MP3
MPEG audio gives a set of standards to lossy audio compression. Algorithms are classified in three layers sorted by complexity and efficiency. They are contained both in MPEG-1 (MPEG1) and MPEG-2 (MPEG2). These standards allow to work on high (32, 44.1, 48 kHz) and low sampling frequencies (16, 22.05, 24 kHz) respectively.
Usually MP3 codecs use a non-uniform quantization on frequency domain driven by perceptual model to compress PCM audio signal into a standard bit stream at various bit rate values.
The time to frequency transform is built by means of a polyphase filter bank and by cascading it with MDCT (hybrid filter bank). Polyphase filter bank gets samples from PCM streaming and represents them, for long window, in 32 frequency sub-bands, further subdivided into 18 finer sub-bands by MDCT.
Psychoacoustic model generates the SMR (Signal to Mask Ratio), this index tells to the quantization block about the bits number that should be allocated for each frequency sub-band in order to get an inaudible quantization noise (Zwicker,2001).
The output of filter banks and perceptual model are the input of non-uniform quantization process. This process decides how to quantize every frequency sub band to respect the SMR value. Huffman lossless compression is performed before bit stream packing. Although MPEG-2 layer 3 frame contains only one granule per frame, MPEG-1 Layer 3 frame is made by two granules.
Further information about MPEG standards can be found in (MPEG1; MPEG2; Noll,1997; Pan,1995).