U.S. patent application number 13/397597 was filed with the patent office on 2012-06-07 for system and method for processing an audio signal.
Invention is credited to Ludger Solbach, Lloyd Watts.
Application Number | 20120140951 13/397597 |
Document ID | / |
Family ID | 38750618 |
Filed Date | 2012-06-07 |
United States Patent
Application |
20120140951 |
Kind Code |
A1 |
Solbach; Ludger ; et
al. |
June 7, 2012 |
System and Method for Processing an Audio Signal
Abstract
Systems and methods for audio signal processing are provided. In
exemplary embodiments, a filter cascade of complex-valued filters
are used to decompose an input audio signal into a plurality of
frequency components or sub-band signals. These sub-band signals
may be processed for phase alignment, amplitude compensation, and
time delay prior to summation of real portions of the sub-band
signals to generate a reconstructed audio signal.
Inventors: |
Solbach; Ludger; (Mountain
View, CA) ; Watts; Lloyd; (Mountain View,
CA) |
Family ID: |
38750618 |
Appl. No.: |
13/397597 |
Filed: |
February 15, 2012 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
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11441675 |
May 25, 2006 |
8150065 |
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13397597 |
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Current U.S.
Class: |
381/94.3 |
Current CPC
Class: |
H04R 2430/03 20130101;
G10L 19/0204 20130101; H04R 25/505 20130101 |
Class at
Publication: |
381/94.3 |
International
Class: |
H04B 15/00 20060101
H04B015/00 |
Claims
1. A method for processing audio signals, comprising: filtering an
input signal with a complex-valued filter of a filter cascade to
produce a first filtered signal; subtracting the first filtered
signal from the input signal to derive a first sub-band signal;
filtering the first filtered signal with a next complex-valued
filter of the filter cascade to produce a next filtered signal; and
subtracting the next filtered signal from the first filtered signal
to derive a next sub-band signal.
2. The method of claim 1 wherein the complex-valued filter and the
next complex-valued filter are single pole, complex-valued
filters.
3. The method of claim 1 further comprising performing phase
alignment on one or more of the sub-band signals.
4. The method of claim 3 further comprising disposing of an
imaginary portion of the one or more phase aligned sub-band
signals.
5. The method of claim 1 further comprising performing amplitude
compensation on one or more of the sub-band signals.
6. The method of claim 1 further comprising performing a time delay
on one or more of the sub-band signals for cross-sub-band
alignment.
7. The method of claim 6 further comprising summing the delayed one
or more sub-band signals to generate a reconstructed audio
signal.
8. The method of claim 1 further comprising pre-processing the
input signal prior to filtering the input signal with the
complex-valued filter of the filter cascade.
9. The method of claim 1 further comprising modifying one or more
of the sub-band signals based on an analysis path from the filter
cascade.
10. The method of claim 1 wherein the sub-band signals are
frequency components of the input signal.
11. A system for processing an audio signal, comprising: an audio
processing engine comprising a filter cascade of complex-valued
filters configured to derive a plurality of sub-band signals from
an input signal, the set of complex-valued filters arranged in the
filter cascade whereby an output of each complex-valued filter is
passed to a next complex-valued filter in the filter cascade.
12. The system of claim 11 wherein the complex-valued filters are
single pole, complex-valued filters.
13. The system of claim 11 wherein the audio processing engine
further comprises a reconstruction module configured to perform
phase alignment on one or more of the sub-band signals.
14. The system of claim 11 wherein the audio processing engine
further comprises a reconstruction module configured to perform
amplitude compensation on one or more of the sub-band signals.
15. The system of claim 11 wherein the audio processing engine
further comprises a reconstruction module configured to perform a
time delay on one or more of the sub-band signals.
16. The system of claim 11 wherein the audio processing engine
further comprises a modification module configured to modify one or
more of the sub-band signals based on an analysis path from the
filter cascade.
