Abstract: A speech activity detection circuit and method are described in which a digitized sample of the analog speech signal in a communications system is taken n times in a period of time t. The magnitude M of each sample is calculated and the largest value of M which is encountered over a given period t is stored as a peak magnitude M(t). This value is compared to a value M(t-1) which is the smallest value of M(t) from prior periods. The lesser of the two values is stored as the new value for M(t-1). The process is continued for i periods and at the end of the i.sup.th period, the value M(t-1) is multipled by a constant k and the product is stored as the threshold value for the next i periods of time t.

Abstract: Recognition of continuous speech by comparison with prestored isolated words may be confused by the merging together of spoken adjacent words (coarticulation). Improved recognition is attained by generating overlap-words, e.g., words whose first phoneme is the end phoneme of the preceding word in a string of words. The reference candidate series of overlap-words is transformed under dynamic time warping so as to time-match the utterance series of overlap-words.

Abstract: This PARCOR-type speech synthesizer replaces a ten-stage lattice type filter with a pipeline multiplier and feedback loop, and provides a loss circuit (for bandwidth broadening) using subtraction circuits for multiplication.

Abstract: A speech controlled dialing circuit identifies input utterances which may be a command word (mode select), repertory word (dialing name or number), or non-recognized ("Other"). Responsive to the identification of each occurring input utterance, a set of predetermined templates are selected to identify the next occuring utterance. A programmed microprocessor system is described to implement the main controller function.

Abstract: A speech recognition system is disclosed which employs a network of elementary local decision modules for matching an observed time-varying speech pattern against all possible time warpings of the stored prototype patterns. For each elementary speech segment, an elementary recognizer provides a score indicating the degree of correlation of the input speech segment with stored spectral patterns. Each local decision module receives the results of the elementary recognizer and, at the same time, receives an input from selected ones of the other local decision modules. Each local decision module specializes in a particular node in the network wherein each node matches the probability of how well the input segment of speech matches the particular sound segments in the sounds of the words spoken. Each local decision module takes the prior decisions of all preceding sound segments which are input from the other local decision modules and makes a selection of the locally optimum time warping to be permitted.

Abstract: An electronic cash register includes a synthetic speech sound generation system for providing an audible announcement related to a bargain article of the present day and a next service day. The synthetic speech sound generation system is activated when a total key is actuated to complete one set of registration of transaction data assigned to one customer. The synthetic speech sound generation system is alternatively actuated at preselected times. The next service day information is also applied to a printer system for printing out the next service day on a receipt slip. The bargain article to be announced and the date information of the next service day are preset through the use of function keys and numeral keys included in the electronic cash register.

Abstract: This electrical waveform synthesizer uses a digital feedback loop circuit which includes selectable inputs for multiplex or recirculation operation, and a shift register whose shift rate is controlled by an adjustable precision clock so as to accurately adjust the frequency of the output waveform. The adjustable clock features a precision oscillator whose pulse rate is divided by a selectable-logic rate-divider circuit.

Abstract: In a formant speech analysis synthesis system, formant extraction to control a recursive digital all-pole filter encounters the problem that pole-pairs are not orderly arranged and that real poles may occur which are not representative of formants. The problem is solved by transforming the coefficients of the second-order sections of the filter to coefficients which can be easily ordered and by means of which it is simple to assign formants to the real poles.

Abstract: An apparatus and method are provided for the recognition of speech produced by various vocal pitches capable of recognition and classification of speech articulation at a real time rate. The apparatus and method assume that articulation of a given sound in an individual's speech can be approximated as the output of a specific linear filter, corresponding to the condition of the individual's vocal tract at the time of articulation, in response to an input of one or more source impulses. The invention selects one of a library of sounds, in response to a speech waveform input, by means of a bank of vocal tract inverse filters, each of which is connected to the speech waveform input. Each vocal tract inverse filter has a complex Fourier transfer function that is the reciprocal of a particular vocal tract transfer function corresponding to a specific speech sound. Thus there is one vocal tract inverse filter for each speech sound as spoken by the particular individual.

Abstract: Simultaneous interpretation apparatus using matrix circuits whereby, even when an additional original language is used other than the specified original languages for the apparatus, simultaneous interpretation of that additional original language into the specified original languages is accomplished with ease and without hindrance. The present invention is particularly adapted to the interpretation of four specified original languages, for example Japanese, English, French and Spanish, and is particularly useful when an additional original language is used, such as Chinese for example. The matrix circuitry of the present invention enables translation of the additional original language to proceed even though the apparatus is primarily adapted for use in interpreting and translating the four specified original languages. Without this matrix circuitry the apparatus could not adapt to the additional original language since certain translators would not be alerted to perform their translating duties.

Abstract: Computation of the partial correlation coefficients (PARCOR K.sub.i) of a signal, using less cascaded hardware, is implemented by first deriving a sequence of auto-correlation coefficients (v.sub.j) which are then transformed into a sequence of K.sub.i using a single section digital filter plus recirculating circuitry for data iteration.

