Audio signal processing method and apparatus

• US 9,832,585 B2
• Filed: 03/19/2015
• Issued: 11/28/2017
• Est. Priority Date: 03/19/2014
• Status: Active Grant

• Current Assignee: GCOA Company Limited, Wilus Institute of Standards & Technology, Inc. (Good Day to Invent, Inc.)
• Original Assignee: Wilus Institute of Standards & Technology, Inc. (Good Day to Invent, Inc.)
• Inventors: Oh, Hyunoh, Lee, Taegyu, Kwak, Jinsam, Son, Juhyung
• Primary Examiner(s): Ramakrishnaiah, Melur
• Application Number:
  US15/124,029
• Publication Number:
  US 20170019746A1
• Time in Patent Office: 985 Days
• Field of Search: 381 17- 22, 704500
• CPC Class Codes: G10L 19/20 using sound class specific coding, hybrid encoders or object based coding
• 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 2400/03 Aspects of down-mixing multi-channel audio to configurations with lower numbers of playback channels, e.g. 7.1 -> 5.1 H04S2400/01 takes precedence
• CPC Class Codes: H04S 2400/11 Positioning of individual sound objects, e.g. moving airplane, within a sound field H04S2420/13 takes precedence
• 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 2420/07 Synergistic effects of band splitting and sub-band processing
• 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 an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesize an object signal and a channel signal and effectively perform binaural rendering of the synthesized signal.

To this end, provided are a method for processing an audio signal, which includes: receiving an input audio signal including a multi-channel signal; receiving truncated subband filter coefficients for filtering the input audio signal, the truncated subband filter coefficients being at least some of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal and the length of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using reverberation time information extracted from the corresponding subband filter coefficients; obtaining vector information indicating the BRIR filter coefficients corresponding to each channel of the input audio signal; and filtering each subband signal of the multi-channel signal by using the truncated subband filter coefficients corresponding to the relevant channel and subband based on the vector information and an apparatus for processing an audio signal by using the same.

Claims

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

receiving an input audio signal including a multi-channel signal;

receiving truncated subband filter coefficients for filtering the input audio signal, the truncated subband filter coefficients being at least some of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal and the length of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using reverberation time information extracted from the corresponding subband filter coefficients;

obtaining vector information indicating the BRIR filter coefficients corresponding to each channel of the input audio signal; and

filtering each subband signal of the multi-channel signal by using the truncated subband filter coefficients corresponding to the relevant channel and subband based on the vector information.

2. The method of claim 1, wherein when BRIR filter coefficients having positional information matching with positional information of a specific channel of the input audio signal are present in a BRIR filter set, the vector information indicates the relevant BRIR filter coefficients as BRIR filter coefficients corresponding to the specific channel.

3. The method of claim 1, wherein when BRIR filter coefficients having positional information matching with positional information of a specific channel of the input audio signal are not present in a BRIR filter set, the vector information indicates BRIR filter coefficients having a minimum geometric distance from the positional information of the specific channel as BRIR filter coefficients corresponding to the specific channel.

4. The method of claim 3, wherein the geometric distance is a value obtained by aggregating an absolute value of an altitude deviation between two positions and an absolute value of an azimuth deviation between the two positions.

5. The method of claim 1, wherein the length of at least one truncated subband filter coefficients is different from the length of truncated subband filter coefficients of another subband.

6. An apparatus for processing an audio signal for performing binaural rendering for an input audio signal, the apparatus comprising:

a parameterization unit configured to generate a filter for the input audio signal; and

a binaural rendering unit configured to receive the input audio signal including a multi-channel signal and filter the input audio signal by using parameters generated by the parameterization unit,

wherein the binaural rendering unit is further configured to;

receive truncated subband filter coefficients for filtering the input audio signal from the parameterization unit, the truncated subband filter coefficients being at least some of subband filter coefficients obtained from binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal and the length of the truncated subband filter coefficients being determined based on filter order information obtained by at least partially using reverberation time information extracted from the corresponding subband filter coefficients,

obtain vector information indicating the BRIR filter coefficients corresponding to each channel of the input audio signal, and

filter each subband signal of the multi-channel signal by using the truncated subband filter coefficients corresponding to the relevant channel and subband based on the vector information.

7. The apparatus of claim 6, wherein when BRIR filter coefficients having positional information matching with positional information of a specific channel of the input audio signal are present in a BRIR filter set, the vector information indicates the relevant BRIR filter coefficients as BRIR filter coefficients corresponding to the specific channel.

8. The apparatus of claim 6, wherein when BRIR filter coefficients having positional information matching with positional information of a specific channel of the input audio signal are not present in a BRIR filter set, the vector information indicates BRIR filter coefficients having a minimum geometric distance from the positional information of the specific channel as BRIR filter coefficients corresponding to the specific channel.

9. The apparatus of claim 8, wherein the geometric distance is a value obtained by aggregating an absolute value of an altitude deviation between two positions and an absolute value of an azimuth deviation between the two positions.

10. The apparatus of claim 6, wherein the length of at least one truncated subband filter coefficients is different from the length of truncated subband filter coefficients of another subband.