Systems and methods for producing a linear-phase digital FIR filter from two sub-filters for an audio signal. In one method, the sub-filters are provided as sub-sets having numbers of coefficients, a lower cutoff frequency of the particular sub-filter being greater than the sampling frequency of the audio signal divided by the number. The sub-sets are linearly convoluted with one another so as to form a total set having a number of coefficients greater than the numbers, and the total set is symmetrically reduced to a number less than the number, so as to form a reduced total set of the filter. A linear-phase digital FIR filter for an audio signal is created by the method.

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount. To this end, provided are a method for processing a multimedia signal including: receiving a multimedia signal having a plurality of subbands; receiving at least one proto-type filter coefficients for filtering each subband signal of the multimedia signal; converting the proto-type filter coefficients into a plurality of subband filter coefficients; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and filtering the multimedia signal by using the truncated subband filter coefficients corresponding to each subband signal and an apparatus for processing a multimedia signal using the same.

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 method includes receiving an input signal at a filter, where the filter includes a plurality of filter taps, and where each of a first filter tap and a second filter tap has a weighting coefficient. The method also includes shutting down the first filter tap based on the weighting coefficient of the first filter tap being below a threshold and the weighting coefficient of the second filter tap being below the threshold, where the second filter tap is next to the first filter tap.

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount.

To this end, provided are a method for processing a multimedia signal including: receiving a multimedia signal having a plurality of subbands; receiving at least one proto-type filter coefficients for filtering each subband signal of the multimedia signal; converting the proto-type filter coefficients into a plurality of subband filter coefficients; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and filtering the multimedia signal by using the truncated subband filter coefficients corresponding to each subband signal and an apparatus for processing a multimedia signal using the same.

A method includes receiving an input signal at a filter, where the filter includes a plurality of filter taps, and where each of a first filter tap and a second filter tap has a weighting coefficient. The method also includes shutting down the first filter tap based on the weighting coefficient of the first filter tap being below a threshold and the weighting coefficient of the second filter tap being below the threshold, where the second filter tap is next to the first filter tap.

The invention relates to a method for producing a linear-phase digital FIR filter (4) from two sub-filters (6a, b) for an audio signal (AU, AF), in which method the sub-filters (6a, b) are provided as sub-sets (Ta, b) having numbers (Ma, b) of coefficients, a lower cutoff frequency (Ga, b) of the particular sub-filter (6a, b) being greater than the sampling frequency of the audio signal (AU, AF) divided by the number (Ma, b), the sub-sets (Ta, b) are linearly convoluted with one another so as to form a total set (SU) having a number (L) of coefficients greater than the numbers (Ma, b), and the total set (SU) is symmetrically reduced to a number (Z) less than the number (L), so as to form a reduced total set (SK) of the filter (4). A linear-phase digital FIR filter (4) for an audio signal (AU, AF) is created by the method. A signal processor (2) for an audio signal (AU, AF) contains the filter (4).

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.

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount.

To this end, provided are a method for processing a multimedia signal including: receiving a multimedia signal having a plurality of subbands; receiving at least one proto-type filter coefficients for filtering each subband signal of the multimedia signal; converting the proto-type filter coefficients into a plurality of subband filter coefficients; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and filtering the multimedia signal by using the truncated subband filter coefficients corresponding to each subband signal and an apparatus for processing a multimedia signal using the same.

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount.

To this end, provided are a method for processing an audio signal including: receiving multi-audio signals including multi-channel or multi-object signals, each of the multi-audio signals including a plurality of subband signals, and the plurality of subband signals including a signal of a first subband group having low frequencies and a signal of a second subband group having high frequencies based on a predetermined frequency band; receiving at least one parameter corresponding to each subband signal of the second subband group, the at least one parameter being extracted from binaural room impulse response (BRIR) subband filter coefficients corresponding to each subband signal of the second subband group; and performing tap-delay line filtering of the subband signal of the second subband group by using the received parameter and an apparatus for processing an audio signal using the same.