|November 25th, 2011, 23:31||#10|
Join Date: Sep 2011
Location: Australia, Tasmania
Here is a table of musical note frequencies:
and here is some ASCII art I created to try and explain it better:
volume +1 |--------| | | | | | | 0 | | | | | | volume -1 |--------| 0 1 second <-----------------> time
It is entirely possible for the waveform to have more cycles per second (> 1Hz), or less cycles per second (< 1Hz).
So a waveform with a frequency of 120.5Hz means it 'fits' 120.5 full cycles of itself in a 1 second time period.
Now, the sample rate is how many samples, or discrete parts of an analog signal, is read per second to re-create a digital version of the original analog signal.
So the more samples per second (Hz) you have when sampling an analog signal, the better, or smoother the reproduction is (and better sound quality). The downside is that it takes more memory (or disk space) to store that sound because of more samples per second.
But that is not an issue for us as we are only playing it in memory and it won't hang around very long.
See here for a good description of sampling rate:
Does this help a bit?