🎚️ Issue #5: The Glossary You Need to Survive (and Thrive)

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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)

• 3:1 Rule (of Microphone Placement)

• Acoustic Treatment

• All-Pass Filter (APF)

• Barkhausen Stability Criterion

• Bleed (Signal Bleed / Spill)

• Clipping / Distortion

• Comb Filtering

• Compression / Compressor

• Crossover Frequency

• DAW (Digital Audio Workstation)

• dBFS (Decibels Full Scale)

• DI (Direct Input / Direct Box)

• Dynamic EQ

• EQ (Equalization / Equalizer)

• Fader

👋 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

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.

3:1 Rule (of Microphone Placement)

• In Plain English and How to Use It: When using multiple microphones, especially on things like a choir or across a stage, the 3:1 rule helps prevent a thin, hollow, or "phasey" sound. It means if one microphone is 1 foot away from its sound source (like a singer), the next closest microphone should be at least 3 feet away from that first microphone.

  • How to Use It: When miking a choir with several mics, or placing mics for multiple background vocalists near each other, try to keep each mic three times farther from its neighboring mic than it is from the singer it's aimed at. This helps each mic pick up its intended source clearly, reducing "bleed" and weird cancellations between mics.

• Advanced but Still Usable: The 3:1 rule is a practical guideline to minimize audible comb filtering and phase interference between microphones capturing multiple sources or a widespread source. By maintaining a distance ratio where the secondary microphone is at least three times farther from the primary sound source than the primary microphone is, the level of the primary source in the secondary microphone is attenuated by approximately 9-10 dB (due to inverse square law and distance). This lower level of bleed means that when the signals combine in the mix, the phase differences cause less destructive interference, resulting in a clearer and more focused sound with better separation between sources. While not a rigid law, it’s a very effective starting point for array miking.

2. Acoustic Treatment

• In Plain English and How to Use It: This means adding materials to a room (like foam panels, thick curtains, or diffusers) to control how sound bounces around. Too many reflections can make the room sound "echoey" or muddy, and can even make feedback worse.

  • How to Use It: If your church sounds overly live, boomy, or you struggle with persistent feedback despite good mic technique, discuss adding acoustic treatment with your leadership. Even simple additions can make a noticeable difference in clarity for the congregation and reduce problems at the mixing desk.

• Advanced but Still Usable: Acoustic treatment involves the strategic application of absorptive, diffusive, and reflective materials to modify the acoustic response of a space. The goal is to achieve a balanced reverberation time (RT60​) across the frequency spectrum, control problematic reflections that cause comb filtering or flutter echo, and minimize standing waves (room modes). Effective treatment improves speech intelligibility, musical clarity, and can increase Potential Acoustic Gain (PAG) by reducing the amount of diffuse sound energy returning to microphones. Materials are chosen based on their absorption coefficients or diffusion characteristics at specific problematic frequencies identified through measurement (e.g., using software like REW or Smaart) and critical listening.

3. All-Pass Filter (APF)

• In Plain English and How to Use It: Imagine a special EQ that doesn't change how loud or quiet any sound is, but it can slightly delay certain sounds (frequencies) to help them line up better with other sounds. This is used by advanced engineers to fix some tricky phase cancellation problems without messing up the tone.

  • How to Use It: This is typically a tool found on higher-end digital consoles. If you suspect phase issues between, for example, a DI and a mic on a bass amp, and simply flipping the polarity doesn't fully solve it, an experienced engineer might use an APF to subtly shift the timing of certain frequencies to get them working together. For most volunteers, knowing this exists is enough; its application requires significant expertise.

• Advanced but Still Usable: An All-Pass Filter (APF) is a signal processing filter that alters the phase relationship between frequencies in a signal without altering their amplitudes. It introduces a frequency-dependent phase shift. APFs are used to correct phase anomalies, often to improve coherence between summed sources (e.g., aligning a mic'd signal with a DI, or different drivers in a loudspeaker). By carefully adjusting the filter's center frequency and Q (which determines the steepness of the phase shift), engineers can mitigate specific phase cancellations or improve the alignment of acoustic sources. Unlike minimum-phase EQs, which inherently link amplitude and phase changes, APFs allow for targeted phase manipulation. However, one must be mindful of the introduced group delay, as excessive or poorly targeted phase shifts can negatively impact transient response.

