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526 results about "Signal transfer function" patented technology

The signal transfer function (SiTF) is a measure of the signal output versus the signal input of a system such as an infrared system or sensor. There are many general applications of the SiTF. Specifically, in the field of image analysis, it gives a measure of the noise of an imaging system, and thus yields one assessment of its performance.

Gesture synthesizer for electronic sound device

A MIDI-compatible gesture synthesizer is provided for use with a conventional music synthesizer to create musically realistic<DEL-S DATE="20020416" ID="DEL-S-00001" / >ally<DEL-E ID="DEL-S-00001" / > sounding gestures. The gesture synthesizer is responsive to one or more user controllable input signals, and includes several transfer function models that may be user-selected. One transfer function models properties of muscles using Hill's force-velocity equation to describe the non-linearity of muscle activation. A second transfer function models the cyclic oscillation produced by opposing effects of two force sources representing the cyclic oppositional action of muscle systems. A third transfer function emulates the response of muscles to internal electrical impulses. A fourth transfer function provides a model representing and altering virtual trajectory of gestures. A fifth transfer function models visco-elastic properties of muscle response to simulated loads. The gesture synthesizer outputs <DEL-S DATE="20020416" ID="DEL-S-00002" / >MIDI-compatible<DEL-E ID="DEL-S-00002" / > continuous pitch data, tone volume and tone timbre information. The continuous pitch data is combined with discrete pitch data provided by the discrete pitch generator within the conventional synthesizer, and the combined signal is input to a tone generator, along with the tone volume and tone timbre information. The tone generator outputs tones that are user-controllable in real time during performance of a musical gesture.
Owner:LONGO NICHOLAS

Low power programmable digital filter

A low power programmable digital filter adapted for use with a telecommunications system transceiver. The digital filter includes a first finite impulse response filter section for receiving an input signal and having a first transfer function. An infinite impulse response filter section is connected to the first finite impulse response filter section and has a second transfer function. A second finite impulse response filter section is connected to the infinite impulse response filter section and outputs a filtered output signal in response the receipt of the input signal by the programmable digital filter. The second finite impulse response filter section has a third transfer function. A programmable coefficient is included in the first, second, and / or the third transfer function. In a specific embodiment, the first transfer function has a first programmable coefficient. The second transfer function has a second programmable coefficient and the third transfer function has a third programmable coefficient. The programmable digital filter further includes a processor for providing a control signal. A memory provides the first, second, and / or third programmable coefficients in response to the control signal. A high-pass filter section provides input to the first finite impulse response filter section. A multiplexer selectively bypasses the high-pass filter in response to a bypass control signal from the processor. The processor generates the bypass control signal in response to bias occurring in the input signal. In the illustrative embodiment, the first finite impulse response filter section includes a first jammer filter, a second jammer filter, and a third jammer filter for removing telecommunications jammer signals in the input signal. The first, second, and third jammer filters have first, second, and third jammer filter transfer functions with the first programmable coefficient, a fourth programmable coefficient, and a fifth programmable coefficient, respectively. The infinite impulse response filter section includes a first equalization filter and a second equalization filter.
Owner:QUALCOMM INC

FEXT determination system

Operational data is utilized to determine the FEXT interference induced by one line into the other DSL line. FEXT interference can be calculated using the NEXT interference measured between the two lines at the upstream ends of the loops and the downstream channel transfer function of one of the loops. Because the NEXT and transfer function constitute a linear time-invariant system, as does the FEXT interference between the lines, the NEXT interference and line transfer function can be multiplied (if in linear format) or added (if in logarithmic format) to approximate the FEXT interference between the lines. The collection of data, calculations and other functions performed in these techniques may be performed by a system controller, such as a DSL optimizer. An Xlog(u,n) quantity is a decibel-magnitude representation of the insertion-loss equivalent of FEXT transfer functions and is defined as the ratio of (1) a line u's source power into a matched load of 100 Ohms when no binder is present to (2) the power at the output of the subject line when line u is excited with the same source and the binder is present. Xlin(u,n) is the linear equivalent of Xlog(u,n). The Xlog(u,n) and Xlin(u,n) quantities may be represented in specific formats that assist in their use in DSL and other systems. When defined as a line's insertion loss, Xlin (or equivalently Xlog) does not include the effect of any transmit filter.
Owner:ASSIA SPE LLC CO THE CORP TRUST CO

Amplifier linearizer

The present invention provides an advanced adaptive predistortion linearization technique to dramatically reduce nonlinear distortion in power amplifiers over a very wide instantaneous bandwidth (up to 2 GHz) and over a wide range of amplifier types, input frequencies, signal types, amplitudes, temperature, and other environmental and signal conditions. In an embodiment of the invention, the predistortion linearization circuitry comprises (1) a higher-order polynomial model of an amplifier's gain and phase characteristics—higher than a third-order polynomial model; (2) an adaptive calibration technique; and (3) a heuristic calibration technique. The higher-order polynomial model is generated by introducing, for example, a plurality of multi-tone test signals with varying center frequency and spacing into the power amplifier. From the power amplifier's corresponding output, the nonlinearities are modeled by employing a higher-order curve fit to capture the irregularities in the nonlinear transfer function. Different distortion transfer functions can be implemented for different operating conditions. The adaptive calibration technique is based on a feedback analysis technique, which updates the applicable distortion transfer function by analyzing the error signal between the introduced input signal and the output signal in real-time. The heuristic calibration technique implements different distortion transfer functions based on historical operating conditions and optimal configurations of the power amplifier.
Owner:TM IP HLDG LLC
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