Hi all. As an experiment, I converted a .wav file to mp3 format and then back into wav again, just to see what happens: $ ffmpeg -i file1.wav file1.mp3 $ ffmpeg -i file1.mp3 file2.wav I've always heard and read that the first step produces a loss in quality. So I would expect that to be seen in a reduction of size. Instead, I was suprised to see that file1.wav and file2.wav are both 154M large. Also the output of `ffmpeg -i' is almost the same for the two: in both cases, there is: Duration: 00:15:10.84, bitrate: 1411 kb/s Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s So I wonder, and am asking to you listers, in where that quality loss is shown and how it can be detected. Or maybe should we think and conclude that the original quality is restored with the second step...? Thanks for any help, Rodolfo