Low delay modulated filter bank

• US 9,722,578 B2
• Filed: 11/17/2016
• Issued: 08/01/2017
• Est. Priority Date: 02/18/2009
• Status: Active Grant

• Current Assignee: Dolby International AB (Dolby Laboratories Incorporated)
• Original Assignee: Dolby International AB (Dolby Laboratories Incorporated)
• Inventors: Ekstrand, Per
• Primary Examiner(s): Yaary, Michael D
• Application Number:
  US15/353,914
• Publication Number:
  US 20170070211A1
• Time in Patent Office: 257 Days
• Field of Search: None
• CPC Class Codes: G06F 17/10 Complex mathematical operations function generation by table look-up G06F1/03; evaluation of elementary functions by calculation G06F7/544
• CPC Class Codes: G06F 17/11 for solving equations , e.g. nonlinear equations, general mathematical optimization problems optimization specially adapted for a specific administrative, business or logistic context G06Q10/04
• CPC Class Codes: G10L 19/008 Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
• CPC Class Codes: G10L 19/0204 using subband decomposition
• CPC Class Codes: G10L 19/0208 Subband vocoders
• CPC Class Codes: G10L 19/26 Pre-filtering or post-filtering
• CPC Class Codes: G10L 19/265 Pre-filtering, e.g. high frequency emphasis prior to encoding
• CPC Class Codes: G10L 21/00 Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility G10L19/00 takes precedence
• CPC Class Codes: H03H 17/0201 Wave digital filters
• CPC Class Codes: H03H 17/0248 Filters characterised by a particular frequency response or filtering method
• CPC Class Codes: H03H 17/0266 Filter banks
• CPC Class Codes: H03H 17/0275 comprising non-recursive filters
• CPC Class Codes: H03H 17/0294 Variable filters; Programmable filters
• CPC Class Codes: H03H 2017/0297 Coefficients derived from input parameters
• CPC Class Codes: H04R 3/04 for correcting frequency response
• CPC Class Codes: H04S 7/307 Frequency adjustment, e.g. tone control H04S7/301 takes precedence

Abstract

The document relates to modulated sub-sampled digital filter banks, as well as to methods and systems for the design of such filter banks. In particular, the present document proposes a method and apparatus for the improvement of low delay modulated digital filter banks. The method employs modulation of an asymmetric low-pass prototype filter and a new method for optimizing the coefficients of this filter. Further, a specific design for a 64 channel filter bank using a prototype filter length of 640 coefficients and a system delay of 319 samples is given. The method substantially reduces artifacts due to aliasing emerging from independent modifications of subband signals, for example when using a filter bank as a spectral equalizer. The method is preferably implemented in software, running on a standard PC or a digital signal processor (DSP), but can also be hardcoded on a custom chip. The method offers improvements for various types of digital equalizers, adaptive filters, multiband companders and spectral envelope adjusting filter banks used in high frequency reconstruction (HFR) or parametric stereo systems.

Claims

1. An apparatus for generating complex-valued output samples, the apparatus comprising:

memory that stores real-valued input audio subband samples, a windowed sample vector, and complex-valued output audio samples; and

a complex-valued low delay analysis filter bank that generates the complex-valued output audio samples in response to the real-valued input audio subband samples, the windowed sample vector, and prototype filter coefficients, wherein the complex-valued low delay analysis filter bank;

shifts a first subset of the real-valued input audio subband samples in the memory;

multiplies the first subset of the real-valued input audio subband samples by the prototype filter coefficients to generate windowed samples;

combines the windowed samples to generate the windowed sample vector;

multiplies the windowed sample vector by a complex-valued exponential modulation matrix to generate complex-valued output audio subband samples; and

stores the complex-valued output audio subband samples in the memory.

2. The apparatus of claim 1 wherein the prototype filter coefficients represent a low delay asymmetric low-pass prototype filter.

3. The apparatus of claim 1 wherein the real-valued input audio subband samples are shifted by 32 positions in the memory.

4. The apparatus of claim 1 wherein the prototype filter coefficients comprise an array of 640 elements and the windowed samples comprise an array of 320 elements.

5. The apparatus of claim 1 wherein the complex-valued low delay analysis filter bank is a QMF filter bank.

6. The apparatus of claim 1 wherein the real-valued input audio subband samples are time domain audio samples.

7. The apparatus of claim 1 wherein the complex-valued low delay analysis filter bank is part of a high frequency reconstruction (HFR) module that reconstructs high frequency components of the audio signal with low processing delay.

8. The apparatus of claim 1 further comprising a phase shifter that shifts the phase of the complex-valued output audio subband samples.

9. The apparatus of claim 1 wherein the complex-valued output audio subband samples and the real-valued input audio samples each comprise arrays of 32 samples.

10. A method for generating for generating complex-valued output samples, the method comprising:

storing real-valued input audio subband samples in a memory;

generating the complex-valued output audio subband samples in response to the real-valued input audio subband samples, a windowed sample vector, and prototype filter coefficients using a complex-valued low delay analysis filter bank; and

storing the complex-valued output audio subband samples in the memory,

wherein the complex-valued low delay analysis filter bank;

shifts a first subset of the real-valued input audio subband samples in the memory;

multiplies the first subset of the real-valued input audio subband samples by the prototype filter coefficients to generate windowed samples;

combines the windowed samples to generate the windowed sample vector;

multiplies the windowed sample vector by a complex-valued exponential modulation matrix to generate complex-valued output audio subband samples; and

stores the complex-valued output audio subband samples in the memory.

11. The method of claim 10 wherein the prototype filter coefficients represent a low delay asymmetric low-pass prototype filter.

12. The method of claim 10 wherein the real-valued input audio subband samples are shifted by 32 positions in the memory.

13. The method of claim 10 wherein the prototype filter coefficients comprise an array of 640 elements and the windowed samples comprise an array of 320 elements.

14. The method of claim 10 wherein the complex-valued low delay analysis filter bank is a QMF filter bank.

15. The method of claim 10 wherein the real-valued input audio subband samples are time domain audio samples.

16. The method of claim 10 wherein the complex-valued low delay analysis filter bank is part of a high frequency reconstruction (HFR) module that reconstructs high frequency components of the audio signal with low processing delay.

17. The method of claim 10 further comprising a phase shifter that shifts the phase of the complex-valued output audio subband samples.

18. The method of claim 10 wherein the complex-valued output audio subband samples and the real-valued input audio samples each comprise arrays of 32 samples.