Abstract: A first CN code (silence code) obtained by encoding a silence signal, which is contained in an input signal, by a silence compression function of a first speech encoding scheme is transcoded to a second CN code of a second speech encoding scheme without decoding the first CN code to a CN signal. For example, the first CN code is demultiplexed into a plurality of first element codes by a code demultiplexer, the first element codes are each transcoded to a plurality of second element codes that constitute the second CN code, and the second element codes obtained by this transcoding are multiplexed to output the second CN code.

Abstract: The invention enables a source speech signal to be coded in an optimal way. An adaptive codebook (Ba) stores a series of signal vectors of a past speech signal. A vector extraction means extracts a signal vector and neighboring vectors adjacent thereto. A high-order long-term prediction synthesis filter produces a long-term predicted speech signal (Sna−1) from the signal vector and its neighboring vectors. A filter coefficients calculation means calculates filter coefficients of the long-term prediction synthesis filter. A perceptual weighting synthesis filter obtains a reproduced coded speech signal (Sna) from the long-term predicted speech signal (Sna−1). An error calculation means calculates the error (En) of the reproduced coded speech signal (Sna) with reference to the speech signal (Sn). A minimum error detection means finds a minimum error point that yields the smallest error among the calculated errors.

Abstract: Disclosed is a voice encoding method having a synthesis filter implemented using linear prediction coefficients obtained by dividing an input signal into frames each of a fixed length, and subjecting the input signal to linear prediction analysis in the frame units, generating a reconstructed signal by driving said synthesis filter by a periodicity signal output from an adaptive codebook and a pulsed signal output from an algebraic codebook, and performing encoding in such a manner that an error between the input signal and said reproduced signal is minimized, wherein there are provided an encoding mode 1 that uses pitch lag obtained from an input signal of a present frame and an encoding mode 2 that uses pitch lag obtained from an input signal of a past frame. Encoding is performed in encoding mode 1 and encoding mode 2, the mode in which the input signal can be encoded more precisely is decided frame by frame and encoding is carried out on the basis of the mode decided.

Abstract: Disclosed is a voice code conversation apparatus to which voice code obtained by a first voice encoding method is input for converting this voice code to voice code of a second voice encoding method. The apparatus includes a code separating unit for separating, from the voice code based upon the first voice encoding method, codes of a plurality of components necessary to reconstruct a voice signal, code converters for dequantizing the codes of each of the components and then quantizing the dequantized values by the second voice encoding method to thereby generate codes, and a code multiplexer for multiplexing the codes output from respective ones of the code converters and transmitting voice code based upon the second voice encoding method.