Converting Audio to AAC using ffmpeg
I have the following setup for FFmpeg:
ffmpeg version N-54790-g1816f55-syslint Copyright (c) 2000-2013 the FFmpeg developers
built on Jul 17 2013 21:34:32 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3)
configuration: --prefix=/usr/local/cpffmpeg --enable-shared --enable-nonfree --enable-gpl --enable-pthreads --enable-libopencore-amrnb --enable-decoder=liba52 --enable-libopencore-amrwb --enable-libfaac --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --extra-cflags=-I/usr/local/cpffmpeg/include/ --extra-ldflags=-L/usr/local/cpffmpeg/lib --enable-version3 --extra-version=syslint
libavutil 52. 40.100 / 52. 40.100
libavcodec 55. 18.102 / 55. 18.102
libavformat 55. 12.102 / 55. 12.102
libavdevice 55. 3.100 / 55. 3.100
libavfilter 3. 81.101 / 3. 81.101
libswscale 2. 4.100 / 2. 4.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
I am trying to use FFmpeg to convert my m4a or mp3 files to
AAC, Low Complexity Profile (LC)
I am really struggling to find a command line that works for me. Is it even possible with my FFmpeg setup?
Thanks
AAC-LC is the default for all of the AAC encoders supported by ffmpeg
.
Example:
ffmpeg -i input.m4a -codec:a aac output.aac
See FFmpeg Wiki: AAC Encoding Guide for more details and examples.
To re-encode any format to AAC-LC in an ADTS container (.aac file) using FFmpegs's native AAC encoder (second best after non-free Fraunhofer's libfdk_aac according to https://trac.ffmpeg.org/wiki/Encode/AAC -- doesn't support any HE-AAC though), you also need to specify -strict experimental
(or -strict -2
):
ffmpeg -i input.mp3 -strict experimental -c:a aac -b:a 128k output.aac
When converting to .aac from a source in m4a/mp4, you don't even need to re-encode:
ffmpeg -i input.m4a -c:a copy output.aac
Note: FFmpeg tries to guess the output format from the output file name. If you need to force the format for ADTS (.aac file), use -f adts
(e.g. when working with piped streams instead of files):
cat input.wav | ffmpeg -i pipe:0 -c:a aac -c:b 128k -f adts pipe:1 > output.aac