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DESCRIPTION JP2009100181

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DESCRIPTION JP2009100181
An operation of a voice switch is stabilized even in a situation in which a delay occurs on a
transmission path of a voice signal to prevent disconnection of a call. SOLUTION: The delay time
due to the audio signal processing of the first and second audio signal processing units 50, 51 is
offset by the delay time by the first and second delay means 60, 61. Therefore, even when the
first and second voice signal processing units 50 and 51 are performing voice signal processing,
an error is less likely to occur in the call state estimation process of the voice switch VS, and the
call voice due to the blocking of the voice switch VS It can prevent the interruption. [Selected
figure] Figure 1
Loudspeaker talker
[0001]
The present invention relates to a speakerphone (interphone, telephone, PHS, etc.) used in a
house, an office, a factory, etc.
[0002]
Conventionally, it is not necessary to have a handset at the time of a call, and an audio signal
transmitted from the other party's call terminal is transmitted to the caller from the other party
by a speaker to the caller away from the call terminal. There has been provided a speech
communication apparatus that enables half-duplex communication by collecting sound and
transmitting it to the other party's communication terminal.
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1
In such a loudspeaker communication apparatus, a two-wire to four-wire conversion hybrid
circuit is required when the component is a combination of a speaker and a microphone, and an
audio signal transmission path is formed in a two-wire form. A closed loop is formed on the call
path by the wraparound from the transmit signal path to the receive signal path caused by the
impedance mismatch at the end of the loop and by the acoustic coupling between the speaker
and the microphone at the other party's call terminal. Howling occurs when the ratio is more
than doubled, and if the howling occurs, the call can not be continued, so a means for
suppressing this is needed.
[0003]
Therefore, in the conventional loudspeaker communication apparatus, it is determined whether
the call state is the reception state or the transmission state by monitoring the transmission
signal and the reception signal, and the transmission signal path is determined according to the
determined communication state. Alternatively, voice switches have been widely used, which
reduce the loop gain of the closed loop and prevent howling by inserting attenuation means in at
least one of the reception signal paths. The basic operation of the voice switch is to estimate the
powers of the transmission signal and the reception signal, compare their magnitude
relationship, and insert a predetermined loss amount to the side of smaller instantaneous power.
In addition, in the case where use in an environment where the ambient noise level is high is
expected, means for determining whether the transmission signal and the reception signal are
voice or non-voice is required. Furthermore, when the acoustic coupling gain and the line
feedback gain when viewed from the voice switch are large, and the transmission signal and the
reception signal include many acoustic coupling components and circuit wraparound
components as well as the near-end speech signal and far-end speech signal. Requires an
algorithm to reduce the influence of acoustic coupling components and circuit wraparound
components because it is impossible to estimate the call state with high accuracy simply by
comparing the power of the transmission and reception signals. It becomes.
[0004]
What is described in Patent Document 1 as a voice communication apparatus equipped with a
voice switch in consideration of the use under the high ambient noise level as described above
and the application to a system having a large acoustic coupling gain and a loop return gain is
there.
[0005]
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2
FIG. 5 is a block diagram showing a voice communication system including the voice
communication device (base device M) of Patent Document 1 and a call terminal (child device S)
connected to the base device M by a two-line transmission path. FIG. 6 is a block diagram
showing the detailed configuration of the voice switch VS mounted on the parent device M. As
shown in FIG.
The parent device M includes a microphone 1, a speaker 2, a two-wire to four-wire conversion
hybrid circuit 3, a microphone amplifier G2 for amplifying a transmission signal from the
microphone 1, and a line output provided on the output side of the transmission side attenuator
4. An amplifier G1, a line input amplifier G3 for amplifying a received signal from a line, a
speaker amplifier G4, and an audio switch VS. Further, the child device S comprises a microphone
1 ', a speaker 2', a two-wire four-wire conversion hybrid circuit 3 ', a microphone amplifier G2'
and a speaker amplifier G4 '.
