🎚️ Issue #6: The Glossary You Need to Survive (and Thrive) -- Part 2

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THIS WEEK’S SETLIST

What You’ll Learn Today

• The Pro's Vocabulary: Key FOH terms defined in plain English for quick understanding.

• Beyond the Basics: Advanced insights into essential sound concepts, made usable.

• Your Essential FOH Lexicon: Building the foundational language you need to mix with confidence and skill.

Just for today… (more coming next week)

16. Fast Fourier Transform (FFT)

17. Feedback

18. Feedback Stability Margin (FSM)

19. Gain / Preamp Gain / Trim / Sensitivity

20. Gain Structure

21. Gating / Gate

22. Haas Effect (Precedence Effect)

23. Headroom

24. High-Pass Filter (HPF) / Low Cut

25. Impulse Response (IR)

26. Inverse Square Law

27. Line Level

28. Magnitude-Squared Coherence Function (γ2(f))

29. Meters (Audio Meters)

30. Mids / Mid Frequencies / Midrange

👋 WELCOME

BUILT FOR VOLUNTEERS IN THE BOOTH

Every week in SundayMix, you’ll get one no-fluff FOH tactic to help you run tighter, cleaner, more confident worship mixes — even when service starts in five and chaos is in the air.

Let’s be real — no one’s probably showing you how to do this. You’re in the booth because you said yes… and now Sunday’s coming.

Let me know what you need most. Got a topic you’re struggling with behind the board? Hit reply and tell me. We read every response — and I’ll build future issues to solve it.

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SUNDAYMIX MAIN LESSON

The Ultimate FOH Glossary — Part 2

You ever been in a situation where everyone around you seems to speak a different language? Years ago, I dove into ballet. My instructor, bless her heart, would start barking out these French terms – plié, tendu, jeté – and I was utterly lost. Being dyslexic and learning disabled didn't exactly help matters. My survival tactic? Frantically watch the other dancers, trying to mirror their moves just to keep from looking like a complete fool.

Here’s the kicker for us sound techs: when you’re in that FOH booth, there’s often no one to model. You either know what that knob does, what that term means, or you don’t. And let’s be brutally honest, the grace extended to performers often doesn't reach the sound booth. The worship pastor might miss a note or forget lyrics, and it’s often overlooked. But if you unleash a feedback squeal during a quiet prayer or the mix sounds like mud, the consequences can be swift. You might just stop getting asked to serve, or if it's a paid gig, you might find yourself looking for a new one. It’s not always fair, but it’s often the reality of the job. So, the solution? Skill up.

My ballet journey didn't click until college when I finally took Ballet I and bought a textbook. Suddenly, seeing the terms defined, understanding the why behind the movements – it was like a lightbulb went on. Knowing the names of things, the actual language, is incredibly powerful. Amateurs often have their own slang for things, and that’s fine for casual conversation. But the pros? They operate with a precise, shared vocabulary. And that’s the language I want to equip you with, so you're not just surviving, but ready to step into any environment, any room, with confidence.

In today’s issue, I’m giving you a foundational glossary – the common terms I desperately wish someone had handed me when I was starting out. My advice? Print this. Turn it into flashcards. Drill them. The more you internalize the language of sound, the faster you’ll be able to diagnose problems, implement solutions, and ultimately, mix at the speed of thought instead of fumbling and thinking, "Wait, what was that again?"

Let's get to it.

16. Fast Fourier Transform (FFT)

• In Plain English and How to Use It: This is a fancy math process that audio analysis software (like Smaart or REW) uses to break down a complex sound into all the individual frequencies that make it up. It's like taking a smoothie and figuring out exactly how much strawberry, banana, and spinach went into it.
  

  • How to Use It: You won't do FFTs yourself, but the graphs you see in audio measurement software (showing which frequencies are loud or quiet in a room, or when analyzing phase) are often created using FFTs. It helps engineers "see" sound problems.

• Advanced but Still Usable: The Fast Fourier Transform (FFT) is an algorithm that efficiently computes the Discrete Fourier Transform (DFT), converting a time-domain signal (amplitude over time) into its frequency-domain representation (amplitude and phase per frequency). In audio, FFT analysis is fundamental to tools like Real-Time Analyzers (RTAs) and dual-channel FFT measurement systems (e.g., Smaart, REW). These tools use FFTs to display the spectral content of signals, measure frequency response, analyze phase relationships, identify room modes, and assess system coherence. The parameters of the FFT (e.g., FFT size, windowing function, averaging) affect the resolution and accuracy of the measurement in both frequency and time.

