In simple terms
A friendly intro before the formal notes — no formulas yet.
From Wave to Bits
Computers can't store smooth sound waves directly, so they take quick snapshots of the wave's height and record them as numbers. The more snapshots you take, and the more precise the numbers are, the better the digital sound quality.
Imagine trying to copy a curvy line onto a piece of graph paper. You can't draw the smooth curve, but you can put dots on the grid where the curve passes. To play the sound back, you just connect the dots. The more dots you plot (sampling rate) and the finer the grid lines (sampling resolution), the closer your dot-to-dot drawing will be to the original smooth curve.
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Start with the original analogue sound wave, which is continuous.
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At regular, tiny time intervals, measure the amplitude (height) of the wave. This is called sampling.
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Each measurement is approximated to the nearest available level, a process called quantisation.
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This level is then stored as a binary number. The number of bits used determines the precision.
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Key formulas
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Full topic notes
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From Analogue Waves to Digital Data
To digitise sound, we must convert the continuous analogue wave into a series of discrete numerical values. This crucial task is performed by an Analogue-to-Digital Converter (ADC). The microphone captures the sound wave and converts it into an analogue electrical signal. The ADC then takes this signal and transforms it into a stream of binary data that the computer can understand. To play the sound, the process is reversed by a Digital-to-Analogue Converter (DAC), which turns the binary data back into an analogue signal sent to your speakers.
The Process of Sampling
The core of digitisation is sampling. This involves measuring the amplitude (height) of the sound wave at very regular time intervals. The frequency at which these samples are taken is called the sampling rate, measured in Hertz (Hz). For example, a sampling rate of 44,100 Hz (or 44.1 kHz), the standard for CD audio, means that the sound wave's amplitude is measured 44,100 times every second.
Higher Sampling Rate: More samples are taken per second, creating a more accurate representation of the original wave. This leads to higher fidelity sound but also a larger file size.
Lower Sampling Rate: Fewer samples are taken, potentially missing details of the wave. This results in lower quality sound but a smaller file size.
Quantisation and Sampling Resolution
Once a sample is taken, its amplitude must be recorded as a number. This is where sampling resolution (also known as bit depth) comes in. The resolution determines how many bits are used to store the value for each sample. This process of assigning a discrete numerical value to each sample's amplitude is called quantisation. A higher bit depth provides a greater number of possible values to represent the amplitude, allowing for a more precise measurement.
An n-bit resolution provides possible levels. For example, 8-bit audio has levels, while 16-bit audio has levels.
Higher Resolution (Bit Depth): Results in a greater dynamic range (the difference between the quietest and loudest sounds) and less quantisation error (noise). The file size increases.
Lower Resolution (Bit Depth): Fewer levels are available, which can lead to quantisation error being audible as a hiss or distortion. The file size is smaller.
Calculating Sound File Size
For uncompressed audio formats like WAV, the file size can be calculated directly from its properties. It's a product of how often you sample, how much data each sample holds, the duration of the audio, and the number of audio channels.
File Size (bits) = Sampling Rate (Hz) Sampling Resolution (bits) Time (s) Channels
Examiners love file size calculations. Always show your full working, including the formula. Pay close attention to units: convert kHz to Hz, minutes to seconds, and be careful when asked for the final answer in bytes, KB, MB, or GiB. Remember that 1 KB = 1024 bytes, not 1000, unless the question specifies otherwise.
Ensuring Quality: Nyquist's Theorem
How do we choose an appropriate sampling rate? The answer lies in Nyquist's Theorem. It states that to accurately reconstruct an analogue signal, the sampling rate must be at least twice the highest frequency of the signal. Since the range of human hearing is approximately 20 Hz to 20,000 Hz (20 kHz), the sampling rate for high-fidelity audio must be at least 40,000 Hz. This is why the CD audio standard is 44.1 kHz, providing a small margin above this theoretical minimum.
Nyquist's Rule: Sampling Rate 2 Highest Frequency.
Sampling below this rate (undersampling) leads to a phenomenon called aliasing, where the reconstructed audio contains frequencies that were not in the original, causing distortion.
Worked examples
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Calculate the file size in kilobytes (KB) for a 25-second mono audio clip sampled at 22,050 Hz with a 16-bit sampling resolution. Give your answer to one decimal place. (Assume 1 KB = 1024 bytes).
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- Identify the formula:<br>File Size (bits) = Sampling Rate Resolution Time Channels<br><br>2. Substitute the values:<br>File Size (bits) = 22,050 Hz 16 bits 25 s 1 (mono)<br><br>3. Calculate the total bits:<br>File Size = 8,820,000 bits<br><br>4. Convert bits to bytes:<br>File Size (bytes) = 8,820,000 / 8 = 1,102,500 bytes<br><br>5. Convert bytes to kilobytes (KB):<br>File Size (KB) = 1,102,500 / 1024 = 1076.66... KB<br><br>6. Round to one decimal place:<br>Final Answer: 1076.7 KB
A 4-minute stereo music track is recorded with a sampling rate of 48 kHz and a bit depth of 24 bits. Calculate the file size in megabytes (MB). Give your answer to two decimal places. (Assume 1 MB = 1024 KB, 1 KB = 1024 bytes).
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- Convert all units to base units:<br>Time: 4 minutes 60 seconds/minute = 240 s<br>Sampling Rate: 48 kHz 1000 = 48,000 Hz<br>Channels: 2 (stereo)<br>Resolution: 24 bits<br><br>2. Use the formula:<br>File Size (bits) = 48,000 24 240 2<br><br>3. Calculate total bits:<br>File Size = 552,960,000 bits<br><br>4. Convert bits to bytes:<br>File Size (bytes) = 552,960,000 / 8 = 69,120,000 bytes<br><br>5. Convert bytes to kilobytes (KB):<br>File Size (KB) = 69,120,000 / 1024 = 67,500 KB<br><br>6. Convert kilobytes to megabytes (MB):<br>File Size (MB) = 67,500 / 1024 = 65.9179... MB<br><br>7. Round to two decimal places:<br>Final Answer: 65.92 MB
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Glossary
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Revision flashcards
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What is sampling in the context of sound?
The process of taking measurements of the amplitude of an analogue sound wave at regular time intervals. These measurements are then converted into digital values.
Key takeaways
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Higher Sampling Rate: More samples are taken per second, creating a more accurate representation of the original wave. This leads to higher fidelity sound but also a larger file size.
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Lower Sampling Rate: Fewer samples are taken, potentially missing details of the wave. This results in lower quality sound but a smaller file size.
Practice — then mark it
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Practice Sound Calculations
Practice Sound Calculations
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