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Claims  |
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What is claimed is:
1. An acoustic echo canceller comprising:
a receive signal input terminal,
a receive signal output terminal,
a transmit signal input terminal,
a transmit signal output terminal,
a variable coefficient digital filter comprising an artificial impulse
response register processed in N divided coefficient blocks, where N is a
positive integer, in response to a receive signal from the receive signal
input terminal and for generating an artificial acoustic echo signal,
a subtractor located between the transmit signal input terminal and the
transmit signal output terminal for subtracting the artificial acoustic
echo signal generated by the variable coefficient digital filter from an
acoustic echo component of the receive signal input from the receive
signal output terminal via an acoustic echo path to the transmit signal
input terminal to find a difference signal,
a coefficient correction amount calculation circuit, to which the receive
signal from the receive signal input terminal and the difference from the
subtractor are supplied, for sequentially updating division coefficients
so as to minimize the difference signal,
a first switch located between the receive signal input terminal and the
receive signal output terminal,
a second switch located between the transmit signal input terminal and the
transmit signal output terminal, the first and second switches being
connectable to an associated party terminal,
a code generator for generating a code for observing an external acoustic
echo characteristic in a state in which the first and second switches are
disconnected from the associated party terminal,
operation means for calculating the external acoustic echo characteristic
when the code is input from the receive signal output terminal via the
acoustic echo path to the transmit signal input terminal and for
generating an operation result,
storage means for storing data, the data stored in the storage means
including the operation result generated by the operation means,
means, when the first and second switch means are connected to the
associated party terminal after a predetermined time has elapsed, for
calculating a sum power for each block in response to the data stored in
the storage means,
means for comparing sum power values between blocks of the artificial
impulse response register which are contiguous and for generating a
comparison result, and
update means for arbitrarily setting an update frequency of each division
coefficient in response to the comparison result.
2. The acoustic echo canceller of claim 1, wherein the operation means
comprises:
a synchronous adder for calculating an impulse response of the code input
from the receive signal output terminal via the acoustic echo path to the
transmit signal input terminal, and
a convolution integral operation circuit for performing a convolution
integral operation on an output of the synchronous adder and the code to
calculate an impulse response of the acoustic echo path.
3. An acoustic echo canceller comprising:
a receive signal input terminal,
a receive signal output terminal connected to the receive signal input
terminal,
a transmit signal input terminal,
a transmit signal output terminal,
a variable coefficient digital filter having an artificial impulse response
register divided into N blocks to which a receive signal through the
receive signal input terminal is input and for generating an artificial
acoustic echo signal, where N represents a positive integer, the impulse
response register storing a plurality of coefficients,
a coefficient series block selector for selecting n blocks to be updated at
a time among a total of N divided blocks of the artificial impulse
response register,
a subtractor located between the transmit signal input terminal and the
transmit signal output terminal for subtracting the artificial acoustic
echo signal generated by means of the variable coefficient digital filter
from an acoustic echo component of the receive signal input from the
receive signal output terminal via an acoustic echo path to the transmit
signal input terminal to find a difference signal,
a coefficient correction amount calculation circuit, to which the receive
signal from the receive signal input terminal and the difference from the
subtractor are supplied, for sequentially updating coefficient series so
as to minimize the difference signal,
a first power calculation circuit for finding a power of each of the
plurality of coefficients stored in the impulse response register, and
a power comparator for totaling power values for each block output by the
first power calculation circuit and comparing the totaled power values,
wherein an update frequency of each divided block adapted to a sound field
characteristic is determined in response to the comparison result of the
power comparator and coefficient correction is executed with P fixed
position blocks and Q (n-P) variable position blocks in response to the
determined update frequency.
4. The acoustic echo canceller of claim 3, comprising:
a second power calculation circuit for finding a power value of each
coefficient correction amount generated by the coefficient correction
amount calculation circuit for a fixed position block assigned the highest
one of update frequencies adapted to a sound field characteristic
according to the power comparator,
a coefficient divider for dividing each power value output by the second
power calculation circuit by each output of the first power calculation
circuit for the fixed position block, and
a coefficient change detector for comparing the calculation result given by
the coefficient divider with a predetermined threshold,
wherein when the output value of the coefficient divider is greater than
the threshold in the coefficient change detector, coefficient correction
operation processing in the same divided blocks as the fixed position
blocks is performed for the variable position blocks.
