U.S. patent application number 11/041275 was filed with the patent office on 2005-06-30 for digital signal sub-band separating/combining apparatus achieving band-separation and band-combining filtering processing with reduced amount of group delay.
This patent application is currently assigned to MATSUSHITA ELECTRIC INDUSTRIAL CO., LTD.. Invention is credited to Banba, Yutaka, Ito, Masayuki, Taniguchi, Shohei.
Application Number | 20050143973 11/041275 |
Document ID | / |
Family ID | 18301885 |
Filed Date | 2005-06-30 |
United States Patent
Application |
20050143973 |
Kind Code |
A1 |
Taniguchi, Shohei ; et
al. |
June 30, 2005 |
Digital signal sub-band separating/combining apparatus achieving
band-separation and band-combining filtering processing with
reduced amount of group delay
Abstract
An apparatus having a band-separating filter bank for separating
a digital signal into a plurality of sub-band signals, to be
processed or transmitted, and a band-combining filter bank for
subsequently combining the resultant sub-band signals into a single
digital signal, wherein each of the band-separating filter bank and
band-combining filter bank incorporates a FIR low pass filter
having an asymmetric impulse response, as the prototype filter of
the filter bank. A significant reduction can thereby be achieved in
the amount of overall group delay that results from the processing
performed by these filter banks.
Inventors: |
Taniguchi, Shohei;
(Yokohama, JP) ; Ito, Masayuki; (Yokohama, JP)
; Banba, Yutaka; (Yokohama, JP) |
Correspondence
Address: |
MCDERMOTT WILL & EMERY LLP
600 13TH STREET, N.W.
WASHINGTON
DC
20005-3096
US
|
Assignee: |
MATSUSHITA ELECTRIC INDUSTRIAL CO.,
LTD.
Osaka
JP
|
Family ID: |
18301885 |
Appl. No.: |
11/041275 |
Filed: |
January 25, 2005 |
Related U.S. Patent Documents
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Application
Number |
Filing Date |
Patent Number |
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11041275 |
Jan 25, 2005 |
|
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09662823 |
Sep 15, 2000 |
|
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6856653 |
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Current U.S.
Class: |
704/200.1 |
Current CPC
Class: |
H03H 2021/0041 20130101;
H03H 17/0266 20130101 |
Class at
Publication: |
704/200.1 |
International
Class: |
H04B 001/66; G10L
019/00 |
Foreign Application Data
Date |
Code |
Application Number |
Nov 26, 1999 |
JP |
11-336698 |
Claims
1. A digital signal low-delay sub-band separating apparatus formed
of a plurality of filters corresponding to respective frequency
bands, for separating an input digital signal into a corresponding
plurality of sub-band signals, and decimation means for applying
digital signal sample decimation to each of said sub-band signals;
wherein said apparatus comprises, as a prototype filter of said
plurality of filters, a FIR (finite impulse response) low-pass
filter which is configured to have a response characteristic that
is asymmetric with respect to the origin.
2. A digital signal low-delay sub-band combining apparatus formed
of a plurality of filters corresponding to respective frequency
bands, for filtering corresponding ones of a plurality of sub-band
signals supplied thereto, interpolation means for applying digital
signal sample interpolation to each of said sub-band signals
subsequent to said filtering, and means for additively combining
respective output sub-band signals from said interpolation means to
form a single digital signal; wherein said apparatus comprises, as
a prototype filter of said plurality of filters, a FIR (finite
impulse response) low-pass filter which is configured to have a
response characteristic that is asymmetric with respect to the
origin.
3. A digital signal encoder apparatus comprising sub-band
separating means for converting an input digital signal to a
plurality of sub-band signals and encoding means for respectively
encoding said sub-band signals and combining resultant encoded data
into a data stream to be transmitted or processed; wherein said
sub-band separating means comprises a sub-band separating apparatus
as claimed in claim 1.
4. A digital signal decoder apparatus for operating on an input
data stream formed of combined encoded data of a plurality of
sub-band signals, comprising decoding means for recovering said
plurality of sub-band signals from said data stream and sub-band
combining means for combining said recovered sub-band signals into
a single output digital signal; wherein said sub-band combining
means comprises a sub-band combining apparatus as claimed in claim
2.
5. A digital signal encoder apparatus according to claim 3, wherein
said encoding means performs encoding processing by applying a
high-speed algorithm which utilizes the periodicity of a cosine
function.
6. A digital signal decoder apparatus according to claim 4, wherein
said decoding means performs decoding processing by applying a
high-speed algorithm which utilizes the periodicity of a cosine
function.
7. A digital signal encoder apparatus according to claim 3 wherein
said input digital signal is a PCM (pulse code modulation) digital
audio signal and wherein said encoding comprises compression
encoding of said PCM digital audio signal.
8. A digital signal low-delay decoder apparatus according to claim
4, wherein said plurality of sub-band signals express a single PCM
(pulse code modulation) digital audio signal and have been encoded
by compression encoding processing, and wherein said decoding
processing performed by said decoder apparatus comprises expansion
decoding to recover said plurality of sub-band signals from said
data stream and combining said sub-band signals into a single
digital PCM audio signal.
9. A transmitter apparatus for a digital wireless microphone
system, having analog-to-digital converter means for converting an
analog audio signal from a microphone to a digital audio signal and
encoder means for converting said digital audio signal to an
encoded compressed data stream, wherein said encoder means
comprises a digital signal encoding apparatus as claimed in claim
3.
10. A receiver apparatus for a digital wireless microphone system,
for receiving a data stream formed by compression encoding of a
plurality of sub-band signals derived from an original digital
audio signal, having decoder means for recovering said original
digital audio signal from said data stream, wherein said decoder
means comprises a digital signal decoder apparatus as claimed in
claim 4.
