Bars with surge suppression
built in such as this one.
7
E. Lightning protection
When lightning strikes the earth or an object on the earth, highly-charged thunderstorm clouds pulse
"leaders" downward toward the earth. They are seeking a path to electrical ground. Objects on the
ground, such as buildings, trees, power lines and radio towers, emit different amounts of electrical
activity during this event. Streamers (lines of particles) are launched upward from some of these
objects. Some of the downward-going leaders connect with some of the upward-reaching streamers. It
is at this point that the circuit is completed and current flows. This creates the visible arc we call
lightning.
Many 1000s of amps = 1000s of volts
The lightning charge is seeking ground and has huge electrical potential. A perfect connection to
ground is not possible, especially on objects such as towers which have many metal pieces joined
together. Any material connected to ground will have some DC resistance and some AC impedance.
Even a small amount of resistance in a connection to ground (earth) will create extremely large
voltages when thousands of amps of current, such as from a direct lightning strike, are forced through
them. These immense voltages and currents will cause great damage to radio stations if proper
grounding is not installed. It is essential that proper grounding be installed at a radio station.
Damage to radio stations can happen in different ways;
A direct hit on an antenna or transmission line can cause great physical damage. Where the lightning
strikes, metal and other materials generally melt, fail, and cannot be repaired. A lightning strike close
by but not direct can cause a magnetic field to build up which will create voltages in power lines.
These voltages are not as high as in a direct strike but can still be high enough to do damage.
For information on how to ground the tower/mast, see the manual How to
Install a Broadcast Transmitter.
Proper ground rod installation
In the case of the transmitter being located in the same building as the
studio, common-point grounding is the safest and most effective method!
8
At the Transmitter Building
All conductors providing ground/earth potential and that enter or leave a transmitter/studio building,
including transmission lines shields and conduits, should be bonded (connected) to a ground system
similar to the one at the tower. This system should include two or three ground rods or plates
connected together. These rods should also have a ground cable connecting them to the safety
ground rod of the electrical system. There then should be one large conductor from them coming into
the transmitter building. This ground cable should be brought in as straight and as short a distance as
possible. All ground/earth conductors in the building should then be connected back to this ground
connection. This creates a “common point” ground or “star” ground. This is crucial in the building to
minimize damage from surges and lightning.
Different Grounds Mean Trouble!
Beware! Double check all the grounds on the electrical system even in the service entrance. Make
sure all ground leads are connected back to the service entrance, then to the ground. Any ground
wires that are not connected to the system service entrance and the single point ground may spell
trouble in an electrical storm. Do Not Assume!
Providing a single, low-impedance/resistance path to ground here as well for the energy in lightning
strikes is vital to minimizing damage. Time and money spent on this will pay off in the long run!
This is a typical North American system. The 240 VAC transformer is located on the utility pole. The
neutral lead is grounded there as well as at service entrance. More than one Grounding “Electrode”
should be used for a radio station as outlined in the text. Note that inside the building, the neutral and
ground buses are connected together at the service panel ground connection and not before.
9
Section 2: Assessing the Need
A. How many channels are needed in the audio mixer?
It is important to carefully consider how many microphone channels will be needed in the on-air mixer
(audio console). Thinking too big for the size of station may cost more money than your budget can
afford. But if you do not have enough microphone channels it can severely limit the abilities of the
station and affect the on-air sound if some voices are not close enough to a microphone. Generally a
basic on-air studio should have at least three microphones: one for the main presenter, one for a
co-presenter and an extra channel for guest speaker. The third channel will also serve as a back up if
either of the other two channels should stop working. Be sure to match the characteristic impedance
of the microphones to the input of the mixer.
B. Machine inputs
The same principle regarding the number of mic inputs also applies to machine inputs. It is best to
count up all the sources of audio that will be used and add two or three more channels to that. It is
common for a new station to install an audio console for on-air use and it be filled or nearly filled when
it is installed. This is not a good situation as there is no room for expansion and no spare channels if
one should stop working. When counting the number of channels needed, sources such as telephone
line interferences (hybrids) must be counted as well. Take into account CD players, cassette players,
outputs from computers, mini-disk players, any portable equipment which may be plugged in directly
to the console, and lines from other studios, if so planned.
C. Budget issues
The budget for starting a station is always a major factor in selecting equipment. Next to tower,
transmitter, and studio-to-transmitter link system, one of the most expensive items is the audio mixer
or console which will be used on-air. There are two types of audio mixers which can be considered.
One is the sound reinforcement type which is used in churches and public buildings. The other is the
actual broadcast console type of mixer. The sound reinforcement type mixer has more controls for
changing the sound and routing sound to different outputs. This type of mixer varies greatly in quality
depending on manufacturer and price.
Audio mixers which are specifically built for broadcasting tend to have fewer tone controls and outputs
than the sound reinforcement units but are built and labeled specifically for broadcasting. The
components used are generally of much higher quality and they are built much sturdier and will last
many years. However they also cost a great deal more money than sound reinforcement mixers.
Some brands cost US $6000 for just six channels. With this in mind there are two radically divergent
philosophies regarding mixers. Some insist that the big names in broadcaster mixers; Ward Beck,
McCurdy, Wheatstone etc. can't be beat, and that the small sound reinforcement style mixers allow
announcers to fiddle with too many controls and they just don't look like a proper radio studio mixer.
10
Increasingly, new radio engineers are favouring Mackie-style PA mixers. They are inexpensive
(very much less expensive than a big broadcast board), light, small, and they generate little heat. They
are quiet in two respects: their audio specs are amazing and they don't require cooling fans. For the
price of one broadcast mixer, several of these can be purchased.
There are some very inexpensive brands of mixers available on the market. However Galcom's
experience with some of these mixers has been very poor. One very popular German brand in
particular has good sound but often will only last a short time. It is essential they have good surge
protection ahead of the power supply. They tend to be susceptible to surges damaging the power
supplies and the mixer will then stop operating.
Another good brand of sound reinforcement mixer is Soundcraft. The model M8 has been used with
great success. They cost a bit more than the Mackie or Behringer but are better built than the
Behringer and do have some features which are desirable for a radio station. Both Mackie and
Soundcraft come with built in filtering against radio signals and other electrical noise which may be
introduced on the power or mic lines.
When considering the purchase of a mixer for a radio station it may be possible to purchase a used
one and save some money. Caution must be exercised when purchasing used equipment. Used
sound reinforcement type mixers should only be considered if they are in nearly new condition or if
the mixer is one that is known to someone at the radio station and is known to be in excellent
condition.
If a used broadcast type console is available have it checked to make sure that it is fully functional and
nothing functional is damaged. These units can often be refurbished by a qualified technician and give
several years of good use. If it is not possible to try a used mixer for a time or have a reliable
technician check it out then it is better to buy a new one.
Studio A, Radio Lumiere, Port- Au-Prince, Haiti.
Using a larger Broadcast console.
11
SELECTION OF EQUIPMENT
There are two basic studios used in radio, an on-air studio and a production studio. The On-Air
studio is the one which is used to put music and programs on the air, or to talk live on the air. The
production studio is used to record material that will be played on the air at a later time. It is very
important to know your budget when planning every part of the studio.
Budget
The budget of a station usually is the single most limiting factor in the equipment selected. This
course is limited to the small budget station. It is important to know how much money can be spent
for the on-air studio and plan out the selection of equipment before ordering any of the pieces. When
funding is limited, modern consumer level equipment is often used with excellent results.
At the time of writing, a basic manually operated studio could be installed for just under US $ 1500.
This would include a small to medium size audio mixer/console often used in sound reinforcement
systems such as the Mackie VLZ1202 or VLZ1402 or the Behringer MX1604A or MX 2004A. Any of
these mixers will handle four or more line level sources such as cassette decks or CD players as
well as at least four microphones. This easily forms the basis for a functional on-air studio. The
larger the mixer the more microphones and sources of pre-recorded material it can handle. This
makes the presenter’s job a bit easier as he/she can be cuing material (getting ready to play) while
other material is being played on-air.
Items needed in a manually operated studio;
i) one Audio mixer
ii) two CD players
iii) two Cassette players
iv) Mini disk players can be used if available in your area but digital recorders are taking over
v) two microphones (minimum)
vi) microphone stands - desk top or floor stands depending on need. One for each mic.
vii) stereo headphones
viii) powered speakers for monitoring. (Good quality multimedia speakers that are used for
home computers work)
ix) one power bar with surge protection
x) a radio receiver to monitor on air. This can be a regular radio or a Galcom fix tuned radio.
xi) optional items if funding available;
- one on-air light
- one audio activated relay to turn on the on-air light.
xii) Backup power source
All of these items must be considered. Prices for each must be obtained as well as any shipping,
taxes, and custom fees that will apply. After the prices are obtained, the equipment which fits into
the budget can be ordered or purchased. It is of no use to order one item which consumes 90% of
the total budget and not leave enough money to purchase the other items needed.
Selection of pieces of equipment
1) Audio mixer or console
There are many audio mixer units on the market varying greatly in quality and features. It is best that
a good quality low noise unit be used. Small mixers designed for home entertainment use will work
in an emergency or on a temporary basis but are not suited for long term every day use. For
broadcasting it is very important that each piece of equipment used causes the lowest noise level
possible. The noise generated from each piece used, adds together during broadcast. For best
noise reduction the mixer should be in a metal case or a large portion of the case should be metal. If
12
the studio is in a location where there is a lot of electrical noise (motors, air conditioners, etc.), built in
filtering is needed. Some models do include this. If the studio and transmitter are located in the same
building and are close to each other, Radio Frequency Filtering is needed. Normally the studio and
transmitter should be located as far apart as possible when they are located in the same building in
order to avoid the radio signal getting back into the audio equipment and causing feedback and hum.
Mixers such as the Mackie VLZ1202 have radio frequency filtering built in from the factory and are in
an all-metal case.
A crucial part of the mixer to be used in broadcasting or high quality recording is the microphone
inputs. The circuitry used must be very sensitive in order to pick up weak signals from microphones,
but also must not generate much noise. The best way to insure this is to use a very low impedance
input to the circuit. The professional version of this circuit uses balanced inputs. The microphones
which work best with this arrangement most often use the three-pin XLR connectors. Professional or
good quality sound reinforcement mixers can handle a wide range of signal levels for different
microphones. The most common way of doing this is for each microphone channel to have a trim
control at the input. This control can be set according to the strength of the signal coming into that
channel.
One of the more obvious considerations in selecting a mixer is the required number of inputs; the
number of microphones, CD, Cassette, etc. The auxiliary returns on the mixer will easily add another
stereo or mono line-level source to your main mix. Here are some examples of sources you might like
to route through the aux returns;
- tape deck
- CD player
- audio tracks from a VCR if needed
- audio from a satellite receiver
2) Cassette deck
Audio Cassettes are now an old technology and have been used for many years for recording and
playing music. They are very economical to purchase and use. The sound quality from good quality
cassette decks is reasonable for use in radio stations. One of the biggest faults with some cassette
tapes and machines is the high amount of hiss that is generated. Cassettes can also be difficult to cue
to an exact point on the tape, ie the start of a song etc. If cassettes are to be used in the radio station,
only high quality metal or chrome tapes and high quality home entertainment or professional grade
machines should be used. A very important feature for a cassette machine is “Automatic Music
Search” or AMS. This can help with cueing a selection on the tape.
At the time or writing, some dealers of audio equipment were no longer carrying consumer cassette
decks. Newer digital players such as minidisk and MP3 players have started to take over this market/
function.
