Abstract: A speech detector uses a signal classifier (19) to identify portions of a representation of the average magnitude of a group of signal samples indicative of either speech or noise. A controller (33) in the signal classifier follows a four state sequence using appropriate time constants for signal measures in a variety of signal conditions in defining the speech and noise portions of the representation. A level estimator (21) uses selectively obtained signal measures from the defined portions of the representation to provide adaptively variable decision levels. A speech definer (16) compares the representation to a first decision level and the signal samples to a higher decision level to indicate the occurrence of speech signal activity when either decision level is exceeded. In a two way transmission arrangement, a receive trunk speech detector uses a stretcher (133) to prevent adaptation of the transmit speech detector thresholds when echo signals are present.

Abstract: This closed loop multiplex communication system uses a timing unit to establish a general multiplex format consisting of 1024 frames in a multiframe. Each subscriber station transmits one message bit per frame (i.e. 1024 message bits per multiframe), but transmits only 32 signalling bits per multiframe (i.e. the eight-bit non-message blocks in four corresponding frames).

Abstract: The accuracy of segmenting an utterance into words is improved by the use of sequence-defining signals. An utterance is recognized as one of a plurality of reference word sequences in an arrangement wherein a set of signals is generated to define the syntax, i.e., word arrangements, of the sequences. Each sequence corresponds to a selected series of reference words. A signal is generated to identify the successive word positions of the sequences. Responsive to the sequence defining signals, the utterance, and the reference words, a set of signals is produced, each representing the correspondence between said utterance and one of the sequences. The sequence having the closest correspondence to the utterance is selected.

Abstract: Code converting circuits for facsimile and other information communications systems in which run length codes are converted to codes having variable lengths and the variable length codes are converted into a word code train, wherein each word comprises N bits, and the converted word code train is stored or transmitted. The code converting circuit consists of a code converter for converting parallel input codes (run length codes of fixed lengths) into variable length codes, a multiplexer and shift register for converting the variable length codes (and dummy codes where needed) into word length codes of fixed length, a multiplexer for outputting the word length codes to a memory interjecting a synchronizing code at predetermined intervals, and control signal generating circuitry for controlling the various circuit components. A decoding circuit is also disclosed for reconstructing the original run length codes from the word length codes.

Abstract: A synthetic-speech calculator includes a keyboard consisting of digit keys and function keys and, if desired, or more mode selectors, a desired number of registers for storing numerical information entered by the depression of selected ones of the digit keys, a read-only-memory for storing a large number of digital codes as sound quantizing information, counter means for specifying the address of the memory so as to take a specific digital code out of the memory, a digital-to-analogue converter for convering the specific digital code taken out of the memory into an audible sound signal, and a loud speaker driven by the audible sound signal and producing an audible sound. There are further provided means for producing audible sounds indicative of not only numerical information but also its associated conditional information having a particular meaning with respect to that numerical information, for example, position information, index information, tabulation information, etc.

Abstract: Speech analysis apparatus transduces speech sounds to electrical signals which are fed via a selectable low pass to extract the pitch frequency. The pitch frequency representing signals are fed via a frequency to voltage transducer to a visual display device.

Abstract: A digital signal mixing apparatus for mixing digital signals in S input channels to obtain digital mixed signals in T output channels is provided which includes a memory in which a matrix of S columns by T rows of elements are stored in digital form, a circuit which determines the value of the elements of the S by T matrix in response to a desired mixing ratio between the S-channels of the digital input signals and supplies these matrix elements to the memory, and a circuit which multiplies the digital signals in the S input channels by corresponding matrix elements read out from the memory to generate the digital mixed signals in the T output channels.

Abstract: A speech detector for use in an adaptive hybrid circuit is disclosed. The speech detector detects the instantaneous sign of current and voltage in a hybrid circuit and thereby determines the direction of power flow in the circuit. The direction of power flow is indicative of the presence or absence of a near end talker. The hybrid circuit enables or disables adaption based upon the presence or absence of the near end talker in the telephone system.

Abstract: A stroboscope device for diagnosis of larynx emissions. The system uses a saw tooth wave for generating in synchronization with a vocal cord vibration saw tooth waves whose voltages are constant in minimum and maximum values. A reference voltage generating circuit is used whose output voltage is variable and a discharge tube light emission control signal generating circuit compares the instantaneous voltage of each saw tooth wave with the output of the reference voltage generating circuit to generate a discharge tube light emission control signal in a phase shift range with respect to said vocal cord vibration. The discharge tube light emission control signal may be generated in a desired phase of said vocal cord vibration every period of said vocal cord vibration or at a repetition rate of multiple periods of the vocal cord vibration.

Abstract: An electronic circuit is disclosed which is capable of producing synthesized sounds under the control of a microprocessor or other type computer. A plurality of switch inputs are provided whereby the sound to be produced can be selected. The processor, under program control, then generates such sounds utilizing one or more basic waveforms stored in a memory. These basic waveforms are modified in specified ways and then outputted through an I/O device to a digital to analog converter for application to an audio system. Eight input control lines to the microprocessor also feed an 8-input NAND gate, such that a momentary change of the normally-high control inputs will cause the NAND gate to activate an interrupt request.

Abstract: There is disclosed a digital gain control particularly adapted for use with digital signals propagating in a telephone line circuit. The gain control is positioned in the circuit to respond to a digital signal manifesting a plurality of words, each having a given number of bits N and indicative of a weighted value of an analog signal. The digital signal is multiplied by a word coefficient having a number of bits M indicative of a gain factor to be imparted to the digital signal. A product signal is provided containing M+N bits. The least significant bits are removed to provide an output signal of N bits possessing the desired gain. The discarded bits are employed via a feedback loop and are added to the next word product, which is then used to provide a next output signal. The feedback loop assures that each output signal is a close approximation to the input signal as gain controlled by the desired gain factor.

