A method of designing a digital filter for example for use in an FBMC/OQAM telecommunications system, with a target overlapping factor and meeting a specified signal to interference ratio is described, whereby a candidate filter design defined by an impulse response, satisfying the Nyquist criterion and having an overlapping factor higher than the target is selected, and the time and frequency coefficients of its impulse response inverted to define a new filter design; and

truncating the impulse response defining said new filter design to the minimum number of coefficients achieving said specified signal to interference ratio.

Provided is a method for processing data samples from a plurality of data channels. The method may include obtaining a plurality of data samples from the plurality of data channels. Obtaining the plurality of data samples may involve successively obtaining a data sample from each data channel of the plurality of data channels. Successively obtaining a data sample from each data channel may be performed a plurality of times during a specified time period. Each data sample of the plurality of data samples may be associated with a respective sample time, and each respective sample time may be relative to a single specified reference point in time. The method may further include, for each data sample of the plurality of data samples, determining a time-dependent coefficient value that may correspond to the sample time associated with the data sample, and applying the determined time-dependent coefficient value to the data sample.

A common sub-expression elimination method for simplifying hardware logic of a hardware filter circuit by eliminating a common sub-expression included in a plurality of sub-expressions is provided. Each of the sub-expressions includes a corresponding two or more of inputs constituting a plurality of coefficients used by the hardware filter circuit. The method is implemented on a computing device and includes: identifying for each coefficient of the plurality of coefficients, a combination of the inputs constituting the coefficient; counting occurrences of the sub-expressions in each of the coefficients; identifying one or more of the sub-expressions having a maximum one of the counts and including the corresponding two or more of the inputs; selecting one of the one or more of the sub-expressions as the common sub-expression; eliminating the common sub-expression; and repeating these steps to eliminate more of the sub-expressions common to multiple ones of the coefficients.

A method of increasing the sample rate of a digital signal by creating intermediate sample points between adjacent neighbouring sample points comprising the step of populating each of the intermediate sample points depending on a weighted influence of a predetermined number of the neighbouring sample points, the weighted influence being calculated by representing the digital signal or filter at the predetermined number of sample points at least in part by its cosine components, which are each represented by absolute values of a cosine function in the time domain substantially limited to half a waveform cycle at its mid-point; combining the aforementioned cosine components at each of the neighbouring sample points to obtain waveforms at each of the neighboring sample points; determining values for each of the waveforms at the intermediate sample points and combining the determined values at the intermediate sample point to derive the weighted influence.

A method of processing a digital signal includes providing a digital filter including neighbouring sample points and performing a sample rate increase on the digital filter to provide intermediate sample points between adjacent neighbouring sample points, said intermediate points being populated dependent on a weighted influence determined in the time domain of a predetermined number of the neighbouring sample points. The digital filter is applied to the signal where: i) one of the neighbouring sample points of the filter is applied to a corresponding sample point of the signal; ii) offset and neighbouring sample points of the signal are defined either side of the corresponding sample point, said offset points being offset in the time domain relative to the respective neighbouring sample points of the filter; and iii) the neighbouring sample points of the filter are applied to respective of the offset and neighbouring sample points of the signal.

Digital signal processing (“DSP”) block circuitry on an integrated circuit (“IC”) is adapted for use, e.g., in multiple instances of the DSP block circuitry on the IC, for implementing finite-impulse-response (“FIR”) filters that are dynamically adjustable. Advantages of such DSP block circuitries may include an increase in performance and a reduction in logic and memory usage for multi-standard FIR filters.

A system and method for representing quasi-periodic waveforms, for example, representing a plurality of limited decompositions of the quasi-periodic waveform. Each decomposition includes a first and second amplitude value and at least one time value. In some embodiments, each of the decompositions is phase adjusted such that the arithmetic sum of the plurality of limited decompositions reconstructs the quasi-periodic waveform. Data-structure attributes are created and used to reconstruct the quasi-periodic waveform. Features of the quasi-periodic wave are tracked using pattern-recognition techniques. The fundamental rate of the signal (e.g., heartbeat) can vary widely, for example by a factor of 2-3 or more from the lowest to highest frequency. To get quarter-phase representations of a component (e.g., lowest frequency “rate” component) that varies over time (by a factor of two to three) many overlapping filters use bandpass and overlap parameters that allow tracking the component's frequency version on changing quarter-phase basis.

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 including an input of the FIR filter that receives an RF input signal, a clock input configured to receive a clock signal, an output of the FIR filter that provides a filtered output signal, a plurality of signal paths including a plurality of sample-and-hold circuits and a plurality of multipliers arranged in parallel, each signal path including a respective sample-and-hold circuit and a respective multiplier being configured to receive the RF input signal and the clock signal to provide a modulated output signal, an adder configured to receive n modulated output signals from the plurality of signal paths and combine the n modulated output signals to produce the filtered output signal, and a controller.

An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.