17. The system of claim 11 further comprising a conditioning module
configured to pre-process the input signal prior to filtering the
input signal with the filter cascade.
18. A machine-readable medium having embodied thereon a program,
the program being executable by a machine to perform a method for
processing an audio signal, the method comprising: filtering an
input signal with a complex-valued filter of a filter cascade to
produce a first filtered signal; subtracting the first filtered
signal from the input signal to derive a first sub-band signal;
filtering the first filtered signal with a next complex-valued
filter of the filter cascade to produce a next filtered signal; and
subtracting the next filtered signal from the first filtered signal
to derive a next sub-band signal.
19. The machine-readable medium of claim 18 wherein the
complex-valued filter and the next complex-valued filter are single
pole, complex-valued filters.
20. The machine-readable medium of claim 18 wherein the method
further comprises performing phase alignment on one or more of the
sub-band signals.
21. The machine-readable medium of claim 18 wherein the method
further comprises performing amplitude compensation on one or more
of the sub-band signals.
22. The machine-readable medium of claim 18 wherein the method
further comprises performing a time delay on one or more the
sub-band signals.
23. The machine-readable medium of claim 18 wherein the method
further comprises pre-processing the input signal prior to
filtering the input signal with the filter cascade.
Description
CROSS-REFERENCE TO RELATED APPLICATIONS
[0001] The present application is a continuation of U.S.
application Ser. No. 11/441,675 filed on May 25, 2006 entitled
"System and Method for Processing an Audio Signal." The present
application is also related to U.S. patent application Ser. No.
10/613,224 entitled "Filter Set for Frequency Analysis" filed Jul.
3, 2003; U.S. patent application Ser. No. 10/613,224 is a
continuation of U.S. U.S. patent application Ser. No. 10/074,991,
entitled "Filter Set for Frequency Analysis" filed Feb. 13, 2002,
which is a continuation of U.S. patent application Ser. No.
09/534,682 entitled "Efficient Computation of Log-Frequency-Scale
Digital Filter Cascade" filed Mar. 24, 2000; the disclosures of
which are incorporated herein by reference.
BACKGROUND OF THE INVENTION
[0002] 1. Field of the Invention
[0003] Embodiments of the present invention are related to audio
processing, and more particularly to the analysis of audio
signals.
[0004] 2. Related Art
[0005] There are numerous solutions for splitting an audio signal
into sub-bands and deriving frequency-dependent amplitude and phase
characteristics varying over time. Examples include windowed fast
Fourier transform/inverse fast Fourier transform (FFT/IFFT) systems
as well as parallel banks of finite impulse response (FIR) and
infinite impulse response (IIR) filter banks. These conventional
solutions, however, all suffer from deficiencies.
[0006] Disadvantageously, windowed FFT systems only provide a
single, fixed bandwidth for each frequency band. Typically, a
bandwidth which is applied from low frequency to high frequency is
chosen with a fine resolution at the bottom. For example, at 100
Hz, a filter (bank) with a 50 kHz bandwidth is desired. This means,
however, that at 8 kHz, a 50 Hz bandwidth is used where a wider
bandwidth such as 400 Hz may be more appropriate. Therefore,
flexibility to match human perception cannot be provided by these
systems.
[0007] Another disadvantage of windowed FFT systems is that
inadequate fine frequency resolution of sparsely sampled windowed
FFT systems at high frequencies can result in objectionable
artifacts (e.g., "musical noise") if modifications are applied,
(e.g., for noise suppression). The number of artifacts can be
reduced to some extent by dramatically reducing the number of
samples of overlap between the windowed frames size "FFT hop size"
(i.e., increasing oversampling). Unfortunately, computational costs
of FFT systems increase as oversampling increases. Similarly, the
FIR subclass of filter banks are also computationally expensive due
to the convolution of the sampled impulse responses in each
sub-band which can result in high latency. For example, a system
with a window of 256 samples will require 256 multiplies and a
latency of 128 samples, if the window is symmetric.