3) CD player
The CD (Compact Disc) is the currently the most popular medium for music. The discs are much
smaller and easier to handle than the old vinyl records of years ago. They also store much more
music or information. Because the recording on these discs is digital, the quality is higher. They also
tend to generate much less noise than vinyl or cassettes. Another advantage of their use in
broadcasting is that they can easily be cued to the start of a song just by looking at the display on the
front of the playback machine. Each song has it’s own number and the machine displays this number.
The very large volume of CD discs and CD players on the market today has brought the price down to
the point of being very affordable. Having two CD players in the budget mentioned above is quite
reasonable.
A very helpful feature to look for in a CD player is a count-down timer. Nearly all players give the
length of a song on the display at the start of a song. As the song is played they display how long the
song has played. In broadcasting it is much more important to know the time remaining in a song
13
which is playing. This will let the on-air presenter know how much time he/she has before the next
song must be played or they have to open the microphone and talk.
4) Minidisc
This medium has become more popular since it was introduced. The portable player/recorders are
very small and convenient to use and to carry. They can be plugged into an amplifier or audio mixer to
playback or the disks can be removed and inserted into a larger minidisc deck which can be used for
editing and playback in a larger system. Minidisc players can cue to the exact start of a file or to an
exact spot in a file. This is very useful for on air use. It can take the place of the old tape-based “cart
machines”. Another great feature is that many of the portable recorder players now have the MP3
format which allows a large amount of material to be stored on one disc.
5) Digital Audio Tape or “DAT”
This format caused some excitement when it was first introduced but has not gained great popularity
in broadcasting. In general the record and play machines are expensive to buy. Also different
manufacturers have different standards for putting the information on the tapes and there are
problems with one kind being compatible with another manufacturer. This causes dropouts or silent
spots during playback. For the money spent, there are other systems that will deliver better quality.
6) Digital Recorders
At the time of writing, digital handheld recorders have become very popular. They can record on a
number of media including flash memory cards. These memory cards are very small and can hold up
to 4 hours of stereo recording per card depending on the capacity. These units have built in
microphones or professional microphones can be plugged into them. The sound is converted to digital
and can be copied into a personal computer to be edited and played later. The cost of these units is
about the same as a professional portable cassette recorder.
7) Microphones
A microphone converts acoustic energy (sound) into electrical energy. Microphone selection can play
a large role in our attempts to reduce the potential for feedback from monitor speakers, if used, and
background noise. If it were able to do so with absolutely equal sensitivity to all frequencies, we would
say it had a “flat” or “linear” frequency response. But a peak would indicate it is more sensitive to
some frequencies than to others. This peak may cause feedback to occur (where monitor speakers
are used) before the required system gain (volume) can be achieved or pick up unwanted noise.
14
When monitor speakers are used, low frequency feedback problems can often be reduced by using a mic with a less
extended low-frequency response. The Low Cut filter on each channel of many mixers will give you this control where
you need.
Microphone Patterns
Omni-directional microphones pick up sounds from all directions equally. These are especially
helpful when recording music groups, choirs, etc in a small studio where there may not be enough
microphones for every
member of the group. An
Omni or non-directional
microphone is not a
good choice for an on air
presenter. It will pick up
undesired sounds of
shuffling papers,
background talking, and
the click of pushing
buttons on a tape player
etc. This can be irritating
to listeners.
15
Uni-directional microphones, sometimes called “cardioid”, pick up sound from mostly one direction.
These microphones are a better choice for on-air presenters. They are very sensitive to sound from
directly in front and out to approximately 45 degrees then the sensitivity decreases until they have a
“null” or very low sensitivity point at the back of the microphone. This helps them to greatly reduce
picking up background noise. These microphones work best when close to a small sound source
(a human mouth, for example). Because there is no boost of signals arriving from a distance around,
pickup of background noise is greatly reduced. Feedback potential is also greatly reduced. This is
very important for on air presenters while live on air or recording material.
Bi-directional microphones also
exist. They are not as popular as
Omni- and Uni-directional
microphones. They pick up sounds
from directly in front and back of the
microphone. They can be used in
small studios for interviews with two
people or two sound sources to be
recorded or aired.
16
More on Microphones
Generally the better the quality of microphone used the better the sound will be on-air. Cheap cone
shaped microphones used in sound reinforcement systems will work but will not give good clear
sound. Sony's SM 58 is extremely popular for sound reinforcement and very durable. It has become
very popular for radio stations on small budgets as well. Although it is popular, affordable, and gives
reasonable sound, it is not the best sounding microphone for a radio station. One to try is the Audio
OM2. It is said to have far superior fidelity and barrel noise reduction, and it's less expensive.
Professional microphones with low impedance will use XLR jacks (three leads). Most microphones are
called dynamic; these do not require a power source. Some microphones are called condenser and
require a power source fed through two of the leads from mixers that offer "phantom power". In
general condenser microphones are great for recording subtle or distant sounds, neither of which are
desired in a radio. The other thing about radio studios is that we are more interested in eliminating
noise and sounding good than in high sound fidelity. You sound good when far off sounds from fan
vibrations and table tapping are not picked up (not audible). This is accomplished by using a
low-sensitivity mic; and using shock mounts. This will also help reduce popping and sibilance. Proper
microphone placement helps with this as well. Large diaphragm mics tend to produce the full audio
spectrum.
Now on to some other popular favourites. The Electro-Voice RE20 microphone is widely popular in
larger budget stations. It is a large diaphragm dynamic microphone with built in noise canceling. The
RE20 retailed for over $800 in Canada, it is discontinued now and has been replaced by the RE30
and RE40. You can get one slightly used on eBay for US$325-$400. A better deal is its lesser-known
cousin the PL20; its guts and specs are identical but it was marketed to the US musician market rather
than radio and it has a slightly different finish: US$275-325 on eBay. In the same quality league is the
Shure SM7 (replaced by the SM7a and later the SM7b with superior hum-bucking in the presence of
fluorescent lights and computer screens). It's about the same price as the PL20. Electro-Voice makes
another microphone for stations with bigger budgets: the RE27 has three frequency cut-off switches,
greater sensitivity than the RE20, and superior hum bucking and pop filtering. One theory is that many
good mics designed to be kick-drum mics would also serve well in capturing voices for radio; the
relatively cheap Audio Technica ATM25 looks like a good candidate. At the time of writing the AT2030
was also being investigated. Canadian prices: about $200.
RE 20 by E lectrovoice
PL 20 by Electrovoice
17
8) Computer - automation & live-assist.
If the budget allows, a station can include computers for recording and playing music or any material
that will be broadcast. A Personal Computer or “PC” now has the capability to record and play music
and spoken word in CD quality. With good software and a good quality sound card a PC can be a
very big asset to production and on-air play. Used as a record and play back unit it has a vast
storage capacity. The PC can also be used for automated play for part of the broadcast day or
overnight programming. This is the medium many radio stations are now using. There are many
computer programs (software) available to record and edit voice and music. Software is also
available to play music or announcements manually or totally automated. One combination is “Zara
Radio” with a recording program such as Cool Edit or Gold Wave. Zara will play any song,
announcement or ID manually or will play a long list automatically. Zara will even switch to the
line-in on a computer sound card in order to rebroadcast things like a signal from a satellite receiver.
The nice thing about Zara is there is a free download version. The other programs mentioned will
record and edit music or other material which is to be broadcast. There are many such programs
available.
When selecting a computer system for a radio station, first select the software (computer program)
which will do what is needed. Then look at the system requirements listed for that software and
select a computer/s which will exceed the requirements. It is important to have a computer with
enough speed, memory and hard drive space for the software to work properly. After this, compare
the total cost to the budget available. It is wise to ask assistance of someone who has worked with
computers and music programs in helping to select and install a system. It is very important to have
as much Hard Disk Drive Space as possible. This is needed for storing a music library and programs
for a couple of days of automated play-out.
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9) Capacity of studio required
Studio Room or Building
In many cases, the studio may have to “fit” into an existing building. Depending on the size and
shape of the rooms available, they may serve for many years. In some cases existing rooms will
serve to get the station started. It is best to have at least three rooms if possible, one room for the
on-air studio, one for the production studio, and one for reception and office. Each of these rooms
should be set up to be not too hot or too cold. During summer months or in tropical climates it may
be necessary to install air-conditioning. Care must be taken to not place the air conditioning unit so
that the noise will be picked up by any microphones. This will usually mean installing window style
units “behind” where the microphones are facing during broadcast and as far away from the
microphones as possible. Another very important item is to plug in this type of air-conditioner to it’s
own electrical outlet. Do not share this outlet with studio equipment. An air conditioner requires a lot
of electricity and creates a lot of electrical noise, especially when it in the process of turning on. This
can cause electrical noise and power “dips” which can interfere with the broadcasts. If central
air-conditioning is installed or can be installed in another location in the building and condition the
studios, that is the best solution.
The studio rooms also need to be mostly sound proof. This is for two main purposes: 1) to keep
outside noise from being picked up by the on air or production microphones and interfering with the
broadcasts, and 2) to deaden any echo effect there may be in the room acoustics. It can be irritating
to listen to a presenter’s voice which has a constant echo or boomy quality to the sound. Sound
proofing can be accomplished by installing acoustic tiles or fiber glass insulation if the budget
permits. For smaller budgets, egg cartons, carpet, or burlap bags can be hung on the studio walls in
order to limit the echo sound and reduce outside noise. (See Section 6, page 39)
In the case of “making due” or “fitting” the studios into an existing room you have to take what you
get. If there is a larger and smaller room, the smaller would normally be the on air studio and the
larger would be the production studio. Keep in mind that the production studio needs to be divided
into two parts. The studio are where the people who are speaking or singing into the microphones
and the control room where the operator is working the controls to record the sound. In the case of
building or renovating a building, a very efficient layout can be realized. A three meter by three meter
room can serve as the on air studio but one that is four meters square would be more comfortable.
This room needs to be able to contain the studio now and for some time into the future as the station
grows. There needs to be room for a “desk” and chair for the presenter as well as the audio mixer/
console, CD players, tape players, microphones, etc. Also, there needs to be space for organizing
any CDs or other media which may be played on air. Also if a computer is used or will be used in the
near future, specific space for it must be allowed.
The production studio needs to be a different layout than the on air studio. The production studio
normally has a studio area and a control room. The studio area needs to be very sound proof in
order to keep outside sounds out. It should also not have a boomy type of sound or echo in the
recordings. The size of this room can vary tremendously depending on budget and the intended
uses of it. A smaller room of four meters by four meters can work well for recording preachers and
teachers, soloists, or interviews of one or two people. For recording choirs or singing groups and
musical instruments, a larger room will be needed. Four meters square may not be large enough. A
six meter square room may be needed. Anything larger than this will need to be well sound insulated
to remove the echo and not give a sound effect of being “lost” in the room when only one person is
being recorded. For both the on air studio and the control room in the production studio, different
layouts will work but the “L” or “U” shaped operator positions have been popular over the years.
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Photo courtesy of
Radio Iliria, Albania .
INSTALLATION OF EQUIPMENT
In order for a radio station to function properly, all the equipment has to be placed and wired
correctly. Each piece of equipment needs to be installed so it is being used as intended. Good
practice in laying out and wiring an on-air or production studio are very important. If proper care is
taken in wiring a studio, many potential problems can be eliminated.
Normally the studio and transmitter should be located as far apart as possible when they are located
in the same building in order to avoid the radio signal getting back into the audio equipment and
causing feedback and hum.
Placement of equipment is very important. The operator/presenter’s position must be comfortable
and easy to use. Each machine that will be used to play music, etc. on air must be easy to see and
use while sitting in front of the microphone.
1) Placement of furniture and equipment
When deciding on the layout (floor plan) of a studio it is important to consider the chiropractic
comfort of the operator/presenter. This means that if a computer is used in the studio the screen
(CRT) must be positioned straight ahead from where it will be used. The keyboard must be the
correct height for comfort and reaching the keys without bending wrists too much. A good chair with
proper back support is needed. Sitting for some hours without back support can cause fatigue and
soreness.
