Chapter 5.22
Church Sound
Advice
Traditional vs;
Contemporary – Stereo vs; Monophonic - Audio Boards – Mirroring – Wired &
Wireless Microphones – Speakers – Cue Systems – Meters – Foldback – Feedback -
Resonant Frequencies – Phasing – Headroom – Wind screens - Distortion – Noise
Gates – Peak Controller’s - Spectrum Analyzers – Notch Filters – Best Operating
Practices
Early
Audio
You have heard the term “Cutting a
Record”; in my early studio days we had record recorders which the
stylus actually cut into the vinyl the commercial that the disc jockey would
later play on the air. The shavings
looked like hair or whiskers as they peeled off behind the stylus. They were fun to stick in your nose to “break up”
the DJ while he was live on the Air.
The process was called the “Hill and Dale”
method for monophonic recordings, with the stylus moving up and down; later
when two sources of sound presented itself, it was called “Binaural Sound” which later became
known as stereo. In addition to the Hill and Dale method of cutting a disc, the stylus would
now also move laterally left and right to produce the second channel of audio. So
now the magnetic
phono pickup cartridge had to accommodate the stylus moving up and down
as well as left and right.
After a few passes of a record being
played, scratches would appear and create high frequency noise in the sound; so the
industry created a “pre-emphasis amplifier” that would record the
music or commercial with the highs extremely increased and when the record was
then played back it went back through a special “phono pre-amp” that de-emphasized the highs back down to normal levels. Now when the record got scratches, those high frequency
scratches were pushed way down; de-emphasized
greatly and were not so noticeable. That’s why a record player has its own special
“Phono Pre-amp”.
Later the bare needle was replaced with a diamond tip to reduce wear and
scratches and a spray was developed to spray on to clean and “in effect”
lubricate the stylus movement.
Conclusion: Amplitude Modulated Noise has always been a problem for the audio engineer. With today’s technology moving into the
Digital format, unwanted noise has not just been reduced, but eliminated in
recordings. During pauses in passages
you do not hear tape hiss or high frequency scratches; you hear “the Sound of
Silence”.
As a fun note:
in the early years, most Radio station owners had both an AM & FM station.
The first stereo
radio transmissions where accomplished by sending the left audio
signal to an AM
radio station and the right channel signal to an FM radio
station. Folks at home at the
advertised hour would place both AM & FM receivers on a table and then tune
in both stations to the different frequencies and would sit back in amazement
at the different sounds. The first one I
heard was “The
Theme from Picnic” which has two different, but complementing themes
running throughout the selection & presented on the separate speakers.
The recording medium
“standard” in the 50’s became vinyl tape coated with iron oxide. As an Audio Engineer for Educational Radio
and TV, my job was to edit audio tapes, taking out coughs, words pronounced
wrong and unwanted sounds from the outside. To do so, you had to physically CUT THE TAPE and remove the
unwanted section and then you re-inserted Room Tone back
in its place, so that the passages would sound natural.
ROOM TONE was recorded for
two minutes after the recording session ended.
Everybody in the studio set silent the entire time. This recorded room tone was then re-inserted into the tape so that you would not
notice any change in the background sound; sometime it meant inserting only 3/8
inch of tape, if it was not enough, you took it out and added another 1/4 inch
or so to it.
This tedious time consuming job was not everybody’s cup of tea; you must
love audio to put up with what some would call foolishness. I call it pride, doing your work as into the Lord.
Colossians 3:23
– The Book – Do your
work as unto the Lord
23 whatsoever ye do, work heartily,
as unto the Lord; not unto men;
Audio
Boards
In the early days the audio boards
contained tubes with the power supplies remotely located in a rack to reduce
hum and was fan cooled; everything was hard wired into the
board. Every microphone or device passed
through a patch
panel bay first. The only thing that could be plugged
in was a microphone in the studio. The
audio consoles were made by GE or RCA and could do only three things. 1) Turn the mic off
and on 2) Put
the mic in audition /cue mode to hear it on a cue speaker and lastly, 3) vary the gain sent to the audio board with a rotary POTENTIOMETER which for short
was called a ‘POT’”,
with only one main VU meter. (To some
today, a POT is something you smoke.)
Today’s Audio Board inputs are not hard
wired, but have balanced audio connectors with the “XLR“ being the most popular, but
also 1/4
inch TRS jacks (Tip, Ring, Sleeve) as well as
unbalanced RCA inputs. Some have
the inputs on the top of the console instead of the back for ease of setup and
teardown. A two conductor input is said
to be “unbalanced”
and is usually high impedance; and a three conductor is said to be “balanced”
and has a low impedance. The “balanced input”
reduced amplitude modulated interference problems such as 60 cycle hum.
VU is Volume
Units and
is measured in DB’s or decibels. A db was
the amount a human ear could detect a change.
An increase of 3db doubled the
sound that the ear distinguished and a decrease of 3db cut the sound in half as
the ear could detect.
The Standard Reference Level was said to
be “0”db which provided an adequate feed to recording devices,
amplifiers and transmitters. Later this
standard would move to +8db which was much
better concerning pushing down noise
on transmissions lines such as was provided by the telephone company
that connected the studio to the transmitter site.
Level Matching became the
challenge for the audio engineer. If
your standard reference level is 0db and you have a recorder that has a +4db output
and another one that has a +8db output and
cart machines that have 0db output, you must PAD down every device above 0db before
they get to the audio board, making all levels the same at the point of input.
