Abstract: A speech synthesizer circuit for use in an automobile. The circuit includes a dedicated microprocessor and a speech synthesizer integrated circuit which has an output couplable to the speakers of the automobile's radio. The microprocessor monitors various automobile status conditions and, based upon these conditions, causes various preprogrammed voice messages to be transmitted to the motorist. The circuit includes various novel speech synthesizer features. These features includes novel circuit energization, condition sensing, failure detection, and voice activation mechanisms.

Abstract: A digital speech coding circuit makes use of linear predictive coding, vector quantization and difference, Huffman coding, and excitation estimation to produce digital representations of human speech having bit rates low enough to be transmitted over such channels as telephone lines and at the same time being capable of being synthesized in the receiver portion of the circuit to produce analog speech of high intelligibility and quality. The transmitter portion of the circuit comprises a series connection of a low pass filter, analog to digital converter, linear predictive coding module comprising five resonators for establishing five center frequencies and bandwidths of the analog speech, vector quantization module comprising binary representation of the likely combinations of resonances found in human speech, Huffman coding module, a variable bit rate to fixed bit rate converter, and optionally, an encryption module.

Abstract: A speech recognition method and apparatus employ a speech processing circuitry for repetitively deriving from a speech input, at a frame repetition rate, a plurality of acoustic parameters. The acoustic parameters represent the speech input signal for a frame time. A plurality of template matching and cost processing circuitries are connected to a system bus, along with the speech processing circuitry, for determining, or identifying, the speech units in the input speech, by comparing the acoustic parameters with stored template patterns. The apparatus can be expanded by adding more template matching and cost processing circuitry to the bus thereby increasing the speech recognition capacity of the apparatus. The speech processing circuitry establishes overlapping time durations for generating the acoustic parameters and further employs a sinc-Kaiser smoothing function in combination with a folding technique for providing a discrete Fourier transform.

Abstract: A speech recognition method and apparatus employ a speech processing circuitry for repetitively deriving from a speech input, at a frame repetition rate, a plurality of acoustic parameters. The acoustic parameters represent the speech input signal for a frame time. A plurality of template matching and cost processing circuitries are connected to a system bus, along with the speech processing circuitry, for determining, or identifying, the speech units in the input speech, by comparing the acoustic parameters with stored template patterns. The apparatus can be expanded by adding more template matching and cost processing circuitry to the bus thereby increasing the speech recognition capacity of the apparatus. Template pattern generation is advantageously aided by using a "joker" word to specify the time boundaries of utterances spoken in isolation, by finding the beginning and ending of an utterance surrounded by silence.

Abstract: Speech recognition calculations are decreased by deactivating (or activating) a word in a grammar graph at the "kernel" level. A word is a sequence of acoustic kernels, each kernel a phoneme spectral vector with min-max duration data on a template.

Abstract: In speech recognition, an unknown input signal is analyzed for its feature pattern sequence A.sub.i which is compared to prestored reference patterns grouped in categories. Each category m has a representative reference pattern B.sup.m which represents the average value of the n patterns in the catagory B.sub.n.sup.m. A first comparison (prematching) to the representative patterns is followed by a second matching to the actual reference patterns in the most likely category. The representative pattern is used in pre-matching and matching, thereby simplifying calculation and increasing accuracy.

Abstract: In a speech recognition system, the beginning of speech versus non-speech (a cough or noise) is distinguished by reverting to a non-speech decision process whenever the liklihood cost of template (vocabulary) patterns, including silence, is worse than a predetermined threshold, established by a Joker Word which represents a non-vocabulary word score and path in the grammar graph.

Abstract: A speech recognition method and apparatus employ a speech processing circuitry for repetitively deriving from a speech input, at a frame repetition rate, a plurality of acoustic parameters. The acoustic parameters represent the speech input signal for a frame time. A plurality of template matching and cost processing circuitries are connected to a system bus, along with the speech processing circuitry, for determining, or identifying, the speech units in the input speech, by comparing the acoustic parameters with stored template patterns. The apparatus can be expanded by adding more template matching and cost processing circuitry to the bus thereby increasing the speech recognition capacity of the apparatus. The template matching and cost processing circuitries provide distributed processing, on demand, of the acoustic parameters for generating through a dynamic programming technique the recognition decision.

Abstract: Speaker-independent word recognition is performed, based on a small acoustically distinct vocabulary, with minimal hardware requirements. After a simple preconditioning filter, the zero crossing intervals of the input speech are measured and sorted by duration, to provide a rough measure of the frequency distribution within each input frame. The distribution of zero crossing intervals is transformed into a binary feature vector, which is compared with each reference template using a modified Hamming distance measure. A dynamic time warping algorithm is used to permit recognition of various speaker rates, and to economize on the reference template storage requirements. A mask vector with each reference vector on a template is used to ignore insignificant (or speaker-dependent) features of the words detected.

