[linux-audio-user] lame and low sample-rate mp3's

[Marco Scoffier]

I am writing here because my problem involves many tools.

I have been trying to make extremely compact mp3's for download.

$lame PO_audio.wav -m m -b 8 PO_audio-low.mp3
LAME version 3.93 MMX  (http://www.mp3dev.org/)
CPU features: i387, MMX (ASM used), SIMD
Autoconverting from stereo to mono. Setting encoding to mono mode.
Resampling:  input 32 kHz  output 8 kHz
Using polyphase lowpass  filter, transition band:  2742 Hz -  2839 Hz
Encoding PO_audio_kino.wav.wav to PO_audio-low.mp3
Encoding as 8 kHz   8 kbps single-ch MPEG-2.5 Layer III (16x) qval=2
     Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA   
1038/1040  (100%)|    0:05/    0:05|    0:05/    0:05|   13.322x|    0:00 
average:   8.0 kbps

Everything looks right except that when I play the resulting file in xmms or 
mgp123 it plays much too fast.


Could the headers on the mp3 be wrong? how do I check? they show up correctly 
in xmms and mpg123.

mpg123 PO_audio-low.mp3
( ... )
Playing MPEG stream from PO_audio-low.mp3 ...
MPEG 2.5 layer III, 16 kbit/s, 11025 Hz mono

I know I used to play lower than CD quality sample rates.

Does anyone have an idea what is going on?

I have tried differing combinations of ecasound and lame with combinations of 
-sr and -rate switches, to often similar results.

I am using the alsa drives from cvs for my audiofile 2496,

Thanks,

--Marco