Abstract: A system for recognizing degraded pseudo-random noise (PN) synchronization in a spread-spectrum receiver. The system uses a signal that indicates the correlation of the locally generated PN sequence with the received PN sequence. The correlation signal can be a symbol-length integration of the output from a square-law detector, or an appropriate similar signal. If the correlation signal is not degraded by demodulating with a deliberately shifted copy of the PN sequence, there is an indication that the unshifted PN sequence was itself not correctly synchronized. A sufficiently degraded correlation signal indicates that the receiver's PN synchronization is correct. To prevent the loss of transmitted data during the testing, each transmitted frame contains a Measurement field (that contains no payload data) for assessing the synchronization in this manner. The PN sequence is shifted only during this specific portion of the received frame.

Abstract: A method for configuring the receiver with an IF delay value that indicates the timing of symbol transitions in a received signal processed by the receiver. The receiver recovers a timing that has the same period as the symbol period, but which is out of phase with the received symbols. The received symbols are members of a constellation with elements that have purely I or purely Q components. A symbol-quality signal is generated by constructing the quantity .vertline..vertline.I.vertline.-.vertline.Q.vertline..vertline.. This quantity is a maximum when the detected symbols are aligned with the expected points in the symbol constellation, and decreases if the detected symbols are rotated away from these constellation points. The method determines an optimal delay value by which the symbol clock should be shifted from the recovered timing by using the symbol-quality signal to evaluate test delays and to successively refine them until the optimal delay value is found.

Abstract: A speech compression/decompression system and method which do not require special hardware are described. The compression unit represents an input audio signal as a collection of parameters, wherein the parameters are a remnant excitation pulse sequence, a set of spectral coefficients and a set of pitch parameters. The decompression unit utilizes the pitch parameters and remnant excitation pulse sequence to produce a reconstructed excitation signal. The decompression unit also utilizes the spectral coefficients to filter the reconstructed excitation signal into a speech waveform. The compression unit includes a short-term predictor, a two-step long-term predictor and a multi-pulse analyzer.

Abstract: A system for matching an input signal, comprising non-white noise and a patterned signal corrupted by said non-white noise, to a plurality of reference signals, the system including an estimator estimating noise features of said non-white noise and producing from the noise features at least one noise whitening filter; a filter generally simultaneously filtering the input signal and the plurality of reference signals using the noise whitening filter and producing a filtered input signal, having a white noise component, and a plurality of filtered reference signals; and a pattern matcher generally robust to white noise for matching the filtered input signal to one of the filtered reference signals.

Abstract: A data reception apparatus and technique for use with a modem and including the functions of receiving via the modem a signal including a plurality of datapoints and including therein desired data and inherently redundant data, wherein upon reception of each datapoint, the modem produces quality information regarding how accurately the datapoint was received and identifying and correcting errors introduced into the signal through use of the inherently redundant data and the quality information.

Abstract: Noise in a speech-plus-noise input signal is suppressed by splitting the input signal into spectral channels and decreasing the gain in the each channel which has a low signal-to-noise ratio (SNR). A voice operated switch (VOX) acts to detect noise-only input to gate a background noise (input signal) estimator and also to gate a residual noise (output signal) estimator. The gain in each of the channels is controlled by the current value (a posteriori) input signal SNR estimate, modified by the prior value (a priori) input signal SNR estimate, and smoothed as a function of the residual (output noise signal) estimate.

Abstract: A voice operated switch employs digital signal processing techniques to examine audio signal frames having harmonic content to identify voiced phonemes and to determined whether the signal frame contains primarily speech or noise. The method and apparatus employ a multiple-stage, delayed-decision adaptive digital signal processing algorithm implemented through the use of commonly available electronic circuit components.

Abstract: An encoder apparatus for communication and pattern recognition systems employing a nearest neighbor search method for signal and data compression, based on a vector encoding quantization technique which optimizes systems performance with substantially reduced computational complexity using a fast geometrically-oriented search procedure. The apparatus comprises pre-procesing apparatus for providing off-line reorganization of a codebook having a set of reference vector patterns constituting codevectors with which the input vector is to be compared in a search procedure, and on-line apparatus for encoding the random input vector through quantization in accordance with the search procedure in the codebook.

Abstract: Pre-recorded speech is played back at a different rate, without pitch change. Adjacent signal segments are combined with best match processing. Method and apparatus process time domain speech signals containing speech information, the rate of reproduction of which is to be varied without changing pitch, wherein the input signal is processed by capturing input time domain speech samples in frames wherein the number of samples per frame is a function of a desired speech change factor, forming blocks from the frames, additively cross correlating input blocks with prior-processed or output blocks, preferably by means of an Average Magnitude Difference Function, to obtain a time relation of best match for the rate of reproduction, adding consecutive input and output blocks at the point of maximum correlation, and applying a window function between the overlapping portions of the output block and the input block to obtain a new output block. The method does not require multiplication or division.