201results about "Digital filters" patented technology

An interpolator, method of interpolating a one-bit signal and resampler employing the interpolator and method. The interpolator employs a cascaded architecture to interpolate a one-bit input signal received at an input thereof. In one embodiment, the interpolator includes a multiple order interpolation filter that generates a sample range from at least three input samples associated with the one-bit input signal. The interpolator further includes a linear interpolation filter, associated with the multiple order interpolation filter, that develops a plurality of samples within the sample range.

An enhancement system improves the perceptual quality of a processed speech. The system includes a delay unit that delays a signal received through a discrete input. A spectral modifier linked to the delay unit is programmed to substantially flatten the spectral character of a background noise. An adaptive filter linked to the spectral modifier adapts filter characteristics to match a response of a non-delayed signal. A programmable filter is linked to the delay unit. The programmable filter has a transfer function functionally related to a transfer function of the adaptive filter.

A programmable logic device can be configured as a finite impulse response (FIR) filter capable of operating in either interpolation mode or decimation mode and of switching between those modes at run time. The FIR filter structure can be mapped onto a specialized processing block of the programmable logic device that includes multipliers and adders for adding the products of the multipliers. The FIR filter structure minimizes the number of multipliers used by reusing various calculations that are repeated as a result of the interpolation or decimation operation, using multiplexers or other run-time-controllable selectors to select which current or stored multiplier outputs to use.

A specialized processing block for a programmable logic device incorporates a fundamental processing unit that performs a sum of two multiplications, adding the partial products of both multiplications without computing the individual multiplications. Such fundamental processing units consume less area than conventional separate multipliers and adders. The specialized processing block further has input and output stages, as well as a loopback function, to allow the block to be configured for various digital signal processing operations.

A digital control circuitry for a MEMS gyroscope is provided. The digital control circuitry comprises a digital primary loop circuitry configured to process a digitized primary signal, a digital secondary loop circuitry configured to process a digitized secondary signal and a digital phase shifting filter circuitry configured to generate two phase shifted demodulation signals from the digitized primary signal. The digital secondary loop is configured to demodulate the digitized secondary signal using the two phase shifted demodulation signals.

A filter apparatus for filtering a time domain input signal to obtain a time domain output signal, which is a representation of the time domain input signal filtered using a filter characteristic having an non-uniform amplitude/frequency characteristic, comprises a complex analysis filter bank for generating a plurality of complex subband signals from the time domain input signals, a plurality of intermediate filters, wherein at least one of the intermediate filters of the plurality of the intermediate filters has a non-uniform amplitude/frequency characteristic, wherein the plurality of intermediate filters have a shorter impulse response compared to an impulse response of a filter having the filter characteristic, and wherein the non-uniform amplitude/frequency characteristics of the plurality of intermediate filters together represent the non-uniform filter characteristic, and a complex synthesis filter bank for synthesizing the output of the intermediate filters to obtain the time domain output signal.

A complex low-pass filter that reduces the influence of component mismatch. The filter includes a first filter section for effecting a first single pole transfer function and a second filter section for effecting a second single pole transfer function, where the first and the second single pole transfer functions collectively define a conjugate pair of poles. In higher order low-pass filters, an optimal cascade order follows a shoestring pattern.

A multi-channel receiver comprising an ADC and a multi-band, multi-channel selector. The ADC converts a broad-band multi-channel signal into a digital signal. The digital signal is then broken into sub-bands each containing a plurality of channels. A channel selector selects desired channels from the appropriate sub-band. The multi-channel receiver may deliver simultaneous channels equal to the number of channel selectors that have been implemented. The multi-channel receiver may be implemented on a single integrated circuit.

An efficient configurable signal filter. The filter includes a first mechanism for receiving a first signal of a first type and a second signal of a second type. A second mechanism selectively filters the first signal during a first mode of operation, and filters the second signal during a second mode of operation. A third mechanism generates control signals. A fourth mechanism automatically configures the second mechanism to operate in the first mode of operation or the second mode of operation based on the control signals. In a specific embodiment, the first type of signal is characterized by a first rate, and the second type of signal is characterized by a second rate. The first signal and the second signal are digital ADC outputs. The second mechanism includes plural filter blocks, each having one or more Multiply-Accumulate (MAC) pipes. Each of the one or more MAC pipes include one or more MAC blocks that are each associated with a coefficient memory data structure of a coefficient memory. The third mechanism or a user selects coefficients from each memory data structure to apply to each MAC block, thereby selectively affecting filter response. The control signals direct multiplexers or switches to configure the MAC pipes in a serial configuration or a parallel configuration corresponding to the first mode of operation or the second mode of operation, respectively.

A filter device and method for suppressing effects of Adjacent-Channel Interference of a received signal in a Frequency-Division-Multiple-Access system by filtering a baseband signal of the received signal. The filter device comprises an interference filter, which is a complex digital Single-Input-Multiple-Output, SIMO, filter that is adapted to simultaneously generate a first signal filtered at an upper-frequency-band and a second signal filtered at a lower-frequency-band, wherein the first signal is separate from the second signal. The filter device also comprises a selector adapted to select one of the signals as the output from the filter device.

