Abstract: To prevent adaptation to misrecognized words in unsupervised or on-line automatic speech recognition systems confidence measures are used or the user reaction is interpreted to decide whether a recognized phoneme, several phonemes, a word, several words or a whole utterance should be used for adaptation of the speaker independent model set to a speaker adapted model set or not and, in case an adaptation is executed, how strong the adaptation with this recognized utterance or part of this recognized utterance should be performed. Furtheron, a verification of the speaker adaptation performance is proposed to secure that the recognition rate never decreases (significantly), but only increases or stays at the same level.

Abstract: It is intended to increase the recognition rate in speech recognition and image recognition. An observation vector as input data, which represents a certain point in the observation vector space, is mapped to a distribution having a spread in the feature vector space, and a feature distribution parameter representing the distribution is determined. Pattern recognition of the input data is performed based on the feature distribution parameter.

Abstract: A discriminant or identification function is used for pattern recognition in which the highest performance can be offered when adaptation is made. Learning is carried out while a discriminant or identification function is adapted to a learning sample. For example, a standard pattern of the character "A" used as an identification function is learned such that when the character "A" slanting in the right or left direction is input, the standard pattern of the character "A" is rotated (adapted) in accordance with the slanting of the input learning sample.

Abstract: A discriminant or identification function is used for pattern recognition in which the highest performance can be offered when adaptation is made. Learning is carried out while a discriminant or identification function is adapted to a learning sample. For example, a standard pattern of the character "A" used as an identification function is learned such that when the character "A" slanting in the right or left direction is input, the standard pattern of the character "A" is rotated (adapted) in accordance with the slanting of the input learning sample.

Abstract: A parameter converting method and a speech synthesizing method for synthesizing speech with a voice quality similar to that of an input voice includes a parameter converting function constituted of a weighting function for setting weighting coefficients on an input sound spectrum parameter space and a plurality of sub-converting functions. Conversion outputs of the respective sub-converting functions are given weighting coefficients such that the sum of the weighted conversion outputs is used as the parameter converting function to convert M sound spectrum parameter to a single sound spectrum parameter. In this way, the freedom of adaptation with respect to the parameter converting function can be more properly set, so that the parameter converting function can provide an accuracy in accordance with an amount of speech data inputted for learning. It is therefore possible to provide a sound spectrum parameter for generating a voice quality much closer to that of an input voice.

Abstract: A method for speech analysis applicable to a speech analysis/synthesis system employed for producing a synthetic speech. Voiced and unvoiced segments of input speech signals X(n) are discriminated. An amplitude information A(.omega.) and a phase information P.sub.X (.omega.) are extracted from the voiced segments of the input speech signals. A pitch period is detected from the voiced segments of the input speech signals. A pulse train S(n) as a sound source information is generated so that its period corresponds on the time scale to the detected pitch period of the input speech signals. A phase information P.sub.S (.omega.) is extracted from the pulse train S(n). A difference P(.omega.) between the phase information P.sub.S (.omega.) of the pulse train S(n) and the phase information P.sub.X (.omega.) of the input speech signal is found and is supplied as the phase information of the desired one-pitch period within the input speech signals.

Abstract: A signal encoding apparatus for cutting out blocks of an input signal waveform at a predetermined time interval and converting the in-block signal by mutually independent conversion axes for encoding, wherein the waveform cutting block length along the time axis is changed according to the input signal.

Abstract: A system for encoding audio signals, transmitting or recording the encoded audio signals, and then decoding the transmitted audio signal includes a circuit for determining the block length of the input audio signals based upon detected values of predetermined portions of the input signal and then performing an orthogonal conversion of the determined block length signal, in which the signal is orthogonally transformed from a time-domain signal to a frequency-domain signal. Following quantization, the converted and encoded signal is transmitted and upon performing reverse quantization, a reverse orthogonal converting circuit converts the signal back to the time-domain in keeping with the signal providing information concerning the various block lengths that were determined during encoding. The converted time-domain signals are then connected into a smooth waveform to form output audio signals.

