I want to add a little to the good response about connecting the
DAC to a LM386.  You might find the sound to be somewhat wraspy due to the
fact that you also need a low-pass filter between the DAC and the final
audio stages.  This is to smooth out the steps caused by the digital
sampling.  If, for instance, the sampling rate is 8,000 samples per second,
the highest audio frequency that will be correctly reproduced is 4,000 HZ.
You will be able to hear the 8,000 HZ sampling quite easily as a harsh
and high-pitched counterpart to the signals you want to hear.

        When recording the sounds you want to reproduce, you will need a
similar low-pass filter on the input to the system to prevent aliasing
or the result will be a bad recording from the beginning.

        This may sound like complexity for the sake of complexity, but
the filter can be anything from a RC circuit set to the correct time constant
to a more elaborate low-pass filter with sharper cut-off, but the idea is
to record your sound with the highest input audio frequency about half that
of the sampling rate and to play your recordings through a filter whose
cut-off frequency is just above half the sampling rate.  This is the
Nyquist frequency and is based upon the minimum number of samples it takes
to accurately describe a wave form.

        It won't hurt a thing to try it without the filter at least on
playback and see if you like it, but I'd leave some room for it in case
it sounds too bad.  You should definitely filter the input to the original
recording.

Martin McCormick WB5AGZ  Stillwater, OK
OSU Center for Computing and Information Services Data Communications Group