5. An acoustic echo canceller comprising:
a receive signal input terminal,
a receive signal output terminal connected to the receive signal input
terminal,
a transmit signal input terminal,
a transmit signal output terminal,
a variable coefficient digital filter for generating an artificial echo
signal in response to a receive signal input through the receive signal
input terminal,
a subtractor located between the transmit signal input terminal and the
transmit signal output terminal for finding a difference between an echo
signal from the transmit signal input terminal and the artificial echo
signal generated by the variable coefficient digital filter,
a coefficient correction amount calculation circuit, to which a receive
signal from the receive signal input terminal and the difference from the
subtractor are supplied, for performing division processing in which a
coefficient series of the variable coefficient digital filter is divided
into N blocks and the entire coefficient series is automatically updated
in M steps, where N and M represent positive integers, and
a coefficient series block selector for selecting at least one block to be
updated among the N blocks of the coefficient series,
wherein a coefficient correction amount fitted to an attenuation
characteristic of an impulse response in a sound field is set for the
block selected by the coefficient block selector.
6. The acoustic echo canceller of claim 5, wherein the variable coefficient
digital filter comprises an artificial impulse response register which
stores the coefficient series of the variable coefficient digital filter,
and a sum-of-products operation circuit which performs a convolution
integral operation on contents of the artificial impulse response register
and the receive signal from the receive signal input terminal.
7. The acoustic echo canceller of claim 6, wherein the variable coefficient
digital filter further comprises a receive signal input register which
stores the receive signal from the receive signal input terminal.
8. An acoustic echo canceller comprising:
a receive signal input terminal,
a receive signal output terminal connected to the receive signal input
terminal,
a transmit signal input terminal,
a transmit signal output terminal,
a variable coefficient digital filter for generating an artificial echo
signal in response to a receive signal input through the receive signal
input terminal,
a subtractor located between the transmit signal input terminal and the
transmit signal output terminal for finding a difference between an echo
signal from the transmit signal input terminal and the artificial echo
signal generated by the variable coefficient digital filter,
a coefficient correction amount calculation circuit, to which the receive
signal from the receive signal input terminal and the difference from the
subtractor are supplied, for applying a correction amount to a coefficient
series of the variable coefficient digital filter, and
a coefficient block selector for sending an instruction for selecting one
of the blocks in sequence for performing a coefficient update operation to
the coefficient correction amount calculation circuit so that the
coefficient series of the variable coefficient digital filter is divided
into N blocks for automatically updating the entire coefficient series in
a total of M steps, where N and M are positive integers, wherein an update
frequency is set in each block and wherein a correction loop gain
interpolated into the coefficient correction amount calculation circuit is
set in response to the update frequency set in each block.
9. The acoustic echo canceller of claim 8, wherein the variable coefficient
digital filter comprises an artificial impulse response register which
stores the coefficient series of the variable coefficient digital filter,
and a sum-of-products operation circuit which performs a convolution
integral operation on contents of the artificial impulse response register
and the receive signal from the receive signal input terminal.
10. The acoustic echo canceller of claim 9, wherein the variable
coefficient digital filter further comprises a receive signal input
register which stores the receive signal from the receive signal input
terminal. |
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Claims |
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Description |
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BACKGROUND OF THE INVENTION
1. Field of the Invention
This invention relates to an acoustic echo canceller for use with
communication lines, indoor sound field controllers, and high-quality
speech communication conference systems, and for cancelling an acoustic
echo component produced when a signal on a receiving communication line
appears on a transmitting communication line via an acoustic echo path.
2. Description of the Related Art
Generally, the acoustic echo cancellers are roughly classified into those
for cancelling an echo produced due to an impedance mismatch of a 2-wire
4-wire converter on long-distance telephone lines using a communication
satellite and submarine cables and those for cancelling an echo produced
due to acoustic coupling of speaker speech at a loudspeaking telephone set
in a TV conference system, etc., each of which includes a correction
amount calculation circuit, a variable coefficient filter for generating
an artificial acoustic echo, and a subtractor. The basic operation of the
acoustic echo canceller will be described hereinafter.
