Abstract: A process of speech synthesis from diphones stored in a dictionary as waveforms, for text-to-speech conversion, comprises supplying a sequence of phoneme codes and respective prosodic information, and, for each phoneme, analyzing and synthesizing each phoneme, and then concatenating the synthesized phonemes. For each phoneme, two diphones are selected among the stored diphones and the presence of voicing is determined. For voiced phonemes, the respective waveforms of the two diphones constituting the phoneme are filtered by a window which is centered on a point of the selected waveform representative of the beginning of a pulse response of vocal cords to excitation thereof. The window has a width substantially equal to twice the greater of the original fundamental period and the fundamental synthesis period and has an amplitude progressively decreasing from the center of the window.

Abstract: A process and apparatus for recalculating the contents of a spreadsheet's cells when they are affected by a modification of one or more cells in the spreadsheet is disclosed. After a spreadsheet has been modified, each cell in the spreadsheet is examined only once to determine the order of the cell recalculation and only the affected cells are recalculated. A list indicating which cells have been modified is maintained at all times in the computer prior to recalculation. When the computer is instructed to update the values in the spreadsheet cells, the recalculation method begins by placing an indication of cells affected by modification, one by one, on either of two lists independently maintained by the computer in storage: the final list (L LIST) if all cells in its Dependency Set (the set of cells affected by a modification to this cell) are already in the final list, or the intermediate list (R LIST) if it has a cell in its Dependency Set which is not already in the final list.

Abstract: An apparatus and method for modifying a speech waveform using sinusoidal speech model parameters, includes finding a net masked threshold for each sinusoid for a normal-hearing subject, and adding the effects of impairment and obtaining an impaired masked threshold. The method also includes finding gain needed for each sinusoid so that its distance above the impaired masked threshold is equal to the distance above normal masked threshold, and multiplying sinusoid amplitudes by the gain. The sinusoidal model is used to address the problem of spread of masking within internal speech components by determining the amount of masking that occurs between surrounding sinusoids. The masked threshold for each sinusoid is determined based on the additive effects of masking by other sinusoids in each frame.

Abstract: A speech coding system operated under a known code-excited linear prediction (CELP) coding method. The CELP coding is achieved by selecting an optimum pitch vector P from an adaptive codebook and the corresponding first gain and, at the same time, selecting an optimum code vector from a sparse-stochastic codebook and the corresponding second gain. Special code vectors are loaded in the sparse-stochastic codebook, which code vectors are hexagonal lattice code vectors each consisting of a zero vector with one sample set to +1 and another sample set to -1.

Abstract: The present invention relates to a digital signal encoding apparatus in which an input digital signal has its frequency range divided into a number of frequency bands so that the bandwidths are broader at higher frequencies. A first allowable noise level based on the energy levels in the respective bands, and a second allowable noise level based on the energies of the signals temporally adjacent to the signals of a frequency band under consideration, are set. Signal components of each frequency band are quantized with a number of bits corresponding to a difference between an output synthesized from the first and second noise levels and the energy level of each frequency band. In this manner, the bit rate may be lowered while deterioration in the sound quality is minimized.

Abstract: A voice recognition system comprises a handset and a hands-free microphone for generating an input audio signal, a high-pass filter for eliminating low frequency components from the signal from the handset or hands-free microphone, a signal level controller for adjusting the level of the high-pass signal in response to the user of either the handset or hands-free microphone, a storer for storing the speech data and a controller for controlling the storer so that a user's utterance is stored or the user's utterance is recognized by comparing the utterance to speech data already stored. The handset hook switch provides an on-hook control signal to reduce amplifier gain during hands-free microphone operation.

Abstract: A digital transmission system has a transmitter (3,6,9) and a receiver (13,16,19). The transmitter includes a coder for subband coding of the digital signal, by dividing the digital signal band into successive subbands having approximately equal bandwidths. The receiver recontructs a replica of the digital signal by merging the subbands to the digital signal band. The transmitter includes analysis filter means (6) and receiver includes synthesis filter means (16). The filter coefficients in the analysis filter means are unequal to the filter coefficients in the synthesis filter means.

Abstract: A low-bitrate (typically 8 kbit/s or less), low-delay digital coder and decoder based on Code Excited Linear Prediction for speech and similar signals features backward adaptive adjustment for codebook gain and short-term synthesis filter parameters and forward adaptive adjustment of long-term (pitch) synthesis filter parameters. A highly efficient, low delay pitch parameter derivation and quantization permits overall delay which is a fraction of prior coding delays for equivalent speech quality at low bitrates.

Abstract: In a voice coding system, the baseband or residual signal is encoded at a lower rate by finding a best estimate at a lower rate. The voice terminal signal x(n) is split into a low-pass filtered band signal y1(n) and a high-pass filtered band signal y2(n). Both y1(n) and y2(n) signals are coded into lower-rate sub-sequences of samples x1(n), x2(n) and x3(n), x4(n) respectively. The sequence of samples to be representative of x(n) is selected among x1(n), x2(n), x3(n) and x4(n) for being the closest to x(n).

Abstract: Disclosed is a system and method for generating text from a voice input that divides the processing of each speech event into a dictation event and a text event. Each dictation event handles the processing of data relating to the input into the system, and each text event deals with the generation of text from the inputted voice signals. In order to easily distinguish the dictation events from each other and text events from each other the system and method creates a data structure for storing certain information relating to each individual event. Such data structures enable the system and method to process both simple spoken words as well as spoken commands and to provide the necessary text generation in response to the spoken words or to execute an appropriate function in response to a command. Speech recognition includes the ability to distinguish between dictation text and commands.