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4. Barkhausen Stability Criterion

• In Plain English and How to Use It: This is the science rule that explains why feedback happens. It says if a sound from a speaker gets picked up by a mic, amplified, and comes out the speaker again loud enough AND timed just right (in phase) to reinforce itself, it will loop and squeal.

  • How to Use It: You don't directly "use" this rule at the console, but understanding it helps you know why turning down a mic, moving a mic away from a speaker, or using EQ to cut a specific ringing frequency stops feedback. You're breaking one of the conditions (loudness or timing) that the rule says causes feedback.

• Advanced but Still Usable: The Barkhausen stability criterion states that for a closed-loop system (like a sound system with a mic and speaker) to oscillate (feedback), two conditions must be met at a given frequency:

  1. The magnitude of the loop gain (∣Aβ∣) must be equal to or greater than unity (0 dB).

  2. The total phase shift around the loop must be an integer multiple of 360 degrees (i.e., the returning signal is in phase with the original). This means if a signal re-entering the microphone from the loudspeaker is as loud or louder than the original signal at the microphone, and its phase aligns, sustained oscillation will occur. FOH engineers combat feedback by reducing loop gain (e.g., lowering faders, preamp gain, or system volume) or by altering the phase response (less common directly, but EQ adjustments have phase implications) or frequency response (e.g., notching out resonant frequencies) to ensure these conditions aren't met.

5. Bleed (Signal Bleed / Spill)

• In Plain English and How to Use It: Bleed is when a microphone picks up sounds it's not supposed to. For example, your singer's microphone picking up the drums, or the acoustic guitar mic picking up the electric guitar amp. Too much bleed can make your mix sound messy and can make it harder to control individual instruments.

  • How to Use It:

    • Move mics closer to their intended sound source (e.g., singer closer to their mic).

    • Use directional mics (like cardioids) and aim them carefully.

    • Place instruments and amps further apart on stage if possible.

    • Use headphones during soundcheck to listen to each mic in solo – you'll hear the bleed clearly.

• Advanced but Still Usable: Bleed, or spill, is the leakage of sound from one source into a microphone intended for another source. While some bleed can add a sense of "liveness" or cohesion, excessive bleed reduces channel separation, complicates individual source processing (EQ, dynamics), can introduce comb filtering if the leaked sound combines out-of-phase with the direct sound, and lowers the gain-before-feedback threshold. Strategies to manage bleed include strategic mic placement (closer to source, utilizing nulls of polar patterns), choice of microphone polar pattern, physical separation of sources, use of acoustic barriers (gobos), and judicious use of gates or expanders.

6. Clipping / Distortion

• In Plain English and How to Use It: This is that horrible, crunchy, fuzzy sound you hear when an audio signal is too loud for the equipment to handle. It often happens if the preamp gain is set way too high, or if a fader is pushed way past its limit. You'll often see red lights on your mixer when this is happening.

  • How to Use It: Always watch your input meters when setting gain. If you see red, turn the preamp gain (trim) knob down until you only see green and occasionally yellow on the loudest parts. If you hear distortion, immediately check your gain structure – starting from the preamp.

• Advanced but Still Usable: Clipping occurs when an audio signal's amplitude exceeds the maximum voltage capacity of an electronic circuit (analog) or the maximum digital value (digital, 0 dBFS). This results in the peaks of the waveform being squared off, generating a harsh-sounding distortion rich in odd-order harmonics. In analog systems, some soft clipping might be musically tolerable or even desirable in small amounts, but hard clipping is generally undesirable. In digital systems, any clipping above 0 dBFS results in data loss and unpleasant, unrecoverable distortion. Proper gain staging throughout the signal path is critical to prevent clipping and maintain adequate headroom.

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7. Comb Filtering

• In Plain English and How to Use It: This is a weird, hollow, or "swooshing" sound that happens when two microphones pick up the same sound source but at slightly different times (because one is a bit further away). When these signals combine, some frequencies get boosted and others get cancelled out, creating a sound like looking through a comb – some teeth, some gaps.

  • How to Use It: This often happens with multiple mics on a drum kit, or two mics on a guitar amp, or when a mic picks up both the direct sound and a strong reflection off a nearby wall. To fix it, try moving one of the mics slightly, flipping the polarity (⊘) on one mic, or if possible, using fewer mics.

