Method for generating filter for audio signal and parameterizing device therefor

• US 10,204,630 B2
• Filed: 10/22/2014
• Issued: 02/12/2019
• Est. Priority Date: 10/22/2013
• Status: Active Grant

• Current Assignee: Electronics and Telecommunications Research Institute, Industry Academic Cooperation Foundation, WILUS Inc.
• Original Assignee: Electronics and Telecommunications Research Institute, Industry Academic Cooperation Foundation, WILUS Inc.
• Inventors: Lee, Taegyu, Oh, Hyunoh, Seo, Jeongil, Lee, Yongju, Beack, Seungkwon, Kang, Kyeongok, Jang, Daeyoung, Park, Youngcheol, Youn, Daehee
• Primary Examiner(s): Maung, Thomas
• Application Number:
  US15/031,274
• Publication Number:
  US 20160275956A1
• Time in Patent Office: 1,574 Days
• CPC Class Codes: G10H 2250/111 Impulse response, i.e. filters defined or specifed by their temporal impulse response features, e.g. for echo or reverberation applications reverberation effects G10H2210/281
• CPC Class Codes: G10H 2250/145 Convolution, e.g. of a music input signal with a desired impulse response to compute an output transforms, i.e. mathematical transforms into domains appropriate for musical signal processing, coding or compression G10H2250/215
• CPC Class Codes: G10L 19/008 Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
• CPC Class Codes: H04R 3/00 Circuits for transducers , loudspeakers or microphones
• CPC Class Codes: H04R 5/033 Headphones for stereophonic communication details thereof, e.g. relating to batteries, cables or control elements H04R1/10
• CPC Class Codes: H04S 2400/01 Multi-channel, i.e. more than two input channels, sound reproduction with two speakers wherein the multi-channel information is substantially preserved
• CPC Class Codes: H04S 2420/01 Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD]
• CPC Class Codes: H04S 2420/03 Application of parametric coding in stereophonic audio systems
• CPC Class Codes: H04S 3/00 Systems employing more than two channels, e.g. quadraphonic H04S5/00, H04S7/00 take precedence
• CPC Class Codes: H04S 3/002 Non-adaptive circuits, e.g. manually adjustable or static, for enhancing the sound image or the spatial distribution control circuits for electronic adaptation of the sound field H04S7/30
• CPC Class Codes: H04S 3/004 For headphones
• CPC Class Codes: H04S 3/008 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

Abstract

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.

Claims

1. A method for processing an audio signal, comprising:

receiving an input audio signal;

receiving one or more binaural room impulse response (BRIR) filter coefficients in a time domain corresponding to at least one position in a virtual reproduction space;

converting the BRIR filter coefficients into a plurality of sets of subband filter coefficients;

truncating each set of subband filter coefficients based on a filter order obtained by at least partially using characteristic information extracted from each set of subband filter coefficients, wherein a length of a set of truncated subband filter coefficients of at least one subband is different from a length of a set of truncated subband filter coefficients of another subband;

generating FFT filter coefficients by fast Fourier transforming (FFT) each set of truncated subband filter coefficients by a predetermined block size in a corresponding subband, wherein the predetermined block size is determined to be a smaller value between first and second values, the first value being obtained by multiplying a reference filter length of a corresponding set of truncated subband filter coefficients by 2, the second value being a predetermined maximum FFT size; and

performing block-wise fast convolution on each subband signal of the input audio signal by using the FFT filter coefficients corresponding thereto.

2. The method of claim 1, wherein the characteristic information includes reverberation time information of the corresponding set of subband filter coefficients, and the filter order has a single value for each subband.

3. The method of claim 1, wherein the reference filter length represents any one of a true value and an approximate value of the filter order in a form of power of 2.

4. The method of claim 3, wherein when the reference filter length is N and the predetermined block size corresponding thereto is M, the M is a power of 2 value and 2N=kM (k is a natural number).

5. The method of claim 1, wherein the generating FFT filter coefficients further comprising:

partitioning each set of truncated subband filter coefficients by a half of the predetermined block size;

generating temporary filter coefficients of the predetermined block size by using the partitioned filter coefficients, a first half part of the temporary filter coefficients being constituted by the partitioned filter coefficients and a second half part of the temporary filter coefficients being constituted by zero-padded values; and

fast Fourier transforming the generated temporary filter coefficients.

6. An apparatus for processing an audio signal, comprising:

a processor configured to;

receive an input audio signal;

receive one or more binaural room impulse response (BRIR) filter coefficients in a time domain corresponding to at least one position in a virtual reproduction space;

convert the BRIR filter coefficients into a plurality of sets of subband filter coefficients;

truncate each set of subband filter coefficients based on a filter order obtained by at least partially using characteristic information extracted from each set of subband filter coefficients, wherein a length of a set of truncated subband filter coefficients of at least one subband is different from a length of a set of truncated subband filter coefficients of another subband;

generate FFT filter coefficients by fast Fourier transforming (FFT) each set of truncated subband filter coefficients by a predetermined block size in a corresponding subband, wherein the predetermined block size is determined to be a smaller value between first and second values, the first value being obtained by multiplying a value twice a reference filter length of a corresponding set of truncated subband filter coefficients by 2, the second value being a predetermined maximum FFT size; and

perform block-wise fast convolution on each subband signal of the input audio signal by using the FFT filter coefficients corresponding thereto.

7. The apparatus of claim 6, wherein the characteristic information includes reverberation time information of the corresponding set of subband filter coefficients, and the filter order has a single value for each subband.

8. The apparatus of claim 6, wherein the reference filter length represents any one of a true value and an approximate value of the filter order in a form of power of 2.

9. The apparatus of claim 8, wherein when the reference filter length is N and the predetermined block size corresponding thereto is M, the M is a power of 2 value and 2N=kM (k is a natural number).

10. The apparatus of claim 6, wherein the processor is further configured to:

partition each set of truncated subband filter coefficients by a half of the predetermined block size;

generate temporary filter coefficients of the predetermined block size by using the partitioned filter coefficients, a first half part of the temporary filter coefficients being constituted by the partitioned filter coefficients and a second half part of the temporary filter coefficients being constituted by zero-padded values; and

fast Fourier transform the generated temporary filter coefficients.