11. (canceled)
12. A digital signal low-delay sub-band separating apparatus formed
of a plurality of filters corresponding to respective frequency
bands, for separating an input digital signal into a corresponding
plurality of sub-band signals, and decimation means for applying
digital signal sample decimation to each of said sub-band signals;
wherein said apparatus comprises, as a base filter of said
plurality of filters, a FIR (finite impulse response) low-pass
filter having an impulse response formed by N filter taps, and
having an impulse response coefficient h[k] that satisfies a
condition: h[k].noteq..+-.h[N-1-k], where 0.ltoreq.k.ltoreq.S k
int(N/2)-1, with int(N/2) being the integer part of (N/2).
13. A digital signal low-delay sub-band combining apparatus formed
of a plurality of filters corresponding to respective frequency
bands, for filtering corresponding ones of a plurality of sub-band
signals supplied thereto, interpolation means for applying digital
signal sample interpolation to each of said sub-band signals
subsequent to said filtering, and means for additively combining
respective output sub-band signals from said interpolation means to
form a single digital signal; wherein said apparatus comprises, as
a base filter of said plurality of filters, a FIR (finite impulse
response) low-pass filter having an impulse response formed by N
filter taps, and having an impulse response coefficient h[k] that
satisfies a condition: h[k].noteq.+h[N-1-k], where
0.ltoreq.k.ltoreq.int(N/2)-1, with int(n/2) being the integer part
of (N/2).
Description
BACKGROUND OF THE INVENTION
[0001] 1. Field of the Invention
[0002] The invention relates to a sub-band separating/combining
apparatus having a band-separating filter bank for converting a
digital signal to a plurality of sub-band signals and a
band-combining filter bank which receives the sub-band signals
after processing or transmission thereof, for combining these to
recover the original digital signal or a processed version of that
signal
[0003] 2. Description of the Related Art
[0004] There are various applications in which a digital signal is
supplied to a set of filters of a band-separating filter bank
(sometimes referred to as an analyzing filter bank) to be
spectrally divided into a plurality of sub-band signals, i.e.,
respectively corresponding to different frequency bands, with the
sample rate of each of the sub-band signals then being reduced by
decimation (i.e., down-sampling). Processing or transmission of the
resultant low-bandwidth sub-band signals can then be efficiently
performed. After processing or transmission of the sub-band
signals, they are supplied to a band-combining filter bank
(sometimes referred to as a synthesizing or a reconstructing filter
bank), to be each subjected to interpolation processing (i.e.,
up-sampling), then inputted to respective ones of a set of filters
whose outputs are additively combined to recover the original
digital signal (or a processed version of that signal).
[0005] A prior art example of such a combination of a
band-separating filter bank and a band-combining filter bank is
shown in FIG. 7. Here, a band-separating filter bank 701, a
processing section 703 and a band-combining filter bank 702
successively operate on an input digital signal designated as x(n).
The processing section 703 may for example perform such operations
as data encoding/decoding, echo cancellation processing, etc.
[0006] The band-separating filter bank 701 divides the input
digital signal x(n) into a total of M channels of sub-band signals,
whose respective frequency bands will be numbered as bands 0 to
(M-1) respectively. 710.about.71n designate the respective
band-separating filters of the filter bank 701, respectively
corresponding to frequency bands 0 to (M-1), with their respective
Z-transform transfer functions (referred to in the following simply
as transfer functions) designated as G.sub.O
(z).about.G.sub.M-1(z). The output sub-band signals from these
filters 710.about.71n are supplied to respective ones of a set of
decimators 720.about.72n, with the resultant down-sampled sub-band
signals being supplied to the processing section 703. The
band-combining filter bank 702 includes a set interpolators
740.about.74n which respectively receive the processed sub-band
signals produced from the processing section 703, while
730.about.73n are band-combining filters respectively corresponding
to the frequency bands 0.about.M-1 and having respective transfer
functions K.sub.0(z).about.K.sub.M-1(z), which receive the
corresponding ones of the interpolated sub-band signals which are
produced from the interpolators 740.about.74n.
[0007] The decimation and interpolation factor is indicated as D.
That is to say, one in every D samples of a sub-band signal is
selected by the decimation processing, while (D-1) fixed sample
values (e.g., zero values) are inserted following each sample of a
processed sub-band signal, by the interpolation processing.
[0008] It will be assumed that the number M of sub-bands into which
the input digital signal is divided is identical to the
aforementioned decimation and interpolation factor D.
[0009] The output sub-band signals from the filters 730.about.73n
are additively combined in an adder 704, to obtain a digital signal
y(n) as the output signal.
[0010] This is a recovered version of the original digital signal
(possibly modified as a result of the operation of the processing
section 703). If it is assumed that the processing 703 section
performs a type of processing such as echo cancellation, which
requires the use of DFT filter banks for band separation and
combining, then the respective transfer functions G.sub.k(z) and
K.sub.k(z) of the k-th band-separating filter and k-th
band-combining filter are expressed as follows by equations (1) and
(2) respectively:
G.sub.k(z)=G.sub.0(zW.sub.M.sup.k) (1)
K.sub.k(z)=W.sub.M.sup.-kK.sub.0(zW.sub.M.sup.k) (2)
[0011] Here, W.sub.M.sup.k=exp(-j2.pi.k/M), with 0<k<M-1, and
each of G.sub.0(z) and K.sub.0(z) represents the transfer function
of the prototype filter of a DFT (Discrete Fourier Transform)
filter bank. The term "prototype filter" as used herein in relation
to a band-separating filter bank or band-combining filter bank
signifies a low-pass filter which handles the lowest frequency
band, such as filter 710 of the band-separating filter bank 701 in
FIG. 7.