• Advanced but Still Usable: Comb filtering is a spectral distortion resulting from the coherent summation of a signal with a delayed version of itself. This creates a series of deep nulls and peaks in the frequency response, spaced harmonically (linearly in frequency). The first null occurs at f=1/(2Δt), where Δt is the time delay. Subsequent nulls occur at odd multiples of this frequency, and peaks occur at even multiples. Comb filtering can be caused by:

  • Multiple microphones at different distances from a single source.

  • A microphone picking up both a direct sound and a strong, short-delay reflection from a nearby surface.

  • Incorrectly summed signals in a console. It typically results in a colored, "hollow," "phasey," or "metallic" sound. Mitigation involves adjusting mic positions to minimize path length differences, applying precise delays to align signals, using absorptive materials to reduce reflections, or employing the 3:1 rule.

8. Compression / Compressor

• In Plain English and How to Use It: A compressor is like an automatic volume controller. It makes the loud parts of a sound quieter and can also help make the quiet parts seem louder, making the overall sound more even and controlled.

  • How to Use It: Use it on vocals to keep them from getting lost when they sing softly or jumping out too much when they sing loudly. Use it on bass guitar to even out the notes. Be gentle – too much compression can make things sound squashed and unnatural. Common controls are Threshold (when it starts working), Ratio (how much it turns things down), Attack (how quickly it reacts), and Release (how quickly it stops reacting).

• Advanced but Still Usable: A compressor is a dynamics processor that reduces the dynamic range of an audio signal. When the input signal exceeds a user-defined Threshold, the compressor attenuates the signal by a user-defined Ratio (e.g., a 4:1 ratio means for every 4dB the signal goes over the threshold, the output only increases by 1dB). Attack time determines how quickly the compressor reacts to signals exceeding the threshold, and Release time determines how quickly it returns to normal after the signal falls below the threshold. Makeup Gain is often used to restore the overall level reduced by compression. Compressors are used to control peaks, increase sustain, improve intelligibility (by making quieter parts more audible), shape transients, and achieve a more "glued" or cohesive mix. Over-compression can lead to a lifeless, "pumping" sound.

9. Crossover Frequency

• In Plain English and How to Use It: In a speaker system with multiple types of speakers (like big subwoofers for bass and smaller speakers for highs), the crossover frequency is the specific sound frequency where the job of making sound is "crossed over" from one speaker to another. For example, sounds below 100 Hz might go to the subwoofer, and sounds above 100 Hz go to the main speakers.

  • How to Use It: You usually don't set this yourself unless you're setting up a more complex PA system. But it's good to know that if your subs and main speakers aren't "playing nicely" together (e.g., a weird lack of punch or a boomy spot in the bass), it might be related to how their crossover points are aligned in terms of level and time/phase.

• Advanced but Still Usable: The crossover frequency is the frequency at which a crossover network divides an audio signal into different bands to be sent to different drivers (e.g., woofers, mid-range drivers, tweeters) in a loudspeaker system or between main loudspeakers and subwoofers. The goal is for each driver to operate only within its optimal frequency range, improving efficiency and reducing distortion. Proper crossover design also involves careful consideration of filter slopes (e.g., 12 dB/octave, 24 dB/octave Linkwitz-Riley) and, critically, the phase and time alignment between drivers at the crossover point to ensure coherent summation and avoid cancellations or lobing in the coverage pattern. Misalignment here is a common source of poor sound system performance.

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10. DAW (Digital Audio Workstation)

• In Plain English and How to Use It: This is software used on a computer for recording, editing, and playing back audio. Think of programs like Pro Tools, Logic Pro, Ableton Live, or Reaper. Churches often use DAWs to play backing tracks, click tracks for the band, or even to record services.

  • How to Use It: If your church uses backing tracks, they're likely coming from a DAW. Ensure you know how to start/stop tracks, see which tracks are playing, and that the outputs from the computer are correctly routed into channels on your mixing console for level control and any necessary EQ.

• Advanced but Still Usable: A Digital Audio Workstation (DAW) is an integrated software and (often) hardware system for recording, editing, mixing, processing, and playing back digital audio. In live sound, DAWs are commonly used for playback of multitrack stems, click tracks, and virtual instruments. They offer precise control over playback elements and allow for complex arrangements. Key considerations for live use include stable operation, low latency (especially if processing live inputs or using virtual instruments triggered live), and robust I/O management to interface with the mixing console.

