Abstract: A noise suppress processing apparatus has a speech input section for detecting speech uttered by the speaker at different positions, an analyzer section for obtaining frequency components in units of channels by frequency-analyzing speech signals in units of speech detecting positions, a first beam former processor section for obtaining target speech components by suppressing noise in the speaker direction by filtering the frequency components in units of channels using filter coefficients, which are calculated to decrease the sensitivity levels in directions other than a desired direction, a second beam former processor section for obtaining noise components by suppressing the speech of the speaker by filtering the frequency components for the plural channels obtained by the analyzer section to set low sensitivity levels in directions other than a desired direction, an estimating section for estimating the noise direction from the filter coefficients of the first beam former processor section, and estimating

Abstract: In a synthesis unit generator, a plurality of synthesis speech segments are generated by synthesizing training speech segments labeled with phonetic contexts and input speech segments while altering the pitch/duration of the input speech segments in accordance with the pitch/duration of the training speech segments. Typical speech segments are selected from the input speech segments on the basis of a distance between the synthesis speech segments and the training speech segments, and are stored in a storage. In addition, a plurality of phonetic context clusters corresponding to the synthesis units are generated on the basis of the distance, and are stored in a storage. A synthesis speech signal is generated by reading out, from the storage, those of the synthesis units, which correspond to the phonetic context clusters including phonetic contexts of input phonemes, and connecting the selected synthesis units in a speech synthesizer.

Abstract: A speech information processing apparatus previously stores a plurality of representative patterns corresponding to each cluster to which prosody unit belongs. A clustering section classifies a plurality of prosody units in speech data to each cluster according to attribute data of the prosody unit. An extraction section extracts pitch pattern corresponding to the prosody unit classified to each cluster from the speech data. A transformation parameter generation section generates a transformation parameter by evaluating error between the pitch pattern and transformed representative pattern by unit of the cluster. A representative pattern generation section updately generates the representative pattern by calculating an evaluation function of the pitch pattern and the transformation parameter by unit of the cluster.

Abstract: In a synthesis unit generator, a plurality of synthesis speech segments are generated by synthesizing training speech segments labeled with phonetic contexts and input speech segments while altering the pitch/duration of the input speech segments in accordance with the pitch/duration of the training speech segments. Typical speech segments are selected from the input speech segments on the basis of a distance between the synthesis speech segments and the training speech segments, and are stored in a storage. In addition, a plurality of phonetic context clusters corresponding to the synthesis units are generated on the basis of the distance, and are stored in a storage. A synthesis speech signal is generated by reading out, from the storage, those of the synthesis units, which correspond to the phonetic context clusters including phonetic contexts of input phonemes, and connecting the selected synthesis units in a speech synthesizer.

Abstract: In a background noise/speech classification method, whether a digital input signal input through an input terminal is background noise or speech is decided by a background noise/speech decision section on the basis of calculated frame power and a calculated LSP coefficient which are obtained by supplying the input signal to a feature amount calculation section and. estimated frame power and an estimated LSP coefficient obtained by an estimated feature amount update section. Thereafter, the estimated feature amount update section updates the estimated frame power and the estimated LSP coefficient by using the frame power and the LSP coefficient obtained by the feature amount calculation section to prepare for the next frame.

Abstract: In a background noise/speech classification method, whether a digital input signal input through an input terminal is background noise or speech is decided by a background noise/speech decision section on the basis of calculated frame power and a calculated LSP coefficient which are obtained by supplying the input signal to a feature amount calculation section and estimated frame power and an estimated LSP coefficient obtained by an estimated feature amount update section. Thereafter, the estimated feature amount update section updates the estimated frame power and the estimated LSP coefficient by using the frame power and the LSP coefficient obtained by the feature amount calculation section to prepare for the next frame.

