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CHAPTER 3

LIVE SOUND SYSTEM FLOW CHART

SIGNAL FLOW

Audio signal flow is the path an audio signal takes from source to output.

In typical home stereo systems, the signal flow is usually short and simple,
with only a few components. However, in recording studios and performance
venues, the signal flow can often be quite complicated, with a large number of
components, each of which may cause the signal to fail to reach its desired
output. Knowing each component in the signal flow becomes increasingly
difficult and important as system size and complexity increases.

Ex. (1) the following example traces the signal flow of a vocalist performing
in a church.

The signal flow begins from the singer, microphone, microphone preamplifier,
EQ, and compressor. For this example, this signal then flows into a mixing
board, which allows the signal to be routed to various outputs. The mixing
board includes facilities for a main mix bus, which we will send to the house
sound system, a monitor mix bus, which we will use to create a monitor mix
for the singer, and an auxiliary mix bus, which we will use to create a second
mix to be sent to the lobby and nursery.

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Ex. (2)

The most important skill you can have as a live sound engineer is a solid
understanding of signal flow. Without it, there’s no music, and you’re not
much of an engineer at all, actually.

You need to know where the signal is coming from, where it needs to go and
how to get it there at all times. In order to do that, you need to know how all
the pieces of the PA system work together.

For the purposes of this article, we’re going to use a passive analog PA
system to run front of house and monitors, in order to demonstrate how
everything works in full detail.

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Mixing Console

The mixer is your home base. It’s where all of your inputs (mics, instruments,
FX, etc) are controlled, processed and routed to the appropriate outputs.
Every console is different, but they all serve the same function.

Most consoles have 8-32 channels, a few subgroups, stereo “main” outputs
and multiple “aux” outputs for stage monitors and outboard effects.

Generally, analog consoles are outfitted with semi-parametric EQs on every


channel, and outboard processors are used for dynamics and effects
processing (gates, comps, reverbs, delays, etc).

Note: For contrast, most modern digital consoles feature fully parametric EQ,
gates and compressors on every channel, in addition to multiple effects
processors.

After signal processing, all channels are eventually routed to the stereo main
outputs and sent to the PA system for the crowd to enjoy, but first, you need
to connect the console to the speakers.

Graphic Equalizers

Console Main Outputs > Graphic EQs

Graphic EQs (GEQs) are used to correct the frequency response of a speaker.

Connect the main outputs of your console to a GEQ and use it to “tune the
room” or correct problem frequencies caused by standing waves in the room.

Connect each of your aux outputs to a GEQ before sending signal to the stage
and use it to “ring out” the monitors or eliminate frequencies that are likely to

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cause feedback.

Note: Most digital desks include digital GEQs for all Main and Aux outputs.
Everything is done internally on the desk so there’s no need for outboard
gear.

Crossovers

Console Main Outputs > Graphic EQs > Crossover

Crossovers are devices that split your signal in two — low frequencies go to
the subs and everything else goes to the mains.

Plug the main outputs of your console into the GEQ, then into the inputs of
the crossover. Set the frequency at which you’d like to split the signals, and
then connect the “low” outputs to the power amps for the subs and the “high”
outputs to the power amps for the mains.

Some crossovers include 3-way splitting for low, mid and high speakers, but
these are for more advanced systems typically only found in venues where
they were professionally installed.

Note: Active speakers have built-in crossovers, so there’s no need for an


outboard unit.

Power Amps

Console Main Outputs > Graphic EQs > Crossover > Power Amps

Power amps have one job: they supply power to passive speakers. There’s
nothing fancy about them — the only control is volume output. Many
engineers choose to run their power amps at full blast, although if not

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properly gain staged it could lead to a low signal to noise ratio.

Power amps are vitally important to your PA system. Using the wrong power
amp could cause your speakers to blow, the amp to catch fire or both.

In order to properly pair speakers and power amps, you need to know
the following:

 Power Amp Impedance

 Power Amp Power

 Speaker Impedance

 Speaker Power

Ohms are a measurement of resistance and they tell us how resistant the
power amps and speakers are to electricity. The most important thing is that
the speakers and the power amps have the same ohm rating.