FIG. 1 shows the basic configuration of an acoustic echo canceller. A
receive signal input terminal 1 is connected to a receive signal output
terminal 2 and a receive signal at the receive signal input terminal 1 is
branched to a variable coefficient digital filter 3 for generating an
artificial echo. A transmit signal from a transmit signal input terminal 4
and the artificial acoustic echo which is an output of the variable
coefficient digital filter 3 are input to a subtractor 5 for cancelling
the acoustic echo component in the transmit signal. An output of the
subtractor 5 is sent to a transmit signal output terminal 6. An output of
the transmit signal output terminal 6 and the signal at the receive signal
input terminal 1 are input to a correction amount calculation circuit 7.
The filter coefficient of the variable coefficient digital filter 3 is
corrected in response to an output of the correction amount calculation
circuit 7. In the variable coefficient digital filter 3, the receive
signal is input to a receive signal input register 8 and a sum-of-products
operation on the receive signal in the receive signal input register 8 and
an artificial impulse response in an artificial impulse response register
9 is performed by a sum-of-products operation circuit 10. The result of
the sum-of-products operation circuit 10 is output as an artificial
acoustic echo. The receive signal output terminal 2 and the transmit
signal input terminal 4 are connected to a 2-wire 4-wire converter on a
long-distance telephone line or connected to a loudspeaker and a
microphone in a loudspeaking telephone system.
Assume that the signal propagation characteristic of an acoustic echo path
can be represented as a linear form and by an FIR type digital filter. Let
its impulse response be h(t), input receive signal be x(t), and sampling
time interval be T. Acoustic echo at time kT, y.sub.k, is represented as
follows:
Y.sub.k =h.sub.k 'x.sub.k ( 1)
where
h=[h.sub.1, h.sub.2, . . . , h.sub.n ]'
x=[x.sub.k-1, . . . , x.sub.k-n ]' (2)
': Inversion of vector
On the other hand, assuming that an estimated value of h at time kT is
hs.sub.k, an estimated value of y.sub.k, ys.sub.k is given as follows:
ys.sub.k =hs.sub.k 'x.sub.k ( 3)
When a speech signal exists at the receive signal input terminal 1 and only
an acoustic echo with no speech signal exists at the transmit signal input
terminal 4, the acoustic echo canceller performs echo cancel operation as
an adaptive operation state. Generally, a learning method for
identification ("A Learning Method for System Identification" by Atuhiko
NODA and Jin-ichi NAGUMO, Measurement and Control, Vol. 7, No. 9, pp.
597-605 (1968)) is adopted as an algorithm of the adaptive operation.
Sequential correction of hs.sub.k by the learning method for
identification is performed according to
hs.sub.k+1 =hs.sub.k +.alpha.(x.sub.k e.sub.k)/x.sub.k 'x.sub.k ( 4)
where
e.sub.k =y.sub.k -ys.sub.k, 0<.alpha..ltoreq.1 (5)
e.sub.k is called a remaining acoustic echo. Such calculation operation is
performed in the coefficient correction amount calculation circuit 7. A
variable coefficient series hs.sub.k is stored in the artificial impulse
response register 9. .alpha. is a correction loop gain for determining
sensibility of estimation; the nearer to 1.0 the value, the greater given
the correction amount, enabling an acoustic echo to be cancelled at a high
speed. However, for actual use, the value must be changed depending on
near-end noise and the line state. It is common practice to determine the
correction loop gain .alpha. according to a rule of thumb at present.
