About half of my music library is encoded as AAC while the other half is MP3... fine for my iPod... but I'm thinking of buying a small flash player for when I go the gym. These players only play mp3s (or wma, but who wants that?)... I have a very general knowledge of DSP and can't seem to understand why converting a music file from one format to another would result in a loss of quality. If it's a digital source, and the sampling frequencies are the same (44.1 kHz), shouldn't the data transfer smoothly? I understand the danger in convert from one bit-rate to another, but assuming I keep everything at 128 bps, how could I lose quality and where (amplitude, frequency response, other?). Thanks.