Louis Dunn Fielder

High-quality audio transform coding at 128 kbits/s

An approach to wideband digital audio compression of CD-quality signals at data rates of 128 kb/s channel and below is presented. A form of adaptive transform coding, this technique features a nonuniform frequency division and coding scheme to exploit known characteristics of human perception. The algorithm has low computational complexity and can be adapted for use at other bit rates. A windowed overlap-add process is used with the forward/inverse transforms, which have been efficiently implemented using FFTs. Transform coefficients are converted into a subband block-companded format consisting of exponent words and associated mantissas, which are then coded with an adaptive quantizer. A real-time, single-chip programmable digital signal processing (DSP) implementation encodes 480-kHz-sampled stereo audio signals at a variety of bit rates. At 128 kb/s, the coders subjective performance is appropriate for highest-quality 15-kHz professional audio applications.<<ETX>>

ATSC Video and Audio Coding

In recent decades, digital video and audio coding technologies have helped revolutionize the ways we create, deliver, and consume audiovisual content. This is exemplified by digital television (DTV), which is emerging as a captivating new program and data broadcasting service. This paper provides an overview of the video and audio coding subsystems of the Advanced Television Systems Committee (ATSC) DTV standard. We first review the motivation for data compression in digital broadcasting. The MPEG-2 video and AC-3 audio compression algorithms are described, with emphasis on basic concepts, system features, and coding performance. Next-generation video and audio codecs currently under consideration for advanced services are also presented.