17. Feedback

• In Plain English and How to Use It: Feedback is that awful squealing, howling, or ringing sound that happens when a microphone picks up the sound from a loudspeaker it's connected to, and then that sound gets amplified and comes out the speaker again, and the mic picks it up again, creating a loop that gets louder and louder.
  

  • How to Use It (to prevent/stop it):

    • Keep mics away from speakers.

    • Turn down the volume of the mic or the speaker.

    • Make sure singers aren't pointing mics at speakers or cupping the mic head.

    • Use an EQ to carefully cut the specific frequency that is ringing (this is called "ringing out" the system).

• Advanced but Still Usable: Feedback, or acoustic loop oscillation, is a positive feedback condition in a sound reinforcement system. It occurs when the acoustic output from a loudspeaker is picked up by a microphone in the same system, re-amplified, and reproduced by the loudspeaker, with the loop gain exceeding unity and the phase aligning (see Barkhausen Stability Criterion). The frequency of the feedback is typically a resonant frequency of the system, influenced by microphone/speaker placement, their frequency responses, and room acoustics (room modes). Control strategies include:
  

  • Reducing gain (mic preamp, fader, amplifier).

  • Increasing distance between mics and speakers (D1​).

  • Decreasing distance between source and mic (Ds​).

  • Using directional microphones and speakers, aiming them strategically.

  • Applying narrow notch filters (parametric EQ) at feedback frequencies.

  • Improving room acoustics.

18. Feedback Stability Margin (FSM)

• In Plain English and How to Use It: When engineers calculate how loud a sound system can get before feedback (called Potential Acoustic Gain or PAG), they usually subtract a safety buffer, typically around 6 dB. This buffer is the Feedback Stability Margin. It's like leaving a little extra room in your suitcase just in case.
  

  • How to Use It: You don't directly adjust the FSM. It's a concept that informs good system design and setup, ensuring the system isn't constantly on the verge of feedback, making it more stable and forgiving to mix on.

Advanced but Still Usable: The Feedback Stability Margin (FSM) is a crucial safety factor, typically 6 dB, subtracted from the calculated Potential Acoustic Gain (PAG) of a sound reinforcement system. This margin accounts for real-world variations in acoustic conditions, microphone positioning, and system response that might otherwise lead to feedback if the system were operated at its absolute theoretical maximum gain. Maintaining an FSM ensures that the system operates below the threshold of instability, providing a more robust and reliable performance with less susceptibility to unexpected feedback.

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19. Gain / Preamp Gain / Trim / Sensitivity

• In Plain English and How to Use It: This is the very first volume control a sound goes through when it enters the mixer. It's usually a knob at the top of each channel, often labeled "Gain," "Trim," or "Sensitivity." Its job is to take the weak signal from a microphone or instrument and boost it to a healthy, usable level for the rest of the mixer to work with. Setting this correctly is THE most important first step for good sound.
  

  • How to Use It: During soundcheck, while a musician or singer is at their loudest, adjust this knob so the channel's meter averages around -18 to -12 dBFS (green/yellow) and never hits red (clipping). A good gain setting means you can keep your faders around "0" (unity) for better control. Too little gain makes things noisy when you turn them up later; too much gain causes distortion and feedback.

• Advanced but Still Usable: Gain, in the context of a mixer input channel, refers to the amplification applied by the preamplifier (preamp) stage. This control (often labeled "Gain," "Trim," or "Sensitivity") adjusts the amount by which the initial weak signal (e.g., from a microphone) is boosted to an optimal internal operating level, often referred to as line level, before further processing like EQ and fader adjustments. Proper gain structure involves setting the preamp gain to achieve a strong signal-to-noise ratio and sufficient headroom, preventing both noise ingress (if gain is too low and later compensated for) and clipping/distortion (if gain is too high). This is foundational to a clean, dynamic, and stable mix.

20. Gain Structure

• In Plain English and How to Use It: This refers to how you set all the different volume controls (gain, faders, EQs that boost, compressors with makeup gain, master faders, etc.) throughout your sound system, from the microphone all the way to the speakers. The goal is to make sure the sound is amplified cleanly and consistently at each step, without adding noise or distortion.
  