Modern Audio Boards do not have rotatory POTS, they have individual strips with sliders or
faders which do the same thing, but each slider/fader is now on its own input,
and have supporting “Pots” that can PAN the amount of signal sent
into left and right stereo channels. For
“SWEETENING” they also have
Frequency Band
Equalizers, Reverbs, Rumble Filters, Routing Program switches, Noise Gates,
as well as input
attenuator TRIM Pots or PADS
to bring the levels into your operating standard. There is also auxiliary “Pre-fade” pots that send a feed to the performers fold back speaker
or in-ear monitor. This is independent
of the house mix.
A mixer used for simple remotes might have 6
channels where a night club might have 24 channels and a concert some 96
channels; a church might have 16 inputs which would be fine for a traditional
service, but it would be pushing it for a Contemporary Service with a Praise Team, multiple guitars, base
guitar, drums, keyboard, piano and special effect instruments. Some Churches have two separate services; Traditional
and Contemporary
and two different audio board operators. It’s just two
different sounds.
The audio engineer can set all the faders to the mid position and then
adjust the input attenuator for 100 percent modulation on its INDIVIDUAL VU METER which is a modern enhancement.
Now a HOT input signal
does not force you to run the fader at the bottom of its range and a WEAK signal does not force the fader to the
top of its range.
NOTE: Early Pots or faders were made of carbon
and the actual audio passed through them and over time, the Pots would become
noisy as you raised and lowered the level.
Today the fader can control a light with a photo cell or control a
transistor’s output and are operational NOISE FREE.
Noise
Gates
Noise Gates are a preset threshold that a signal must
exceed in order for the gate to open and allow the signal to pass. When it is closed, you hear nothing. The term “Noise” applies to any signal that is
below the signal threshold. Gates
eliminate the background noise from “on stage” amplification systems, when a
vocalist is not singing or an instrument is not in use. Noise gates can be “chained”
so that one gate can cause other gates to open if desired.
This is not a
noise rejection circuit; if noise is present when the threshold gate is
opened, the noise will also pass. The
adjustments are Attack,
Hold and Release (fade out). Noise gates can “chatter” (open and close) which is caused by spurious undesirable
sound fluctuations A “Hysteresis” circuit in the more expensive audio
consoles greatly reduces this “chatter”.
Subgroup
Routing
Specific inputs can be mixed on a
subgroup channel and then inserted into the main mix. For instance, drums in an enclosure could
have a number of microphones with individual faders that are grouped on just one fader for insertion into the final house mix. Instead of adjusting many faders for the
drums, you would just adjust just one; same for the vocals.
Meters
VU meters (with needles) indicate the
levels for each channel; Peak Meters are
often LED’s
on each channel that light up different colors so that at a glance you can tell
if the signal is too HOT and is clipping the audio. Some systems use GREEN to indicate signal
presence and YELLOW
for a signal approaching clipping and RED to indicate clipping/distortion.
Cue
Systems
Early audio boards cue system was selected by
rotating the pot all the way down until it clicked into the cue position which
then could be heard on the control room cue speaker. When the DJ opened the Control room mic, it would
trip an “ON
THE AIR” light outside the door and cut the cue speaker and send the
cue signal to his headphones. He could
then cue up records as he talked live on the air. When he opened a studio mic; the fold back speaker would be silenced in the studio and a “Quiet”
or “On the
Air” sign would light outside the studio doors.
It was not unusual to see the audio board
raised up about 5 inches to accommodate mercury switches below each pot that
would turn on the appropriate TURN TABLES. However, the most common method was to back
cue the record
1/4 turn and then hold the “Platter” and let it “Slip” on the turntable felt until
you were ready and then you just released it.
Today’s cue system utilizes a PFL “Pre-Feed Listen”
pot also known as a “Solo” pot or just simply “Cue”. There is also a SIP “Solo in Place” button which
literally mutes all other inputs so that you can only hear that specific audio
input and troubleshoot it; CAUTION this function would be a disaster if depressed during a live
performance.
Talk-Back systems utilize
a microphone at the audio console for the sound engineer to communicate with an
announcer in a announce booth or a performer on stage which may be some
distance away. Talk-Back might utilize
speaker monitors, or heard silently in headphones or an earpiece. This can be observed on TV newscasts where the
anchors wear an earpiece with a stretchable coil of wire located just behind
their neck.
TEST
TONE oscillators
are on some audio consoles that are used for troubleshooting and testing
channels during set up. Many audio
engineers prefer “Unity Gain” between pieces of electronic
equipment; meaning the same level coming in is the same lever coming out.
Mirroring
When mixing sound for a “Live”
performance in an auditorium and also mixing sound for broadcast, some consoles
can Mirror each other, making both consoles
exact duplicates of each other. The one
for broadcast or recording can then be re-mixed as needed.
In the event of hardware
failure TV
networks will have Mirror Audio Consoles so the listener never knows of a
problem. Many preamps have dual
amplifiers, so that if one side fails the other continues and the audio only
drops -3db which is only half the signal. The audio engineer just raises the fader to
increase the signal level back to normal.
Headroom
Headroom simply stated
is a term used for how much louder a signal can get above normal before
distortion sets in. For instance a 0db
audio board may have 10db of head room which means that when a signal “over modulates”
an amplifier, it can go as much as an additional 10db before distortion or
clipping occurs.
Wireless microphones have limited headroom
because of the transmission “bandwidth” of the transmitter and receiver; so
some hand held mics have selectable attenuators on the mic themselves. Recording devices are also limited depending
on the medium that it is being transferred to as well as Radio and Television
Transmitters which have limited bandwidth.