Abstract: A very small, very flexible, high-quality, linear predictive vocoder has been implemented with commercially available integrated circuits. This fully digital realization is based on a distributed signal processing architecture employing three commercial Signal Processing Interface (SPI) single chip microcomputers. One SPI implements a linear predictive speech analyzer, a second implements a pitch analyzer while the third implements the excitation generator and synthesizer.

Abstract: Speech synthesis is selectable in two modes: "normal" mode in which the synthesis filter is responsive to prestored parameters, and "test" mode which executes a checking procedure based on test words from an outside source. Each mode has corresponding instructions prestored in a memory.

Abstract: Natural quality and bit-rate for LPC speech synthesis is improved by encoding the LPC residual signal in a prescribed multipulse format formed for each LPC frame. Voiced, unvoiced, and mixed (hiss plus periodic) excitation is inherent. The speaking-rate is changed by adding, deleting, or repeating pitch-periods, and the pitch (intonation) is changed by adding or deleting zeros in the multipulse excitation signal.

Abstract: A synthetic-speech calculator includes a calculating circuit inclusive of a plurality of memory locations to allow successive or chain calculations by some calculations equations and calculating numerical data, and voice synthesizer for providing pronunciation of selected calculating information. The calculator further comprises a first circuit for loading the some calculation equations and the calculating numerical data to the selected memory locations, a second circuit for indicating or pronouncing the some calculation equations and the calculating numerical data which have been utilized for the calculation and stored in the memory locations, and a third circuit for repeating the calculation by the some calculation equations and the calculating numerical data all stored in the memory locations.

Abstract: Repeated announcements by a talking timepiece are selectable in two modes: predetermined intervals selected by a selector switch, or arbitrary intervals selected by a keyboard.

Abstract: Circuitry usable in a tamper protected recording system for characterizing audio signals at the time of recording including a filter for selecting a portion of the audio signal, a zero crossing detector to detect zero crossings, a threshold detector to determine when the threshold of the detector is satisfied, a first microprocessor for counting the zero crossing detections for successive time intervals and determining for each of the time intervals whether the threshold was satisfied and a second microprocessor for converting the counts and threshold information into data signals that can be recorded with the audio signal.

Abstract: Audio signals are analyzed for predictable components (reflection coefficients) and non-predictable (residual) components. The original signal state, over a short-term interval of samples (packet) is defined as one of four states: Silence, Hiss, Sigma, or Peaky. The state determines the step-size encoding of the residual quantized signal, which can therefor be encoded more efficiently.

Abstract: A voice control system for controlling execution of a computer program includes a microphone (10) operative to receive voice commands having the output thereof connected to a clipping circuit (12) which amplifies and clips the microphone output to generate a zero-crossing signal. The output of circuit (12) is connected to a microprocessor (14) which on detecting speech input interrupts program execution in a CPU (15) and "freezes" display device (16) on which the game is displayed. Microprocessor (14) then processes the input signal to recognize the voice command by determining the relative frequency content of discrete portions of the command. Once it has recognized the command, microprocessor (14) prompts the CPU (15) to execute the command and resume execution of the game program at the point of interrupt. By suspending the progress of the game during the speech input and recognition interval, the voice control system minimizes the time-constraint problems associated with voice input.

Abstract: A coding scheme which uses Shannon-Fano coding for data headers to identify the type of command signal, uses a first set of formant data in the command signal to generate second sets of formant data for sound class initialization, and uses delta modulation to update the initialized sound class and for sound type transitions. The header indicates initialization of sound classes, repeat of the previous command, updating the previous command or end of word. Given types of command signals and sound classes have the same header and the data portion of the command signal defines which type of command signal is present. A unique delta modulation scheme is used wherein an increment, decrement or no change is indicated by a 11, a 00 or a 10 or 01 wherein each pair represents the delta modulation bit for a parameter, one in the present frame and one in the previous frame for that parameter. A repeat code with no data is used when the delta modulation bit for all parameters change from the previous frame.

Abstract: A signal compression system includes a plurality of channels. A plurality of these channels include a bandpass filter (for filtering out all but a portion of an input signal), an intensity detector (for deriving a spectrally weighted estimate of the intensity of a broader spectral portion of the input signal than the bandpass filtered spectral portion), and a divider (for compressing the bandpass filtered spectral portion using a variable gain having a preselected functional relationship to the spectrally weighted intensity estimate). The signal compression system preserves cross-channel information.

Abstract: An improved speech analysis and synthesis system wherein LPC parameters and a modified residual signal for excitation is transmitted: the excitation signal is the cross correlation of the residual signal and the LPC-recreated original signal.