Linux.com: Using Advanced Audio Coding Under Linux
Jan 06, 2001, 13:07 (3 Talkback[s])
(Other stories by Mukund)
"...MP3 has had its time in the sun. In fact, more than its share of the sun. Do you know how old the MP3 format is? MP3 means MPEG-1 Audio, layer-3 compression, which is part of the ISO MPEG-1 suite, which was published in 1996. Lots of other encoding technologies have made their debut since then, some of which, such as Ogg Vorbis, are far better than MP3."
"Advanced Audio Coding (AAC) is one such audio coding format, which made its debut as a part of MPEG-2, and was revised and included in the MPEG-4 specification. AAC is probably the most advanced audio encoding system in the world. To quote MP3'Tech, "Advanced Audio Coding (AAC), also known as MPEG-2 NBC represents the actual state of the art in audio coding. It is able to include up to 48 audio channels, 15 low frequency enhancement channels, 15 embedded data streams and has multi-language capability. It also offers a better compression ratio than [MPEG-1] layer-3. MPEG formal listening tests have demonstrated it is able to provide slightly better audio quality at 96 kb/s than [MPEG-1] layer-3 at 128 kb/s or layer-2 at 192 kb/s."
"Freeware Advanced Audio Coder (FAAC) is an opensource GPL'd suite of tools which allow a person to use the AAC format. FAAC tools are available for both Linux and Microsoft Windows platforms. The FAAC project contains an AAC encoder and decoder, and plugins for XMMS and Winamp. The tools are easy and convenient to use, although they need more CPU power than tools for MP3. FAAC needs libsndfile to encode audio from existing formats such as WAV."
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