Texas Instruments Corporation

Mustang Technology Group, L.p.

Raytheon Company

Rustin W Allred

Zhongnong Jiang

James Robert Hochschild

Robert S Young, Jr

Michael Joseph Tsecouras

John B Allen

Hirohisa Yamaguchi

Yoshito Higa

Keith G Gutierrez

John L Snyder, Iii

Stephen R Handley

Hanna E Witzgall

Richard Lowder

Donald C Whitney

Robert Bless, Jr

Mhamed Ibnabdeljalil

Robert R Bless

Anthony L Leatham

A system for estimating a location of the source of a projectile includes a radar system for transmitting a wave and detecting and providing an indication of a wave reflection from the projectile, an acoustic detection system for detecting and providing an indication of at least one sound associated with the projectile, and circuitry for estimating the location in response to the indication of a wave reflection and the indication of at least one sound.

System and method for estimating location of projectile source or shooter location

A distance determining system for use in a movable device. The system comprises circuitry for capturing an image of an object external from the movable device. The system also comprises circuitry, responsive to the image, for determining an angle between a direction of movement of the movable device and the object. Lastly, the system also comprises circuitry for determining a distance to the object in response to at least the angle.

System and method for determining range in response to image data

Systems and methods for determining coefficients of an Finite Impulse Response (FIR) filter are disclosed. The FIR filter coefficients are computed by determining a sine of an input value and an inverse of the input value. The sine of the input signal and the inverse of the input signal are multiplied together to form a sinc value of the input value. The sinc value is employed to determine the coefficient. The system and method can be repeated to compute any number of FIR filter coefficients in real-time. The sine of the input signal is computed utilizing a memory lookup table. The memory lookup table includes pairs of uniformly distributed values for the sine and cosine functions in the range of 0 to &#x3c0;. The inverse of the input value is computed using an inverse memory lookup table, a most significant digit and a remainder. The coefficient is then computed from a product of the sine of the input signal and the inverse of the input signal. Thus, the coefficient is computable in real-time without the use of previously computed and stored coefficients.

Efficient real-time computation of FIR filter coefficients

An electronic dB-to-linear gain conversion system (). The system comprises an input () for receiving a gain index signal (GI) representing a desired dB value. The desired dB value is selected from a set having an integer number S of dB values. The system also comprises a storage circuit () for storing an integer number V of linear gain values and circuitry for producing a linear gain signal (LG) in response to the gain index signal and to one of the V linear gain values. In the preferred embodiment, V is less than S.

Device with dB-to-linear gain conversion

A system () for providing an integer number N of filters. The system comprises an input (D) for receiving a digital audio signal and an output (D) for providing a filtered audio signal. The system also comprises circuitry () for storing at least a first set of fixed filter coefficients and circuitry for storing estimation data. The system also comprises circuitry () for estimating a number of sets of estimated filter coefficients in response to the estimation data and the fixed filter coefficients. The system also comprises circuitry () for applying a transfer function to the digital audio signal and in response for providing the filtered audio signal. The circuitry for applying the transfer function applies a set of filter coefficients selected from the first set of fixed filter coefficients and the sets of estimated filter coefficients. Also, the transfer function is selected from a transfer function set consisting of a high pass filter transfer function and a low pass filter transfer function.

Variable digital high and low pass filters

A digital signal system () for determining an approximate logarithm of a value of x having a base b is described. The system comprises circuitry for storing x as a digital representation, identifying a most significant digit (MSD) of the digital representation, a table for storing a set of predetermined logarithms having the base b, circuitry for addressing the table in response to a first bit group (t) of the set of bits in respective lesser significant bit locations and for outputting a one of the predetermined logarithms corresponding to a first number (l) in the set of numbers, and circuitry for outputting the approximate logarithm of the value of x in response to the one of the predetermined logarithms and in response to a function estimation between logarithms at a first and second endpoint.

Circuits, systems, and methods implementing approximations for logarithm, inverse logarithm, and reciprocal

A digital signal system () determines an approximate antilog x from a value of &#x192;(x), where x has a base b.

Circuits, systems, and methods implementing approximations for inverse logarithm

A self-adaptive graphic equalizer operable to equalize the affects of an audio system on an audio signal includes an adaptive graphic equalizer having a plurality of equalizing filters, where the plurality of equalizing filters have different center frequencies equidistant from one another and spanning a predetermined audio bandwidth. Each equalizing filter is operable to filter an isub-band of the audio signal. A plurality of first filters are coupled to the audio system, each first filter is operable to filter an isub-band of an output signal of the audio system. A plurality of second filters are operable to filter an isub-band of the audio signal. A gain adjuster is operable to adjust the isub-band of the adaptive graphic equalizer in response to a difference in the isub-band of the filtered output signal from the plurality of first filters and the isub-band of the filtered audio signal from the plurality of second filters.

