This paper will describes a digital speech signal processor(DSSP) designed for a narrow band and a medium band speech coding system, both of them are recently developed at the E.C.L., N.T.T.. The narrow band system is LSP(Line Spectrum Pair) vocoder, which is based on the frequency domain representation of LPC parameters. The LSP parameters are superior to conventional LPC parameters in view of its quantization and interpolation properties. The medium band system is a split band adaptive predictive coding with adaptive bit-allocation, APC-AB. This system can achieve a toll quality coding at 16 Kbit/s. We have made an extensive software simulation of LSP and APC-AB, and decided the specification of the DSSP. The DSSP can handle arithmetic and shifting operations in both 12 and 24 bit mode. Multiplications are done by a 12- by-12 parallel multiplier. Multiplier, ALU and shifters can operate concurrently in a pipeline mode at the minimum interval of 0.5 μs per multiplication-accumulation.
|ジャーナル||ICASSP, IEEE International Conference on Acoustics, Speech and Signal Processing - Proceedings|
|出版ステータス||Published - 1982|
|イベント||1982 IEEE International Conference on Acoustics, Speech, and Signal Processing, ICASSP 1982 - Paris, France|
継続期間: 1982 5 3 → 1982 5 5
ASJC Scopus subject areas
- Signal Processing
- Electrical and Electronic Engineering