Abstract: In a speech signal encoding system comprising a maximum similarity series extracting unit (50) for producing a series of excitation pulses appearing at an equidistant time interval and an identical amplitude, the maximum similarity series extracting unit sums up, as a waveform, a series of autocorrelation coefficients to produce a series of summation result coefficients. The autocorrelation coefficients correspond to polarized pulses which are equal to one another in pulse interval and pulse amplitude. The polarized pulses form a plurality of pulse sequences which have phases different from one another.

Abstract: In a multipulse processing device for achieving a high encoding efficiency without using a high sampling frequency for an input signal and with a great degree of freedom given to positions of multipulses, an input speech signal is subjected by an LPC analyzer/processor 3 to LPC analysis of each analysis frame for extraction of LPC coefficients after sampled by an A/D converter 2. Multipulses are retrieved as a result of decision by a multipulse analyzer 20 with a degree of freedom given relative to sampling points of a sampled speech signal supplied through an auditorily weighting filter 4. Encoded by an encoder 41 and together with k parameters used as an example of the LPC coefficients, retrieved multipulses are multiplexed by a multiplexer 42 for delivery to a synthesis side. A multipulse waveform synthesizer 45 synthesizes a waveform by using decoded multipulse data and the LPC coefficients.

Abstract: In a speech signal encoding system comprising an analyzer and a synthesizer, the analyzer is supplied with an input analog signal to preliminarily select a sequence of digital signals within an analysis frame, to extract a sequence of excitation pulses, from the analysis frame which is divided into a plurality of time intervals each of which is subdivided into a plurality of phases, correlations are calculated between autocorrelations of impulse responses within the analysis frame and cross correlations between the digital signals and the impulse responses to detect, by a maximum similarity series searching circuit, the sequence of the excitation pulses which has a maximum similarity between the autocorrelation coefficients and the cross correlations. The excitation pulses appear at an equidistant time interval and an identical amplitude and have a selected one of the phases and variable polarities.

Abstract: In an encoding device (100) operable in response to an input speech signal by means of an adaptive transform coding to produce an output encoded speech signal, the input speech signal is partitioned into data blocks by a partition circuit (113). Each of data blocks is decomposed into a plurality of frequency components by a Fourier transformer (114). A spectral envelope calculator (120) estimates intensity of a spectral envelope of the input speech signal. In cooperation with a scalar spectral calculator (115) and a bit assignment determiner (121), a quantizer (116) quantizes or encodes the frequency components with phase information selectively removed from a part of the frequency components on the basis of the intensity of the spectral envelope. In a decoding device, a phase information assignor assigns pseudo-phase information to each of the frequency components from which the phase information is selectively removed.

Abstract: A frequency bandwidth of a speech signal is divided into a plurality of partial bandwidths. Formant information is extracted on the basis of LPC information developed for the respective partial bandwidths. At least one partial bandwidth may overlap upon the preceding bandwidth. The boundary frequencies of the partial bandwidths can be determined based on the frequency envelope of the speech signal.

Abstract: In a modulated signal transmission system operable in response to an input digital signal of a baseband to produce a modulated signal through a nonlinear amplifier, a nonlinear converter is controlled by a control signal generator to compensate for not only a nonlinear distortion but also a linear distortion. The control signal generator is given a difference signal between a linearly detected signal and the input digital signal to which an estimated linear distortion is added by a linear filter having filter coefficients controlled by a filter coefficient estimator. An additional nonlinear converter is located prior to the linear filter. Alternatively, the linear filter may be located after the nonlinear converter.

Abstract: A speech signal communication apparatus used in a confidential communication system, which performs blockwise processing to achieve a high level of confidentiality, free from waveform discontinuity at the block boundary in a restored speech signal. This apparatus includes a calculating unit for calculating linear predictive coefficients of the input speech signal and a filter to inversely filter the speech signal by using the linear predictive coefficients. The filter flattens the spectral envelope of the input speech signal and produces a predictive residual signal therefrom. This apparatus also includes a removing unit for adaptively removing a low power frequency component from the frequency components of the predictive residual signal. The removing unit uses the linear predictive coefficients to determine which frequency components are to be removed. The linearly predictive coefficients are then converted into a signal having the removed frequency components therein.

Abstract: Speech signal encoding at a low bit rate is realized by decreasing the number of arithmetic operations required to determine LPC parameters in a backward processing unit. These parameters are determined and updated by filtering, rather than by impulse response. In this manner, it is not necessary to use previously-calculated parameters in updating.

Abstract: In a data transmission system where a first signal partially leaks as an echo from a first transmission line to a second transmission line through a hybrid circuit to form a mixed signal of the echo and a second signal on the second transmission line, an echo canceller is used for cancelling the echo by producing an echo replica at a transversal filter according to filter coefficients and subtracting the echo replica from the mixed signal. In order to reliably generate the filter coefficient for a reduced time duration, a series of autocorrelation coefficients of the first signal and a series of cross-correlation coefficients between the first and the mixed signals are calculated at calculators and the filter coefficients are determined from both of the autocorrelation and the cross-correlation coefficient series at a coefficient determining circuit. The coefficient determining circuit may be an arithmetic circuit for solving simultaneous linear equations.