[0012] If however the processing section 703 performs processing
which requires the use of cosine modulation filter banks as the
band-separating filter bank and band-combining filter bank, then
the respective transfer functions of the k-th band-separating
filter and k-th band-combining filter are obtained as follows from
equations (3) and (4) respectively:
G.sub.k(z)=a.sub.k*c.sub.kP(zW.sub.2M.sup.(k+1/2))+a.sub.k*c.sub.k*P(zW.su-
b.2M.sup.-(k+1/2)) (3)
K.sub.k(z)=a.sub.k*c.sub.kP(zW.sub.2M.sup.(k+1/2))+a.sub.k*c.sub.k*P(zW.su-
b.2M.sup.-(k+1/2)) (4)
[0013] Where W.sub.2M=exp(-j.pi./M), with 0<k<M-1,
a.sub.k=exp(j.theta..sub.k), C.sub.k=W.sub.2M.sup.(k+1/2)(N-1)/2,
.theta..sub.k=(2k+1).pi./4, N is the number of taps of the
prototype filter, the * symbol indicates the complex conjugate, and
P(z) designates the transfer function of the prototype filter of a
cosine modulation filter bank.
[0014] In the prior art, a FIR low-pass filter having a symmetric
impulse response is used as the prototype filter in such a type of
filter bank. An example of such a symmetric impulse response is
shown in FIG. 8.
[0015] However with such a prior art type of apparatus which uses a
filter in which each of the prototype filters of the sub-band
separating filter bank and sub-band combining filter bank is a FIR
(finite impulse response) low-pass filter having a symmetric
impulse response, designating the number of taps of such a
prototype filter as N, an amount of delay will be produced by the
operation of a filter bank that is equal to the total of the group
delays of (N-1) taps. In many applications, such an amount of delay
becomes a serious disadvantage. For example, if the processing
section 703 in FIG. 7 performs echo canceller processing, then it
is essential to minimize the sub-band separating and sub-band
combining filter delays, in order to achieve a suitably high speed
of control response together with stability of operation.
SUMMARY OF THE INVENTION
[0016] It is an objective of the present invention to overcome the
above problem by providing an improved sub-band separating
apparatus and sub-band combining apparatus, each apparatus having
at least one band-separating filter bank and at least one
band-combining filter bank, wherein an amount of delay which
results from filtering performed successively by said filter banks
is reduced, by comparison with prior art types of filter bank
utilized for a sub-band separating and combining apparatus.
[0017] To achieve the above objective, the invention provides a
sub-band separating apparatus and sub-band combining apparatus
wherein each of respective basic filters of a band-separating
filter bank and a band-combining filter bank is configured to have
a symmetric impulse response, to thereby achieve a lower amount of
group delay for each prototype filter and thereby reduce an overall
amount of delay which results from filtering by the filter
banks.
[0018] The invention further provides a digital signal encoder
apparatus comprising sub-band separating means for converting an
input digital signal to a plurality of sub-band signals and
encoding means for respectively encoding said sub-band signals and
combining resultant encoded data into a data stream to be
transmitted or processed, in which the sub-band separating means
consists of a low-delay sub-band separating apparatus as described
above, and similarly provides a corresponding decoder apparatus
which utilizes a low-delay sub-band combining apparatus as
described above The invention moreover enables such an encoder
apparatus and decoder apparatus to each perform efficiently by
providing a high-speed algorithm which utilizes the periodicity of
a cosine function to minimize an amount of processing which is
required to implement the respective functions of the various
band-pass filters of such an apparatus
[0019] Such an encoder apparatus and decoder apparatus are
particularly suitable for use in compression encoding and
subsequent expansion decoding of a PCM digital audio signal.
[0020] The invention further enables an improved digital wireless
microphone system to be configured, in which a digital audio signal
which is to be transmitted by radio as a data stream is
compression-encoded by an encoding apparatus utilizing a low-delay
sub-band separating apparatus and is subsequently decoded upon
reception, by using a decoding apparatus similarly utilizing a
low-delay sub-band combining apparatus according to the
invention.
[0021] The apparatus moreover provides an echo canceller apparatus
in which a digital audio signal received from a remote location to
be audibly reproduced by a loudspeaker is subjected to sub-band
separation and then adaptive filtering of the respective sub-band
signals, a digital audio signal obtained from a microphone which
may be adjacent to the loudspeaker is also converted to a set of
sub-band signals, the differences between these signals and the
adaptively filtered sub-band signals are obtained as respective
error signals and applied to update the coefficients of the
adaptive filters, and are also subjected to band-combining filter
processing to obtain an output digital signal that is returned to
the remote location, in which the sub-band separating and sub-band
combining processing are performed using low-delay sub-band
separating apparatuses and a low-delay sub-band combining apparatus
according to the invention. As a result, due to the reduced amounts
of filter delay, more effective suppression can be achieved of a
signal that is returned to the remote location as an echo.
BRIEF DESCRIPTION OF THE DRAWINGS
[0022] FIG. 1 is a general system block diagram of a digital signal
sub-band separating apparatus and sub-band combining apparatus
according to a first embodiment of the invention;
[0023] FIG. 2 is a graph illustrating an asymmetric impulse
response of a FIR filter;
[0024] FIG. 3A is a graph for comparing respective
amplitude/frequency characteristics of a conventional type of FIR
low pass filter and of a FIR low pass filter having an asymmetric
impulse response;
[0025] FIG. 3B is a graph for comparing respective group
delay/frequency characteristics of a conventional type of FIR low
pass filter and of a FIR low pass filter having an asymmetric
impulse response;
[0026] FIG. 4 is a general system block diagram of an embodiment of
a PCM digital audio signal encoder apparatus and decoder apparatus,
utilizing a sub-band separating apparatus and sub-band combining
apparatus according to the invention, for use in transmitting or
storing digital audio signal data in compressed encoded form;
[0027] FIG. 5 is a general system block diagram of an embodiment of
a wireless microphone transmitter system, having a PCM digital
audio signal encoder apparatus and decoder apparatus, utilizing a
sub-band separating apparatus and sub-band combining apparatus
according to the invention;
[0028] FIG. 6 is a general system block diagram of an embodiment of
an echo canceller apparatus which utilizes sub-band separating
apparatuses and a sub-band combining apparatus according to the
invention;
[0029] FIG. 7 is a general system block diagram of an example of a
prior art sub-band separating apparatus and sub-band combining
apparatus; and
[0030] FIG. 8 is a graph illustrating a symmetric impulse response
of a FIR filter as used in the apparatus of FIG. 7.
