FOH Terms Flashcards
(31 cards)
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.
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.
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.
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.
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.
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.
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.
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).
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.
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.
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.”
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).
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.
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.
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.
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.
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.
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.
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.
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:
Start by setting your preamp gain correctly for each input (healthy level on meters, no red!).
Keep channel faders around “0” (unity) for good mixing control.
Make sure any boosts from EQs or makeup gain on compressors aren’t causing clipping further down the line.
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.
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.
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.
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).
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.
