Abstract: A harmonic speech coding arrangement where vector quantization is used to improve speech quality. Parameters are determined at the analyzer of an illustrative coding arrangement to model the magnitude and phase spectra of the input speech. A first codebook of vectors is searched for a vector that closely approximates the difference between the true and estimated magnitude spectra. A second codebook of vectors is searched for a vector that closely approximates the difference between the true and the estimated phase spectra. Indices and scaling factors for the vectors are communicated to the synthesizer such that scaled vectors can be added into the magnitude and phase spectra for use at the synthesizer in generating speech as a sum of sinusoids.

Abstract: Various functions (or portions thereof) are associated with some of the words or instructions recognizable by a speaker independent voice recognition device (128). This association is presented to an operator via one or more menus (200a-200d) so that the operator may select any of several functions by use of a limited set of speaker independent commands.

Abstract: A communication system having an encoder device used in combination with a decoder device for encoding a sequence of digital speech signals into a sequence of output signals, using a spectrum parameter and a pitch parameter. A subsidiary parameter of the digital speech signals is detected and monitored by a monitoring circuit. Digital speech signals are classified into voiced sound or voiceless sound and into vocality, nasal, fricative, or explosive durations at every frame. When a voiced sound, i.e., a vocality is detected, a predetermined number of excitation pulses are calculated during a representative subframe and are produced as primary sound source signals. A subsidiary information signal is produced during the remaining subframes to represent phase and amplitude correction factors in each of the subframes. When a voiceless sound, i.e.

Abstract: A method and apparatus utilizing electrodermal response as a control mechanism. A specific embodiment of the method and apparatus utilizing a personal computer having a game paddle input port is described. The time rate of change in the electrodermal response of a biologic individual is sensed independently of the absolute value of the electrodermal response and continuously adjusted toward a time rate of change approaching the average time rate of change which the biologic individual cna consciously achieve in such electrodermal response. The resultant time rate of change is utilized as the input to a computer in order to provide the control.

Abstract: A speech analyzing and synthesizing apparatus samples a speech signal and its coding section outputs one or more codes of variable length per sampling according to the differential value between two mutually adjacent sampled signals to represent this differential value. The apparatus also includes a decoding section which decodes the outputted codes to obtain the differential value and a limiting circuit for limiting the number of outputtable codes per sample according to the differential value and codes representing differential values previously outputted from the coding section.

Abstract: A recording and reproducing apparatus wherein an externally input voice stating an alarm time such as month, day and time, together with an associated message, is converted into voice data for storage in a voice data memory. Only that portion of the voice indicating the alarm time is recognized by a voice alarm time recognition circuit and is also stored in an alarm time data memory as an alarm time. When the alarm time is reached, the corresponding voice data stored in the voice data memory is read out and audibly reproduced so as to sound the alarm time and the associated message.

Abstract: A signal vector quantizing coder (CELP) is provided with an adaptive codebook originally loaded with preselected codewords. The codebook is split into a fixed contents portion and a fixed length adaptive contents portion. During coding operations, the codewords dynamically selected for coding the coder input signal are shifted into the fixed length adaptive codebook section for codebook contents updating purposes.

Abstract: Apparatus for order entry in a restaurant includes a host and remote portable terminals. Orders are entered on the remote terminals and sent by radio frequency communication to the host systems. The portable terminals include a bar code template of menu items and a light pen for scanning menu item selections. Orders are entered by scanning selections in a manner similar to the use of a conventional restaurant order pad. Terminals and printers are connected to the host system and are used to display orders entered at portable terminals for the cooks in the kitchen and to print order receipts.

Abstract: An orthogonalizing time filter section is arranged in place of a Gram Schmidt orthogonalizing section. The orthogonalizing time filter section is constituted by a plurality of filters for performing smoothing processing and differential processing. The orthogonalizing time filter section obtains an average pattern of acquired learning patterns, and smoothes the average pattern along the time base to obtain a dictionary of a first axis. The section differentiates the average pattern along the time base to obtain a dictionary of a second axis. The above processing is repeated for each category, thus generating an orthogonalized dictionary.

Abstract: A speech recognition apparatus including a memory for storing with respect to each feature specific to a particular phoneme a name of a process and a procedure of the process which is performed in order to search whether the presence of a feature specific to a certain type of speech is included in a feature vector series, and for storing a table in which the names of the processes in a performed for all the categories of speech to be recognized. The information stored in the memory is used to discriminate between two categories and provides ways for interpreting the results of the process. The recognition processes performed the discrimination is done in accordance with the information stored in the table.

