Audio through a sound system begins and ends as a series of vibrations.  Vocal cords or instruments create a vibration, and our ears perceive those vibrations and sends signals to our brain for interpretation.  The vibrations take the form of multiple sinusoidal waves or sine waves, depending on the frequency of the tone.  Audio can be analogous to water. Think of sound waves like ripples on a pond, spreading out from the source.  Sound requires some type of medium to pass through, typically air, although it can pass through solid material, liquid, and other gases, but it cannot pass through a vacuum, so it is true that ‘In Space, No One Can Hear You Scream’.  Temperature and humidity can also have an effect on how sound travels through the air.

Cupping your hands by your mouth, like using a megaphone, encourages your voice to veer in one direction so it can carry farther, a sound system is an electronic reproduction of this.

The human hearing range is limited between 20 to 20,000 Hz, at best.  There is quite a variation in the hearing range between individuals. The average young person can hear up to 18,000 Hz. Unfortunately, our ability to hear higher frequencies declines with age after 25.  By the age of 55, some men can’t hear above 5,000 Hz and some women can’t hear above 12,000 Hz.  Women tend to have better hearing than men at high frequencies. It is worth noting that the speaking voice of a man typically has a fundamental frequency between range of 85 to 180 Hz, and a woman 165 to 255 Hz.

Humans have binaural hearing, augmented by the shape of our ears, and how they are designed to capture sound, which allows us to determine the direction from which higher-pitched sound originates.  This was useful for hunting, or to avoid being hunted.

The ear’s sensitivity varies significantly with frequency. The human ear is most sensitive to frequencies in the range of 2,000-5,000 Hz. Our hearing has its peak sensitivity around 3,500-5,000 Hz. This frequency is associated with the resonance of the ear canal.  At frequencies above 10,000 Hz our hearing sensitivity declines.

Higher frequencies oscillate more rapidly (more times per second) and tend to be more directional due to the size and shape of the sound source. The higher the frequency, the shorter the wave. These higher frequencies can be easily absorbed by soft surfaces like fabrics and even humans.

Conversely, lower frequencies are longer waves, contain a lot of energy, can pass through and around most objects and are not as directional.  The lowest frequencies are more felt than heard with the ear, and for that reason, humans have a difficulty determining the direction from which low frequency sound emanates.

Midrange are the frequencies which fall in between Highs and Lows. According to the Laws of Physics: energy cannot be created nor destroyed, and when sound energy hits a surface, what is not reflected is turned into a tiny, imperceptible amount of heat.

Each frequency has a unique wavelength, which is what determines its pitch. More wavelengths completed per second means a higher pitch. Throw a big rock into our pond, you get a big, long wave, which probably will make it all the way to shore.  A smaller pebble makes a smaller wave which will dissipate with distance.

Signal Flow is the path sound takes through your system.

• Input devices, including wireless mics, CD Players, Sound Effect Generators, essentially anything that goes into the console
• The Console or Mixer
• Processing, including effects like reverb, gates, limiters, etc.
• Equalizer
• Crossover
• Amplifiers
• Speakers

 

The mixer has multiple channel strips which run vertically.  These channel strips allow you to modify the signal and ‘mix’ it with other signals to achieve the best overall output.

In the mixer, as each input is passed through the preamp, the Gain Knob, which is typically at the top of the Channel Strip, is used to adjust the level of the signal.

Through a cable, this small amount of electricity enters the mixer and goes through a preamplifier, which is used to amplify the small amount of electricity into something more robust and usable by audio equipment. The quality of the preamp is probably the most critical component in an audio mixer, because if the preamp isn’t doing a good job, there is no chance of making it better afterwards.

The microphone utilizes a diaphragm, known as the ‘Mic Element’ or ‘Mic Capsule’, to capture the vibrations of voice or instrument and converts those vibrations into a very small amount of electrical current, in the form of a sine wave.  The mic diaphragm needs to be very sensitive to function effectively, and abuse can degrade that functionality.  Tapping on the head of a microphone sends a massive shockwave to the element and is not a good way to test a mic.  Dropping the Mic makes a great visual statement; however, it is terrible for the microphone.  NO ONE who has ever spent their own money to purchase a microphone would purposefully drop one.  Instead, one of the best ways to test a mic is to snap your fingers in front of it from the distance the performer will be.

