Sound Design Technology Flashcards

Question 1

What is sound in physical terms, and how does it travel?

Accepted Answer

Sound is a mechanical vibration that travels as a pressure wave through a medium such as air, water, or solid material. A vibrating source pushes neighboring particles together and apart, and each particle passes the disturbance along without itself moving far from its resting position.

Question 2

What does frequency measure, and what unit is used?

Accepted Answer

Frequency measures how many complete cycles of a vibration occur each second, and it determines the perceived pitch of a sound. It is expressed in hertz, where one hertz equals one cycle per second. Higher frequencies sound higher in pitch, while lower frequencies sound deeper.

Question 3

What is amplitude, and how does it relate to perceived loudness?

Accepted Answer

Amplitude is the magnitude of a sound wave's displacement or pressure variation from its resting level. Greater amplitude generally produces a louder perceived sound, though loudness also depends on frequency and duration. Sound pressure is often measured in decibels, a logarithmic scale that compares a level to a chosen reference value.

Question 4

What is the difference between a sine wave and a complex waveform?

Accepted Answer

A sine wave is a pure tone with a single frequency and no overtones. A complex waveform combines several sine components, each with its own frequency and amplitude. Timbre, the quality that distinguishes a violin from a flute playing the same note, comes from the balance of those harmonic components.

Question 5

What is the speed of sound in air, and what factor most affects it?

Accepted Answer

Sound travels through air at roughly 343 meters per second at about 20 degrees Celsius. Temperature has the greatest effect on this speed, since warmer air carries the pressure wave faster. Speed also differs between media, traveling much faster through water and solids than through air.

Question 6

What is the Doppler effect, and where does it appear in sound design?

Accepted Answer

The Doppler effect is the apparent change in a sound's frequency when the source and listener move relative to each other. An approaching source sounds higher in pitch, and a receding source sounds lower. Sound designers use this effect to make vehicles, flybys, and passing objects feel like they are moving through a space.

Question 7

Why does the same sound seem different when heard in a small room compared with an open field?

Accepted Answer

In a small room, sound reflects off nearby walls, ceilings, and floors, creating many closely spaced echoes that blend into reverberation. In an open field, reflections are sparse, so the sound decays quickly and feels dry and direct. The listener's sense of space depends largely on this pattern of early and late reflections.

Question 8

What is the inverse square law, and how does it affect sound levels with distance?

Accepted Answer

The inverse square law states that sound intensity from a point source falls in proportion to the square of the distance. Doubling the distance from a source reduces intensity to about one quarter, which is roughly a six decibel drop. This principle explains why distant sounds feel quieter.

Question 9

How does a dynamic microphone convert sound into an electrical signal?

Accepted Answer

A dynamic microphone uses a diaphragm attached to a small coil that sits within a magnetic field. When sound pressure moves the diaphragm, the coil moves through the field and induces a small voltage that mirrors the waveform. Dynamic microphones are rugged, handle high sound pressure well, and need no external power.

Question 10

How does a condenser microphone differ from a dynamic microphone in sensitivity?

Accepted Answer

A condenser microphone uses a thin diaphragm placed close to a fixed backplate, forming a capacitor whose capacitance changes as the diaphragm moves. Condenser microphones are generally more sensitive and capture finer detail and high frequencies than dynamic designs. They need external power, typically phantom power from a mixer or interface.

Question 11

What does the polar pattern of a microphone describe?

Accepted Answer

A polar pattern describes how sensitive a microphone is to sound arriving from different directions around it. Cardioid patterns favor the front and reject sound from the rear, omnidirectional patterns pick up sound equally from all sides, and figure eight patterns capture the front and back while rejecting the sides.

Question 12

What is proximity effect, and why does it matter when recording close to a microphone?

Accepted Answer

Proximity effect is the boost in low frequencies that directional microphones produce when a sound source sits very close to the capsule. The bass response rises noticeably as distance shrinks, which can make voices sound warm but muddy. Engineers manage it by adjusting microphone distance, using a high-pass filter, or choosing an omnidirectional pattern.

Question 13

Why do recording engineers place a microphone at different distances from a sound source?

Accepted Answer

Close placement captures a direct, detailed sound with little room coloration and strong isolation from surroundings. Distant placement picks up more of the space, including reflections and ambience, which can add realism but also more noise and less clarity. Engineers choose a distance by balancing direct sound against room sound.

Question 14

What is the purpose of a pop filter, and where is it usually placed?

