Resampling output signals of QMF based audio codecs

Patent metadata

Field	Value
Publication number	US-12154583-B2
Application number	US-202318378028-A
Country	US
Kind code	B2
Filing date	Oct 9, 2023
Priority date	Aug 12, 2010
Publication date	Nov 26, 2024
Grant date	Nov 26, 2024

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

The invention claimed is:

1. An apparatus for processing an audio signal, comprising:
a configurable first audio signal processor for processing an audio signal to acquire a processed audio signal,
an analysis filter bank comprising a first number of analysis filter bank channels,
a synthesis filter bank comprising 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,
a second audio processor being adapted to receive and process the audio signal comprising a predetermined sampling rate, and
a controller for controlling the first number of analysis filter bank channels in accordance with a configuration setting provided to the configurable first audio signal processor, so that an audio signal output of the synthesis filter bank comprises 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 an analysis filter bank input signal,
wherein the apparatus is adapted to receive the configuration setting at run time.

2. The apparatus according to claim 1 ,
wherein the analysis filter bank is adapted to transform the analysis filter bank input signal being represented in a time-domain into a first time-frequency domain audio signal comprising 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 comprising a plurality of second subband signals from the first time-frequency domain audio signal based on the 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 , wherein 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 , wherein 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 , wherein the signal adjuster is adapted to generate a zero signal as additional subband signal.

6. The apparatus according to claim 1 , wherein 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 , wherein 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 , wherein the apparatus furthermore comprises an additional resampler being adapted to receive a synthesis filter bank output signal comprising a first synthesis sampling rate, and wherein the additional resampler resamples the synthesis filter bank output signal to generate a resampled output signal comprising a second synthesis sampling rate.

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

10. The apparatus according to claim 1 , wherein 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 , wherein 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 acquire a processed audio signal,
controlling a first number of analysis filter bank channels of an analysis filter bank in accordance with a configuration setting, so that an audio signal output of a synthesis filter bank comprises 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 an analysis filter bank input signal, wherein the synthesis filter bank transforms an output of the analysis filter bank from the time-frequency domain to the time domain by the synthesis filter bank, and
processing the audio signal output comprising the predetermined sampling rate,
wherein the configuration setting is received at run time.

13. A non-transitory computer-readable medium comprising a computer program for performing the method for processing an audio signal, said method comprising:
processing an audio signal to acquire a processed audio signal,
controlling a first number of analysis filter bank channels of an analysis filter bank in accordance with a configuration setting, so that an audio signal output of a synthesis filter bank comprises 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 an analysis filter bank input signal, wherein the synthesis filter bank transforms an output of the analysis filter bank from the time-frequency domain to the time domain by the synthesis filter bank, and
processing the audio signal output comprising the predetermined sampling rate,
wherein the configuration setting is received at run time.

Assignees

• Fraunhofer Ges Forschung

Inventors

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

Classifications

• H03H17/0211

  using specific transformation algorithms, e.g. WALSH functions, Fermat transforms, Mersenne transforms, polynomial transforms, Hilbert transforms (correlation computation G06F17/156) · CPC title

• G10L19/26

  Pre-filtering or post-filtering · CPC title

• G10L19/0204

  using subband decomposition · 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

• G10L19/02

  using spectral analysis, e.g. transform vocoders or subband vocoders · CPC title