When the acoustic echo characteristic in a loudspeaking sound field is
represented by such FIR type digital filter, a large configuration of
several hundreds to several thousands of taps results and the operation
amount involved in updating the correction amount of the variable
coefficient series hs.sub.k becomes enormous and cannot be covered by a
small-scaled hardware. Thus, the variable coefficient series hs.sub.k is
divided into several stages for processing and the operation amount for
updating in one step is reduced (for example, Japanese Patent Unexamined
Publication No. Sho. 63-246934). As an example, FIG. 2 shows the acoustic
echo cancellation characteristic with an autoregressive signal as an input
when 2-division processing is performed for the variable coefficient
series divided into first and latter halves. For comparison, a case where
no division processing is performed is also shown. In the figure, "ERLE"
is short for echo return loss enhancement. Assuming that the total of
variable coefficient series is N, the division contents become as follows:
hs1.sub.k : 0 to N/2
hs2.sub.k : N/2+1 to N
By applying the above-mentioned division range, from expression (4), update
algorithm can be represented as
hs1.sub.k+1 =hs.sub.1k +.alpha.(x.sub.k e.sub.k)/x.sub.k 'x.sub.k ( 6)
hs2.sub.k+1 =hs.sub.2k +.alpha.(x.sub.k e.sub.k)/x.sub.k 'x.sub.k ( 7)
which is an adaptive algorithm for updating all variable coefficient series
hs.sub.k at M of two or in two steps (where M is the number of steps for
updating all the coefficient series). Therefore, the operation amount in
one step can be reduced to a half; of course, if the division count N is
increased, the operation amount can be reduced to 1/N accordingly.
If processing of updating the correction amount of variable coefficient
series hs.sub.k is performed with division, the operation amount involved
in the updating is reduced, but the variable coefficient not updated in
one step causes an estimation error to occur on generation of an
artificial acoustic echo; resultantly, the remaining echo increases and
the acoustic echo cancellation characteristic is degraded. As shown in
FIG. 2, as compared with processing in which all variable coefficients are
updated at a time, the updating processing with division requires about
double time until saturation, and the convergence speed lowers to a half.
When the convergence speed lowers, the remaining echo at a large level
exists on the line, causing the communication state to be degraded. Also,
if a path fluctuation occurs on the echo path during talking, the
follow-up characteristic to that state worsens and the acoustic echo
cancellation characteristic changes rapidly, causing rasping remaining
voice to occur, so that high-accuracy and high-quality acoustic echo
cancellation cannot be performed.
SUMMARY OF THE INVENTION
It is therefore an object of the invention to provide an acoustic echo
canceller which compensates degradation of the convergence speed involved
in updating the coefficient correction amount with division and improves
in stability of two-way communication detection.
It is another object of the invention to provide an acoustic echo canceller
which is excellent in operation stability, has a high follow-up
characteristic to acoustic echo path fluctuation, provides a high-speed
acoustic echo cancellation characteristic, and always maintains a large
acoustic echo cancellation amount for performing acoustic echo control in
a sound field.
To these ends, according to the invention, there is provided an acoustic
echo canceller including a receive signal input terminal, a receive signal
output terminal connected to the receive signal input terminal, a transmit
signal input terminal, a transmit signal output terminal, a variable
coefficient digital filter for generating an artificial echo signal in
response to a receive signal input through the receive signal input
terminal, a subtractor being located between the transmit signal input
terminal and the transmit signal output terminal for finding a difference
between an echo signal from the transmit signal input terminal and the
artificial echo signal generated by the variable coefficient digital
filter, a coefficient correction amount calculation circuit, to which the
receive signal from the receive signal input terminal and the difference
from the subtractor are supplied, for performing division processing in
which a coefficient series of the variable coefficient digital filter is
divided into N blocks for automatically updating the entire coefficient
series in M steps, and a coefficient block selector for selecting a block
to be updated among the N blocks of the coefficient series, wherein a
coefficient correction amount fitted to an attenuation characteristic of
an impulse response in a sound field is set for the block selected by the
coefficient block selector.
