An apparatus for modifying a sampling rate includes a forward transformer for forming a first version of a spectrogram by means of transformation with a first transformation length from an information signal with a first sampling rate. The apparatus includes a processor for forming a second version of the spectrogram with a lower bandwidth than the first version. The apparatus includes a reverse transformer for forming a coarsely pre-modified information signal with a second sampling rate that is reduced with respect to the first sampling rate, by means of reverse transformation of the second version of the spectrogram with a second transformation length that is reduced with respect to the first transformation length. The apparatus includes a time domain interpolator for acquiring an information signal with a third sampling rate that is modified with respect to the second sampling rate, by means of interpolation of the pre-modified information signal.

A Finite Impulse Response (FIR) filter that reduces the complexity of the hardware required for a filter with a high decimation factor while achieving similar performance of prior art poly-phase filters of greater complexity. The FIR filter includes a small number of multiply-and-accumulate (MAC) units connected in parallel to each other between an input stream and an output stream. The MAC units are provided with coefficients from a memory. In an example implementation, the memory is addressed by a counter and the output of the memory selected by a multiplexer for suppling the coefficients.

A method includes receiving an input digital signal and applying the input digital signal to digital filter processing with a corner frequency to produce a filtered output digital signal. The digital filter processing includes a set of multiplication operations using a set of filter multiplication coefficients. The set of multiplication operations is performed by alternately using a first set of approximate multiplication coefficients and a second set of approximate multiplication coefficients different from the first set of approximate multiplication coefficients. The approximate multiplication coefficients in the first set of approximate multiplication coefficients and the second set of approximate multiplication coefficients approximate multiplication coefficients in the set of filter multiplication coefficients as a function of negative power-of-two values. The alternating of multiplication operations results in digital filter processing with average corner frequency approximating the corner frequency.

Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a linear subband of the audio signal. The method also includes determining a filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.

A radio communication receiver and a method performed by the radio communication receiver for configuring a Notch filter of the radio communication receiver. The method comprises retrieving stored and previously determined filter coefficients from a set of filter coefficients, where the retrieved filter coefficients constitute a fraction of the total number of filter coefficients; and setting the rest of the filter coefficients to one. The method further comprises normalising the retrieved filter coefficients; and transforming the filter coefficients such that the Notch position ends up at the one or more frequencies to be filtered out.

Various embodiments are generally directed to techniques for optimizing convolution filters. Generally, embodiments may determine, based on an analysis of a plurality of values of a convolution filter, an optimization operation to optimize at least one value of the plurality of values of the convolution filter. Embodiments may perform the optimization operation on the values of the convolution filter to generate an optimized convolution filter. Embodiments may also perform a convolution operation by a convolution logic based on the optimized convolution filter and an input data.

A receiving device includes: a resampler to convert a sampling rate of a reception signal, and output a first signal that is a signal having been subjected to sampling rate conversion; an equalizer to perform an adaptive equalization process using the first signal as an input, and output a second signal that is a signal having been subjected to the adaptive equalization process and having a sampling rate that is an integer fraction of an input signal; a correlation calculator to calculate a correlation function between the first signal and the second signal; and a rate controller to control a rate conversion ratio for sampling rate conversion in the resampler on a basis of the correlation function.

A parallel transfer rate converter inputs first parallel data with number of samples being S1 pieces in synchronism with a first clock, and outputs second parallel data with number of samples being S2=S1(m/p) pieces (p is an integer equal to or larger than 1) in synchronism with a second clock having a frequency which is p/m times of a frequency of the first clock. A convolution operation device inputs the second parallel data in synchronism with the second clock, generates third parallel data with number of samples being S3=S2(n/m) pieces (S3 is an integer equal to or larger than 1) by executing a convolution operation with a coefficient indicating a transmission characteristic to the second parallel data, and outputs the third parallel data in synchronism with the second clock.

Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a linear subband of the audio signal. The method also includes determining a filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.

A parallel transfer rate converter inputs first parallel data with number of samples being S1 pieces in synchronism with a first clock, and outputs second parallel data with number of samples being S2=S1(m/p) pieces (p is an integer equal to or larger than 1) in synchronism with a second clock having a frequency which is p/m times of a frequency of the first clock. A convolution operation device inputs the second parallel data in synchronism with the second clock, generates third parallel data with number of samples being S3=S2(n/m) pieces (S3 is an integer equal to or larger than 1) by executing a convolution operation with a coefficient indicating a transmission characteristic to the second parallel data, and outputs the third parallel data in synchronism with the second clock.