Abstract
A novel audio coding sheme is presented for the transmission of music signals with about 20 kHz bandwidth in asynchronous-transfer-mode-based networks. The coding algorithm is based on subband division and embedded ADPCM (adaptive differential pulse-code modulation) techniques in order to minimize the quality degradation by packet loss and to meet the various service grade demands on performance. The advantages of the proposed algorithm are investigated theoretically and compared to those of conventional embedded ADPCM. The SNR improvements are computed using actual radio signals. Rate control strategies are briefly discussed in case of packet loss.
| Original language | English |
|---|---|
| Pages (from-to) | 188-191 |
| Number of pages | 4 |
| Journal | ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings |
| Volume | 1 |
| Publication status | Published - 1989 Dec 1 |
| Event | 1989 International Conference on Acoustics, Speech, and Signal Processing - Glasgow, Scotland Duration: 1989 May 23 → 1989 May 26 |
ASJC Scopus subject areas
- Software
- Signal Processing
- Electrical and Electronic Engineering