|Current AAC Encoders|
(Most to least recommended)
|1||Apple AACM W|
|2||Fraunhofer FDK AACS L M W|
|3||Nero AACL W|
|4||FFmpeg 3.0+ AAC encoderS L M W|
|5||FAACS L M W|
|6||Libav (pre-3.0 FFmpeg) AAC encoderS L M W|
S Source code available, L Linux, M Mac OSX, W Windows
|List of AAC encoders|
The multi-codec library, libavcodec, includes an AAC encoder and decoder. The library was created as part of FFmpeg, and forked with Libav. Both projects maintain a separate version of libavcodec.
The original native AAC encoder was written by Konstantin Shishkov, and released under version 2.1 of the LGPL. It was considered experimental and poor quality compared to non-free/commercial encoders, but at least free. For GSoC 2015, Claudio Freire and Rostislav Pehlivanova did a significant amount of work on the AAC encoder that would be included in FFmpeg 3.0 (February 2016), and FFmpeg has declared this encoder stable (at least for CBR) and ready for common use. Libav has not merged this new work, and continues to use the original experimental encoder.
The FFmpeg/Libav frameworks can also use other encoding libraries, if they are available.
- Fraunhofer FDK AAC, via libfdk-aac. Both frameworks recommend using this encoder if it is available, but it is non-free.
- FAAC, via libfaac.
- libvo-aacennc, the very poor VisualOn AAC encoder. (support removed in FFmpeg 3.0)
- libaacplus, the very old Coding Technologies HE-AAC[v2] encoder. (support removed in FFmpeg 3.0)
- FFmpeg 3.0+
- Supports sample rates for input of 7350, 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, and 96000 Hz.
- FFmpeg 3.0+
|4||C, L+R, B|
|5||C, L+R, BL+BR|
|5.1||C, L+R, BL+BR, LFE|
|7.1||C, FL+FR, SL+SR, BL+BR, LFE|
The plus sign (+) denotes paired "stereo" channels.
Gapless decoding support
- See Gapless playback for more on the general topic.
AAC encoding groups samples into frames of 512/1024/2048 samples. There may be padding of extra samples at the start, due to encoder delay, or at the end if there less than the exact number or samples needed to fill a frame. These extra samples will be decoded as additional silence if they are not trimmed. AAC provides no way to signal how many samples need to be trimmed, so this information is stored in the container, likely in the mp4 iTunSMPB atom.
- FFmpeg's decoder will trim from the beginning, but not the end
- Libav's decoder will not trim at all
Embedded album art
Album art embedding is not supported the same way it is for MP3. Use -vn if the source file contains album art, or it will try to convert it to MP4 video and fail.