DESCRIPTION OF PREFERRED EMBODIMENTS
First Embodiment
[0031] FIG. 1 is a general system block diagram of a first
embodiment of the invention, which is a combination of a sub-band
separating apparatus and a sub-band combining apparatus, for use
with a processing (or transmitting/receiving) system. In FIG. 1, a
band-separating filter bank 1 performs filtering of respective
frequency bands of an input PCM digital signal x(n), then
decimation is applied, using a decimation factor of D. The
band-separating filter bank 1 is formed of a set of band-separating
filters 010.about.01n and a corresponding set of down-samplers
020.about.02n, each of which applies decimation by a fixed factor D
(i.e., selecting one in every D successive samples) to the output
sub-band signal from the corresponding one of the band-separating
filters 010.about.01n. After the resultant decimated sub-band
signals have been subjected to predetermined processing in a
processing section 3, they are inputted to a band-combining filter
bank 2, which effects interpolation of the signals, i.e., by
inserting (D-1) interpolation values for each value of an input
signal, and filtering of the resultant interpolated sub-band
signals by respective filters. The resultant filtered interpolated
sub-band signals are then additively combined by an adder 4, to
recover the original PCM digital signal y(n) or a processed version
of that signal.
[0032] The band-combining filter bank 2 is formed of a set of
up-samplers 040.about.04n which receive and apply interpolation by
the aforementioned factor D to respectively corresponding ones of
the sub-band signals which are outputted from the processing
section 3, and a set of band-combining filters 030.about.03n
respectively corresponding to the frequency bands of the sub-band
signals, which receive and filter respectively corresponding ones
of the interpolated sub-band signals which are outputted from the
up-samplers 040.about.04n, with the resultant sub-band signals
being supplied to the adder 4.
[0033] If the processing section 3 performs a type of processing
such as echo cancellation, for which each of the band-separating
filter bank 1 and band-combining filter bank 2 should be DFT filter
banks, then the respective transfer functions of the k-th
band-separating filter of filter bank 1 and the k-th band-combining
filter of the filter bank 2 are expressed as follows by equations
(5) and (6) respectively:
H.sub.k(z)=H.sub.0(zW.sub.M.sup.k) (5)
F.sub.k(z)=W.sub.M.sup.-kF.sub.0(zW.sub.M.sup.k) (6)
[0034] Where W.sub.M.sup.k=exp(-j2.pi.k/M), with 0<k <M-1
[0035] In equations (5), (6), H.sub.0(z) and F.sub.0(z)
respectively express the transfer functions of the prototype
filters 010, 030 respectively of the band-separating filter bank 1
and band-combining filter bank 2 of this embodiment, for the case
in which each of these is a DFT filter bank.
[0036] If the processing section 3 on the other hand performs a
type of processing which requires that each of the band-separating
filter bank 1 and band-combining filter bank 2 be a cosine
modulation filter bank, then transfer functions of the k-th
band-separating filter of filter bank 1 and the k-th band-combining
filter of the filter bank 2 are expressed as follows by equations
(7) and (8) respectively:
H.sub.k(z)=e.sup.j.theta..sup.kP(W.sub.2M.sup.(k+1/2)z)+e.sup.-j.theta..su-
p.kP(W.sub.2M.sup.(k-1/2)z) (7)
F.sub.k(z
)=e.sup.j.PSI..sup.kP(W.sub.2M.sup.(k+1/2)z)+e.sup.-j.PSI..sup.k-
P(W.sub.2M.sup.(k-1/2)z) (8)
[0037] where 0<k<M-1, and where, designating the group delay
of the prototype filter as k.sub.d, the following relationships are
true:
.theta..sub.k(z)=(M-k.sub.d)(2k+1).PI./(4M)
.PSI..sub.k(z)=(-M-k.sub.d)(2k+1).PI./(4M)
[0038] In equations (7)and (8), P(z)expresses the transfer function
of the prototype filter of a cosine modulation filter bank, i.e.,
in this case, the transfer function of each of the prototype
filters 010 and 030 of the band-separating filter bank 1 and
band-combining filter bank 2 respectively.
[0039] For encoding efficiency, the decimation/interpolation factor
D is preferably made equal to the separation factor M, i.e., made
equal to the number of sub-band signal channels.
[0040] FIG. 2 shows an example of the impulse response of each of
the prototype filters 010, 030 of this embodiment. As shown, this
is an asymmetric impulse response, as opposed to the symmetric
impulse response shown in FIG. 8.
[0041] The operation of the sub-band separating/combining apparatus
having the configuration set out above will be described referring
to FIG. 1. The input PCM digital signal x(n) is supplied to the
band-separating filter bank 1, to be subjected to convolution
processing in respective frequency bands by the band-separating
filters 010.about.01n, to be thereby separated into respective
sub-band signals which are outputted from these filters.
[0042] Each of these sub-band signals is then subjected to
decimation by the factor D (i.e., through extraction of one out of
every D successive samples of a sub-band signal) by the
corresponding one of the down-samplers 020.about.02n. The resultant
decimated sub-band signals are then subjected to some form of
signal processing by the processing section 3, where the term
"processing" is to be interpreted as having a broad significance
which can for example include encoding a signal for transmission or
storage, followed by decoding upon reception or read-out
[0043] The resultant processed sub-band signals which are produced
from the processing section 3 are supplied to respectively
corresponding ones of the up-samplers 040.about.04n in the
band-combining filter bank 2, and each of the resultant
interpolated sub-band signals is then subjected to convolution by
the corresponding one of the band-combining filters 030.about.03n.
The resultant filtered sub-band signals are then additively
combined by the adder 4, to obtain as output a recovered PCM
digital signal, i.e., in general, a modified version of the
original PCM digital signal x(n), as determined by the processing
applied by the processing section 3.
