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.
|ジャーナル||ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings|
|出版ステータス||Published - 1989 12月 1|
|イベント||1989 International Conference on Acoustics, Speech, and Signal Processing - Glasgow, Scotland|
継続期間: 1989 5月 23 → 1989 5月 26
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