  • How to Use It:

    1. Start by setting your preamp gain correctly for each input (healthy level on meters, no red!).

    2. Keep channel faders around "0" (unity) for good mixing control.

    3. Make sure any boosts from EQs or makeup gain on compressors aren't causing clipping further down the line.

    4. Ensure your main output faders are also at a good level, not too low or too high. Good gain structure makes your mix sound cleaner, gives you more headroom (room for loud parts), and reduces feedback.

• Advanced but Still Usable: Gain structure refers to the systematic management of signal levels at each amplification or attenuation stage within an audio signal path, from input transducer (e.g., microphone) to output transducer (e.g., loudspeaker). The objective is to maximize the signal-to-noise ratio (SNR) and dynamic range while maintaining adequate headroom to prevent clipping at any stage. This involves optimizing preamp gain, ensuring EQs and dynamics processors are operating at their ideal levels, using faders effectively around their unity gain positions, and properly setting levels for buses, master outputs, and amplifiers. Poor gain structure can lead to noise, distortion, reduced dynamic range, and a higher risk of feedback.

21. Gating / Gate

• In Plain English and How to Use It: A gate is like an automatic mute button. You set a volume level (called a threshold), and if the sound going into that channel is quieter than that level, the gate "closes" and mutes the sound. When the sound gets louder than the threshold, the gate "opens" and lets the sound through.
  

  • How to Use It: Gates are often used on drums to cut out the sound of other drums bleeding into a specific drum mic when that drum isn't being hit (e.g., stop the cymbals from bleeding into the tom mics). Use them carefully, as a badly set gate can cut off quiet wanted sounds or open/close unnaturally.

• Advanced but Still Usable: A noise gate is a dynamics processor that silences or significantly attenuates an audio signal when its level falls below a user-defined Threshold. When the signal exceeds the threshold, the gate "opens," allowing the signal to pass. Key parameters include:
  

  • Threshold: The level at which the gate opens/closes.

  • Attack: How quickly the gate opens once the threshold is exceeded.

  • Hold: How long the gate stays open after the signal falls below the threshold before starting to close.

  • Release/Decay: How quickly the gate closes once the signal is below the threshold (and hold time has passed).

  • Range/Floor: The amount of attenuation applied when the gate is closed. Gates are used to reduce background noise, eliminate bleed between microphones (especially on drums), or create special dynamic effects. Incorrect settings can result in "chattering" or unnaturally cutting off the decay of sounds.

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22. Haas Effect (Precedence Effect)

• In Plain English and How to Use It: This is a cool hearing trick: if the same sound comes from two different places (like two speakers), your brain will mostly notice the sound that arrives first, even if the second sound is a little louder and arrives just a tiny bit later (within about 5-35 milliseconds). Your brain uses this to tell where sounds are coming from.
  

  • How to Use It: FOH engineers use this to make sound seem like it's coming from the stage, even if some of it is coming from delay speakers further back in a large room. By making the sound from the main stage speakers arrive just a few milliseconds before the sound from the delay speakers, the audience will "hear" the sound originating from the stage, creating a more natural experience.

• Advanced but Still Usable: The Haas effect, or precedence effect, is a psychoacoustic phenomenon where, if a sound is followed by another sound separated by a sufficiently short time delay (typically below ~35-40 milliseconds), listeners perceive a single fused auditory event whose spatial location is dominated by the first-arriving sound (the "leading" sound). The later-arriving sound(s), even if slightly louder (up to a certain limit, often around 10 dB), will not be perceived as a distinct echo but rather as contributing to the loudness and fullness of the perceived sound from the direction of the leading source. This effect is fundamental in sound system design for localization, particularly when using delay speakers in large venues to maintain sonic imaging towards the stage.

23. Headroom

• In Plain English and How to Use It: Headroom is the safety zone between your normal sound levels and the loudest possible sound your system can handle before it distorts (clips). Think of it like the space between the top of your head and the car roof – you need some room so you don't bang your head on bumps!
  

  • How to Use It: By setting your preamp gains correctly (e.g., averaging -18 to -12 dBFS), you leave plenty of headroom. This means if a singer suddenly gets much louder or a drummer hits a cymbal extra hard, the sound won't distort because you have that safety zone. Not enough headroom leads to clipping and a squashed, unpleasant sound.

• Advanced but Still Usable: Headroom is the difference, in decibels, between the nominal operating level of an audio system and the maximum level that system can pass without unacceptable distortion (clipping). Adequate headroom (typically 15-20 dB in professional systems) is essential to accommodate unexpected transient peaks in the signal without causing overload. It provides a safety margin, allowing for dynamic variations in the audio material. In digital systems, headroom is the range between the nominal level (e.g., -18 dBFS or -20 dBFS, aligned with 0 VU in analog) and 0 dBFS (the point of digital clipping).

