Masayuki Nishiguchi

TL;DR: In this article, a cycle changing part 3 changes the output cycle of the encoded parameter by compressing or expanding the time base, and a decoding part 4 performs the interpolation processing for the encoded parameters outputted in the cycle changed by cycle changing parts 3 to calculate the changed encoded parameter corresponding to a disired time and outputs a synthesized voice signal from an output terminal 201 by synthesizing the voice signal based on the change encoded parameter.

TL;DR: In this paper, a speech codec is comprised of a r.m.s (root means square) value of an input speech signal inputted from an input terminal 1 and processed by filtering, a steady level calculating part 3 for calculating a steady-level of the r. m.s value, a division operator 4 for calculating the under-mentioned quotient rmsg by dividing the rm.ms output by the output min-rms of the steady level, and a fuzzy inference part 9 for outputting a decision flag decflag from a logarithmic amplitude difference

TL;DR: An audio-signal time-axis expansion/compression method for subjecting an audio signal to timeaxis expansion and compression at a time domain has been proposed in this paper.

TL;DR: In this article, the authors proposed to simply switch the bit rate by providing multiple coding sections for the vector quantization of the time base wave-form, reference-inputting the quantization error of the (N-1)th stage for the coding of the Nth stage, and selecting the quantisation output of each stage to switch bit rate.

TL;DR: In this article, the alpha-parameters are converted by an alpha -parameter to LSP converting circuit 13 into linear spectral pair (LSP) parameters and a vector of these LSP parameters is vector-quantized by a quantizer.