OCR · GCSE · Computer Science
Course version: J277 · Version from Sept 2022
Data storage: How sound can be sampled and stored in digital form
Sound is a continuous pressure wave, so a computer cannot record every instant of it.
Checked against the OCR J277 specification on .
- Point 12 of 13
- 4 key terms
Study notes
From the checked teaching notes for this specification point.
The idea
Sound is a continuous pressure wave, so a computer cannot record every instant of it. Instead, a microphone converts the wave into a varying analogue voltage, and an analogue-to-digital converter measures that voltage at regular intervals. Each measurement is a sample, and the number of samples taken each second is the sample rate, measured in hertz (Hz). Each sample is rounded to the nearest available level, and the number of bits used per sample is the bit depth. The resulting stream of binary values is stored as a file, for example a WAV file. Playback reverses the process: a digital-to-analogue converter turns each stored value back into a voltage, and a speaker produces the sound. A higher sample rate and bit depth capture the wave more accurately but create a larger file.
What earns the marks
- Sound is a continuous analogue wave, so it must be converted into discrete digital values before a computer can store it.
- A microphone converts the sound wave into a varying analogue electrical signal.
- An analogue-to-digital converter (ADC) samples the analogue signal at regular time intervals.
- Each sample is a measurement of the signal's amplitude at that instant, and it is quantised to the nearest representable binary value.
- The sample rate is the number of samples taken per second, measured in hertz (Hz).
- The bit depth is the number of bits used to store each sample, which determines how many amplitude levels are available.
- The sequence of binary sample values is stored in a sound file, such as a WAV file.
- Playback uses a digital-to-analogue converter (DAC) to rebuild an analogue signal that a speaker outputs as sound.
Common mistakes and how to put them right
Thinking the whole sound wave is stored continuously.
Put it right: Describe sampling as taking regular measurements of amplitude.
Confusing sample rate with bit depth.
Put it right: Link rate to time and depth to amplitude precision.
Believing a higher sample rate alone gives perfect reproduction.
Put it right: Explain that both sample rate and bit depth affect accuracy.
Exam tips
- Use the full sequence microphone → ADC → binary storage → DAC → speaker when explaining the process.
- Define sample rate in Hz and bit depth in bits, and keep the two terms clearly separate in your answer.
- When asked to compare, refer to the shape of the reconstructed wave and to file size rather than saying simply 'better quality'.
Try the lesson
The start of the Data storage: How sound can be sampled and stored in digital form lesson, made from checked teaching notes for this specification point.
Have a go first
What is the name of the process that converts a continuous sound wave into a series of digital values?
Listen
Sound is a continuous pressure wave, so a computer cannot record every instant of it. Instead, a microphone converts the wave into a varying analogue voltage, and an analogue-to-digital converter measures that voltage at regular intervals. Each measurement is a sample, and the number of samples taken each second is the sample rate, measured in hertz (Hz). Each sample is rounded to the nearest available level, and the number of bits used per sample is the bit depth. The resulting stream of binary values is stored as a file, for example a WAV file. Playback reverses the process: a digital-to-analogue converter turns each stored value back into a voltage, and a speaker produces the sound. A higher sample rate and bit depth capture the wave more accurately but create a larger file. A strong answer earns marks for these points. Sound is a continuous analogue wave, so it must be converted into discrete digital values before a computer can store it. A microphone converts the sound wave into a varying analogue electrical signal. An analogue-to-digital converter (ADC) samples the analogue signal at regular time intervals. Each sample is a measurement of the signal's amplitude at that instant, and it is quantised to the nearest representable binary value. A common slip: Thinking the whole sound wave is stored continuously: the wave is sampled at intervals, so only discrete values are kept. Instead, describe sampling as taking regular measurements of amplitude. Exam tip: Use the full sequence microphone → ADC → binary storage → DAC → speaker when explaining the process.
Key words
Say what each word means before you turn the card.
Analogue to digital conversion
Try answering aloud or in your head before turning the card.
Sampling and quantisation
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Binary representation of sound
Try answering aloud or in your head before turning the card.
File size and quality trade-offs
Try answering aloud or in your head before turning the card.
See every definition
- Analogue to digital conversion
- A microphone converts the sound wave into a varying analogue voltage, and an analogue-to-digital converter measures that voltage at regular intervals.
- Sampling and quantisation
- Each measurement is a sample, and each sample is rounded to the nearest available level.
- Binary representation of sound
- The resulting stream of binary values is stored as a file, for example a WAV file.
- File size and quality trade-offs
- A higher sample rate and bit depth capture the wave more accurately but create a larger file.
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