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

Patent metadata

Field	Value
Publication number	US-12014746-B2
Application number	US-202217580578-A
Country	US
Kind code	B2
Filing date	Jan 20, 2022
Priority date	Jul 28, 2014
Publication date	Jun 18, 2024
Grant date	Jun 18, 2024

Abstract

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

The invention claimed is:

1. A method for processing an audio signal, the method comprising:
filtering audio frames of the 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, wherein filtering the audio frames of the audio signal comprises filtering a past frame of the audio signal using a set of first filter parameters for obtaining a filtered past frame, and filtering a current frame of the audio signal using a set of second filter parameters for obtaining a filtered current frame, and
removing a discontinuity between the filtered past frame and the filtered current frame of the audio signal using a linear predictive filter,
wherein initial states of the linear predictive filter are obtained by filtering a last part of the past frame using the set of second filter parameters.

2. The method of claim 1 , further comprising estimating the linear predictive filter using the audio signal.

3. The method of claim 2 , wherein estimating the linear predictive filter comprises estimating the linear predictive filter based on the past and/or current frame of the audio signal or based on the filtered past frame of the audio signal using the Levinson-Durbin algorithm.

4. The method of claim 1 , wherein the linear predictive filter comprises a linear predictive filter of an audio codec.

5. A non-transitory digital storage medium having stored thereon a computer program product for performing, when run on a computer, a method for processing an audio signal, the method comprising:
filtering audio frames of the 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, wherein filtering the audio frames of the audio signal comprises filtering a past frame of the audio signal using a set of first filter parameters for obtaining a filtered past frame, and filtering a current frame of the audio signal using a set of second filter parameters for obtaining a filtered current frame, and
removing a discontinuity between the filtered past frame and the filtered current frame of the audio signal using a linear predictive filter,
wherein initial states of the linear predictive filter are obtained by filtering a last part of the past frame using the set of second filter parameters.

6. An apparatus for processing an audio signal,
wherein the apparatus comprises a processor configured to:
filter audio frames of the 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, wherein filtering the audio frames of the audio signal comprises
filtering a past frame of the audio signal using a set of first filter parameters for obtaining a filtered past frame, and
filtering a current frame of the audio signal using a set of second filter parameters for obtaining a filtered current frame, and
remove a discontinuity between the filtered past frame and the filtered current frame of the audio signal using a linear predictive filter,
wherein initial states of the linear predictive filter are obtained by filtering a last part of the past frame using the set of second filter parameters.

7. An audio decoder, comprising an apparatus of claim 6 .

8. An audio encoder, comprising an apparatus of claim 6 .

Assignees

• Fraunhofer Ges Forschung

Inventors

• Ravelli Emmanuel
• Jander Manuel
• Pietrzyk Grzegorz
• Dietz Martin
• Gayer Marc

Classifications

• G10L19/12

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

• H04B1/1027

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

• G10L19/03

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

• G10L19/005

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

• G11B27/038

  Cross-faders therefor · CPC title