11. dBFS (Decibels Full Scale)

• In Plain English and How to Use It: This is a measurement for how loud a sound is in a digital audio system (like most modern mixers). 0 dBFS is the absolute loudest a signal can be before it clips (distorts badly). You want your average sounds to be well below this, like around -18 to -12 dBFS on your input meters, to leave room for unexpected loud moments.

  • How to Use It: When setting your preamp gain for a microphone or instrument, watch the channel meter. Aim for the level to average in the -18 to -12 dBFS range, with the loudest peaks maybe hitting -6 dBFS but NEVER hitting 0 dBFS (which is usually a red light). This gives you good "headroom."

• Advanced but Still Usable: Decibels Full Scale (dBFS) is a unit of measurement for amplitude levels in digital audio systems. 0 dBFS is defined as the maximum possible digital level. Unlike analog dB measurements (like dBu or dBV which are referenced to specific voltages), dBFS is relative to this maximum. All dBFS values will be negative or zero. Exceeding 0 dBFS results in digital clipping, which is typically very harsh and unrecoverable. Maintaining an average signal level around -18 to -12 dBFS (with peaks up to -6 dBFS) during recording and mixing is common practice to ensure sufficient headroom and optimize the signal-to-noise ratio of digital converters and processors.

12. DI (Direct Input / Direct Box)

• In Plain English and How to Use It: A DI box (or a DI input on a mixer/stage box) lets you plug an instrument like an acoustic guitar, bass guitar, or keyboard directly into the mixing console without using a microphone. It changes the instrument's signal to be the right type and level for the mixer.

  • How to Use It: Use a DI for instruments that have a pickup (like acoustic/electric guitars or basses) or an electronic output (keyboards, laptops). It gives a clean, clear signal. Often, you'll blend a DI signal (e.g., clean bass) with a mic'd signal (e.g., bass amp speaker) for a fuller sound, but be aware of potential phase issues (see Phase Cancellation).

• Advanced but Still Usable: A Direct Input (DI) box converts an unbalanced, high-impedance instrument-level signal (from sources like electric guitars, basses, or keyboards) into a balanced, low-impedance microphone-level signal. This allows the signal to be sent over long cable runs without significant signal degradation or noise pickup, and properly interfaces with the mic preamps on a mixing console. Passive DIs use a transformer for this conversion and impedance matching, while active DIs use electronics (requiring power, often phantom power) for the same purpose, sometimes offering higher input impedance or gain. Many DIs also include a ground lift switch to help eliminate hum caused by ground loops.

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13. Dynamic EQ

• In Plain English and How to Use It: Imagine an EQ that only kicks in when a certain sound (frequency) gets too loud. For example, if a singer has a harsh sound only when they sing a really loud high note, a dynamic EQ could be set to turn down that harshness only at that moment, without affecting their sound the rest of the time.

  • How to Use It: This is a more advanced tool. If you have a problem frequency that only appears at certain times or levels (like sibilance on a vocal that's only bad on loud notes, or a boomy note on a bass that only rings out sometimes), a dynamic EQ can be more transparent than a regular EQ cut that would affect the sound all the time.

• Advanced but Still Usable: Dynamic EQ combines the precision of an equalizer with the level-dependent behavior of a dynamics processor. Unlike a static EQ cut or boost which is always active, a dynamic EQ band only applies its gain change when the signal level within that specific frequency band crosses a set threshold. This allows for targeted tonal adjustments that adapt to the changing dynamics of the material. For instance, it can be used to control sibilance that only becomes problematic at high vocal levels, tame resonant frequencies on drums that only ring out on loud hits, or de-ess vocals more transparently than a broadband de-esser. Key parameters include frequency, Q (bandwidth), threshold, ratio/range (amount of cut/boost), attack, and release.

14. EQ (Equalization / Equalizer)

• In Plain English and How to Use It: EQ is like a super-powered tone control for each sound. Instead of just "bass" and "treble," you can adjust very specific ranges of sound (frequencies) to make an instrument or voice clearer, less muddy, less harsh, or to help it fit better with other sounds in the mix.

  • How to Use It:

    • HPF (High-Pass Filter)/Low Cut: Use this on almost everything except kick and bass to remove low rumble.

