* -s 8/11.025/12/16/22.05/24/32/44.1/48 sampling frequency

When I'm using -q0 -b 64 -s 44.1 --noreplaygain -mj - %d
for Foobar2000 encoder, I have 24000Hz MP3 on output
Why does "-s 44.1" not works?
When I'm using lameXP I can force samplerate using corresponding option
P.S. Sorry for my English

so it works in lamexp, but not in foobar2000, right? if so, it is better to ask foobar/plugin developer.

The "-s 44.1" tells LAME that, if you feed in RAW PCM data, it has a samplerate of 44.1 kHz.
To control the output samplerate add "--resample 44.1" to overrule the default selected one.

Try --resample 44.1
-s sets the sample frequency of the input file, in case of a raw file.

http://lame.cvs.sourceforge.net/*checkout*...ml/switchs.html

Thanks. It works.
But this is a double resampling (44.1(input file)->24->44.1) and this may affect the quality. I understand that 64kbps is very low quality but I don't like unnecessary resamplings

No, if the input and output sample frequencies are the same, then there is no resampling happening here.

Is it possible to force output sample rate always the same as the input?