A DI box (direct box, direct injection box, or direct input box) is a device which converts an unbalanced signal into balanced and provides the impedance bridge between high-impedance outputs and low-impedance inputs.

The Input Gain knob works in conjunction with the Channel Fader to control output. 

When starting a new mix, turn the Input Gain knob all the way counterclockwise to the lowest setting, -10, 0 (zero) or ∞ (infinity), depending on the console.

Many consoles have a button below the Gain Knob called the high-pass filter or HPF. The HPF makes the effort to cutting off frequencies below a certain point (a threshold, or corner frequency.) For example, the HPF may cut frequencies below 100 Hz (Hertz), which can be useful for vocals. I usually engage the HPF for vocal mics, and instruments that do not produce tones below the cutoff frequency.  Be sure to disengage, typically the up position, for lower range musical instruments and other inputs such as prerecorded music or special effects when the low-end signal is desired.

Most consoles have a PAD (Passive Attenuation Device) button. This should be engaged for devices that produce a strong enough signal to not require the preamp (like Line Level signals.) If you have an input that is extremely loud even with the Gain Knob turned down, engage the PAD button.

Some microphones, such as Condenser Microphones require Phantom Power to function which is typically equal to 48V. Often, this is turned on with a single button per channel. However, some mixers have a ‘global’ Phantom Power switch for the whole board.  If you have forgotten to turn Phantom Power on in advance, it is a good idea to mute the channel before doing so.  If your Phantom Power is ‘Global’, it is a good idea to turn down, or mute the outputs before engaging.

Below the Input Gain knob and HPF are Parametric Equalizers, and they help compensate for signals that are less than perfect. We will talk about those more later. When starting a mix, it is a good practice to have these knobs straight up at “12:00” so they do not affect tone as you are trying to set volume levels.

Gain is a term used to refer to every place the audio technician can add or subtract from the signal to achieve the desired results.

Gain Structure is how we balance the gain controls to maximize signal output and minimize the noise floor inherent in all electronic devices.

It is a good practice to keep your Channel Fader at Unity, typically indicated as a 0 (zero) about ¾ of the way up the fader path, or with heavier hash marks.

There are practical reasons for this:

Keeping the fader at or near unity gives us the greatest resolution for changes. Note that 6 dB takes up quite a bit more space near Unity compared to down by -30 dB. This means we can make much smaller changes much easier.

If you are required to quickly pull a fader down to mute a mic for a cough, or to mitigate feedback, returning it to zero is visually easy to accomplish.

Keeping our faders at or near Unity also means that we are imparting or changing the signal as little as possible – which is usually a good thing.

With the Channel Fader at -inf, have the performer or a stand-in speak or sing at stage volume while you adjust the Input Gain knob at the top of the Channel Strip. Watch that channel’s meter to be sure you’re getting signal into the right channel and that the level is good (typically, kissing yellow.) Only after you believe you’ve got the signal level at a good place should the fader be brought to Unity, carefully. It is typically best practice to adjust your gain first, and then put your faders in place. If you find that the system is far too loud with a healthy signal level and the faders at Unity, it’s likely the loudspeaker system needs to be turned down. This is the best way to keep your signals undistorted, clean, and optimum.

Be mindful of the performer’s or musician’s dynamic range, the quietest to loudest they will be during a performance, and plan accordingly.  You will come to learn that many performers rehearse at a low volume and perform at a much higher level.

Once you have a volume you like, if necessary, use the Equalizers, or more accurately Semi-Parametric EQ to adjust the particular parts of a person’s voice you wish to enhance or minimize. Try fixing the sound at the source (e.g., the mic placement, how the person is singing, etc.). EQ is a band-aid and should be approached as a “well, I have no other options right now.”

Think of the Parametric Equalizers as individual gain controls for certain frequency ranges.