    • Cut before you boost: If something sounds bad, try to find the annoying sound and turn it down (cut) rather than just turning other things up (boost). For example, cut some "boxy" mids (around 250-500 Hz) from a vocal if it sounds muffled.

    • Make small changes and listen carefully.

• Advanced but Still Usable: Equalization (EQ) is the process of altering the frequency response of an audio signal. EQs allow for the adjustment of amplitude (gain) of specific frequency bands. Common types include:

  • Parametric EQ: Offers control over center frequency, gain, and Q (bandwidth).

  • Shelving EQ: Boosts or cuts all frequencies above (high shelf) or below (low shelf) a selected frequency.

  • Graphic EQ: Provides fixed frequency bands with individual gain sliders. EQ is used for corrective purposes (e.g., mitigating room resonances, reducing harshness, removing mud) and creative shaping (e.g., adding "air" or "warmth"). Subtractive EQ (cutting frequencies) is often prioritized for problem-solving to maintain headroom and achieve a more natural sound. All analog and many digital IIR (Infinite Impulse Response) EQs introduce phase shift, which can impact clarity and coherence, especially when multiple EQ'd sources are summed. Linear Phase EQs avoid this phase shift but introduce pre-ringing and latency.

15. Fader

• In Plain English and How to Use It: The fader is that sliding knob (or sometimes a round knob on simpler mixers) on each channel that controls how much of that channel's sound goes to the main mix that the congregation hears. It's your primary volume control for each instrument and voice after you've set the initial gain with the preamp.

  • How to Use It: Once your preamp gain is set correctly (getting good levels on the meter, usually around -18 to -12 dBFS average), you use the faders to balance all the sounds together. The "0" mark (often labeled "U" for unity) is a good starting point for faders if the gain is set well. If you have to push a fader all the way up, your preamp gain is probably too low. If you can barely crack it open, your preamp gain is too high.

• Advanced but Still Usable: A fader is a control (typically a linear slider) on a mixing console that adjusts the level of the signal from a channel (or group, or master output) being sent to its respective bus. It operates post-preamp, post-EQ, and often post-dynamics (though signal flow can vary). The fader's scale is usually logarithmic and calibrated in decibels (dB). The "unity gain" position (often marked "0" or "U") indicates that the fader is neither attenuating nor boosting the signal passing through it at that point in the signal chain. Proper gain staging aims to have faders operate around this unity position for optimal resolution and control. Faders are the primary tool for creating the dynamic balance of the mix.

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SUNDAYMIX ACTION STEPS

3 THINGS TO IMPLEMENT THIS NEWSLETTER TODAY

1. Build Your Lexicon - Flashcard Drill.

   • Action: Pick 10 terms from today's glossary that you feel the least confident about, or that are newest to you. Create flashcards (physical or digital). Write the term on one side, and on the other, summarize the "In Plain English" definition and one key "How to Use It" tip in your own words. Drill yourself for 15-20 minutes today.

   • Why: Passive reading isn't enough for mastery. Active recall (like with flashcards) and rephrasing cements these critical terms in your mind, enabling you to think and react faster when it counts.

2. Connect Glossary to Console - The "Try It Out" Challenge.

   • Action: Choose 3-5 terms from the glossary that relate directly to functions or observable issues on your specific mixing setup (e.g., "HPF," "Gain Structure," "Polarity ⊘," "EQ," "Meters," "Feedback"). During your next soundcheck or practice session, consciously locate these controls or listen for these phenomena. As you make adjustments, say the term and its purpose out loud (or in your head) based on the glossary.

   • Why: Linking the new vocabulary to physical actions and real-time auditory feedback transforms abstract knowledge into practical skill. This is how you move from knowing about a term to knowing how to use the concept.

3. Deepen Your Understanding - Targeted Research Dive.

   • Action: Select ONE term from the "Advanced but Still Usable" definitions that particularly intrigued you or that you'd like to explore further (e.g., "Comb Filtering," "Potential Acoustic Gain," "Room Modes," "All-Pass Filter"). Spend 15 minutes today doing a quick online search for that specific term + "live sound explained" or "FOH audio tutorial." Read a reputable article or watch a short explanatory video.

   • Why: The glossary gives you the foundation. Taking the initiative to explore a concept further from another expert source will provide different perspectives, solidify your understanding, and start building the deeper expertise needed for more complex environments.

Until next time,

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