A properly laid-out studio will also have each of the machines for playing music and programs within
easy reach of the operating position. It is much less stressful and tiring to operate a studio where
these items are within easy arm’s reach. This is also conducive to making fewer errors. Microphone
stands and racks for “stacking” the equipment are very helpful in this. The basic operating position
can be laid out in a few different ways. The shape of the letter L or U are very popular to achieve
this. See the photos below of studios in existing radio stations.
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2) Lighting
Proper lighting in a studio will enable the operator to see the controls of the equipment and his
“script” papers without causing glare or shadows. Sometimes two light sources from each side of the
operating position work well. Florescent lights are popular because of the amount of light they give
and filling a room with light without having one spot much brighter than the rest. This is good but
caution must be exercised. The “Balasts for florescent lights can create a great amount of electrical
noise. In some cases this can be picked up by studio equipment and get into the recordings and
even go on-air. This is very undesirable. Make sure that in the electrical wiring of the building and
studio room all ground connections are made and in good condition. Check all connections on the
shielded cables of the audio equipment. All signal cables should be shielded and connected to
signal ground. If the hum and noise is bad it may be necessary to install the balasts for the
florescent lights outside the studio room and bring the wires from them into the fixture in the studio.
The tubes MUST still be mounted in their fixtures. This may relieve some of the hum noise problem.
Echo in a room used for the on-air or production studio can cause serious problems for listeners.
The echo of a presenters’ own voice can make it difficult to understand what he or she is saying.
The “boomy” quality of a room with an echo can make listeners strain to hear what is said. This can
be tiring and is sometimes called “Listener Fatigue”. One solution to this is to install sound proof
materials on the walls, floors, and ceiling of the studio. In some cases entire walls should be covered
and in other cases only portions of the walls need to be covered with these materials in the form of
baffles. If the money is available to use professional sound insulation, that is the best. If funds are
limited then other materials can be used. Curtains can be hung over cement walls in order to absorb
some sound and prevent it from hitting the cement and bouncing around the room. Carpet normally
used as flooring can also be used in this fashion. It can be installed on cement floors and walls and
will give the desired effect. If funds are really limited egg trays, foam and even burlap sacs can help
deaden or take out some of the echo.
3) Feedback
Inverse Square Law - One of the greatest tools is the rule which describes the behavioral
characteristic of sound. Understanding the Inverse Square Law can help you make intelligent
microphone placement decisions which can lead to reduced risk of feedback and dramatically
improve recordings.
The Inverse Square Law refers to the way sound levels decrease as you move away from their point
of origin. We discuss sound as ‘sound pressure level’ or SPL. The unit used to compare the levels
between sounds is called a decibel, ‘dB’.
0dB SPL is the theoretical ‘threshold of hearing’ - the softest sound level that can be heard by
sensitive ears. 0dB does not indicate the absence of sound, just that there is zero difference from
the reference level. The intensity of the sound varies inversely according to the square of the
distance it moves away. Simply stated, measured sound level will drop 6dB for each doubling of the
distance from the source. And when we hear a sound that we perceive to be twice the level of
another, that difference is about 10dB. Room acoustics play a part.
Whenever monitor speakers are used in a studio, on-air or production, they must be placed behind
the microphones used. (It is preferable to use headphones for the on-air studio or mute the speakers
when a microphone is open). They must not be close enough to cause feed back. If a speaker, for
example, stays 10" away from a microphone which is picking up the sounds from that speaker (i.e.
monitors), the result is feedback. This happens because the sound from the monitor speaker is
picked up (heard) by the microphone that is used and is then fed back through the system again.
This causes the system to howl or create other tones at a high volume.
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Section 3: Wiring the Mixer
Before starting to wire a studio together it is essential to have a basic understanding of the signal flow
or the path that the sound takes. Connecting microphones and other devices to the proper inputs on
the audio mixer/console is crucial to good quality sound and proper operation of the studio. Likewise,
the mixer needs the proper connections to the recording equipment or broadcast transmitter for
optimum sound quality. Connections made between pieces of equipment must be of high quality.
Loose wires, ill fitting connectors, and unshielded cables can cause hum, hiss, crackling, and even
loss of sound. These traits are NOT acceptable for radio
broadcasting.
A. Cables and Connectors
Without them you would just have a pile of electronic
equipment doing nothing! A connector can also be
called a ‘plug’. The receptacle that a plug plugs into is
called a ‘jack’ or ‘socket’. Good ones cost more, but by
not compromising on cables and connectors, you will
eliminate a critical variable when it comes to
troubleshooting problems.
1) 1/4 inch Phone Connector - There are two types of
1/4" phone plugs; TS and TRS.(Mini plugs and jacks are
the same configuration as 1/4 inch just smaller in size.)
Two-conductor phone plug - has a tip and a sleeve,
thereby also called a TS plug. The tip carries the actual
signal and the sleeve acts as a ground, or return. This is
an unbalanced connection and is mainly used for
short-distance runs between high-impedance equipment. The major disadvantage of an unbalanced
line is that it’s more susceptible to picking up noise.
Three-Conductor phone plug - has a tip, a ring, and a sleeve, thereby
also called a TRS plug. There are two common ways to use this type of
connector. First, as a balanced connection which uses two of the
conductors (tip and ring) carrying the same signal, while the sleeve acts
as a shield. This provides far better noise rejection than an unbalanced
connector. Second, as an unbalanced connection where the tip and the
sleeve are used to make connection, the tip being live and the sleeve
being ground/return.
2) RCA/Phono Connector -is the standard for consumer stereo components such as CD players,
cassette decks and more. Their advantage is that they’re
extremely inexpensive and can be grouped together in
small areas. The disadvantages of RCA/Phono plugs are
that they’re an unbalanced connection, and tend to be very
fragile. You will usually find they are needed for Tape In
and Tape Out sockets because of their common use in
consumer grade CD and cassette players.
3)XLR Connectors - (sometimes called Cannon Connectors) are barrel-shaped plugs most often associated with
microphones. Most XLR connectors are balanced and have three wires connected to three separate
pins. These connectors are almost always used for low-impedance, balanced connections such as
microphones. The advantage is that they are extremely sturdy and reliable.
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However, they are more expensive than other types. It will be wired so that Pin 1 is ground, Pin 2 is
‘hot’ (positive), and Pin 3 is ‘cold’ (negative). Occasionally you will encounter audio equipment that
requires a different configuration of hot/cold/ground. Make sure to consult the owner’s manual if the
standard XLR configuration doesn’t work.
Standard wiring of an XLR type connector.
Note both male and female connectors in the back of this amplifier. When installing
XLR connectors make sure of which gender is needed for the equipment involved.
4) Adapters
Audio adapters are available for just about every possible combination of connectors. When one piece
of equipment needs to be connected to another but the connectors are different, an adapter allows
this connection to be made. For example, a 1/4 inch TRS male to a XLR female.
RCA to TS1/4 inch is used in both professional and consumer grade equipment. Mini-phone to 1/4
inch TRS phone mini plugs are generally used on consumer grade
equipment or on mini disc players and discmans.
5) Audio cable
i. Coaxial - This type of cable looks similar for both unbalanced
microphone and radio signal. However, their electrical characteristics
are very different. They both consist of a center conductor (wire)
surrounded by insulating material, then an outer conductor which is of
screen-like construction. This is all covered by a protective jacket. The
size of conductors and the insulative material between them determines
the characteristics of these cables making them suitable for either audio or
small signal radio frequencies.
ii. Two-conductor shielded - For balanced microphones and many other
connections between audio equipment, two-conductor shielded cable is
used. There are different sizes and qualities. These are not coaxial type
cables. Better quality cables have a foil surrounding the two inner wires,
then a strip of bare wire laying on the foil to form 100% shield. This again is
contained inside a protective jacket. The two inner wires are insulated from
each other and the shield and insulation on them is
color-coded.
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6) Soldering
Extreme caution should be used when soldering audio cables and connectors in every part of the
radio station. A high quality, low wattage (25-40 watt) soldering iron should be used. An iron that is too
hot will melt insulation on wires causing short circuits. It can also melt insulation inside connectors
causing them to be unusable. A good quality, high density sponge, slightly moist, should be used for
cleaning the tip of the iron. Never use acid-core or automotive type solder in audio or radio station
equipment. The best solder to use is 60/30/10 tin/lead/silver resin core. If this is not available, 60/40 is
second choice.
Note: Be sure BOTH ends of the cable are disconnected from equipment before preparing for
soldering. The act of stripping insulation to get ready to solder can produce short circuits which could
result in damage to the equipment. Soldering a cable that is connected at the other end can cause
heat damage or a 'ground loop' through the soldering iron, resulting in damage to the equipment.
B. Balanced or Unbalanced
There are two basic types of connections for audio signals to audio and radio equipment (other than
optical); balanced and unbalanced. Unbalanced is generally intended to be used for shorter cables, 6
meters or less, where it is not in close proximity to strong interfering signals such as high power radio,
wireless communication, or high power electrical motors. Balanced connections are intended for
situations where a much greater rejection of interference is crucial. They also allow much longer lines
to be run between pieces of equipment without degrading signals.
Unbalanced cables and connectors only use two
conductors. One is the so-called 'hot' (live) wire. The
other is called 'return' (ground). Typically, the cable
running between the pieces of equipment is a
co-axial style. The center conductor is the hot lead,
which is surrounded by insulating material, which is
then surrounded by a second conductor which is the
return. The return conductor is most often formed
into a screen-like fashion in order to shield outside
electrical signals away from the hot lead.
Balanced cables and connectors use three conductors. Two of these are hot and the third is return. The
two hot wires carry the desired signal between
pieces of equipment. The third conductor does not
carry any of the desired signal but only acts as a
shield to keep out external interfering signals. Any
interfering signal which does get past the shield will
then be present on both of the hot wires at the same
time. Because these wires are connected to a
transformer which cancels out anything that is not
desired, interfering signals are not heard.
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C. Inputs
Caution:
The Mixer or Console
Most connections on a mixer are straight forward, mic-in is where the microphone connects. Line-in is
where a CD player or cassette machine will plug in. Their signal level is much higher than a
microphone and should only be plugged where designated. Plugging a CD or Cassette (line level
device) into a mic input will overload the input and cause severe distortion. Plugging a microphone
into a line input will give a weak and muffled signal which will not be useable. NEVER put speaker
outputs of an amplifier into any of these inputs. This will cause severe damage to the mixer.
Some true broadcast consoles set their input level using a jumper on the circuit board inside. It can be
set to line level or microphone level. This must be done before the cables are wired to the mixer or
signals are applied to the inputs.
1) Types – microphone or machine/line
On an analog broadcast type mixer there are two types of inputs for sound sources. Microphone
inputs which tend to be mono and low impedance and high sensitivity. Line level inputs tend to be
stereo and higher input impedance. Line level inputs generally can not be used for microphones as
the sensitivity is too low. When ordering these type of mixers, balanced or unbalanced inputs have to
be specified for the line level inputs. Analog sound reinforcement mixers tend to have mono channels
which can be used for microphone inputs or mono line level inputs. The “trim” adjustments are used to
set the sensitivity for the given channel.
2) Impedance high- or low- levels
Broadcast mixers tend to have jumper wires or switch settings to select the input impedance for a
given channel as well as the input level. On line level channels the input signal level has to be set for
the type of equipment being connected. Consumer equipment has an nominal output of -10dbm,
professional equipment has a nominal output of +4dbm.
