Audio signal processing method and device

Patent metadata

Field	Value
Publication number	US-10129685-B2
Application number	US-201815974689-A
Country	US
Kind code	B2
Filing date	May 9, 2018
Priority date	Apr 2, 2014
Publication date	Nov 13, 2018
Grant date	Nov 13, 2018

Abstract

To this end, the present invention provides a method for processing an audio signal, including: receiving an input audio signal including at least one of a multi-channel signal and a multi-object signal; receiving type information of a filter set for binaural filtering of the input audio signal, the type of the filter set being one of a finite impulse response (FIR) filter, a parameterized filter in a frequency domain, and a parameterized filter in a time domain; receiving filter information for binaural filtering based on the type information; and performing the binaural filtering for the input audio signal by using the received filter information, wherein when the type information indicates the parameterized filter in the frequency domain, in the receiving of the filter information, a subband filter coefficient having a length determined for each subband of a frequency domain is received, and in the performing of the binaural filtering, each subband signal of the input audio signal is filtered by using the subband filter coefficient corresponding thereto and an apparatus for processing an audio signal by using the same.

First claim

What is claimed is:

1. A method for processing an audio signal, the method comprising:
receiving an input audio signal;
obtaining block length information and number of blocks information of filter coefficients for each subband;
receiving filter coefficients for each of a subband index, a binaural filter pair index, a block index in the number of blocks, and a time slot index in each block having a length according to the block length information, wherein a total length of filter coefficients for a same subband index and a same binaural filter pair index is determined based on a filter order of the corresponding subband; and
filtering each subband signal of the input audio signal by using the received filter coefficients corresponding thereto.

2. The method of claim 1 , wherein the filter order is determined to be variable in a frequency domain.

3. The method of claim 1 , wherein the filter order is determined based on characteristic information extracted from filter coefficients of the corresponding subband.

4. The method of claim 1 , wherein the filter order has a single value for each subband.

5. The method of claim 1 , wherein the filter coefficients for each of the indexes include a left output channel filter coefficient of a real value, a left output channel filter coefficient of an imaginary value, a right output channel filter coefficient of the real value, and a right output channel filter coefficient of the imaginary value.

6. The method of claim 1 , wherein the number of blocks in a subband is determined based on a value obtained by dividing a reference filter length in the subband by the length according to the block length information, and
wherein the reference filter length is determined based on a filter order of the corresponding subband.

7. The method of claim 1 , wherein the filter coefficients are received in a unit of a block having a length according to the block length information.

8. A method for processing an audio signal, the method comprising:
receiving an input audio signal;
receiving fast Fourier transform (FFT) length information for each subband;
obtaining block length information of filter coefficients for each subband based on the FFT length information;
receiving number of blocks information of filter coefficients for each subband;
receiving filter coefficients for each set of indexes, wherein the set of indexes includes a subband index, a binaural filter pair index, a block index in the number of blocks, and a time slot index in each block having a length according to the block length information, wherein a total length of filter coefficients for a same subband index and a same binaural filter pair index is determined based on a filter order of the corresponding subband; and
filtering each subband signal of the input audio signal by using the received filter coefficients corresponding thereto.

9. The method of claim 8 , wherein the filter order is determined to be variable in a frequency domain.

10. The method of claim 8 , wherein the block length is determined as a value of power of 2 having an FFT length of the corresponding subband as an exponent value.

11. An apparatus for processing an audio signal, the apparatus comprising:
a fast convolution unit configured to perform filtering one or more subband signals of an input audio signal,
wherein the fast convolution unit is configured to:
receive an input audio signal,
obtain block length information and number of blocks information of filter coefficients for each subband,
receive filter coefficients for each of a subband index, a binaural filter pair index, a block index in the number of blocks, and a time slot index in each block having a length according to the block length information, wherein a total length of filter coefficients for a same subband index and a same binaural filter pair index is determined based on a filter order of the corresponding subband, and
filter each subband signal of the input audio signal by using the received filter coefficients corresponding thereto.

12. The apparatus of claim 11 , wherein the filter order is determined to be variable in a frequency domain.

13. The apparatus of claim 11 , wherein the filter order is determined based on characteristic information extracted from filter coefficients of the corresponding subband.

14. The apparatus of claim 11 , wherein the filter order has a single value for each subband.

15. The apparatus of claim 11 , wherein the filter coefficients for each of the indexes include a left output channel filter coefficient of a real value, a left output channel filter coefficient of an imaginary value, a right output channel filter coefficient of the real value, and a right output channel filter coefficient of the imaginary value.

16. The apparatus of claim 11 , wherein the number of blocks in a subband is determined based on a value obtained by dividing a reference filter length in the subband by the length according to the block length information, and
wherein the reference filter length is determined based on a filter order of the corresponding subband.

17. The apparatus of claim 11 , wherein the filter coefficients are received in a unit of a block having a length according to the block length information.

18. An apparatus for processing an audio signal, the apparatus comprising:
a fast convolution unit configured to perform filtering one or more subband signals of an input audio signal,
wherein the fast convolution unit is configured to:
receive an input audio signal,
receive fast Fourier transform (FFT) length information for each subband,
obtain block length information of filter coefficients for each subband based on the FFT length information,
receive number of blocks information of filter coefficients for each subband,
receive filter coefficients for each set of indexes, wherein the set of indexes includes a subband index, a binaural filter pair index, a block index in the number of blocks, and a time slot index in each block having a length according to the block length information, wherein a total length of filter coefficients for a same subband index and a same binaural filter pair index is determined based on a filter order of the corresponding subband, and
filter each subband signal of the input audio signal by using the received filter coefficients corresponding thereto.

19. The apparatus of claim 18 , wherein the filter order is determined to be variable in a frequency domain.

20. The apparatus of claim 18 , wherein the block length is determined as a value of power of 2 having an FFT length of the corresponding subband as an exponent value.

Assignees

• Wilus Inst Standards & Tech Inc

Inventors

• Lee Taegyu
• Oh hyun oh

Classifications

• H04S7/303

  Tracking of listener position or orientation · CPC title

• H04R2499/15

  Transducers incorporated in visual displaying devices, e.g. televisions, computer displays, laptops · CPC title

• H04S3/008 Primary

  in which the audio signals are in digital form, i.e. employing more than two discrete digital channels (data reduction aspects thereof based on psychoacoustics G10L19/02) · CPC title

• H04S7/306

  For headphones · CPC title

• G10L19/008

  Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing · CPC title