BRIEF DESCRIPTION OF THE DRAWINGS
The accompanying drawings, which are incorporated in and constitute a part
of this specification, illustrate embodiments of the invention and,
together with the description, serve to explain the objects, advantages
and principles of the invention. In the drawings,
FIG. 1 is a block diagram showing the basic configuration of a conventional
acoustic echo canceller using a general learning method for
identification;
FIG. 2 is a graph showing an acoustic echo cancellation characteristic for
an autoregressive signal when division processing is performed for a
conventional coefficient correction amount update;
FIG. 3 is a block diagram showing the configuration of an acoustic echo
canceller according to a first embodiment of the invention;
FIG. 4 is a graph showing an example of an impulse response characteristic
in a sound field;
FIG. 5 is a graph showing an acoustic echo cancellation characteristic;
FIG. 6 is a diagram showing an example of a weighted division processing
procedure in the first embodiment;
FIG. 7 is a graph showing an acoustic echo cancellation characteristic for
an autoregressive signal;
FIG. 8 is a block diagram showing the configuration of an acoustic echo
canceller according to a second embodiment of the invention;
FIG. 9 is a graph showing an example of coefficient series stored in an
artificial impulse response register in the case of using an adaptive
interpolation correction loop gain;
FIG. 10 is a graph showing an example of coefficient series stored in an
artificial impulse response register in the case of using a fixed
interpolation correction loop gain;
FIG. 11 is a graph showing an example of power of coefficient series
corresponding to FIG. 9;
FIG. 12 is a graph showing an example of power of coefficient series
corresponding to FIG. 10;
FIG. 13 is a graph showing an example of error signal power transition when
voice of a woman is used as a reference signal in the case of using an
adaptive interpolation correction loop gain;
FIG. 14 is a graph showing an example of error signal power transition when
voice of a woman is used as a reference signal in the case of using a
fixed interpolation correction loop gain;
FIG. 15 is a block diagram showing the configuration of an acoustic echo
canceller according to a third embodiment of the invention;
FIG. 16 is a diagram showing state evaluation combinations for determining
divided update block mapping;
FIG. 17 is a graph showing an example of error signal running average power
transition with white noise as a reference input in the case of using a
divided update block mapping;
FIG. 18 is a graph showing an example of error signal running average power
transition with white noise as a reference input without a divided update
block mapping;
FIG. 19 is a graph showing an example of acoustic echo cancellation
characteristics with white noise as a reference input;
FIG. 20 is a block diagram showing the configuration of an acoustic echo
canceller according to a fourth embodiment of the invention;
FIG. 21 is a graph showing an example of acoustic echo cancellation
characteristics with white noise as a reference input; and
FIG. 22 is a graph showing an example of acoustic echo cancellation
characteristics when rapid acoustic echo path fluctuation exists with
white noise as a reference input.
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS
Referring now to the accompanying drawings, there are shown preferred
embodiments of the invention.
FIG. 3 is a block diagram showing the configuration of an acoustic echo
canceller according to a first embodiment of the invention. As shown in
FIG. 3, the acoustic echo canceller in the embodiment has a coefficient
block selector 11 in addition to the same configuration as the
conventional acoustic echo canceller including a receive signal input
terminal 1, receive signal output terminal 2, variable coefficient digital
filter 3, transmit signal input terminal 4, subtractor 5, transmit signal
output terminal 6, coefficient correction amount calculation circuit 7,
receive signal input register 8, artificial impulse response register 9,
and sum-of-products operation circuit 10 and adopting the learning method
for identification as an adaptive algorithm.
That is, the acoustic echo canceller in the first embodiment includes the
receive signal input terminal 1, transmit signal output terminal 6,
receive signal output terminal 2 for sending a receive signal arriving at
the receive signal input terminal 1 to an unknown acoustic echo path,
transmit signal input terminal 4 for collecting the signal sent from the
receive signal output terminal 2 and an echo responding to the receive
signal, artificial impulse response register 9 which stores coefficient
series of an adaptive digital filter, sum-of-products operation circuit 10
for performing a convolution integral operation on the contents of the
artificial impulse response register 9 and the contents of the receive
signal input register 8 which stores the receive signal, subtractor 5 for
calculating the difference between the echo and an artificial echo
generated by the sum-of-products operation circuit 10, and coefficient
correction amount calculation circuit 7 for performing processing so that
the coefficient series in the artificial impulse response register 9 is
automatically updated so that the adaptive digital filter supplies an
approximate value of the echo. The added coefficient block selector 11 is
provided for dividing the artificial impulse response register 9 into a
plurality of blocks and selecting one of the blocks in sequence for
performing coefficient update operation, thereby performing division
processing of updating the coefficient correction amount fitted to the
impulse response attenuation characteristic in a sound field.
