Please describe the problem:
Rhythmbox plays all songs with a constant sample rate, apparently with the 
highest available rate. This breaks the playback of wave files which contain 
sound in DTS or AC3 format. The correct output over spdif is only possible if a 
certain sample rate (usually 44,1kHz or 48kHz) is set before playback.
The sample rate should be adjusted for every song file according to its sample 
rate (totem does this), or at least there should be an option to enable such a 
behaviour.

Steps to reproduce:
1. verify that your sound card supports 48khz over spdif
2. start rhythmbox
3. start the playback of a 44.1khz file 



Actual results:
and there's a 48khz signal on the spdif connector

Expected results:
there shouldn't be any resampling - the sample rate of the signal on spdif 
should be the same as the sample rate of the file being played

Does this happen every time?
yes, both with alsa and esd gstreamer output plugins.

Other information:
there is a nasty workaround for playing dts 44,1khz files - you can limit the 
maximum sample rate by modifying the .asoundrc file