Abstract: Digital delay apparatus for variable delay uses a shift register fixed delay driving a random access memory variable delay. A variable modulus counter controls the read-write addressing, whereby the difference between addresses sets the variable delay of the random access memory. The random access memory capacity is small relative to the shift register for increased efficiency.

Abstract: In a preferred embodiment of the invention the audio signal is bandwidth limited before being differentiated to provide either a positive or negative signal (depending on the slope of the audio signal) to a comparator which provides a corresponding logical one or zero at its output. A digital logic circuit connected to the comparator output is clocked to provide a digital output signal transition from a logical one to a zero or vice versa if its input has changed from a logical one to a zero or vice versa at least once during the clock period. The clocked digital output of the logic circuit is provided as an input to a computer where it is stored. The digital data is subsequently read out of the computer and provided to a filter and integrator to provide an audio signal replica of the filtered input audio signal.

Abstract: An analog signal processing system is described for changing the frequency of analog signals in a real-time manner. The system includes at least two analog shift registers, (specifically, three analog shift registers), an input circuit connecting the input end of all the registers in parallel, an output circuit selectively connectable to the output ends of the registers, a high frequency clock pulse source, a lower frequency clock pulse source, and cyclically operable switching means effective to load from the input circuit at least one analog shift register at the rate corresponding to the frequency of one of the clock pulses, while unloading at least one other register into the output circuit at the rate corresponding to the frequency of the other clock pulses.Two applications are described, for purposes of example. One application is as a hearing aid device to reduce the frequency of audio signals in a real-time manner for a person having a limited-frequency audibility.

Abstract: This speech signal digital interpolator increases data (and date rate) by first using an interface unit for rate-changing and zero-stuffing a 32 Kiloword/sec (13 bits/word) signal to form a 1.024 Megaword/sec (13 bits/word) signal, which is then smoothed by a low-pass filter and approximated with 3-bit words. The interface circuit repeats each word a given number of times at the increased rate with each word separated by an all zero bit word having the same number of bits as the digital word to provide an output signal of a specified pattern for each word in the digital signal. A low pass recursive filter responds to the output signal from the interface means and is operative to provide a filtered output digital signal having smooth transitions between each output digital word with said filter output signal at said increased rate and with each output filtered digital word being of a lesser number of bits than the bits in said original signal.

Abstract: A digital to analog converter is employed in the digital line circuit of a telephone system and operates to convert a digital signal indicative of an analog speech signal back into a replica of the analog signal. The converter operates with an interpolated input digital signal to detect by means of a sign bit, the characteristic of an input digital word as being indicative of a positive or negative level. An error correcting signal is provided by the converter which is added to the next digital word to provide a compensated word having a sign bit determined by the remainder and the sign bit of the previous digital word. This word is then processed in sequence to produce an output pulse stream from the sign detector indicative of successive positive or negative values as defined by the input digital words, each of which are modified according to the error correcting signal.

Abstract: A sampled speech compression and expansion system, for two-dimensional prssing of speech or other type of audio signal, comprises transmit/encode apparatus and receive/decode apparatus.The transmit/encode apparatus comprises a low-pass filter, adapted to receive an input signal, for passing through low-frequency analog signals. A converter is connected to the low-pass filter for converting the analog signal into a digital signal. A buffer memory, whose input is connected to the converting means, stores the digitized signals.A correlator, having inputs from the A/D converter and the buffer memory, correlates the digital signal received directly from the converter with a delayed signal from the buffer memory. An "interval-select" circuit, whose input is connected to the output of the correlator, uses the autocorrelation value as a basis for comparison with subsequent peaks in the correlation value which are greater than a specified fraction of the autocorrelation value.

Abstract: For the multiplex transmission of coded speech signals in periodic frames, single segments (blocks of coded samples) are selectively suppressed for redundancy reduction, and are replaced on the receiver side by optimally correlated subsections of equal length from previously transmitted segments. On the transmitter side, a multiplicity of compare operations are made for each speech signal, between the respective newest coded segment and a step wise shifted subsection window of previous segments, to determine the best correlated subsection, i.e. the one which is most suitable as replacement and the respective relative offset and correlation. From a group of speech signals, the one signal, or several signals, having the best correlation is selected for suppression of a segment before transmission. Instead of each missing segment an indication of the corresponding optimum offset is transmitted in the frame header.

Abstract: An audible output device useful in timepiece or calculator devices, features a prestored and preselected order of digital codes representing speech words and pauses, to be outputted through gate circuitry responsive to the pause codes.

Abstract: A phoneme based speed synthesizer that utilizes an analog delay line (ADL) vocal tract which simulates the variations in the acoustical characteristics of the human vocal tract which occur as a result of changes in the cross-sectional area of the human vocal tract at different points along its length. The ADL vocal tract comprises a plurality of T-sections having a resistor as the series component and a frequency dependent negative resistance (FDNR) as the shunt component. Both the series and shunt components are readily tunable via electric control signals, although in the preferred embodiment only the series resistive elements are tuned. Except for the vocal excitation source, the ADL vocal tract is driven entirely by digital circuitry, including novel digital transition circuitry for producing gradual variations in the values of the control signals as they change from phoneme to phoneme.