Abstract: An input speech signal to an input terminal is supplied to a speech synthesizer section through a speech analyzer section and frequency parameter quantizer section to form a synthesis filter, and the input speech signal is expressed by quantized LPC coefficients representing the characteristics of the synthesis filter and an excitation signal for exciting the synthesis filter. In this case, in a pulse excitation section, a pulse position selector selects pulse position candidates from the integer pulse positions and non-integer pulse positions stored in a pulse position codebook, and an integer position pulse generator and non-integer position pulse generator respectively generate integer position pulses set at sampling points of the excitation signal and non-integer position pulses set at positions located between sampling points. These pulses are synthesized into a pulse train serving as a source of an excitation signal.

Abstract: An input speech signal to an input terminal is supplied to a speech synthesizer section through a speech analyzer section and frequency parameter quantizer section to form a synthesis filter, and the input speech signal is expressed by quantized LPC coefficients representing the characteristics of the synthesis filter and an excitation signal for exciting the synthesis filter. In this case, in a pulse excitation section, a pulse position selector selects pulse position candidates from the integer pulse positions and non-integer pulse positions stored in a pulse position codebook, and an integer position pulse generator and non-integer position pulse generator respectively generate integer position pulses set at sampling points of the excitation signal and non-integer position pulses set at positions located between sampling points. These pulses are synthesized into a pulse train serving as a source of an excitation signal.

Abstract: A sound signal processing method includes emphasizing a first sound signal based on a plurality of sound signals produced by a plurality of microphones arranged at intervals, determining a frequency by an arrival direction of a second sound signal other than the first sound signal and the interval between the microphones, and removing a frequency band including the frequency determined, from the first sound signal emphasized.

Abstract: A method for encoding speech wherein an input speech signal is separated by a component separator into a first component mainly constituted by speech and a second component mainly constituted by a background noise at each predetermined unit of time, a bit allocation selector selects bit allocation for each component based on the first and second components from among a plurality of predetermined candidates for bit allocation, a speech encoder and a noise encoder encode the first and second components from the component separator based on the bit allocation according to predetermined different methods for encoding, and a multiplexer multiplexes encoded data of the first and second components and information on the bit allocation and outputs them as transmitted encoded data.

Abstract: A speech encoding method in which information representing characteristics of a synthesis filter is generated based on an input speech signal in units of one frame. A pitch vector is generated from an adaptive codebook containing past excitation signals, and a first number of reduced pulse position candidates are generated by selecting a first number of pulse positions from a number of possible pulse positions in each of sub-frames obtained by dividing the frame, where a density of the reduced pulse position candidates is high where the pitch vector has a large power and decreases in accordance with a decrease in the power.

Abstract: A constraint relieving section receives an input vector and transforms it with a predetermined transform function to generate a target vector obtained by subtracting a constraint vector representing a predetermined constraint from the input vector. An error evaluation section selects from a codebook a code vector constituting a quantization vector having a minimum error with respect to the target vector and outputs an index representing this code vector.

Abstract: A speech encoding method includes generating information representing characteristics of a synthesis filter, and generating an excitation signal for exciting the synthesis filter, the excitation signal including a pulse train generated by setting one or more pulses at a predetermined number of pulse positions selected from a plurality of pulse position candidates adaptively changed in accordance with the characteristics of the speech signal. A speech decoding method includes inputting the excitation signal to a synthesis filter for reconstructing a speech signal.

Abstract: A method for encoding speech wherein an input speech signal is separated by a component separator into a first component mainly constituted by speech and a second component mainly constituted by a background noise at each predetermined unit of time, a bit allocation selector selects bit allocation for each component based on the first and second components from among a plurality of predetermined candidates for bit allocation, a speech encoder and a noise encoder encode the first and second components from the component separator based on the bit allocation according to predetermined different methods for encoding, and a multiplexer multiplexes encoded data of the first and second components and information on the bit allocation and outputs them as transmitted encoded data.

Abstract: A speech encoding method including generating a reconstruction speech vector by using a code vector extracted from a codebook storing a plurality of code vectors for encoding a speech signal. In addition an input speech signal to be encoded is used as a target vector to generate an error vector representing the error of the reconstruction speech vector with respect to the target vector, and the error vector is passed through a perceptual weighting filter having a transfer function including the inverse characteristics of the transfer function of a filter for emphasizing the spectrum of a reconstructed speech signal. Thus a weighted error vector is generated, the codebook for a code vector that minimizes the weighted error vector is searched, and an index corresponding to the code vector found as an encoding parameter is output.

Abstract: The coding apparatus comprises an adaptive codebook storing excitation signals as vectors, a synthesis filter for forming a synthesis signal, referring to the vectors stored in the adaptive codebook, a similarity computation circuit for computing a similarity between the synthesis signal obtained by the synthesis filter and a target signal, and a coding scheme determining circuit for deciding one coding scheme from a plurality of coding schemes respectively having coding bit rates different from each other, on the basis of the similarity obtained by the similarity computation circuit.

Abstract: Adjusting the shape of a spectrum of a speech signal includes the steps of using a first filter with pole-zero transfer function A(z)/B(z) for subjecting a speech signal to a spectrum envelop emphasis and a second filter cascade-connected with the first filter, for compensating for a spectral tilt due to the first filter, independently deriving two filter coefficients used in the second filter for compensating for the spectral tilt from the pole-zero transfer function, and compensating for the spectral tilt corresponding to the pole-zero transfer function according to the derived filter coefficients.

Abstract: A speech encoding method and apparatus including analyzing, using a codebook expressing speech parameters within a predetermined search range, an input speech signal in an audibility weighting filter corresponding to a pitch period longer than the search range of the codebook, and searching, from the codebook, on the basis of the analysis result, a combination of speech parameters by which the distortion of the input speech signal is minimized, and encoding the combination. The apparatus uses an adaptive codebook of pitch and a noise codebook. The codebooks search a group formed by extracting vectors of predetermined length from one original code vector, while sequentially shifting position so that the vectors overlap each other. The search group is further restricted and another preselection is made before the final search. Search is based on inversely convoluted, orthogonally transformed vectors.

Abstract: In a vector quantization apparatus for expressing a target vector by using a code vector designated by an index, an error evaluating section performs error evaluation for a code vector without considering a code error of the index and error evaluation with considering the code error, a first selecting section selects a small number of indexes from a larger number of indexes on the basis of an evaluation result without considering the code error, and a second selecting section selects, on the basis of an evaluation result with considering the code error, an index used to express the target vector from a small number of indexes selected by the first selecting section.

Abstract: A vector quantizing apparatus includes a first search section for obtaining an approximate vector X1 which is approximated to a desired vector R, a residual vector calculator for calculating a residual vector Rv from the desired vector R and the approximate vector X1, a weighting section for obtaining weighted vectors X2 to XN of code vectors x2 to xN, and a second search section for calculating an estimation value which is the magnitude of a projection vector of the residual vector Rv with respect to the vector space formed by the approximate vector X1 and the weighted vectors X2 to XN, and searching a code vector which maximizes this estimation value.