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Why does audio store so many samples of the audio? I mean, 1 sample per second should theoretically hold the same amount of audio as a 48,000 sample/sec audio file, I don't understand it, I get the bit size though AKA 16/24bit, that's not what I'm asking about.

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closed as off topic by Karan, 8088, Scott, TFM, Dave M Apr 1 '13 at 20:49

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1 sample per second should theoretically hold the same amount of audio as a 48,000 sample/sec Mathematics says NO. You don't get more with less, you get less with less. – Fiasco Labs Apr 1 '13 at 16:19
How? a sample 1 second long, would hold just as much info as a sample 48,000 samples. – MarcusJ Apr 1 '13 at 23:44
Yep, sure. <grin> do a little studying on the theory. You've missed by a whole galaxy. – Fiasco Labs Apr 2 '13 at 0:05
up vote 4 down vote accepted

The range of most human hearing is 20Hz to 20,000Hz.

Sound is when something oscillates back and forth, vibrating the air, which your ear picks up as sound.

If we have a system where a device can take 8-bit bytes from a file, convert them into analog voltages which control a speaker, to reproduce the highest frequency possible, you will need two bytes, one at the minimum value (0), and one at the maximum value. (Encoding this way is called PCM - we're assuming 8-bit PCM for this.)

So if you have a file consisting of 255, 0, 255, 0, 255, 0, it will cause the speaker to be vibrated as fast as possible. You need some difference in the values to actually move the speaker and create sound (i.e. "oscillate"). If your file is nothing but 255, 255, 255, 255, the speaker is going to stay in one position and not create any sound.

And if you want to output the full range of human frequency response, your output device needs to be able to move that speaker at least 20,000 times a second. hence at least 40,000 bytes per second.

I don't know why 44,100 was selected as a standard for CD's over 40,000Hz. 48,000Hz, which is usually really 49,152Hz, was probably selected because it's easily divisible by powers of 2 and more easily handled by digital circuits.

1 sample per second could only record a sound with a maximum of 0.5Hz - not very useful.

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Oh, my god how didn't I think of that? That makes SO much sense, thanks man. – MarcusJ Apr 1 '13 at 23:46

The basis for that is the Nyquist–Shannon sampling theorem. It says that the sampling rate has to be twice as high as the signals spectrum. This means that with 48,000 samples/sec you can sample audio signals up to 24kHz, given we start our lowest frequency at 0Hz.

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Phone service can use 8k samples/sec due to intelligible speech requiring a much narrower bandwidth. Doesn't support having nice music playback though, so 4x that is required to get the full audio spectrum. – Fiasco Labs Apr 1 '13 at 16:18

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