[0044] FIG. 3A shows a comparison between the amplitude/frequency
response of a prototype filter (i.e., a FIR low pass filter) having
an asymmetric impulse response as utilized with the present
invention, as indicated by the full-line curve, and a prototype
filter having a symmetric impulse response as used in the prior
art, as indicated by the broken-line curve. Both of the filters are
formed with 128 taps, and differ only with respect to the impulse
response.
[0045] FIG. 3B shows a comparison between the group delay/frequency
characteristic of a digital signal sub-band separating/combining
apparatus such as that of FIG. 1 (i.e., with respect to the total
amount of group delay which occurs from input to a sub-band
separating filter bank to output from a sub-band combining filter
bank, and results only from the effects of these filter banks), for
the case in which both of the filter banks utilizes a prototype
filter having an asymmetric impulse response as illustrated in FIG.
2, in accordance with the present invention, with that
delay/frequency characteristic being shown as a full-line curve,
and for the case in which the filter banks each utilize a prototype
filter having a symmetric impulse response, as illustrated in FIG.
8, with that delay/frequency characteristic being shown as a
broken-line curve.
[0046] As is clear from FIG. 3A, with this embodiment of the
invention, the attenuation/frequency characteristic of the
prototype filter is closely similar to that of prior art type of
filter used as a prototype filter of a digital signal sub-band
separating/combining apparatus. However as can be seen from FIG.
3B, a significant improvement is obtained with regard to reducing
the amount of group delay which is incurred in the
separating/combining processing.
[0047] Thus with this embodiment of the invention, by using a FIR
low-pass filter having an asymmetric impulse response as each of
the respective prototype filters of a band-separating filter bank
and band-combining filter bank of a digital signal sub-band
separating/combining apparatus, the amount of overall delay which
results from transfer of a digital signal through such an apparatus
can be substantially reduced, without causing significant
deterioration of the attenuation/frequency characteristic of the
apparatus.
Second Embodiment
[0048] FIG. 4 shows a second embodiment of the invention, which is
a PCM digital audio signal compression encoding/decoding apparatus.
It should be understood that the invention could of course be
applied to various other types of digital signal encoding
apparatus. In FIG. 4, an encoder 101 receives as input a PCM
digital audio signal, performs sub-band separating processing, and
uses human psycho-acoustic response characteristics etc., to
perform compression encoding processing. The encoder 101 is formed
of a band-separating filter bank 102, psycho-acoustic model section
103, quantization/encoding section 104 and frame forming section
105.
[0049] The band-separating filter bank 102 is formed as described
hereinabove for the band-separating filter bank 1 of the first
embodiment, for the case in which this is a cosine modulation
filter bank in which the band-separating filters are configured in
accordance with equation (7) above, with a decimation factor D that
is identical to the separation factor (i.e., is equal to the number
of sub-band signal channels.
[0050] The bit stream that is produced from the encoder 101 is
inputted to the demodulator 106, in which the original sub-band
signals are subjected to dequantization and sub-band combining
processing in accordance with the frame information, to thereby
recover the original PCM digital audio signal. The demodulator 106
is formed of a frame analyzing section 107, a
dequantization/decoding section 108 and a band-combining
filter-bank 109.
[0051] The band-combining filter bank 109 is formed as described
hereinabove for the band-separating filter bank 1 of the first
embodiment, for the case in which this is a cosine modulation
filter bank in which the band-separating filters are configured in
accordance with equation (8) above, with a decimation factor D that
is identical to the number M of sub-band signal channels.
[0052] The operation of the encoder 101 and demodulator 106 will be
described referring to FIG. 4. Firstly, a PCM digital audio signal
is inputted to the encoder 101, and is converted to D channels of
sub-band signals (i.e., respective sequences of decimated samples)
by the band-separating filter bank 102 as described hereinabove for
the first embodiment.
[0053] These sub-band signals are inputted to the
quantization/encoding section 104 and the psycho-acoustic model
section 103, to be processed in parallel by these. In the
psycho-acoustic model section 103, the input PCM digital audio
signal is subjected to frequency analysis by a method such as FFT
processing, etc., to calculate scale factor information from the
sub-band signals and to calculate a masking level for the
quantization error based on a psycho-acoustic model of human
auditory characteristics. Bit allocation information is thereby
calculated for each of the frequency bands respectively
corresponding to the sub-band signals. However it should be noted
that it would be equally possible to calculate only the bit
allocation information at this time, without performing FFT
processing.
[0054] In the quantization/encoding section 104, quantization and
encoding are performed in accordance with the bit allocation
information which is calculated by the psycho-acoustic model
section 103, and the resultant encoded data are combined with
externally supplied ancillary data in the frame forming section
105, to obtain successive data frames which are outputted from the
encoder 101.
[0055] These data frames are then transmitted via a transmission
path 110 to be inputted to the demodulator 106. In the demodulator
106, the frame analyzing section 107 first performs frame analysis
to separate out the ancillary data of the frames, and also
separates the bit allocation information and the sub-band sample
information for the respective sub-bands, from the side information
which has been transmitted within the frames. The
dequantization/decoding section 108 then recovers the original set
of sub-band signals, and these are inputted to the band-combining
filter bank 109. Here, filtering and interpolation of samples are
applied to the respective sub-band signals, and additive combining
of the resultant sub-band signals, are performed as described
hereinabove for the band-combining filter bank 2 of the first
embodiment, to recover the original PCM digital audio signal.
[0056] With this embodiment of the invention, in which the
band-separating filter bank 102 and band-combining filter bank 109
each achieve a low amount of group delay, a PCM digital signal
compression encoding/decoding apparatus can be realized which has a
reduced amount of overall system delay.
[0057] In the above description, an example is given in which
low-delay sub-band separating/combining is performed in the case of
PCM digital audio signal compression encoding/decoding. However it
would be equally possible to apply the principles described above
to a quantization algorithm for modifying images, i.e., to
compression encoding and decoding of a digital video signal.