Abstract: A speech processing system such as a variable frame length type vocoder and a pattern matching vocoder of the same type capable of improving the reproduced speech. Representative frames replacing a plurality of frames in a given section are developed from among the frames in the given frame, or the frames in the given frame and the final representative frame developed in the preceding section. First frames to be replaced by the representative frames, and second frames, located between the neighboring different representative frames, which are to be approximated by interpolation between the neighboring different representative frames, are determined under the condition the lengths of the first and second frames be variable.

Abstract: A linear predictive speech synthesis apparatus which receives a pitch parameter and linear predictive coefficients. The synthesis apparatus includes a device for producing an exciting source signal in response to the pitch parameter, and a device for filtering the exciting source signal in response to the linear predictive coefficients to produce a synthesized speech signal. A cascade frequency characteristic between spectral envelope frequency characteristics of the producing device and the filtering device is designated such as to correspond to a spectral envelope characteristic of an input speech signal.

Abstract: A pattern matching vocoder includes first and second reference pattern memories, a pattern matching processor, and a frame selector. The first pattern memory stores reference vector patterns clustered by a distribution of the number of times of occurrence for spectral envelope vectors of an input speech signal. The second reference pattern memory stores reference vector patterns clustered by pole frequencies, pole bandwidths and a bandwidth of the input speech signal. The pattern matching processor divides the bandwidth of the speech signal into frequency regions and performs pattern matching using, as spectral envelope vectors, power ratios between the frequency regions. The frame selector performs frame selection using, as an evaluation value, a total distortion consisting of a vector distortion caused by pattern matching and a time distortion caused by frame selection with a DP (Dynamic Programming) scheme.

Abstract: An input speech signal is converted into sampled data with a first sampling frequency in each of a plurality of analysis frames. The sampled data are produced as filtered data through a digital filter having a high cut-off frequency smaller than the highest frequency of the speech signal. The filtered data are decimated into decimated signals which are sampled at a second sampling frequency smaller than the first sampling frequency and which are used to develop multi-pulses representative of an exciting source information of the input speech signal. Each of the analysis frames is divided into a plurality of subframes. At most one multi-pulse is developed in one subframe and the other multi-pulses are subsequently developed for the subframes other than the subframe where the one multi-pulse has been developed.

Abstract: The system utilizes a linear predictive coding (LPC) analyzer, an Attenuator, a line spectrum pair (LSP) analyzer, a reference pattern memory and a pattern matching device. The LPC analyzer derives LPC parameters from an input speech signal. The LPC parameters are attenuated in the attenuator and fed to the LSP analyzer for deriving LSP parameters which are in turn fed to the pattern matching device. The reference pattern memory stores a plurality of reference patterns composed of a sequence of LSP parameters for a variety of predetermined speech samples. The pattern matching device is connected to the LSP analyzer and the reference pattern memory to select the reference pattern which most closely resembles the input pattern from the LSP analyzer and to provide a label code as an output thereof. On the decoding side, a decoder is responsive to the label for generating LPC parameters corresponding to the reference pattern of the label.

Abstract: From a speech signal, spectrum information as a plurality of line spectrum data, pitch position data and amplitude data are extracted. Each of the sinusoidal wave signals of different frequencies is allotted to the predetermined line spectrum data. The frequency of the sinusoidal wave signal is changed with the pitch position being the boundary. The plurality of sinusoidal wave signals are added and the added result is modulated by the amplitude data to transmit the modulated signal as the transmission data. The line spectrum data, the pitch position data and amplitude data are extracted from the modulated signal. The replica of the speech is produced on the basis of these extracted data.

Abstract: A vocoder for matching an input speech signal with a reference speech signal on the basis of mutual angular data developed through spherical coordinate conversion of a plurality of formant frequencies obtained from the input and reference speech signals.

Abstract: A digital speech signal sampled at a predetermined interval is stored in a memory. An LPC (Linear Prediction Coefficient) DDK is developed from the speech signal and thus developed LPC coefficient specifies coefficients of a recursive filter. The speech signal read out from the memory is backwardly supplied to the recursive filter in the reverse order to the sampling order of the speech signal. A plurality of multi-pulses are determined on the basis of the crosscorrelation coefficients (between the speech and an impulse response of the recursive filter) obtained by the recursive filter.

Abstract: In an encoder for encoding a speech signal having a spectrum envelope into a plurality of excitation pulses, a spectrum emphasis unit emphasizes peak components of the spectrum envelope to produce an emphasized speech signal. As a result of a spectrum emphasis operation, the emphasized speech signal has an emphasized spectrum envelope which substantially comprises a plurality of line spectra. Responsive to the emphasized speech signal, a pulse producing unit produces a plurality of excitation pulses by the use of a pulse search method.

Abstract: In a multipulse-excitation system, the number of pulses versus the number of quantization levels are adjusted as a function of speech signal power, for example, voiced sound is high power and needs only a few pulses, but large number of quantization levels, versus the reverse for unvoiced sound.

Abstract: A speech recognition system for recognizing the words spoken by identified speakers and unidentified speakers, based upon the use of three dimensional, polar coordinate expressions of formant information, which eliminate the differences in word patterns encountered among various speakers, and the content of a reference pattern memory which stores formant information as a reference pattern in polar coordinates. The input speech is subjected to LPC analysis for each of several analysis degrees and the resulting coefficients are time normalized against the reference pattern using a DP matching process, which may be on the basis of city block distances in order to determine the most appropriate analysis degree that represents the input speech.