Accepted Answer

A pop filter is a mesh or foam screen that reduces plosive bursts of air from consonants such as p and b before they strike the diaphragm. It is usually placed a few centimeters in front of the microphone, between the performer and the capsule. Using one prevents sharp low-frequency thumps that are difficult to remove in editing.

Question 15

What is gain staging, and why does it matter during recording?

Accepted Answer

Gain staging is the practice of setting input and output levels at each stage of a signal chain so that the signal stays strong without clipping. Recording too quietly raises the noise floor, while recording too hot causes distortion that cannot be undone. Setting healthy levels at the start gives editing and mixing enough headroom to work cleanly.

Question 16

What is the difference between a room mic and a close mic in a multitrack session?

Accepted Answer

A close mic sits near a single source and captures its direct sound with clarity and isolation from other instruments. A room mic is placed farther away to record the acoustic space itself, including reflections, decay, and the sense of the room. Mixing the two lets producers blend precise detail with natural ambience.

Question 17

What is sampling, and what does sample rate determine?

Accepted Answer

Sampling is the process of measuring an analog sound wave's amplitude at regular intervals and storing each measurement as a number. Sample rate is the number of these measurements taken per second, and it determines the highest frequency a digital recording can represent.

Question 18

What is bit depth, and how does it affect a recording?

Accepted Answer

Bit depth is the number of binary digits used to store each sample, and it sets the resolution of amplitude values. Higher bit depths provide a greater range between the quietest and loudest representable levels, which lowers the quantization noise floor. Common production formats use 16 or 24 bits.

Question 19

What is aliasing, and how is it prevented?

Accepted Answer

Aliasing is distortion that occurs when frequencies above half the sample rate are recorded and appear as false lower frequencies. It is prevented by applying an anti-aliasing low-pass filter before sampling, which removes content above the Nyquist limit. Oversampling also reduces the risk by pushing filtering artifacts beyond the audible range.

Question 20

What is the difference between lossless and lossy audio compression?

Accepted Answer

Lossless compression reduces file size while allowing the original audio data to be perfectly reconstructed. Lossy compression discards information judged less perceptible to the listener, producing smaller files at the cost of permanent data loss. Sound designers usually keep working masters in lossless or uncompressed formats and reserve lossy formats for delivery.

Question 21

What is dithering, and when is it applied?

Accepted Answer

Dithering is the addition of very low-level noise before reducing bit depth, which randomizes quantization errors that would otherwise create audible distortion. It is typically applied when exporting a 24-bit mix to a 16-bit file for CD or streaming delivery, which avoids gritty distortion in quiet passages.

Question 22

What is the difference between a digital clip and analog saturation?

Accepted Answer

A digital clip occurs when a signal exceeds the maximum value a system can represent, so peaks are flattened abruptly and produce harsh harmonic distortion. Analog saturation happens gradually as a circuit or tape approaches its limits, compressing peaks smoothly and often adding pleasing harmonics. Digital systems offer no such gradual transition.

Question 23

What is latency, and why does it matter when recording and monitoring?

Accepted Answer

Latency is the delay between a sound entering an audio system and its output emerging from the speakers or headphones. Excessive latency makes performers hear themselves late, which disrupts timing and feels unnatural. Latency comes from converters, buffer size, and processing load, and it can be reduced by lowering the buffer size.

Question 24

What is a sample library, and how do designers use one?

Accepted Answer

A sample library is a collection of recorded sounds, often organized by instrument, action, or category, that designers can trigger, layer, and manipulate. Designers use libraries as raw material, selecting, editing, and processing recordings to build new effects or musical parts. Many also record original sources for a distinct character.

Question 25

What is subtractive synthesis, and how does it shape a sound?

Accepted Answer

Subtractive synthesis starts with a harmonically rich waveform, such as a sawtooth or square wave, and removes frequencies using a filter. Cutoff frequency and resonance determine which harmonics pass through, so sweeping the filter changes brightness and character over time. Envelopes then shape how the sound evolves over time.

Question 26

What do the attack, decay, sustain, and release stages of an ADSR envelope control?

Accepted Answer

The attack stage sets how quickly a sound reaches its peak after a key press, decay sets how fast it falls to a sustained level, sustain holds that level while the key remains down, and release controls how long the sound fades after the key is released. Together these four values describe how a note's loudness or filter brightness changes over time.

Question 27

What is a low-pass filter, and what effect does it have on a sound?

Accepted Answer

A low-pass filter allows frequencies below a chosen cutoff point to pass while attenuating frequencies above it. Lowering the cutoff makes a sound darker and more muffled, as bright upper harmonics are removed. Designers use low-pass filters to simulate distance, muffle objects behind walls, or soften harsh material.

