I use lame to encode wav audio.
When i transform a 48khz wav file in a 24 khz mp3 file (abr or vbr), the file duration is doubled and the bitrates is halved. Winamp plays the file well, but other applications have problems to read the file. I have no problem if i don't resample (48 khz) or if i resample at 32 khz. Somebody tell me that it is a lame bug.
What can i do to solve this problem?
Thanks
HG

a 24Khz (16khz/22khz) mp3 file is "Mpeg 2 layer 3", whereas 32khz/44.1khz/48khz is "Mpeg 1 layer 3".

Winamp supports "mpeg 1 layer 3 (1/2)" "mpeg 2 layer 3 (1/2?)" and "mpeg 2.5 layer 3 (1/2?)"

I doubt the file has any problem, but instead the decoder doesn't recognize those other formats (flags rather...).
So, which are the programs that you are having problems with?

I have problem with Windows media player: if i make a video with this kind of audio (using Nandub) the video stream is played twice faster (the audio has no problem). If i see the idtag of the original audio the file seem to be long twice and with bitrate halved. I have no problem with CBR audio.

I was told awhile back when having such a problem with a video I made with NanDub, being told that for a video of 24fps or higher, audio at 32kHz or higher must be used. I think 24kHz and under audio works for 15fps videos, but I guess that's useless information to you.

Windows Media Player sucks. I say forget about it and start using Zoom Player. It has all WMP's capabilities plus TONS, TONS more. Way better. So, maybe with Zoom Player you won't get this problem.
(if you download it try using 2.80 first instead of 2.90 beta1 to get the feel of a "final" zp release)