3) Types of connection
On broadcast consoles the connections tend to be terminal strips which the cables from the audio
sources are wired in rather than ¼ inch or XLR connectors used on other types. Some also have
special connectors which the cables are wired into that are then plugged into the mixer.
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D. Outputs
1) Balanced or unbalanced
Outputs from both broadcast and sound reinforcement mixers tend to have both balanced and
unbalanced outputs. The main outputs tend to be balanced and the sub-channels or monitor channels
tend to be unbalanced. The main channels in a radio station should be fed to the transmitter and the
monitor channels should be used for cue and monitor speakers. Professional transmitters will require
balanced lines.
2) Impedance
Generally both types of mixers will have low impedance outputs. Most often the output is nominally
600 ohms. It is best to check with the owners manual for the given transmitter to see if it has 600 ohm
input. Most often the transmitter will have a higher impedance than 600 ohm but is compatible with a
600 ohm output from the mixer. It is best to check with the manual of the mixer to see if it requires a
600 ohm termination. Some transmitters will have this as an option. Having this termination set
properly will give the correct audio levels into the transmitter.
3) Levels
Again, both types of mixers tend to have a setting or a switch for the output level to select between;
10dbm and +4dbm. Most professional transmitters require a nominal +4dbm input level. Some sound
reinforcement mixers do not have this option and are set at -10dbm.
4) Types of connection
Most broadcast mixers use wire terminal connections for the output lines the same way as the input
lines are wired. The balanced lines will have three terminals; hot, return and shield. Some newer
smaller lower priced broadcast mixers are using three pin 1/4inch connectors. These tend to be
cumbersome for permanent connections and can be prone to having wires break when they are left
hanging from the 1/4inch connectors. Most sound reinforcement mixers will generally use XLR
connectors for the main outputs.
5) Number
In general, broadcast mixers will have their outputs labeled for the function they perform. Some even
have small amplifiers and speakers built in which can be used for cue and monitoring. They will also
have one or two of these secondary outputs available for control room monitor speakers or even
recording room monitoring speakers.
6) Multi-purpose
Sound reinforcement mixers tend to have more outputs than broadcast mixers. They have monitor
outputs and effects outputs. These are all line level and can be used for what they are intended but
are really multi purpose and can be used for the cue bus and monitoring just the same as a broadcast
console. It is even possible to use one of these output “buses” to feed into a voice activated relay
(such as manufactured by Radio Design Labs) which can bring on an on-air light. This would be set on
only the microphone channels. This copies the “tally” relay built into broadcast mixers which costs
more money.
7) On-air
A convenient feature of the broadcast mixer is the on/off button on each channel. This allows the
operator to leave the fader set to a given level and toggle the audio on and off if it is needed; such as
the presenter's microphone being set to optimum level but needing to be switched off while someone
else is speaking into another microphone or while music is playing. Sound reinforcement mixers have
a mute button on each channel which essentially performs the same function; a feature which is not
lacking in the less expensive.
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8) Recording
The multiple outputs of both the broadcast and sound reinforcement mixers mentioned above can be
useful for recording programming while simultaneously broadcasting when this is desired. Broadcast
consoles with fewer actual output lines (buses) may in some cases need to feed the recording
equipment through a distribution amplifier. This would be necessary when the output is already being
sent to other equipment, such as monitor speakers. Without a distribution amplifier it is possible that
this signal may not be strong enough to give quality sound to more than one system at the same time.
Sound reinforcement consoles, as already mentioned, tend to have more output lines. This makes it
very convenient to record programs at the same time they are being broadcast. Any one of the line
level outputs, such as Aux send, Monitor send, or Effects send, can be dedicated for this task. In
general, a distribution amplifier would not be needed in this case.
9) Cue bus
Cue bus is used to prepare the next item for broadcast, whether music or speaking program, while
something else is on the air. The cue channel on the mixer does not affect the on-air channel but
sends the audio to a separate output which can be headphones or speakers installed only in the
control room. The operator can then cue the next media (CD, cassette, computer) to the beginning of
the song or program which is to go on the air next. Any material played on the 'cue bus' does not go
out over the air. Once material is cued, the switch for monitoring would then be put back to the on-air
signal.
Most broadcast mixers have a built-in cue bus and the monitor switch has a position labeled for it.
Some have a small amplifier and speakers built in. Other smaller mixers require separate amplifier
and speakers or powered speakers to hear what is played on cue bus.
Sound reinforcement mixers do not have a label 'cue bus' or a built-in amplifier and speakers. But, as
mentioned above, any one of the line level outputs labeled 'aux1, aux2, effects send, monitor send'
can be used for this function. Whichever is selected can be relabeled as 'cue bus'.
E. Testing the Signal through the System
When all of the inputs and outputs to the mixer have been wired correctly and it is time to play audio
through the system, go through the following basic steps to ensure the system is working
properly:
Step 1: Set channel slider (level control) to half volume. Never turn a level control to full for an initial
test or for troubleshooting. This could lead to sudden bursts of very high volume sound which can
damage microphones and speakers and hurt the human ear.
Step 2: Turn headphone or control room monitor speakers to approximately ¼ volume.
Step 3: Route the signal from the channel being used to the cue bus.
Step 4: Give the mixer audio input via a computer or CD player. It is preferable to use music for a
more constant signal for this test.
Step 5: The music should now be heard through the speakers connected to the cue bus at a low to
comfortable volume. If the sound is not distorted and is constant, the system is working properly.
Adjust speaker volume for comfortable listening.
Step 6: Adjust channel slider until a proper reading is seen on the cue bus meter (average -6vu to
0vu) with no red light blinking showing that the signal is peaking (distorting) and going above +6vu.
27
Step 7: If the music is not heard, turn the power off and recheck all connections between the mixer
and audio sources. Check that the wiring on the outputs from the mixer are correct. Begin at step 1
again.
Step 8: Before going on-air, if at all possible, send a 1000 Hz tone through the mixer. Set the output to
make the cue bus meter read 0. The manual for the mixer should tell what the level is in dbm when
the meter reads 0.
Step 9: Make sure the transmitter is switched off. Then switch to program and do the same as step 8.
Step 10: Adjust the master output control on the audio mixer until the correct nominal input level is
given to the transmitter.
Caution: Before switching on any radio transmitter, make sure the antenna and feed line are not
damaged and are connected properly. (See How to Wire a Transmitter Manual compiled by Galcom
International in partnership with Africa by Radio.)
F. Metering
Broadcast mixers often have both input and output metering. Whereas sound reinforcement mixers
rarely have input metering and often have a series of LEDs used as an output level meter.
A. Inputs
Broadcast mixers which have analogue or meters with moving needles will also have a red light to
indicate clipping (distortion – signal coming in is too strong). These meters work well but do need
annual calibration. Sound reinforcement mixers, while not having input meters, do have trim controls
(sensitivity control) plus a red LED which does indicate clipping on each channel. The trim control
should be adjusted for the audio source on that channel so that the average sound input gives 0vu
with the red LED never coming on. Broadcast mixers have switch settings or jumpers which determine
the level of input signal for each channel.
B. Outputs
Both types of audio mixer have level meters, as previously mentioned. Both the analogue type of
meter and the row of LEDs work well to give an indication of the signal leaving the mixer. The LED
style meters tend to respond more quickly and do not require maintenance.
1. Program Bus Meter – Measures the audio level leaving the mixer going to the transmitter (or to the
transmitter of the Studio-to-Transmitter Link system.) When this meter reading is averaging 0vu, the
system should be set up to be giving the transmitter the correct input level. For example, +4dbm.
2. Audition Bus Meter – This bus is used when there is a need for more constant monitoring (such as
listening to a receiver on the frequency of the station). It can also be used for recording programs at
the same time as a broadcast is going on-air. Hence, the Audition Bus Meter is indicating the level of
this material and not what is going out over the air. Its calibration, however, is the same as the
Program Bus Meter.
3. Cue Bus Meter – The function of Cue Bus is described in Section D-9. The meter indicates only
what is played through the Cue Bus. Again, its levels and calibration are the same as the Program and
Audition meters.
One disadvantage of sound reinforcement mixers is that they do not have the flexibility in metering
that a broadcast mixer has. It is quite easy to set up the functions of program, audition, and cue bus,
but in most cases the only actual metering that is built-in is the output meter.
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Analog Peak Reading Meter
Analogue Volume Units Meter—Classic Style
29
Section 4: Studio Operation
One of the best ways to become comfortable with the mixer console is to spend a little time with it in
private. The purpose is to learn cause and effect at some more appropriate time than on-air. Send
your program source through an input channel, play with the various channel controls, and monitor
the results. Connect your source to a stereo channel to do some of your tests and then connect to a
mono input to discover the differences. Focus on the controls you’re actually using the most. During
programming, relatively few controls need to be touched. Many of the controls on your console are
used to pre-adjust levels or determine the destinations to which signals will be routed. Most mixer
controls can be identified as belonging to either an input section (channel strip), or a master section.
This picture is of a channel section.
1. Level
Trim/Gain - Before you begin playing with any of the other mixer
controls, you need to first adjust input trim levels. Here you adjust
the level of the incoming signal to that which is optimum for
mixing. By proper trim adjustment you can prevent the addition of
noticeable noise and avoid overload distortion. It is used to adjust
incoming signal levels to be nearly the same for each source and
establish a starting point for the mix.
Some true broadcast consoles do not have a variable trim control.
The input level is selected by a jumper in the circuit board inside.
It can be set to line level or microphone level.
If the output is predominantly low frequencies, you may suffer
some intelligibility loss, or may end up with a “muddy” sound,
even after you have adjusted the gain control. If so, turn down the
low-frequency control on the channel EQ to restore a natural
quality. Microphones designed specifically for close vocal use are
often designed to have a rolled-off low frequency response when
measured at a distance. When the mic is up close, as intended,
proximity effect restores the low end.
To set Trim (Unity Gain) on a channel: first turn the input trim
control of the desired channel all the way down. Next, connect the
mic or line input to that channel. (Turn down or mute all other
channels.) Press the channel’s solo button - make sure it’s set to
PFL mode if you have a PFL/AFL option. Put music or speaking
(i.e. someone talking or CD or cassette playing) through the
channel, and turn up the channel’s input trim. You should see the
input level on the mixer’s meters. Adjust the input trim until the
meter level is around zero dB. Now you can adjust the channel’s
volume - with either fader or gain control - to suit your needs.
(Make sure the master volume is up!) In about ten seconds per
channel, you’ve optimized the mixer for maximum headroom,
minimum noise, and loads of extra gain above Unity. This will
allow you to get the most out of your mixer.
2. Fader
This is the slide control that allows you to adjust the amount of
signal that appears in the main mix. The fader allows you to adjust
the relative levels of all channels being mixed. (If your mixer has a
rotary, rather than slide control, it may be referred to as a “gain” or
simply “level” control.)
30
3. Tone
EQ (Equalizer) - The Equalizer is a way of increasing or decreasing the amplitude of a certain
frequency range within a signal. The bass and treble knobs on your home stereo are essentially
EQ’s. Some mixers have Low Cut Filters that cut all signal below a certain frequency. These tend to
be used on channels with mics because they can help eliminate mic thumps, wind noise, etc.
This is the area you will want to spend the most time with. Listen to a variety of program material to
determine how your mixer’s channel EQ affects the sound of the signal. Practice and listen with
different sources; male speaking voice, solo piano, violin and guitar, and test your EQ for the best
sound. For broadcasting each channel should be set for the most natural sound in a good set of loud
speakers in the studio or control room. When each program source sounds as natural as possible,
go on-air and monitor the sound from a radio tuned to your radio station. The sound may not be as
good as in the studio but should still have a very natural quality.
