Abstract: A new way of determining correlation coefficients for stochastic codebook vectors for CELP coding of speech takes advantage of the sparsely populated nature of stochastic codebook vectors. N valued input signals (e.g., convolution vectors) to be correlated with N valued codebook vectors are fed to an N by N multiplexer or other selection means and the signal values either passed to an accumulator or not according to the state of N select inputs or other identification means determined from a memory store (e.g., an EPROM) whose entries correspond to the non-zero values of the codebook vectors. The accumulator output is the correlation of the codebook vector with the input signal. A sequencer steps through the entire codebook to provide correlation values for each vectors. The results are used to determine the optimum stochastic codebook vector for replicating the particular speech frame being analyzed.

Abstract: A device for compressing a digital audio signal, includes a device for allocating bits available for the transmission or storage of the signal, controlling means for the adaptive quantization of the signal, in order to enable a major reduction in the bit rate while at the same time preserving the quality of the starting signal to the maximum extent. The device includes means for allocating a specific number of bits for the expression of the coefficients of each frequency band of a transformed digital audio signal, as a function of a piece of auxiliary information corresponding to a description of the spectrum of the signal, said device being informed by means for the prior elimination of spectral components of said transformed signal as a function of a psychoauditive criterion.

Abstract: A digitally implemented central dictation system wherein users designated as dictators may input Voice Files for later retrieval and transcription by a second group of users designated as transcriptionists. A system Supervisor has the capability to monitor operation of the system and allocate system resources. The system further includes an optional Voice Mail capability.

Abstract: A new way of determining autocorrelation coefficients for adaptive codebook vectors for CELP coding of speech simplifies and improves the accuracy of the autocorrelation coefficient determination for the situation where the codebook vector length being analyzed is less than a speech frame length. This is important in synthesizing short pitch period speech. Copy-up of the shortened codebook vector to equal the frame length is not needed and autocorrelation coefficient errors associated with copy-up are avoided. The improved system relies on calculating autocorrelation coefficients of the first (shortest) vector and then obtaining subsequent autocorrelation coefficients for successive vectors of increasing length by a simple end correction technique until the vector length equals the frame length. The autocorrelation coefficients are scaled by multiplying them by the ratio of the frame length to the vector length.

Abstract: A speech encoder is disclosed, in which, of the DCT coefficients after the discrete cosine transformation, a coefficient which has a large absolute value and exerts great influence on the tone quality is selected and encoded and zeros are inserted into the other unselected coefficients, so that selective encoding is carried out which does not seriously deteriorate the tone quality even when the coding rate is 8 kbps or below. In another arrangement, about three to 16 different selection patterns (vector patterns per frame) are used for the selective coding and a pattern which minimizes the coding error is selected and encoded to ensure optimum coding.

Abstract: Method for time-scale modification ("TSM") of a signal, for example, a voice signal, wherein starting positions of blocks in an input signal, referred to as analysis windows, are varied and an output signal is reconstructed by overlapping analysis windows using fixed window offsets, i.e., the duration of overlap between analysis windows is fixed during reconstruction. This is done by searching for segments of the input signal which are similar to the previous portion of the output signal. In one embodiment of the present invention a cross-correlation is used as a similarity measure to evaluate such similarity and the cross-correlation uses a fixed, predetermined minimum number of samples. The starting position of the analysis window which results in the greatest similarity in overlapping regions is determined as the starting position which provides the largest value of cross-correlation in the overlapping regions.

Abstract: A new way of CELP coding speech simplifies the recursive loop used to poll code adaptive book vectors by reducing the number of autocorrelation operations that must be performed with the K vectors of the codebook each having N entries. Autocorrelation is initially performed for only a small number P<<N autocorrelation coefficients in each codebook vector and the values found are used to scan through all the codebook vectors looking for those S vectors (S<K) which give the best match to the input speech. The autocorrelation function for the S vectors is then recalculated for R entries (P<R.ltoreq.N) in the codebook vectors to determine which of the S codebook vectors and associated gain gives the best match to the input speech.

Abstract: A CT apparatus for reducing aliasing in reconstructed images uses an x-ray tube with a translatable focal spot to double the spatial sampling rate, over that achieved by a conventional CT machine, by acquiring a first and second projection corresponding to two different focal spot positions. The amount of gantry rotation and the translation distance of the focal spot are coordinated so that the projections are interlaced and the resulting combined projection is geometrically indistinguishable from a conventional projection with twice the spatial sampling rate. The distance the focal spot is translated is further adjusted to eliminate redundant projections and provide adequate data acquisition time.

Abstract: A phoneme estimator in a speech-recognition system includes energy detect circuitry for detecting the segments of a speech signal that should be analyzed for phoneme content. Speech-element processors then process the speech signal segments, calculating nonlinear (powers and products) representations of the segments. The nonlinear representation data is applied to speech-element modeling circuitry which reduces the data through speech element specific modeling. The reduced data are then subjected to further nonlinear processing. The results of the further nonlinear processing are again applied to speech-element modeling circuitry, producing phoneme isotype estimates. The phoneme isotype estimates are rearranged and consolidated, that is, the estimates are uniformly labeled and duplicate estimates are consolidated, forming estimates of words or phrases containing minimal numbers of phonemes. The estimates may then be compared with stored words or phrases to determine what was spoken.

