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Audio Sampling - Nyquist Shannon Theorem

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satheeshvelu

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Hi everyone,

The Audio signal frequency is limited to 20 to 20kHz.
If we need to convert Analogue Audio signal to Digital, the signal should be sampled at 40kHz. (as per Sampling Theorem)

But why is there 44.1kHz?

I have gone through the below link, which I could not follow.

Tech Talk 3: All About Sampling

Someone please explain me in a simple way.

Thank you very much.

Regards,
Satheesh
 

msabeel

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According to the Sampling Theorem, the MINIMUM sampling rate should be twice the highest frequency. The minimum sampling rate is called Nyquist Rate. When the sampling rate is greater than the Nyquist rate, the signal is said to be oversampled.

Although we can sample a signal exactly at the Nyquist rate ( which in your case is 40KHz), but it is much better to oversample a signal. The reason being that sampling at exactly Nyquist Rate requires a high performance analog filter which are usually difficult to design and also they are expensive.
 

venky yadav

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Sampling exactly at 40Khz will obviously give you better bandwidth efficiency but your tradeoff would be with the filters which will require sharp cutoff
whereas relaxing the cutoff for filter i.e sampling at 44.1KHz will provide a guard band but the tradeoff would be sacrifice of bandwidth
 
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