Abstract: A decoder for reproducing information from an encoded signal that has been frequency-domain transformed by a transform conversion circuit in response to audio signal samples grouped into time-domain blocks based on selected block time lengths, includes a reverse conversion transformation circuit that groups the frequency-domain transfer coefficients into transform blocks based on the block time length information used in the encoding operation and applies the inverse discrete transform function to generate time-domain signal sample blocks therefrom.

Abstract: An apparatus for predictive coding processing a digital signal comprises a selective predictive filter, responsive to an input signal for outputting a predictive residual signal, an adaptive predictive filter, responsive to the predictive residual signal, for outputting an optimal predictive residual signal, and a re-quantizing unit means for re-quantizing and outputting the optimal predictive residual signal. The positions of the selective and adaptive predictive filters may be interchanged and the processing apparatus may be either feed forward or feed back type.

Abstract: A signal encoding apparatus for cutting out blocks of an input signal waveform at a predetermined time interval and converting the in-block signal by mutually independent conversion axes for encoding, wherein the waveform cutting block length along the time axis is changed according to the input signal.

Abstract: In an adaptive predictive coding (APC) encoder for compression of quantized digital audio, the signal-to-quantization noise ratio is improved and more complicated input signal spectra can be processed through use of an input signal frequency spectrum detector employing linear predictive coding (LPC) analysis for setting the coefficients of the requantization error noise filter. The detector operates in synchronism with the predictive and noise filters so as to minimize the number of processing calculations while maximizing processing speed.

Abstract: A method and system for radio communication between at least one parent communication unit and a plurality of slave communication units. A slave communication unit transmits an automatic answer signal in response to an incomming call receipt signal transmitted from the parent communication unit. When one of the slave communication units makes a called-person's answer to the incomming call receipt signal, it transmits a called-person's answer signal. If interference occurs between the automatic and called-person's answer signals, the parent communication unit provides a command, causing the slave communication units to stop the automatic and called-person's answer signals and permits the slave communication unit which transmits the called-person's answer signal to communicate with the parent communication unit.

Abstract: This invention relates to a digital signal encoding apparatus in which the width of the range in selected to be wider for the higher frequency range of the digital signals divided into a plurality of regions and in which the encoded signals are synthesized for the respective ranges, wherein encoding is controlled as a function of the output detecting the characteristics of the components of the divided frequency ranges and the detection time interval is selected to be longer for the lower frequency to enable more efficient encoding to be performed as a function of the properties of input digital signals.

Abstract: A method and an apparatus for processing a digital signal by calculating a predictive error, then requantizing the predictive error thus obtained, and correcting the spectral form of requantization noise generated in quantizing the predictive error, wherein the spectral form of such requantization noise is approximated to that of the input signal while being suppressed in the lower frequency band thereof.

Abstract: A method and system for radio communication between at least one parent communication unit and a plurality of slave communication units. A slave communication unit transmits an automatic answer signal in response to an incoming call receipt signal transmitted from the parent communication unit. When one of the slave communication units makes a called-person's answer to the incoming call receipt signal, it transmits a called-person's answer signal. If interference occurs between the automatic and called-person's answer signals, the parent communication unit provides a command, causing the slave communication units to stop the automatic and called-person's answer signals and permits the slave communication unit which transmits the called-person's answer signal to communicate with the parent communication unit.

Abstract: This invention relates to a digital signal encoding apparatus in which the .[.width of the range in.]. .Iadd.digital signal is divided into frequency components in plural frequency bands and the bandwidth of the frequency bands is .Iaddend.selected to be wider for .[.the.]. higher .[.frequency range.]. .Iadd.frequencies .Iaddend.of the digital .[.signals divided into a plurality of regions.]. .Iadd.signal .Iaddend.and in which the encoded signals are synthesized for the respective .[.ranges.]. .Iadd.frequency bands .Iaddend.wherein encoding is controlled as a function of the output detecting the characteristics of the .Iadd.frequency .Iaddend.components .[.of.]. .Iadd.in .Iaddend.the .[.divided.]. frequency .Iadd.bands .Iaddend.and the detection time interval is selected to be longer for the lower frequency .Iadd.bands .Iaddend.to enable more efficient encoding to be performed as a function of the properties of input digital signals.