24. High-Pass Filter (HPF) / Low Cut

• In Plain English and How to Use It: This is one of your best friends on the mixing board! An HPF (or "Low Cut") is a special EQ that cuts out all the really low, rumbly sounds from a channel, while letting all the higher sounds pass through.
  

  • How to Use It: Turn this ON for almost every channel – vocals, guitars, keyboards, snare, cymbals, etc. The only things you usually leave it OFF for are kick drum, bass guitar, and maybe some very low synths or tracks that need that deep bass. It cleans up muddiness from things like stage vibrations, mic handling noise, and breath "p"ops, making your whole mix clearer and tighter.

Advanced but Still Usable: A High-Pass Filter (HPF), also known as a Low-Cut Filter, attenuates frequencies below a specified cutoff frequency while allowing frequencies above it to pass largely unaffected. The "steepness" of this attenuation is determined by the filter's slope (e.g., 6 dB/octave, 12 dB/octave, 18 dB/octave, 24 dB/octave). HPFs are widely used to remove unwanted low-frequency content such as stage rumble, handling noise, plosives, wind noise, and low-frequency mud from sources that don't have significant information in those regions. This cleans up the mix, increases clarity, and conserves headroom in the overall system. The cutoff frequency should be set as high as possible without negatively affecting the desired tone of the source.

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25. Impulse Response (IR)

• In Plain English and How to Use It: An impulse response is like a sonic photograph or "fingerprint" of a space (like a room or concert hall) or an audio device (like a speaker or reverb unit). It's captured by sending a very short, sharp sound (an impulse, like a loud click or a starter pistol) into the space/device and recording how it responds over time – all the echoes and changes to the sound.
  

  • How to Use It: Engineers use IRs with special software (like Smaart or REW) to "see" how a room affects sound, helping them to EQ the PA system or place acoustic treatment. IRs are also used in digital reverb units and speaker cabinet simulators to realistically recreate the sound of real spaces and gear.

• Advanced but Still Usable: An Impulse Response (IR) is the output of a system (be it an acoustic space, electronic device, or digital processor) when presented with a brief input signal called an impulse (approximating a Dirac delta function). The IR characterizes the linear, time-invariant properties of the system. In acoustics, an IR recorded in a room reveals information about reflections, reverberation time (RT60​), and modal behavior. In signal processing, IRs are used to create convolution reverbs (by convolving a dry audio signal with the IR of a real space) and to model the behavior of EQs, speaker cabinets, and other audio hardware. Analysis of an IR in the time domain (e.g., ETC - Energy Time Curve) or frequency domain (via FFT) is a fundamental tool for system measurement and tuning.

26. Inverse Square Law

• In Plain English and How to Use It: This science rule says that sound gets much quieter very quickly as you move away from the sound source. Specifically, if you double your distance from a sound, it's not just half as loud, it's actually about four times quieter (which is a 6 dB drop in sound level).
  

  • How to Use It: This is super important for mic placement!

    • Vocals: A singer moving just a few inches further from their mic makes their voice significantly quieter, forcing you to turn up their gain (risking feedback and bleed). Encourage singers to stay a consistent 2-4 inches from their mic.

    • Feedback: The further a mic is from a speaker, the quieter that speaker's sound is at the mic, reducing feedback risk.

• Advanced but Still Usable: The Inverse Square Law states that for a point source radiating sound spherically in a free field (an environment without reflections), the sound intensity (power per unit area) is inversely proportional to the square of the distance from the source. This means that doubling the distance reduces the sound intensity to one-quarter, which corresponds to a Sound Pressure Level (SPL) decrease of approximately 6 dB (20log10​(1/2)≈−6.02 dB for pressure, 10log10​(1/4)≈−6.02 dB for intensity). This principle is critical for understanding microphone placement (distance to source affects direct-to-reverberant ratio and signal strength), loudspeaker coverage, and gain-before-feedback calculations (PAG/NAG). Deviations occur in non-free-field conditions or with non-point sources.

27. Line Level

• In Plain English and How to Use It: Line level is the standard strong signal strength that most audio equipment (like mixers, amplifiers, effects processors) uses to talk to each other after a weak signal (like from a microphone) has been boosted by a preamp. It's much stronger than a mic signal but usually weaker than the signal going to power the actual speakers.
  