In the early days of Radio, broadcasting
was on AM
Amplitude
Modulated
frequencies and the owner of the station would want his station to sound louder
than the other stations in the market.
The higher the signal pumped into the transmitter the more the
Transmitter stayed at maximum power. The
results were the DJ would always have the audio pumping in the RED; blowing it through.
The FCC Federal
Communications Commission didn’t agree
with that and would issues costly citations to the station for “Over Modulation”. A
television station’s CONSTRUCTION
PERMIT PAPER would be worth
one million dollars before anything had ever been purchased. Fragrant “Over Modulation” violations
could result in the loss of the station’s license to broadcast; so over
modulation was a BIG DEAL.
This resulted in the invention of “Peak Controllers”
that would limit or push down the loudest audio signals to prevent over
modulation. This just gave the AM radio stations the license to SIN;
so they increased the audio levels even more making the Peak Controller work even harder.
It would suck up the lowest
levels and then compress down the
highest levels causing “Pumping” or “Sucking” of the audio.
To the audio fanatic, which I am; being raised as a child on
classical music in the 40’s and 50’s, this was JUST AWFUL because it destroyed the “Dynamic Range”;
this is the range between soft passages and loud passages, THEY ARE JUST LOST. Today audio pumping as well as distortion is injected
as special effects into the music.
Since the introduction of FM radio, the frequency
is Modulated, not the Amplitude Modulated. Therefore there is no need to SLAM the audio, because it gets no louder, the power stays
the same. Today Radio and Television
stations use an ALC
Automatic Level Control at the studio that gently raises or lowers
the base
platform of the average signal
plus or minus 10db. The low passages of
an old western movie would be gently and slowly raised as it was needed to the
standard level. Then at the transmitter
site a “Peak
Controller” or limiter would attack the high levels spikes of audio.
Many stations use a five band
loudness controller and limiter before
the program audio is sent by the microwave STL (Studio Transmitter
Link) to the Transmitter site.
This separates the audio into five discreet bands and attacks them
individually leaving the other frequencies unaffected. Sweet
Microphones
Microphones come in a variety of
configurations, but the most common is the Omni-directional which picks up 360
degrees around the mic and the others are directional.
Some you see at football games pointing backward into a parabolic dish
with filters that only pass the mid frequency range of the human voice. It rejects or greatly reduces all the other
sounds that are picked up “off axis” to what the operator is pointing
at. That’s how you can hear the calls of
the players. (Today law enforcement can bounce a laser beam off
your window pane and hear what is being said in the room, via the vibrations on
the glass pane.)
Phantom Power is power that is
supplied to a condenser or capacitor microphone. The mic cable is plugged into the power
supply first then to the audio console.
A DC voltage is sent via the audio cable to the microphone to power it
and the audio returns on the same cable.
Caution:
this must be a dedicated line to that specific microphone.
The one that is most used in churches would be the microphone that produces
a cardioid
pattern in one direction. This greatly
reduces the sounds originating on each side of the mic; it is a must for the
audio operator whether it be a vocal or a violin. Note; every microphone MUST be in phase with
each other. (see phasing)
Microphones are additive; every time you add another mic you are
increasing the sound that other microphones are also picking up, only at a
reduced level, but additive none the less. So the goal of the audio engineer is to make
each mic invisible or dead to the other mics. This is done through placement or
proximity. A vocalist who holds the
microphone near their mouth aids in making their mic invisible to the vocalist
beside them.
The caveat is that the vocalist needs to know how to “Work the Mic”.
Every singer has a frequency range in which they are effective, and anything higher or
lower than their range, their volume level falls off. So the
mic is moved in closer at the end of their range and further away in the middle
of it. Just a movement of an inch or two
WILL have a profound effect at the audio
board. When the vocalist BELTS
IT OUT, the microphone should be “worked” a good distance away. This is a SECOND NATURE ART that must be
learned and felt that only time can teach. The audio operator will either smile or
cringe.
A base vocalist uses lots of
air to produce a low note and is sung very soft otherwise he will run out of
air; the results is that he needs to almost swallow the mic. They CUP their hand from mouth to their ear just
to hear themselves. They may also be
seen moving the mic rapidly in and out close to his mouth creating a vibrato
sound. A guitarist will do the same
thing as he shakes his guitar on certain notes.
Presence
Frequency Range is dependent
upon the vocalist as well as the type of instrument being played; a tuba has a
lower range than that of a flute and obviously vocalist by their very own
nature find themselves singing as a soprano, alto, tenor or base; their voice
ranges are just different.
Presence is a term that
is used to indicate that something is displaying additional quality of
sound. The Loudness Control on your stereo system ENHANCES THE LOWS while playing at low volume levels. It does not make the overall sound louder, it
just makes the frequency response “Fuller”. When a microphone is used close to
the performer’s mouth, the lower frequencies are enhanced; they have a better “Presence”. Lower frequencies are vocalized softer and
require more AIR to get the sound waves moving.
Wind screens are used inside and
outside in order to prevent the explosive sounds that occur when a burst of air
emits from the vocalist on certain sounds, particularly the “P” sound.
Wireless Microphones
Wireless Microphones in their
infancy were a joke. You would be
listening to the Pastor using his wireless mic and out of nowhere you would
hear “10-4
Rubber Ducky… I’ll catch you on the Flip Flop”. A wireless mic might only have a 1/4 watt
transmitter and a Public Service Vehicle or an Armature Radio operator might
have a 35 watt transmitter which is no match for the wireless receiver.
