Digital filterbank for spectral envelope adjustment
AssigneeDolby International AB
InventorPer Ekstrand
Life of this patent
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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.
Claims7
A signal processing device for filtering and performing high frequency reconstruction of an audio signal, the signal processing device comprising:
- an analysis filter bank that receives real valued time domain input audio samples and generates complex-valued subband samples;
- a unit that generates modified complex-valued subband samples by modifying gains of the complex valued subband samples according to current spectral envelope adjuster settings;
- and
- a synthesis filter bank that receives the modified complex-valued subband samples and generates time domain output audio samples,
- wherein the synthesis filter bank comprises M synthesis filters fk(n) that are complex exponential modulated versions of a prototype filter p0(n) according to:
- f
- k
-
- (
- n
- )
- =
- p
- 0
-
- (
- n
- )
-
- exp
-
- {
- i
-
-
- π
- M
-
- (
- k
- +
- 1
- 2
- )
-
- (
- n
- -
- D
- 2
- )
- }
- ,
- 0
- ≤
- n
- <
- N
- ,
- 0
- ≤
- k
- <
- M
- wherein the analysis filter bank comprises K analysis filters, with K being different from M, that are complex exponential modulated versions of a prototype filter obtained from the prototype filter p0(n) by resampling the prototype filter p0(n) through either decimation or interpolation, wherein the prototype filter p0(n) has a length N, and wherein the analysis filter bank and synthesis filter bank have a system delay of D samples that represents a latency of a signal passing through the analysis filter followed by the synthesis filter, and D is smaller than the prototype filter length N.
A method performed by an signal processing device for filtering and performing high frequency reconstruction of an audio signal, the method comprising:
- receiving real-valued time domain input audio samples;
- filtering the real-valued time domain input audio samples with an analysis filter bank to generate complex valued subband samples;
- generating modified complex valued subband samples through a high frequency reconstruction process by modifying gains of the complex valued subband samples according to current spectral envelope adjuster settings;
- filtering the modified complex valued subband samples with a synthesis filter bank to generate time domain output audio samples,
- wherein the synthesis filter bank comprises M synthesis filters fk(n) that are complex exponential modulated versions of a prototype filter p0(n) according to:
- f
- k
-
- (
- n
- )
- =
- p
- 0
-
- (
- n
- )
-
- exp
-
- {
- i
-
-
- π
- M
-
- (
- k
- +
- 1
- 2
- )
-
- (
- n
- -
- D
- 2
- )
- }
- ,
- 0
- ≤
- n
- <
- N
- ,
- 0
- ≤
- k
- <
- M
- wherein the analysis filter bank comprises K analysis filters, with K being different from M, that are complex exponential modulated versions of a prototype filter obtained from the prototype filter p0(n) by resampling the prototype filter p0(n) through either decimation or interpolation, where the prototype filter p0(n) has a length N, and the analysis filter bank and synthesis filter bank have a system delay of D samples that represents a latency of a signal passing through the analysis filter followed by the synthesis filter, and D is smaller than the prototype filter length N.
Dependencies are inferred from the claim text.