Abstract: A speech synthesis device is adapted to provide an audible indication of numerical information through the utilization of a predetermined number of phonemes. Those phonemes are stored within a read only memory on a single large scale integrated circuit chip. A desired length of pause or silence is provided depending upon the kind and location of information to be audibly outputted. The necessity for the pause period is stored in digitally encoded signals within the read only memory in the same manner as with the phonemes.

Abstract: A speech recognition technique is disclosed for recognizing words that are spoken at speeds that approach the speed of continuous speech.To avoid the cost of a rigorous approach wherein all possible words of stored vocabulary are correlated against each group of speech samples, this invention uses relatively long duration inter-string pauses to detect string boundaries, and relatively short duration inter-segment pauses to determine speech segment boundaries which are used as a limited number of start or end points.

Abstract: An apparatus for the acquisiton of a raw speech signal and the essentially simultaneous acquisition of a transform of the speech signal, wherein said transform covaries as a function of changes in one or more parameters in the speech signal and is indicative of a predetermined selected speech characteristic, such as nasalization, pitch or intensity. The apparatus includes a microphone for producing first signals representative of raw speech, and a second transducer, such as, for example, an accelerometer for generating second signals essentially simultaneous to the production of the first signals, with the second signals being indicative of a selected parameteric characteristic of the human speech, such as, for example, nasalization. The first and second signals are applied to data processing circuits which analyzes the first and second signals to produce transform signals based on arithmetic combinations thereof.

Abstract: A speech synthesis circuit capable of being implemented in an integrated circuit is disclosed. The speech synthesis circuit has an input port for receiving frames of data consisting of speech coefficients, a memory for storing interpolated values of the speech coefficients and an interpolator circuit coupled to the input port and to the memory. A synchronous timing circuit is provided for generating a data frame timing signal, interpolation count timing signals and parameter count timing signals. The rate of the parameter count timing signals is a multiple of the rate of the interpolation count timing signals, which is in turn a multiple of the rate of the data frame timing signal. These signals occur at predetermined times and are generated in the disclosed embodiment by Programmed Logic Arrays (PLA's). The data frame timing signal controls the receipt of a new frame of data at the input port.

Abstract: A control interface system for use with a memory device executing variable length instructions and implementable utilizing integratable field effect semiconductor devices is disclosed. One example of such a system is a speed synthesis device controlled by the central processing unit of a computer. A control latch is utilized to release the control device, such as a central processing unit, after an instruction has been initiated. A plurality of latches, each set by the conclusion of the execution of a previous instruction, are utilized to reset the control latch. An attempted second access of the memory device during the execution of a previous command will result in suspending the operation of the control device until such time as the previous instruction is concluded and the new instruction is initiated.

Abstract: Systems analysis approach to measure and to reconstruct the sound energy flux distribution characteristic of "live" situations. The present invention relates to a method of measurement of acoustical fields and the functional relationships in audio systems for enhancing the reproduction of sound. The environmental acoustic simulator is a system which generates at least two signals having different combinations of time delays from each of a stereo input signal pair and for deriving therefrom a set of not less than four output channels. At least forty five time delays at nonuniform intervals spanning a time period of not less than two seconds with different frequency equalizations are derived. Diffuse sound fields are created through electronic mixing and by the employment of not less than four loudspeaker groups.

Abstract: The speech/silence discriminator is used on a telephone line to distinguish between periods of speech and periods of silence. A signal derived from the speech signal is, in the speech state compared to an adaptive threshold which is a fraction of the maximum, to be eventually quantized, reached by said signal during the period of speech which is considered; the speech to silence transition being determined when the signal decreases below said threshold level, the threshold level being determined during each speech period as a function of the maximum attained during the period. The adaptive threshold has a minimum value of -60 dBm. It is also comprised of a noise level evaluation circuit which determines the threshold decision level (.gtoreq.-60 dBM) for the transition from silence to speech which once reached, in the period of silence, enables the discriminator to go from the silent state to the speech state. It is useable in speech interpolation systems.

Abstract: To provide alternative operation at 2.4 or 9.6 Kb/s bit rates, a multirate igital processor generates a data frame of 216 bits representing sync and voice encoded at both rates, with the 2.4 Kb/s embedded as a subset within the frame. Rate conversion is made by inserting or deleting 162 bits per frame. Speech signal is transformed by linear prediction into coefficients and residual signals. The coefficients are transmitted similarly at both 2.4 and 9.6 Kb/s rates. However, the residual is processed into a simple pitch/hiss excitation signal for 2.4 Kb/s, but is processed into a Fourier transform of its baseband (250-1500 Hz) for 9.6 Kb/s excitation signal transmission.

Abstract: A highly simplified speech synthesizer that is capable of producing quality speech. The present speech synthesizer is adapted to be driven by an 8-bit digital input command word. Six of the bits are used for phoneme selection and the remaining two bits for inflection control. In a first embodiment, the system is adapted to generate twelve parameter control signals for each phoneme, with one of the parameters being utilized to control both high and low frequency fricative injection into the vocal tract. This embodiment also provides asynchronous excitation of the vocal tract by including a second fricative excitation control circuit that is adapted to inject white noise in parallel into the second and third resonant filters under the control of the vocal amplitude control signal. In a second embodiment, one of the twelve signal parameters is utilized as two separate control signals thus effectively providing thirteen control signal parameters.