AAC and MP3 audio quality
The AAC format is designed to enhance MP3 in the following ways:
More sample rates (8 kHz to 96 kHz) than MP3 (16 kHz to 48 kHz)
Up to 48 channels (MP3 supports up to two channels in MPEG-1 mode and up to 5.1 channels in MPEG-2 mode)
Arbitrary bit rate and variable frame length. Constant bit rate normalized with bit bank.
More efficient and simpler filter banks (AAC uses pure MDCT instead of hybrid MP3 encoding)
More efficient encoding for still signals (AAC uses a block size of 1024 or 960 samples, which is more efficient than MP3’s 576-sample block)
Higher encoding accuracy for transient signals (AAC uses a 128- or 120-sample block size, allowing more accurate encoding than MP3’s 192-sample block)
Spectral leakage can be removed using a Kaiser-Bessel derived window function, but at the expense of expanding the main lobe
Better audio handling above 16 kHz
More flexible joint stereo (different methods can be used in different frequency ranges)
Added additional modules (tools) to improve compression efficiency: TNS, backward prediction, PNS, etc. These modules can be combined to form different encoding profiles.
In general, the AAC format gives developers more flexibility in codec design than MP3 and can correct many of the design choices made in the original MPEG-1 audio specification. This greater flexibility generally results in more concurrent encoding strategies and thus more efficient compression.
The MP3 specification, although outdated, has proven to be surprisingly robust despite its many flaws. AAC and HE-AAC outperform MP3 at low bit rates (typically less than 128 kb/s). This is especially true at extremely low bit rates, where excellent stereo encoding, pure MDCT, and more ideal conversion window sizes prevent MP3 from competing. However, as the bitrate increases, the efficiency of the audio format becomes less important relative to the efficiency achieved by the encoder, and the inherent advantages of AAC over MP3 no longer dominate.