Abstract: A speech synthesis apparatus synthesize a speech signal by filtering a speech source signal through a synthesis filter. A speech source signal codebook stores a plurality of speech source signals as a code vector. A unit dictionary memory stores a plurality of synthesis units corresponding to phonemic symbols, each synthesis unit comprising an index of the code vector in the speech source codebook and a shift number for the code vector to decode the speech source signal. A unit selection section selects a synthesis unit corresponding to phonemic symbols to be synthesized from the unit dictionary memory. A synthesis unit decoder selects the code vector corresponding to the index in the synthesis unit from the speech source signal codebook, and shifts the code vector according to the shift number in the synthesis unit.

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 recognition synthesis based encoding/decoding method recognizes phonetic segments, syllables, words or the like as character information from an input speech signal and detects pitch periods, phoneme or syllable durations or the like, as information for prosody generation, from the input speech signal, transfers or stores the character information and information for prosody generation as code data, decodes the transferred or stored code data to acquire the character information and information for prosody generation, and synthesizes the acquired character information and information for prosody generation to obtain a speech signal.

Abstract: In a formant emphasis method of emphasizing the formant as the spectral peak of an input speech signal and attenuating the spectral valley of the input speech signal, a spectrum emphasis filter performs processing for emphasizing the formant of the input speech signal and attenuating the valley of the input speech signal. A first-order variable characteristic filter whose characteristic adaptively changes in accordance with the characteristic of the input speech signal and a first-order fixed characteristic filter compensate a spectral tilt included in an output signal from the spectrum emphasis filter.

Abstract: A preprocessing device used in a video coding apparatus, comprising a motion compensation prediction estimating unit for detecting a change of an video from a current picture and a past picture to generate change data, and a filter for deforming the current picture in accordance with the change data generated by the motion compensation prediction estimating circuit such that a deformed current picture is sent to a motion compensation prediction coding section of the video coding apparatus to be coded.

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: A speech synthesis apparatus includes; a memory for storing a plurality of typical waveforms corresponding to a plurality of frames, the typical waveforms each previously obtained by extracting in units of at least one frame from a prediction error signal formed in predetermined units, a voiced speech source generator including an interpolation circuit for performing interpolation between the typical waveforms read out from the memory means to obtain a plurality of interpolation signals each having at least one of an interpolation pitch period and a signal level which changes smoothly between the corresponding frames, a superposition circuit for superposing the interpolation signals obtained by the interpolation circuit to form a voiced speech source signal, an unvoiced speech source generator for generating an unvoiced speech source signal, and a vocal tract filter selectively driven by the voiced speech source signal outputted from the voiced speech source generator and the unvoiced speech source signal fro

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: A preprocessing device used in a video coding apparatus, comprising a motion compensation prediction estimating unit for detecting a change of an video from a current picture and a past picture to generate change data, and a filter for deforming the current picture in accordance with the change data generated by the motion compensation prediction estimating circuit such that a deformed current picture is sent to a motion compensation prediction coding section of the video coding apparatus to be coded.

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.

Abstract: This invention provides a novel speech coding system which recursively executes a filter-applied "Toeplitz characteristic" by causing a drive signal (i.e., an excitation signal) to be converted into a "Toeplitz matrix" when detecting a pitch period in which distortion of the input vector and the vector subsequent to the application of filter-applied computation to the drive signal vector in the pitch forecast called either "closed loop" or "compatible code book" is minimized. The vector quantization method substantially making up the speech coding system of the invention is characteristically used by the system.

Abstract: A speech signal is input to an excitation signal generating section, a prediction filter and a prediction parameter calculator. The prediction parameter calculator calculates a predetermined number of prediction parameters (LPC parameter or reflection coefficient) by an autocorrelation method or covariance method, and supplies the acquired prediction parameters to a prediction parameter coder. The codes of the prediction parameters are sent to a decoder and a multiplexer. The decoder sends decoded values of the codes of the prediction parameters to the prediction filter and the excitation signal generating section. The prediction filter calculates a prediction residual signal, which is the difference between the input speech signal and the decoded prediction parameter, and sends it to the excitation signal generating section.