4. Noise
Reducing noise during the operation of the studio equipment is a matter of listening and adjusting
levels. The finest electronics, if improperly adjusted, can still produce hiss and hum. The amount of
noise each component (such as a mixer) will generate internally stays constant unless there is a
major breakdown or problem. This noise can be very pronounced at the output if the gain controls
are not set correctly by the user/presenter.
For example, you’re playing music on a CD plugged into a channel on your mixer. The volume level
on that channel is set a bit too low. But instead of raising the level of the CD, you choose to raise the
master level. By doing so, you amplify the CD music to the level you want to listen, but at the same
time, you have amplified the level of electronic noise, thereby making it louder and more
predominant in the program. The solution is to raise the level of the CD through the channel so the
noise won’t be amplified. It is also imperative to have the gain/trim control set properly as mentioned
above.
5. Feedback
Feedback occurs when sound from a speaker re-enters the microphone at a level that is equal to or
greater than the level of the sound arriving from the original sound source i.e. speaker’s mouth. It is
a signal-to-noise ratio problem. The greater the ratio between the levels of the desired signal, and
the unwanted signal the better. By simply decreasing the distance between the sound source and
microphone, you can dramatically improve your signal-to-noise ratio. Other tips include using a
directional microphone and maintaining a minimal number of open microphones to pick up unwanted
noise.
6. Signal Routing
A) Pan – This control is only found on mixers which have stereo output. It is used for placing the
sound from a source to the left channel, right channel, or centered between them. This feature will
only have an effect on listeners to radio stations which broadcast in stereo. This is almost always
done in an FM station which is so equipped.
B) Mute – This button turns off any sound from that channel so that it will not be heard. This feature
is used mostly on microphone channels to “kill” any undesirable sound such as coughing or shuffling
of papers from going on-air. This can be useful for the on-air presenter themselves or for control
over microphones used in interviews with people not familiar with being on the air. In some cases it
may also be necessary to switch between this source and another without changing the level setting
on the microphone.
C) Solo (cue bus) - This feature is used for monitoring the channel on headphones or control room
monitors while not affecting the main outputs. This is ideal to be used for “Cuing” the nest source
while on-air. This means that if CD one is on-air, CD two can be heard on “Cue” (Solo) and get it set
to the beginning of the desired song or program and not be heard on-air.
31
The Cue/Solo feature is then switched off. When it is time for CD two it is ready to go.
D) Aux Sends - These are level controls for additional mixes of your audio sources. They may be
routed to an outboard effect (such as reverb, delay, etc.), and returned to be added to the main mix, or
they may be used to perform a monitor function, or recording material at the same time as it goes to
air. This might be done in the case of a live interview which is needed to be aired again at a later time.
Some mixers have more than one “Aux” per channel. Usually there is a master “Aux Send” in the
Master section of the mixer. This control sets the overall level of all the Aux Sends of each channel.
7. Master Controls
a) Master Fader - This is your master level
control which controls the overall output volume/
level of the mixer. In broadcasting it sets the
level of the studio sound going into the
transmitter.
The Master or Output section of a mixer might
look something like this one. Each of the
controls is labeled with the same names
discussed in the channel strip section and
perform the same function only for the “Master”
or overall sound. In this case that is what goes
on the air.
A mixer used for broadcasting needs to have a
good quality meter indicating output level. It can
be a traditional analog type or led type, either is
good. When setting up the levels for the
broadcast system, the output meter should
indicate an average of around -3 to 0 db on
program material. At this level the system should
be installed to be working properly. This is the
level which the operator should be working in
and should never let the levels get so high that
the meters indicate constantly in the red area
above 0 db. This will cause distortion and other
problems in the equipment and will be very
unpleasant for the listener. The operator needs
to also guard against letting the output levels go
too low as well. If the meters are indicating down
around -15 most of the time the listeners will not
be able to hear some of the program material.
Control of levels is something to be learned and
is VERY important in good operation of a radio
station.
32
b) Metering - Keep an eye on things. If our signals are too strong, they will overload equipment. If they
are too weak, they will not overpower the background noise element in our electronics. We need an
indicator. There are many different types of audio
level meters, but the most common one is the VU
meter (VU stands for volume unit). The VU meter is
an "averaging" meter in that it doesn't respond to
sudden peaks in level - sort of like your ears. You
can get VU meters in the standard, needle movement form, or as an LED or LCD display. They are
calibrated in a scale that ranges typically from -20
VU Meter
PPM Meter
VU to + 3 VU.
Another meter commonly used in Europe (and increasingly, in North America) is the PPM
(peak program meter). This device reads peaks in level rather than averages. The argument
goes that although humans may not hear momentary peaks in loudness, the equipment does.
Therefore, the PPM is better insurance against signal distortion.
LED peak reading
meter
When mixing audio, "ride the gain" so the level stays between -5 and 0 VU; ride a PPM meter at +8
dBm. It's normal for peaks and dips to go beyond this, of course. Mix with a light, fluid hand. Pots
should not be jerked up and down to make adjustments at the slightest fall or rise in loudness.
Changes should be smooth - you'll hear abrupt ones. Once again, this is another good reason to use
the faders to activate channels, instead of the mute buttons.
d) Monitoring - Selector switch in the master section in which you determine what you hear in your
headphones or studio monitor speakers, and level controls that allow you to adjust their volume
without affecting the overall mix or volume.
d) Submix/Assign -You can assign channels to specific “Submixes”or “Group” faders. All of your
backing vocals may be sent to one “submix”, and the rhythm section may be sent to another. This is
sometimes called “group” mixing, as you can control the relative levels of groups of microphones/
sources with just one fader per group. In broadcasting this feature would be used more in the
production studio than on air. It can be used for ease in balance of mix between vocals and
instruments for example when recording a live group.
33
Section 5: Maintenance
How Tape Recorders Work
Brain, Marshall. "How Tape Recorders Work." 01 April 2000. HowStuffWorks.com. <http://electronics.howstuffworks.com/gadgets/audio-music/
A. The Tape Recorder
The simplest tape recorders are very simple indeed, and everything from a Walkman to a high-end
audiophile deck embodies that fundamental simplicity. The basic idea involves an electromagnet that
applies a magnetic flux to the oxide on the tape. The oxide permanently "remembers" the flux it sees. A
tape recorder's record head is a very small, circular electromagnet with a small gap in it, like this:
This electromagnet is tiny -- perhaps the size of a flattened pea. The
electromagnet consists of an iron core wrapped with wire, as shown in
the figure. During recording, the audio signal is sent through the coil of
wire to create a magnetic field in the core. At the gap, magnetic flux
forms a fringe pattern to bridge the gap (shown in red), and this flux is
what magnetizes the oxide on the tape. During playback, the motion of
the tape pulls a varying magnetic field across the gap. This creates a
varying magnetic field in the core and therefore a signal in the coil. This
signal is amplified to drive the speakers. In a normal cassette player,
there are actually two of these small electromagnets that together are
about as wide as one half of the tape's width. The two heads record
the two channels of a stereo program, like this:
When you turn the tape over, you align the other half of the tape with the two electromagnets.
When you look inside a tape recorder, you generally see something like this:
Spindles
Capston
Pinch Roller
Playback Head
34
At the top of this picture are the two sprockets that engage the spools inside the cassette. These
sprockets spin one of the spools to take up the tape during recording, playback, fast forward and
reverse. Below the two sprockets are two heads. The head on the left is a bulk erase head to wipe the
tape clean of signals before recording. The head in the center is the record and playback head
containing the two tiny electromagnets. On the right are the capstan and the pinch roller, as seen
below:
The capstan revolves at a very precise rate to pull the tape across the head at exactly the right speed. The standard speed is
1.875 inches per second (4.76 cm per second). The roller simply applies pressure so that the tape is tight against the
capstan.
In order for cassette players and recorders to perform properly there are two important maintenance
tasks which must be performed regularly.
First is cleaning of the tape path. Particles of oxide from the tapes become deposited on the head
and the pinch roller and other components in the tape path. When these particles are deposited on
the playback head it can cause the sound of the recording to be low and muffled and very hissy. This
is very disturbing to listeners. Other components in the tape path must be cleaned as well as the
particles become deposited on the pinch roller and capstan. If these items have too much build up
they will loose grip on the tape and not be able to pull it over the heads in a consistent manner. This
will also lead to the parts becoming sticky and the tape wrapping around the pinch roller and
damaging or destroying the tape. It is critical that the tape path be kept clean.
The second maintenance item is to demagnetize the tape path. Since the tape is magnetized and
being pulled past metal objects such as the head, capstan and other parts they also become
magnetized. Eventually the magnetic field on these parts will pull on the tape as it passes and the
tape will not pass over the head properly. This in turn will cause the frequency response to
deteriorate.
http://electronics.howstuffworks.com/gadgets/audio-music/cassette2.htm
35
Cleaning the tape path can be done by using “ear buds” sometimes called “Q-Tips” and “high
proof” (90% or more) isopropyl alcohol or high quality tape head cleaner. (Longer “buds” can be used
for hard to reach places in some player/recorders.) Other solvents or cleaners are NOT
recommended. They may damage the pinch roller and cause the tapes to wrap around it or to not run
at a consistent speed. Water should NOT be used! It will drip inside the machine and cause electrical
failure. Water also can not clean some of the parts properly. Rubbing alcohol should not be used due
to the other ingredients such as dyes which leave a sticky film behind. This again can cause tapes to
stick to parts of the tape path especially the pinch roller.
The frequency of cleaning and demagnetizing should be determined by how often the machines are
used and the quality and age of the tapes used in them. For a station which plays tapes on-air
everyday twice a day tape machines most likely should be cleaned everyday and demagnetized every
week. In other situations the frequency of cleaning should be often enough to keep the machines from
letting sound deteriorate due to build up of particles from the tapes and magnetizing of the tape path.
When metal parts of a tape machine are built up with the oxide can be see by looking at the heads
and capston as well as heard in the quality of the sound.
When cleaning the tape path use the clean end of a bud. Never put a used end of a bud into the
alcohol as this will make the alcohol dirty with the particles which come off of the tape machine. This
will then be placed on the next machine you clean rather than cleaning you will be placing dirt from the
last one cleaned.
•
•
•
•
Only dampen the tip of the bud do not soak it. The bud should not be dripping with the alcohol.
Rub the bud on the heads, guides and capston until no more dirt is picked up on the bud. This may
require repeating the process with the clean end of the bud and even another bud.
After these parts are clean engage the tape drive as if there were a tape in the machine and hold
the bud on the pinch roller careful to not let it get torn between the capston and pinch roller. When
the pinch roller is its originally color and does not have a band or ring around it the same color as
the tape stop the tape drive and remove the bud.
Allow the tape path several minutes to dry before loading another tape. This is very important. If a
tape is placed in the machine right away the alcohol may damage it!
Demagnetizing is done with a special tool for the job. It plugs into AC power and creates an
alternating magnetic field at its tip. It then is passed over the metal parts of the tape path in a specific
method. This removes the magnetic field in part of the tape path of the machine. The demagnetizer
needs to have a strong enough field to work on parts as large as the capston. Lower quality units may
not be able to do this.
When demagnetizing;
• Remove tape from and turn power off to the tape player/recorder.
• Make sure there is a plastic end on the tip of the demagnetizer so it does on scratch parts of the
tape path. If this plastic end is lost then cover the end with good quality electrical tape.
• Make sure that the power cord of the unit is in good condition and that it stays on when power is
applied. It will not work properly if power goes off during the process.
• You want to do the head(s), guides, and the capstan. Carefully move the demagnetizer tip in from
a few feet away to the part of the tape path to be demagnetized. Do not contact the parts of the
tape path. Move the tip up and down and in a circular motion in front of the individual part After
this, slowly move the demagnetizer straight away from that part until about one meter away.