[0008] The IIR subclass is computationally less expensive due to
its recursive nature, but implementations employing only
real-valued filter coefficients present difficulties in achieving
near-perfect reconstruction, especially if the sub-band signals are
modified. Further, phase and amplitude compensation as well as
time-alignment for each sub-band is required in order to produce a
flat frequency response at the output. The phase compensation is
difficult to perform with real-valued signals, since they are
missing the quadrature component for straight-forward computation
of amplitude and phase with fine time-resolution. The most common
way to determine amplitude and frequency is to apply a Hilbert
transform on each stage output. But an extra computation step is
required for calculating the Hilbert transform in real-valued
filter banks, and is computationally expensive.
[0009] Therefore, there is a need for systems and methods for
analyzing and reconstructing an audio signal that is
computationally less expensive than existing systems, while
providing low end-to-end latency, and the necessary degrees of
freedom for time-frequency resolution.
SUMMARY OF THE INVENTION
[0010] Embodiments of the present invention provide systems and
methods for audio signal processing. In exemplary embodiments, a
filter cascade of complex-valued filters is used to decompose an
input audio signal into a plurality of sub-band signals. In one
embodiment, an input signal is filtered with a complex-valued
filter of the filter cascade to produce a first filtered signal.
The first filtered signal is subtracted from the input signal to
derive a first sub-band signal. Next, the first filtered signal is
processed by a next complex-valued filter of the filter cascade to
produce a next filtered signal. The processes repeat until the last
complex-valued filters in the cascade has been utilized. In some
embodiments, the complex-valued filters are single pole,
complex-valued filters.
[0011] Once the input signal is decomposed, the sub-band signals
may be processed by a reconstruction module. The reconstruction
module is configured to perform a phase alignment on one or more of
the sub-band signals. The reconstruction module may also be
configured to perform amplitude compensation on one or more of the
sub-band signals. Further, a time delay may be performed on one or
more of the sub-band signals by the reconstruction module. Real
portions of the compensated and/or time delayed sub-band signals
are summed to generate a reconstructed audio signal.
BRIEF DESCRIPTION OF THE DRAWINGS
[0012] FIG. 1 is an exemplary block diagram of a system employing
embodiments of the present invention;
[0013] FIG. 2 is an exemplary block diagram of the analysis filter
bank module in an exemplary embodiment of the present
invention;
[0014] FIG. 3 illustrates a filter of the analysis filter bank
module, according to one embodiment;
[0015] FIG. 4 illustrates for every six (6) sub-bands a log display
of magnitude and phase of the sub-band transfer function;
[0016] FIG. 5 illustrates for every six (6) stages a log display of
magnitude and phase of the accumulated filter transfer
functions;
[0017] FIG. 6 illustrates the operation of the exemplary
reconstruction module;
[0018] FIG. 7 illustrates a graphical representation of an
exemplary reconstruction of the audio signal; and
[0019] FIG. 8 is a flowchart of an exemplary method for
reconstructing an audio signal.
DETAILED DESCRIPTION OF EXEMPLARY EMBODIMENTS
[0020] Embodiments of the present invention provide systems and
methods for near perfect reconstruction of an audio signal. The
exemplary system utilizes a recursive filter bank to generate
quadrature outputs. In exemplary embodiments, the filter bank
comprises a plurality of complex-valued filters. In further
embodiments, the filter bank comprises a plurality of single pole,
complex-valued filters.
[0021] Referring to FIG. 1, an exemplary system 100 in which
embodiments of the present invention may be practiced is shown. The
system 100 may be any device, such as, but not limited to, a
cellular phone, hearing aid, speakerphone, telephone, computer, or
any other device capable of processing audio signals. The system
100 may also represent an audio path of any of these devices.
[0022] The system 100 comprises an audio processing engine 102, an
audio source 104, a conditioning module 106, and an audio sink 108.
