Method and apparatus for processing an audio signal, audio decoder, and audio encoder to filter a discontinuity by a filter which depends on two fir filters and pitch lag

US12165665B2 · US · B2

Patent metadata
FieldValue
Publication numberUS-12165665-B2
Application numberUS-202318537655-A
CountryUS
Kind codeB2
Filing dateDec 12, 2023
Priority dateJul 28, 2014
Publication dateDec 10, 2024
Grant dateDec 10, 2024

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Abstract

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A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filtering.

First claim

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The invention claimed is: 1. A method for decoding an audio signal, the method comprising: receiving an encoded audio signal; and generating a decoded audio signal; wherein generating the decoded the audio signal comprises: decoding the encoded audio signal to acquire the decoded audio signal, filtering audio frames of the decoded audio signal using the following filter: H ( z )=(1− B ( z ))/(1− C ( z )· z −T ) where B(z) and C(z) are two FIR filters and the H(z) filter parameters are the coefficients of the FIR filters B(z) and C(z), and T indicates the pitch lag, and removing a discontinuity between a filtered past frame and a filtered current frame by modifying a beginning portion of the filtered current frame by a signal acquired by linear predictive filtering a predefined signal with initial states of a linear predictive filter, the initial states defined on basis of a last part of an unfiltered past frame filtered using a set of current frame filter parameters for filtering the current frame. 2. The method of claim 1 , further comprising estimating the linear predictive filter on the filtered or non-filtered audio signal. 3. The method of claim 2 , wherein estimating the linear predictive filter comprises estimating the filter based on the past and/or current frame of the audio signal or based on the past filtered frame of the audio signal using the Levinson-Durbin algorithm. 4. The method claim 1 , wherein the linear predictive filter comprises a linear predictive filter of an audio codec. 5. A non-transitory digital storage medium having a computer program stored thereon to perform the method for decoding an audio signal, the method comprising: receiving an encoded audio signal; and generating a decoded audio signal; wherein generating the decoded audio signal comprises: decoding the encoded audio signal to acquire the decoded audio signal, filtering audio frames of the decoded audio signal using the following filter: H ( z )=(1− B ( z ))/(1− C ( z )· z −T ) where B(z) and C(z) are two FIR filters and the H(z) filter parameters are the coefficients of the FIR filters B(z) and C(z), and T indicates the pitch lag, and removing a discontinuity between a filtered past frame and a filtered current frame by modifying a beginning portion of the filtered current frame by a signal acquired by linear predictive filtering a predefined signal with initial states of a linear predictive filter, the initial states defined on basis of a last part of an unfiltered past frame filtered using a set of current frame filter parameters for filtering the current frame, when said computer program is run by a computer. 6. A decoder for decoding an audio signal, the apparatus comprising: an input for receiving an encoded audio signal; and a processor configured to generate a decoded audio signal, wherein, for generating the decoded audio signal, the processor is configured to: decode the encoded audio signal to acquire the decoded audio signal, filter audio frames of the decoded audio signal using the following filter: H ( z )=(1− B ( z ))/(1− C ( z )· z −T ) where B(z) and C(z) are two FIR filters and the H(z) filter parameters are the coefficients of the FIR filters B(z) and C(z), and T indicates the pitch lag, and remove a discontinuity between a filtered past frame and a filtered current frame by modifying a beginning portion of the filtered current frame by a signal acquired by linear predictive filtering a predefined signal with initial states of a linear predictive filter, the initial states defined on basis of a last part of an unfiltered past frame filtered using a set of current frame filter parameters for filtering the current frame.

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  • the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders · CPC title

  • assessing signal quality or detecting noise/interference for the received signal · CPC title

  • Spectral prediction for preventing pre-echo; Temporary noise shaping [TNS], e.g. in MPEG2 or MPEG4 · CPC title

  • Correction of errors induced by the transmission channel, if related to the coding algorithm · CPC title

  • Cross-faders therefor · CPC title

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What does patent US12165665B2 cover?
A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filtering.
Who is the assignee on this patent?
Fraunhofer Ges Forschung
What technology area does this patent fall under?
Primary CPC classification G10L19/20. Mapped technology areas include Physics.
When was this patent published?
Publication date Tue Dec 10 2024 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 2 related publications on this page (citations in our corpus or others sharing the same primary CPC).