Advanced Audio Coding (AAC) is the successor format to MP3, and is defined in MPEG-4 part 3 (ISO/IEC 14496-3). It is often used within an MP4 container format; for music the .m4a extension is customarily used. The second-most common use is within MKV (Matroska) files because it has better support for embedded text-based soft subtitles than MP4. The examples in this guide will use the extensions MP4 and M4A.
FFmpeg can support four AAC-LC encoders (aac, libfaac, libfdk_aac, libvo_aacenc) and two AAC-HE encoders (libaacplus and libfdk_aac). The licenses of libaacplus, libfaac, and libfdk_aac are not compatible with the GPL, so the GPL does not permit distribution of binaries containing code licensed under these licenses when GPL-licensed code is also included. Therefore these encoders have been designated as "non-free", and you cannot download a pre-built ffmpeg that supports them. This can be resolved by compiling ffmpegyourself.
See also GuidelinesHighQualityAudio for general guidelines on FFmpeg audio encoding.
The Fraunhofer FDK AAC codec library. This is currently the highest-quality AAC encoder available with ffmpeg. Requires ffmpeg to be configured with --enable-libfdk_aac (and additionally --enable-nonfree if you're also using --enable-gpl). But beware, it defaults to a low-pass filter of around 14kHz (details). If you want to preserve higher frequencies, use-cutoff 18000. Adjust the number to the upper frequency limit you prefer.
Detailed information about the FDK AAC library (not FFmpeg specific) can be found at the Fraunhofer FDK AAC page in the HydrogenAudio? Knowledgebase.
These settings target a specific bit rate, with less variation between samples. It gives you greater control over file size, and it is compatible with the HE-AAC profile. As a rule of thumb, for audible transparency, use 64kb/s for each channel (so 128kb/s for stereo, 384 kb/s for 5.1 surround sound). Set the bit rate with the -b:a flag.
Convert and audio file to AAC in an M4A (MP4) container:
ffmpeg -i input.wav -c:a libfdk_aac -b:a 128k output.m4a
Convert 5.1 surround sound audio of a video, leaving the video alone:
ffmpeg -i input.mp4 -c:v copy -c:a libfdk_aac -b:a 384k output.mp4
Convert the video with libx264, with a target of fitting a 90-minute movie on a 700MB(=5734400kb) CD-ROM, mixing the audio down to two channels (Windows users should use NULrather than /dev/null):
ffmpeg -y -i input.mp4 -c:v libx264 -b:v 933k -preset:v veryfast -pass 1 -an /dev/null && ffmpeg -i input.mp4 -c:v libx264 -b:v 933k -preset:v veryfast -pass 2 -ac 2 -c:a libfdk_aac -b:a 128k output.mp4
VBR encoding in libfdk-aac is experimental and only works with some parameter combinations (details). Target a quality, rather than a specific bit rate. 1 is lowest quality and 5 is highest quality. This mode is not compatible with AAC-HE. Set the VBR level with the -vbr flag.
VBR modes give roughly the following bit rates per channel (details):
| VBR | kbps/channel |
| 1 | 32 |
| 2 | 40 |
| 3 | 48-56 |
| 4 | 64 |
| 5 | 80-96 |
Convert an audio file to AAC in an M4A (MP4) container:
ffmpeg -i input.wav -c:a libfdk_aac -vbr 3 output.m4a
Convert the audio only of a video:
ffmpeg -i input.mp4 -c:v copy -c:a libfdk_aac -vbr 3 output.mp4
Convert the video with libx264, and mix down audio to two channels:
ffmpeg -i input.mp4 -c:v libx264 -crf:v 22 -preset:v veryfast -ac 2 -c:a libfdk_aac -vbr 3 output.mp4
This is a pair of AAC profiles tailored for low bit rates (version 1 and version 2). VBR mode is not compatible with AAC-HE so a bitrate is declared. AAC-HE version 1 is suited for bit rates below 64kb/s (for stereo audio) down to about 48 kb/s, while AAC-HE version 2 is suited for bit rates as low as 32 kb/s (again, for stereo).
