Abstract: A voice decoding device for reproducing the sound information which is encoded in a predetermined frame unit and packet-transmitted is provided. An interframe-predicting unit continuously predicts a series of data over plural frames. A control unit always monitors whether the absence of a packet occurs or not. When the absence of a packet is detected, a selector is switched, and the decoding processing of the voice signal is performed using a series of predicted data over plural frames obtained at the interframe-predicting unit instead of a series of received data used in the usual voice decoding processing.

Abstract: Methods for determination of parts of speech of words in a text or other non-verbal record are extended to include so-called Viterbi optimization based on stored statistical data relating to actual usage and to include noun-phrase parsing. The part-of-speech tagging method optimizes the product of individual word lexical probabilities and normalized three-word contextual probabilities. Normalization involves dividing by the contained two-word contextual probabilities. The method for noun phrase parsing involves optimizing the choices of, typically non-recursive, noun phrases by considering all possible beginnings and endings thereof, preferably based on the output of the part-of-speech tagging method.

Abstract: A method of encoding speech includes a limited search of a tree-code excitation codebook with a closed loop gain calculation for each test path under consideration. The gain calculation occurs when minimizing an error distance measurement between a synthetic signal defined by each test path being considered and the appropriate speech signal by optimizing a scaling factor of the synthetic signal. The encoding method achieves a significant reduction in computational complexity with minimal loss of performance.

Abstract: An interactive patient assistance device houses first and second compartments for storing a first item and a second item away from access by the patient. First and second delivery mechanisms are associated with the first and second compartments for making the first stored item available to the patient in response to a first command signal and for making the second stored item available to the patient in response to a second command signal. The first and second items are delivered to the patient according to schedules stored in resident memory. The schedules may be altered by a prescribed command issued by the patient.

Abstract: A transform encoder, a transform decoder, and a transform encoder/decoder system employ adaptive bit allocation wherein each code word representing spectral information is allocated a fixed number of bits and an adaptive number of bits, except that at least some but not all code words are allocated a fixed number of bits.

Abstract: A speech coding system of the code excited linear prediction (CELP) type includes apparatus (24,26) for filtering digitized speech samples to form perceptually weighted speech samples. Entries in a one-dimensional codebook (110) comprising frame length sequences are filtered in a perceptually weighted synthesis filter (28) to form a one-dimensional filtered codebook. The filtered codebook entries are compared with the perceptually weighted speech signals to obtain a codebook index which gives the minimum perceptually weighted error when the speech is resynthesized. Using a one-dimensional codebook (110) reduces the amount of computation which is required compared to the use of a two-dimensional codebook.

Abstract: A speech coding apparatus which selects an optimum code from a code book, the optimum code giving the minimum magnitude of error signal between the input signal and the reproduced signal obtained by a filter calculation using a linear prediction parameter from a linear predictive analysis unit with respect to the codes of the code book, wherein the code book is formed by thinning to 1/M (M being an integer of two or more) the plurality of sampling values constituting the codes. To compensate for the deterioration of the quality of the reproduced signal caused by thinning the sampling values in this way, an additional linear predictive analysis unit is further introduced and use made of an amended linear prediction parameter instead of the linear prediction parameter from the originally provided linear predictive analysis unit.

Abstract: Speech analysis apparatus for detection and measurement of low frequency amplitude and frequency modulations, tremor, of the fundamental frequency. A microphone signal of sustained phonation is input to the apparatus which demodulates and measures five parameters digitally displayed; fundamental frequency, amplitude demodulated frequency, amplitude demodulated level, frequency demodulated frequency, and frequency demodulated level. Demodulated outputs are provided for external analysis. A variable cutoff low pass filter maintains cutoff at 1.5 times the fundamental input frequency.

Abstract: A noise reduction system for enhancing noisy speech signals by performing a spectral decomposition on the signal, passing each spectral component through a non-linear stage which progressively attenuates lower intensity spectral components (uncorrelated noise) but passes higher intensity spectral components (correlated speech) relatively unattenuated, and reconstituting the signal. Frames of noisy signal are transformed into the frequency domain by an FFT (Fast-Fourier Transform) device, with windowing. Each transformed frame is then processed to effect a non-linear transfer characteristic, which is linear above a soft "knee" region, and rolls off below, and transformed back to a reconstituted time-domain signal with reduced noise by an IFFT (Inverse Fast Fourier Transform) device (with overlapping). A level control matches the signal to the characteristic.

Abstract: A method and apparatus for linear vocal control of cursor position within a computer display system. A microphone is utilized in conjunction with a computer system to detect vocal utterances and each vocal utterance is then coupled to an analysis circuit to detect voiced and unvoiced vocal utterances. Variations in the pitch of each voiced vocal utterance and the virtual frequency of each unvoiced vocal utterance are then utilized to linearly vary the position of a cursor in the computer display system in two axes independently. In a depicted embodiment of the present invention the analysis circuit includes a short delay to ensure that a valid control signal has occurred. Thereafter, increases or decreases in pitch or virtual frequency from an initial value are utilized to initiate movement by the cursor in a positive or negative direction in the two axes. Cursor motion will persist until pitch or virtual frequency return to an initial value or until the utterance ceases.