The top Parametric is for high frequencies (HF), the bottom for low frequencies (LF), and the middle range(s) are often ‘sweepable’, meaning you can choose the frequency to boost or cut.  Some consoles have two Sweepable Mids, High-Mids (HM) and Low-Mids (LM).  I find 125 Hz to often be a problematic vocal frequency, so I typically cut there first.  Speech intelligibility typically resides in the 3000 to 4000 Hz range, often expressed as 3 kHz to 4 kHz (Kilohertz) or 3K to 4K.  If someone sounds muffled, increasing the higher range to ‘brighten’ it up can help.    Interestingly, the frequency of a baby’s cry is between 1,000 Hz and 5,000 Hz, centered at 3,500 Hz, which is a very sensitive frequency range for the human ear. It seems that we are hard wired to hear the cry of our baby.

You may want the vocals to be ‘brighter’, ‘darker’, less nasal, etcetera.  Mastering this will require lots of practice. However, a simple slow sweep left, or right will typically make the sound better or worse, and you will know when you are headed in the correct direction.

After years of practice, most audio technicians learn to distinguish frequencies, which helps ‘dial’ in a mix faster.  There are also tools that can allow us to visually see the frequencies and how loud they are.

Next on the Channel Strip are Auxiliary Sends, or Aux Sends.  These are ways of also routing the signal to stage monitors or other outboard processing.  Some Aux Sends are ‘Pre-Fader’, which means the Fader at the bottom of the Channel Strip (-inf) has no effect on the signal.  Pre-Fader is usually best for stage monitors because you can turn down the house volume without affecting the stage monitors.  Post-Fader Aux Sends are typically used for external effects or other processing and are affected by the fader.  Some Aux Sends have a button to select Pre or Post fader.

If the console has Stereo output, there is likely a Pan Knob.  Keep this at straight up “12:00” unless you are utilizing a Stereo loudspeaker setup and need Left and Right Pan, or have another design reason to do so.

Mute is pretty self-explanatory, and very useful when you need to instantly silence a channel.  Hopefully the Mute Button on your console is illuminated, but if not, be very aware of its position. Not to be confused with a Solo or On button.

You may have a button labeled “PFL” or Pre-Fader Listen. Engaging this button sends only this channel to the headphone jack (or monitor bus) and meter, if applicable.  This is an amazing tool to isolate and specifically listen to a single input through headphones while all other inputs are still active. This does not affect the main outputs to the speakers, so use without worry.

Jumping to the output of the system for a moment and work back to the console.  At the other end of the system are the speakers, which also contain diaphragms, albeit much larger ones.  Speakers have an electrical coil, attached to a diaphragm, known as the Speaker Cone, and the coil resides within or around a magnet.  The diaphragm and magnet are mounted in a cage, known as the Basket, and this assembly makes up the guts of a Speaker.  The speaker is typically placed within an enclosure, or housing, which is also referred to as a Speaker, or Speaker Cabinet.

By sending electricity to the coil, it also becomes a magnet, which is either attracted to, or repulsed from the magnet.  By sending the electricity as a sine wave (from our console), the diaphragm, or cone, pushes out and pulls back in at the same rate as the sine wave, creating a vibration of the air in front of, and behind the cone.  It literally pushes and pulls the air to recreate the vibration that matches the sine wave sent to it.  Whatever frequency sine wave you send, that is the frequency of tone the speaker will emit.

By varying the speed and frequency of the sine wave, you recreate the sound that was input into the system.

It is possible for a single speaker cone to create all frequencies of audible sound; however, it is not the most efficient way to do so.  The smaller the diaphragm, the faster it can move back and forth, so small diaphragm speakers (one variety are called tweeters, another horns) are most efficient at creating high-frequency tones.  In the case of ‘horns’ the driver, or diaphragm is placed at the small end of a specially shaped cone, similar to a megaphone, which increases the output and aims it in the desired direction.  The larger the diaphragm, the more air you can physically move, and these speakers, known as woofers, do better with mid and lower range sound.  The larger the speaker, the more control over the direction of lower frequencies we’ve got.

Some systems utilize a specialized large woofer for very low frequencies, known as subwoofers or subs.  Subwoofers are designed to allow the speaker diaphragm, or cone, to move a greater distance within the speaker Basket, to produce the low range tones we call bass.  Humans ‘feel’ low frequencies and struggle to discern the direction from which very low frequency sounds emanate, so placement of the Subwoofer is not dictated by the audience’s ears, rather the ability to transmit the vibrations most effectively and with the fewest problems.

