How do we convert audio from analogue to digital and back?
Computers and phones can't process analogue signals - but they can convert them.
All sound is made up of a continuous analogue soundwave, but digital kit like computers, phones and CD players can’t process sound in its analogue form, which means it has to be converted into a digital signal – a long chain of numbers, made up of ones and zeros – to be stored and processed digitally. This is done using an analogue-to-digital converter (ADC), while a digital-to-analogue converter (DAC) is used to reverse the process at the other end.
An ADC works by first taking ‘samples’ of the soundwave, which measure its amplitude at regular intervals. In order to put a numerical value on these samples, the amplitude is split into levels along the vertical axis.
The number of levels depends on the bit depth, which is determined by the resolution of your ADC. Most commonly, they are 8-, 16- or 24-bit. The greater the bit depth, the more accurate the signal will be, because every sample measurement is converted to the binary value that intersects it on the waveform.
If it falls between two binary values, it will be rounded up or down accordingly. This is called quantisation, and is how the digital signal deviates from the analogue. However, the higher the resolution, the more binary values there are for samples to be converted to and the less quantisation occurs. With an 8-bit ADC, there are 256 possible binary values. With a 16-bit ADC, there are 65,536.
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As for DACs, they decode the binary digital signal to rebuild the analogue wave, smoothing out the ‘steps’ that are added through quantisation in the process. This is called interpolation, which looks at two points and approximates the values between them in order to fill in the gaps.
Read more about the science of sound:
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