Abstract: A programmable digital circuit is provided for bus control of a fast microprocessor such as a V33 microprocessor. The digital circuit includes a bus controller, a wait state controller and a clock generator generating a processor clock and a system clock with but a small skew between them.

Abstract: A standard pattern comparing system recognizes a data by comparing the data with a standard pattern registered in a dictionary, a plurality of kinds of data being supplied to the standard pattern comparing system. The standard pattern comparing system has a plurality of dictionaries including a plurality of normal dictionaries and one master dictionary. All standard patterns stored in each normal dictionary and some of the standard patterns stored in the master dictionary corresponding to one kind of data. There being no duplicate standard pattern included in more than two dictionaries, and each standard pattern included in the master dictionary being commonly used for at least more than two normal dictionaries to correspond to an arbitrary kind of data. The present invention is especially useful for a voice recognition system in which a speaker dependent voice recognizer is used for a personal computer which manages a plurality of application programs.

Abstract: The present invention discloses a method for encoding and decoding human speech signals by generating a data base which stores a number of human speech signal types. The number of human speech signal types stored is sufficiently high enough to cover substantially all observable human speech. According to a first embodiment of the invention, a set of representative human speech signal curves is taken directly from natural human speech. According to a second embodiment of the invention, a predetermined set of speech signal parameters is used where maximum voice signal segment values are measured. According to a third embodiment of the invention, an adaptive set of speech signal parameters is used, where the encoder transmits a set of signal parameters to the decoder. Although the invention is specifically designed for human speech, it can also be used in connection with other audio signals, such as those of electronic musical instruments.

Abstract: In an apparatus for controlling an industrial robot system comprising a workpiece handling apparatus having first revolute joints for moving a workpiece, and a tool moving apparatus having second revolute joints for moving a tool so as to move the workpiece and the tool on a predetermined trajectory between two adjacent teaching points, there are calculated a position and an attitude of a predetermined coordinate system of a predetermined workpiece attachment reference point based on a coordinate system of a baseplane of said workpiece handling apparatus, and a position and an attitude of a predetermined coordinate system of a predetermined tool attachment reference point based on a coordinate system of a baseplane of said tool handling apparatus, from inputted teaching data. Further, there are calculated joint variables of the first and second revolute joints for moving tile workpiece and the tool on the trajectory, from the calculated data.

Abstract: The set of possible excitation signals is subdivided into a plurality of subsets, the first of which provides the contribution to the coded signal necessary to set up a transmission at a minimum rate guaranteed by the network, while the others supply a contribution which, when added to that of the first subset, causes a rate increase by successive steps. At the receiving side, a decoded signal is generated by using the excitation contribution of the first subset alone if the coded signals are received at the minimum rate, while for rates higher than the minimum rate the contributions of the subsets which have allowed such rate increase are also used.

Abstract: Systems and methods for improved speech acquisition are disclosed including a plurality of linearly arrayed sensors to detect spoken input and to output signals in response thereto, a beamformer connected to the sensors to cancel a preselected noise portion of the signals to thereby produce a processed signal, and a speech recognition system to recognize the processed signal and to respond thereto. The beamformer may also include an adaptive filter with enable/disable circuitry for selectively training the adaptive filter a predetermined period of time. A highpass filter may also be used to filter a preselected noise portion of the sensed signals before the signals are forwarded to the beamformer. The speech recognition system may include a speaker independent base which is able to be adapted by a predetermined amount of training by a speaker, and which system includes a voice dialer or a speech coder for telecommunication.

Abstract: The present invention relates to a data protection (310) for preventing data from being erroneously written in a data holding circuit (307). A data protection circuit (311) receives a data protection set release select signal Sa provided by a processor (330), thereby causing a second timer (316) to start timing. The data protection circuit (311) further detects the level of a chip select signal CS provided by a power supply monitoring circuit (320) after the second timer (316) timed the time interval T2 and provides a data protection signal DP corresponding to the signal Sa to the data holding circuit (307) only in the case that the data protection circuit (311) judged that the main power supply (305) is at normal state.

Abstract: A modular input-output subsystem for a supercomputer is disclosed. Peripheral devices are coupled to the system through channel adaptor interfaces, while communication with the CPU is through high speed data channels. A memory buffer is provided to buffer data transfers between the peripherals and the Central Processing Units (CPUs) and the Solid State Storage Devices (SSDs).

Abstract: A system is provided with a voice activated control apparatus which permits precise control of one or more system variables by means of voice commands uttered by the system operator. When a system variable is in a movement or change mode, the movement or change is terminated by any sound exceeding a preestablished acoustic threshold level. Any value of one or more system variables can be identified and appropriate data stored in a memory to permit the one or more system variables to return to an identified value or state with a single voice command. The control apparatus is combined with a screen monitor and/or an acoustic speaker to provide visible and/or acoustic responses to an operator. The control apparatus is practical for retrofitting existing remotely controllable systems.

Abstract: A disk array controller including a data path architecture which can be configured to perform data transfer operations between a host system and a plurality of disk drives configured as a RAID level 1, 3, 4 of 5 disk array. The data path architecture includes a plurality of array channels for coupling selected disk drives with a host system bus and circuitry for generating parity information during write operations. Each array channel includes a data bus connected through bus drivers to the host bus, its associated disk drive, and the input and output of the parity generation circuitry. The bus drivers control the placement of data unto the array busses and the direction of the data flow on the array busses.