One way to think about this would be a cinema sound system, or a home theater.  You can discern sound coming from front left, right, center, as well as back left and right, however, you feel the bass, so placement is not as critical.

The speakers of an Audio Systems are typically designed as 2-Way High/Low, 3-Way High, Mid, Low, and sometimes 4-way, depending on the desired response and/or “volume”, known as Sound Pressure Level, or SPL.  The more SPL or distance you desire at a particular range of frequencies, the more speakers you need to dedicate to those frequency ranges.

Trying to achieve more “volume” or SPL (Sound Pressure Level) out of a speaker than it can handle will cause distortion, this is where the diaphragm, or cone, are being asked to travel beyond their design capabilities.  This will first cause distortion, but eventually permanent damage.  Remember, the cone travels in and out of the basket thousands of times per second, which wears them down over time.  One extension beyond tolerance could cause damage, thousands could rip the cone apart.

The Speakers are driven by Amplifiers, and each speaker, or pair of like-speakers requires an amplifier to drive them.  A 2-Way system would require an amp to drive the highs, and an amp to drive the lows.  If the system is Stereo, separated Left and Right, the amplifiers need to also have the two sides fed separately.  Powered Speakers have the Amplifier built into the enclosure, but often installed speakers are passive where the amp is mounted in a different location.

The signal that comes from the console going to the speakers contains the entire range of frequencies, so a device called a Crossover is used to separate the highs from the lows before the amplifiers.  In a 2-Way system, the crossover allows you to select the point where signal passes to the amplifier for the high frequency speakers, and what passes to the low frequency speakers.  In a 3-Way system, there are two crossover points, selecting what passes to the highs, the lows, and what is in between goes to the mids.  A Stereo Crossover, or two separate Crossovers, are required to sends the signal from the console left and right outputs to the appropriate amplifiers.

Prior to the Crossover is where you would utilize a Compressor-Limiter, which automatically prevents unexpected overpowering signal level which could damage the speakers.

Upstream of that is the system equalizer, which is used to make corrections for the room acoustics and the loudspeakers. Think of each Band as a small group of frequencies, and each slider as a gain control for the center frequency of that group. The goal of system equalization is to have the same sonic experience at every seat in the venue.

Unless a room had received extensive acoustical treatment, there are likely a few frequencies that will cause you trouble when trying to achieve that ‘perfect mix’.

Almost every room has a particular frequency that is problematic.  Sound leaves your speakers as a series of waves, and each frequency forms a particular wavelength.  One particular length of wave will hit the back wall and reflect back towards the stage where it may hit another surface and reflect again, but this time in the same direction as the original signal, and perfectly match the new output, doubling the energy of that particular frequency.

One of the biggest issues you will have to contend with is the Larsen Effect, named for Danish scientist Søren Absalon Larsen who discovered it.  You probably know the Larsen Effect as the dreaded Feedback, and unlike in the movies, you should take great care to avoided it.

Audio feedback, AKA acoustic feedback, is when an input hears its own output.

A signal received by the microphone is amplified and sent out of the loudspeaker. The sound from the loudspeaker can then be received by the microphone, amplified further, and then sent out through the loudspeaker again, and again, creating an ever-intensifying loop.  This can manifest as a high-pitched squeal, screech, or low rumble.  If feedback is not quickly mitigated, it can overload and damage the system.

Cupping the microphone is not a good way to stop feedback, in fact, it can make it worse.

The room equalizer is used to try and reduce the problematic frequencies, and to enhance any frequencies the room ‘absorbs’.  The best way to ‘EQ’ a room is to use a Pink Noise generator which sends an equal amount of signal at every octave through the speakers, this sounds a lot like static. With the help of a sophisticated program using a process called a dual-channel transfer function, we can see the difference between what is leaving the console and what is coming out of the speakers. But that is another day altogether.

In a pinch, using a Real Time Analyzer or RTA, along with a reference microphone, you can get a visual representation of the end result in the room. It will not tell you WHY certain frequencies it sounds like it does, but at least it’s a little more information than none. Frequencies that are higher than the average can be reduced with the corresponding band on an Equalizer, and frequencies that are below average can be increased in the same way.  This is a noisy process, best done when the room is empty.  Once you have EQ’d with the pink noise, it is a good idea to play music through the system from a well mastered source and reevaluate what you previously cut or added.  I often put back a little bit of what I originally changed.  If you are able to leave the reference mic in place during tech rehearsal, it will also assist with fine tuning the room, as well as individual microphones.