Abstract: Speech is analyzed to derive the parameters of a synthesis filter and the parameters of a suitable excitation which is selected from a codebook of excitation frames. The selection of the codebook entry is facilitated by determining a single-pulse excitation (e.g., using conventional multipulse excitation techniques), and using the position of this pulse to narrow the codebook search.

Abstract: Voice signals amplified in a programmable gain, programmable bandwidth subsystem are digitized and buffered and the digitized signals are transferred over a dedicated high speed bus to a speech processor, thus relieving the speech processor and its resident host of the overhead of high data rate transfers and permitting a relatively low capacity computer to accomplish voice recognition on a real time basis.

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: In a method of and a system for determining the variation of a speech parameter, for example, the pitch, in a speech signal, values ms(i,j) and coupling vectors v(i,j) are calculated for time instants i and a number of values j of the speech parameter f.sub.j for each time instant i by means of an optimization algorithm. Of the values ms(m,j) associated with the last time instant i=m, the optimum (that is to say the smallest or, on the contrary, the largest) value is determined. By use of the coupling vectors, the variation of the speech parameter as a function of time can be obtained by means of a back-tracking procedure. In the calculation of the values ms(i,j), inter alia, a cost parameter k is taken into account which is a measure of the deviation of the speech parameter f.sub.j (i) at the time instant i with respect to a predicted value for the speech parameter at the time instant i.

Abstract: An interrogating device includes a memory for storing forward rules and facts, which are used in a forward inference control system, and a control unit operable to add or delete a fact to the memory. In this interrogating device, the accuracy indicates the probability degree of preparation for a conclusion section of the rule. The fact is added to the conclusion section of the rule and is stored in the memory. Also, the accuracy may be used in a strategy for a conflict resolution of a forward inference control. The fact for the accuracy that is obtained in the backward inference has a structure identical to the fact which the accuracy is added for use in the forward inference. By adding the accuracy negatively to the fact and the rule, an inference which uses uncertain data in the forward inference is possible.

Abstract: A voice recognition method and apparatus in which a scalar time-space field pattern lattice plane having an abscissa and an ordinate, with one being the time axis and the other the space axis, is obtained from a voice signal inputted as a recognition object, and a voice characteristic pattern based upon said time-space pattern is to be formed as a standard pattern or to be matched with a standard pattern to recognize a characteristic of the inputted voice. The scaler time-space field pattern is converted to extract a vector field pattern lattice corresponding to each lattice point of the scalar time-space field pattern lattice plane with each vector of the vector field pattern lattice having a quantity and an angular orientation value.

Abstract: A voice coding system for separating and coding voice information into spectrum envelope information and voice source information, with the intention of compressing the amount of information for efficient coding of vocal audio signals through the control of the voice source information based on the fact that the spectrum envelope information and voice source information highly correlate with each other.

Abstract: A similarity memory stores a syllable recognition candidate and its similarity (j) supplied from a syllable recognition section at a specific storage position defined by reference syllable data and its syllable position (i), stores a subsidiary similarity (j-10) at an (i+1)th syllable position of the same syllable name, and stores a subsidiary similarity (j-20) at an (i+2)th syllable position of the same syllable name. The subsidiary similarity (j-10) is also stored at the (i+1)th syllable position of the same syllable name. A coincidence computing section computes a sum of similarities of a series of n syllables from the similarity memory and computes a coincidence.

Abstract: When a speech sound of at least a predetermined sound pressure is externally input while a time measurement is not being performed, a time measuring circuit starts a time measurement responsive to a signal from a speech detector. When another speech sound of at least a predetermined sound pressure is externally input while a time measurement is being performed by the time measuring circuit, a measurement time measured by the time measuring circuit at this moment is stored in a time information memory. After a predetermined time has elapsed, if a speech recognition circuit recognizes that the externally input speech sound is a "stop" command, the time measurement operation performed by the time measuring circuit is stopped, and the time information stored in the time information memory is read out and displayed as measurement time information on a display unit.

Abstract: A continuous speech recognition system having a speech processor and a word recognition computer subsystem, characterized by an element for developing a graph for confluent links between confluent nodes; an element for developing a graph of boundary links between adjacent words; an element for storing an inventory of confluent links and boundary links as a coding inventory; an element for converting an unknown utterance into an encoded sequence of confluent links and boundary links corresponding to recognition sequences stored in the word recognition subsystem recognition vocabulary for speech recognition. The invention also includes a method for achieving continouous speech recognition by characterizing speech as a sequence of confluent links which are matched with candidate words. The invention also applies to isolated word speech recognition as with continuous speech recognition, except that in such case there are no boundary links.