[FFmpeg-user] -ab : syntax deprecated, use '|' to separate sample rate. ????
sean darcy
seandarcy2 at gmail.com
Thu Apr 11 23:46:38 CEST 2013
I'm transcoding from wav to m4a:
ffmpeg -i $f -vn -acodec aac -ab 128000 -strict experimental $f1.m4a
I get a deprecation warning on the sample rate that I can't understand:
ffmpeg version N-51822-gf17651b Copyright (c) 2000-2013 the FFmpeg
developers
built on Apr 10 2013 20:37:57 with gcc 4.7.2 (GCC) 20120921 (Red Hat
4.7.2-2)
configuration: --prefix=/usr --bindir=/usr/bin
--datadir=/usr/share/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man
--shlibdir=/usr/lib64 --extra-cflags='-Ofast -march=native -mtune=native
-fopenmp -fomit-frame-pointer -pipe' --enable-static --enable-shared
--enable-gpl --enable-nonfree --enable-version3 --enable-postproc
--enable-avfilter --enable-pthreads --enable-x11grab --enable-gray
--enable-vaapi --enable-hardcoded-tables --enable-libaacplus
--enable-frei0r --enable-libfdk-aac --enable-libgsm --enable-libmp3lame
--enable-libopenjpeg --enable-libopus --enable-librtmp
--enable-libschroedinger --enable-libspeex --enable-libtheora
--enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid
--enable-zlib --disable-debug --cpu=amdfam10 --arch=x86_64 --enable-pic
--enable-libopencv --enable-openssl
libavutil 52. 25.100 / 52. 25.100
libavcodec 55. 2.100 / 55. 2.100
libavformat 55. 2.100 / 55. 2.100
libavdevice 55. 0.100 / 55. 0.100
libavfilter 3. 50.103 / 3. 50.103
libswscale 2. 2.100 / 2. 2.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
[wav @ 0x1e899a0] max_analyze_duration 5000000 reached at 5015510
microseconds
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from '9.1_Politics.wav':
Duration: 00:44:52.64, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz,
stereo, s16, 1411 kb/s
[audio format for output stream 0:0 @ 0x1e9d440] This syntax is
deprecated, use '|' to separate sample rate.
Output #0, ipod, to '09.1_Politics.m4a':
Metadata:
encoder : Lavf55.2.100
Stream #0:0: Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo,
fltp, 128 kb/s
From ffmpeg help:
-ab <int> E...A. set bitrate (in bits/s) (from 0 to INT_MAX)
Thanks,
sean
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