• Turn off the demagnetizer.
• Repeat this process for each metal part of the tape path.
36
Decline in popularity
Analog cassette deck sales began to decline with the advent of the compact disc and other digital
recording technologies such as digital audio tape (DAT), and MiniDisc. Philips responded with the
digital compact cassette, a system which was backward-compatible with existing analog cassette
recordings for playback, but it failed to garner a significant market share and was withdrawn. Tascam,
Marantz, Yamaha, Teac, Denon, Sony, and JVC are among the companies still manufacturing
cassette decks in relatively small quantities for professional and niche market use.
Despite the decline in the production of cassette decks, these products are still valued by some. Some
audiophiles believe that cassette deck technology, due to its analog nature, provides sound recordings
superior to current digital technology, such as CD-R and DAT. However, cassette decks are not
considered by most people today to be either the most versatile or highest fidelity sound recording
devices available. One problem with fidelity is the removal of a tape type selector from many budgetoriented cassette decks. Without a tape selector to set proper bias and equalization settings, Type II
[High Bias] and Type IV [Metal Bias] tapes could no longer be used to their best effect. These tapes
were intended for high fidelity reproduction, but without the tape selector, only low grade Normal Bias
tape can be used at its best.
37
B. Reel to Reel Tape Machines
The cleaning and demagnetizing for reel to reel machines is the same as for cassette machines.
The same cleaner and buds can be used. The same demagnetizer can be used. On most reel to
reel machines the parts of the tape path are easier to reach than on cassette machines.
If there are problems after a tape machine has been cleaned and demagnetized do not try to
service the machine yourself. Replacing and aligning parts of the tape path should only be done by
someone who has been trained properly in these procedures. Attempting to service tape machines
without knowing how and some information from the service manuals can lead to making the
problem worse.
C. Compact Disc Players
CD Players/writers either in computers or as stand-alone audio players require less maintenance
than tape machines. There is no demagnetizing involved and cleaning is much simpler. Cleaning
involves using a special disk with brushes which continually sweep over the laser lens assembly in
order to remove any build up of dust. The disc usually has a sound track recorded on it which tells
the operator when to remove the disc. Other than keep dust from the controls and closing the disc
tray when not in use this is all the maintenance that should be done on CD players. Non-trainer
personnel should NEVER attempt repairs to CD players such as aligning lasers. This can cause
eye damage to the person and damage to the equipment if not done properly. CD players which
are not functioning properly should be assessed by a technician.
Photo from:
http://electronics.howstuffworks.com/cd4.htm
38
Section 6: Studio Acoustics
Some of the information presented here is referenced from sources in the United States and Canada
where materials for treatment of acoustic qualities of different rooms is readily available. In some
areas of the world not all of these materials are available or affordable. However basic sound
absorbing materials can be used to reduce undesirable effects in rooms with solid or rigid wall
surfaces. These materials and some good planning can also reduce the noise from outside sources
coming into the on-air studio.
Finish any preparations on the room first such as creosote being applied to any wooden structure
(i.e. the roof supports or other parts.) Decide the layout of the studio before starting construction.
Make decisions on where the presenter will sit and where guests will sit in the studio. Where will the
playback equipment be located? Where are the electrical outlets and how many are needed? Place
lighting so that presenters can see script pages, equipment controls and any computer screens that
will be used. It is best to not have starters and ballasts from florescent lights in the studio due to the
amount of electrical noise they make which can sometimes be heard on-air or in recordings. Any
construction which needs to be done to accommodate these things needs to be done before starting
to treat the room acoustics.
It is best to make basic changes to the room for on-air use before adding acoustic materials. Windows
if possible should be double panes of glass; one on the inside and one on the outside. They should be
installed with putty and wooden frames. If installed in metal frames there should be foam between the
metal and the glass. This is to reduce the sound from outside getting into the studio. This is called
isolation. Windows from the studio to another room to if there are any, should be installed with foam
between the glass and the frame, again for isolation.
Note the location of windows and doors in relation to work area and microphones. Place microphones
as far away from and windows to the outside as practical. This will help reduce noise pickup from
outside sources. Also the presenters’ position should be in the best possible location for ease of reach
to equipment without obstructions such as a chair bumping into furniture or reaching over the mixer to
reach a CD player or other needed equipment.
If possible, position the presenter so he/she is able to see door to the on-air studio while working. It is
a help to be able to see the door without turning around.
Budget for air-conditioning or proper cooling and heating as needed according to the local climate. If
central air-conditioning is unavailable or too expensive then it is best to use “wall mounted” units with
the compressor outside. These units are much quieter and generally provide more cooling than
window units. Window air conditioners can be quite noisy and may be picked up by microphones while
on-air or recording.
Most rooms which have walls of cement or other rigid material will create a great deal of echo and
boom to a wide range of sounds. This is a very negative characteristic for a studio. The best way to
stop echo and boom is to use acoustic material around the room in specific panels and install them
according to the room size and shape. A noise generator and spectrum analyzer is then used to
measure and adjust the acoustic material until the room response is equal at all frequencies and there
are no tones which change amplitude in different places around the room.
39
Acoustic Materials
Sample of “Rigid fiberglass
For information on other types of fiberglass see;
http://insulation.owenscorning.com/homeowners/insulation-products/quietzone.aspx
For acoustic foam see;
http://www.soundprooffoam.com/
[Many other websites have information on this topic as well.
40
In the situation where such test gear and materials are not available, a much simpler approach can be
used. It is not as accurate but will still be of great benefit. Start by clapping you hands while standing
in the middle of the room. Listen to the echo of the clap. Pay attention to the tones which echo the
longest and how many echoes there seems to be. Now do the same from approximately one third of
the way from the wall nearest the presenters position and again at the presenters position. Note each
time the tonal quality of the clap and how many echoes there are after the clap. It may be helpful to
write down what is heard at each location. This is to get a feel for the room acoustics. Now start
gathering the best acoustic material available or that the station budget can afford. Fiberglass (rigid or
bats of insulation), carpet, heavy blankets, heavy drapes or sheets of foam can be used.
Fiberglass is the best. There are now both “fluffy” bats and “ridged” panels available. One model of the
ridged type is 705-FRK by Owns-Corning. There is a series of 700 type fiberglass panels for this
purpose. The ridged panels are more absorbent per thickness. If these panels are not available or too
expensive, the fluffy bats will help also. The fluffy bats also have acoustic ratings also from some
suppliers.
Treat the ceiling and at least three walls of the studio room with these materials. It may be necessary
to install carpet on the floor. The three walls which should be treated are on either side and behind the
presenter. The wall in front of the presenter which would be behind the microphones may or may not
need to be covered. Now repeat the hand clap exercise in the same way as before the acoustic
material was installed. If there is no echo or ringing or noticeable “boomy” characteristic to the clap
then the room has been treated well.
If there is still significant echo or ringing then more material may need to be added to the 4th wall.
Some rooms when treated this way have a very “dead” characteristic sound and even sound
unnatural. When this happens then the room may need to have either the floor or some of the wall
space not covered with the acoustic absorbing material to give it a more “live” characteristic.
Some experts even recommend that rooms that have a very dead sound after being treated to cover
the walls in alternate strips. That is, cover the walls floor to ceiling two feet wide with sound absorbing
material. Then leave a two foot wide strip with either no covering or a harder surface such as plywood
mounted several inches in front of the original wall. This gives a combination of absorbing sound and
some reflection.
A. Methods of installing acoustic material;
1) For very small rooms where space is limited the fiberglass or foam can be glued or screwed to the
walls directly. For appearance and protection of the material cloth or burlap can be stretched over in
front of the acoustic material. This cloth material must not have metallic beads or decorations and
must be porous. This method will deaden the room and prevent echo from coming in the microphone
when on-air.
2) If space and budget allow, build a wooden frame around the room several inches to one foot out
from the original wall. Make two foot rectangular openings between vertical supports. Attach fiberglass
panels between the verticals. Do this on the ceiling also. This will increase the effectiveness of the
sound absorption at lower frequencies. If the fiberglass panels are not available then space the
verticals far enough apart to suspend the material used such as fiberglass bats, acoustic foam, or
heavy curtains.
3) If space or budget does not allow option number 2 but more than option 1, wooden 2x2s or 2x4s
can be used to suspend the material just in front of the wall. Place as many of the “panels” of the
acoustic material as possible 2 or 4 inches in front of the wall. As in option 2 fiberglass, foam or curtains can be attached to the 2x2 or 2x4 and they can be attached to the wall.
41
4)If the clap test mentioned earlier gave a distinct low tone characteristic or if the on-air room/studio
has been set up and there is a noticeable “boomy” characteristic or if any music will be live on air or
recorded in this room, then the following can be added to the acoustic treatment of the room. In each
corner a “bass trap” can be added in order to reduce low frequency reflection that are most likely
causing this.
B. Acoustic Treatment and Design for Recording Studios and Listening Rooms
By Ethan Winer (Used with permission) (This page was last updated on August 10, 2007.)
The following section is quoted from or based on an article on the internet. Some portions have been edited or
removed for the purpose of application smaller in scope. Direct use of the article is by permission from the author, Credits must remain when printing and copying this document. The article is written for the purpose of
acoustic treatment of a small recording studio but the basic principals apply to small radio studios. The desire to
reduce basic “acoustic interference” in a small recording studio is the same principal for an “on-air” studios.
[ Section from original removed for clarity]
BASS TRAPS - OVERVIEW
The most common application of bass traps in recording studios and control rooms is to minimize
standing waves and acoustic interference which skew the room's low frequency response. As you can
see in Figure 1 below, acoustic interference occurs inside a room when sound waves bounce off the
floor, walls, and ceiling, and collide with each other and with waves still coming from the loudspeaker
or other sound source. Left untreated, this creates severe peaks and dips in the frequency response
that change as you move around in the room. At the listening position, there might be near-total
cancellation centered at, say, 100 Hz, while in the back of the room, 100 Hz is boosted by 2 dB but 70
Hz is partially canceled.
Here, a positive wave front from the loudspeaker (left) is reflected off the rear wall
on the right, and the reflection collides
with other waves that continue to
emanate from the loudspeaker. Depending on the room dimensions and the
wavelength (frequency) of the tones, the
air pressure of the reflected waves either
adds to or subtracts from the pressure of
the waves still coming from the speaker.
Worse, different locations in the room
respond differently, with a boost at some
frequencies and a reduction at others.
When waves combine in phase and reinforce each other, the increase in level can be as much as 6
dB. But when they combine destructively, the dip in response can be much more severe. Level
reductions of 25 dB or more are typical in untreated rooms, and near-total cancellation at some
frequencies and locations is not uncommon. Further, most rooms have many peaks and dips
throughout the entire bass range, not just at one or two frequencies. Figure 2 below shows the
frequency response of the 10- by 16-foot untreated control room at a friend's studio. Note the large
number of ripples, and their magnitude, all within just one octave!
[Graph of untreated room response removed for clarity in context of this document.]
42
The action of sound waves colliding and combining in the air is called acoustic interference, and this
occurs in all rooms at all low frequencies - not just those related to the room's dimensions. The only
thing that changes with frequency is where in the room the peaks and nulls occur. The principle is
identical to how phaser and flanger effects work, except the comb filtering happens acoustically in the
air.
The only way to get rid of these peaks and dips is to avoid, or at least reduce, the reflections that
cause them. This is done by applying treatment that absorbs low frequencies to the corners, walls,
and other surfaces so the surfaces do not reflect the waves back into the room. A device that absorbs
low frequencies is called a bass trap. Although it may seem counter-intuitive, adding bass traps to a
room usually increases the amount of bass produced by loudspeakers and musical instruments. When
the cancellations caused by reflections are reduced, the most noticeable effect is increasing the bass
level and making the low frequency response more uniform. As with listening rooms, bass traps are
also useful in studio recording rooms for the same reasons - to flatten the response of instruments
captured by microphones and, with large studios, to improve the acoustics by reducing the low
frequency reverb decay time which makes the music sound more clear.
