Scalable and progressive audio codec

A source coding technique for variable, bandwidth-constrained channels such as the Internet must do two things: offer high quality at low data rates, and adapt gracefully to changes in available bandwidth. Here we propose an audio coding algorithm that is superior on both counts. It is inherently scalable, meaning that channel conditions can be matched without the need for additional computation. Moreover, it is compact: in subjective tests our algorithm, coded at 32 kb/s/channel, outperformed MPEG-1 Layer 3 (MP3) coded at 56 kb/s/channel (both at 44.1 kHz). We achieve this simultaneous increase in compression and scalability through use of a two-dimensional transform that concentrates relevant information into a small number of coefficients