Abstract: A CT or other radiographic scanner (A) generates data that is arranged into sets (32). Each set is convolved (40) with a convolution function (42) and backprojected (44) into an image memory (46) along a corresponding one of a plurality of rays. A corresponding gradient image (52) in which each pixel value has either a one or a zero value is forward projected (54) and compared (60) with a standard. The comparison indicates along which rays data sets including bad data were projected. To subtract the bad data contribution from the image, the image representation is forward projected (90) along the identified rays, convolved (40) with a negative of the convolution function (84), and backprojected (44) along the identified ray into the image memory (46). Further correction may be obtained by replacing the subtracted data with interpolated data.

Abstract: In a speech encoding apparatus, a pitch of an input speech signal is analyzed, and a basic waveform of one pitch of the input speech signal is derived. A number of a pair or pairs of pulse elements of a desired framework is decided, and the desired framework is generated in response to the basic waveform. The generated desired framework is encoded. An inter-element waveform code book contains predetermined inter-element waveform samples which are identified by different identification numbers. Inter-element waveforms which extend between the elements of the framework are encoded by use of the inter-element waveform code book.

Abstract: A system and method are described for automatic analysis of image information, such as dimensional, surface, internal or other information. Analysis is controlled by the electronic analysis of speech signals generated by an operator or by a combination of speech generated signals and devices such as switches and variable potentiometers, capacitors or inductors. Electronic computers are employed to analyze the operator's speech signals and to generate control signals.

Abstract: A voice signal processor features a particular improvement of the S/N ratio. In the voice signal processor, the signal level in the voice band of a signal from which noise is cancelled to some extent is emphasized relative to the signal level in the noise band. Moreover, a cancellation factor is utilized in cancelling the noise, so that the voice level in the voice band is emphasized, or the noise level in the noise band is attenuated, achieving a better noise-suppressed voice signal.

Abstract: Speech recognition is carried out by performing a first analysis of a speech signal using a Hidden Semi Markov Model and an asymmetric time warping algorithm. A second analysis is also performed using Multi-Layer Perceptron techniques in conjunction with a neural net. The first analysis is used by the second to identify word boundaries. Where the first analysis provides an indication of the word spoken above a certain level of confidence, an output representative of the word spoken may be generated solely in response to the first analysis, the second analysis being utilized when the level of confidence falls. The output controls a function of an aircraft and provides feedback to the speaker of the words spoken.

Abstract: A speech signal communication apparatus used in a confidential communication system, which performs blockwise processing to achieve a high level of confidentiality, free from waveform discontinuity at the block boundary in a restored speech signal. This apparatus includes a calculating unit for calculating linear predictive coefficients of the input speech signal and a filter to inversely filter the speech signal by using the linear predictive coefficients. The filter flattens the spectral envelope of the input speech signal and produces a predictive residual signal therefrom. This apparatus also includes a removing unit for adaptively removing a low power frequency component from the frequency components of the predictive residual signal. The removing unit uses the linear predictive coefficients to determine which frequency components are to be removed. The linearly predictive coefficients are then converted into a signal having the removed frequency components therein.

Abstract: A method and apparatus for reading out a feature parameter and a driver sound source stored in a VCV (vowel-consonant-vowel) speech segment file, sequentially connecting the readout parameter and the readout sound source information in accordance with a predetermined rule, and supplying connected data to a speech synthesizer, thereby generating a speech output, includes a memory for storing the average power of each vowel, and a power controller for controlling the apparatus to normalize a VCV speech segment so that powers at both ends of each VCV segment coincide with the average power of each vowel.

Abstract: A method of speech recognition includes the steps of predetermining a series of parameters a.sub.1, a.sub.2, . . . , a.sub.J representative of standard patterns of speeches of preset words, where the letter J denotes a predetermined natural number; deriving parameters x.sub.i representative of data of respective frames of an input signal, the adscript i denotes a frame number; calculating similarities d.sub.j.sup.(i) between the parameters a.sub.j and the parameters x.sub.i, where j=1, 2, . . . , J; calculating parameters R.sub.j.sup.(i) for the respective preset words by referring to the following recurrence formulas:R.sub.1.sup.(i) =d.sub.1.sup.(i)R.sub.j.sup.(i) =d.sub.j.sup.(i) +opt(R.sub.j-1.sup.(i-h),R.sub.j-1.sup.(i-h-1),R.sub.j-1.sup.(i-h-2), . . . ,R.sub.j-1.sup.(i-h-m))where j=2, 3, . . .

Abstract: A signal processing apparatus extracts a speech signal from an inputted noisy speech signal. In the signal processing apparatus, a band division process including a Fourier transformation is performed for an inputted speech signal, thereby outputting spectrum signals of plural channels, and a cepstrum analysis process is performed for the spectrum signals of plural channels, thereby outputting a cepstrum analysis result. Thereafter, a speech signal interval of the inputted noisy speech signal is detected in response to the cepstrum analysis result, and then, a speech signal is extracted from the inputted noisy speech signal according to the detected speech signal interval.

Abstract: A document edition system includes a microphone to input audio information; an analog to digital converter to convert the audio information into digital information; a memory unit to store the digital audio information into an external storage unit as information to be added to document information; a sound pressure display control unit to display the sound pressure information of the digital audio information to a display device; a digital to analog converter to regenerate the digital audio information from the memory unit into speech; and a device to add the audio information to an arbitrary position of the document information. The sound pressure display control unit can display the speech regenerating position from the digital to analog converter onto a sound pressure display waveform and also includes a device for displaying the sound pressure information by a predetermined pattern width or a tone.