For recording engineers, problems caused by standing waves and acoustic interference are often first
noticed when you realize your mixes are not "portable," or do not "translate" well. That is, songs you
have equalized and balanced to sound good in your control room do not sound the same in other
rooms. Of course, variations from different loudspeakers are a factor too. But bass frequencies are the
most difficult to judge when mixing because acoustic interference affects them more than higher
frequencies.
Another problem is that the level and tone quality of bass instruments vary as you walk around the
room. The sound is thin here, too bassy over there, yet not accurate anywhere. Indeed, even if you
own all the latest and most expensive recording gear, your mixes will still suffer if you can't hear what's
really happening in the low end. Aside from portability concerns, it's very difficult to get the bass
instrument and kick drum balance right when acoustic interference and modal ringing combine to
reduce clarity. And when every location in the room has a different low-end response, there's no way
to know how the music really sounds.
[The article here talks about recording engineers. For the purposes of installing an “on-air” studio we are interested in the
basic principal of reducing the low frequency reflections which may be causing a boom effect being picked up by any of the
microphones used in the on-air studio. This should only happen in special cases of on-air studios as mentioned earlier.]
[ Section on size of speakers and equalization removed for clarity.]
[ In on-air studios headphones are often used for monitoring to avoid feedback from the microphone picking up sound from
open loud speakers. Loudspeaker monitors are normally muted when microphones are open. This reduces some of the problems talked about in this article, however in some cases there may still be the kind of acoustic interference talked about here.
Using headphones will only allow hearing each person talking into the microphones but will not cure the acoustic problem in
the room]
43
FIBERGLASS BASS TRAPS
There are a number of ways to create a bass trap. The simplest and least expensive is to install a
large amount of thick rigid fiberglass, spacing it well away from the wall or ceiling. As noted earlier,
705-FRK that is four inches thick and spaced 16 inches away from the wall can be quite effective to
frequencies below 125 Hz. But many rooms have severe problems far below 125 Hz and losing twenty
inches all around the room for thick fiberglass and a large air space is unacceptable to most studio
owners and audiophiles. Fortunately, more efficient bass trap designs are available that are much
smaller. However, studios on a tight budget can apply rigid fiberglass in the room corners as shown in
Figure 3a and lose only the small amount of space in the corners. Since bass builds up the most in the
corners of a room, this is an ideal location for any bass trap.
Wooden 2x2
Acoustic material, rigid fiberglass. [If not available
then see text for other materials as second choice.]
Figure 3a shows the corner viewed from above, looking down from the ceiling. When the rigid
fiberglass is mounted in a corner like this, the large air gap helps it absorb to fairly low frequencies.
For this application 705-FRK is better than 703 because the goal is to absorb as effectively as
possible at low frequencies. However, you can either absorb or deflect the higher frequencies by
facing the paper backing one way or the other, to better control ‘liveness’ in the room. Using 705 fiberglass that is two inches thick does a good job, but using four inches works even better. Note that two
adjacent two-inch panels absorb the same as one piece four inches thick, so you can double them up
if needed. However, if you are using the FRK type you should remove the paper from one of the
pieces so only one outside surface has paper.
Besides the corners where two walls meet as in Figure 3a, it is equally effective to place fiberglass in
the corners at the top of a wall where it joins the ceiling. With either type of corner, you can attach the
fiberglass by screwing it to 1x2-inch wood strips that are glued or screwed to the wall as described
previously. The 1x2 ends of these strips are shown as small black rectangles in Figure 3a above. One
very nice feature of this simple trap design is that the air gap behind the fiberglass varies continuously, so at least some amount of fiberglass is spaced appropriately to cover a range of frequencies.
When mounting 705-FRK directly to a wall - not across a corner - you'll achieve more low frequency
absorption if the paper covered side is facing into the room. However, that will reflect mid and high
frequencies somewhat. One good solution is to alternate the panels so every other panel has the
paper facing toward the room to avoid making the room too dead. Panels attached with the backing
toward the wall should be mounted on thin (1/4-inch) strips of wood to leave a small gap so the
backing is free to vibrate. For fiberglass across a corner as shown in Figure 3a, the backing should
face into the room to absorb more at low frequencies.
44
For a typical unfinished basement ceiling [or cement or other hard surface ceiling] you can take
advantage of the gap between the support beams and the floor above by placing rigid fiberglass
between the beams. Short nails or screws can support the fiberglass, making it easy to slide each
piece of fiberglass into place. Then cover the fiberglass with fabric as shown below in Figure 3b. You
can optionally pack the entire cavity with fluffy fiberglass one foot thick and you'll probably get similar
results. [**Note the fluffy fiberglass as an alternative.]
[Or unfinished ceiling]
[If “Rigid” type not available
this whole area can be filled
with “fully” style fiberglass.]
Figure 3b: 705 between support beams, covered with fabric.
Treating a "dropped" grid ceiling is even easier: Simply lay fluffy fiberglass batts on top of the grid,
above the ceiling tiles. The thicker the fiberglass, the better. One foot thick R38 is perfect for this if you
have the space. If you don't want to bother covering the entire ceiling that way, at least put fiberglass
batts around the perimeter to treat the important wall-ceiling corners. And since the fiberglass is not
exposed to the room and doesn't show, you don't need to cover it with fabric.
**Another great and inexpensive way to make a bass trap - if you have a lot of room - is to place bales
of rolled up [fluffy fiberglass] in the room corners. These bales are not expensive, and they can be
stacked to fill very large spaces. Better still, they are commonly available and you don't even have to
unpack them! Just leave the bales rolled up in their original plastic wrappers, and stuff them in and
near the room corners wherever they'll fit. Stack them all the way up to the ceiling for the most
absorption.
[**] [This type is readily available in most countries of the world.]
OPTIMIZING THE AIR GAP
While increasing the depth of the air gap does
indeed lower the frequency range absorbed,
for thinner panels it can also reduce the
absorption at some higher bass frequencies.
The maximum amount of absorption for a
given frequency occurs when the air gap depth
equals 1/4 the wavelength for that frequency.
Figure 4 below shows the velocity of a sound
wave, which is greatest at its positive and
negative peaks. Because the velocity is
greatest at the peaks, more energy is present
to force the waves through the absorbent
material.
Figure 4: A sound wave reaches its maximum velocity at 1/4 of its length.
But at half the length the velocity is at a minimum. Then it rises again at
3/4 length. This pattern repeats indefinitely.
45
The reason an absorbent material like fiberglass works
better when spaced away from a surface is that sound
waves passing through it have a greater velocity there.
As a wave approaches a boundary, such as a wall, the
velocity is reduced, and when it finally hits the
boundary, the velocity is zero. Imagine a cue ball as it
approaches the side rail on a pool table. The ball could
be travelling 100 miles per hour, but at the exact point
where it hits the rail it is not moving at all.
Likewise, fiberglass placed exactly at a rigid boundary
does nothing because the air particles are not moving
there. And since there's no velocity, the fiberglass has
very little effect. As fiberglass is spaced further from
the wall, the air particles passing through it have
greater velocity. They are slowed down as they pass
through the fiberglass, which converts the sound
energy into heat therefore absorbing some of the
sound.
In practice, you don't necessarily have to measure
wavelengths and calculate air gaps, and the first few
inches of space yield the most benefit. Most people are
not willing to give up two or more feet all around the
room anyway, so just make the gap as large as you
can justify. If you can afford to fill the gap entirely with
material, all the better. And even though the velocity is
indeed highest at 1/4 wavelength, there's still plenty at
1/8th of the wavelength too.
Note that the angle at which sound waves strike a
Figure 6: The higher frequencies (top) are absorbed well befiberglass panel can make the panel and its air gap
cause their velocity peaks fall within the material thickness. The
lower frequency at the bottom does not achieve as much velocappear thicker than they really are. Further, low
frequency waves that strike an absorbing panel at an ity so it's absorbed less.
angle may be absorbed less than when they strike it at
90 degrees, due to a "grazing" effect. The explanations in this section are a simplification and are
correct only for a 90 degree angle of incidence, which is not always the case in some rooms.
[ Small section on wavelength removed from original article for clarity.]
[ Section removed on better bass traps.]
46
ROOM SYMMETRY
Unless you plan to record and mix in mono only, the symmetry of your room and loudspeaker
placement are very important. If both loudspeakers are not situated symmetrically in a room they will
have a different frequency response, and your stereo imaging will not be balanced. In a room that is
longer than it is wide, it's better to place the speakers near the shorter wall so they fire the long way
into the room as shown on the left in Figure 11 below. This puts you farther from the rear wall where
the low frequency peaks and nulls are most severe.
Figure 11: Symmetry matters! In a typical stereo mixing room, the loudspeakers are spaced equally from the walls and corners, and form an
equilateral triangle at the mix position. The arrangement shown on the left above is better than the one on the right because it's more symmetrical within the room. The layout on the right also suffers from a focusing effect caused by the wall-wall junction behind the listener.
Besides positioning the loudspeakers symmetrically, you should also place your console and chair so
your ears are the same distance from each speaker. Likewise, acoustic treatment - whether
absorption or diffusion - should be applied equally on both sides. In many home studios it is not
possible to create a completely symmetrical arrangement, but you should aim for as close to this ideal
as possible. Especially in the critical front part of the room where the first reflections to reach your ears
are those from the side walls, and from the floor and ceiling if they're not treated with absorbent
material. What happens in the rear of the room is probably less important.
Although the sample rooms shown above in Figure 11 are rectangular, I prefer angled walls and an
angled ceiling because that provides deflection which reduces flutter echoes and ringing. Some
people argue that parallel walls are preferred because you can better predict the room modes, and
then treat the inevitable flutter echoes with absorption. But as I explained earlier, simply knowing the
modes is not always that valuable, and with angled walls you can make the average dimensions
comply with the ideal ratios. Further, if a room has parallel walls that must be treated with absorptive
material to avoid echoes and ringing, you may not be able to make the room as live as you'd like.
47
One somewhat controversial aspect of control room design is soffit mounting the main loudspeakers.
Most home studio owners simply put their speakers on stands, or sit them on the mixing desk, and
leave it at that. But many pro studios prefer to install the speakers into the wall so the front surface of
the speaker cabinet is flush with the wall. There are sound scientific reasons to use soffit mounting,
yet some engineers say it's not necessary or that it gives poorer results. Those in favor of soffit
mounting point out that it reduces reflections called Speaker Boundary Interference, or SBIR, that
cause peaks and dips in the low frequency response. If a loudspeaker is out in the room away from
the wall, low frequencies from the rear of the cabinet will bounce off the wall behind it and eventually
collide with the direct sound coming from the front of the speaker. (Even though it may not seem
obvious, very low frequencies do in fact leave a speaker cabinet in all directions.) Proponents also
claim that soffit mounting improves stereo imaging by reducing mid and high frequency reflections.
I happen to side with those in favor of soffit mounting, yet I also respect the opinions of those who
disagree. One thing nobody will dispute is that soffit mounting requires a lot more effort! If you do use
soffit mounting, please understand that the speakers must be built into the real wall. You can't just
apply a lightweight facade around the front of the speaker cabinet and expect the same results.
[ This section is mentioned as some FM stations are stereo and some are mono and monitor in stereo. It is still important to
place the monitoring speakers properly.]
