Abstract: The output of an audio power amplifier fed to a loudspeaker driver is sensed and when this output exceeds a predetermined level, a switch control circuit operates to place a current limiting device, such as a power resistor, in series with one of the driver feed lines, thereby reducing the power fed thereto. An indicator may also be provided to indicate when the overload condition exists. Operation is made more accurate and distortion minimized by employing separate sensing and power channels in the circuit, the first of these channels operating to sense the overload while the second of these channels utilizes a portion of the amplifier output for driving the switch control circuit for placing the power limiter in the speaker line. The sensing channel is designed to sense the average voltage of the amplifier output, thus providing control more directly responsive to power. The power channel is designed to use audio power from the amplifier output only occasionally, thus producing a minimum of distortion.

Abstract: A speech recognition system adaptable to noisy environments is disclosed. The system includes a recognition unit for recognizing input speech signals and a noise measuring unit for measuring the intensity of ambient noises. The system also includes a rejection unit responsive to a rejection standard controlled by the intensity of the measured noise for rejecting the rejection results given from the recognition unit when the rejection standard is exceeded.

Abstract: A processor for audio program material which synthesizes simulated reflection and reverberation signals for use in creating the illusion of a specific modeled listening environment. A preferred embodiment of the processor accepts stereophonic audio signals and simulates the multiple reflections of an acoustically modeled environment by recirculating the signals through a pair of digital signal delays which have different delay times. The analog input signals are sampled and converted to digital form by first attenuating the signals with a digitally controlled step attenuator and then encoding the amplitude of the attenuated signal using a ramp comparison technique. The encoded amplitude signals, and a code representative of the attenuator position, are stored in a random access memory and are retrieved after the desired delay. The retrieved digital data are converted back to analog signals by reversing the encoding process, using the same ramp voltage generator.

Abstract: The accuracy of the conversion of an analog signal to digital form is improved by amplifying the input signal by different factors and automatically utilizing the highest amplified input signal which has a voltage level within operational limitations in the analog to digital conversion process. Output data from a digital computer, based on the input data, is converted from digital form to analog form and is routed to a required location by the use of control logic and optically isolated digital signals used to address other computers.

Abstract: The output signals of a phonograph and of an auxiliary audio signal source, such as a high impedance microphone, electric guitar, or the like, are combined before being fed to the amplifier of the phonograph, whereby the combined signals are amplified without distortion, and are fed to one or more speakers, thus audibly reproducing the phonograph recording with the auxiliary sound signal superimposed thereover. The device which combines the signals includes means for suppressing the base frequencies in the auxiliary signal, and for selectively adjusting the amplitudes of the two signal components.

Abstract: A moving-average notch filter with a well-defined phase characteristic for use in eliminating oscillation frequencies in a sound amplification system, wherein this moving-average filter is designed to have an output Y.sub.K in accordance with the system equation ##EQU1## where b.sub.1 are the weighting coefficients, X.sub.K-i are input samples, and K is a constant. This filter may be designed to have a linear phase characteristic. In one embodiment, the location of the notch in the frequency response of the filter may be made to automatically track the drift of an acoustic oscillation frequency.

Abstract: An apparatus and method of voice encryption uses segment swapping. Features of the invention include weighting the input time-function segments by a Hamming or Hanning Window function, before converting to frequency domain segments.

Abstract: A speech digitizer is disclosed including an analyzer for generating power and filter coefficient parameters representative of an analog speech waveform. The digitizer also includes a pitch detector for generating a digital pitch parameter substantially representing the fundamental periodicity of the waveform and including range restrictor means for restricting the pitch signal to a range of pitches within a predetermined tolerance if the average pitch of the periodicity signal is below a predetermined level. The pitch detector also includes means for determining the number of extreme maximum and minimum points within a predetermined range of an absolute magnitude difference function thereby generating a structure number signal representing a voiced event. The digitizer includes a voicing detector for generating a three-level voicing/unvoicing parameter representing whether the speech waveform is voiced or unvoiced.

Abstract: An antisidetalk signal control circuit for preventing transmission of signals from a first source connected with a first input circuit to an output circuit while allowing the transmission of signals from a second source connected with a second input circuit to the output circuit. The signal control circuit comprises a first switching circuit having an input coupled to the first input circuit, a second switching circuit having an input coupled to the first output circuit through a phase inverter and a third switching circuit having an input coupled to the outputs of the first and second switching circuits. An oscillator drives the first switching circuit with signals having a given phase and the second and third switching circuits with signals of opposite phase. A low-pass filter couples the junction of the first and second switching circuits to the second input circuit and a feedback circuit couples the second input circuit to the junction of the first and second switching circuits.

Abstract: The addition of two digitized audio signals, each of which has been non-linearly compressed and PCM encoded, is accomplished by converting only one of the signal samples (preferably the smaller) to an exponential-format value which may be linearly added to the other.

Abstract: To transmit a number of individual speech channels over a smaller number of transmission channels, the frequency range of each speech channel is broken into sub-channels and each of these is considered separately for operational activity. Composite speech signals are then formed from the active frequency sub-channels of the individual speech channels and these are transmitted with coding signals indicative of their composition.