Power is measured in watts and sometimes referred to as program, continuous


or RMS power rating. You need to use an amp that is 50% more powerful
than your speakers.

For instance, if you’re using two 8 ohm, 1000 watt speakers, you’ll need two
8 ohm, 1500 watt power amps, or a single 8 ohm, 2-channel 3000 watt power
amp.

After connecting your crossover outputs to the power amps, finish the chain
by connecting the power amps to the speakers.

Note: Active speakers have built-in power amps, so there’s no need for

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outboard units.

Main Speakers

Console Main Outputs > Graphic EQs > Crossover > Power Amps >
Main Speakers

The speakers are the very end of your signal chain — the last stop before all
this electricity turns back into music.

Speakers come in two basic types: active and passive.

Active speakers are powered. There’s a power amp built into the unit and all
you need to do is plug the IEC cable into the wall to turn them on. These are
more common in smaller, portable systems.

To connect active speakers to your console, simply run an XLR or TRS cable
from the main outputs on the console to the inputs on the speakers.

Passive speakers are unpowered. They require you to buy and connect an
external power amp to turn them on. These are more common in larger
permanent installations.

To connect passive speakers to your console, run an XLR from their main
outputs of the console to the inputs on your power amps, and then connect the
outputs of the power amps to the inputs of the speakers using TS or Speakon
cables.

Remember, each speaker should be paired with a GEQ and an appropriately


matched power amp.

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Subwoofers

Console Main Outputs > Graphic EQs > Crossover > Power Amps >
Subwoofers

Many systems include multiple subwoofers for extra low end reinforcement.

If only two active subwoofers are used, you can simply route the main
outputs of your console to the inputs of the subwoofers. Then, use the “thru”
outputs to connect the subwoofers to your active “tops” using XLRs.

Active subwoofers include built-in “crossovers”, which are devices that split
your signal in two — low frequencies go to the subwoofers and everything
else goes to the mains.

If passive speakers are being used, you’ll have to purchase an outboard


crossover, which works the same way the ones on active subs do.

Occasionally, more than 2 subwoofers are required for a gig, in which case,
you’ll need to use a speaker management system. These usually include lots
of DSP (digital signal processors) like EQ, compression, limiting and stereo
imaging: but most importantly, they allow you to split a stereo signal to eight
or more outputs, which gives you the freedom to include multiple subs.

Monitors

Console Aux Outputs > Graphic EQs > Power Amps > Monitors

Last but not least, now that you have the PA system up and running, it’s time
to connect the stage monitors so the band can hear themselves.

Stage monitors are connected essentially the same way the mains are
connected, only using the aux outputs instead of the main outputs.

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Each stage monitor will need its own graphic EQ and aux send. Start at the
console, and connect the aux outputs to the GEQs using XLR or TRS cables.
Then, connect the GEQs to the power amps using XLRs or TRS cables, and
finally connect the power amps to the monitors using TS or Speakon cables.

Note: Active monitors have built-in power amps, so there’s no need for
outboard units.

Boot up Procedures

Now that you’ve got everything connected, you need to turn everything on.
Always start at the beginning of the signal chain the console. Then, simply
follow the chain down the line.

Boot up the GEQs next, followed by the crossover and any additional
processing. Finally, turn on the speakers/power amps last; otherwise you’ll
get a nasty pop in the system when you turn on whatever missing component
is behind the speakers.

Channel Strips

Analog consoles can be really intimidating. Some of them have 32 channels,


dozens of knobs and more LED lights than your average Christmas tree. The
important thing to remember is that if you can understand how one channel
strip works, you can understand how roughly 80% of the console works.

Aside from the center, or “master” section, the rest of the console operates
exactly the same as channel 1. If you’re used to mixing in a DAW you’ll be
pleased to find out the channel strips are laid out almost the same way on an
analog console, and while every console is a little bit different, they all work
essentially the same way.
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Preamp / Gain

It all starts with the input of the preamp. Whether you’re plugging mics
directly into the back of the board or using a “stage snake” to feed the signals
from the stage, the first step is amplifying the Mic using the gain knob on the
preamp.

So… while we’re on the subject, what exactly is gain? First of all, let’s talk
about what gain isn’t.