  • How to Use It: You generally don't plug a microphone directly into a "line input" because the mic signal is too weak – it needs a preamp first. Keyboards, CD players, and outputs from other mixers are often at line level. Knowing this helps you plug things into the right jacks on your mixer or snake.

• Advanced but Still Usable: Line level refers to a standardized signal strength for analog audio signals transmitted between components like mixers, effects processors, and amplifiers (before the final power amplification stage). There are two common nominal line levels:
  

  • Consumer Line Level: -10 dBV (approximately 0.316 volts RMS), common for home audio gear.

  • Professional Line Level: +4 dBu (approximately 1.228 volts RMS), common for professional audio equipment. Microphone signals are much weaker (mic level) and require a preamplifier to be boosted to line level. Instrument level (from electric guitars/basses) is also typically weaker than line level and often requires a preamp or DI box. Connecting mismatched levels can result in distortion (if too hot) or a noisy signal (if too weak and excessively amplified later).

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28. Magnitude-Squared Coherence Function (γ2(f))

• In Plain English and How to Use It: This is a very advanced measurement that engineers use with special software (like Smaart) to check how similar two audio signals are at different frequencies. If two mics are supposed to be picking up the same sound and working together, this measurement should be close to "1" (perfectly similar). If it's low, it means the signals are different, which could indicate phase problems, too much bleed, or reflections messing things up.
  

  • How to Use It: As a volunteer, you won't directly use this. But it's one of the tools expert engineers use to diagnose complex phase issues or to check if different parts of a PA system (like mains and subs) are working together correctly.

• Advanced but Still Usable: The Magnitude-Squared Coherence (MSC) function, denoted as γ2(f), is a dual-channel FFT measurement that quantifies the linear relationship between two signals (e.g., an input signal to a system and its output, or signals from two microphones) as a function of frequency. It yields a value between 0 and 1 for each frequency:
  

  • γ2(f)=1: Indicates perfect linear correlation; the output signal is entirely determined by the input signal at that frequency (or the two mic signals are highly related).

  • γ2(f)=0: Indicates no linear correlation.

  • Values between 0 and 1 indicate partial correlation. Low coherence can be caused by noise, distortion, time delays not accounted for in the measurement window, non-linearities in the system, or uncorrelated signals (e.g., excessive bleed from other sources). It's a critical diagnostic tool for assessing the quality of a measurement, identifying frequency ranges where data might be unreliable, and troubleshooting issues like reflections or misalignment in PA systems.

29. Meters (Audio Meters)

• In Plain English and How to Use It: Meters on your mixing console are the little lights or digital bars that show you how loud the audio signal is for each channel and for your main mix. They usually show green for good levels, yellow as a warning that it's getting loud, and red for too loud (clipping/distortion!).
  

  • How to Use It:

    • Input Channels: When setting preamp gain, aim for average levels in the green/low-yellow (e.g., -18 to -12 dBFS on a digital meter, or around 0 on a VU meter if analog). Peaks should stay out of the red.

    • Main Output: Keep your main mix levels healthy, also staying out of the red.

    • Important: Meters only show loudness, not if the mix sounds good (clear, balanced, no feedback). Trust your ears first, but use meters as a helpful guide for gain structure and avoiding distortion.

• Advanced but Still Usable: Audio meters provide a visual representation of signal levels. Common types include:
  

  • VU (Volume Unit) Meters: Primarily show average signal levels, responding relatively slowly. 0 VU is a nominal reference level (e.g., +4 dBu). Good for judging perceived loudness.

  • PPM (Peak Programme Meters): Respond quickly to show peak signal levels, crucial for avoiding transient clipping.

  • dBFS (Decibels Full Scale) Meters: Used in digital systems, where 0 dBFS is the maximum level. These often show instantaneous peak levels. Meters are essential for proper gain staging, ensuring signals are above the noise floor but below clipping points. However, they do not measure subjective audio quality attributes like clarity, tonal balance, or spatial imaging. Critical listening remains paramount, with meters serving as objective indicators of signal integrity.

30. Mids / Mid Frequencies / Midrange

• In Plain English and How to Use It: "Mids" refers to the middle range of sounds that humans can hear – not the very deep bass or the very high shimmery treble. This is where a lot of the main information in voices and many instruments lives. Our ears are very sensitive to these sounds.
  

