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Candidates should be able to:
- explain how sound can be sampled and stored in digital form
- explain how sampling intervals and other considerations affect the size of a sound file and the quality of its playback.
How can sound be sampled and stored in digital form?
A microphone converts sound waves into voltage changes. If a microphone is plugged into a sound card then the voltage can be sampled at regular intervals (the sample rate) and each value converted into a binary number. This digitising of the sound is carried out by the Analogue to Digital Convertor (ADC) on the sound card and the series of binary numbers can be stored as a sound file.
Sampling is therefore the process of measuring the sound level (as a voltage from a microphone) at set intervals of time (the sample interval) and storing the values as binary numbers.
The sound card can recreate the stored sound using a Digital to Analogue Convertor (DAC). This converts the series of binary numbers back into a changing voltage which can make a speaker (in a set of headphones or external speakers) vibrate to reproduce the sound.
What affects the size of a sound file and the playback quality?
Sampling rate/sample interval
The sample rate is the number of samples of the analogue sound wave taken per second. This frequency is measured in Hertz (Hz).
The sample interval is the time period between each sample. It is therefore the reciprocal of the sample rate – as the sample interval decreases, the sample rate increases.
For an audio compact disk (CD) the sampling rate is 44.1KHz or 44100 samples per second so the sample interval would be the reciprocal of this so about 0.024 Milliseconds. At this sample rate, 1 minute of audio would use 10Mb of memory (using 16 bit rate sampling).
The smaller the sample interval then the more often the sound is sampled and the closer the match between the original analogue sound wave and the digitised version.
However, higher sample rates need greater space on storage devices, need faster processors to manipulate the data and files will be slower to transfer over networks and the Internet.
- A higher sample rate (lower sample period) gives a better quality recording but a larger file size.
The bit rate describes the number of memory bits that are used to store each sound sample.
If 1 bit is used then only 2 different levels of sound can be recorded and the sound would be unrecognisable when played back. If 3 bits (as illustrated in the animations above) are used then 8 different levels to be stored but the result would still be an extremely poor quality of digitised sound.
As the bit rate becomes higher, the number of different levels it is possible to record becomes greater and the closer the value stored in binary will be to the actual value that was sampled so the quality of the recording improves.
An audio compact disk (CD) uses 16 bit rate sampling which in theory gives 65,536 different levels of sound, enough for the playback quality to be difficult or impossible to distinguish from the original analogue source.
- A high bit rate gives a better quality recording but a larger file size.
Both lossy and lossless file compression can be used with sound files although the former is far more effective in reducing file sizes.
- The greater the file compression the smaller the file size. If the compression is lossy then the increased file compression will also result in a lower sound quality.