Gain ≠ Volume

If you want to increase the volume of a signal, you should reach for the
channel fader, not the gain knob.

Gain can be thought of as Mic sensitivity. It increases the intensity of what


the microphone can “hear”, and is used to dial in the appropriate intensity of
each instrument.

Too much gain and the signal can clip and distort. Too little gain and the
signal feel weak.

To set the gain for a channel, have the musician start playing their instrument,
and with the channel fader down, slowly turn up the gain knob until you hit
the console’s “sweet spot.”

Every desk is different, but this is usually where the signal turns from green to
yellow or orange. Just make sure you keep it out of the red!

After setting the levels on the preamp, slowly bring up the channel fader until
the instrument is at the appropriate volume.

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Note: The preamp is also home to the phantom power (+48v) switch, and the
phase flip switch.

Inserts

From here, the signal typically travels through the “insert” section, where you
can connect outboard processors (like gates and compressors) before running
through the EQ section. Many analog consoles have a pre/post button that
allows you to move the insert point before or after the EQ.

On most large format consoles, channels have a 1/4″“insert send” output to


send the signal to the processor and a 1/4”“insert return” input to return the
processed signal.

Some more compact consoles use a single 1/4” “y-cable” insert point, which
has three ends: one that plugs into the insert point on the back of the console,
one that plugs into the input of the processor (send) and one that plugs into
the output of the processor (return).

EQ

Up next is the EQ section. Most analog consoles are outfitted with a four
band semi-parametric EQ, which sounds intimidating but isn’t, really.

Typically, you’ll have a fixed low band and high band, with adjustable gain
and frequencies. Occasionally they’ll feature a button to switch between a
shelf and bell shape.

Next, you’ll have bell-shaped low-mid and high-mid bands, with fully
adjustable gain, frequency and Q (bandwidth) parameters.

Finally, there’s usually a fixed high-pass filter switch set to a specific slope
and frequency (usually around 80 Hz).

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Aux Sends

Next, your signal runs through the Aux Send section. Most analog consoles
have 4 or more aux sends, which can be used to duplicate signals and route
them to one of two places: stage monitors, or FX processors (like reverb or
delay).

To send a signal to an Aux Send, simply find the Aux Send section of the
desired channel, and turn up the send to the corresponding Aux.

Of course, you’ll have to make sure the Aux Outputs on the back of the board
are connected to the correct stage monitors and/or FX processors.

Remember, when using outboard FX processors, you can send a signal from
the Aux Sends, but if you actually want to hear the effect you need to return it
as well. Most consoles have dedicated Stereo Aux Returns specifically for
FX, although literally any channel input will do.

There’s also an “Aux Master” section in the center portion of the console,
which has a master volume control for each Aux Send, just in case you have
the balance just right but need a little more or less volume overall. Make sure
it’s turned up (typically to unity) or you won’t be sending any signal!

Pan Knob, Fader and Routing Buttons

Up next is the pan pot, the volume fader and the routing assignment
buttons.

Aside from panning stereo signals left and right, there’s not a lot of use for
panning in live sound. Sure, some engineers choose to tilt their toms a little
this way or move a keyboard over there to get it out of the way in a busy mix,
but overall most signals are panned in the center.

Panning makes sense when everyone is wearing headphones, but when


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everyone in the room has a different place in the stereo spectrum it starts to
cause more problems than it’s worth.

The next piece of the channel strip is the fader. Get to know how the fader
feels under your fingers. Become one with the fader.

From there, you need to tell the signal where to go. Analog consoles usually
have a few buttons (M, 1+2, 3+4, etc) directly next to each fader that allow
you to send the signal to the mains and/or one or more subgroups.

Often times, engineers use channel strips to process individual signals and use
subgroups to balance the mix. Think about it; what are you going to do, pull
up eight mics simultaneously during the drum solo?

By routing similar signals to subgroups (like drums, bass, guitars, vocals, etc),
you can balance levels by controlling multiple mics at once, leaving your
other hand free to mix effects, run lights, or fight off the droves of groupies
that are always fighting to hang out with the sound engineer.

Board Mix

While every engineer does things a little bit differently, here’s an example of
what a typical 16-channel board mix might look like, including common
microphone options, inserts, and aux sends.