Question 28

What is equalization, and how do boost and cut settings differ in practice?

Accepted Answer

Equalization adjusts the level of specific frequency bands in a sound, either raising them with a boost or lowering them with a cut. Boosting a band emphasizes its character but can add harshness or resonance, while cutting it removes unwanted energy such as rumble or boxiness.

Question 29

What does a compressor do to the dynamic range of a sound?

Accepted Answer

A compressor reduces the difference between a sound's loudest and quietest parts by lowering levels above a set threshold by an amount determined by the ratio. Attack and release settings control how quickly the compressor responds and recovers. Compression can tame unpredictable performances or add punch to percussive sounds.

Question 30

What is convolution reverb, and how does it differ from algorithmic reverb?

Accepted Answer

Convolution reverb applies the measured acoustic response of a real space or object, called an impulse response, to an audio signal. Algorithmic reverb generates reverberation mathematically using delay networks and filters, giving the designer adjustable parameters such as decay time and density rather than a captured space.

Question 31

What is a send and return bus, and why do mixers use one?

Accepted Answer

A send and return bus routes a portion of several channels' signals to a shared effect, such as reverb or delay, and then returns the processed result to the mix. Each channel controls how much of its signal is sent, so different sources can receive different amounts of the same effect. This approach saves processing power and keeps the treatment consistent across the mix.

Question 32

How does a ring modulator create new frequencies from two input signals?

Accepted Answer

A ring modulator multiplies two input signals together, producing output frequencies equal to the sum and difference of the originals. If the inputs are not harmonically related, the result sounds metallic, bell-like, or inharmonic, which is useful for robotic voices, alien textures, and unusual percussion.

Question 33

What is a sound design brief, and what should it define?

Accepted Answer

A sound design brief is a document or conversation that defines the purpose, context, and constraints of the audio being created. It should describe the emotional goal, the visual or interactive context, the required format and length, and any technical limits such as file size or playback system. A clear brief prevents wasted effort and gives the designer criteria for judging whether each sound succeeds.

Question 34

Why do sound designers create a spotting list or cue sheet?

Accepted Answer

A spotting list identifies every moment in a project where sound is needed, along with its timing, duration, and description. Designers use it to plan their workload, estimate how many assets must be created, and confirm that nothing is overlooked before production advances.

Question 35

What is layering in sound design, and why do designers layer sounds?

Accepted Answer

Layering combines several sounds, each contributing a distinct quality such as weight, texture, transient, or tail, into one composite effect. Designers layer because a single recording rarely captures every characteristic a moment needs, so blending sources gives more control over realism and emotional impact. Each layer can be processed, balanced, and timed independently.

Question 36

How do film and game sound effects differ in their mixing approach?

Accepted Answer

Film sound is usually mixed to a fixed timeline, so designers can balance each sound precisely against picture and dialogue. Game sound must respond to unpredictable player actions, so designers build variations, randomization, and dynamic mixing systems that adapt in real time.

Question 37

Why is variation important when designing repeated sounds such as footsteps?

Accepted Answer

Repeating the exact same sample makes a sound feel mechanical and quickly becomes noticeable to listeners. Designers solve this by creating several recorded versions and varying pitch, volume, and timing slightly with each playback. Randomized selection from a pool of variations keeps repeated events natural while still sounding consistent.

Question 38

What is loudness normalization, and why do streaming platforms use it?

Accepted Answer

Loudness normalization adjusts playback levels so that different tracks or programs sound roughly equally loud to the listener. Platforms use it to reduce abrupt volume jumps between content and to keep listening levels consistent across an entire playlist. Designers measure integrated loudness in units such as LUFS.

Question 39

What is a stem, and why do mix engineers deliver stems to clients?

Accepted Answer

A stem is a grouped mix of related elements, such as dialogue, music, or effects, bundled into one submix. Delivering stems allows clients and other departments to adjust balance, translate content into other languages, or create alternate versions without requesting new mixes.

Question 40

How do sound designers evaluate whether a finished sound works in context?

Accepted Answer

Designers evaluate a sound by hearing it in its intended context, such as the final scene, level, or interface, rather than in isolation. They check its timing, clarity, emotional effect, and balance against surrounding audio, and they test it on the playback systems the audience will use.

Sound Design Technology Flashcards

1Sound Fundamentals

2Microphones and Recording

3Digital Audio Basics

4Synthesis and Signal Processing

5Applied Sound Design Workflow

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