[0058] Furthermore the invention could be applied to achieve a
higher speed of processing for the band-separating and
band-combining operations by making the number of taps of each
prototype filter twice the separation factor M, i.e., 2M and by
converting each of the above equations (5), (6) to the time domain,
and making use of the fact that a cosine function within each of
the converted equations periodically takes the values 1 and -1 for
successive signal samples, as shown in the following. This also
will enable the hardware and memory requirements for performing the
processing to be reduced. Specifically, equations (5), (6) can be
expressed as respective time-axis functions by the following
equations (7), (8): 1 h k ( n ) = 2 p L ( n ) cos [ ( 2 k + 1 ) 2 M
( n - k d 2 ) - ( 2 k + 1 ) 4 ] ( 7 ) f k ( n ) = 2 p L ( n ) cos [
( 2 k + 1 ) 2 M ( n - k d 2 ) + ( 2 k + 1 ) 4 ] ( 8 )
[0059] In the above, k is a band index, i.e., taking values 0,1, .
. . ,M-1, h.sub.k(n) is the impulse response of the band-separating
filter for the k-th sub-band, f.sub.k(n) is the impulse response of
the band-combining filter for the k-th sub-band, p.sub.L(n) is the
impulse response of each prototype filter, and k.sub.d is the group
delay measured from input to output of the band-separating filter
bank or band-combining filter bank.
[0060] The manner of achieving high-speed processing will be
described only for the case of band-separating operation. The
band-separating processing can be expressed by the following
equation (9): 2 x k ( r ) = n = 0 N - 1 h k ( n ) x ( rM - n ) ( 9
)
[0061] In the above, x.sub.k(r) is the output sub-band signal which
results from filtering and decimation of the k-th band, with r
expressing respective time-axis positions of the signal samples, N
is the number of taps of the filter for the k-th-band, and x(n) is
the input signal to the band-separating filter bank, i.e., with n
expressing the respective time-axis positions of the input signal
samples.
[0062] Designating n=2M_65 +.rho., and inserting the resultant form
of equation (7) into equation (9), the following equation (10) can
be obtained: 3 x k ( r ) = = 0 2 M - 1 = 0 N 2 M - 1 cos [ ( 2 j +
1 ) 2 M ( 2 M + - k d 2 - M 2 ) ] 2 p L ( 2 M + ) x ( Mr - 2 M - )
( 10 )
[0063] Furthermore, designating the cos term in equation (10) as A,
and developing that term A, the following equation (11) can be
obtained: 4 A = cos [ ( 2 k + 1 ) 2 M ( - M 2 - k d 2 ] cos [ ( 2 k
+ 1 ) ] ( 11 )
[0064] In equation (11), the portion cos(2k+1).pi..gamma. takes the
value +1 when .gamma. is even =31 1 when .gamma. is odd. As a
result, equation (10) can be rewritten as follows, as equation
(12): 5 x k ( r ) = = 0 2 M - 1 [ cos [ ( 2 k + 1 ) 2 M - ( - M 2 -
k d 2 ) ] = 0 N 2 M - 1 ( - 1 ) 2 p L ( 2 M + ) x ( Mr - 2 M - ) ]
( 12 )
[0065] Use of equation (12) as the algorithm for deriving each of
the sub-band signals from the input digital signal x(n) enables the
band-separation processing to be performed efficiently. A similar
algorithm can be utilized for operating on each of the sub-band
signals which are to be combined, in the band-combining
processing.
Third Embodiment
[0066] FIG. 5 shows the system configuration of a third embodiment
of the invention. This is a wireless microphone system which uses
sub-band compression encoding/decoding processing having a low
amount of delay, implemented as described above for the second
embodiment. As a result with this system, by comparison with the
prior art, there is a reduced amount of delay between the time at
which a sound is received by a microphone of the system and the
time at which a corresponding amplified sound is emitted from a
loudspeaker.
[0067] In FIG. a transmitter 200 applies A-D conversion to convert
an audio signal from a microphone into a PCM digital audio signal,
then applies compression encoding processing as described
hereinabove for the second embodiment of the invention, to obtain a
compressed bit stream. The bit stream is then subjected to encoding
conversion to reduce the effects of errors which may arise when the
bit stream traverses a transmission path, and the resultant signal
is then applied in digital modulation to obtain a high-frequency
modulated signal which is transmitted as radio waves. The
transmitter 200 is made up of a microphone 202, an analog signal
amplifier 203, an A-D converter 204, a compression encoder 205, a
code conversion/interleaving/error correction circuit 206, a
modulator/amplifier circuit 207 and a transmitting antenna 208. The
compression encoder 205 is is configured in accordance with the
second embodiment of the invention.
[0068] A receiver 201 receives the radio waves which are
transmitted from the transmitter 200, amplifies the resultant
signal and applies frequency conversion and demodulation. The
resultant demodulated signal is then subjected to transmission path
error correction processing, and the resultant encoded compressed
signal is decoded to obtain a digital output signal. That digital
output signal is then subjected to digital-analog conversion to
obtain an analog output audio signal, which can be supplied to
drive a transducer such as a loudspeaker (not shown in the
drawing). The receiver 201 is made up of a receiving antenna 209, a
high-frequency amplifier/frequency converter 210 coupled to receive
a high-frequency signal from the antenna 209, an
intermediate-frequency amplifier 211, a demodulator 212, a code
conversion/de-interleaving/error correction circuit 213, a
compressed signal decoder 214, a D-A converter 215 and a analog
signal amplifier 216. The compressed signal decoder 214 is
configured in accordance with the second embodiment of the
invention.
[0069] The operation of this digital wireless microphone system is
as follows. Firstly, sound waves which reach the microphone 202 are
converted to an analog audio signal which is amplified to an
appropriate level by the analog signal amplifier 203, and the
resultant signal is converted to a PCM digital audio signal by the
A-D converter 204. In the compression encoder 205, the PCM digital
audio signal is subjected to compression encoding with a low amount
of delay, then encoding conversion is applied to reduce the effects
of errors arising in the transmission path, by the error correction
code conversion circuit 206, to obtain the final encoded data
stream. Various schemes for processing data prior to transmission
so that transmission errors can be automatically corrected in the
receiving process, such as BCH encoding, interleaving, etc., which
could by utilized for the operation of the error correction code
conversion circuit 206.
