Method and apparatus for processing an audio signal, audio decoder, and audio encoder for removing a discontinuity between frames by subtracting a portion of a zero-input-reponse

Patent metadata

Field	Value
Publication number	US-12033648-B2
Application number	US-202318339915-A
Country	US
Kind code	B2
Filing date	Jun 22, 2023
Priority date	Jul 28, 2014
Publication date	Jul 9, 2024
Grant date	Jul 9, 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 filter. Removing the discontinuity further comprises processing a beginning portion of the filtered current frame, wherein the beginning portion of the current frame comprises a predefined number of samples being less or equal than a total number of samples in the current frame, and wherein processing the beginning portion of the current frame comprises subtracting a beginning portion of a zero-input-response (ZIR) from the beginning portion of the filtered current frame.

First claim

What is claimed is:

1. A method for processing an audio signal, the method comprising:
filtering a past frame of the audio signal using a set of first filter parameters for obtaining a filtered past frame,
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, and
wherein removing the discontinuity comprises processing a beginning portion of the filtered current frame, wherein the beginning portion of the current frame comprises a predefined number of samples being less or equal than a total number of samples in the current frame, and wherein processing the beginning portion of the current frame comprises subtracting a beginning portion of a zero-input-response (ZIR) from the beginning portion of the filtered current frame.

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 , comprising processing the current frame of the audio signal on a sample-by-sample basis using a recursive filter, and wherein processing a sample of the beginning portion of the current frame comprises:
filtering the sample with the recursive filter using the second filter parameters of the current frame for producing a filtered sample, and
subtracting a corresponding ZIR sample from the filtered sample for producing the corresponding sample of the filtered current frame.

5. The method of claim 4 , wherein filtering and subtracting are repeated until the last sample in the beginning portion of the current frame is processed, and wherein the method further comprises filtering the remaining samples in the current frame with the recursive filter using the second filter parameters of the current frame.

6. The method of claim 1 , comprising generating the ZIR, wherein generating the ZIR comprises:
filtering M last samples of the past frame with a non-recursive or a recursive filter using the second filter parameters used for filtering the current frame for producing a first portion of a filtered signal, wherein M is the order of the linear predictive filter,
subtracting from the first portion of the filtered signal the M last samples of the filtered past frame for generating a second portion of the filtered signal, and
generating a ZIR of a linear predictive filter by filtering a frame of zero samples with the linear predictive filter having initial states equal to the second portion of the filtered signal.

7. The method of claim 6 , comprising windowing the ZIR such that its amplitude decreases faster to zero.

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

9. The method of claim 1 , comprising filtering the current frame of the audio signal using a non-recursive filter for producing the filtered current frame.

10. 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 a past frame of the audio signal using a set of first filter parameters for obtaining a filtered past frame,
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, and
wherein removing the discontinuity comprises processing a beginning portion of the filtered current frame, wherein the beginning portion of the current frame comprises a predefined number of samples being less or equal than a total number of samples in the current frame, and wherein processing the beginning portion of the current frame comprises subtracting a beginning portion of a zero-input-response (ZIR) from the beginning portion of the filtered current frame.

11. An apparatus for processing an audio signal,
Wherein the apparatus comprises a processor configured to:
filter a past frame of the audio signal using a set of first filter parameters for obtaining a filtered past frame,
filter 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, and
wherein removing the discontinuity comprises processing a beginning portion of the filtered current frame, wherein the beginning portion of the current frame comprises a predefined number of samples being less or equal than a total number of samples in the current frame, and wherein processing the beginning portion of the current frame comprises subtracting a beginning portion of a zero-input-response (ZIR) from the beginning portion of the filtered current frame.

12. An audio decoder, comprising an apparatus of claim 11 .

13. An audio encoder, comprising an apparatus of claim 11 .

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