[0006]
The voice switch VS includes a transmitting side attenuator 4 inserted on a transmission signal
line for transmitting to the line an audio signal (transmission signal) collected by the microphone
1, and an audio signal (reception signal) received from the line Receiver attenuator 5 inserted on
the receiver signal line for transmitting the speaker 2 to the speaker 2, and an insertion loss
control unit for controlling the gains of the transmitter attenuator 4 and the receiver attenuator 5
according to the call state And 6. In addition, the insertion loss amount control unit 6 estimates
the instantaneous power of the input signal (signal at point B) to the transmission side attenuator
4, and the input to the reception side attenuator 5. The second instantaneous power estimating
unit 8 for estimating the instantaneous power of the signal (the signal at point C), the input point
B to the transmitting side attenuator 4, the transmitting side attenuator 4 and the looping on the
line side to receive the speech Line feedback gain multiplication means 9 having as a coefficient
a value determined according to the gain of the system fed back to the input point C to the side
attenuator 5 and the reception side attenuator 5 from the input point C to the reception side
attenuator 5 And an acoustic coupling gain multiplication means 10 having as a coefficient a
value determined in accordance with the gain of the path to the input point B to the transmitting
side attenuator 4 through the wraparound on the acoustic side, and the second instantaneous
power estimation The output signal P D 'obtained by inputting the output signal P C of the unit 8
to the acoustic coupling gain multiplication means 10 and the first blink The first comparator 11
for comparing the magnitude relation with the output signal P B of the power estimation unit 7
and the output signal P B of the first instantaneous power estimation unit 7 are obtained by
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inputting them to the line feedback gain multiplication means 9 The second comparator 12 that
compares the magnitude relationship between the output signal P A ′ and the output signal P C
of the second instantaneous power estimation unit 8, and the output signals of the first
comparator 11 and the second comparator 12 There is provided an insertion loss amount
distribution processing unit 13 which determines the call state based on C1 and C2 and controls
the gains of the transmission side attenuator 4 and the reception side attenuator 5.
[0007]
Here, the first and second instantaneous power estimation units 7 and 8 are realized by an
envelope detector, an integration circuit, and the like having characteristics of a sharp rise and a
slow fall, respectively. The instantaneous powers P B and P C of the input signal to and the input
signal to the receiving side attenuator 5 are estimated.
[0008]
The line feedback gain multiplication means 9 also includes a variable coefficient multiplier 9a
having as a coefficient a value Gt equal to the gain of the transmission side attenuator 4 and the
output point of the transmission side attenuator 4 measured in advance on the line side. It has a
fixed coefficient multiplier 9b having a value tt obtained by multiplying the gain of the path
leading to the input point C of the receiving side attenuator 5 through the wraparound by a
predetermined (about 2 to 3 times) margin value as a coefficient.
Furthermore, the acoustic coupling gain multiplication means 10 includes a variable coefficient
multiplier 10a having as a coefficient a value Gr equal to the gain of the receiving side attenuator
5, and a speaker amplifier G4-speaker from the output point of the receiving side attenuator 5
measured in advance. 2-A value obtained by multiplying the gain of the path from the sound
transmission system to the microphone 1 and the microphone 1 to the input point B of the
transmitting side attenuator 5 through the microphone amplifier G2 by a predetermined (about 2
to 3 times) margin value and a fixed coefficient multiplier 10b having η r as a coefficient. Here,
when setting the coefficients ηt and ηr of the fixed coefficient multipliers 9b and 10b, it is
necessary to use the margin value because of variations in acoustic coupling gain due to changes
in reflection conditions in front of the speaker 2 and the microphone 1, and This is to absorb
fluctuations in line side loop gain due to changes in impedance when the other party's
communication terminal (child device S) is viewed from the four-wire conversion hybrid circuit 3.
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[0009]
Next, the operation of the voice switch VS will be described.
[0010]
The first comparator 11 is obtained by inputting the output signal P B from the first
instantaneous power estimation unit 7 and the output signal P C from the second instantaneous
power estimation unit 8 to the acoustic coupling gain multiplication means 10 The output signal
C1 becomes "1" when P B P P D 'and the output signal C 1 becomes "0" when P B <P D'.
