akwatve
May 8th, 2008, 12:18 AM
Hello,
I tried to use ffmpeg to convert (and resample) audio files. But I am facing few problems
1) When I explicitly specify audio codec as mp3, ffmpeg gives codec error
(Wed May 07 19:17:21)/data1> ffmpeg -ab 128k -acodec mp3 -i foo.wma foo.mp3
FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2007 Fabrice Bellard, et al.
configuration: --enable-gpl --enable-pp --enable-swscaler --enable-pthreads --enable-libvorbis --enable-libtheora --enable-libogg --enable-libgsm --enable-dc1394 --disable-debug --enable-shared --prefix=/usr
libavutil version: 1d.49.3.0
libavcodec version: 1d.51.38.0
libavformat version: 1d.51.10.0
built on Mar 12 2008 15:36:03, gcc: 4.2.3 (Ubuntu 4.2.3-2ubuntu4)
Input #0, asf, from 'foo.wma':
Duration: 00:04:34.5, start: 2.600000, bitrate: 96 kb/s
Stream #0.0: Audio: wmav2, 44100 Hz, stereo, 96 kb/s
Output #0, mp2, to 'foo.mp3':
Stream #0.0: Audio: 0x0000, 44100 Hz, stereo, 128 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Unsupported codec for output stream #0.0
However, when I don't specify any codec, conversion completes successfully. I am not sure whats going on... I have mp3lame-dev packages installed.
2) When I convert a file to mp3, I cannot play that file on my cell phone. Is there any difference between format of the mp3 file generated by ffmpeg and format expected by phones ? I have Sony Ericsson w580i
I am using 64 bit kubuntu hardy.
Thanks,
I tried to use ffmpeg to convert (and resample) audio files. But I am facing few problems
1) When I explicitly specify audio codec as mp3, ffmpeg gives codec error
(Wed May 07 19:17:21)/data1> ffmpeg -ab 128k -acodec mp3 -i foo.wma foo.mp3
FFmpeg version SVN-rUNKNOWN, Copyright (c) 2000-2007 Fabrice Bellard, et al.
configuration: --enable-gpl --enable-pp --enable-swscaler --enable-pthreads --enable-libvorbis --enable-libtheora --enable-libogg --enable-libgsm --enable-dc1394 --disable-debug --enable-shared --prefix=/usr
libavutil version: 1d.49.3.0
libavcodec version: 1d.51.38.0
libavformat version: 1d.51.10.0
built on Mar 12 2008 15:36:03, gcc: 4.2.3 (Ubuntu 4.2.3-2ubuntu4)
Input #0, asf, from 'foo.wma':
Duration: 00:04:34.5, start: 2.600000, bitrate: 96 kb/s
Stream #0.0: Audio: wmav2, 44100 Hz, stereo, 96 kb/s
Output #0, mp2, to 'foo.mp3':
Stream #0.0: Audio: 0x0000, 44100 Hz, stereo, 128 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Unsupported codec for output stream #0.0
However, when I don't specify any codec, conversion completes successfully. I am not sure whats going on... I have mp3lame-dev packages installed.
2) When I convert a file to mp3, I cannot play that file on my cell phone. Is there any difference between format of the mp3 file generated by ffmpeg and format expected by phones ? I have Sony Ericsson w580i
I am using 64 bit kubuntu hardy.
Thanks,