Improved interpolator and decimator apparatus and methods, including the addition of an elastic storage element in the signal path. In one exemplary embodiment, the elastic element comprises a FIFO which advantageously allows short term variation in sample clocks to be absorbed, and also provides a feedback mechanism for controlling a delta-sigma modulated modulo-N counter based sample clock generator. The elastic element combined with a delta-sigma modulator and counter creates a noise-shaped frequency lock loop without additional components, resulting in a much simplified interpolator and decimator.

A low power sample rate converter adapted for use with a telecommunications system transceiver. The sample rate converter includes a first circuit that provides an input signal characterized by a first sample rate and a delayed version of the input signal. A second circuit periodically multiplies, at a second sample rate, samples in the input signal by a first predetermined coefficient in accordance with a predetermined transfer function and provides a first signal in response thereto. A third circuit periodically multiplies, at the second sample rate, samples in the delayed version of the input signal by a second predetermined coefficient in accordance with the predetermined transfer function and provides a second signal in response thereto. A fourth circuit combines the first signal and second signal providing a rate-converted version of the input signal as an output signal in response thereto. In a specific embodiment, the delayed version of the input signal is delayed by one sample with respect to the input signal. The sample rate converter further includes a counter that is clocked by a first periodic signal. The first periodic signal has a frequency related to the first sample rate by a predetermined fraction. The counter is cleared by a second periodic signal having a second frequency equivalent to the first sample rate. The counter produces a counter output at the first frequency. In the preferred embodiment, the predetermined fraction is ⅓ and the first predetermined coefficient is equivalent to the sum of 1 and the counter output. The second predetermined coefficient is equivalent to the difference of 2 and the counter output. The fourth circuit includes an adder for adding the first signal and the second signal and providing the output signal in response thereto. The predetermined transfer function is represented by the following coefficient sequence: [1 2 3 2 1].

In order to process a subband signal of a plurality of real subband signals which are a representation of a real discrete-time signal generated by an analysis filter bank, a weighter for weighting a subband signal by a weighting factor determined for the subband signal is provided to obtain a weighted subband signal. In addition, a correction term is calculated by a correction term determiner, the correction term determiner being implemented to calculate the correction term using at least one other subband signal and using another weighting factor provided for the other subband signal, the two weighting factors differing. The correction term is then combined with the weighted subband signal to obtain a corrected subband signal, resulting in reduced aliasing, even if subband signals are weighted to a different extent.

A sampling-rate conversion method receives N input channels which have been digitized at an input sampling rate, and converts the sampling rate of each input channel to produce N output channels at an output sampling rate. The method comprises computing a common FIR interpolating function which depends upon the input and output sample clocks and the instantaneous output time, and applying the common FIR interpolating function to all N input channels to compute all N output channels.

A programmable integrated circuit device such as a programmable logic device can be configured as a finite impulse response (FIR) filter capable of operating in decimation mode. The device includes at least one user-configurable random access memory block, and that user-configurable random access memory is configured as coefficient memories and data sample memories. The memories are large enough to hold up to all of the coefficients of the filter and a plurality of data samples at one time. Because the data samples and coefficients need not be shifted through the filter at the programmable logic device clock rate, overclocking of the filter is not necessary. The filter can run at a clock rate which is the same as the input data rate, while taking advantage of the available random access memory to mimic a shift register.

In order to solve a problem of achieving distortion compensation with high accuracy, a digital filter device includes a first distortion compensation filter unit for conducting distortion compensation of first waveform distortion included in an inputted signal through digital signal processing, a first filter coefficient setting unit for setting a filter coefficient of the first distortion compensation filter unit, a second distortion compensation filter unit for compensating second waveform distortion included in a signal outputted from the first distortion compensation filter unit, and a second filter coefficient setting unit for setting a filter coefficient of the second distortion compensation filter unit based on the filter coefficient set by the first filter coefficient setting unit.

This invention provides a filter system which may be implemented with less hardware and software resources than traditional filters. In addition, the filter system structure reduces the complexities typically associated with filter design by permitting direct specification of the filter frequency response. Thus, the filter system may adaptively change the filter frequency response on the fly without incurring excessive time or computational costs. The filter system may provide a filtered signal output to any subsequent processing system, such as a voice recognition system or audio reproduction system.

An apparatus for demodulating a digital input signal having a pre-assigned symbol rate by using a single sample clock signal is disclosed. The pre-assigned symbol rate is selected from the group consisting of a plurality of symbol rates. The apparatus comprises: a Cascaded Integrated Comb (CIC) Decimator filter configured to perform a filtering-decimation operation to isolate the down converted digital signal having the pre-assigned symbol rate; a Shaper configured to restore an original spectrum of the I and Q components of the CIC output signal; and a CIC tuner loop filter configured to insert the estimated by the Coarse Frequency estimation block frequency offset into a carrier loop in order to complete a carrier recovery of the pre-assigned input digital signal.