Abstract: A digital audio signal generating apparatus suitable for the application to electronic musical instruments, sound effect generators for amusement machines and the like. This digital audio signal generating apparatus has a signal processing section and a memory used to perform a speech synthesis, wherein the signal processing section utilizes a vacant area of the memory to perform a delay processing to add a reverberation sound. Thus, the number of required memories can be reduced and the arrangement of the apparatus can be simplified.

Abstract: Disclosed is an apparatus for checking audio signal processing systems. The apparatus has the following features:the apparatus is provided with a first input connection, to which the input signal of the audio processing system to be checked is transmitted, a second input connection, to which the output signal of said system is transmitted, and a signal processor.

Abstract: A speech signal transmission system for transmitting coded speech signals with variable bit rate is disclosed, which comprises a coder for analyzing digital speech signals inputted in a one-frame period and transforming them into coded data comprising a plurality of parameters indicating characteristics of the inputted speech signals, a data arranging circuit for arranging the coded data outputted by the coder in the order of the priority in the decoding of the speech signals and outputting them, and a bit stealer allowing a series of coded data outputted successively by the data arranging circuit to pass through only in a period of time determined by the transmission bit rate. The bit arranging circuit outputs the parameters in the order of the priority of decomposes each of them in unit of a bit and outputs them with decreasing priority. Parameters or bits of low priority are omitted by the bit stealer, depending on the transmission bit rate.

Abstract: A circuit for generating a sequence of pseudo random numbers, {A.sub.k }. There is an exponentiator (30') in GF(2.sup.m) for the normal basis representation of elements in a finite field GF(2.sup.m) each represented by m binary digits and having two inputs and an output from which the sequence {A.sub.k } of pseudo random numbers is taken. One of the two inputs is connected to receive the outputs {E.sub.k } of maximal length shift register of n stages. There is a switch (32) having a pair of inputs and an output. The switch (32) output is connected to the other of the two inputs of the exponentiator (30'). One of the switch (32) inputs is connected for initially receiving a primitive element A.sub.0 in GF(2.sup.m). Finally, there is a delay circuit (34) having an input and an output.

Abstract: A D-A converter outputs feed pitch error signals on an exposure head. The error signals are inverted to be supplied to a multiplier, and non-inverted to be supplied to a second multiplier. The exposure head radiates multiple beams, and ON/OFF dot signal for odd channels thereof are supplied to the multiplier and ON/OFF dot signals for even channels are supplied to the second multiplier respectively. The multipliers multiply the supplied signals, whereby values of the ON/OFF dot signals are changed in response to the errors. Thus, the beams outputted from the respective channels are controlled in power intensity, whereby diameters of printing spots on photosensitive material are changed per channel.

Abstract: An arrangement for a speech enhancement processor which maintains the processed speech at a constant level regardless of large changes in the associated noise level. The composite speech and noise signal is applied to a first AGC circuit and then to a speech enhancement system which removes tonal, impulse, and wideband noises from the signal. The extracted noise power estimates are subtracted from the constant amplitude signal to provide a gain control signal value to which the gain of a second variable gain amplifier is inversely proportional. The amplifier multiplies the processed speech output from the enhancement system and, because of the variable gain control, provides an output speech signal having short-term amplitude levels which correspond to those of the input speech signal, and having a constant long-term amplitude level.

Abstract: Speech words are recognized by first recognizing each spectral vector identified by a label (feneme), then identifying the word by matching the string of labels against phones using simplified phone machines based on label and transition probabilities and Merkov chains. In one embodiment, a detailed acoustic match word score is combined with an approximate acoustic match word score to provide a total word score for a subject word. In another embodiment, a polling word score is combined with an acoustic match word score to provide a total word score for a subject word. The acoustic models employed in the acoustic matching may correspond, alternatively, to phonetic elements or to fenemes. Fenemes represent labels generated by an acoustic processor in response to a spoken input. Apparatus and method for determining word scores according to approximate acoustic matching and for determining word scores according to a polling methodology are disclosed.

Abstract: Speech synthesis is improved by using normalized values of pitch and amplitude data to modify phoneme signals which are further processed by a quantizing noise filter which computes and interleaves the average value between adjacent samples.

