How do you capture audio cleanly in SQA Higher Music Technology, and what do you need to know about microphones, placement and gain for the question paper?
Developing audio capture skills: choosing and positioning microphones, setting input levels and gain, and recording sources cleanly without distortion or noise.
Audio capture in SQA Higher Music Technology: choosing microphone types (dynamic, condenser, ribbon), reading polar patterns, positioning mics, setting gain to avoid clipping and noise, and recording sources cleanly for later processing and mixing.
Reviewed by: AI editorial process; not yet individually human-reviewed
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What this dot point is asking
Audio capture is the first practical skill in SQA Higher Music Technology: getting sound into the system cleanly so that everything downstream (editing, processing, effects, mixing) has good material to work with. The question paper tests your knowledge and understanding of how capture works, and the assignment tests you doing it. A question on capture asks you to choose a suitable microphone, position it sensibly, and set levels so the recording is free of distortion and excessive noise. This dot point sets out the microphone types, polar patterns, placement principles and gain staging you are expected to understand.
The answer
To capture audio cleanly, match the microphone to the source, position it for the sound you want, and set the gain so the signal sits well above the noise floor but never reaches the clipping point. There are three microphone families to know: dynamic microphones (robust, handle high volume, used on drums and loud amps), condenser microphones (sensitive, wide frequency response, used on vocals and acoustic instruments, need phantom power) and ribbon microphones (smooth, warm, fragile). Each microphone has a polar pattern that describes the directions it picks up from. Placement controls tone and spill, and gain staging sets the level: aim for healthy peaks with headroom below 0 dBFS so the recording is clean and quiet.
Microphone types and when to use them
The three families differ in how they convert sound to a signal, and that shapes their use.
- Dynamic microphones use a moving coil and are robust and inexpensive. They handle high sound-pressure levels without distorting, so they suit loud sources: snare and toms, guitar amplifiers, and live vocals. They are less sensitive to quiet detail.
- Condenser microphones use a charged diaphragm and capacitor, need phantom power (usually plus 48 volts from the interface or desk), and are sensitive with a wide, even frequency response. They capture detail and transients well, so they suit studio vocals, acoustic guitar, piano and overheads.
- Ribbon microphones use a thin metal ribbon in a magnetic field, give a warm, smooth tone with a natural high-frequency roll-off, and are fragile. They suit brass, strings and guitar amps where smoothness is wanted.
Polar patterns
A microphone's polar pattern is the shape of its sensitivity to sound from different directions. The three to know are:
- Cardioid (heart-shaped): most sensitive to the front, rejects the rear. The default for close-miking single sources because it reduces spill from other instruments and the room.
- Omnidirectional: equally sensitive in all directions. Captures the room and is natural-sounding, but picks up everything, so it suits a controlled space or a deliberate room sound.
- Figure-of-eight (bidirectional): sensitive front and rear, rejects the sides. Used for specific techniques and is the natural pattern of most ribbon microphones.
Placement: distance, angle and the proximity effect
Where you put the microphone shapes the recorded tone as much as the microphone itself. Closer placement gives a drier, more detailed, more isolated sound; further placement captures more room and a more natural blend. With directional microphones, moving closer increases bass through the proximity effect, which can warm a vocal or muddy it if overdone. Angling a microphone off-axis to a source reduces harshness and plosives. For a vocal, a typical setup is a cardioid condenser about 15 to 20 cm away with a pop shield to reduce the blasts of air from "p" and "b" sounds (plosives). These choices are deliberate and examinable: a question may ask you to justify a placement.
Gain staging: levels, headroom and the noise floor
Gain staging is setting the input level correctly. Two problems sit at the extremes. Too much gain causes clipping: the signal exceeds 0 dBFS (the digital maximum), the waveform is flattened, and the sound distorts harshly. This cannot be repaired later, so it must be avoided at capture. Too little gain leaves the signal close to the noise floor (the constant low-level hiss of the system), so when you raise it later you also raise the noise. The target is healthy peaks with headroom: set the gain so the loudest part of the performance peaks comfortably below 0 dBFS, commonly aiming for peaks around minus 6 dBFS during a level check, watching the meters and re-checking before the take.
Examples in context
Recording an acoustic guitar for the assignment, you might choose a small-diaphragm condenser (sensitive, wide response), set it to cardioid (to reject room spill), and place it about 30 cm from where the neck meets the body, angled slightly off the sound hole to avoid boom. You run a level check by playing the loudest strum and set the gain so it peaks near minus 6 dBFS, giving headroom against clipping.
Recording a snare drum, you would instead choose a dynamic microphone (robust, handles the high level), place it close to the head angled across the skin, and set a lower gain because the source is loud. The reasoning, microphone family to source, pattern to spill, distance to tone, gain to headroom, is the same in both cases and is exactly what the question paper rewards.