Further, in the second comparator 12, an output signal P A ′ obtained by inputting the output
signal P B of the first instantaneous power estimation unit 7 to the line feedback gain
multiplication means 9 and the second instantaneous power estimation unit 8 The output signal
C2 is "1" when P A '≧ P C, and the output signal C2 is "0" when P A' <P C.
[0011]
On the other hand, the insertion loss amount distribution processing unit 13 determines the call
state based on the binary signals C1 and C2 output from the first and second comparators 11
and 12, and the transmission side according to the determination result. The gains of the
attenuator 4 and the receiving side attenuator 5 are determined. Here, the determination rule of
the call state is a sending state when C1 = C2 = 1, a receiving state when C1 = C2 = 0, and an idle
state when C1 ≠ C2. Then, when the determination result is in the transmission state, the
insertion loss amount distribution processing unit 13 sets the gain of the transmission side
attenuator 4 to the maximum value and sets the gain of the reception side attenuator 5 to the
minimum value. When the determination result is in the receiving state, the gain of the
transmitting side attenuator 4 is set to the minimum value, the gain of the receiving side
attenuator 5 is set to the maximum value, and the determination result is in the idle state. The
gains of the transmit side attenuator 4 and the receive side attenuator 5 are set to equal values
(the square root value of the total gain).
[0012]
According to the above voice switch VS, by appropriately setting the coefficients ηt and ηr of
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the fixed coefficient multipliers 9b and 10b, transmission blocking can be reliably performed
even in a speech communication system having large acoustic coupling gain and line feedback
gain. It is possible to prevent reception blocking. Patent No. 3709739
[0013]
By the way, in the conventional loudspeaker communication apparatus, by performing audio
signal processing on the audio signal collected by the microphone, an effect special to the audio,
for example, an effect of making the voice color and speed (speaking speed) of the speaker
different from actual voice May be performed. In such a case, a large delay (for example, several
hundred milliseconds to several seconds) occurs in the timing at which the loop component due
to line feedback or acoustic coupling is input to the voice switch VS by the time required for the
voice signal processing for the voice signal. Therefore, an error is likely to occur in the process of
estimating the call state in the voice switch, and as a result, the call voice may be interrupted.
[0014]
The present invention has been made in view of the above-mentioned circumstances, and an
object thereof is a voice communication apparatus capable of stabilizing the operation of the
voice switch and preventing the interruption of the call even in the situation where a delay
occurs on the transmission path in the voice signal. To provide.
[0015]
In order to achieve the above object, the invention according to claim 1 provides a microphone
and a speaker, a receiver-side signal path for transmitting a reception signal sent from a call
terminal on the other side to the speaker, and a transmission collected by the microphone. The
voice switch includes a voice switch that switches the call state to reception and transmission by
inserting a loss in the transmission side signal path that transmits the signal and sends it to the
other party's call terminal. The voice switch is located on the transmission side signal path.