For demonstration purposes, let’s assume you’re working with a 5-piece rock
band: drums, bass, 2 guitars, keyboards and 4 vocals.

There are 4 stage monitors: stage left, center stage, stage right and upstage
(next to the drummer). They’re connected to Auxes 1-4, respectively.

You also have 2 FX processors: one for reverb and one for delay. The reverb
unit is connected to Aux 5 and the delay unit is connected to Aux 6.

This template won’t work for every band, but use it as a starting point or a
reference guide when setting up and you should be prepared for most gigs.

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Now that you’ve got PA set up, it’s time for load-in, sound check and show
time!

SOUND SYSTEM CONFIGURATION

Sound system configuration is the setting up and arrangements of musical


equipment in the best way they can interact with each other to produce the
required result.

If you have been tasked with Configuring a sound system for a small band
that wishes to reach an audience of 300 to 500 people, there are various
elements, both strategic and technological, to consider. Audio tech people
have never had such a broad range of sound reinforcement equipment and
techniques at their disposal. The choices of technology and products available
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can be overwhelming, so let's talk about some of the options.

Speaker Selection

Your choice of speakers should be based on coverage requirements and the


size of the venue. There are some things to consider regarding the shape of
the room and how the speakers will interact with boundaries, such as the
walls, the ceiling, and the floor.

You want to get the best speakers your budget will allow. Start by figuring
out what you can afford and then determine what sounds best to you within
that price range. Always listen to the speakers before buying, as not all of
them are made equal. When choosing a speaker, you'll want to consult the
specification sheet, which should be readily available from most reputable
manufacturers. The most important specs to know are the frequency response,
SPL output, and dispersion. If you are using passive speakers, then you'll
need to know the wattage and impedance (ohms resistance).

A full-range speaker with a frequency response of 60 Hz to 18 kHz may be


fine for many genres of music, such as country, folk, or folk-rock, where the
kick drum and bass don't need additional punch. For rock, metal, pop, hip
hop, EDM, etc., you will want a subwoofer. A subwoofer extends the
frequency response down to 45 Hz or lower and will allow the full-range
speakers additional headroom and increased output.

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The sound pressure level of a speaker will determine how loud a speaker is at
a given distance (typically 1 meter). Most spec sheets will show Peak and
Continuous outputs. The peak is how loud the speaker is on loud transients,
while continuous output is the average loudness. This is a good indication of
how the speaker performs, dynamically. Sound pressure levels (SPL) will
attenuate by 6 dB with the doubling of the distance. If a speaker were capable
of 135 dB at 1m, then 2m would have an SPL of 129 dB. By doubling the
distance to 4m, the speaker would output 123 dB and so on. Another
consideration is that doubling up on the speakers will result in a +3 dB
increase. If a speaker has a peak output of 135, by adding another speaker the
output would increase to 138 dB.

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Dispersion is the way the sound is projected horizontally and vertically from
the speaker. This is incredibly useful for determining the placement of
speakers, as you can direct the sound away from boundaries, such as walls
and ceilings. For instance, a speaker with a 60-degree horizontal dispersion
might work well for a narrow room, while adding an additional speaker could
increase the dispersion to 120. The goal is to offer coverage to the entire
audience, while directing the sound off the walls. Many speakers are designed
to couple by utilizing a trapezoidal enclosure, versus a square or rectangular
enclosure. The trapezoidal design allows for easy placement of the speakers,
as they can be placed together in tight-knit group or array, which allows for
coupling with reduced interference between speakers.

The vertical dispersion will determine how high the full-range tops will need
to be to provide proper sound coverage for the audience. There are many
ways to configure a system, in terms of height and whether ground-stacking,
speaker stands, scaffolding, or trussing should be implemented as a way to
get the speaker high enough to offer extended coverage. The higher the
speaker, the farther the sound will travel. If it is too high, there will be a loss
of impact in the front. Not high enough may result in the sound being
uncomfortably loud for the front row.