Further components not related to reconstruction of the audio
signal may be provided in the system 100. Additionally, while the
system 100 describes a logical progression of data from each
component of FIG. 1 to the next, alternative embodiments may
comprise the various components of the system 100 coupled via one
or more buses or other elements.
[0023] The exemplary audio processing engine 102 processes the
input (audio) signals inputted via the audio source 104. In one
embodiment, the audio processing engine 102 comprises software
stored on a device which is operated upon by a general processor.
The audio processing engine 102, in various embodiments, comprises
an analysis filter bank module 110, a modification module 112, and
a reconstruction module 114. It should be noted that more, less, or
functionally equivalent modules may be provided in the audio
processing engine 102. For example, one or more modules 110-114 may
be combined into few modules and still provide the same
functionality.
[0024] The audio source 104 comprises any device which receives
input (audio) signals. In some embodiments, the audio source 104 is
configured to receive analog audio signals. In one example, the
audio source 104 is a microphone coupled to an analog-to-digital
(A/D) converter. The microphone is configured to receive analog
audio signals while the A/D converter samples the analog audio
signals to convert the analog audio signals into digital audio
signals suitable for further processing. In other examples, the
audio source 104 is configured to receive analog audio signals
while the conditioning module 106 comprises the A/D converter. In
alternative embodiments, the audio source 104 is configured to
receive digital audio signals. For example, the audio source 104 is
a disk device capable of reading audio signal data stored on a hard
disk or other forms of media. Further embodiments may utilize other
forms of audio signal sensing/capturing devices.
[0025] The conditioning module 106 pre-processes the input signal
(i.e., any processing that does not require decomposition of the
input signal). In one embodiment, the conditioning module 106
comprises an auto-gain control. The conditioning module 106 may
also perform error correction and noise filtering. The conditioning
module 106 may comprise other components and functions for
pre-processing the audio signal.
[0026] The analysis filter bank module 110 decomposes the received
input signal into a plurality of sub-band signals. In some
embodiments, the outputs from the analysis filter bank module 110
can be used directly (e.g., for a visual display). The analysis
filter bank module 110 will be discussed in more detail in
connection with FIG. 2. In exemplary embodiments, each sub-band
signal represents a frequency component.
[0027] The exemplary modification module 112 receives each of the
sub-band signals over respective analysis paths from the analysis
filter bank module 110. The modification module 112 can
modify/adjust the sub-band signals based on the respective analysis
paths. In one example, the modification module 112 filters noise
from sub-band signals received over specific analysis paths. In
another example, a sub-band signal received from specific analysis
paths may be attenuated, suppressed, or passed through a further
filter to eliminate objectionable portions of the sub-band
signal.
[0028] The reconstruction module 114 reconstructs the modified
sub-band signals into a reconstructed audio signal for output. In
exemplary embodiments, the reconstruction module 114 performs phase
alignment on the complex sub-band signals, performs amplitude
compensation, cancels the complex portion, and delays remaining
real portions of the sub-band signals during reconstruction in
order to improve resolution of the reconstructed audio signal. The
reconstruction module 114 will be discussed in more details in
connection with FIG. 6.
[0029] The audio sink 108 comprises any device for outputting the
reconstructed audio signal. In some embodiments, the audio sink 108
outputs an analog reconstructed audio signal. For example, the
audio sink 108 may comprise a digital-to-analog (D/A) converter and
a speaker. In this example, the D/A converter is configured to
receive and convert the reconstructed audio signal from the audio
processing engine 102 into the analog reconstructed audio signal.
The speaker can then receive and output the analog reconstructed
audio signal. The audio sink 108 can comprise any analog output
device including, but not limited to, headphones, ear buds, or a
hearing aid. Alternately, the audio sink 108 comprises the D/A
converter and an audio output port configured to be coupled to
external audio devices (e.g., speakers, headphones, ear buds,
hearing aid).
