Advanced Audio Coding (AAC)

What is Advanced Audio Coding (AAC)?
Advanced Audio Coding (AAC) is a technique for compressing and encoding digital audio files.

AAC technology can be used to encode audio files with medium to high bit rates. AAC is the logical successor to MP3 (ISO / MPEG Audio Layer -3) and offers better sound quality than its predecessor with the same bit rate.

The AAC technique involves taking advantage of two primary coding strategies to minimize the amount of data required to convey high quality digital audio. Signal components that are irrelevant are discarded. Redundancies in the coded audio signal are canceled.

Encoding digital audio files involves the following steps:

1.) The modified discrete cosine transform (MDCT) is used to convert the signal from the time domain to the frequency domain. Filter banks are used to convert an exact number of time samples into frequency samples.

2.) The frequency range is quantized with a psychoacoustic model and then coded.

3.) Appropriate internal error correction codes are applied.

4.) The signal is stacked or transmitted.

5.) The Luhn Mod N algorithm is used for each frame in order to avoid sample corruption

AAC can sample frequencies in the range from 8 Hz to 96 kHz and up to 48 channels. It is also capable of compressing audio that contains streams of complex pulses and square waves better than MP3.

AAC is an international standard used by some large companies such as Dolby Laboratories Inc., Sony Corp. and Nokia Corp. is used. AAC is also used as the standard audio codec for Apple's .m4v format in iTunes Store video files.

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