Audio codec that outputs Advanced Audio Coding (AAC) data. AAC is one of the most used lossy audio compression formats. It is the successor to the MP3 compression format with improved encoding. The format is proprietary and requires the purchase of a license for commercial use.
More detailed information about the AAC format can be found, for example, at this link: Advanced Audio Coding
The filter is implemented on the basis of the CTransformFilter class and uses the libfaac library (Freeware Advanced Audio Coder library) for compression.
32- and 64-bit versions of the AAC codec are available.
The AAC codec accepts a 16-bit audio stream in PCM format as input.
The required number of channels (from one to 8), as well as the number of frames per second of the input stream (from 8000 to 96000) can be specified via the filter control interface or the property page.
|Number of channels||1, 2, 3, 4, 5, 6, 7, 8|
|Samples per second||96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000|
On the output, the AAC codec outputs a compressed AAC stream.
Compression parameters and additional characteristics of the output stream can be set using the control interface on the filter and the property page.
Limitations of the free version of the filter
Limitiation of the free version is the same as for other DirectShow filters – only one copy of the ACC codec can be created by a process.
|free x86 version|
|free x64 version|
|interface description file (aac.idl)|