[0030] In alternative embodiments, the audio sink 108 outputs a
digital reconstructed audio signal. In another example, the audio
sink 108 is a disk device, wherein the reconstructed audio signal
may be stored onto a hard disk or other medium. In alternate
embodiments, the audio sink 108 is optional and the audio
processing engine 102 produces the reconstructed audio signal for
further processing (not depicted in FIG. 1).
[0031] Referring now to FIG. 2, the exemplary analysis filter bank
module 110 is shown in more detail. In exemplary embodiments, the
analysis filter bank module 110 receives an input signal 202, and
processes the input signal 202 through a series of filters 204 to
produce a plurality of sub-band signals or components (e.g.,
P1-P6). Any number of filters 204 may comprise the analysis filter
bank module 110. In exemplary embodiments, the filters 204 are
complex valued filters. In further embodiments, the filters 204 are
first order filters (e.g., single pole, complex valued). The
filters 204 are further discussed in FIG. 3.
[0032] In exemplary embodiments, the filters 204 are organized into
a filter cascade whereby an output of one filter 204 becomes an
input in a next filter 204 in the cascade. Thus, the input signal
202 is fed to a first filter 204a. An output signal, P1, of the
first filter 204a is subtracted from the input signal 202 by a
first computation node 206a to produce an output D1. The output D1
represents the difference signal between the signal going into the
first filter 204a and the signal after the first filter 204a.
[0033] In alternative embodiments, benefits of the filter cascade
may be realized without the use of the computation node 206 to
determine sub-band signals. That is, the output of each filter 204
may be used directly to represent energy of the signal at the
output or be displayed, for example.
[0034] Because of the cascade structure of the analysis filter bank
module 110, the output signal, P1, is now an input signal into a
next filter 204b in the cascade. Similar to the process associated
with the first filter 204a, an output of the next filter 204b
(i.e., P2) is subtracted from the input signal P1 by a next
computation node 206b to obtain a next frequency band or channel
(i.e., output D2). This next frequency channel emphasizes
frequencies between cutoff frequencies of the present filter 204b
and the previous filter 204a. This process continues through the
remainder of the filters 204 of the cascade.
[0035] In one embodiment, sets of filters in the cascade are
separated into octaves. Filter parameters and coefficients may then
be shared among corresponding filters (in a similar position) in
different octaves. This process is described in detail in U.S.
patent application Ser. No. 09/534,682.
[0036] In some embodiments, the filters 204 are single pole,
complex-valued filters. For example, the filters 204 may comprise
first order digital or analog filters that operate with complex
values. Collectively, the outputs of the filters 204 represent the
sub-band components of the audio signal. Because of the computation
node 206, each output represents a sub-band, and a sum of all
outputs represents the entire input signal 202. Since the cascading
filters 204 are first order, the computational expense may be much
less than if the cascading filters 204 were second order or more.
Further, each sub-band extracted from the audio signal can be
easily modified by altering the first order filters 204. In other
embodiments, the filters 204 are complex-valued filters and not
necessarily single pole.
[0037] In further embodiments, the modification module 112 (FIG. 1)
can process the outputs of the computation node 206 as necessary.
For example, the modification module 112 may half wave rectify the
filtered sub-bands. Further, the gain of the outputs can be
adjusted to compress or expand a dynamic range. In some
embodiments, the output of any filter 204 may be downsampled before
being processed by another chain/cascade of filters 204.
[0038] In exemplary embodiments, the filters 204 are infinite
impulse response (IIR) filters with cutoff frequencies designed to
produce a desired channel resolution. The filters 204 may perform
successive Hilbert transformations with a variety of coefficients
upon the complex audio signal in order to suppress or output
signals within specific sub-bands.
