摘要:
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.
摘要:
Certain aspects of the present disclosure provide circuits for generating high accuracy millimeter wave or radio frequency (RF) wideband in-phase (I) and quadrature (Q) oscillating signals having acceptable amplitude and phase mismatch over process, voltage, and temperature (PVT) variations with reduced cost, area, and power consumption. In one example apparatus, a polyphase filter having a first stage and a second stage is provided. Each stage comprises resistive elements and capacitive elements. Certain aspects of the present disclosure provide for intentional resistive and/or capacitive value mismatch between the resistive or capacitive values of one or multiple stages such that the phase mismatch between the resulting I and Q signals may be reduced without degrading the amplitude mismatch. Certain aspects of the present disclosure provide for replacing the resistive elements in at least one stage with transistors operating in the triode region, where the on-resistance is controlled by a feedback network.
摘要:
Reduction of a circuit size and power consumption for performing digital filtering processing in a frequency domain is realized. The digital filter circuit includes: a complex conjugate generation unit for generating a second complex number signal including conjugate complex numbers of all complex numbers included in a first complex number signal of the frequency domain generated by converting a complex number signal of a time domain by Fourier transform; a filter coefficient generation unit for generating a first and a second frequency domain filter coefficient of a complex number from a first, a second and a third input filter coefficient of a complex number having been inputted; a first filtering unit for performing filtering processing to the first complex number signal by the first frequency domain filter coefficient, and outputting a third complex number signal; a second filtering unit for performing filtering processing to the second complex number signal by the second frequency domain filter coefficient, and outputting a fourth complex number signal; and a complex conjugate combining unit for combining the third complex number signal and the fourth complex number signal, and generating a fifth complex number signal.
摘要:
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.
摘要:
A time-division (TD) decimation filter bank includes two decimation filter units. The first decimation filter unit operates at a system clock and receives a first-stage input data string. Each data in the first-stage input data string has a first part data and second part data. During the odd clock periods, the first part data are filtered and decimated in frequency. During the even clock periods, the second part data are filtered and decimated in frequency. The second decimation filter unit operates at the system clock and 2N clock periods are set as an operation-period unit, N≧2. The second decimation filter unit receives the outputs from the first decimation filter unit and receives several feedback data of the second decimation filter unit by TD, so that the received data are distributed into the 2N clock periods for filtering and decimation and outputting by TD.
摘要:
A finite impulse response filter comprises an input formatter, a plurality of sample registers, a plurality of coefficient registers, an arithmetic unit, a multiply accumulate unit, a crosspoint switch, an interpolator, a control unit, and an output formatter. The input formatter separates the in-phase portion of a complex-number discrete-time sample from the quadrature portion. The sample registers store a plurality of discrete-time samples. The coefficient registers store a plurality of coefficients. The arithmetic unit adds two of the discrete-time samples to create a sum. The multiply accumulate unit includes a multiplier that multiplies the sum by a coefficient to create a product, an adder that adds the product to a sum of products, and a register that stores the sum of products. The crosspoint switch allows communication between the first and second plurality of registers and the arithmetic unit and the multiply accumulate unit. The interpolator inserts a desired number of zeros into the time-sampled data stream to adjust the time-sampled data stream to an increasing sampling rate. The control unit controls the settings of the crosspoint switch, the arithmetic unit, and the multiply accumulate unit. The output formatter combines the in-phase sum of products and the quadrature sum of products to create a filtered complex-number discrete-time sample.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.