Today wireless mics have their own
FM frequency spectrum with squelches, filters and dual receiving antennas. Sometimes a pastor wanders out in front of the
speakers causing “Feedback”.
Technology today allows pastors to wear
an invisible tube that conducts the sound to the pickup coil behind his ear. The extreme proximity of the mic to the mouth
allows the audio operator to lower the gain on his mic making it almost
invisible to the auditorium speakers.
This can be a big deal concerning FEED BACK when the pastor moves around the stage
area or even into the isles.
Wireless receivers come equipped
with a squelch
that kicks in and shuts off the audio if the signal is too weak; otherwise it
would blow white NOISE at a high level into the sound board. This is noticed as the user’s audio cuts in
and out; or it could be DEMONS in the
audio board; maybe.
Antenna placement can be
troublesome in some rooms. It is best to
extend the antenna to the middle of the room in the ceiling or even under the
podium so that when the user moves, he is always moving toward the antenna, not
further from it. I had to mic an Actor
coming onto shore in a boat and then running down a beach for some 400 feet, so
I buried a 400 foot wire in the sand all the way to the antenna on my
receiver. As she talked to the camera in
the boat she was always moving closer to the antenna and then she ran on top of
the wire; I never lost her signal once.
Distortion is often noticed
with wireless mics. Let’s talk
technically for a minute; the quiescent point of a transistor allows the
amplifier to amplify the audio signal positive and negative equally at the same
level. When the batteries get low in the
mic transmitter, the lower voltage causes the quiescent point to move downward,
now the Positive swings are just fine, but the negative swings are clipped,
squared off and we hear it as distortion.
It’s not the mic, it’s the batteries and it’s the audio operators fault.
You must be aware of the time on the
batteries and when it’s time to
change them. When in doubt start fresh. It’s
not the batteries fault; it’s your fault.
Phasing
Phasing is very important in microphones
as well as in speakers. If one
microphone is wired differently that the others, it would be said to be 180
degrees “Out
of Phase” and therefore would “Cancel Out” the same sounds that the other
microphones picked up. It is referred to as “Frequency Discrimination”. This also occurs with MULTIPLE SPEAKERS.
SPEAKERS A sound engineer would walk across the back of an
auditorium and listen to the speakers; if the sound diminished or produced what
is called “Dead
Spots” on his walk, he knew he had a speaker or two that were “out
of phase” and needed to be re-wired. Speakers are color coded at the factory for
“Plus & Minus” or Positive & Negative”.
There is actually no polarity in an “AC” circuit, but it just denotes the polarity for speaker phasing.
NOTE: If
the polarity of a speaker is not marked, you can take a 1 ½ volt battery with
two clip leads and place on the two terminals of the speaker. If the speaker pushes outward the terminal
with the positive lead on it would be positive, if not, reverse the wires to
get the outward movement for positive and then mark it.
MICROPHONES A sound engineer
doing microphone checks before a live concert would set all the mic’s at their
mid-level on the sound board, then place his favorite microphone on a stand and
began to speak into it. As he spoke, he
would bring in a second microphone alongside of it and speak into both of them,
his assistant in the back of the auditorium would give a thumbs up if the sound
“Increased”
and a thumb down if the sound “Decreased”.
If the sound decreased, it would mean that the second microphone was
wired backward; it was “Out of Phase” and needed to be rewired or a “Phasor”
that reverses the wires be placed into the line to get it back “In Phase”.
This process would continue until all
the Microphones were deemed “In Phase”.
In a 3 pin XLR connector, pin 1
is ALWAYS ground, pin 2 is red and pin 3 is black. Pin 2 and Pin 3 may vary depending on the
manufacturer. Just make sure they are
all wired the same. - (In the resistance
color code, Red is always 2.)
In the 50’s I worked as a
sound engineer for Educational Radio and TV.
When running audio for baseball
games, the crowd sounds were so loud that we could barely hear the announcer
sitting in the announce booth speaking into a DESK mic.
My first attempt to correct the
problem, was to put a smaller lavalier microphone on a coat hanger around his
neck sticking up about an inch from his mouth.
Changing
the proximity of the microphone
allowed me to turn his volume down which essentially turned down the volume on
the crowd noise. My second attempt was to place a second microphone on the same coat
hanger facing the opposite direction toward the crowd. It was wired “Out of Phase” then reintroduced or
mixed back into the sound board with just enough audio to “phase out” the “Crowd Noise”.
Today we see announcers with headsets with microphones that almost touch their
lips that have integrated voice canceling phasing technology built in.
Far out FUN note: in the early
days of stereo RECORDS; two
microphones were set up to present separate left and right instrumental
channels and the vocalist would be placed equally on both the left and right
channel. With that said, an audio
engineer could take the phono preamp on the record player and reverse the
wiring on the second channel, which when re-introduced into the sound board,
would “Phase
out” the vocalist; GONE, leaving only
the music be heard.
Conclusion; the reason you
have a boat moving with two paddles synchronized “in phase” is so that you don’t
wind up going in circles. PHASING IS A BIG DEAL
Speakers
Just a word about speakers, we have
already talked about phasing, so next to discuss is their power and impedance
rating. Speakers should have an
impedance rating that matches the amplifier’s output impedance and have a power
rating that matches or exceeds the amplifier’s power rating for each channel. Often there is a desire to build your own
speaker system and money becomes a big part of the decision on what you purchase.