When powering up your system, start from the beginning of the signal flow, any powered input devices, then console, outboard equipment, with amplifiers being the last devices to be powered on.

Powering down is the reverse of that sequence, amps or powered speakers off first, then work your way back. 

The reason we follow this power-up, power-down procedure is because sometimes when electronics power up and always when they power down, electronic capacitors, or caps, discharge electrical current.  If the amps are still on when you power down the board, you will likely get a loud pop out of the speakers, that can potentially cause significant damage.

Radio Frequency or RF output of the body pack or handheld transmitter emanates primarily from a relatively small antenna in all directions in radio waves, so the farther the receiver antenna is from the transmitter, the less signal there is to receive. Signal radiates from the shaft of the antenna, not the tip, so it is best if the antenna points straight up or down when possible.

Water is the enemy of RF signals, and humans are essentially large bags of water, so keep receiver antenna well above the audience.

The RF signal is line-of-sight, so be sure the antennas are located where they will receive maximum signal. Signal is received through the shaft of the antenna, not the tip, so it is best to keep receiver antennas perpendicular to the location of the transmitters.

Most wireless systems manufactured today use what’s called diversity, which means the transmitter is sending two signals, A and B, and the receiver seamlessly switches between whichever signal is strongest.  Individual receivers come with two small antennas, one for A, and one for B.  If that is what you are using, make sure the back of the receivers are facing the stage and the pair of antennas are adjusted to be 45º left and right.  Since the signal from the transmitter is also a wave, having your antennas 90º from each other affords the best chance that either signal A or B will be properly received.

A significant upgrade to having a bunch of individual antennas in the back of your rack, jokingly referred to as an ‘antenna farm’, is to have an antenna distribution system, which eliminates the need for all the little antennas, and allows a larger and better antenna to be mounted in a more optimal location for reception. A high gain antenna on receiver end is superior to a more powerful antenna on transmitter side which might exceed allowable output. If more gain is required, use an active antenna booster.

Most wireless microphones are also Frequency Agile, which means that if you are experiencing another competing transmission, from a TV station, or emergency response group like police of fire departments, you can reassign the frequency to eliminate the issue.

The assigned frequencies need to be properly spaced to ensure one transmitter does not interfere with another transmitter, which in known as intermodulation.  If you have ever had two performers whose wireless mics worked perfectly when on opposite sides of the stage, but one dropped out when they came close together, that is likely intermodulation. When two transmitting antennas are brought close together, they will produce additional transmission frequencies, and if one of those new frequencies happens to be that of the other transmitter, the result is often LOS, loss of signal from one or both transmitters.

The best way to avoid intermodulation is to utilize the frequency groups set up by the wireless manufacturers.  Most receivers offer multiple groups, with 20 to 30 available channels within each group.  If there is an Auto-Scan function, run that, and it will display available groups as well as number of available channels within the group.  Once you have selected a Channel within a specific Group for the scanning receiver, select from that Group with a different Channel for each additional receiver.  Should you need to re-freq a receiver, just stay within the same group, taking care not to duplicate the Channel from another unit.  Once you have all receivers set to a unique channel or frequency, synchronize, or sync each transmitter to a corresponding receiver.

Besides the Antenna Distribution system, better antennas, such as a directional antenna or a parabolic antenna can enhance the RF signal. If available a network switch interconnecting the receivers also makes your job setting up and operating much easier if the receivers have a network port, like an RJ45 (Ethernet).

Mic placement is also critical to good audio reinforcement, so consider the best placement for the microphone element.  Most wireless systems utilize a very small element, so over the ear, or concealing at the hairline work well for theatre.  Clipping a Lavaliere Microphone to clothing can be problematic due to clothing rustle or the performer turning their head away from the element.  Extremely energetic performances such as dancing, or a fight scene present issues with keeping the mic in the correct position, and additional logistic considerations must go into situations when you have performers sharing a mic and transmitter.