I converted a WAV file to MP3. ffmpeg's output states that it's being converted into 128k bitrate, but it ends up with only 32k bitrate.
# ffmpeg -i 3.28.09.WAV -acodec libmp3lame -ab 128k 3.28.09.mp3
ffmpeg version 0.8.6-6:0.8.6-1, Copyright (c) 2000-2013 the Libav developers
built on Mar 24 2013 07:20:17 with gcc 4.7.2
*** THIS PROGRAM IS DEPRECATED ***
This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
[wav @ 0x954f800] max_analyze_duration reached
Input #0, wav, from '3.28.09.WAV':
Duration: 00:27:07.47, bitrate: 2304 kb/s
Stream #0.0: Audio: pcm_s24le, 48000 Hz, 2 channels, s32, 2304 kb/s
Incompatible sample format 's32' for codec 'libmp3lame', auto-selecting format 's16'
Output #0, mp3, to '3.28.09.mp3':
Metadata:
TSSE : Lavf53.21.1
Stream #0.0: Audio: libmp3lame, 48000 Hz, 2 channels, s16, 128 kb/s
Stream mapping:
Stream #0.0 -> #0.0
Press ctrl-c to stop encoding
size= 25430kB time=1627.51 bitrate= 128.0kbits/s
video:0kB audio:25430kB global headers:0kB muxing overhead 0.000495%
The original WAV file is RIFF (little-endian) data, WAVE audio, Microsoft PCM, 24 bit, stereo 48000 Hz.
The output MP3 file is an audio file with ID3 version 2.4.0, contains: MPEG ADTS, layer III, v1, 32 kbps, 48 kHz, Stereo when inspected with the file utility. My PHP library getID3 also state.
# ffmpeg -i 3.28.09.mp3
ffmpeg version 0.8.6-6:0.8.6-1, Copyright (c) 2000-2013 the Libav developers
built on Mar 24 2013 07:20:17 with gcc 4.7.2
*** THIS PROGRAM IS DEPRECATED ***
This program is only provided for compatibility and will be removed in a future release. Please use avconv instead.
[mp3 @ 0x8f56800] max_analyze_duration reached
Input #0, mp3, from '3.28.09.mp3':
Metadata:
encoder : Lavf53.21.1
Duration: 00:27:07.51, start: 0.000000, bitrate: 128 kb/s
Stream #0.0: Audio: mp3, 48000 Hz, stereo, s16, 128 kb/s
At least one output file must be specified
Any ideas what I might be missing here?
ffmpeg
" from a fork. – llogan