📄 Paper 1 · 1.1 Data Representation
1.1.6 Sound Representation and ADC
Cambridge 9618 · International A Level Computer Science · ~13 min read
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Analogue vs Digital Sound

Sound is a continuous analogue signal — a pressure wave that varies smoothly over time. Computers, however, can only store discrete digital values. To store sound digitally, the analogue signal must be converted to digital form using an Analogue-to-Digital Converter (ADC).

The ADC Sampling Process

Sampling is the process of measuring the amplitude (volume) of an analogue sound signal at regular time intervals and storing each measurement as a binary number.

  1. Sample: At each regular time interval, the ADC measures the current amplitude of the analogue signal.
  2. Quantise: The measured amplitude is rounded to the nearest available digital level.
  3. Encode: The quantised value is stored as a binary number.

Key Terms

Sample Rate (Sampling Frequency)

Sample rate is the number of samples taken per second, measured in Hertz (Hz) or kHz.

  • Higher sample rate → more samples per second → better reproduction of the original signal → larger file size
  • CD quality: 44,100 Hz (44.1 kHz)
  • Phone quality: 8,000 Hz (8 kHz)

According to the Nyquist theorem, the sample rate must be at least twice the highest frequency in the sound to reconstruct it accurately. For human hearing (up to ~20 kHz), a sample rate of ≥40 kHz is needed.

Bit Depth (Sample Resolution)

Bit depth is the number of bits used to store each sample. It determines how many different amplitude levels can be represented:

Number of amplitude levels = 2bit depth

  • Higher bit depth → more amplitude levels → finer detail → larger file size
  • CD quality: 16 bits (65,536 levels)
  • Professional audio: 24 bits

Audio File Size Calculation

File size (bits) = Sample rate × Bit depth × Duration (seconds)

For stereo (two channels), multiply by 2.

Worked Example

Calculate the file size of 3 minutes of stereo audio at 44,100 Hz sample rate and 16-bit depth:

  1. Duration = 3 × 60 = 180 seconds
  2. Bits = 44,100 × 16 × 180 = 126,720,000 bits
  3. For stereo: × 2 = 253,440,000 bits
  4. Bytes: 253,440,000 ÷ 8 = 31,680,000 bytes
  5. MB: 31,680,000 ÷ 1,048,576 ≈ 30.2 MB

Quantisation Error

Because amplitude values are rounded to the nearest digital level, there is always a small difference between the true analogue value and the stored digital value. This is called quantisation error or quantisation noise. Higher bit depth reduces quantisation error.

DAC — Digital to Analogue Converter

When playing back audio, the stored digital values are converted back to an analogue signal by a Digital-to-Analogue Converter (DAC). This signal is then amplified and sent to speakers or headphones.

FactorEffect of increasingTrade-off
Sample rateBetter frequency response — captures higher-frequency soundsLarger file size
Bit depthMore amplitude levels — less quantisation noise, better dynamic rangeLarger file size
Exam tip: Know both the sample rate and bit depth formulas. Sample rate determines how many times per second the sound is measured. Bit depth determines how precisely each measurement is stored. File size = sample rate × bit depth × duration (× channels for stereo). Cambridge exam questions often ask for the effect of changing one parameter, or ask you to calculate file sizes.
⚠️ Common Mistakes
  • Confusing sample rate (Hz) with bit depth (bits) — they are different things
  • Forgetting to multiply by 2 for stereo audio
  • Saying a higher sample rate means "more bits per sample" — it does not
  • Not knowing that the Nyquist theorem requires sample rate ≥ 2× the highest frequency
  • Forgetting quantisation error — samples are rounded to the nearest level
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Worksheet — 1.1.6 Sound Representation and ADC

8 questions · instantly marked · Cambridge 9618 standard

Q1Explain why sound must be converted before it can be stored in a computer.[2]
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Sound is an analogue signal (continuous wave) [1]; computers can only store digital (discrete binary) data, so an ADC must convert the analogue signal to digital form [1].
Q2Define 'sample rate' and state the units in which it is measured.[2]
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Sample rate is the number of samples taken per second [1]; measured in Hertz (Hz) or kHz [1].
Q3A sound recording uses a sample rate of 22,050 Hz, 16-bit depth, and lasts for 60 seconds (mono). Calculate the file size in megabytes.[4]
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Bits = 22,050 × 16 × 60 = 21,168,000 bits [1]; Bytes = 21,168,000 ÷ 8 = 2,646,000 bytes [1]; KB = 2,646,000 ÷ 1024 = 2,584 KB [1]; MB = 2,584 ÷ 1024 ≈ 2.52 MB (accept 2.5 MB) [1].
Q4Explain what is meant by 'bit depth' and state its effect on sound quality.[3]
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Bit depth is the number of bits used to represent each sample [1]; greater bit depth means more amplitude levels can be represented [1]; this reduces quantisation error/noise and improves the dynamic range of the recording [1].
Q5Explain what quantisation error is and how it can be reduced.[2]
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Quantisation error is the difference between the actual analogue amplitude and the nearest digital level it is rounded to [1]; it can be reduced by increasing the bit depth (more levels available, so rounding error is smaller) [1].
Q6State the Nyquist theorem and explain its significance for audio sampling.[2]
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The sample rate must be at least twice the highest frequency in the sound [1]; this ensures the sound can be accurately reconstructed — for human hearing (up to ~20 kHz) a sample rate of at least 40 kHz is required [1].
Q7A stereo audio file is recorded at 44,100 Hz with 16-bit depth for 2 minutes. How much larger is this file than a mono version of the same recording?[2]
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Stereo has two channels vs mono's one channel [1]; the stereo file is exactly twice the size of the mono file [1].
Q8Describe the role of a DAC (Digital-to-Analogue Converter) in audio playback.[2]
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A DAC converts the stored digital audio data back into an analogue signal [1]; this analogue signal is then amplified and sent to speakers/headphones so the sound can be heard [1].
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Mini Test — 1.1.6 Sound Representation

10 questions · 10 marks · 10 minutes

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