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For our purposes, I suggest setting the tops at shoulder to head level, about 5
to 6 feet from the floor. If you are utilizing subwoofers, you might try ground-
stacking the tops on top of the subs. Many speakers offer pole mounts for use
with speaker stands. This is the simplest way to get proper height, especially
if you don't have multiple subs to create a ground stack. At the very least, you
want your high-frequency driver above the heads of the people in the
audience.

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Active versus Passive

There are pros and cons to both active and passive speaker designs. Active
speakers are the easiest to deploy with built-in amplifiers that are matched to
the speaker components (woofers, mid-range, and tweeters-typically
compression drivers). They also feature crossovers, which isolate and route
frequency ranges to each component, and built-in limiters for protecting the
drivers. A three-way active speaker will have two or more built-in crossovers,
which isolate the high, mid, and low frequencies. The advantage of active
speakers is the ease of setup and operation. They only require a line level
input and you won't have to use separate amplifiers to power them.

Passive speakers require amplification, speaker cables, and may require an


outboard crossover and other signal processing. Some passive speakers will
utilize an internal crossover network, which functions much like the active
speakers. Other speakers are designed to be bi-amped or tri-amped, which
can be a benefit, as this allows greater control over the speaker components,
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but also requires a separate amplifier for each component of the speaker. If
you decide to go with a passive speaker design, you'll need to look at the
specification sheet provided by the manufacturer to determine the correct
amplifier(s).

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The input range of a speaker is typically given in continuous, program, and
peak wattage measurements. You will most likely see the continuous output
and either program or peak. The general rule is a doubling of the continuous
results in program, while doubling the program will give the peak
performance. For instance, a speaker with a continuous input of 1400 watts
will offer a program of 2800 watts and a peak of 5600 watts. The larger the
amplifier, the more headroom will be available. Do you really need to match
5600 watts to this speaker? Most professionals will say no. A good formula is
to aim for 1.5 x the continuous input. A 1400 watt input x 1.5 is 2100 watts
and should be the bare minimum for this speaker. A safer bet is to match the
speaker to the program output of 2800 watts.

Another consideration is the impedance or ohms resistance for the speaker.


You will need to consult the amplifier specifications to determine how much
power an amplifier is able to produce at given impedance. Most
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manufacturers will boast the highest output of both channels at the lowest
resistance. When matching your amplifier to your speaker, it's important to
consider the ohms rating and wattage. For instance, an amplifier that is rated
at 4000 watts (2000 watts per channel) at 2 ohms will realistically deliver
1400 watts at 4 ohms and 850 watts at 8 ohms. We could certainly use this
amp with our 1400-watt speaker, but at its continuous output rate, it doesn't
leave much headroom. Without headroom, it is entirely possible we could
drive the amplifier into clipping and potentially damage the speakers.

Some amplifier manufacturers will indicate power draws as 1/8 power, 1/3
power, and full power. 1/8 power delivers the amplified signal below the
built-in clip limiters, while 1/3 power will have the clip limiters occasionally
flashing. Full power will have the limiters in constant activity. When engaging
the clip limiters, you are actually rounding off the audio signal to prevent
distortion, but the signal of the audio will be compromised. I prefer to run the
amplifiers at 1/8 power, which will give plenty of headroom without squaring
off the waveforms. You may also use a higher-rated amplifier at 1/8 power
without fear of damaging your speakers. Remember, the quickest way to
blow a speaker is to under power it.

Subwoofers also come in active and passive options with the same pros and
cons. There are many different designs that can offer pretty outstanding
results. The best bang-for-the-buck I recommend is a Yorkville LS801P. It is
a self-powered, single 18" tapped-horn design with a tremendous output,
rivaling or surpassing most double 18" designs.

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Depending on your setup and how many subs you have, you will have more
consistent results by placing all the subs together. Placing two subs together
will yield a 3 dB gain in SPL and they will couple without interference. A
stereo sub configuration may create null points in the room where certain
frequencies cancel each other out. Other tricks to maximize bass are to place
the subs near a wall or corner, as each of the boundaries will reinforce the
sound and help load the room. My favorite configuration is to center-cluster
four subs together (2 wide x 2 tall).