[0039] FIG. 3 is a block diagram illustrating this signal flow in
one exemplary embodiment of the present invention. The output of
the filter 204, y.sub.real[n] and y.sub.imag[n] is passed as an
input x.sub.real[n+1] and x.sub.imag[n+1], respectively, of a next
filter 204 in the cascade. The term "n" identifies the sub-band to
be extracted from the audio signal, where "n" is assumed to be an
integer. Since the IIR filter 204 is recursive, the output of the
filter can change based on previous outputs. The imaginary
components of the input signal (e.g., x.sub.imag[n]) can be summed
after, before, or during the summation of the real components of
the signal. In one embodiment, the filter 204 can be described by
the complex first order difference equation
y(k)=g*(x(k)+b*x(k-1))+a*y(k-1) where b=r_z*exp(i*theta_p) and
a=-r_p*exp(i*theta_p) and "y" is a sample index.
[0040] In the present embodiment, "g" is a gain factor. It should
be noted that the gain factor can be applied anywhere that does not
affect the pole and zero locations. In alternative embodiments, the
gain may be applied by the modification module 112 (FIG. 1) after
the audio signals have been decomposed into sub-band signals.
[0041] Referring now to FIG. 4, an example log display of magnitude
and phase for every six (6) sub-bands of an audio signal is shown.
The magnitude and phase information is based on outputs from the
analysis filter bank module 110 (FIG. 1). That is, the amplitudes
shown in FIG. 4 are the outputs (i.e., output D1-D6) from the
computation node 206 (FIG. 2). In the present example, the analysis
filter bank module 110 is operating at a 16 kHz sampling rate with
235 sub-bands for a frequency range from 80 Hz to 8 kHz. End-to-end
latency of this analysis filter bank module 110 is 17.3 ms.
[0042] In some embodiments, it is desirable to have a wide
frequency response at high frequencies and a narrow frequency
response at low frequencies. Because embodiments of the present
invention are adaptable to many audio sources 104 (FIG. 1),
different bandwidths at different frequencies may be used. Thus,
fast responses with wide bandwidths at high frequencies and slow
responses with a narrow, short bandwidth at low frequencies may be
obtained. This results in responses that are much more adapted to
the human ear with relatively low latency (e.g., 12 ms).
[0043] Referring now to FIG. 5, an example of magnitude and phase
per stage of an analytic cochlea design is shown. The amplitude
shown in FIG. 5 is the outputs of filters 204 of FIG. 2 (e.g.,
P1-P6).
[0044] FIG. 6 illustrates operation of the reconstruction module
114 according to one embodiment of the present invention. In
exemplary embodiments, the phase of each sub-band signal is
aligned, amplitude compensation is performed, the complex portion
of each sub-band signal is removed, and then time is aligned by
delaying each sub-band signal as necessary to achieve a flat
reconstruction spectrum and reduce impulse response dispersion.
[0045] Because the filters use complex signals (e.g., real and
imaginary parts), phase may be derived for any sample.
Additionally, amplitude may also be calculated by A= {square root
over (((y.sub.real[n]).sup.2+(y.sub.imag[n]).sup.2))}{square root
over (((y.sub.real[n]).sup.2+(y.sub.imag[n]).sup.2))}. Thus, the
reconstruction of the audio signal is mathematically made easier.
As a result of this approach, the amplitude and phase for any
sample is readily available for further processing (i.e., to the
modification module 112 (FIG. 1).
[0046] Since the impulse responses of the sub-band signals may have
varying group delays, merely summing up the outputs of the analysis
filter bank module 110 (FIG. 1) may not provide an accurate
reconstruction of the audio signal. Consequently, the output of a
sub-band can be delayed by the sub-band's impulse response peak
time so that all sub-band filters have their impulse response
envelope maximum at a same instance in time.
[0047] In an embodiment where the impulse response waveform maximum
is later in time than the desired group delay, the filter output is
multiplied with a complex constant such that the real part of the
impulse response has a local maximum at the desired group
delay.
