Advanced audio coding
(AAC) is a technique used for compressing and encoding scheme digital audio
files.
AAC technology can be used for coding audio
files at medium to high bit rates. AAC is built to be the logical successor to
MP3 (ISO/MPEG Audio Layer -3) and claims to offer better sound quality than its
predecessor at the same bit rate.
AAC technique involves
exploiting two primary coding strategies in order to minimize the amount of
data required to impart high-quality digital audio. Signal components that are
irrelevant are discarded. Redundancies in the coded audio signal are wiped out.
The encoding of digital audio files involves the following steps:
The encoding of digital audio files involves the following steps:
- Modified discrete cosine
transform (MDCT) is used to convert the signal from time-domain to
frequency-domain. Filter banks are used to convert an exact number of time
samples into frequency samples.
- The frequency domain is
quantized using a psychoacoustic model and is then encoded.
- Appropriate internal error
correction codes are applied.
- The signal is stacked or
transmitted.
- Luhn mod N algorithm is used
for each frame to avoid sample corruption
AAC can sample frequencies ranging from 8Hz to
96kHz and up to 48 channels. It is also able to compress audio that contains
streams of complex pulses and square waves better than MP3.
AAC is an international standard used by some major companies like Dolby Laboratories Inc., Sony Corp. and Nokia Corp. AAC is also used as a default audio codec for .m4v format by Apple in iTunes Store video files.
AAC is an international standard used by some major companies like Dolby Laboratories Inc., Sony Corp. and Nokia Corp. AAC is also used as a default audio codec for .m4v format by Apple in iTunes Store video files.
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