Digital filterbank for spectral envelope adjustment

• US 10,460,742 B2
• Filed: 01/22/2018
• Issued: 10/29/2019
• 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): Vo, Huyen X
• Application Number:
  US15/876,613
• Publication Number:
  US 20180146288A1
• Time in Patent Office: 645 Days
• Field of Search: 704500-504, 704200-229, 711147
• 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

An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

Claims

1. An apparatus for processing an audio signal, the apparatus comprising:

an input interface for receiving real-valued time-domain samples;

a digital filterbank including an analysis part and a synthesis part, wherein the analysis part converts the real-valued time-domain samples to complex-valued subband samples, and the synthesis part converts the complex-valued subband samples to time-domain output samples;

a first phase shifter for shifting a phase of the complex-valued subband samples by an amount;

a spectral envelope adjuster for modifying at least a portion of a spectral envelope of the audio signal by applying gains to the complex-valued subband samples;

a second phase shifter for unshifting a phase of the complex-valued subband samples by the amount; and

an output interface for outputting the time-domain output samples,

wherein the analysis part includes M_a=32 analysis filters formed by complex-exponential modulation of a prototype filter having a length of N=640, and the analysis part further includes a decimator for maximally decimating the real-valued time-domain input samples,

wherein the synthesis part includes M_s=64 synthesis filters formed by complex-exponential modulation of the prototype filter, and the synthesis part further includes an interpolator for interpolating the complex-valued subband samples,

wherein the amount of shifting and unshifting is chosen to reduce a complexity of the digital filterbank,

wherein the digital filterbank has a system delay D that represents a latency of a signal passing through the analysis part followed by the synthesis part, and D is smaller than the prototype filter length N, and

wherein the apparatus is implemented with one or more hardware elements.

2. The signal processing device of claim 1 wherein the prototype filter is a symmetric low pass prototype filter or an asymmetric low pass prototype filter.

3. The apparatus of claim 1 wherein the analysis filters and the synthesis filters are Quadrature mirror filters (QMF).

4. The apparatus of claim 1 wherein an order of the prototype filter equals the system delay D of the digital filterbank.

5. The apparatus of claim 1 further comprising a high frequency reconstructor or parametric stereo processor for modifying the complex-valued subband samples.

6. The apparatus of claim 1 wherein the prototype filter comprises coefficients of Table 1 or coefficients derivable from the coefficients of Table 1 by any of rounding, truncating, scaling, subsampling or oversampling.

7. The apparatus of claim 1 wherein the one or more hardware elements comprise a digital signal processor, a microprocessor, or a memory.

8. The apparatus of claim 1 wherein the first phase shifter and the second phase shifter are combined into a consolidated unit.

9. A method for processing an audio signal, the method comprising:

receiving real-valued time-domain samples;

filtering the real-valued time-domain samples with a digital filterbank including an analysis part and a synthesis part, wherein the analysis part converts the real-valued time-domain samples to complex-valued subband samples, and the synthesis part converts the complex-valued subband samples to time-domain output samples;

shifting a phase of the complex-valued subband samples by an amount;

modifying at least a portion of a spectral envelope of the audio signal by applying gains to the complex-valued subband samples;

unshifting a phase of the complex-valued subband samples by the amount; and

outputting the time-domain output samples,

wherein the analysis part includes M_a=32 analysis filters formed by complex-exponential modulation of a prototype filter having a length of N=640, and the analysis part further includes a decimator for maximally decimating the real-valued time-domain input samples,

wherein the synthesis part includes M_s=64 synthesis filters formed by complex-exponential modulation of the prototype filter, and the synthesis part further includes an interpolator for interpolating the complex-valued subband samples, and

wherein the amount of shifting and unshifting is chosen to reduce a complexity of the digital filterbank,

wherein the digital filterbank has a system delay D that represents a latency of a signal passing through the analysis part followed by the synthesis part, and D is smaller than the prototype filter length N, and

wherein the method is performed with one or more hardware elements.

10. A non-transitory computer readable medium containing instructions that when executed by a processor perform the method of claim 9.