[0048] As shown, sub-band signals 602 (e.g., S.sub.0, S.sub.n, and
S.sub.m) are received by the reconstruction module 114 from the
modification module 112 (FIG. 1). Coefficients 604 (e.g., a.sub.0,
a.sub.n, and a.sub.m) are then applied to the sub-band signal. The
coefficient comprises a fixed, complex factor (i.e., comprising a
real and imaginary portion). Alternately, the coefficients 604 can
be applied to the sub-band signal within the analysis filter bank
module 110. The application of the coefficient to each sub-band
signal aligns the phases of the sub-band signal and compensates
each amplitude. In exemplary embodiments, the coefficients are
predetermined. After the application of the coefficient, the
imaginary portion is discarded by a real value module 606 (i.e.,
Re{ }).
[0049] Each real portion of the sub-band signal is then delayed by
a delay Z.sup.-1 608. This delay allows for cross sub-band
alignment. In one embodiment, the delay Z.sup.-1 608 provides a one
tap delay. After the delay, the respective sub-band signal is
summed in a summation node 610, resulting in a value. The partially
reconstructed signal is then carried into a next summation node 610
and applied to a next delayed sub-band signal. The process
continues until all sub-band signals are summed resulting in a
reconstructed audio signal. The reconstructed audio signal is then
suitable for the audio sink 108 (FIG. 1). Although the delays
Z.sup.-1 608 are depicted after sub-band signals are summed, the
order of operations of the reconstruction module 114 can be
interchangeable.
[0050] FIG. 7 illustrates a reconstruction graph based on the
example of FIG. 4 and FIG. 5. The reconstruction (i.e.,
reconstructed audio signal) is obtained by combining the outputs of
each filter 204 (FIG. 2) after phase alignment, amplitude
compensation, and delay for cross sub-band alignment by the
reconstruction module 114 (FIG. 1). As a result, the reconstruction
graph is relatively flat.
[0051] Referring now to FIG. 8, a flowchart 800 of an exemplary
method for audio signal processing is provided. In step 802, an
audio signal is decomposed into sub-band signals. In exemplary
embodiments, the audio signal is processed by the analysis filter
bank module 110 (FIG. 1). The processing comprises filtering the
audio signal through a cascade of filters 204 (FIG. 2), the output
of each filter 204 resulting in a sub-band signal at the respective
outputs. In one embodiment, the filters 204 are complex-valued
filters. In a further embodiment, the filters 204 are single pole,
complex-valued filters.
[0052] After sub-band decomposition, the sub-band signals are
processed through the modification module 112 (FIG. 1) in step 804.
In exemplary embodiments, the modification module 112 (FIG. 1)
adjusts the gain of the outputs to compress or expand a dynamic
range. In some embodiments, the modification module 112 may
suppress objectionable sub-band signals.
[0053] A reconstruction module 114 (FIG. 1) then performs phase and
amplitude compensation on each sub-band signal in step 806. In one
embodiment, the phase and amplitude compensation occurs by applying
a complex coefficient to the sub-band signal. The imaginary portion
of the compensated sub-band signal is then discarded in step 808.
In other embodiments, the imaginary portion of the compensated
sub-band signal is retained.
[0054] Using the real portion of the compensated sub-band signal,
the sub-band signal is delayed for cross-sub-band alignment in step
810. In one embodiment, the delay is obtained by utilizing a delay
line in the reconstruction module 114.
[0055] In step 812, the delayed sub-band signals are summed to
obtain a reconstructed signal. In exemplary embodiments, each
sub-band signal/segment represents a frequency.
[0056] Embodiments of the present invention have been described
above with reference to exemplary embodiments. It will be apparent
to those skilled in the art that various modifications may be made
and other embodiments can be used without departing from the
broader scope of the invention. Therefore, these and other
variations upon the exemplary embodiments are intended to be
covered by the present invention.
* * * * *