An effect addition device includes at least one processor that executes a time domain convolution process of convolving a first time domain data part of impulse response of sound effects with a time domain data on an original sound, a frequency domain convolution process of convoluting a second time domain data part of the impulse response data with the time domain data on the original sound, a convolution extension process of extending a convolved state(s) of an output signal(s) resulting from the time domain convolution process and/or the frequency domain convolution process by arithmetic processing which corresponds to an all-pass filter and/or arithmetic processing which corresponds to a comb filter, and a synthesized sound effect addition process of adding a sound effect which is synthesized by execution of the time domain convolution process, the frequency domain convolution process and the convolution extension process to the original sound.

The present invention relates to a method and an apparatus for binaural rendering an audio signal using variable order filtering in frequency domain. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving a set of truncated subband filter coefficients for filtering each subband signal of the input audio signal, the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing FFT by a predetermined block size; generating at least one subframe for each subband; generating at least one filtered subframe for each subband; performing inverse FFT on the filtered subframe for each subband; and generating a filtered subband signal by overlap-adding the transformed subframe for each subband and an apparatus for processing an audio signal using the same.

The present disclosure presents a novel approach to automatic transcription of piano music in a context-dependent setting. Embodiments described herein may employ an efficient algorithm for convolutional sparse coding to approximate a music waveform as a summation of piano note waveforms convolved with associated temporal activations. The piano note waveforms may be pre-recorded for a particular piano that is to be transcribed and may optionally be pre-recorded in the specific environment where the piano performance is to be performed. During transcription, the note waveforms may be fixed and associated temporal activations may be estimated and post-processed to obtain the pitch and onset transcription. Experiments have shown that embodiments of the disclosure significantly outperform state-of-the-art music transcription methods trained in the same context-dependent setting, in both transcription accuracy and time precision, in various scenarios including synthetic, anechoic, noisy, and reverberant environments.

The present invention relates to a method and an apparatus for binaural rendering an audio signal using variable order filtering in frequency domain. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving a set of truncated subband filter coefficients for filtering each subband signal of the input audio signal, the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing FFT by a predetermined block size; generating at least one subframe for each subband; generating at least one filtered subframe for each subband; performing inverse FFT on the filtered subframe for each subband; and generating a filtered subband signal by overlap-adding the transformed subframe for each subband and an apparatus for processing an audio signal using the same.

An effect addition device includes at least one processor that executes a time domain convolution process of convolving a first time domain data part of impulse response of sound effects with a time domain data on an original sound, a frequency domain convolution process of convoluting a second time domain data part of the impulse response data with the time domain data on the original sound, a convolution extension process of extending a convolved state(s) of an output signal(s) resulting from the time domain convolution process and/or the frequency domain convolution process by arithmetic processing which corresponds to an all-pass filter and/or arithmetic processing which corresponds to a comb filter, and a synthesized sound effect addition process of adding a sound effect which is synthesized by execution of the time domain convolution process, the frequency domain convolution process and the convolution extension process to the original sound.

The present invention relates to a method and an apparatus for binaural rendering an audio signal using variable order filtering in frequency domain. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving a set of truncated subband filter coefficients for filtering each subband signal of the input audio signal, the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing FFT by a predetermined block size; generating at least one subframe for each subband; generating at least one filtered subframe for each subband; performing inverse FFT on the filtered subframe for each subband; and generating a filtered subband signal by overlap-adding the transformed subframe for each subband and an apparatus for processing an audio signal using the same.

A signal processing apparatus has a first memory in which plural pieces of FIR coefficient data used for implementing an FIR filter algorithm are stored, a second memory which stores plural pieces of input data to be subjected to the FIR filter algorithm, and a processor implements the FIR filter algorithm using the plural pieces of FIR coefficient data stored in the first memory and the plural pieces of input data stored in the second memory as many times as the number corresponding to a designated filter order, in which filter algorithm each piece of coefficient data and each piece of input data are multiplied together and resultant products are summed up. The signal processing apparatus is provided, which can implement plural sorts of FIR filter algorithms of filter order which can be changed flexibly.

The present invention relates to a method and an apparatus for binaural rendering an audio signal using variable order filtering in frequency domain. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving a set of truncated subband filter coefficients for filtering each subband signal of the input audio signal, the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing FFT by a predetermined block size; generating at least one subframe for each subband; generating at least one filtered subframe for each subband; performing inverse FFT on the filtered subframe for each subband; and generating a filtered subband signal by overlap-adding the transformed subframe for each subband and an apparatus for processing an audio signal using the same.

The present invention relates to a method and an apparatus for processing a signal, which are used to effectively reproduce an audio signal, and more particularly, to a method for generating a filter for an audio signal, which are used for implementing a filtering for input audio signals with a low computational complexity and a parameterization apparatus therefor.

To this end, provided are a method for generating a filter of an audio signal, including: receiving at least one proto-type filter coefficient for filtering each subband signal of an input audio signal; converting the proto-type filter coefficient into a plurality of subband filter coefficients; truncating each of the 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 generating FFT filter coefficients by fast Fourier transforming (FFT) the truncated subband filter coefficients by a predetermined block size in the corresponding subband and a parameterization unit using the same.

A signal processing apparatus has a first memory in which plural pieces of FIR coefficient data used for implementing an FIR filter algorithm are stored, a second memory which stores plural pieces of input data to be subjected to the FIR filter algorithm, and a processor implements the FIR filter algorithm using the plural pieces of FIR coefficient data stored in the first memory and the plural pieces of input data stored in the second memory as many times as the number corresponding to a designated filter order, in which filter algorithm each piece of coefficient data and each piece of input data are multiplied together and resultant products are summed up. The signal processing apparatus is provided, which can implement plural sorts of FIR filter algorithms of filter order which can be changed flexibly.