Speaker Processors

Regardless of whether you are using active speakers or passive speakers with
an amplifier, you should invest in a speaker processor. In my opinion, it is the
most important piece of gear and will save you time, money, and headaches.
A speaker processor combines a number of processors into a rack mount
signal processor. You will find gain, EQ, delay, crossovers, and limiting for
both input and output. A typical processor might have a stereo input and six
outputs. The inputs will feature a 6- to 8-band parametric EQ and/or a graphic
EQ, as well as a system delay. Each output on your processor will offer gain,
a 4-band parametric EQ, a delay for time-aligning speaker components in a
bi-amp or tri-amp application, or a full-range speaker and a subwoofer.

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You will also find digital crossovers featuring Bessel, Butterworth, and
Linkwitz-Riley band-pass filters. Each crossover offers a high-frequency and
low-frequency setting with selectable filter types. For tops, I typically set the
HPF to 96 Hz on a 24 dB per octave Linkwitz-Riley filter and the LPF to off.
For subs, I set the LPF to 96 Hz on a 24 dB per octave Link-Riley filter and
the HPF to 30 Hz on a 48 dB Butterworth filter. The major focus is the
crossover between the LPF of the sub and the HPF of the top. A 24-dB-per-
octave Link-Riley filter keeps the frequency response flat where both the
subwoofer and top are crossing over.

Note: A speaker processor delay is not a digital delay effect, as it is intended


to literally delay a signal by a set amount and does not offer a "wet/dry"
setting. If a manufacturer offers time delay settings for your speakers, you can
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use those to time align the tops and subs. The LS-801P has a 3 ms delay, so
adjusting the tops to match the inherent delay of the subs will provide a
coherent and phase-accurate wave front. If you don't have the specs, you can
invest in a measurement system like SMAART. If you have an iPhone or
iPad, you can purchase the Audio Tools app by Studio Six Digital, which can
help you, measure and calibrate your sound system.

Analog Mixers versus Digital Mixers

Analog mixers are the mainstay of any audio system, and range in price and
features. There are some diehard analog enthusiasts who will not move to a
digital mixing board, as they believe the analog components sound superior to
digital. If you are mixing a live band, you will want some additional signal
processors to shape the sound of each instrument. Most analog mixing
consoles will offer a built-in four band parametric EQ, which helps balance
the tonal sound and carves out space for each instrument in the mix. It is rare
to find analog consoles with built-in dynamics available on every channel.
Therefore, an all-analog setup will require several racks of gear to
accommodate the additional signal processing, such as compression and gates
for each channel.

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Another aspect to consider is the use of wedge monitors or stage monitors.
These are speakers that are typically on the floor and angled up toward the
performers, offering a dedicated mix, which allows the musicians to hear
themselves on stage. Feedback can become a problem, so the use of graphic
EQs will be needed to remove the frequencies that are feeding back. Add in
additional signal processors like multi-effects, delays, and reverbs and you
can see the analog setup may sound better, but will cost more money with the
additional signal processing, plus there are additional racks, cabling,
troubleshooting, and maintenance involved.

Digital mixers have made some considerable advances in recent years


regarding the quality of the sound, and pricing that is comparable to many
moderately priced analog consoles. Digital mixers offer the best solution for
any touring band, with a large channel count and each channel packing four-
band EQ, compression, and gating. Additionally, each output features graphic
EQ for ringing out monitors. Many mixers feature internal effects with up to
eight insert slots for use with internal sends. You can still use your favorite
outboard gear, but the digital platform reduces the amount of gear
substantially. Another benefit of the digital mixer is the wireless control
options. Many mixers offer iOS and Android control apps.

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If the FOH position is in a less than desirable place, the engineer can move
about the room to make informed adjustments based on the audience's
perspective. This also allows the engineer to tweak monitors from the stage,
while standing next to the musicians. Many mixer platforms will allow
multiple device setups in which band members may adjust their own mix in
real time, allowing the FOH engineer to focus on the main mix. Other features
now incorporated in the digital platform include spectral analysis and a real-
time analyzer (RTA) for making adjustments to monitors or to the entire mix.
However, I still recommend a dedicated speaker processor for tuning the
sound system.

