How can I convert to AAC?
This is the most basic FFmpeg command to convert input to AAC audio using the highest quality AAC FFmpeg has.
libfdk_aac is the Fraunhofer AAC encoder, and it is available when you compiled FFmpeg with support for it.
ffmpeg -i input.mp3 -c:a libfdk_aac output.m4a
To change the quality, you have two options:
For variable bitrate (VBR), use the
-vbr option. For example,
-vbr 4 is a good choice (roughly 128 kBit/s for stereo audio). Higher means better. Values range from 1 to 5.
ffmpeg -i input.mp3 -c:a libfdk_aac -vbr 4 output.m4a
For fixed bitrate (CBR), use the
-b:a option, for example
-b:a 128k. 128 kBit/s should be good enough for most situations. You can often choose something lower as well.
ffmpeg -i input.mp3 -c:a libfdk_aac -b:a 128k output.m4a
You can also use HE-AAC (v1/v2) instead of the default AAC-LC (low complexity) profile, which is best suited for low bitrates:
ffmpeg -i input.mp3 -c:a libfdk_aac -profile:a aac_he_v2 -b:a 32k output.m4a
What if I don't have
Some versions of FFmpeg do not have
libfdk_aac (for licensing reasons). If that's the case, you can use the built-in AAC encoder:
ffmpeg -i input.mp3 -c:a aac -b:a 192k output.m4a
This encoder does not support VBR properly, so if you need that or HE-AAC, read on.
You can compile ffmpeg yourself and add
libfdk_aac support. There are guides on the FFmpeg Wiki. Compiling is easy on Linux, moderately easy on OS X, and rather hard on Windows. When you follow the compilation guides and install the appropriate libraries before, FFmpeg now gives you the option to use
There were some problems with your original approach:
alac is not AAC. ALAC is the Apple Lossless Audio Codec, whereas AAC is Advanced Audio Coding.
That's why your output is larger than the input, because in contrast to MP3, ALAC is still compressed, but it needs to be lossless – that's why it needs to store more data.
.aac is not an output container for ALAC audio. If you use AAC, that should work. I would use MP4 or M4A though.