Insertion loss control for inserting the transmitting side attenuation means, receiving side
attenuation means inserted on the receiving side signal path, and the gain of the transmission
side attenuation means and the receiving side attenuation means according to the call state The
insertion loss amount control unit estimates the instantaneous power of the input signal to the
receiving side attenuation means, and the first instantaneous power estimation unit that
estimates the instantaneous power of the input signal to the transmission side attenuation means
The second moment to do A value determined according to the gain of the system that returns
from the input point to the transmitting side attenuation means to the input point to the
receiving side attenuation means via the transmission side attenuation means and the looping on
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the line side Determined according to the gain of the line feedback gain multiplier with
coefficients, the input point to the receiver attenuation means, the receiver attenuation means,
and the wraparound on the sound side to the input point to the transmission attenuation means
And an output signal obtained by inputting an output signal of the second instantaneous power
estimating unit to the acoustic coupling gain multiplying unit, and an output signal of the first
instantaneous power estimating unit. Between an output signal obtained by inputting the output
signal of the first instantaneous power estimating unit to the line coupling gain multiplying unit
and an output signal of the second instantaneous power estimating unit; A second comparator
that compares the magnitude relationship with the first ratio And an insertion loss amount
distribution processing unit for determining the speech state based on the output signals of the
comparator and the second comparator and controlling the gains of the transmitting side
attenuation means and the receiving side attenuation means, and line feedback gain multiplying
means A variable coefficient multiplier having a coefficient substantially equal to the gain of the
transmitting attenuation means, and a gain of a path from the output point of the transmitting
attenuation means to the input point of the receiving attenuation means through the loop on the
line side The acoustic coupling gain multiplication means comprises a variable coefficient
multiplier having a coefficient substantially equal to the gain of the receiver attenuation means,
and the receiver attenuation A fixed coefficient multiplier having as a coefficient a value obtained
by multiplying a predetermined margin value by the gain of the path from the output point of the
means to the input point of the transmitting side attenuation means via the acoustic coupling
system, Slows the input signal by the time corresponding to the group delay in the feedback gain
A first delay means for delaying is provided before or after the line feedback gain multiplying
means, and a second delay means for delaying the input signal by a time corresponding to a
group delay in the acoustic coupling gain is provided before the acoustic coupling gain
multiplying means. Alternatively, it is characterized in that it is provided at a later stage.
[0016]
According to the invention of claim 1, the first delay means for delaying the input signal by the
time corresponding to the group delay in the line feedback gain is provided at the front or rear of
the line feedback gain multiplying means, and the group delay in the acoustic coupling gain Since
the second delay means for delaying the input signal by the time corresponding to h is provided
before or after the acoustic coupling gain multiplication means, for example, the delay (group
delay) on the transmission path to the transmission signal input from the microphone Even in the
situation that occurs, the first time delay means cancels the relative time difference between the
reception signal and the transmission signal by delaying the reception signal received from the
other party's call terminal by the time corresponding to the delay (group delay). As a result, the
operation of the voice switch can be stabilized even in a situation where the voice signal is
delayed on the transmission path, and call interruption can be prevented.
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[0017]
According to the second aspect of the present invention, in the first aspect of the present
invention, audio signal processing with delay is performed on at least one of a transmission
signal and a reception signal input to the audio switch, and the presence or absence of the audio
signal processing is selected. The audio signal processing means is switchable in one and at least
one of the first and second delay means is characterized in that the input signal is delayed only
when the audio signal processing means performs audio signal processing. I assume.
[0018]
According to the second aspect of the present invention, malfunction of the voice switch due to
unnecessary delay can be prevented by delaying the delay means only when delay occurs in the
voice signal.
[0019]
According to the invention of claim 3, in the invention of claim 1 or 2, an echo canceller having
an adaptive filter is provided at least one of the front stage and the rear stage of the voice switch,
and the first and second delay means are echo cancelers. The delay time is adjusted in
accordance with the filter coefficient of the adaptive filter possessed by
[0020]
According to the invention of claim 3, since the first and second delay means adjust the delay
time according to the filter coefficient of the adaptive filter of the echo canceller, the delay time
is accurately estimated to operate the voice switch. Can be reliably stabilized, and the time and
effort required to obtain a delay time corresponding to the group delay in advance becomes
unnecessary.
[0021]
According to the present invention, for example, even in the situation where a delay (group
delay) occurs on the transmission path in the transmission signal inputted from the microphone,
the first delay means only for the time corresponding to the delay (group delay) By delaying the
reception signal received from the call terminal, the relative time difference between the
reception signal and the transmission signal can be offset, and as a result, the operation of the
voice switch is stabilized even in the situation where the voice signal is delayed on the
transmission path. It is possible to prevent the interruption of the call.
[0022]
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Embodiment 1 Hereinafter, an embodiment of the present invention will be described in detail
with reference to the drawings.
However, since the basic configuration of the loudspeaker communication device according to
the present embodiment is the same as that of the conventional example (the base unit M)
described in Patent Document 1, the same components are denoted by the same reference
numerals, I omit explanation.
