real problem in sampling audio signal...

Hi, all

when doing the sampling for the audio, do we really care about the sampling rate?

I mean as long as we satsify the nyquist conditoin, we will always get back the original audio?

The process is done on a PC, I tired several times, it seems this is true.

If so, does this mean the reconstruction filter I have build is really flexible?

You can get intelligible speech at 8k sampling rate.
And I believe I've seen that rate for music streamed over the internet (such as Youtube).

However it is not hi-fidelity. It may resemble a cheap AM radio. An audiophile will tell you a great deal is lost, as far as sonic quality.

That is why the music industry chose a 44.1k sampling rate. Studios use even higher rates.

I once tried recording music into the computer at a 22k sample rate. The playback sounded like the original FM broadcast to my ears (note: to my ears).

There are claims that our listening experience is enhanced when we are exposed to the ultrasonic frequencies in the music, even though tests show we do not 'hear' them.

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As for mp3's I don't know how they do it. They compress the data unbelievably, yet retain excellent sound quality.
The 160 kbps (or 128k) is good enough to get widespread acceptance.