NOTE: AAC-HE version 2 only handles stereo. If you have mono, or want to down-mix to mono, use AAC-HE version 1.
Unfortunately, many devices that can play AAC-LC (the default profile for libfdk_aac) simply cannot play either version of AAC-HE, so this is not recommended for surround sound audio, which normally needs to be compatible with such hardware players. If you are only going to play it on your computer, or you are sure that your hardware player supports AAC-HE, you can aim for a bit rate of 160kb/s for version 1, or 128kb/s for version 2. As always, experiment to see what works for your ears.
ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a
ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he_v2 -b:a 32k output.m4a
By default when encoding AAC files using libfdk_aac the metadata ('moov' atom) is written after the audio stream ('mdat' atom) at the end of the file. In order to enable streaming of the encoded file the 'moov' atom has to be moved before the 'mdat' atom. In addition some AAC player implementations have issues decoding such files.
FFmpeg offers the option '-movflags +faststart' covering that functionality which can be used during encoding:
ffmpeg -i input.wav -c:a libfdk_aac -movflags +faststart output.m4a
Existing m4a files can be modified with the "qt-faststart' program which is distributed with FFmpeg in the 'tools' directory
qt-faststart input.m4a output.m4a
The native FFmpeg AAC encoder is included with ffmpeg and is does not require an external library like the other AAC encoders described here. Note that you will not get as good results as with libfdk_aac. This is considered an experimental encoder, so -strict experimental or -strict -2 is required.
ffmpeg -i input.wav -strict experimental -c:a aac -b:a 240k output.m4a
Note: -cutoff 15000 was previously recommended for this encoder, but this is now automatically applied since 2012 July.
ffmpeg -i input.wav -strict experimental -c:a aac -q:a 6 output.m4a
Effective range for -q:a is around 0.1-10. This VBR is experimental and likely to get even worse results than the CBR. If ffmpeg ignores -q:a then get a newer build (see ticket #1346).
VisualOn AAC encoding library. Requires ffmpeg configuration with --enable-libvo-aacenc. This has the advantage of not being non-free, and is included by some distributors, but is a rather poor encoder compared to libfdk_aac and even the native FFmpeg AAC encoder according to Quality Assessment of FFmpeg AAC and a mailing list post. This encoder supports up to 2 channels, is CBR only, and does not work with -q:a/-qscale:a.
ffmpeg -i input.wav -c:a libvo_aacenc -b:a 128k output.m4a
Freeware Advanced Audio Coder. Requires ffmpeg configuration with --enable-libfaac --enable-nonfree. Note that you will not get as good results as with libfdk_aac.
ffmpeg -i input.wav -c:a libfaac -q:a 100 output.m4a
Range for -q:a is 10-500 and is similar to using the -q option in standalone faac. 100 is a good value to try.
ffmpeg -i input.wav -c:a libfaac -b:a 192k output.m4a
libfaac does not support a true Constant Bit Rate (CBR) mode.
You can add metadata to any of the examples on this guide:
ffmpeg -i input ... -metadata author="FFmpeg Bayou Jug Band" -metadata title="Decode my Heart (Let's Mux)" output.mp4
For AAC-LC the likely answer is: libfdk_aac > libfaac > Native FFmpeg AAC encoder (aac) > libvo_aacenc.
If you require a low audio bitrate, such as ≤ 32kbs/channel, then AAC-HE would be worth considering if your player or device can support AAC-HE decoding. Anything higher may benefit more from AAC-LC due to less processing. If in doubt use AAC-LC. All players supporting AAC-HE also support AAC-LC.
Some encoders, such as the native FFmpeg AAC encoder (aac), are considered experimental and require the addition of -strict experimental or -strict -2 (same thing, different name) to your command as an output option. Otherwise you may see:
The encoder 'aac' is experimental but experimental codecs are not enabled, add '-strict -2' if you want to use it.
So, for example, given a command line such as:
ffmpeg -i file.avi file.mov
you may need to invoke ffmpeg as follows for the experimental codec to be enabled:
ffmpeg -i file.avi -strict -2 file.mov
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