[0070] The resultant encoded data stream is sent to the
modulator/amplifier circuit 207, in which it is applied in digital
quadrature modulation such as .pi./4-DQPSK (direct quadrature phase
shift keying) modulation, to be converted to a modulated RF signal.
This is then amplified to a sufficient level by an amplifier, and
supplied to the transmitting antenna 208 to be transmitted as radio
waves.
[0071] In the receiver 201, the radio waves are received by the
receiving antenna 209, the resultant signal is amplified by the
high-frequency amplifier/frequency converter 210, and converted to
an intermediate-frequency signal by the high-frequency
amplifier/frequency converter 210, then is amplified by
intermediate-frequency (IF) amplifier 211 to a sufficiently high
level for performing demodulation. The resultant IF signal is then
demodulated by the demodulator 212. The demodulated signal is
subjected to error correction processing to eliminate code errors
which may have arisen in the transmission path, by the error
correction code conversion section 213, to obtain an
error-corrected signal. The compressed signal decoder 214 then
applies low-delay decoding, to recover an original set of sub-band
signals, and additive combination of these sub-band signals to
recover the original PCM digital audio signal.
[0072] In some cases it may be possible for the receiving apparatus
to directly output only that recovered PCM digital audio signal.
However since it may be necessary to drive an analog type of
amplifier apparatus such as a high-power audio amplifier, it is
preferable to perform D-A conversion so that an analog output
signal can also be provided. With this embodiment the PCM digital
audio is converted to an analog audio signal by the D-A converter
215, which is then amplified by the analog signal amplifier 216 to
obtain an output analog audio signal.
[0073] With this embodiment of the invention, a band-separating
filter bank and band-combining filter bank each having a low amount
of filtering delay are utilized, in a digital signal compression
encoding/decoding apparatus. As a result it is possible to
implement a digital type of wireless microphone system which
enables sounds to be produced in amplified form from a loudspeaker
with a minimum of delay between reception of the sounds by a
microphone and emission of the sounds from the loudspeaker.
[0074] Furthermore, when compression encoding is performed using
digital modulation, frequency can be effectively utilized, so that
it becomes possible to simultaneously use a plurality of wireless
microphones.
Fourth Embodiment
[0075] FIG. 6 shows the configuration of an echo canceller which
uses sub-band separating/combining processing in accordance with
the first embodiment of the invention described above. For the
purpose of description, it is assumed that the apparatus shown in
FIG. 6 receives an input digital audio signal x(k) which is
transmitted from a distant location (referred to in the following
as the far-end location) as a result of speech sound waves that are
produced by an individual (referred to in the following as the
far-end individual) entering a microphone, with the resultant audio
signal being converted to the digital audio signal x(k) and
transmitted via some form of communication link. The location of
the apparatus shown in FIG. 6 will be referred to as the near-end
location, and will be assumed to be an enclosed room. In FIG. 6,
numeral 323 denotes a combination of a D/A converter for converting
the digital audio signal x(k) to analog form, a loudspeaker, and an
audio amplifier which amplifies the analog audio signal to drive
the loudspeaker, however for brevity of description that
combination will be referred to simply as the loudspeaker 323. Also
in FIG. 6, numeral 324 denotes a combination of a microphone and an
A/D converter for converting an analog audio signal from the
microphone to a digital audio signal, with that combination being
referred to in the following simply as the microphone 324. The
purpose of the microphone 324 is to enable an individual at the
near-end location to communicate with the far-end individual,
however for the purpose of the following description, only those
sounds which reach the microphone 324 from the loudspeaker 323 will
be considered.
[0076] The objective of the echo canceller is to prevent sound
waves which are emitted from the loudspeaker 323 as a result of the
input digital audio signal x(k) and enter the microphone 324 (in
accordance with a transfer function of the room at the far-end
location, with respect to transmission of sound waves from the
loudspeaker 323 to the microphone 324) from being transmitted back
to the far-end individual in delayed form, as echoes. Basically,
the echo canceller estimates the transfer function of the far-end
location, and controls a set of adaptive filters accordingly such
as to cancel any audio signal components from the microphone 324
that result from the audio signal being applied to the loudspeaker
323.
[0077] The echo canceller apparatus is formed of a first
band-separating filter bank 320, whose configuration and operation
are as described hereinabove for the band-separating filter bank 1
of the first embodiment, a set of adaptive FIR filters
300.about.30n which respectively receive the decimated sub-band
signals produced from the band-separating filter bank 320, a set of
coefficient updating sections 310.about.31n each of which operates
on a corresponding one of the adaptive FIR filters 300.about.30n to
adjust the tap coefficients of that corresponding filter, a set of
adders 330.about.33n whose respective outputs are supplied to
corresponding ones of the coefficient updating sections
310.about.31n and which each receives at a first input thereof an
output signal produced from a corresponding one of the adaptive FIR
filters 300.about.30n, a second band-separating filter bank 321,
whose operation and configuration are also in accordance with the
band-separating filter bank 1 of the first embodiment and which
receives the aforementioned digital audio signal produced from the
microphone 324 and inputs each of the resulting decimated sub-band
signals to a subtraction input of a corresponding one of the adders
330.about.321, a band-combining filter bank 322 whose operation and
configuration are in accordance with the band-combining filter bank
2 of the first embodiment and which receives respectively outputs
produced from the adders 330.about.321, and an adder 340 which
additively combines the interpolated sub-band signals which are
produced from the band-combining filter bank 322 to obtain a
digital audio output signal y(n), to be transmitted back to the
far-end location via a communication link.