Abstract: Waveform significant event parameters based on the duration of a first running average of the duration of positive slope portions and a second running average of the duration of negative slope portions of the wave, include occurrences of dwells, creeps, and rapid changes in the values of the running averages.

Abstract: Speech versus music is decided by comparing the presence of pauses greater than 32, 60 and 500 ms in the lower input spectrum, and 4 ms in the higher input spectrum.

Abstract: In order to provide a visual indication of the location of error in stored programs and allow easy and prompt correction for the error, a programmable calculator is provided with both a display device (preferably, a cathode ray tube) capable of displaying the contents of a plurality of consecutive program steps in a program memory at the same time and adapted for selectively displaying the plurality of the consecutive program steps in a stepwise manner and a sound or voice synthesizer for sequentially providing verbal indications of the contents of the consecutive program steps being visually displayed.

Abstract: The process uses a set of data characteristic of the speech signal, supplied by processing circuits: measurements of the time intervals between zero crossovers and measurements of the energy in the half-waves of this signal. The test procedure implemented by a microprocessor selects the half-waves whose energies exceed thresholds characterizing pitch period beginnings. These thresholds are predetermined for the first two successive sums selected, then depend on the energy values of the preceding half-waves selected differently according as to whether the voiced character of the signal is acquired or not. Complementary tests are used for minimizing detection errors.

Abstract: Compression and synthesis techniques and related apparatus for time domain signals, particularly signals whose information content resides in the power spectrum such as speech and more particularly signals whose amplitude is aperiodic, such as unvoiced speech sounds. Compression techniques include eliminating serially redundant segments of information. Synthesis, particularly of unvoiced sounds which are sensitive to injected artificial periodicity, involves repeating sequentially portions of the same segmentrepresentative of the source signal, including commencing and terminating at different points each repetition, varying the length of the portion and reproducing the portion forwards and backwards in time. The invention finds application in speech compression and compact speech synthesis devices.

Abstract: Adaptive bit allocation optimizes the encoded transmission of speech signal parameters. Allocation is controlled by a voiced/unvoiced decision signal derived from occurrence rate distributions of Partial Correlation Coefficient K1.

Abstract: Integrated circuit speech synthesis system utilizing complementary metal-insulator-semiconductor technology to achieve low voltage operation, wherein a pluse width modulated digital-to-analog converter is employed to provide for accurate conversion of digital signals into analog signals even though the low voltage operation prohibits the large voltage swings normally required for digital-to-analog converter circuitry. The speech synthesis system includes a linear predictive filter as a speech synthesizer which utilizes coded reflection coefficients to produce digital signals representative of human speech. A microprocessor controls the access of digitized speech data which is stored in a memory. The speech synthesizer and microprocessor along with the pulse width modulated digital-to-analog converter are implemented in complementary metal-insulator-semiconductor technology.

Abstract: A voice discriminating system is disclosed which includes input circuitry for a pair of microphones, circuitry for detecting the presence of voiced sounds inputted to the microphones, selectively enabled circuitry for detecting the presence of unvoiced sounds inputted to the microphones, and a microcontroller which executes functions under the control of the presence or absence of voiced or non-voiced sounds inputted to the microphones. The disclosed system further includes a display circuit and a sound circuit that are controlled by the microcontroller for the purpose of playing games wherein the players provide spoken commands to the microphones.

Abstract: This improved voice-excited speed synthesizer expands the voice-band spectrum to generate high-frequency band excitation signal by using two separate distortion networks: a full-wave rectifier for lower frequencies, and a limiter for higher frequencies. Channel and formant type voice excited embodiments are shown.

Abstract: A speech synthesis system utilizing a linear predictive filter. Voiced and unvoiced excitations are applied to the filter to produce a digital signal representative of human speech. The voiced excitation is provided by a repeating chirp function stored in memory. The unvoiced excitation consists of two excitation signals of opposite sign, stored in programmable memory and randomly addressed. The programmable storage of unvoiced excitation signals allows gain scaling between voiced and unvoiced excitation to be easily accomplished.

Abstract: In a communication system an input predictive type signal is analyzed in successive time intervals to generate a set of prediction signals for the interval. A predictive residual signal is produced jointly responsive to the input signal and the interval prediction signals. The predictive residual signal is quantized and encoded. The quantization includes dividing the predictive residual signal into a plurality of distinct portions and selectively quantizing the distinct portions to improve intelligibility.

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.