Try this
Q1. Why would you choose a dynamic microphone over a condenser for a snare drum? [2 marks]
- What the marker wants. Dynamic microphones are robust and handle high sound-pressure levels without distorting, so they suit a loud source like a snare; a sensitive condenser could distort and is less rugged.
Q2. What is the proximity effect, and how can it be used? [2 marks]
- What the marker wants. With directional microphones, moving closer to the source boosts low frequencies; it can be used to warm a thin vocal, but overuse makes the sound boomy.
Q3. What does "headroom" mean and why does a recordist leave it? [2 marks]
- What the marker wants. Headroom is the gap between the peak signal level and 0 dBFS; it is left so transient peaks do not clip and distort, keeping the capture clean.
A note on sources
This guide is AI-written and not individually human-reviewed. The audio capture skills follow SQA's Higher Music Technology course specification (C851 76); verify current detail against the SQA Higher Music Technology documents at sqa.org.uk.
Exam-style practice questions
Practice questions written in the style of SQA exam questions on this dot point, with worked answer explainers. The year tag is the paper they imitate, not the source.
Higher specimen4 marksDescribe two factors you would consider when capturing a vocal recording, and explain how each affects the quality of the recording. (4 marks)Show worked answer →
A skills-knowledge question on audio capture. Two marks reward a named factor; two reward a clear effect on quality, so pick factors you can explain precisely.
Microphone choice is the safest first factor: a condenser microphone is a strong choice for vocals because its wide frequency response and high sensitivity capture detail and breathiness, giving a clear, full recording. Gain staging is a strong second factor: setting the input gain so the loudest sung note peaks near minus 6 dBFS leaves headroom, so the recording avoids clipping (digital distortion) while staying well above the noise floor.
Other valid factors are microphone placement and distance (a pop shield and roughly 15 to 20 cm reduce plosives and excessive proximity-effect bass), and the recording environment (a treated or quiet room reduces unwanted reflections and background noise). The discriminator is the move from naming a factor to explaining its effect on quality, never just listing equipment.
Higher 20193 marksExplain what is meant by clipping and how a recordist avoids it. (3 marks)Show worked answer →
A definition-and-application question. One mark defines clipping, the rest reward how it is avoided.
Clipping is the distortion that happens when a signal exceeds the maximum level the system can represent (0 dBFS in digital), so the tops of the waveform are flattened and the sound becomes harsh and broken. It is avoided by gain staging: setting the input gain so the loudest part of the source peaks with headroom below 0 dBFS (commonly aiming for peaks around minus 6 dBFS), watching the meters during a level check, and re-recording or lowering gain if the meter hits the red.
A weak answer confuses clipping with simply being loud, or says only "turn it down" without the idea of headroom and metering. The point is that clipping is a hard digital ceiling, so capture leaves room below it.
Related dot points
- Understanding the audio signal path and hardware: how sound travels from source through microphones, cables, preamps, audio interfaces and mixing desks to the recording, and how monitoring works.
The audio signal path in SQA Higher Music Technology: how sound travels from source through microphones, cables and connectors, preamps and audio interfaces, mixing desk channel strips, to the recording and monitors, including analogue-to-digital conversion.
- Using hardware and software to manipulate audio: editing in a DAW (cutting, copying, comping, fades, crossfades), and correcting timing and pitch (quantise, pitch correction, time-stretching), with sample rate and bit depth.
Editing and manipulating audio in SQA Higher Music Technology: DAW editing (cut, copy, comp, fades, crossfades), timing and pitch correction (quantise, pitch correction, time-stretching), looping, and the meaning of sample rate and bit depth for audio quality.
- Processing audio: using equalisation (EQ) to shape frequency content and dynamics processing (compression, limiting, gating, normalisation) to control level, and knowing what each does and why.
Audio processing in SQA Higher Music Technology: equalisation (EQ) to boost or cut frequency bands and shape tone, and dynamics processing (compression, limiting, gating, normalisation) to control the loud-quiet range, with the key controls (threshold, ratio, attack, release) explained.
- Mixing and sequencing: balancing levels and panning, using MIDI sequencing and virtual instruments, automation, and producing a stereo mixdown, plus an awareness of mastering.
Mixing and sequencing in SQA Higher Music Technology: balancing levels and panning across the stereo field, MIDI sequencing and virtual instruments, automation of parameters over time, the stereo mixdown (bounce), and an awareness of mastering.
- Explaining how technological developments relate to 20th and 21st century music: how recording, amplification, electronic instruments, multitrack, sampling, MIDI, the DAW and digital distribution changed how music was made and heard.
How technological developments shaped 20th and 21st century music for SQA Higher Music Technology: recording and amplification, electric and electronic instruments, multitrack recording, sampling and drum machines, MIDI, the DAW, and digital distribution and streaming, and how each changed musical style.
Sources & how we know this
- Higher Music Technology Course Specification (C851 76) — SQA (2024)
- Music Technology subject page (Higher) — SQA (2026)