Example; You have a 100 watt stereo amplifier
with 8 ohm outputs; that’s 50 watts of power for each channel, so each speaker
must be able to handle 50 watts of power at 8 ohms to obtain maximum transfer
of power without frequency discrimination. You have found an awesome deal on some 8 ohm
speakers with a FULL
RANGE
of 20 to
20,000 cycle’s frequency response, but they are only 12 watts each.
So you figure you can connect
four of these speakers for EACH channel; 12 watts times 4 speakers would handle
48 watts; a pretty good match for 50 watts per channel. But the problem comes in, how do you wire the
speakers, Parallel or Series?
If you wire all 4 in
parallel;
the power is additive; you have 48w, but the impedance reduces; 8 ohms divided
by 4 speakers, (8/4=2), the load would be just 2 ohms which would burn up
the 8 ohm amplifier at maximum power.
If you wire all 4 in
series; the power is additive; you still have 48w.
BUT the impedance adds (8+8+8+8=32) you
would get 32 ohms and it would work, but you would not get the maximum transfer
of power, plus you would get frequency discrimination.
Solution; Wire TWO SETS of two speakers
in parallel (8 ohms divided by 2 speakers) that would give you a 4 ohm load for
each set of two speakers. Then wire the
TWO SETS of parallel
speakers in SERIES (4+4= 8 ohms) and now you get an 8 ohm load for each
channel. You now have a total of 8
speakers for both channels that can handle 96 watts of power.
Constant voltage
speaker systems are utilized when there are a number of speakers
in a distribution system that are needed to send sound to church nurseries,
overflow areas and hallways.
There are 25, 50, 70 and 100 volt system. I prefer the 70 volt system. Each speaker has a 70 volt transformer with
TAPS that can select the wattage the speaker uses. The higher input voltage helps with signal
loss on long runs as well as matches the impedance of the loaded speakers to
the output of the amplifier for maximum signal transfer and optimum frequency
response. Since all lines are connected
in parallel, the installation is made quite simple. If an amplifier does not have a 70 volt
output, an external
step-up transformer can be purchased
and connected to the 8 ohm output of an auxiliary amplifier. You might be surprised, but typically a
hallway speaker only uses ½ watt of power each.
A 50 watt amplifier can handle 100 speakers connected at ½ watt each, or 50 speakers at 1 watt each, or 2 speakers at 10 watts each (20), plus 5 at 2 watts each (10), plus 15 at 1 watt each (15) and
10
at ½ watt each (5). 20 + 10 + 15 + 5 = 50 watts. GET IT;
its mix and match.
Speaker
Placement
Stereo or Mono
speakers for
a church sanctuary? Depends on who you
ask, there are argument on both sides of the fence. The question becomes what is desired
and what is required.
Mono cost less and gives a naturally
better intelligibility
system. Stereo cost more, but most think
it gives a better experience in the worship service. But the wider the sanctuary, that experience
is lost unless you duplicate the stereo speaker on each side and that
introduces phasing problems throughout.
Low
Frequencies are not directional so they radiate out all sides of a speaker,
therefore smaller churches have one MONO base speaker
mounted in front, high and centered on staged area with the main Left and Right
stereo speakers located angled down to the opposite corners of the auditorium.
Half space loading occurs when a
speaker is close to or touches a hard surface causing it to boost the low
frequency energy. Placing a speaker
against the wall, the low frequencies are increased 6db and
when placed in a corner it can be as much as 12db gain; so these two placements
should be avoided for the flattest frequency response in the room.
Sound originating at the speaker and
arriving at the listener can be reduced or even cancelled out because of signals
bouncing off the walls and arriving out of phase at the
listener’s ear; likewise the signals can also be reinforced making specific
sounds louder.
Ideally the speakers are focused on the
congregation avoiding the walls; a laser can be placed on the center top of a
speaker to adjust the angle just as it hits the back row.
In large auditoriums, more rows of
speakers may be required and since sound travels faster in wires than in the
air, the extra rows of speaker will need a delay system to make sure the listener hears the
sound from the main and secondary speakers in phase, otherwise it can dampen
the intelligibility
of the sound. Some sound board mixers have a sub-group output delay just for
this purpose; if not, a separate “add-on” processor can be purchased.
Where balconies or extra rooms with a low
ceiling exist, additional speakers must be added above and below these
areas to prevent dead spaces and complaints.
Most will also need to be phased, delayed as well as have their own
independent amplifier feed, so that the level can be adjusted.
Sound
with a Twist
Donald Leslie invented a
device that would twist and flange “the sound of music”. He sold the rights to CBS who then sold it to the Hammond Organ Company.
The Hammond organ’s “Rotating Leslie” was a high
frequency horn and a base speaker that rotated in a full circle, controlled
with a motor at two different speeds to produce a vibrato much like a singer
does when they sustain a note for a period of time. This technique, if not overdone by the
vocalist, is just more pleasant to listen to.
The rotation made the sound
waves twist and arrive at the ear going in and out of phase much like a vibrato, also
mimicking the sound of a pipe organ. The
slow speed was known as choral and the fast speed tremolo. The Rotating Leslie could be purchased
separately as an “add on”. Today there
are electronic versions on the market that do the same thing.
It’s also fun to know that the Allen
Organ Company was the first ELECTRIC
ORGAN
Company to introduce an “air escape” sound when you pressed a key that
mimicked the momentary sound that is first produced on an actual pipe organ as
the large tubes filled with air. The
Allen electric organ sounded like a real pipe organ.
In the early days of
tape recording, on a Reel to Reel tape recorder, there was a distance of
about an inch between the record head and the
playback head. During a live recording, the audio engineer
could give a delayed echo to the original sound by injecting the sound picked
up by the playback head into the audio board.
