Audio signal processing method and device

US10469978B2 · US · B2

Patent metadata
FieldValue
Publication numberUS-10469978-B2
Application numberUS-201816159624-A
CountryUS
Kind codeB2
Filing dateOct 13, 2018
Priority dateApr 2, 2014
Publication dateNov 5, 2019
Grant dateNov 5, 2019

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  1. Title

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  2. Abstract

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  3. Assignees and inventors

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  4. Key dates

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  5. First independent claim

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  6. CPC / IPC classifications

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  7. Citations and related patents

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Abstract

Official abstract text for this publication.

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

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What is claimed is: 1. A method for processing an audio signal, the method comprising: receiving an input audio signal; receiving a set of filter coefficients for each subband; determining a scalability factor for performing a binaural rendering of the input audio signal; obtaining a filter order for each subband based on the scalability factor, wherein the filter order is determined to be variable for each subband; editing the set of filter coefficients for each subband based on a filter order of the corresponding subband, wherein a length of the edited set of filter coefficients is determined based on the filter order of the corresponding subband; and filtering each subband signal of the input audio signal by using the edited set of filter coefficients corresponding thereto. 2. The method of claim 1 , wherein the scalability factor indicates energy ratio which determines an energy decay time of the corresponding set of filter coefficients. 3. The method of claim 2 , wherein the filter order is obtained based on an energy decay time extracted from a set of proto-type filter coefficients for the corresponding subband. 4. The method of claim 1 , wherein the energy decay time includes a reverberation time. 5. The method of claim 4 , wherein the scalability factor is determined among reverberation time (RT)10, RT20, RT40. 6. The method of claim 1 , wherein the received set of filter coefficients has a length according to a base filter order corresponding to a maximum scalability factor, and wherein the editing is performed by truncating the set of filter coefficients based on the filter order corresponding to the scalability factor determined for performing the binaural rendering. 7. An apparatus for processing an audio signal, the apparatus comprising: a binaural renderer configured to perform filtering one or more subband signals of an input audio signal, wherein the binaural renderer is configured to: receive an input audio signal; receive a set of filter coefficients for each subband; determine a scalability factor for performing a binaural rendering of the input audio signal; obtain a filter order for each subband based on the scalability factor, wherein the filter order is determined to be variable for each subband; edit the set of filter coefficients for each subband based on a filter order of the corresponding subband, wherein a length of the edited set of filter coefficients is determined based on the filter order of the corresponding subband; and filter each subband signal of the input audio signal by using the edited set of filter coefficients corresponding thereto. 8. The apparatus of claim 7 , wherein the scalability factor indicates energy ratio which determines an energy decay time of the corresponding set of filter coefficients. 9. The apparatus of claim 8 , wherein the filter order is obtained based on an energy decay time extracted from a set of proto-type filter coefficients for the corresponding subband. 10. The apparatus of claim 7 , wherein the energy decay time includes a reverberation time. 11. The apparatus of claim 10 , wherein the scalability factor is determined among reverberation time (RT)10, RT20, RT40. 12. The apparatus of claim 7 , wherein the received set of filter coefficients has a length according to a base filter order corresponding to a maximum scalability factor, and wherein the binaural renderer performs the editing by truncating the set of filter coefficients based on the filter order corresponding to the scalability factor determined for performing the binaural rendering.

Assignees

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Classifications

  • Positioning of individual sound objects, e.g. moving airplane, within a sound field (H04S2420/13 takes precedence) · CPC title

  • Synergistic effects of band splitting and sub-band processing · CPC title

  • 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 title

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

  • H04R3/04Primary

    for correcting frequency response · CPC title

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What does patent US10469978B2 cover?
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 filt…
Who is the assignee on this patent?
Wilus Inst Standards & Tech Inc
What technology area does this patent fall under?
Primary CPC classification H04R3/04. Mapped technology areas include Electricity.
When was this patent published?
Publication date Tue Nov 05 2019 00:00:00 GMT+0000 (Coordinated Universal Time) (B2). Legal status and post-grant events are not shown on this page.
What related patents are in patentsdb?
We list 11 related publications on this page (citations in our corpus or others sharing the same primary CPC).