Provided is a digital filter circuit in which a filter coefficient can be easily changed, for which circuit scale and power consumption can be reduced, and which carries out digital filter processing in a frequency domain. This digital filter circuit includes: a separating circuit for separating a first complex number signal, of a frequency domain that was subjected to Fourier transform, into a real number portion and an imaginary number portion; a filter coefficient generating circuit for generating a first frequency domain filter coefficient from a first input filter coefficient and a third input filter coefficient, and for generating a second frequency domain filter coefficient from a second input filter coefficient and the third input filter coefficient; a first filter that filters the separated real number portion using the first frequency domain filter coefficient; a second filter that filters the separated imaginary number portion using the second frequency domain filter coefficient; and a combining circuit for combining the output from the two filters.

Provided herein is an implementation of a finite impulse response (FIR) filter that uses a distributed arithmetic architecture. In one or more example, a data sample with multiple bits is processed through a plurality of bit-level multiply and accumulate circuits, wherein each bit of the data sample corresponds to a bit of the data sample. The output of each bit-level multiply and accumulate circuit can then be shifted by an appropriate amount based on the bit placement of the bit of the data sample that corresponds to the bit-level multiply and accumulate circuit. After each output is shifted by the appropriate amount, the outputs can be aggregated to form a final FIR filter result.

Briefly, in accordance with one or more embodiments, a processor receives an incoming data stream that includes alpha channel data, and a memory stores an application programming interface (API). The API is to route the alpha channel data to a fixed point blending unit to perform one or more blending operations using fixed point representation of the alpha channel data. The API is further to route the incoming data stream to a floating point blending unit to perform operations involving floating point representation of the incoming data.

A finite impulse response (FIR) filter includes a plurality of registers. The data input terminal of each register is directly coupled to the input of the FIR filter. A new data value is passed to each register on each clock cycle of a filter clock signal. Only one of the registers processes the data value on each clock cycle. A ring counter is coupled to the registers and determines which register processes the data value on each dock cycle.

Biquad stage systems and methods include receiving at biquad sections a signal sample, generating, by each biquad section, a pair of output values based on the signal sample, including a first value based on fixed-point processing path and a second value emulating a floating-point processing path, and accumulating the pair of output values from each of the plurality of biquad sections to generate an output signal. The biquad stage receives an N-bit input signal, which is processed by a biquad section. Delay elements delay the signal sample before input to other biquad sections. The delayed signal sample is input to the first processing path and the second processing path of a corresponding biquad stage. By performing the processing based on two paths, a more accurate result can be found when using a reduced word length in the multiply operations resulting in a lowering of the power consumption.

A Finite Impulse Response (FIR) filter is configured to minimize delay and maximize passband power by adjusting the filter coefficients applied to the sampled values. The FIR filter obtains an input signal and samples the input signal to generate a set of sampled input values. The FIR filter generates a set of filter coefficients, with each filter coefficient based on a corresponding sampled input value in the set of sample input values. The FIR filter selects a subset of sampled input values that have been most recently sampled from the input signal, and selects a subset of filter coefficients corresponding to sampled input values that are not the most recently sampled. The subset of sampled input values is combined with the subset of filter coefficients to generate an output value for the FIR filter.

A polyphase decimation FIR filter apparatus including a modulo integrator circuit configured to integrate input samples and to provide integrated input samples; and a polyphase FIR filter circuit configured to process the integrated input samples, the polyphase FIR filter circuit including a plurality of multiplier accumulator circuits, each configured to accumulate products of coefficients and respective integrated signal samples, wherein each of the multiplier accumulator circuits receives a subset of FIR filter coefficients, wherein the FIR filter coefficients are derived as the nth difference of original filter coefficients, where n is a number of integrators in the integrator circuit, and wherein the FIR filter circuit is configured to perform computation operations with modulo arithmetic.

A finite impulse response (FIR) filter including a delay line and a plurality of arithmetic units. Each arithmetic unit is coupled to a different one of a plurality of tap points of the delay line, is configured to receive a respective signal value over the delay line, and is associated with a respective coefficient. Any given one of the arithmetic units is configured to receive a respective control word. The respective control word specifying: (i) a plurality of trivial multiplication operations, and (ii) a plurality of bit shift operations. Any given one of the arithmetic units is further configured to estimate or calculate a product of the respective signal of the arithmetic unit respective signal value and the respective coefficient of the arithmetic unit by performing the trivial multiplication operations and bit shift operations that are specified by the respective control word that is received at the given arithmetic unit.

Biquad stage systems and methods include receiving at biquad sections a signal sample, generating, by each biquad section, a pair of output values based on the signal sample, including a first value based on fixed-point processing path and a second value emulating a floating-point processing path, and accumulating the pair of output values from each of the plurality of biquad sections to generate an output signal. The biquad stage receives an N-bit input signal, which is processed by a biquad section. Delay elements delay the signal sample before input to other biquad sections. The delayed signal sample is input to the first processing path and the second processing path of a corresponding biquad stage. By performing the processing based on two paths, a more accurate result can be found when using a reduced word length in the multiply operations resulting in a lowering of the power consumption.

A digital filter structure and related method of digital filtering are presented. The digital filter structure is arranged to receive one or more clocked input signals having a first clock rate, and which is driven at a second clock rate higher than said first clock rate. The digital filter structure has a plurality of delay elements and multiplexing circuitry arranged to selectively engage the delay elements such that, at every clock cycle of the digital filter structure, a filter operation is performed on a different stream of data. The disclosure can be applied in many different contexts. One particular implementation example is that of an adaptive noise cancellation (ANC) system using sigma-delta infinite impulse response filters. In this context the present disclosure minimizes latency and hardware implementation area by requiring only one filtering circuit for multiple channels of data to be filtered.