On Thu, Jun 22, 2017 at 3:20 PM, Rodolfo Medina <[email protected]> wrote:
> Hi all. > Hi! > > 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...? > > The conversion from WAV to MP3 is lossy, reducing the filesize by controlling the bitrate. "expanding" the MP3 back to a WAV file, it restores the large bitrate of 1411 kb/s, but it is sampling from the MP3 file, thus "expanding" the filesize back to 154MB. The original quality is lost in the last WAV file - the file sizes are just the same because they are the same bitrate. If you want to see what really happened, import both files into Audacity, use a filter to invert one of the files, and slide it around until they are 1:1 (it's off slightly IIRC) and then hit "Play" or downmix to a single track. If both tracks are aligned correctly, you will hear the discarded portions of the original WAV file during the MP3 conversion. > Thanks for any help, > > Rodolfo > > > Steve _______________________________________________ ffmpeg-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email [email protected] with subject "unsubscribe".
