> hi, > > I get this segfault whenever I try to encode with lame 3.91: > > Current directory is /home/felix/src/lame-3.91/frontend/ > GNU gdb 19990928 > Copyright 1998 Free Software Foundation, Inc. > GDB is free software, covered by the GNU General Public License, and you are > welcome to change it and/or distribute copies of it under certain conditions. > Type "show copying" to see the conditions. > There is absolutely no warranty for GDB. Type "show warranty" for details. > This GDB was configured as "i686-pc-linux-gnu"... > (gdb) set args ~/music/eigene/test.wav > (gdb) r > Starting program: /home/felix/src/lame-3.91/frontend/lame ~/music/eigene/test.wav > LAME version 3.91 (http://www.mp3dev.org/) > Resampling: input 44.101 kHz output 44.1 kHz > Using polyphase lowpass filter, transition band: 15115 Hz - 15648 Hz > Encoding /home/felix/music/eigene/test.wav > to /home/felix/music/eigene/test.wav.mp3 > Encoding as 44.1 kHz 128 kbps j-stereo MPEG-1 Layer III (11x) qval=5 > Frame | CPU time/estim | REAL time/estim | play/CPU | ETA > 0/65 ( 0%)| 0:00/ 0:00| 0:00/ 0:00| 0.0000x| 0:00 0 0 404 >404
It is a bug in the resampling routine - I guess it was never tested when resampling from 44101 Hz to 44100 Hz! Quick workaround: edit the .wav header and change the samplerate from 44101 to 44100. That's only an error of 1 second for every 12 hours of music. Or check the program you used to make the test.wav, since I'm sure the intended samplerate was 44100? Temporary fix: I'm going to modify the code so that it doesn't bother to resample if the input and output rates agree to 4 digits. We should fix this bug, but what should really be done is replace the resampling code with a well designed resample + lowpass + highpass system. Mark Mark _______________________________________________ mp3encoder mailing list [EMAIL PROTECTED] http://minnie.tuhs.org/mailman/listinfo/mp3encoder
