Audio conversion quality problems

conorlennon

I have recorded part of the live stream from Dublin City Anna Livia FM.
According to mplayer the stream is in the following format:

AUDIO: 44100 Hz, 2 ch, s16le, 32.0 kbit/2.27% (ratio: 3998->176400)
Selected audio codec: [ffwmav2] afm: ffmpeg (DivX audio v2 (FFmpeg))

Also using mplayer, I have dumped the recording to a WAV file.
I'm assuming that this conversion is lossless.

Using lame, I have converted this WAV file to an MP3 file as follows:

lame -b 32 -m s audiodump.wav --resample 44.1 recording.mp3

As far as I can see, the MP3 file should be the same 32kbps 44.1kHz stereo as the original stream.
However, the quality is nowhere near as good.
Even if I bump the bit rate in the MP3 to 64kbps, the quality still isn't as good.

I know that MP3 encoding is lossy, but I don't think that accounts for the big difference in quality.

Am I doing something wrong, or is this just something I have to live with?

esquier

32 or 64 bit/s is very low quality, try 192 or 224 bit/s

stereoroid

conorlennon wrote:

I know that MP3 encoding is lossy, but I don't think that accounts for the big difference in quality.

Am I doing something wrong, or is this just something I have to live with?

If the new MP3 sounds worse than the WAV, then I think the problem is probably the re-conversion in to MP3. Not all MP3 encoders are the same, and you don't know what Anna Livia used to encode it. Even the same encoder might do more damage to the file than it originally did - it works by analysing the file as it is, not as it was...

conorlennon

Thanks for the feedback.

Part of the problem I'm having is that I don't know a lot about audio codecs.
Does a particular bit rate in one codec (e.g. MP3) necessarily mean similar audio quality to the same bit rate in a different codec?