Stage Snakes and Stage Boxes

A stage box or multi-channel snake is highly beneficial for reducing clutter on


the stage. Some larger stage setups use a splitter that splits the signal from all
the sound sources on stage between FOH and monitors. Most mid-level bands
typically don't have a dedicated monitor engineer, so the FOH engineer will
perform both main mixing and monitor duties. With an analog setup, you'd be
working with a 16- to 24-channel audio snake with a cable run of 100+ feet.
A drum kit may have 8 to 12 microphones set up to capture the sound, so a
dedicated sub-snake allows for shorter Mic-cable runs and a much cleaner
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stage setup.

Utilizing stage sub-snakes before going to the main stage snake will keep the
cable clutter on stage to a minimum. Many digital mixers offer digital stage
boxes that function like an analog stage snake, only instead of a 16- to 24-pair
multi-channel cable, the digital snake will use a single CAT5 cable to connect
to the mixer in the FOH position. This cuts down considerably on the weight
and setup time of the entire system.

Microphones

In order for musicians to be heard, microphones are used to capture vocals,


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guitar amplifiers, and drums. The mainstay of live music is the use of dynamic
microphones. There are many microphone manufacturers, but the favorite of
most clubs is still the Shure SM58 for vocals and SM57 for instruments. They
have proven their value over time by sounding good and being incredibly
rugged. They can literally take a beating and still function. If there is the
budget and desire for wireless microphones, I personally recommend the
Shure QLX-D series digital microphones. They offer clean, clear sound
without any artifacts, and with a simple setup.

In-Ear Monitors

Many bands prefer to forgo the use of stage monitors and opt for in-ear-
monitors (IEM). I've used the entry-level PSM300 Shure Personal Monitoring
System for years and have had excellent results. With a digital mixer, the
setup and operation is even simpler, resulting in very happy musicians who
are able to set their own monitor mix-and without excessive stage volume.

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In Conclusion to this topic

You can see there are many directions one can choose when setting up a
sound system for your band or event: analog mixers versus digital mixers;
passive speakers and subs versus active designs. Each has its pros and cons.
The most important thing is to use your ears when making decisions. Always
listen to speakers before purchasing and, if possible, demo speakers and subs
together, especially if you are using different brands. I can't recommend
enough the importance of having a dedicated speaker processor for any
system, regardless of size or budget.

TROUBLE SHOOTING

Troubleshooting is a systematic approach to problem solving. Also it means


to find source of problems and correct them.

Top 4 troubles you may face in live audio engineering:

Monitor Feedback

Acoustic feedback occurs when the amplified sound from any loudspeaker re-
enters the sound system through any open microphone and is amplified again
and again and again. We’ve all heard it – it’s that sustained, ringing tone,
varying from a low rumble to a piercing screech

The best way to troubleshoot feedback is to quickly figure out the channel
strip by muting your strips one after the other. Once you get the strip causing
the hum, turn down the gain a little bit and go through the frequencies to find
out where you’ve gone wrong.

Another way is to mute idle microphones. Mute inputs like guitar when they
are idle because you don’t know if they are leaning on the amp.

Don’t turn mics up on stage monitors after proper sound check. If you must
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do that let it be on request because then it’s got your full attention.

Buzzes

Buzzes can be cause by bad signal cables, when the instrumentalist unplugs
his/her input with the strip on, you will get a buzz. A dirty signal cable will
give you a constant buzz.

The solution is to trace out the cable and change it. In the case of unplugging
input with channel strip on, you already know! Keep idle strips muted. If
possible create mute groups.

No sound check

There are two major reasons why there may be no sound check before the
show begins

Late setup and when Talent comes late to a show.

In any of the two cases just be calm. Know that trying to be in a hurry may
cause more trouble. Just be as fast as you can but be calm. Even if you have
only one strip tested, one by one check all inputs starting from the vocal mics
and then the instruments. Gain them up on the FOH and then the monitors.

In the case of one single talent coming late, that’s easy! Just do same thing
buy putting him on the house then his monitor. Gain him gradually on the
monitor until he indicates he’s okay. Then you can take your time to do other
tweakings as the show goes on.

Monitor levels dangerously loud

When the monitor levels are too loud, simply turn down the master Aux or
Bus of that monitor and increase the input you want to have more on it. It’s

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that simple.

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