Abstract: The present invention is embodied in a method and algorithm for rapidly quantifying phagocytic functions using computer image analysis (CIA) of video light microscopic images. The method and algorithm involve sequential acquisition of bright field or phase contrast and epi-fluorescence video microscopic images of respective field, addition of the images, decision making, object referencing, morphological feature extraction, arithmetic operations, and statistical analysis. This invention provides significantly faster phagocytic functions analysis than manual microscopic examination and more detailed quantitative morphological data than flow cytometery.

Abstract: A speech detector has an intensity detector that indicates whether the intensity of a PCM signal exceeds a first threshold, and a normal-zero-crossing-count detector that indicates whether the zero-crossing count of the PCM signal exceeds a second threshold. The outputs of the intensity detector and normal-zero-crossing-count detector are combined by AND logic to produce the output of the speech detector. The second threshold is set well below the minimum zero-crossing count occurring in normal speech, the function of the normal-zero-crossing-count detector being to disable speech detection during line faults.

Abstract: An electronic system and a method substantially eliminates any delays caused by buffering of a signal containing speech. The electronic system comprises a receiver for receiving signals, a buffer, coupled to the receiver, a buffer controller coupled to the buffer, and an audio section for presenting signals received to a listener. The method comprises the following steps. The receiver receives an input signal representing speech, and produces an output signal representing the input signal after a delay intentionally introduced in the system. The delay corresponds to the length of the buffer. The buffer stores the input signal in the buffer at an input rate, and produces the output signal at an output rate.

Abstract: A process is disclosed for digitizing (more precisely encoding) speech for the purpose of reducing information rate and bandwidth relative to that of prior art means to digitize speech, while enjoying a high signal to noise ratio. Using the the same encoding techniques the process can be used for storage of speech and machine recognition of speech. The process depends on detecting audio waveform zero crossings and generating uniform pulses at the time of the zero crossings. The uniform pulses are created through a regenerative process and are independent of the actual waveform shape save the time of zero crossing. Transmission (or storage) of these uniform pulses permits reconstruction of highly intelligible speech. Ratios of times between zero crossings is used in a new technique for machine word recognition; said ratios allowing recognition regardless of the speakers actual speech rate.

Abstract: A method and apparatus for address setting for speech recording and replay, includes an apparatus for storing a direct current (DC) voltage with a high frequency signal in the dynamic random access memory (DRAM) as an index for replay. During recording, the user can enter speech of different durations. During replay, the system automatically repeats the exact contents without any other background noise or noise speech at the end of the desired message. The apparatus includes a wave clearer, a direct current voltage generator, a high frequency signal generator, a DC voltage detecting and time setting circuit, a resetter, and a recording and replay signal generator.

Abstract: A parametric signal processing structure, which receives autocorrelation coefficients, produces, in an additive sparse superlattice embodiment, quantities .lambda..sub.i.sup.+, defined as (1-k.sub.m) (k.sub.m-1 +1), and in a subtractive sparse superlattice embodiment, quantities .lambda..sub.m.sup.-, defined as (1+k.sub.m) (1-k.sub.m-1), where the quantities k.sub.i correspond to the lattice predictor coefficients, or PARCORS of the signal under study. The additive and subtractive sparse superlattice processing structures are strictly optimal in terms of hardware and complexity, yet lend themselves to implementation in a fully parallel, fully sequential, or parallel partitioned manner.

Abstract: An electrical process transforms an object electrical signal into a compact time-varying graphical representation whereby study of the time-varying spectral properties of said signal may be efficiently pursued.

Abstract: An isolated speech word is extracted from an input speech signal. First, the speech signal is divided into a high frequency band signal and a low frequency band signal. Next, the power levels of each of the high and low frequency band signals are independently compared with respective threshold levels. Finally, the front and the end of the speech word are detected from these independent comparisons.

Abstract: A digital signal encoding apparatus in which input digital signals are divided into a plurality of frequency bands so that the bandwidth of each frequency band will be greater the higher the frequency band, the allowable noise level is set for each frequency band on the basis of the energy value of each frequency band, and components of each frequency band are quantized with the number of bits consistent with the level difference between the energy of each frequency band and the preset allowable noise level. The output information volume following the quantization is detected and the number of bits of allocation for quantization is corrected in dependence upon the error between the detection output and the target value to render the information volume constant over a predetermined time period to enable bit rate adjustment or bit packing with lesser signal deterioration by a simplified construction.

Abstract: A method and apparatus for signal prediction using the estimate-maximize ) algorithm in a time-varying signal system is provided. A time function is used to appropriately weight, in complementary fashion, the significance of both the complete and incomplete data sets used by the EM algorithm over a time period of interest. Initially, the EM solution is based solely on the complete data set. As time progresses, the significance of the complete data set in the solution decreases while the significance of the incomplete data set increases. By the end of the time period of interest, the EM solution is based solely on the incomplete data set. The rate of decrease of significance of the complete data set, and complementary increase in significance of the incomplete data set, are controlled by the characteristics of the time function.

Abstract: A speech recognition apparatus of the speaker adaptation type operates to recognize an inputted speech pattern produced by a particular speaker by using a reference pattern produced by a voice of a standard speaker. The speech recognition apparatus is adapted to the speech of the particular speaker by converting the reference pattern into a normalized pattern by a neural network unit, internal parameters of which are modified through a learning operation using a normalized feature vector of the training pattern produced by the voice of the particular speaker and normalized on the basis of the reference pattern, so that the neural netowrk unit provides an optimum output similar to the corresponding normalized feature vector of the training pattern.