Digital self-adapting graphic equalizer and method

An audio loudness compensation system includes a level sensor receiving an audio input signal and operable to estimate a level of the audio input signal over a first predetermined time period, and a level mapper receiving the estimated level and operable to map the estimated level to a raw audio gain in response to a slope setting and an offset setting. The system further includes an attack and decay filter receiving the raw audio gain and operable to smooth out increasing and decreasing changes in the raw audio gain in response to a second and, possibly a third predetermined time period, and a compensation filter receiving the smoothed raw audio gain and operable to modify the audio input signal in response to the smoothed raw audio gain, a center frequency setting and a bandwidth setting, and generate a loudness compensated audio output signal.

Configurable digital loudness compensation system and method

A graphametric equalizer has graphic and parametric equalization capabilities within a single non-redundant system. A translation function capability converts user selected inputs for center frequency, bandwidth and gain into allpass filter parameters to realize an allpass filter-based equalization filter structure capable of performing graphic and/or parametric equalization on-the-fly. The graphametric equalizer has a softening function capability to time user inputs and increment filter parameters gracefully such that the graphametric equalizer can be recharacterized with new filter parameters on-the-fly without incurring undesirable audible artifacts.

Digital graphametric equalizer

Systems and methods for determining coefficients a filter are disclosed. The filter coefficients are computed by determining a sine of an input value and an inverse of the input value. The sine of the input signal and the inverse of the input signal are multiplied together to form a sinc value of the input value. The sinc value is employed to determine the coefficient. The system and method can be repeated to compute any number of filter coefficients in real-time. The sine of the input signal is computed utilizing a memory lookup table. The memory lookup table includes pairs of uniformly distnbuted values for the sine and cosine functions in the range of 0 to &#x3c0;. The inverse of the input value is computed using an inverse memory lockup table, a most significant digit and a remainder. The coefficient is then computed from a product of the sine of the input signal and the inverse of the input signal. Thus, the coefficient is computable in real-time without the use of previously computed and stored coefficients.

Efficient real-time computation

An audio compensation system () for producing a sound compensated output signal (S). The system comprises a volume control circuit () for producing a volume-adjusted signal (S) by applying a volume adjustment to an audio signal in response to a volume setting, wherein the sound compensated output signal is responsive to the volume-adjusted signal. The system also comprises circuitry () for producing an amplified signal (S) by amplifying a selected bandwith of signals in response to the volume setting. The sound compensated output signal is also responsive to the amplified signal.

Volume-responsive loudness compensation circuits, systems, and methods

A signal generator (). The signal generator comprises circuitry () for producing at least a first input noise signal (N), wherein the first input noise signal has a statistically insignificant autocorrelation. The signal generator also comprises an allpass filter circuit () coupled to receive and filter the first input noise signal and for outputting in response a white noise signal () having a Gaussian probability distribution function

Gaussian noise generator

Digital audio tone control implemented using Shelving filters for the digital audio treble tone control exhibits artifacts (noise, distortion, etc.) as the tone control settings are changed. This was previously accomplished by changing filter coefficients in the traditional small equal (on a dB scale) filter steps of a fraction of 1 dB. While this worked for bass filters, artifacts were still present for treble. This invention eliminates these artifacts by changing the filter steps to small equal steps on a linear scale. Additionally, where the steps became too large for the resolution required, additional filter steps are added. Approximately 150 filter steps are used for treble control and 128 filter steps are used for bass tone control. An efficient way of implementing the filter steps for digital tone control stores () one set of filter coefficient values and a small amount of additional information and then increments the coefficients between all the other steps. This reduces the memory required by as much as 95% or the machine cycles for implementing filter coefficients by 40-200 times. This new efficient method is accomplished by: 1) defining the filter coefficients, 2) piecewise linearizing the plot of filter coefficients versus filter step to define the linear regions, 3) defining the initial set of filter coefficients, and 4) defining increments between the filter steps.

Digital tone control with linear step coefficients

A method for generating digital filters for tuning a hearing aid to enhance hearing ability. Data is provided, for a range for a target response curve representative of enhanced hearing ability of sound level versus frequency. Second digital data is also generated, for an initial response curve or audiogram of an individual's hearing ability. The first digital data is compared with the second digital data and it is determined whether the initial response curve is within the tolerance range. If not, digital audio filters are iteratively generated, and the digital audio filters are applied to digital data representing received sound signals to generate third digital data to enhance hearing. Parameters of the digital audio filters are automatically optimized until the compensated response curve is within the tolerance range or a predetermined number of digital audio filters has been reached.

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Plano, TX, United States of America

Rustin W Allred