Muggy wrote: > I think that both those method would change the frequency content of > the signal.
Not nearly as much as your example. > By adding zeros between samples, the frequency content is > mirrored/repeated at higher frequencies and this can then be removed by > filtering leaving an upsampled signal with the same frequency content as > the original. > > Taking your example:- > 0, 2, 4, 6, 7 @44.1kHz > becomes > 0, 0, 2, 0, 4, 0, 6, 0, 7, 0 @88.2kHz > this is then passed through a low-pass digital filter with a cut-off > frequency set to the bandwidth of the original signal. The result will > be an upsampled signal with the same spectral content as the original. Take both the original data and your list with zeros interspersed, and plot it using your favorite plotting tool. You will see that you are adding a huge amount of very high frequency "signal" that is 100% spurious. To do a 0, 2, 0, 4, 0 requires extremely high frequency, and since real PCM is sinusoidal, it has to have -2, -4, or something close because with sine waves, you have to have negative values to match the positive, unless you have a really big low frequency signal that takes something that is really 0, +1, +2, +1, 0, -1, -2, -1, 0, +1, +2, +1, 0, -1, -2, -1, 0, into something that is positive for a long time. Anything that goes 0, 2, 0, 4, 0 has signals at least twice the frequency of a signal that goes 0, 2, 4, ... This is all just Harry Nyquist and Claude Shannon's work. -- Pat http://www.pfarrell.com/music/slimserver/slimsoftware.html _______________________________________________ audiophiles mailing list [email protected] http://lists.slimdevices.com/lists/listinfo/audiophiles