It is known that the impulse response in a sound field presents an
attenuation characteristic in which the coefficient amplitude lowers with
the time, as shown in FIG. 4 (for example, "Improvement on Adaptation of
an Echo Canceller in a Room" by Shoji MAKINO and Nobuo KOIZUMI,
Electronics Communication Society, Technical Report, EA89-3 (1989)). Which
part of the attenuation process of the impulse response contributes to the
acoustic echo cancellation characteristic in what degree was checked. The
result is shown as the cancellation characteristic in FIG. 5. The order N
of a variable coefficient series was fixed in the following manner
1) 0-N (full tap)
2) 0-3N/4 (3/4 tap)
3) 0-N/2 (1/2 tap)
4) 0-N/4 (1/4 tap)
and acoustic echo cancellation operation was performed for a white noise
signal. As a result, if the first half of the variable coefficient series,
namely, the portion of large impulse response coefficient power is
identified, cancellation amount of about 70% of that when all the taps are
updated is obtained and the start-up time becomes faster than that when
all the taps are updated. Therefore, in division processing, the blocks
are also weighted and the coefficient correction amount is updated
intensively in low-order portions of the variable coefficient series. FIG.
6 shows an example where the variable coefficient series is divided into
four blocks to make three pairs, each of which consists of two blocks, and
N/2 tap update operation is performed in one step for updating the full
tap in four steps. Block 1 is updated in all steps 1-4, block 2 is updated
in steps 1 and 3, block 3 is updated in step 2, and block 4 is updated in
step 4. In other words, block 1 is updated every time, block 2 is updated
twice every four times (once every twice), and blocks 3 and 4 are updated
each once every four times.
FIG. 7 is a graph showing the acoustic echo cancellation characteristic
when an autoregressive signal is used. As a result of performing weighted
division processing (proposed method), substantially equal performance to
that of the system of updating all the taps of the variable coefficient
series (normal method) can be provided.
As discussed in detail, according to the first embodiment of the invention,
the following excellent effects can be expected.
(1) Since degradation of the convergence speed of the acoustic echo
cancellation characteristic due to division processing of updating the
coefficient correction amount can be corrected, the acoustic echo can be
cancelled at a high speed.
(2) Since the internal operation amount of the adaptive algorithm can be
reduced drastically without degrading the acoustic echo cancellation
performance, the hardware is provided as a small-scaled configuration.
(3) Fluctuation of the echo path characteristic is caused by a spacial move
of human bodies and objects approaching a microphone or a loudspeaker.
That is, in the invention for intensively updating low-order taps of
impulse response, the follow-up characteristic to echo path fluctuation is
very excellent and communication lines can be placed in the stationary
state rapidly.
(4) The operation amount related to updating a variable coefficient series
can be reduced to a half or less while the high quality of communication
lines can be provided.
(5) Since portions with large coefficients are identified preferentially,
the semi-stationary state is held and remaining echo signal at a
comparatively large level does not exist on the communication line, so
that two-way communication detection can be made easily and speech
degradation such that the head of transmit voice is cut is eliminated,
ensuring the high sound quality.
FIG. 8 is a block diagram showing the configuration of an acoustic echo
canceller according to a second embodiment of the invention. As shown in
FIG. 8, the acoustic echo canceller in the embodiment includes a receive
signal input terminal 1, receive signal output terminal 2, variable
coefficient digital filter 3, transmit signal input terminal 4, subtractor
5, transmit signal output terminal 6, coefficient correction amount
calculation circuit 7a, receive signal input register 8, artificial
impulse response register 9, sum-of-products operation circuit 10, and
coefficient block selector 21.
That is, the acoustic echo canceller in the second embodiment includes the
receive signal input terminal 1, receive signal output terminal 2,
transmit signal input terminal 4, transmit signal output terminal 6,
variable coefficient digital filter 3 to which a receive signal input
through the receive signal input terminal 1 is input, artificial impulse
response register 9 which stores coefficient series of the variable
coefficient digital filter 3, sum-of-products operation circuit 10 for
performing a convolution integral operation on the contents of the
artificial impulse response register 9 and the input signal through the
receive signal input terminal 1, subtractor 5 for calculating the
difference between an artificial echo generated by the sum-of-products
operation circuit 10 and an acoustic echo input through the transmit
signal input terminal 4, coefficient correction amount calculation circuit
7a for applying a correction amount to the coefficient series in the
artificial impulse response register 9 so that the variable coefficient
digital filter 3 supplies an approximate value of the acoustic echo, and
coefficient block selector 21 for sending an instruction for selecting one
of blocks in sequence for performing coefficient update operation to the
coefficient correction amount calculation circuit 7a so that the
coefficient series in the artificial impulse response register 9 is
divided into N blocks for automatically updating the entire coefficient
series in a total of M steps.