  • How to Use It: EQing the mids is crucial.

    • Too much in the "low-mids" (around 200-500 Hz) can make things sound muddy, boxy, or like they're in a barrel. Cutting a bit here often cleans things up.

    • Too much in the "upper-mids" (around 2 kHz - 5 kHz) can sound harsh, tinny, or make your ears tired.

    • Not enough mids can make things sound hollow or distant. Small adjustments in the midrange can have a big impact on clarity.

• Advanced but Still Usable: The midrange frequencies, typically considered to span from approximately 250 Hz to 4 kHz, are critical for the intelligibility of speech and the characteristic timbres of most musical instruments. The human ear is most sensitive in the upper part of this range (around 1-4 kHz).
  

  • Low Mids (approx. 250 Hz - 500 Hz): Contribute to warmth and fullness, but excess can cause "muddiness" or "boxiness."

  • Center Mids (approx. 500 Hz - 2 kHz): Define the body and presence of many instruments; can sound "honky" or "nasal" if problematic.

  • Upper Mids (approx. 2 kHz - 4 kHz): Crucial for intelligibility, attack, and perceived loudness; excess can lead to "harshness" or "sibilance." Careful EQ sculpting in the midrange is essential for achieving clarity, separation, and a balanced mix.

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SUNDAYMIX ACTION STEPS

3 THINGS TO IMPLEMENT THIS NEWSLETTER TODAY

• Build Your Feedback-Fighting & EQ Vocabulary.

  • Action: From this week's list (terms 16-30), let's laser-focus on these 5 critical terms for your situation: Feedback (#17), Gain/Preamp Gain (#19), Headroom (#23), High-Pass Filter (HPF) (#24), and Mids/Mid Frequencies (#30). Create simple flashcards (a small notebook works great too!). On one side, write the term. On the other, jot down in your own words:

    • A super simple "what it means."

    • One key way it helps you "stop squeals (feedback)" or "make voices/instruments clearer (EQ)." Carry these with you or review them for 10 minutes before your next soundcheck.

  • Why: These specific terms are your foundational toolkit for tackling those frustrating feedback squeals and muddy sound. Understanding them clearly and quickly means you can diagnose and solve problems much faster, especially when things get hectic before or during a service. This isn't just theory; it's about practical control.

• Hear It, See It, Fix It - Console Connection for Your Church.

  • Action: Next time you're at your church's mixing board (even if it's just for a quiet practice session, not necessarily a full soundcheck):

    • A. Gain & Feedback Hunt: Pick one microphone (like the main speaking mic). With the fader down, slowly turn up its Gain/Preamp (#19) knob. Watch the Meters (#29). Keep going until you just barely hear Feedback (#17) start. Note that gain level. Now, back the gain off a bit (maybe 1/8th of a turn) – this creates some Headroom (#23) and is a safer starting point.

    • B. HPF for Clarity: On that same mic, find the High-Pass Filter (HPF) or Low Cut (#24) button/knob. Turn it on. Listen carefully to how it cleans up any low-end rumble or "boominess" without making the voice thin. Most smaller church PAs will thank you for this!

    • C. Mid-Range Mission: If that mic sometimes sounds "honky," "boxy," or feeds back in a particular tone, gently experiment with the Mids/Mid Frequencies (#30) EQ knob(s). Try a small cut (turn it to the left a bit). Does it make the voice clearer or reduce a problem feedback tone? Small changes here make a big difference.

  • Why: Connecting these terms to the actual knobs on your board and hearing the immediate results in your room is the absolute best way to learn. This isn't just about definitions; it's about building practical skills to directly address the EQ and feedback challenges you're facing.

• Unlock a Key Secret to Less Feedback: The "Inverse Square Law" (#26).

  • Action: From the list, the Inverse Square Law (#26) might not sound like an EQ or feedback term, but it's a game-changer. This week, spend just 10-15 minutes looking this up. Search YouTube or Google for "Inverse Square Law live sound easy explanation" or "microphone distance and feedback." Find a simple explanation.

  • Why: Understanding this one simple physics principle will give you a massive "Aha!" moment about why getting microphones closer to the person speaking/singing is SO important. It directly impacts your Gain Structure (#20), reduces the chance of Feedback (#17), and often means you'll need less drastic EQ. It's a foundational piece that makes everything else easier for a volunteer in a smaller church setting.

Until next time,

Church sound that slaps. Built for the volunteers in the booth, not the guys in suits.