That is a simple question:
All sound that passes into analogue domain using digital audio converters (DACs) have a SNR inferior to about 113 at best, and more probably 96-100... everything recorded using microphones will have inherant noise, unless the recording has been copy pasted onto a silent track leaving intervals of silence.
So what you need to do to have a very clean signal, is to take a totally digital sound which was generated with i.e. sine waves and has not had reverb and echo effects that raise the noise in the silent time(s) of the recording, and digital synthesizers which return to 0 after the note envelope. You can go though a track that contains elements of silence in Audacity or a more easy to zoom editor, and check that you are having times in the track where the amplitude = 0.0; or program a fast test code to count that the track has amplitudes of 0.0 lasting for more than 500ms or something.
if you take any recording and vary the volume of the track, for example you take an AC-DC rock song, and apply an artificial envelope that fades the track to 0.00 regularly i.e. every couple of seconds, it also makes a test file with totally silent intervals.
you can also write a code that sais : if volume < than 0.05, volume =0; to artifically make any track you want totally silent at times.
find digital recordings on freesound and measure them for 0.00 intervals in between sounds, check digital music artists that boast about 192khz recording equipment for a particular album, get a feel of which artists use state of the art production equipment, check standard commercial tracks that are used to measure monitor speakers, there are many options.