[0023]
FIG. 1 is a block diagram showing the main part of the loudspeaker communication device
(master device M) of this embodiment.
In the present embodiment, as shown in FIG. 1, a first audio signal processing unit 50 that
performs audio signal processing with a delay on a transmission signal input to the audio switch
VS, and a reception signal input to the audio switch VS. On the other hand, the second voice
signal processing unit 51 for performing voice signal processing with delay, the first delay means
60 provided at the rear stage of the line feedback gain multiplication means 9, and the rear stage
of the acoustic coupling gain multiplication means 10 The second embodiment is characterized
in that the second delay means 61 is provided.
[0024]
The first and second audio signal processing units 50 and 51 perform audio frequency
conversion processing for converting the frequency (voice color) of audio, speech speed
conversion processing for converting the speed of audio, and the like, and are not shown. The
presence or absence of the audio signal processing can be alternatively switched according to the
operation input received by the operation input receiving means.
For example, in the case of performing the audio frequency conversion processing in the first
audio signal processing unit 50, the relatively high voice of a woman or a child is converted to a
relatively low voice such as a male, so that a push sale etc. can be performed. It can be done with
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confidence.
Alternatively, if the speech speed conversion process is performed by the second audio signal
processing unit 51, it is possible for the hearing impaired person, the elderly person, etc. to make
the sound from the speaker 2 at a slow speed suitable for audio listening.
However, since any voice signal processing of voice frequency conversion processing and speech
speed conversion processing can be realized by a conventionally known technology (for example,
TDHS <Time Domain Harmonic Scaling> algorithm etc. in speech speed conversion processing)
The description is omitted.
[0025]
Here, the first and second audio signal processing units 50 and 51 perform audio signal
processing with delay such as audio frequency conversion processing and speech speed
conversion processing on the audio signal temporarily stored in the buffer, A large delay (for
example, several hundreds of milliseconds to several seconds) occurs in the timing at which the
loop component due to line feedback or acoustic coupling is input to the voice switch VS with or
without voice signal processing.
Then, when such a large delay occurs, there is a possibility that the process of estimating the call
state in the voice switch VS becomes unstable to cause a malfunction.
[0026]
Therefore, in the present embodiment, the first delay means 60 is provided at the rear stage of
the line feedback gain multiplication means 9 of the voice switch VS, and the second delay means
61 is provided at the rear stage of the acoustic coupling gain multiplication means 10. When the
audio signal processing unit 50 is performing audio signal processing, the first delay means 60
delays the reception signal input to the audio switch VS by a predetermined delay time (first
delay time), and the second audio is generated. When the signal processing unit 51 is performing
audio signal processing, the second delay means 61 delays the transmission signal input to the
audio switch VS by a predetermined delay time (second delay time).
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However, the first delay time is set to a time corresponding to the group delay in the acoustic
coupling gain (delay time caused by the audio signal processing of the first audio signal
processing unit 50), and the second delay time is for the line feedback gain. The time
corresponding to the group delay (the delay time generated by the audio signal processing of the
second audio signal processing unit 51) is set.
[0027]
Thus, when the first voice signal processing unit 50 is performing voice signal processing, the
input to the voice switch VS of the sneak component by acoustic coupling is delayed, but the line
feedback gain is only for the time corresponding to the delay time. By delaying the timing at
which the output signal P A ′ of the multiplication means 9 is input to the second comparator
12 by the first delay means 60, the delay due to the audio signal processing of the first audio
signal processing unit 50 is first The delaying means 60 cancels out the delay due to the delay
means 60, and there is no time lag between the input timings of the two signals P A ′ and P C
compared with the second comparator 12.
Similarly, when the second voice signal processing unit 51 is performing voice signal processing,
the input to the voice switch VS of the wraparound component by line feedback is delayed, but
the acoustic coupling gain multiplication is performed for a time corresponding to the delay time.