Running the tape deck at a speed of 7 1/2 inches per second gave a slow
reverb and running it at 15 inches per second would give a faster reverb. Today
electronic reverbs are used not only on vocals, but instruments such as
guitars.
Impedance
Impedance is resistance to the flow of
electrons. For something to have maximum
signal transfer of power, the impedance is said to be “matched”. For instance, if an audio amplifier has a
speaker output of 8 ohms impedance then it is looking for a speaker that also
has an 8 ohm impedance, this allow for the maximum transfer of power to be obtained as well as prevent frequency
discrimination.
Mismatch: At maximum
power; loading a 4 ohm speaker on an amplifier with an 8 ohm output could burn
up the amplifier. It is said to be “Loaded Down”. However on the other hand, putting an 8 ohm
speaker on an amplifier with a 4 ohm output works just fine, it said to be “Bridging”
because the load of the impedance or resistance is higher than the
amplifier. In all cases the Load must
be matched or be higher than the audio source.
Audio lines come basically
in two configurations, the first is composed of ONE WIRE inside a flexible shield said to be “high
impedance” and used in economically low level devices such as microphones; the
second is TWO
WIRES that
are also shielded and said to be low impedance and used with low level devices
such as microphones as well as all high level devices.
Belden Cable was the first
to introduce audio cables that had aluminum “Z” foil folded around the wires
which further reduced unwanted induced signals. Multi Audio cable could
be purchased with eight color coded audio lines in once cable. The sound engineer would then make a 50-100
foot SNAKE
with a box of EIGHT connectors on
one end and EIGHT XLR plugs on
the other.
Loss & Noise; normally we have
no concern for loss or noise until the audio lines start exceeding 100 feet in
length. Audio is Amplitude Modulated and
so is noise as well as the common 60 cycle hum that
is produced by electrical devices. The
longer the cable run the more loss of audio you have and the more chance of Hum,
“Snap,
Crackle and Pop” due to electrical devices being turned
off and on and radiating into the cables. Sometimes pre-amps are
used to raise the audio source levels BEFORE it travels down
the cables.
This not only gives a hotter signal, but it pushes any external noise
generated down to where it is not noticed. It also allows the microphone lines to be split so
as to provide a feed to the live auditorium board as well as a separate feed to
recording devices which in the best sense, should have a separate audio board
for separate controlled results. Recording devices and Live speaker
presentations are just two different ducks.
The larger churches will have two audio
sound consoles.
Have you ever noticed a POP when a
projection screen is lowered or raised?
The switch radiates electrical energy when it “makes”
and “breaks”.
It needs a non-polarized capacitor
across the “End
of Travel” limit switches to absorb the arc.
Foldback
Foldback is a term used
for sound which is currently being processed by the audio board and is sending
it back into the studio or the stage area.
It is used as monitors on the stage for the performers, so they can hear
themselves singing or playing their instruments. The foldback speaker mix is different from
the House speaker mix and can give the audio operator headaches as it also
becomes part of the overall sound level in the auditorium or sanctuary. The sound engineer’s job is to make sure the
performers are HAPPY; so set them up
first before optimizing the main mix. Don’t be surprised that they want more “feed”
once the room is running at its optimum level.
Caveat; this can contribute to FEEDBACK.
Many performers have
their own Foldback systems and are a MUST for their individual
performance. Just LOVE THEM and make sure they are happy before you
start your house mix, because their foldback sound levels are additive to what
is heard in the auditorium.
Caveat;
an instrumentalist may hear their music just fine and NOT be heard at all in the overall mix. It is the Audio operator’s responsibility to WORK THE BOARD and bring them forward, it’s not SET AND FORGET.
Feedback
Audio Frequencies all travel at the same speed at the same
temperature. When traveling through Air
at 68 degrees, they travel at 1,125 feet per second, but the wavelength of the
frequencies vary. The length of a 1,000 cycle note is approximately 13 inches. The “A” string on an instrument is 440
cycles/Hertz and the wave length is
approximately 2 1/2 feet; 100 Hz
is 11.25 feet and 50 Hz is
a whopping 22 1/2 feet.
The reason this matters is because the
surfaces or materials the frequencies encounter either reflect or absorb the sound.
Many churches have the seats padded so that they will all reflect or absorb the
same whether they are “occupied” or not.
Example: The 100 Hz
bounces off a wall traveling right back along the same path bouncing off the
other wall. A 100 Hz note will become excited if it encounters something else
that is 11.25 feet or a multiple of that, reinforcing or adding to the sound. Notch filters can help greatly with this… (See Resonant Frequencies)
Automatic Feedback Suppression is accomplished
by “Frequency
Shifting”, “Adaptive Filtering”, “Automatic Notch Filtering” and “Delay”.
Most devices are invasive and noticeably
COLOR the sound except for the Auto Notch
technique which is the most popular method.
Suicide Speakers are those
located behind the stage presentation area limiting the amount of gain that can
be utilized on a microphone before feedback starts to occur. Church auditoriums are rectangular and many
have the stage area constructed in ONE CORNER forming a “V”; thus
reducing the reflection of sound off adjacent walls and complementing the sound
distribution like a megaphone.
Resonant
Frequencies
A Resonant Frequency is the
frequency of which a speaker or a room becomes sympathetic. As a teenager, I would sit in a room at
church with a bunch of other kids and I would hum at a certain frequency and
the room would become sympathetic and would join in and the sound
would reverberate
and you could not tell where the sound was coming from. I thought I was pretty funny until the
teacher made me sit up front.