In a more typical room I recommend a mix of hard and soft surfaces for the walls, with no one large
area all hard or all soft. I suggest applying absorbent material to the walls using stripes or a
checkerboard pattern to alternate between hard and soft surfaces every two feet or so. This makes
the room uniformly neutral everywhere. You can make the spacing between absorbent stripes or
squares larger or smaller to control the overall amount of liveness. If you are using 705-FRK rigid
fiberglass or an equivalent product, you can cover more of the wall and still control the liveness by
alternating the direction of the paper backing. That is, one piece of fiberglass will have the paper
facing the wall to expose the more absorbent fiberglass, and the next piece will have the paper facing
out to reflect the mid and high frequencies. In fact, when the paper is facing into the room the lower
frequencies are absorbed even better than when it is faces the wall.
LIVE OR DEAD - WHICH IS BEST AND WHERE?
[Section deleted for clarity.]
[Paragraph on plywood bass traps removed as not mention in this document.]
[ Paragraph of cello teacher and ceilings removed as not applicable.]
48
NOISE CONTROL
Reducing noise and sound leakage is beyond the scope of this article, but I will share a few tips studio
owners may find useful. If your studio has forced air ventilation, be sure to place the microphones
away from the vents while recording. If the vents have adjustable deflectors, set them to direct the air
away from where you normally place your microphones. Better, allow the room to get to the desired
temperature before you start recording so you can turn off the blower. You can turn it on again
between takes if needed. Likewise, radiators often make creaking sounds due to expansion and
contraction as they warm up and cool down, so use them before you start recording.
Another troublesome noise source in many studios is the fan noise from a computer. You can buy a
low noise replacement power supply from PC Power and Cooling and other companies. Easier, buy a
computer from one of the better manufacturers because they often have much less fan noise than the
cheaper brands. My last three computers were Dells, and they have all been very quiet. The small
premium you pay for a better brand is easily gained back by not replacing the power supply or having
to build or buy a sound proof enclosure.
I also attached 703 fiberglass wrapped with fabric to the rear and underside of my desk, as shown in
the photo below (left), to absorb the fan noise rather than reflect it into the room. Between the Dell's
quiet power supply and the fiberglass, I can record myself playing the cello or acoustic guitar while
sitting in front of the computer, with the mikes pointed right at me and the computer, and still pick up
very little noise. A second piece of 703 (right) can be placed in front of the computer to reduce the
noise even further while recording [or on air with an open mic].
One easy way to reduce noise from a computer is to line the surrounding surfaces with absorbent material.
If you've done all you can to reduce ambient noise and it's still too loud in a recording, consider using
digital noise reduction. Many programs are available that do a remarkable job of removing any type of
steady noise - not just hiss, but hum and air conditioning rumble too - after the fact. I use Sonic
Foundry's Noise Reduction plug-in, but other affordable programs are available that also do an
excellent job.
[This is only in the case of recording. For on-air anything that is picked up by the microphone will have
to be eliminated by the methods discussed in planning the layout or the room or in the acoustics.]
49
MORE RESOURCES
I have tried to make this article as complete as possible, but it is impossible to cover every aspect of
acoustics. Many books have been written about acoustics and studio design, and my goal here has
been to cover only the issues that are most important to recording engineers and audiophiles. Further,
acoustics is as much an art as a science, and surely mine are not the only valid opinions. Fortunately,
the Internet offers many resources for more information including my own Acoustics forum at EQ
Magazine, John Sayer's Studio Design forum, the SAE web site, the Acoustics newsgroup, and Angelo Campanella's Acoustics FAQ. Perhaps the most valuable resource of all is Google, where you
can find web pages that cover nearly any topic.
SIDEBAR: WHY THEY'RE CALLED STANDING WAVES
If you've ever used an ultrasonic cleaner to clean jewelry or small electronic components, you've
probably seen standing waves in action. When you drop a pebble into a pond, a series of waves is
created that extends outward from the point of impact. Since a pond is large, the waves dissipate
before they can reach the shore and be reflected back to the place of origin. But in a contained area
like the tub of an ultrasonic cleaner, the waves bounce off the surrounding walls and create a pressure
front that makes them literally "stand still" within the cleaning solution. The exact same thing happens
in your control room when your loudspeakers play a sustained bass tone. Static nodes develop at
different places in the room depending on the loudspeaker position, the room's dimensions, and the
frequency of the tone.
50
REVISIONS
March 5, 2003: Initial release.
March 16, 2003: The paragraph just above Figure 5 was clarified by Wes Lachot. Thanks Wes!
March 20, 2003: Added Figure 3b and accompanying text showing how to install rigid fiberglass in an unfinished basement ceiling. Also
added a new paragraph under Table 1 explaining the importance of density in absorbing materials.
March 22, 2003: Added a new last sentence to the second paragraph after Figure 7 offering a better reason panel bass traps must be sealed
air tight.
March 25, 2003: Added the sidebar The Numbers Game to expand on the role of edges when testing absorbing materials.
March 28, 2003: Changed the formula for calculating the ideal 1/4 wavelength air gap (below Figure 5) from 1100 to 1130 feet per second,
which is more accurate.
April 5, 2003: Added the sidebar Big Waves, Small Rooms to better debunk the myth that low frequencies require a minimum room size.
April 28, 2003: Added text (just below Figure 3b) describing using large bales of fluffy fiberglass as a bass trap.
May 14, 2003: Added the sidebar Hard Floor, Soft Ceiling to explain in more depth why that combination is better than a carpeted floor with a
standard drywall ceiling.
June 10, 2003: Added a link to my Graphical Room Mode calculator program in the text below Table 2.
June 23, 2003: Added three sentences to the end of the paragraph under Figure 6 explaining how the angle of incidence can affect the absorption of a fiberglass panel. Thanks to Eric Desart for bringing this to my attention.
June 26, 2003: Added, clarified, and edited throughout to remove what some have perceived as my bias against foam treatment.
July 12, 2003: Added the online manual for ModeCalc (see June 10 above) as a sidebar, so you don't have to download and run the program just to read the explanation of modes.
August 28, 2003: Added the sidebar Creating a Reflection Free Zone above.
November 23, 2003: Added two paragraphs about diffusors in the section Diffusors and Absorbers, directly under the photo of the wood
diffusor made by ACID acoustics.
January 14, 2004: Removed the qualification that the front wall angle must be "at least 35 degrees" in the RFZ sidebar because Wes Lachot
says the exact minimum angle depends on several factors. Also made a few other minor edits to clarify the text.
April 21, 2004: Added a few more manufacturers of rigid fiberglass, and explained that FRK stands for Foil Reinforced Kraft paper.
April 26, 2004: Clarified the difference between a deflector and a true diffusor. Also, clarified the importance of density in porous absorber
materials like rigid fiberglass.
May 4, 2004: Minor edits to the RFZ sidebar to make it clearer that early reflections from either side wall are a problem, not just the left wall
with the right speaker and vice versa.
May 12, 2004: Added text under the Room Symmetry heading explaining the importance of having loudspeakers fire the long way into the
room. It was a slow day so I also added several other minor enhancements and clarifications.
May 27, 2004: Replaced the graph I had made showing absorption versus air gap with a more compelling graph from Everest's Master
Handbook of Acoustics. Also added test data from fiberglass manufacturer Johns-Manville showing how increasing the density of rigid fiberglass enhances its performance at low frequencies.
December 30, 2004: Enhanced ModeCalc to allow entering feet and inches (12'6") rather than having to use decimal feet (12.5).
February 11, 2005: Added a link to my Density Report in the section about fiberglass density. Added a new section under Bass Traps Overview to explain modal ringing. Added a paragraph to enhance the explanation of flutter echo.
February 21, 2005: Numerous minor edits, additions, and clarifications.
August 19, 2005: Added a paragraph to explain that two rigid fiberglass panels 2-inch thick can be combined to work as well as one panel 4
inches thick. Also added a warning not to have FRK paper between the layers, or on both the front and rear.
September 21, 2005: Clarified the difference between fiberglass bass traps that act on velocity versus wood panel traps that absorb via
damping wave pressure.
February 3, 2006: Clarified that density is only one factor that affects a material's usefulness as an absorber, and that at some point the density can be too high.
March 9, 2006: Added text and a graph to the sidebar Creating an RFZ showing how comb filtering is an equally important problem that is
solved by placing absorption at the first reflection points.
51
June 18, 2006: Added text near the end of the section Fiberglass Bass Traps explaining how to treat the cavity above a hung ceiling.
October 12, 2006: Updated the ModeCalc sidebar to link to the new Windows version, and updated the text to match.
August 10, 2007: Updated Figure 4 with an improved version submitted by Markus Mehlau.
Ethan Winer is a reformed rock 'n' roll guitarist who sold his successful software business in 1992 at
the tender age of 43 to take up the cello. Ethan has, at various times, earned a living as a studio musician, computer programmer, audio engineer, composer/arranger, technical writer, and college instructor. He has had more than 70 feature articles published in computer and audio magazines including
Mix, PC Magazine, EQ, Electronic Musician, Audio Media, Computer Language, Microsoft Systems
Journal, IBM Exchange, Strings, Keyboard, Programmers Journal, The Strad, Pro Sound News,
prorec.com, Recording, and Sound On Sound. He now heads up RealTraps, manufacturer of high
performance acoustic treatment, and also hosts the EQ Magazine Acoustics Forum at the MusicPlayer web site.
In addition to technical writing, Ethan has produced two popular Master Class videos featuring renowned cellist Bernard Greenhouse, as well as five CDs for Music Minus One including a recording of
his own cello concerto. Besides writing and recording many pop tunes, Ethan has composed three
pieces for full orchestra, all of which have been performed. He lives in New Milford, Connecticut, with
his wife Elli and cat Bear, and plays in both the Danbury Symphony and the Danbury Community Orchestra where he serves as principal cellist. You can read more about Ethan's musical exploits on his
Music and Articles pages. Top
Entire contents Copyright © 2003-2008 by Ethan Winer and RealTraps, LLC. All rights reserved.
Ethan Winer
34 Cedar Vale Drive
New Milford, CT 06776
860-350-8188
[email protected]
Used by permission for Galcom International, August 2008.
52
Conclusion
God is providing many exciting opportunities for ministry through low power radio broadcasting.
Radio is still the most affordable and effective medium in reaching people for Christ.
Assembling a broadcast studio takes long hours, hard work, and thousands of dollars. It also
requires great care to keep it faithfully maintained and functioning properly. This can be a struggle at
times, but it is a worthy struggle. “Let us not become weary in doing good, for at the proper time we
will reap a harvest if we do not give up.” Gal. 6:9, NIV.
The rewards of such commitment are only known in part, but the part we know is already exhilarating. We will only know the full impact in Heaven. Let us seek God’s guidance together in using this
powerful tool for His Kingdom work.
53
Bibliography
1) Mackie Church Sound Note Book
http://www.mackie.com/
Used by permission.
2) Build a Better Bass Trap
www.ethanwiner.com - since 1997
By Ethan Viner
Used by permission.
3) Acoustic Treatment and Design for
Recording Studios and Listening Rooms
This page was last updated on August 10, 2007
By Ethan Viner
Used by permission.
4) BEST PRACTICES
The Quest for Excellence in Canadian Christian Radio
This book is dedicated to the memory of Larry Kayser,
Chief Engineer of United Christian Broadcasters in Belleville,
who passed away to be with the Lord on
October 5, 2004. OCTOBER 2004
5)Community Radio Course by Roger Stoll
Radio World Wide
Used by Permission
Websites
http://www.danalee.ca
http://electronics.howstuffworks.com/gadgets/audio-music/cassette2.htm
Non copy write material.
http://www.soundfirst.com/cleandemag.html
http://en.wikipedia.org/wiki/Cassette_deck#External_links
Non copy write material.
54