Abstract: A neural network architecture consisting of input weight multiplications, product summation, neural state calculations, and complete connectivity among the neuron processing elements. Neural networks are modelled using a sequential pipelined neurocomputer producing high performance with minimum hardware by sequentially processing each neuron in the completely connected network model. An N neuron network is implemented using multipliers, a pipelined adder tree structure, and activation functions. The activation functions are provided by using one activation function module and sequentially passing the N input product summations sequentially through it. One bus provides N.times.N communications by sequentially providing N neuron values to the multiplier registers. The neuron values are ensured of reaching corresponding multipliers through a tag compare function. The neuron information includes a source tag and a valid signal.

Abstract: A speech coding apparatus compares the closeness of the feature value of a feature vector signal of an utterance to the parameter values of prototype vector signals to obtain prototype match scores for the feature vector signal and each prototype vector signal. The speech coding apparatus stores a plurality of speech transition models representing speech transitions. At least one speech transition is represented by a plurality of different models. Each speech transition model has a plurality of model outputs, each comprising a prototype match score for a prototype vector signal. Each model output has an output probability. A model match score for a first feature vector signal and each speech transition model comprises the output probability for at least one prototype match score for the first feature vector signal and a prototype vector signal.

Abstract: A method and system are provided for time aligning speech. Speech data is input representing speech signals from a speaker. An orthographic transcription is input including a plurality of words transcribed from the speech signals. A sentence model is generated indicating a selected order of the words in response to the orthographic transcription. In response to the orthographic transcription, word models are generated associated with respective ones of the words. The orthographic transcription is aligned with the speech data in response to the sentence model, to the word models and to the speech data.

Abstract: A computer-aided expert system comprising a hypermedia structured expert system and a hypermedia-structured multi-level electronic document archive linked thereto. The expert system includes a knowledge base, an inference engine that interacts with the knowledge base to generate advice. A hypermedia interface having knowledge base transfer means is provided on a display that includes selectable areas that provide access to active and passive documents in the document archive. The document archive includes active and passive electronic documents interlinked by means of the hypermedia interface and linked to the expert system by means of the hypermedia interface. Each active and passive electronic document includes active document transfer means that provide for transfer from the active document to another active document, a passive document, or a knowledge base linked thereto.

Abstract: A neural network employing absolute difference calculating synapse cells comprising a pair of floating gate devices coupled in parallel between an internal cell node and column line of the network. The network further includes a switched-capacitor circuit for summing all of the charges generated by all of the synapse cells within a column of the network. The circuit operates in response to a sequence of applied voltage pulses such that each cell generates a charge representing either the input, the weight, or the minimum/maximum of either the weight or the input. The accumulation of these charges represents the sum of the absolute value difference between the input voltages and the stored weights for a single column of the array.

Abstract: A dictionary order sorter resorts character strings of recognition candidates stored in a high-ranking candidate memory in the order of distance into a dictionary order (character code order). Upon receipt of a sort termination signal a display controller displays the character strings of recognition candidates stored in the high-ranking candidate memory in the dictionary order and their ranking numbers in order on a display. Where an attribute-dependent sorter is provided in place of the dictionary order sorter, the character strings of recognition candidates stored in the distance order in the high-ranking candidate memory are sorted (grouped) according to attribute information of categories, such as parts of speech, concepts, etc., which are stored in a template memory for each of the recognition candidates and then displayed classified into groups. The categories of attribute information can arbitrarily be specified by the user.

Abstract: Speech coding of the code excited linear predictive type is implemented by providing an excitation vector which comprises a set of a pre-determined number of pulse patterns from a codebook of P pulse patterns, which have a selected orientation and a pre-determined delay with respect to the starting point of the excitation vector. This requires modest computational power and a small memory space, which allows it to be implemented in one signal processor.

Abstract: A code excited linear predictive coder and decoder well suited to speech recording, transmission and reproduction, especially in voice messaging systems, provides backward adaptive gain control of stored codevectors to be applied to a synthesis filter prior to being compared with sequences of input speech signals. Simplified linear predictive parameter quantization using efficient table lookup procedures, efficient codevector storage and search all contribute in an illustrative embodiment to high quality coding and decoding with reduced computational complexity.

Abstract: A method and apparatus for the automatic analysis, synthesis and modification of audio signals, based on an overlap-add sinusoidal model, is disclosed. Automatic analysis of amplitude, frequency and phase parameters of the model is achieved using an analysis-by-synthesis procedure which incorporates successive approximation, yielding synthetic waveforms which are very good approximations to the original waveforms and are perceptually identical to the original sounds. A generalized overlap-add sinusoidal model is introduced which can modify audio signals without objectionable artifacts. In addition, a new approach to pitch-scale modification allows for the use of arbitrary spectral envelope estimates and addresses the problems of high-frequency loss and noise amplification encountered with prior art methods.

Abstract: A speech signal coding apparatus inputs a pitch period generated by coding a speech signal, and outputs information on a range for said pitch period, together with the pitch period. A speech signal decoding apparatus inputs the pitch period and the above information on the range, and determines whether or not the pitch period is within the range. When the pitch period is determined to be within the range, the speech signal decoding apparatus outputs the above pitch period. When the pitch period is determined not to be within the range, the speech signal decoding apparatus outputs as a pitch period a predetermined value within the range.