When the coefficient series stored in the artificial impulse response
register 9 is divided into N blocks and the entire coefficient series is
updated in M steps, a large interpolation loop gain .alpha..sub.0
interpolated into the coefficient correction amount calculation circuit 7a
is applied to blocks for which high update frequency is set so that the
blocks are to be updated every time and an extremely small interpolation
correction loop gain .alpha..sub.M-1 is applied to blocks for which low
update frequency is set so that the blocks are to be updated once every M
times. An interpolation correction loop gain .alpha..sub.m smaller than
the interpolation correction loop gain .alpha..sub.0 set in the blocks
updated every time and larger than the interpolation correction loop gain
.alpha..sub.M-1 set in the blocks updated once every M times is applied to
the n-th block for which update frequency is set so that the n-th block is
to be updated twice to less than M times every M times.
The relationship in value among the interpolation correction loop gains can
be represented as follows:
0<.alpha..sub.M-1 <. . . <.alpha..sub.m <. . . <.alpha..sub.0 .ltoreq.1 (8)
If a sequential update algorithm is formed using the interpolation
correction loop gains, it can be shown as expression (9) from expression
(4):
hs1.sub.k+1 =hs1.sub.k +.alpha..sub.0 (x.sub.k e.sub.k)/x.sub.k 'x.sub.k
hsn.sub.k+1 =hsn.sub.k +.alpha..sub.m (x.sub.k e.sub.k)/x.sub.k 'x.sub.k
hsN.sub.k+1 =hsN.sub.k +.alpha..sub.M-1 (x.sub.k e.sub.k)/x.sub.k 'x.sub.k
(9)
where 1<n<N. If variable coefficient series matrix hsn (n=1, 2, . . . , N)
of N blocks in the sequential update algorithm shown in expression (9) is
arranged in the update frequency ascending order of the blocks, it can be
shown as expression (10):
hsN<. . . <hsn<. . . hs1 (10)
Of course, the blocks may become the same in update frequency depending on
setup scheduling of divided update. In such a case, whether or not the
interpolation correction loop gain of the same value is to be adopted may
be determined in response to operation characteristics.
In the embodiment, division processing of updating the coefficient
correction amount is performed using each interpolation correction loop
gain stored in the coefficient correction amount calculation circuit 7a
conforming to the setup update frequencies under the above-mentioned
conditions.
FIGS. 9 and 10 show the observation results of coefficient series in the
artificial impulse response register 9 in which an interpolation
correction loop gain is set conforming to the update frequency of division
processing according to this embodiment and in the register in which a
fixed interpolation correction loop gain is set. The result of using the
adaptive interpolation correction loop gain according this embodiment in
FIG. 9 indicates that the impulse response presents an attenuation
characteristic, while the result of using the fixed interpolation loop
gain in FIG. 10 indicates that the coefficient portion of the long delay
time has a large value and the impulse response does not present an
attenuation characteristic as a whole. The impulse responses are observed
in the same step on simulation by inputting the same reference signal.
FIGS. 11 and 12 show the results of finding power in FIGS. 9 and 10
respectively. The result of adopting the fixed interpolation correction
loop gain in FIG. 12 shows that power is distributed throughout the
coefficient series. The power distribution of impulse response in an
actual sound field becomes one where the coefficient power in a long delay
time portion as shown in FIG. 11 is extremely small as compared with the
coefficient power in a short delay time portion. This fact also shows that
the invention is effective in division processing in which intermittent
update is executed.
FIGS. 13 and 14 show the results of observing power displacement of error
signal when actual voice of a woman is input as a reference signal. The
result of adopting the adaptive interpolation correction loop gain shown
in FIG. 13 indicates that the error signal, namely, acoustic echo is
attenuated conscientiously, but the result of adopting the fixed
interpolation correction loop gain in FIG. 14 shows that error signal
power is large and that erroneous cancellation occurs. The result shown in
FIG. 14 is a very rasping reverberation sound aurally.
Thus, according to the second embodiment of the invention, the following
effects are produced.