Abstract: A multi way speaker system comprises an amplifier, frequency dividing network including a plurality of filters, a plurality of speakers respectively responsive to the output signals of the filters, and a summing circuit responsive to the output signals of the filters for producing a composite signal to be fed back to the input of the amplifier to establish a negative feedback loop. Because of the feedback loop undesirable influences by various elements included in the filters are diminished and thus the speakers reproduce audio frequencies with high fidelity. At least one equalizer may be employed in the feedback loop, while auxiliary filters may be used in such a manner that the output terminals of the auxiliary filters are connected to only the summing circuit so that the output signals of the auxiliary filters are added to an output signal of at least one filter of the frequency dividing network.

Abstract: Disclosed is apparatus for and a method of decreasing the time duration of playback of recorded speech without distortion. Speech recorded at a normal rate is played back at a higher rate using two variable length buffers in alternating fashion to receive sampled speech segments. Threshold analysis logic circuitry is provided to continually compare the output voltage levels of both buffers and to switch the output between the buffers to a using device when the voltage level of both buffers is approximately zero and of an equal amplitude. In this way output to the using device consists of segments from both buffers joined with a minimum of generated noise and no associated dead zone without regard to any predetermined segment length.

Abstract: A pre-processing system for a speech recognition system nullifies the distortion in a received signal caused by the transmission line by forming a normalized received signal whereby the distortion is cancelled: the received signal is spectrally averaged to form a mean spectrum which is used to adjust a filter to a characteristic inverse to the distortion; this filter then filters the received signal to form the normalized received signal s.sub.j '(.omega.,t)=S.sub.j (.omega.,t)/S.sub.j (.omega.) which is equal to the normalized source signal =S.sub.i (.omega.,t)/S.sub.i (.omega.).

Abstract: An adaptor is disclosed for use with audiometers to enable the measurement of the masking level difference (MLD) in a person's hearing capability. The adaptor comprises a signal channel and a noise (masker) channel which are connected respectively with the audiometer outputs and the earphones. Means are provided to apply the signal voltage and the noise voltage to respective mixers in the two channels and for selectively inverting the phase of the signal voltage at one of the mixers and for selectively inverting the phase of noise voltage at one of the misers. Various MLD test conditions can be established including binaural and monaural conditions and adjustable intensity at one ear.

Abstract: Each of a pair of ear pads is provided with an electrically-operated earphone. The earphones are connected to each other by a circuit having a cord for connecting the earphones to an external audio-signal source. The circuit includes a pair of Zener diodes connected in back-to-back or opposed parallel relation to each other, and in parallel to the earphones so that if an incoming AC audio signal exceeds a predetermined value one of the diodes will fire to short the input signal, thus limiting the output (in decibels) of the earphones. A field-effect transistor is connected in series with the earphones to limit the current load thereto.

Abstract: A TDMA multiplexer-demultiplexer includes a plurality of input/output ports and common control equipment. Each input port is capable of accepting information asynchronous to the TDMA clock and at typical terrestrial clock rates. The input/output ports perform all the necessary signal processing for a TDMA terminal including pulse stuffing, changing continuous asynchronous low rate information to high bit rate burst form as well as forward acting error correcting encoding, if desirable. By incorporating all the signal processing in the multiplexer-demultiplexer, much equipment duplication is eliminated along with eliminating a requirement for multiple high frequency control and timing signal line drivers and buffers. More particularly, the transmit side of each port includes a separate elastic buffer to raise the input rate to a synchronous common higher rate. In addition, on the receive side, clocking on the low speed side of the expansion buffer is derived from the station's transmit clock.

Abstract: This invention recovers the time waveform envelope of a human voice signal ithout the loss of subtleties providing speaker recognition cues and voice quality.This is accomplished by rectifying the voice envelope and then obtaining slope information about the original time waveform to detect peaks which are successively held by sample and hold circuitry.

Abstract: In a speech processing arrangement for synthesizing more natural sounding speech, a speech signal is partitioned into intervals. For each interval, a set of coded prediction parameter signals, pitch period and voicing signals, and a set of signals corresponding to the spectrum of the prediction error signal are produced. A replica of the speech signal is generated responsive to the coded pitch period and voicing signals as modified by the coded prediction parameter signals. The pitch period and voicing signals are shaped responsive to the prediction error spectral signals to compensate for errors in the predictive parameter signals whereby the speech replica is natural sounding.

Abstract: Apparatus, and a related method, for use with a speech analysis system, which derives coefficient parameters by linear prediction and utilizes an intermediate correlation technique that produces a set of auto-correlation coefficients. The apparatus includes an estimate register, circuitry for comparing the contents of the estimate register with a first auto-correlation coefficient and setting the estimate register to the value of the first coefficient if the first coefficient is less than or equal to the estimate register contents, and circuitry for incrementing the contents of the estimate register if the first coefficient is greater than the estimate register contents.