Much though and engineering goes into the
design of auditoriums and theaters, with angled walls and ceiling panels that
reduce reflection and other panels to absorb sound, but rarely is a church
sanctuary designed for sound; it is designed for the aesthetic
appeal rather than acoustics.
Church Auditoriums have a resonant
frequency; most of the time two or three of them. When music is displayed, the room enhances
the sound at certain frequencies. For
the sound man this can be a nightmare of feedback. NOTE: A resonant frequency
of 400 cycles has a sympathetic vibration at 800 cycles even though its
reproduction sound level is greatly reduced and then again at 1600 cycles, etc.
So killing the 400 cycle demon is the best solution.
Solution: Have a licensed and insured system integrator
audio professional come out and “optimize” the auditorium. This is accomplished
with a 32 or 64 set of notch filters. White noise or Pink noise is injected into the sound system through any
input, filling the room with a noise level much higher than normal. A specially designed microphone is connected
to a spectrum
analyzer and it breaks down the
white noise into 32 or 64 discrete frequencies and displays their levels on a
display screen. Note;
today you can get a spectrum analyzer application for your cell
phone; some are free.
By adjust the filters you can bring
down the band of frequencies till they are all virtually at the same level,
this is said to be a “Flat Response”. Usually no more than eight filters would have
to be purchased to optimize the house. This compensates for the Audio Board, the
Speakers as well as the Room imperfections. Resonant frequencies would
be vastly minimalized and the room would not “BOOM”; feedback
tendencies would be greatly reduced making the Board operations much easier.
Speakers have their own resonant frequency. Try to select one that is below what the human
ear can detect which is 20 to 20,000 cycles. So if the speaker resonant frequency is 60
cycles, you have just bought a speaker that is sympathetic to the frequency of
all AC powered devices which operate on 60 cycles… The not so technical term is
called HUM….
Grooving
Performer’s body motions in a church
setting can be deemed EXCESSIVE AND SHOWY. Point
out to those that would complain, that performers feel their music and their
emotions are transmitted to their body and that’s a good thing. They are in “THEIR
ZONE”
praising God which is the ultimate end for which man was created. SELAH
A Praise and Worship Leader once told me
that in the Psalms, the word SELAH is often used by the chief musician, indicating
that when the foregoing passive is somber in thought, that you should in
reverence “PONDER IT”, but when it’s joyful you should just Cut Loose and Do Your Thing. It’s
in the Book
Psalm 30:12 –
Our ultimate end is to offer praise to God
12 To the end that
my glory may sing praise to You and
not be silent. O LORD
my God, I will give thanks to You forever.
(NKJV)
Blending
Vocals & Instruments
The Blend you desire on
vocals and instruments has a number of factors that enters into the
equation. As with the audio board you
either have to RAISE all the LOW level vocals or instruments to 0db or LOWER the HIGH
level devices to 0db.
The acoustical
drums would
probably rate as the highest level instrument, already being too loud when they
are still being unpacked. Don’t be
surprised if the drummer wants a mic on the “high hat” to add in those high
frequency sounds. The mic should be
positioned extremely close so that the input can be turned way down reducing
the other drum sounds.
The real solution is to put the
drums in a glass enclosure to reduce their sound and then mic them
independently. IF NOT you must
increase the levels of all the other instruments on stage to match the drum
set. Some considerate drummers put a
pillow inside the base drum, others stick a little rubber mastic on the symbols
and snare drums to help reduce some of the sound, while still others chunk the
real drums and get electronic drums which introduce an irritating
SMACK of the sticks on each device.
Noise Gates can be used on
drums with great success by reducing the “bleeding” between drum mics; making them
almost invisible to each other. – (See Noise Gates)
Best
Operating Practices
There is a total “LOUDNESS”
level
of the room that varies depending on the operator, so sometime it is advisable
to obtain a settable loudness meter so
that a deacon or elder does not MESS WITH YOU about how loud
it is; just point him to the meter. A
ship has only ONE CAPTAIN; find out who that captain is and that’s the one
you please, hopefully it’s a performer who has wisdom as well as knowledge.
Color coding should be used
on all microphones on the stage. Color
Electrical Tape can be purchased at Lowe’s or Home Depot in the Electrical
Department; place a strip of color on each microphone as well as a matching
strip on the audio board. It’s also a
good practice on foldback speakers as well.
In case of a
microphone failure; a spare color coded microphone, corded or wireless must be
located in plain view as a “Stand by Mic.” It’s the operator’s job to confirm its
operational status as well as educate those on stage of its physical location.
Wires should be dressed with PRIDE on all stands as well as along the floor
to prevent accidents. To avoid HUM, power cords should NOT be dressed parallel to audio lines although they
can be crossed without a problem.
Seasoned operators can be seen using masking tape across the bottom of the audio console writing what’s
on each channel. Anybody can come in
behind them and easily pick up the operation.
If your audio console does not have INPUT attenuators on
each channel, then masking tape is recommended to place arrows for where the
normal position of the fader should be. You can then easily return to your “default value”
setting whenever desired.
Microphones that are not in
use, should be left turned off so that if they are accidently bumped it will
not be heard on the sound system. Ease
the sound up and down in making corrections.
When only one microphone is in use the majority of the time, all the
others should be off. My personal
preference is to run the fader pots to the bottom so I can SNEAK them ON when I
need them, not slam them on; then everybody knows you were late.