(1) Since degradation of the convergence speed of the acoustic echo
cancellation characteristic due to division processing of updating the
coefficient correction amount can be corrected, the acoustic echo can be
cancelled at a high speed.
(2) Since the internal operation amount of the adaptive algorithm can be
reduced drastically without degrading the acoustic echo cancellation
performance, the hardware can be provided as a small-scaled configuration
and costs can be reduced.
(3) Fluctuation of the echo path characteristic is caused by a spacial move
of human bodies and objects approaching a microphone or a loudspeaker.
That is, in the invention for raising the update frequency of low-order
taps of impulse response and adapting large interpolation correction loop
gains, the start-up speed of the acoustic echo cancellation characteristic
is fast, thus the follow-up characteristic to echo path fluctuation is
very excellent and communication lines can be placed in the stationary
state rapidly.
(4) The operation amount related to updating a variable coefficient series
can be reduced to a half or less while the high quality of communication
lines can be provided.
(5) Since amplitude fluctuation of error signal due to erroneous
cancellation scarcely occurs, the semi-stationary state is held and
remaining echo signal at a comparatively large level does not exist on the
communication line, so that two-way communication detection can be made
easily and speech degradation such that the head of transmit voice is cut
is eliminated, ensuring the high sound quality.
FIG. 15 is a block diagram showing the configuration of an acoustic echo
canceller according to a third embodiment of the invention. As shown in
FIG. 15, the acoustic echo canceller in the embodiment includes a receive
signal input terminal 1, receive signal output terminal 2, variable
coefficient digital filter 3, transmit signal input terminal 4, subtractor
5, transmit signal output terminal 6, coefficient correction amount
calculation circuit 7b, receive signal input register 8, artificial
impulse response register 9, sum-of-products operation circuit 10,
coefficient block selector 31, code generator 32, first selection switch
33, synchronous adder 34, second selection switch 35, linear convolution
integral operation circuit 36, observation impulse response register 37,
state selector 38, and block power evaluation circuit 39.
That is, the acoustic echo canceller in the third embodiment includes the
receive signal input terminal 1, receive signal output terminal 2,
transmit signal input terminal 4, transmit signal output terminal 6,
variable coefficient digital filter 3 to which a receive signal input
through the receive signal input terminal 1 is input, artificial impulse
response register 9 which stores coefficient series of the variable
coefficient digital filter 3, sum-of-products operation circuit 10 for
performing a convolution integral operation on the contents of the
artificial impulse response register 9 and the input signal through the
receive signal input terminal 1, subtractor 5 for calculating the
difference between an artificial echo generated by the sum-of-products
operation circuit 10 and an acoustic echo input through the transmit
signal input terminal 4, coefficient correction amount calculation circuit
7b for applying a correction amount to the coefficient series in the
artificial impulse response register 9 so that the variable coefficient
digital filter 3 supplies an approximate value of the acoustic echo, and
coefficient block selector 31 for sending an instruction for selecting one
of blocks in sequence for performing coefficient update operation to the
coefficient correction amount calculation circuit 7b so that the
coefficient series in the artificial impulse response register 9 is
divided into N blocks for automatically updating the entire coefficient
series in a total of M steps.
The echo canceller further includes the code generator 32 for generating a
series code having no correlation with the receive signal input through
the receive signal input terminal 31, first selection switch 33 for
outputting either the receive signal input to input terminal a or the
series code input to input terminal b through the receive signal output
terminal 2, synchronous adder 34 for calculating an impulse response of
the series code input through the transmit signal input terminal 4 via the
acoustic echo path, second selection switch for switching output terminals
a and b in synchronization with the first selection switch 33, linear
convolution integral operation circuit 36 for performing a convolution
integral operation on an output of the synchronous adder 34 and the series
code to calculate an impulse response of the acoustic echo path,
observation impulse response register 37 which stores observation
coefficient series of the impulse response output by the linear
convolution integral operation circuit 36, state selector 38 for
synchronously controlling the first and second selection switches 33 and
35 and issuing a discharge instruction of the impulse response coefficient
series stored in the observation impulse response register 37, and block
power evaluation circuit 39 for dividing the impulse response coefficient
series discharged from the observation impulse response register 37 in
resp | | |