Abstract: A channel bank speech synthesizer for reconstructing speech from externally-generated acoustic feature information without using externally-generated voicing or pitch information is disclosed. An N-channel pitch-excited channel bank synthesizer (340) is provided having a first low-frequency group of channel gain values (1 to M) and a second high-frequency group of channel gain values (+1 to N). The first group controls a first group of amplitude modulators (950) excited by a periodic pitch pulse source (920), and the second group controls amplitude modulators excited by a noise source (930). Both groups of modulated excitation signals are applied to the bandpass filters (960) to reconstruct the speech channels, and then combined at the summation network (970) to form a reconstructed synthesized speech signal. Additionally, the pitch pulse source (920) varies the pitch pulse period such that the pitch pulse rate decreases over the length of the word.

Abstract: A music tone pitch shift apparatus which converts an original audio signal into digital data by way of pulse code modulation (PCM), shifting the pitch, and converting the pitch shifted digital data into an analog signal. The PCM digital data is stored in a ring memory at a given sampling speed, and is read out of the memory by a pair of identical read circuits at a common read addressing speed corresponding to the desired pitch. One of the read circuits starts reading from the opposite address location to the other on the ring memory. Since the read addressing speed is set faster than the write addressing speed when increasing the pitch, and vice versa, overtaking or lapping between the addresses could occur.

Abstract: A voice recognition system is disclosed which has incorporated therein information on the phonological effects on syllables. The system receives a voice signal and tentatively identifies syllable arrays and provides a collection of data on syllables in the arrays. The data are used to generate hypothetical syllable arrays from the tentatively identified arrays. The hypothetical arrays are evaluated via arithmetic operations, taking into consideration the effects of context, the speaker's habit and dialect, thereby determining a reliable representation of the input voice signal.

Abstract: A pattern recognition apparatus using the hidden Markov model technique in which parameters for defining a mean vector representing the probability density function in each one of plural states for composing the hidden Markov model vary with time. Accordingly, the recognition precision may be enhanced when this apparatus is used in, for example, voice recognition.

Abstract: A broadcast digital sound processing system includes an ISA (Industry Standard Architecture) bus compatible personal computer with a hard disk drive and a sound processor board installed in an expansion slot of the computer. The board includes a stereo input, analog to digital converter (ADC) and a stereo set of digital to analog converters (DAC's) interfaced to a digital signal processor (DSP) chip. A stereophonic audio signal is converted to digital data by the ADC and communicated to the computer by the DSP chip through a two port record first-in/first-out (FIFO) buffer for storage on the disk. A program is played back by communicating a program data file through a two port playback FIFO buffer to the DSP and from thee to the DAC's for reconstruction to a stereo set of analog signals. The reconstructed audio signals may then be used as a modulating signal for radio broadcasting.

Abstract: A spiral audio spectrum display system and technique. A plurality of signals are provided which represent power spectral density of an audio signal as a function of frequency. These signals are transformed to polar coordinates and combined with a fixed spiral reference signal in polar coordinates. The combined signal is displayed to provide a spiral representation of the audio spectrum signal. In specific embodiments, each octave span of the audio signal is displayed as a revolution of the spiral such that tones of different octaves are aligned and harmonic relationships between predominant tones are graphically illustrated. The system and display technique of the present invention allows both pitch and harmonic content of an audio signal to be visually identified.

Abstract: A method of operating an autocorrelation pitch detector for use in a vocoder overcomes the pitch doubling and tripling problem using a heuristic rather than an analytic approach. The process tracks the times of occurrence of a highest and a second-highest autocorrelation peak. The amplitudes of the highest and the second-highest autocorrelation peaks are compared and, when these peaks are within a predetermined percentage difference in amplitude, the ratio of the time position (IPITCH2) of the second-highest peak to the time position (IPITCH) of the highest peak is checked to determine if that ratio is 1/3, 1/2 or 2/3, within a predetermined error limit .epsilon.. If so and if the ratio is either 1/2 or 1/3, then IPITCH is set equal to IPITCH2 as reepresentative of the pitch period while, if the ratio is 2/3, then IPITCH is divided by three in order to represent the pitch period.

Abstract: A speech recognition apparatus having reference pattern adaptation stores a plurality of reference patterns representing speech to be recognized, each stored reference pattern having associated therewith a quality value representing the effectiveness of that pattern for recognizing an incoming speech utterance. The method and apparatus provide user correction actions representing the accuracy of a speech recognition, dynamically, during the recognition of unknown incoming speech utterances and after training of the system. The quality values are updated, during the speech recognition process, for at least a portion of those reference patterns used during the speech recognition process. Reference patterns having low quality values, indicative of either inaccurate representation of the unknown speech or non-use, can be deleted so long as the reference pattern is not needed, for example, where the reference pattern is the last instance of a known word or phrase.

Abstract: A harmonic signal is created from a limited spectral representation of a voice signal. The harmonic signal is combined with the at least a portion of the limited delayed spectral signal to provide a reconstructed speech signal having perceptually improved audio quality.

Abstract: The speech signal of one speaker in a plurality of signals of speakers can be isolated and recognized by using non-linear oscillators which lock on the fundamental and harmonic frequencies of the one speaker, whereby extraction of the one speaker's signals and speech recognition is done.