By delaying the timing at which the output signal P D ′ of the means 10 is input to the first
comparator 11 by the second delay means 61, the delay due to the audio signal processing of the
second audio signal processing unit 51 is second There is no time lag between the input timings
of the two signals P D ′ and P B which are offset by the delay by the delay means 61 and
compared in the first comparator 11. Therefore, even when the first and second voice signal
processing units 50 and 51 are performing voice signal processing, an error is less likely to occur
in the call state estimation process of the voice switch VS, and the call voice due to the blocking
of the voice switch VS It can prevent the interruption. However, when the first and second audio
signal processing units 50 and 51 do not perform audio signal processing, the first and second
delay units 60 and 61 are also not delayed the audio signal so that the audio due to an
unnecessary delay is generated. It is desirable to prevent the switch VS from malfunctioning. In
the present embodiment, the first and second delay means 60 and 61 are provided at the
subsequent stages of the line feedback gain multiplication means 9 and the acoustic coupling
gain multiplication means 10, respectively. I don't care.
[0028]
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Embodiment 2 In this embodiment, as shown in FIG. 2, an acoustic side echo canceller (not
shown) for suppressing an acoustic echo generated by acoustic coupling between the
microphone 1 and the speaker 2 and an acoustic coupling in the other party's call terminal Or a
line-side echo canceller 30 that suppresses line echoes caused by signal wraparound in the
transmission system, and a delay time estimation unit 62 that estimates a delay time according to
the filter coefficient of the adaptive filter 31 of the line-side echo canceler 30 A voice switch VS is
disposed between the acoustic side and the line side echo canceller 30. However, since the basic
configuration of the present embodiment is the same as that of the first embodiment, illustration
and description of the common components are omitted as appropriate.
[0029]
As shown in FIG. 3, the line-side echo canceller 30 reflects the impedance mismatch between the
two-wire / four-wire conversion hybrid circuit 3 and the transmission line, and reflects between
the speaker 2 'and the microphone 1' in the door phone handset S. Adaptive filter 31 that
adaptively identifies the impulse response of the feedback path (line echo path) H LIN formed by
acoustic coupling, and the echo component (line echo) estimated from the reference signal
(transmission signal) from the reception signal And a subtractor 32 for subtraction. The adaptive
filter 31 updates filter coefficients based on a predetermined algorithm (for example, a learning
identification method). The acoustic echo canceller (not shown) also has the same configuration
as the line echo canceller 30.
[0030]
Here, the relationship between the tap number in the adaptive filter 31 and the filter coefficient
(count value) is shown in FIG. 4, but D taps in FIG. 4 correspond to the group delay τ T in the
echo path. Assuming that the sampling period is Ts, the following relationship is established.
[0031]
Therefore, the delay time estimation unit 62 calculates the group delay τ T based on the above
equation when the line side echo canceller 30 is in the convergence state, and the group delay τ
T is the first delay. The delay time in the means 60 is set.
[0032]
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As described above, in the present embodiment, the delay time of the first delay means 60 is
adjusted according to the filter coefficient of the adaptive filter 31 of the line-side echo canceler
30, so that the delay time can be estimated accurately. As a result, the operation of the circuit can
be reliably stabilized, and the trouble of obtaining the delay time corresponding to the group
delay in advance becomes unnecessary.
However, although only the echo canceller on the line side (the line side echo canceller 30) has
been described in this embodiment, the echo canceler on the acoustic side (acoustic side echo
canceller) is similarly configured, and the adaptive filter of the acoustic side echo canceller It is
possible to adjust the delay time of the second delay means 61 according to the filter coefficient.
[0033]
It is a block diagram showing Embodiment 1 of the present invention.
It is a block diagram of the principal part which shows Embodiment 2 of this invention. It is a
block diagram which shows the channel side echo canceller 30 in the same as the above. It is
operation | movement explanatory drawing of the delay time estimation part in the same as the
above. It is a block diagram which shows a prior art example. It is a block diagram which shows
the audio | voice switch in same as the above.
Explanation of sign
[0034]
M master unit (voice speaker communication apparatus) VS voice switch 50 first voice signal
processing unit 51 second voice signal processing unit 60 first delay means 61 second delay
means
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