[0078] Each of the band-separating filter banks 320 and 321 and the
band-combining filter bank 322 is configured with a prototype
filter which is a FIR low-pass filter having an asymmetric impulse
response, with each of the band-separating filter banks 320, 321
being a DFT filter bank which is formed in accordance with equation
(5) above and with the band-combining filter bank 322 being a DFT
filter bank which is formed in accordance with equation (6)
above.
[0079] The operation of this echo canceller apparatus is as
follows. The input audio signal which is sent from the far-end
individual passes over a transmission path and arrives as the
digital audio signal x(k) which is supplied to the band-separating
filter bank 320 and to the loudspeaker 323. In the band-separating
filter bank 320, the input signal is subjected to convolution
processing in respective frequency bands by the band filters, and
the resultant signals are subjected to decimation processing by a
decimation factor D which is no greater than the separation factor
M, as described for the first embodiment. The resultant
down-sampled sub-band samples are inputted to the adders
330.about.33n. Sound waves which are emitted by the loudspeaker 323
are received by the microphone 324, and the resultant audio signal
is inputted to the band-separating filter bank 321.
[0080] In the band-separating filter bank 321, the input signal is
subjected to convolution processing in respective frequency bands
by the band filters, and the resultant sub-band signals are
subjected to decimation processing by a decimation factor D which
is no greater than the separation factor M, as described for the
first embodiment. The resultant down-sampled sub-band signals are
inputted to respectively corresponding ones of the adaptive FIR
filters 300.about.30n. The resultant output signal from each of the
adaptive FIR filters 300.about.30n has the corresponding one of the
sub-band signals from the band-separating filter bank 320
subtracted therefrom, in the corresponding one of the adders
330.about.33n, to thereby obtain an error signal. These error
signals are inputted to respectively corresponding ones of the
coefficient updating sections 310.about.31n, and also inputted to
respectively corresponding ones of the up-converters of the
band-combining filter bank 322. Thus, the signal y(n) which is
obtained from the band-combining filter bank 322 will be reduced in
amplitude in accordance with reduction of the error signals.
[0081] A known type of coefficient updating algorithm such as the
NLMS algorithm can be utilized for the operation of each of the
coefficient updating sections 310.about.31n. Such an algorithm is
of a type whereby, at each sample time point of the input digital
signal to a FIR filter, the coefficients of the filter are updated
by adding thereto an updating amount, which is determined in
accordance with a preceding history of errors between the actual
output values produced from the filter and respective ideal values
which would be produced by an ideal FIR filter. With a LMS (least
mean-square) type of adaptive algorithm, only the error resulting
from the immediately preceding digital signal input to the filter
is utilized, in general, with the updating being successively
performed starting from an initial assumed set of coefficients
(e.g., all zero). The coefficients of a filter are processed as a
vector quantity, as are each of the digital signal values.
[0082] Basically, designating successive input signal sample time
points as 0,1 . . . k, (k+), and the corresponding values of the
sets of coefficients of a filter as the vectors w(0), w(1) . . .
w(k), w(k+1), . . . , e.g., with w(0) being predetermined as zero,
and designating as .delta.w(k)an updating amount, each of
successive coefficient values w(k+1) are obtained as:
w(k+1)=w(k)+.mu...delta.w(k) (13)
[0083] Here, .mu. is an adaptation constant, generally referred to
as the step size, which controls the size of the updating amount,
at each update. The updating amount .delta.w(k) is derived based on
an amount of error between the preceding output value w(k) from the
filter and the output which would have been produced from an ideal
filter, which can be considered as being identical to the value of
the target signal at that time (i.e., with this embodiment, the
output value of the corresponding one of the sub-band signals from
the sub-band separating filter bank 321 at that time).
[0084] With the NLMS (normalized least mean-square) method, the
step size is normalized, i.e., is automatically adjusted based on
the power of the input signal to the filter. Typically, the NLMS
algorithm may be expressed as follows:
W(k+1)=w(k)+(.alpha./(x(k).sup.Tx(k)+.beta.))e(k)x(k) (14)
[0085] Here, .alpha. determines the maximum step size
(0<.alpha.<2), D is a small-magnitude value for the purpose
of preventing division by zero, x(k) is the (preceding) input
signal value to the filter, e(k) is the aforementioned error amount
between the target signal and the signal resulting from the
adaptive filtering of the preceding input signal value, and
superscript T denotes the transposed matrix.
[0086] With this embodiment, the sub-band signals of respective
frequency bands which are produced from the band-separating filter
bank 320 are subjected to convolution processing by respectively
corresponding ones of the adaptive filters 300.about.30n, with
updating of the filter coefficients being performed for each of
these by the corresponding one of the coefficient updating sections
31n as described above, and the resultant error signals
respectively corresponding to these filtered sub-band signals from
the adaptive filters, obtained from the adders 330.about.33n, are
inputted to the band-combining filter bank 322. In the
band-combining filter bank 322, these signals are subjected to
interpolation processing, using the same interpolation factor D as
the factor M used for band separation, and convolution with the
band-combining filters is then applied for the respective frequency
bands.
[0087] The additive combination of the resultant signals is then
obtained by the adder 340, as the digital audio signal y(n), which
is the output signal from the echo canceller. That signal is
transmitted over a communication link to the far-end
individual.
[0088] With this embodiment, due to the fact that each of the
filter banks is configured to utilize a FIR low-pass filters having
an asymmetric impulse response as the prototype filter, thereby
achieving a lower amount of group delay for each of the filter
banks than is possible in the prior art, it is found that greater
effectiveness can be achieved in suppressing a spurious digital
audio signal that may be produced as a component of the output
signal y(n) due to sound waves from the loudspeaker 323 reaching
the microphone 324, and thereby returned to the far-end individual.
Hence, greater effectiveness in echo suppression can be achieved
than is possible in the prior art.
[0089] It should be noted that although the invention has been
described in the above referring to specific embodiments, it should
be understood that various modifications to the described
embodiments could be envisaged, which fall within the scope claimed
for the invention in the appended claims.
* * * * *