When “hunting” a microphone don’t in
frustration slam your fader or pot to the top of their range and then start
turning on mics; you will blow out the audience when you find it. With the attenuators set at their “default value”;
then open the mics, find it, then adjust.
Baby Grand
Pianos can be a task to mic. I find a pair of Omni-directional microphones
placed 24 inches apart and about 10 inches above the strings and about 6 inches
behind the hammers does an excellent job.
It is sent and mixed as a mono signal to
the audio board. If you get too close to
the hammers you will hear the “attack” as the hammers hit the strings.
As a compromise, one mic on a piano can
also be utilized; since the low frequencies do not need as much help; place the
mic on the floor near the right leg pointing up at the high frequency strings
of the piano. Caveat;
you may hear the foot petals being depressed, which is very annoying.
There are Piano mic systems (PianoMic) that use a compression pole to suspend
two microphones above the strings that is designed to eliminate the ” Proximity”
effect, allowing all the strings to sound equal. Don’t get frustrated, at this writing there is
no single STANDARD for mic
placement on a piano.
Drums believe it or
not, do need a mic on instruments that are used such as bongo drums,
tambourines, cowbells, shakers, etc. The
audio operator’s job is easier if they are physically located away from the
main drum set, and for balance, the single mic is placed further from the
stronger instrument and closest to the weaker.
An Announce
mic is
occasionally used on stage and sometimes you are surprised when it comes into
use. As a best practice it would normally be turned off. A mic with a switch on it should be provided.
When it’s off, it adds nothing to the
overall sound and when it is switched on and the fader is set at the default
value, your job has already been done for you, the level is perfect and nobody
is embarrassed. “In line” switches that plug into the
mic are also available as an “add on”.
As a rule, Orchestras and large Choirs
utilize a pair of directional microphones generally placed 13 feet up and 13
feet out and placed 13 feet apart. In a
church settings that have no orchestras, the microphones are placed much closer
to the smaller choir.
NOISE Caveat; Live
microphones on stands transmit the sounds of walking on the floor as well as
hands touching the stand. Lavalier
microphones on lanyards, rustle on clothes when they move. When clothes move on
top of the microphone the sound becomes muffled. If a lapel is chosen for a microphone, it
should be placed on the side which the performer is most likely to be turned
to. Ideally the mic on the host that is
most preferred for minimum sound level changes, is located on a tie or in the
center of their apparel.
Interesting Note; in “on stage
performances”, four or more directional “SHOTGUN” mics are laid on the floor on foam rubber
at the stage edge. Some principal actors
will wear wireless microphone with almost invisible clear tubes in their hair
just above their forehead to provide “presence” to their presentation. Stage actors PROJECT their voice
anyway.
It is your responsibility to tell your
pastor, that his wireless mic must be considered HOT all the time, wherever he is,
including the restroom. Did you notice that the POTUS does not wear a
cordless mic?
If someone else can assist with the video
presentations and audio recordings, you can do your job much
more proficiently.
One Omni directional microphone suspended in the rear center of the
auditorium is great for adding in the overall sound produced as well as
congregational or audience participation.
Some orchestra leaders want the feed in an ear piece to know which
sections to get louder or softer.
Hard of hearing church members
can use wireless
receivers and ear pieces to enjoy
the sermon better. It is the audio
operator’s responsibility to check the levels being sent as well as the
batteries; there is no “Do Over” to provide an Exceptional
Worship Experience. Remember
low or weak battery voltage causes distortion in the ear piece.
Taped Recordings are already
limited by their band width, so ALC’s and peak controllers are
recommended; while you are busy on your job on the audio console, the
recordings are a SET AND FORGET operation.
Silent Servant
Trying to balance vocals with the other
instruments usually gets unsolicited HELP
from those who think they know how to do your job, but won’t. Be patient and LOVE
them anyway; “Speak
in Tongues” so they won’t know that
you are frustrated. A seasoned operator
knows that as soon as the sanctuary or auditorium fills up, their clothing
changes the acoustics, requiring many adjustments as the program proceeds. That means the sound board is NOT PLUG AND PLAY, it requires
continuously “Working
the Board” to get the blend that the PERFORMERS DESERVE.
In a studio or professional stage
environment
the audio engineer does this job every day and is super proficient. But in a Church environment, audio is run by
anybody who is willing to do it. Some
just “step
in” at the last minute and have “enough going” to turn it Off
& On. Those who commit to running the board for
long periods of time, give up one night during the week to practice, then give
up Adult classes on Sunday morning to practice and then operate the board for
the live service; some churches have two and three services. This person is a SILENT
SERVANT serving
in the background. It’s
quite a commitment.
Andrew was one of the twelve Disciples, the only thing
you know about him, was that he served God in the BACKGROUND, bringing people to meet Jesus. He brought his brother Peter and 3,000 more people got
saved. The
unnoticed Audio person makes a difference in lives.
John 6:8-9 – Andrew
brings the boy with two sardines to Jesus
8 Another of his
disciples, Andrew,
Simon Peter's brother, spoke up, 9 "Here is a boy with five small barley
loaves and two small fish, but how far will they go among so many?" You are an
Andrew; a Silent Servant
It’s the Holy Spirit’s job to plant the seed; the Audio and Praise and Worship
Team’s job is to water the seed, and it’s God’s job to harvest the seed. We are only responsible for watering.
Worship lets you
get in on what God is doing; we “water” with our talents.
NOBODY WILL NOTICE WHEN YOU DO YOUR JOB RIGHT.
But
just mess up one time and EVERYBODY will notice it.
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