Resampling output signals of QMF based audio codecs

Patent metadata

Field	Value
Publication number	US-11804232-B2
Application number	US-202117172335-A
Country	US
Kind code	B2
Filing date	Feb 10, 2021
Priority date	Aug 12, 2010
Publication date	Oct 31, 2023
Grant date	Oct 31, 2023

Abstract

An apparatus for processing an audio signal includes a configurable first audio signal processor for processing the audio signal in accordance with different configuration settings to obtain a processed audio signal, wherein the apparatus is adapted so that different configuration settings result in different sampling rates of the processed audio signal. The apparatus furthermore includes n analysis filter bank having a first number of analysis filter bank channels, a synthesis filter bank having a second number of synthesis filter bank channels, a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate, and a controller for controlling the first number of analysis filter bank channels or the second number of synthesis filter bank channels in accordance with a configuration setting.

First claim

What is claimed is:

1. An apparatus for processing an audio signal, comprising:
a configurable first audio signal processor for processing the audio signal to obtain a processed audio signal, wherein the apparatus is adapted to control the sampling rate of the processed audio signal in accordance with a configuration setting received at run time,
an analysis filter bank having a first number of analysis filter bank channels, wherein the analysis filter bank is adapted to transform an analysis filter bank input signal from a time-domain to a time-frequency domain,
a synthesis filter bank having a second number of synthesis filter bank channels, wherein the synthesis filter bank is adapted to transform an output of the analysis filter bank from the time-frequency domain to the time domain, and
a second audio processor being adapted to receive and process an audio signal having a predetermined sampling rate,
wherein the apparatus comprises a controller for controlling the first number of analysis filter bank channels, so that an audio signal output of the synthesis filter bank has the predetermined sampling rate or a sampling rate being different from the predetermined sampling rate and being closer to the predetermined sampling rate than a sampling rate of the analysis filter bank input signal.

2. The apparatus according to claim 1 ,
characterized in that the analysis filter bank is adapted to transform the analysis filter bank input signal being represented in the time-domain into a first time-frequency domain audio signal having a plurality of first subband signals, wherein the number of first subband signals is equal to the first number of analysis filter bank channels,
wherein the apparatus further comprises a signal adjuster being adapted to generate a second time-frequency domain audio signal having a plurality of second subband signals from the first time-frequency domain audio signal based on a configuration setting, such that the number of second subband signals of the second time-frequency domain audio signal is equal to the number of synthesis filter bank channels, and wherein the number of second subband signals of the second time-frequency domain audio signal is different from the number of subband signals of the first time-frequency domain audio signal, and
wherein the synthesis filter bank is adapted to transform the second time-frequency domain audio signal into a time domain audio signal as the audio signal output of the synthesis filter bank.

3. The apparatus according to claim 2 , characterized in that the signal adjuster is adapted to generate the second time-frequency domain audio signal by generating at least one additional subband signal.

4. The apparatus according to claim 3 , characterized in that the signal adjuster is adapted to generate at least one additional subband signal by conducting spectral band replication to generate at least one additional subband signal.

5. The apparatus according to claim 3 , characterized in that the signal adjuster is adapted to generate a zero signal as additional subband signal.

6. The apparatus according to claim 1 , characterized in that the analysis filter bank is a QMF analysis filter bank and wherein the synthesis filter bank is a QMF synthesis filter bank.

7. The apparatus according to claim 1 , characterized in that the analysis filter bank is an MDCT analysis filter bank and wherein the synthesis filter bank is an MDCT synthesis filter bank.

8. The apparatus according to claim 1 , characterized in that the apparatus furthermore comprises an additional resampler being adapted to resample a synthesis filter bank output signal having a first synthesis sampling rate to output a resampled output signal having a second synthesis sampling rate.

9. The apparatus according to claim 1 , characterized in that the apparatus is adapted to feed a synthesis filter bank output signal having a first synthesis sampling rate into an analysis filter bank as an analysis filter bank input signal.

10. The apparatus according to claim 1 , characterized in that the controller is adapted to determine the first number of analysis filter bank channels or the second number of synthesis filter bank channels based on a tolerable error.

11. The apparatus according to claim 10 , characterized in that the controller comprises an error comparator for comparing the actual error with a tolerable error.

12. A method for processing an audio signal, comprising:
processing an audio signal to obtain a first processed audio signal, wherein a sampling rate of the first processed audio signal is controlled in accordance with a configuration setting received at run time,
controlling a first number of analysis filter bank channels of an analysis filter bank, so that an audio signal output by the synthesis filter bank has the predetermined sampling rate or a sampling rate being different from the predetermined sampling rate and being closer to the predetermined sampling rate than the sampling rate of an input signal into the analysis filter bank, and
processing the audio signal output having the predetermined sampling rate,
wherein the analysis filter bank transforms the input signal into the analysis filter bank from a time-domain to a time-frequency domain, and
wherein the synthesis filter bank transforms an output of the analysis filter bank from the time-frequency domain to the time domain.

13. A non-transitory computer-readable medium having stored thereon a computer program for implementing the method of claim 12 , when the computer program is executed by a computer or a signal processor.

Assignees

• Fraunhofer Ges Forschung

Inventors

• Lohwasser Markus
• Jander Manuel
• Neuendorf Max
• Geiger Ralf
• Schnell Markus
• Hildenbrand Matthias
• Chalupka Tobias

Classifications

• G10L19/24 Primary

  Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding · CPC title

• G10L19/008 Primary

  Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing · CPC title

• G10L19/0204

  using subband decomposition · CPC title

• G10L19/16

  Vocoder architecture · CPC title

• G10L19/26

  Pre-filtering or post-filtering · CPC title