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Quick Guide To Digital Audio

How is sound digitally recorded? How important are sample rates and resolutions? We look at the crystal clear world of digital audio.

Digitising audio is the process of converting sound into a series of numbers. The hardware that does the job is known, naturally enough, as an audio-to-digital converter, usually abbreviated to ADC. A digital-to-audio converter (DAC) reverses the process, converting digital audio in a sampler or on a hard drive, for example, to sound which we can hear.

The two most important aspects during conversion are the sample rate and sample resolution and these determine the overall quality of the material.

Sample rate

To convert sound to a digital format, the ADC measures or samples it so-many times per second. The more samples taken in a given time, the more accurate the digital representation of the sound. This can be clearly seen in the following diagrams that use a sine wave for a sound source.

This is clearly a sine wave but you can see the steps in the waveform showing the points at which it has been sampled.

This sine wave has been recorded at a higher sample rate and is therefore more accurate and closer to the original sine wave.

Although the second example is closer to a perfect sine wave, the steps are still evident and on playback it might sound a little 'rough'. The next two illustrations show what happens with extremely high and low sample rates.

This wave has been sampled at a very high sample rate and although the steps are there we can see that it is virtually a pure sine wave.

This has been sampled at a very low sample rate. In fact the samples are so far apart the sine wave shape is only barely distinguishable. On playback, it wouldn't sound anything like a sine wave.

The Nyquist limit

So, we can clearly see that the higher the sample rate, the more accurate the digital representation of the sound. Mathematician Harry Nyquist showed that to accurately digitise a sound we need only to sample it at twice its frequency. Assuming the limit of human hearing is around 20kHz, we should be able to capture our full audio spectrum by sampling at 40kHz. So the 44.1kHz sample rate of audio CDs ought to give us a little headroom.

From Harry's calculations we get the Nyquist Limit which is half the frequency of the sample rate. If you sample a frequency beyond the Nyquist Limit - that is, more than half the sampling rate - the sample is 'folded over' and stored at a value lower than it actually is. This creates an effect called aliasing which produce frequencies that were not in the original recording resulting in a distorted sound. Not something you want in a recording.

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